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Subject: Re: Fwd: Re: Define a unicast RTP session.
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: "Strahm, Bill" <bill.strahm@intel.com>,
	Mark Baugher <mbaugher@passedge.com>,
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From: Vincent_Virgilio@nmss.com
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The literal understanding of the section 3 definition includes the
observation that the definition is singular, thus saying "For each
participant, _the_ session . . .".  It does not say "For each participant,
_a_ session . . .".  Subtle, but unambiguous.  So the session (not
necessarily the destination address) is common to all, regardless of the
caste.  Further proof: section 7.1, paragraph 1.  There, clouds are equated
to sessions, and a unicast cloud is defined by a pair of unicast addresses.
It requires no interpretation to call those destination addresses.
Therefore a unicast session is defined by two destination addresses (where
the destination port is still common).

This causes no problems for the MIB.  I see this might not be true for SDP.
Perhaps you could elaborate.  Here's my contribution:

I agree with you that, under RFC 1889, an SDP invite and its OK will create
two sessions if the destination ports differ.  But if they don't (can this
be done predictably?), there should be (is) one session.  Later drafts of
RFC 1889 do specify that the destination ports in a session may differ, and
thus might eventually cause trouble for SIP/SDP if its notion of a session
changes, if there's not trouble now.  Then it might become difficult for
SIP/SDP to correlate the two directions of flow (is this necessary?).
Anyway, this came after the definition of SDP (RFC 2327), so was not there
to muddy the waters when the SDP authors were considering the nature of a
session.  Bidirectional unicast sessions could have been explicitly
permitted from the beginning.

Vince Virgilio





Jonathan Rosenberg <jdrosen@dynamicsoft.com> on 09/01/2000 12:40:49 AM

To:   "Strahm, Bill" <bill.strahm@intel.com>
cc:   "'Vincent_Virgilio@nmss.com'" <Vincent_Virgilio@nmss.com>, Mark
      Baugher <mbaugher@passedge.com>, rem-conf@es.net
Subject:  Re: Fwd: Re: Define a unicast RTP session.




"Strahm, Bill" wrote:
>
> I think I'll step in as another one of the RTP MIB authors who had to
> wrangle with this issue...
>
> We took the definition from 1889 quite literally From section 3
>
>    RTP session: The association among a set of participants
>         communicating with RTP. For each participant, the session is
>         defined by a particular pair of destination transport addresses
>         (one network address plus a port pair for RTP and RTCP). The
>         destination transport address pair may be common for all
>         participants, as in the case of IP multicast, or may be
>         different for each, as in the case of individual unicast network
>         addresses plus a common port pair.  In a multimedia session,
>         each medium is carried in a separate RTP session with its own
>         RTCP packets. The multiple RTP sessions are distinguished by
>         different port number pairs and/or different multicast
>         addresses.
>
> Therefore in the case of unicast there ARE two sessions, because there
are
> two different destination transport addresses.  Did we get it right, I
hope
> so (we are on track to have an RFC issued).

This was my understanding, and is exactly the reason I claimed it was
two in the first place. Also, if one assumes that a single m line in SDP
means one session, then with SIPs usage of SDP we also get two sessions
- thats because one SDP is sent in the INVITE (indicating the receive
port pairs of the caller), and another, different SDP in the 200 OK
(indicating the receive port pairs of the callee). This is what I meant
by mentioning that the annexes in SIP should be checked for more info.

-Jonathan R.
--
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com







From rem-conf Fri Sep 01 20:20:06 2000 
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Vincent_Virgilio@nmss.com wrote:
> 
> I agree with you that, under RFC 1889, an SDP invite and its OK will create
> two sessions if the destination ports differ.  But if they don't (can this
> be done predictably?), there should be (is) one session.  Later drafts of
> RFC 1889 do specify that the destination ports in a session may differ, and
> thus might eventually cause trouble for SIP/SDP if its notion of a session
> changes, if there's not trouble now.

There is no trouble now. Whether you call it one or two sessions in any
case seems completely academic. The operation in either case, as I can
tell, is the same.

>  Then it might become difficult for
> SIP/SDP to correlate the two directions of flow (is this necessary?).

Sorry? How do you arrive at this conclusion? As the caller, I choose the
ports I will receive media on, and I am told the ports to send media
two. They are correlated since they are part of the same SIP call. Its
been working in SIP devices for a while.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From rem-conf Sat Sep 02 23:20:50 2000 
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From rem-conf Sun Sep 03 14:55:34 2000 
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With the addition of the congestion control to the RTP draft I was
wondering if anyone in the group has references or pointers on work
examining subjective and/or objective performance of adaptive coding
schemes?  i.e. the effects, real or simulated, from an end-user
perspective.  Any pointers would be welcome.

cheers
- Orion.





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Orion Hodson wrote:
> With the addition of the congestion control to the RTP draft I was
> wondering if anyone in the group has references or pointers on work
> examining subjective and/or objective performance of adaptive coding
> schemes?  i.e. the effects, real or simulated, from an end-user
> perspective.  Any pointers would be welcome.

It's been bought to my attention that this question is a too vague,
that was the intention, but I clearly went too far.  I'm interested in
work assessing the subjective, or objective, quality at the receiver
when the encoder has multiple operational modes (and/or bit-rates)
with a modulation mechanism that controls the encoder.

As an example, consider MPEG-4 audio.  It includes several distinct
families of audio encoder operating at different data rates and over
different signal bandwidths.  An application might adapt the mean data
rate in response to a congestion indicator.  You might (reasonably)
expect transitions between encoder family to be harder to conceal than
small differences in mean data rate by the encoder.  This type of
information might influence an adaptation algorithm.

Is there work that other members of the group are aware of that
quantifies those types of transitions by subjective or objective
assessment?

- Orion.








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===========================================================================



*** PRELIMINARY PROGRAM AND CALL FOR PARTICIPATION ***
                          LCN 2000
   The 25th Annual IEEE Conference on Local Computer Networks

                    November 8 - 10, 2000
             Embassy Suites USF, Tampa, Florida
                   http://www.ieeelcn.org

Sponsored by the IEEE Computer Society with support from GTE Data
Services,
Paradyne Corporation, and the University of South Florida College of
Engineering.  The IEEE LCN conference is the premier conference on
leading
edge and practical computer networking.  LCN 2000 will have three
tutorials,
two keynote addresses, three panels, and 105 papers (72 full and 33
short)
in six sessions and four tracks.  Full papers are 10 pages and 20
minutes
presentation, short papers are 2 pages and 10 minutes.  The preliminary
program is below.

CONFERENCE PRE-REGISTRATION: The conference fees include a copy of the
proceedings, four breaks, two luncheons, and the Thursday evening
dinner.
The pre-registration deadline is October 31, 2000.

          IEEE member Non-member Student
Conference    $495      $610      $300
Tutorial #1   $645      $745      $645
Tutorial #2   $350      $500      $350
Tutorial #3   $350      $500      $350

Online registration is at http://www.ieeelcn.org/lcn25registration.html.

Contact Ken Christensen at christen@csee.usf.edu for additional
information.

HOTEL PRE-REGISTRATION: Hotel registration is through the USF Embassy
Suites
(rooms are $114 US per night).  See http://www.ieeelcn.org or call the
hotel
at 1 813-977-7066.  The hotel pre-registration deadline is October 15,
2000.

****************************************************************
Wednesday - November 8, 2000
****************************************************************
9:00 - 5:00: Tutorial #1: Security Topics and Techniques (Gary Kessler,
Champlain College)

9:00- 12:30: Tutorial #2: Virtual Private Networks (Dr. Tim Strayer, BBN

Technologies and Dr. Ruixi Yuan, GTE Internetworking)

12:30 - 1:30: Lunch

1:30: - 5:00: Tutorial #3: Managing Internet Quality of Service (QoS)
(Dr. Sanjay Jha, University of New South Wales)

****************************************************************
Thursday - November 9, 2000
****************************************************************
7:30 - 8:30: Registration

8:30 - 9:00: Opening and welcome session

9:00 - 10:00: Keynote address #1 - IPsec: How and Why (Dr. Stephen Kent,
BBN)

10:00 - 10:30: Best paper presentation
Anonymization Services for IP Multicast (full) C. Grosch

10:30 - 11:00: Coffee break

11:00 - 12:00: Panel #1 - Security (chair: Gary Kessler)

12:00 - 1:20: Lunch (sponsored by USF College of Engineering)

1:20 - 2:20: Panel #2 - Home access methods

2:20 - 2:40: Short break

2:40 - 4:00: Session #1 (Tracks A, B, C, and D)

>>> Track A - Wireless LANs
Transmission Power Control for Multiple Access Wireless Packet Networks
(full) J. Monks
A Novel Approach to ARQ Error Control Mechanisms for Wireless LANs
Communications (full) N. T. Almeida, S. A. Abrantes
Wireless Packet Scheduling with Signal-to-Noise Ratio Monitoring (full)
H. Aida, Y. Tamura, Y. Tobe, H. Tokuda
Comparison of TCP Reno and Vegas in Wireless Mobile Ad Hoc Networks
(short)
S. Xu, T. Saadawi, M. Lee
Adaptive Approaches to Enhance Throughput of IEEE 802.11 Wireless LAN
with
Different Channel Quality (short) S. Ci, H. Sharif

>>> Track B - Differentiated Services and QoS 1
Layered Network QoS Signalling - Motivation, Implementation &
Measurements
(full) J. Schmitt, M. Karsten, R. Steinmetz
Class-Based QoS Control Scheme by Flow Management in the Internet Router

(full) K. Minami, H. Tode, K. Murakami
An Analysis of Resource Scheduling with Prioritization for QoS in LANs
(full)
D. H. Summerville, L. Edmunds
Adaptive Bandwidth Reservation Mechanism Using Mobility Probability in
Mobile
Multimedia Computing Environment (full) C. H. Choi, M. I. Kim, T. J.
Kim,
S. J. Kim

>>> Track C - Network security 1
Techniques for Intrusion-Resistant Ad Hoc Routing Algorithms (TIARA)
(full)
R. Ramanujan, A. Ahamad, J. Bonney, R. Hagelstrom, K. Thurber
Network Security(Security in Large Networks) (full) M. Singh, S. Singh
IGMPv3-Based Method for Avoiding DoS Attacks in Multicast-Enabled
Networks
(short) A. F. Gomez-Skarmeta, A. L. M. Martinez, P. M. R. Martinez
Distributed
Weakness in Virtual Private Networks (short) S. Patton, D. Doss, W.
Yurcik

>>> Track D - Reliability and fault-tolerance
Algorithms for Minimal Cutsets Enumeration of Networks by Graph Search
and
Branch Addition (full) H. Suh, C. Chang
Provisioning Multilayer Restoration for Convergence (full) J. Kroculick,

C. Hood
Fault-Tolerant Ethernet Middleware for IP-Based Process Control Networks
(full)
S. Song, J. Huang, P. Kappler, R. Freimark, T. Kozlik
Simulative Reliability Analysis of SCI Ring-Based Topologies (full)
M. Sarwar, A. George, D. Collins

4:00 - 4:30: Cookie break

4:30 - 5:50: Session #2 (Tracks A, B, C, and D)

>>> Track A - Wireless LANs / Mobile IP
Mobile Stations Localization in a WLAN (full)
F. Peyrard, C. Soutou, J. J. Mercier
Handoff Support for Mobilty with IP over Bluetooth (full) S. Baatz, M.
Frank,
R. Gpffarth, D. Kassatkine, P. Martini , M. Schetelig, A. Vilavaara
A Portable Mobile IP Implementation (full) H. Haverinen, A. Kuikka,
T.Maattanen
Security of Routing Cache Updates in Cellular IP (short) S. Baatz, W.
Hansmann,
J. Tlle Location of Mobile Stations in a Wireless LAN (short)
E. Llusca, T. Val, C. Normand, J. Mercier

>>> Track B - Differentiated services and QoS 2
A Core-Stateless Buffer Management Mechanism for Differentiated Services

Internet (full) Y. T. Hou, D. Wu, Z.-L. Zhang, T. Chujo
Co-operation and Comparison of DiffServ and IntServ: Performance
Measurements
(full) J. Harju, P. Kivimaki An Optimal Bandwidth Allocation Scheme and
Real-
Time Performance Analysis for LTPB Network (full)
Z. Qiang, L. Zhiqiang, L. Qiao, X. Huagang
Programmable Resource Control in Global Active IP Networks (short)
H. Guo, T. Becker
On Integrating Differentiated Services and UMTS Networks(short)
J. Diederich, T. Lohmar, M. Zitterbart, R. Keller

>>> Track C - Network security 2
Multi-Dimensional Prefix Matching Using Line Search (full) M. Waldvogel
On Key Distribution In Secure Multicasting (full) K.-P. Wu, F. Lai,
C.-K. Tseng
Group Management Strategies for Secure Multicasting on Active Virtual
Private
Networks (full) C. Labonte, S. Srinivas
Open Source Versus Commercial Firewalls: Functional Comparison (short)
S. Patton, D. Doss, W. Yurcik
A Virtual Private Network Deployment Framework (short)
S. Patton, B. Smith, D. Doss, W. Yurcik

>>> Track D - Resource control
Efficient Congestion Avoidance Mechanism (full) A. Dracinschi, S. Fdida
Quality of Service using Traffic Engineering over MPLS: An Analysis
(full)
P. Bhaniramka, W. Sun, R. Jain
DSRED: A New Queue Management Scheme for Next Generation Networks (full)

B. Zheng, M. Atiquzzaman
Achieving Moderate Fairness for UDP Flows by Path-Status Classifcation
(full)
Y. Tobe, Y. Tamura, A. Molano, S. Ghosh, H. Tokuda

5:50 - 7:00: Social hour

7:00: Dinner

****************************************************************
Friday - November 10, 2000
****************************************************************

7:30 - 8:30: Registration

8:30 - 9:30: Keynote address #2 - The Care and Feeding of Network
Interfaces
(Denton Gentry, Sun Microsystems)

9:30 - 9:40: Short break

9:40 - 10:40: Panel #3 - Multicast

10:40 - 11:10: Coffee break

11:10 - 12:30: Session #3 (Tracks A, B, C, and D)

>>> Track A - Wireless / cellular networks
Indoor Location Technique for 2G and 3G Cellular/PCS Networks (full)
D.J. Shyy, B. Rohani
Real Time Multicast in Wireless Networks (full) A. Benslimane
DRiVE-ing to the Internet: Dynamic Radio for IP Services in Vehicular
Environments (full)
L. Xu, R. Tonjes, T. Paila, W. Hansmann, M. Frank, M. Albrecht
iNetwork - a GSM Compliant IP-Based Communication Platform (short)
C. W. Cheng, W. C. Liu, C. G. Chung
Optimization Paging Cost by Dynamic Cell Grouping in Mobile Network
(short)
G. Y. Lee, Y. Lee

>>> Track B - Differentiated services and QoS 3
Performance Measurements and Analysis of TCP Flows in a Differentiated
Services
WAN (full) J. Harju, Y. Koucheryavy, J. Laine, S. Saaristo, K. Kilkki,
J. Ruutu,
H. Waris, J. Forsten, J. Oinonen
Proportional Packet Loss Differentiation and Buffer Management for
Differentiated Services in the Internet (full)
M. Markaki, M. P. Saltouros, I. S. Venieris
End-to-end QoS Management for Delay-Sensative Scalable Multimedia
Streams over
DiffServ (full) M. Albrecht, M. Koster, P. Martini, M. Frank
Integrated Service Mappings on Differentiated Service PHB :Guaranteed
Service to
Expedited Forwarding (short) B. Budiardjo, B. Nazief, D. Hartanto
Intra-domain Bandwidth Management in Differentiated Services Network
(short)
S. Jha, M. Hassan, P. Nanda

>>> Track C - Video communications
Index Control and Bandwidth Distribution for Layered Video Stream
Transmission
on the Internet (full) C. Zhiyong, L. Zhang
Joint Synchronization between Live and Stored Media in Multicast
Communications
(full) Y. Ishibashi, S. Tasaka, I. Miyamoto
A Comparative Survey of Sychronization Algorithms for Continuous Media
in
Network Envirionments (full) Y. Ishibashi, S. Tasaka
An Adaptive Dynamic Congestion Control of Compressed Video Traffic over
ATM
Networks (short) G. Sisodia, L. Guan, S. De, M. Dowlatshahi
The Performance of Ethernet Under a Combined Data/Real-Time Traffic
(short)
M. A. Aboelaze, A. Elnagar

>>> Track D - Multicast 1
A New Paradigm for Multipoint-to-Multipoint Communication Using XTP
(full)
G. Ramasivan, J. W. Atwood
Multicast Routing Based on Ant-Algorithm For Delay-Bounded and
Load-Balancing
(full) L. Guoying, L. Zemin
Systematic Testing of Protocol Robustness: Case Studies on Mobile IP and
MARS
(full) S. Begum, M. Sharma, S. Gupta, A. Helmy
Optimal Placement of Multicast Nodes in All-Optical Networks (short)
M. Ali, J. Deogun

12:30 - 1:50: Lunch

1:50 - 3:10: Session #4 (Tracks A, B, C, and D)

>>> Track A - Wireless networks
Interesting Optimization Problems in the Planning of Microwave
Transmission
Networks (full) S. Puthenpura
Satellite over Satellite (SOS) Network: A Novel Concept of Hierarchical
Architecture and Routing in Satellite Network (full)
J. W. Lee, T. W. Kim, D. W. Kim
The Effect of GA Parameters on PCS Network Planning (full)
J. Li, Y. L. Guan, B. H. Soong
Auto-Configuration of Wireless Cell-site (short) S. Djoko, H. Lee, K.
Basu
Experiment with the Validation of WAP Systems (short) O. Kone, A. R.
Cavalli

>>> Track B - TCP and other protocols 1
TCP Tunnels: Avoiding Congestion Collapse (full)
B. P. Lee, R. K. Balan, L. Jacob, W. K. G. Seah, A. L. Ananda
The Performance of TCP over ATM on Lossy ADSL Networks (full)
G. Lu, R. Simmonds, X. Zhonge, B. Unger, C. Williamson
Analytical Modeling and Performance Evaluation of the HIPERLAN CAC Layer

Protocol for Real-Time Traffic (full) C. Coutras, P.-J. Wan, O. Frieder
Simple Reliable Multicast for Parallel Processing in Extended LANs
(short)
J. Mulik, Y. Shi, P. Conrad
TCP Performance of DMT-based ADSL System (short) X. He

>>> Track C - Network management
A Methodology for Performance Management of Networks (full)
A. B. Drago, A. S. Garcia, M. E. Monteiro
Switching Theory Approach to Alarm Correlation in Network Management
(full)
E. Aboelela, C. Douligeris
Comparisons of Different Approaches for Capacity Management in ATM
Networks
(full) S. O. Larsson
Open Network Platform for Multiprotocol Communication (short)
D. Jozic, T.Osmanlic, D. Huljenic, V. Sinkovic
Investigation of SNMP Impact on a Target Device (short) N. Matsushita

>>> Track D - Multicast 2
Tree-Based Reliable Multicast in Combined Fixed/Mobile IP Networks
(full)
W. Yoon, D. Lee, C. Yu
An Architecture for Seamless Access to Multicast Content (full) P.
Liefooghe
Multi-Cores Uni-Directional Shared Trees Multicast Routing Protocol
(short)
M. Yulu, C. Shiduan
Delay Analysis for Multicast Switch under Burst and Poisson Traffic
(short)
C. Khemapatapan, P. Prapinmongkolkarn, S. Theerawoot
Optical QoS in Multicast Wavelength-Routed Networks Part I
-PowerConsiderations
(short) M. Ali, J. Deogun

3:10 - 3:40: Cookie break

3:40 - 4:40: Session #5 (Tracks A, B, C, and D)

>>> Track A - Virtual networks
Performance Impact of Data Compression on Virtual Private Network
Transactions
(full) McGregor
Implementation of a Bandwidth Broker for Dynamic End-to-End Resource
Reservation
in Outsourced Virtual Private Networks (full) I. Khalil, T. Braun
Inter Bridge VLAN Registration Protocol for IP Subnet VLAN (short)
C. Kok, M. SalimBeg, N. Gehlot
Performance Evaluation of Software Virtual Private Networks (VPN)
(short)
C. Pena, J. Evans

>>> Track B - TCP and other protocols 2
TCP Models in Simple Cases (full) T. Elteto, P. Benko
TCP Performance over Heterogeneous Networks (full) M. Arpaci, H.
Uzunalioglu, J.
A. Copeland
Effects of Competing Traffic on the Performance of TCP/IP over
Asymmetric Links
(short) K. Phanse, L. A. DaSilva, K. Kidambi
On Some Metrics of TCP (short) A. Trinh, T. Eteto, L. Gyorfi

>>> Track C - Optical networks
Performance of Multihop Networks using Optical Buffering and Deflection
Routing
(full) A. G. Fayoumi, A. Jayasumana, J. Sauer
Practical Traffic Grooming Scheme for Single-Hub SONET/WDM Rings (full)
X.-Y. Li, L. Liu, P.-J. Wan, O. Frieder
A Route Design Algorithm for Multiple-Encoding Optical CDM Switching
Network
(full) H. Tode, Z. Tianwei, S. Nishi, K. Murakami

>>> Track D - Routing
A Qos Routing Algorithm Based on Ant Algorithm (full) Z. Subing, L.
Zemin
Hierarchial Qos Routing in Delay-Bandwidth Sensitive Networks (full)
K.-S. Lui, K. Nahrstedt, S. Chen
IP Route Lookups as String Matching (full) A. Donnelly, T. Deegan

4:40 - 4:50: Short break

4:50 - 5:50: Session #6 (Tracks A, B, C, and D)

>>> Track A - ATM
Delay Estimation and Control of Available Bit Rate (ABR) Sources (full)
P. Zhou, C. Thompson
Link-based Performance Monitoring of ATM Networks (full)
B. Thurm, H. R. Wiltfang
Measurement of ATM Frame Latency (full) A. Durresi, R. Jain, G. Babic
ATM Network Connection Management Using Mobile Agents (short)
S. Lazar, S. Kodeswaran, R. Varadharaj, D. Sidhu

>>> Track B - Load balancing and caching
A Distributed Internet Caching System (full) T. Tay, Y. Feng, M.
Wijeysundera
Load-Balanced Routing and Scheduling for Real-Time Traffic in
Packet-Switched
Networks (full) S. Bak, A. Cheng, J. Cobb, E. Leiss
Performance Evaluation of New Methods of Automatic Redirection for Load
Balancing of Apache Servers Distributed in the Internet (full)
K. Suryanarayanan, K. Christensen

>>> Track C - Traffic characterization
Generalized Autoregressive Moving Average Modeling of the Bellcore Data
(full)
R. Ramachandran, V. R. Bhethanabotla
High-Speed Calculation Method of the Hurst Parameter Based on Real
Traffic
(full) T. Hagiwara, H. Doi, H. Tode, H. Ikeda
The Dependence of Internet User Traffic Characteristics on Access Speed
(full)
N. Vicari, S. Khler, J. Charzinski
Nonlinear Time Series Model for VBR Video Traffic (full)
J. L. Davis, K. Chandra, C. Thompson

>>> Track D - High-speed switching and routing
Scheduling Algorithms for A High-Speed Switch Supporting Real-Time
Periodic
Traffic Sources (full) J. Liu, L. Xia, R. Tsang, A. Pavan, D. H. C. Du
IP Forwarding Engine with VC Merging in MPLS System (full) S. Kang,
Y.-K. Lee
High-Speed IP Forwarding Mechanism for an MPOA-Based Switched Router
(short) J.
A. Jun, H. C. Kim, K.-H. Lee, D. K. Sung
A New Route Selection Approach Using Scaling Techniques: An Application
to
Hierarchical QoS-Based Routing (short)
M. P. Saltouros, M. E. Markaki, A. K. Taskaris, M. E. Theologou, and I.
S.
Venieris

*******************************************************************************

--
_____________________________________________________________________
Prof. Dr. Peter Martini

University of Bonn                     Tel.:   +49-228-73-4334
Institute of Computer Science IV       Fax:    +49-228-73-4571
Roemerstr. 164                         E-mail: martini@cs.uni-bonn.de
D-53117 Bonn, Germany                  http:   www.cs.uni-bonn.de/IV/





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From: Bezalel Gavish <gavishb@mail.cox.smu.edu>
Subject: 9th International Conference on Telecommunication Systems
  Conference - CFP
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Attached is a call for papers for the conference which will be held in 
Dallas on March 15-18, 2001. I will appreciate it very much if you will 
bring it to the attention of interested parties.

Sincerely yours,
Bezalel 
--=====================_247764807==_
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			C A L L  for  P A P E R S
	9th International Conference on Telecommunication Systems,
			  Modeling and Analysis
			    March 15-18, 2001
			    Dallas, Texas USA

Sponsors:	American Telecommunication Systems Management Association
		British Telecommunications plc
		Edwin L. Cox School of Business, Southern Methodist University
		IFIP WG 7.3 "Computer system modeling and performance evaluation"
		INFORMS Technical Section on Telecommunications
		INFORMS College of Information Systems
                The University of Texas at Dallas
		

The 9th International Conference on Telecommunication Systems - Modeling and 
Analysis will be held in Dallas, Texas, on March 15-18, 2001.  The conference will
build on the tradition of the earlier conferences. The general idea is to 
encourage informal interaction and exchanges of ideas by limiting the number of 
participants, concentrating on a few topics, and by presenting new problems and 
problem areas. The objective is to advance the state of the modeling and 
analysis in telecommunications by stimulating research activity on new and 
important problems.

The conference will be divided into segments with each segment devoted to a 
specific topic.  This will allow for little conflict between segments. Papers 
will be screened by the Program Committee to ensure the quality of 
presentations. A decentralized paper handling process will be used. Abstracts 
and papers should be submitted directly to a Program Committee member who will 
handle its review.  It is expected that this will expedite the paper review 
process.  Social and cultural activities will be included in the 2001 agenda. 
The conference will be held at the Edwin L. Cox School of Business, at Southern
Methodist University, near Downtown Dallas.

Listed below are some of the potential segments:

-- Configuration of ATM Networks
-- DSL and Cable Based Systems
-- Internet and its Impact on Commerce
-- Internet and Intranet
-- Mobility and Nomadicity
-- Multimedia modeling and analysis
-- Pricing and Economic Analysis of Internet and E-commerce
-- Topological Design and Network Configuration Problems
-- Design and Analysis of Local Access Networks and Outside
     Plant Problems
-- Low and Medium Earth Orbit Satellite Communication Systems
-- Cellular Systems and PCS Modeling and Configuration
-- Time Dependent Expansion of Telecommunication Systems
-- Network Reliability, Availability and Survivability
-- Network Design Problems in Gigabit and Terabit Networks
-- LAN, WAN Global Network Interconnection
-- Artificial Intelligence/Heuristics in Telecommunication Systems
-- Quantitative Methods in Network Management
-- Pricing and Economic Analysis of Telecommunications
-- Impact of Telecommunications on Industrial Organization
-- Performance Evaluation of Telecommunication Systems
-- Distributed Computing and Distributed Data Bases
-- Security and Privacy Issues in Telecommunications
-- Virtual Reality, Multimedia and their Impact
-- Standards

The Program Committee is open to any ideas you might have regarding additional 
topics or format of the conference.  The intention is, whenever possible, to 
limit the number of parallel sessions to three.  The conference is scheduled 
over a weekend so as to reduce teaching conflicts for academic participants, and 
to enable participants to take advantage of weekend hotel and airfare rates and 
of the many events that take place in the Downtown area.

Members of the Program Committee include:

Marco Ajmone Marsan, Politecnico di Torino, Italy <ajmone@polito.it>
Kemal Altinkemer, Purdue University, USA <kemal@mgmt.purdue.edu>
Suk-Gwon Chang, Hanyang University, Republic of Korea <changsg@email.hanyang.ac.kr>
Jerome Chifflet, France Telecom, France <jerome.chifflet@rd.francetelecom.fr>
Imrich Chlamtac, University of Texas at Dallas, USA <chlamtac@utdallas.edu>
Laurie G. Cuthbert, Queen Mary & Westfield College, UK <l.g.cuthbert@elec.qmw.ac.uk>
Lou Dellaverson, Motorola, Radio Research Lab, USA <FLD100@email.mot.com>
Piet Demeester, University of Ghent - IMEC, Belgium <demeester@intec.rug.ac.be>
Armin Eberlein, University of Calgary, Canada <eberlein@enel.ucalgary.ca>
Mark Epstein, Qualcomm, Inc., USA <mepstein@qualcomm.com>
Romano Fantacci, Universita di Firenze, Italy <fantacci@lenst.det.unifi.it>
Bezalel Gavish (Chairman), Southern Methodist University, USA <gavishb@mail.cox.smu.edu>
Luis Gouveia, Universidade de Lisboa, Portugal <lgouveia@fc.ul.pt>
Horst W. Hamacher, University of Kaiserslautern, Germany <hamacher@mathematik.uni-kl.de>
Richard J. Harris, Royal Melbourne Institute of Technology, Australia
  <richard@catt.rmit.edu.au>
Frank Huebner, Concert Technologies, USA <frank.huebner@concert.com>
Mladen  Kos, University of Zagreb, Croatia <kos@tel.fer.hr>
Joakim Kalvenes, Southern Methodist University, USA <kalvenes@mail.cox.smu.edu>
Konosuke Kawashima, NTT Advanced Technology Corp., JAPAN <shima@mitaka.ntt-at.co.jp>
Hans Kruse, Ohio University, USA <hkruse1@ohiou.edu>
Lorne G. Mason, Nanyang Technological University, Singapore <ELGMason@ntu.edu.sg>
Nikos E. Mastorakis,  Hellenic Naval Academy, Greece <mastor@softlab.ece.ntua.gr>
Armin R. Mikler, University of North Texas, USA <mikler@cs.unt.edu>
Sverrir Olafsson, BT Laboratories, UK <sverrir.olafsson@bt.com>
June S. Park, The University of Iowa, USA <june-park@uiowa.edu>
Hasan Pirkul, The University of Texas at Dallas, USA <hpirkul@utdallas.edu>
Guy Pujolle, University of Versailles, France <guy.pujolle@prism.uvsq.fr>
Dimitrios N. Serpanos, University of Patras, Greece <serpanos@ee.upatras.gr>
Yutaka Takahashi, Kyoto University, Japan <takahashi@i.kyoto-u.ac.jp>
J. L. van den Berg, KPN Research, The Netherlands <j.l.vandenberg@research.kpn.com>
Stefan Voss, University of Braunschweig, Germany <stefan.voss@tu-bs.de>
Lipo Wang, Nanyang Technological University, Singapore <elpwang@ntu.edu.sg>
Lars C. Wolf, Darmstadt University of Technology, Germany <lars.wolf@kom.tu-darmstadt.de>
Bill Yurcik, Illinois State University, USA <wjyurci@ilstu.edu>


Due to the limit on the number of participants, early conference registration is
recommended.  To ensure your participation, please use the following steps:

1. Send to a Program Committee member, by October 30, 2000, a paper
(electronic submissions are preferable), or titles and extended abstracts for 
potential presentations to be considered for the conference. Sending more than 
one contribution is encouraged, enabling the Program Committee to have a wider 
choice in terms of assigning talks to segments. Use E-mail to expedite the 
submission of titles and abstracts and papers. If you would like to organize a 
panel on a specific subject, please send the proposal to Bezalel Gavish.

2.  Use the form at the end of this message to pre-register for the 
conference.  Let us also know if you would like to have a formal 
duty during the conference such as: Session Chair, or Discussant.

3.  You will be notified by December 15, 2000, which abstract(s)/paper(s) 
have been selected for the conference. Detailed instructions on how to prepare 
camera-ready copies will be sent to authors of accepted presentations.  February 
1, 2001, is the deadline for sending a final version of the paper. Participants 
will receive copies of the collection of papers to be presented.

4.  All papers submitted to the conference will be considered for publication in 
the "Telecommunication Systems" journal. If you do not wish for your paper to 
be submitted for publication consideration in the "Telecommunication Systems" 
journal, please specify it in the cover letter of your submission.

The Program Committee looks forward to receiving your feedback/ideas. Feel free 
to volunteer any help you can offer.  If you have suggestions for Segment 
Leaders (i.e., individuals who will have a longer time to give an overview/state 
of the art talk on their segment subject) please E-mail them to Professor 
Gavish.  Also, if there are individuals whose participation you view as 
important, please send their names and E-mail addresses to a Program Committee 
Chairman, or forward to them a copy of this message.

I look forward to a very successful conference.

Sincerely yours,
Bezalel Gavish


-----------------------------------------------------------------------
				 Cut Here
-----------------------------------------------------------------------
	Ninth International Conference on Telecommunication Systems
			   Modeling and Analysis
			     REGISTRATION FORM
		Dates: March 15, 2001 (afternoon) to March 18, 2001

							Date: ______________

Name: ________________________________________	Title: _____________________

Affiliation:	____________________________________________________________

Address:	____________________________________________________________

		____________________________________________________________

		____________________________________________________________

Phone:		____________________________  FAX:	____________________

E-mail:	____________________________________________________________________

Potential Title of Paper(s): _______________________________________________

____________________________________________________________________________


I would like to Volunteer as			  Comments
A Session Chair:		Yes  No	____________________________________
A Discussant:			Yes  No	____________________________________
Organize a Session:		Yes  No	____________________________________
					____________________________________

REGISTRATION RATES and DEADLINES:
(Included in the registration fee are: Conference proceedings, a reception, two 
dinners, two lunches, coffee breaks, and cultural events.)

		Last Applicable		Academic	Industry	Corporate
		   Date			Rate		Rate		Rate
		---------------		--------	--------	--------
1. Pre-registration
		Until  Dec. 31, 2000	$ 430		$ 600		$1,500
2. Registration
		Until  Feb.  1, 2001	$ 530		$ 750		$1,500
3. On Site Registration
		After  Feb.  1, 2001	$ 630		$ 850		$1,500

As part of the conference registration dues you can become a member of the 
"American Telecommunication Systems Management Association". Please mark an "X" 
in the following entry if you wish to become an ATSMA member.

____	Yes, I wish to become an ATSMA member.
____	No, I do not wish to become an ATSMA member.

Mail your registration form and check to:

		Ms. Dru Lundeng
		ATSMA, Inc.
		Edwin L. Cox School of Business
		Southern Methodist University
		P.O. Box 750333
		Dallas, TX  75252, USA

Checks should be made payable to:

		ATSMA, Inc., Ninth Telecommunication Conference

Refund Policy: Half refund, for requests received by January 15, 2001.
No refund after January 15, 2001.


--=====================_247764807==_
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Professor Bezalel Gavish
Eugene J. and Ruth F. Constantin Distinguished Chair in Business
Chairman of the ITOM Department
Edwin L. Cox School of Business
Southern Methodist University
P.O. Box 750333
Dallas, TX 75275-0333
E-mail: gavishb@mail.cox.smu.edu


Tel: Office (214) 768-8258         FAX (214) 768-8812     Assistant (214) 
768-8256
       Home (615) 370-0813



--=====================_247764807==_--




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Subject: Re: Fwd: Re: Define a unicast RTP session.
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: "Strahm, Bill" <bill.strahm@intel.com>,
	Mark Baugher <mbaugher@passedge.com>,
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From: Vincent_Virgilio@nmss.com
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Hi Jonathan,

Whether there are one or two sessions is not completely academic, which is
close to the subject of this thread.  The point becomes concrete under
operation of the RTP MIB.  I think the SIP folks  would populate the MIB
session table differently than the H-media folks.  The former would put two
entries in the table where the latter would put one.

Vince Virgilio





Jonathan Rosenberg <jdrosen@dynamicsoft.com> on 09/01/2000 10:06:22 PM

To:   Vincent_Virgilio@nmss.com
cc:   "Strahm, Bill" <bill.strahm@intel.com>, Mark Baugher
      <mbaugher@passedge.com>, rem-conf@es.net
Subject:  Re: Fwd: Re: Define a unicast RTP session.




Vincent_Virgilio@nmss.com wrote:
>
> I agree with you that, under RFC 1889, an SDP invite and its OK will
create
> two sessions if the destination ports differ.  But if they don't (can
this
> be done predictably?), there should be (is) one session.  Later drafts of
> RFC 1889 do specify that the destination ports in a session may differ,
and
> thus might eventually cause trouble for SIP/SDP if its notion of a
session
> changes, if there's not trouble now.

There is no trouble now. Whether you call it one or two sessions in any
case seems completely academic. The operation in either case, as I can
tell, is the same.

>  Then it might become difficult for
> SIP/SDP to correlate the two directions of flow (is this necessary?).

Sorry? How do you arrive at this conclusion? As the caller, I choose the
ports I will receive media on, and I am told the ports to send media
two. They are correlated since they are part of the same SIP call. Its
been working in SIP devices for a while.

-Jonathan R.
--
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com








From rem-conf Tue Sep 05 14:28:14 2000 
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From: "Rex Coldren" <coldrenr@agcs.com>
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To: Flemming Andreasen <fandreas@cisco.com>
CC: rem-conf <rem-conf@es.net>
Subject: Re: Optional DTMF support in RFC 2833
References: <39A2C95B.290E8189@cisco.com>
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Flemming,
   Thanks for the heads-up.  The implementation note conflicts with what
is stated in RFC 2833, by the way.  RFC 2833 says the following:

   Since all implementations MUST be able to receive events 0 through
   15, listing these events in the a=fmtp line is OPTIONAL.

   I guess I'm wondering why the implementation notes say that the 0 thru
15 numbers MUST be declared when the RFC says declaring them is
optional.  Also, for my information, do the implementation notes override
the contents of the RFC?

Regards,
Rex Coldren

Flemming Andreasen wrote:

> Folks
>
> Just wanted to draw all RFC 2833 interested parties to the following
> implementation note (http://www.cs.columbia.edu/~hgs/rtp/payload.html):
>
> "Implementations can support events 0 through 15 (DTMF) by simply
> ignoring the packets, but MUST declare all event numbers that are
> meaningful to it in the fmtp parameter, including 0 through 15."
>
> Thanks
>
>         Flemming
>
> --
> Flemming Andreasen
> Cisco Systems




From rem-conf Tue Sep 05 15:28:17 2000 
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Rex Coldren wrote:

> Flemming,
>    Thanks for the heads-up.  The implementation note conflicts with what
> is stated in RFC 2833, by the way.

I guess that depends on how you interpret the words.

> RFC 2833 says the following:
>
>    Since all implementations MUST be able to receive events 0 through
>    15, listing these events in the a=fmtp line is OPTIONAL.
>
>    I guess I'm wondering why the implementation notes say that the 0 thru
> 15 numbers MUST be declared when the RFC says declaring them is
> optional.

I should probably let Henning answer, but the idea is to not require all RFC
2833 implementations to implement DTMF support. The reason is, that RFC 2833
is a general and very useful mechanism, and requiring all use of it to
include support for DTMF is overly restrictive. The note provides a way out
of that and I believe will be included in RFC 2833 when it progresses to
Draft Standard.



> Also, for my information, do the implementation notes override
> the contents of the RFC?

They clarify them at least.

-- Flemming

>
>
> Regards,
> Rex Coldren
>
> Flemming Andreasen wrote:
>
> > Folks
> >
> > Just wanted to draw all RFC 2833 interested parties to the following
> > implementation note (http://www.cs.columbia.edu/~hgs/rtp/payload.html):
> >
> > "Implementations can support events 0 through 15 (DTMF) by simply
> > ignoring the packets, but MUST declare all event numbers that are
> > meaningful to it in the fmtp parameter, including 0 through 15."
> >
> > Thanks
> >
> >         Flemming
> >
> > --
> > Flemming Andreasen
> > Cisco Systems

--
Flemming Andreasen
Cisco Systems





From rem-conf Tue Sep 05 17:30:11 2000 
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From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
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CC: coldrenr@agcs.com, rem-conf <rem-conf@es.net>
Subject: Re: Optional DTMF support in RFC 2833
References: <39A2C95B.290E8189@cisco.com> <39B563EB.E7999097@agcs.com> <39B57367.1FCB5071@cisco.com>
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Flemming Andreasen wrote:
> 
> Rex Coldren wrote:
> 
> > Flemming,
> >    Thanks for the heads-up.  The implementation note conflicts with what
> > is stated in RFC 2833, by the way.
> 
> I guess that depends on how you interpret the words.
> 
> > RFC 2833 says the following:
> >
> >    Since all implementations MUST be able to receive events 0 through
> >    15, listing these events in the a=fmtp line is OPTIONAL.
> >
> >    I guess I'm wondering why the implementation notes say that the 0 thru
> > 15 numbers MUST be declared when the RFC says declaring them is
> > optional.
> 
> I should probably let Henning answer, but the idea is to not require all RFC
> 2833 implementations to implement DTMF support. The reason is, that RFC 2833
> is a general and very useful mechanism, and requiring all use of it to
> include support for DTMF is overly restrictive. The note provides a way out
> of that and I believe will be included in RFC 2833 when it progresses to
> Draft Standard.

It is also backward compatible for now, in that declaring support is
currently redundant. 

Practically speaking, with DTMF, support is also a rather vague term.
Support could mean anything short of core dumping. If the other side
silently swallows type 0-15, is that support? Presumably, this would
only occur in real systems if DTMF doesn't make sense. Maybe it would be
helpful is somebody could provide examples of RFC 2833 systems that
would have no use for DTMF. If you're building a system that logically
should support DTMF, but doesn't, having the protocol police chase after
you is probably not your major concern.

> 
> > Also, for my information, do the implementation notes override
> > the contents of the RFC?
> 
> They clarify them at least.

No, they obviously don't override the RFC. However, RFC 2026 says

   A Proposed Standard specification is generally stable, has resolved
   known design choices, is believed to be well-understood, has received
   significant community review, and appears to enjoy enough community
   interest to be considered valuable.  However, further experience
   might result in a change or even retraction of the specification
   before it advances.

Thus, implementors can't completely rely on every word staying the same.
The implementation note is meant to tell implementors "indicate DTMF
support now explicitly as it will prevent problems later". 

A typical way of resolving this would be to ask around, at Draft time,
whether any known implementation is having problems with that change.


-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



From rem-conf Tue Sep 05 19:29:07 2000 
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From: "Koji Imura" <imura@telecom.mci.mei.co.jp>
To: <rem-conf@es.net>, "Colin Perkins" <colin@isi.edu>
Cc: <4on2andIP-sys@advent.ee.columbia.edu>
References: <200008251549.QAA01356@purple.cs.ucl.ac.uk>
Subject: Re: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt
Date: Wed, 6 Sep 2000 11:21:24 +0900
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Dear all,

I reviewed the draft and had a question.
In the section 3.1, there is a description about the marker bit and
time stamp for the MPEG-4 visual.
According to this description, it is possible to include some VOPs
in one RTP packet.
In such a case, the only time stamp for first VOP is included in the
RTP header, but there is no time information for rest of VOPs in the
same RTP packet.
The temporal interval between VOPs is variable in MPEG-4 visual
So it is difficult to estimate the time stamp of the other VOPs.

If there is a system that presentation/composition time is decided
by the RTP time stamp, how do the system estimate the time stamp
of rest of the VOPs?

Best Regards,

Koji

-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
Koji Imura(Koji.Imura@yrp.mci.mei.co.jp)
Matsushita Communication Industrial
Phone:+81-468-40-5684 Fax: +81-468-40-5184
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=


----- Original Message -----
From: Colin Perkins <colin@isi.edu>
To: <rem-conf@es.net>
Cc: <4on2andIP-sys@advent.ee.columbia.edu>
Sent: Saturday, August 26, 2000 12:49 AM
Subject: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt


> To the IETF AVT working group and the ad-hoc group on transport of
MPEG4-on-IP:
>
> The IETF Audio/Video Transport working group has received a request to
publish
>     http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-mpeg4-es-03.txt
> the "RTP payload format for MPEG-4 Audio/Visual streams" as a proposed
standard.
>
> Accordingly, this message starts a two week working group last call for
comments
> on this draft. All comments should be sent to the AVT working group
mailing list
> <rem-conf@es.net> by 8th September 2000. At this time we will make our
decision
> whether or not to request IESG approval for publication, based on the
consensus
> of replies.
>
> We particularly value input from the MPEG committee as to technical
correctness
> and the appropriate nature of this specification. It is our intent that
this
> payload format is complementary to the ongoing work in MPEG defining a
general
> framework for transport of the MPEG-4 Systems environment. If this intent
is not
> reflected in the draft, we value comment with detailed and specific
suggestions
> for change.
>
> Note that we had originally planned to issue this working group last call
after
> the forthcoming MPEG meeting. However, time constraints imposed by the
ITU-T
> schedule for publication of the next version of H.323 require that we
issue this
> last call early. We hope that this will not cause undue problems, and look
forward
> to your input.
>
> Thanks,
> Colin Perkins (IETF AVT co-chair)
>
>




From rem-conf Tue Sep 05 19:41:11 2000 
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CC: Flemming Andreasen <fandreas@cisco.com>, rem-conf <rem-conf@es.net>
Subject: Re: Optional DTMF support in RFC 2833
References: <39A2C95B.290E8189@cisco.com> <39B563EB.E7999097@agcs.com> <39B57367.1FCB5071@cisco.com> <39B58F64.EE30FEAF@cs.columbia.edu>
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Flemming, Henning,

I agree with both of you.  DTMF support should not be required.  Realistically,
there will likely be zero implementations that will implement all of the possible
combinations of events and tones.  So how do we deal with trying to push RFC
2833 toward Draft Standard when none of the implementations support the full
set of capabilities?  Do the IETF rules for deprecating options require that all of
the interoperable implementations support option sets that are a union or an
intersection of working options of the common group (forgive my ignorance of
the rules; RFC 2026 didn't help me much here)?  I am hoping that a simple
intersecting set is the answer.  Otherwise RFC 2833 will disintegrate.  Thoughts?
.
Cheers,
Rex Coldren

Henning Schulzrinne wrote:

> Flemming Andreasen wrote:
> >
> > Rex Coldren wrote:
> >
> > > Flemming,
> > >    Thanks for the heads-up.  The implementation note conflicts with what
> > > is stated in RFC 2833, by the way.
> >
> > I guess that depends on how you interpret the words.
> >
> > > RFC 2833 says the following:
> > >
> > >    Since all implementations MUST be able to receive events 0 through
> > >    15, listing these events in the a=fmtp line is OPTIONAL.
> > >
> > >    I guess I'm wondering why the implementation notes say that the 0 thru
> > > 15 numbers MUST be declared when the RFC says declaring them is
> > > optional.
> >
> > I should probably let Henning answer, but the idea is to not require all RFC
> > 2833 implementations to implement DTMF support. The reason is, that RFC 2833
> > is a general and very useful mechanism, and requiring all use of it to
> > include support for DTMF is overly restrictive. The note provides a way out
> > of that and I believe will be included in RFC 2833 when it progresses to
> > Draft Standard.
>
> It is also backward compatible for now, in that declaring support is
> currently redundant.
>
> Practically speaking, with DTMF, support is also a rather vague term.
> Support could mean anything short of core dumping. If the other side
> silently swallows type 0-15, is that support? Presumably, this would
> only occur in real systems if DTMF doesn't make sense. Maybe it would be
> helpful is somebody could provide examples of RFC 2833 systems that
> would have no use for DTMF. If you're building a system that logically
> should support DTMF, but doesn't, having the protocol police chase after
> you is probably not your major concern.
>
> >
> > > Also, for my information, do the implementation notes override
> > > the contents of the RFC?
> >
> > They clarify them at least.
>
> No, they obviously don't override the RFC. However, RFC 2026 says
>
>    A Proposed Standard specification is generally stable, has resolved
>    known design choices, is believed to be well-understood, has received
>    significant community review, and appears to enjoy enough community
>    interest to be considered valuable.  However, further experience
>    might result in a change or even retraction of the specification
>    before it advances.
>
> Thus, implementors can't completely rely on every word staying the same.
> The implementation note is meant to tell implementors "indicate DTMF
> support now explicitly as it will prevent problems later".
>
> A typical way of resolving this would be to ask around, at Draft time,
> whether any known implementation is having problems with that change.
>
> --
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs




From rem-conf Tue Sep 05 19:59:38 2000 
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From: "Rex Coldren" <coldrenr@agcs.com>
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Subject: Re: Optional DTMF support in RFC 2833
References: <39A2C95B.290E8189@cisco.com> <39B563EB.E7999097@agcs.com> <39B57367.1FCB5071@cisco.com> <39B58F64.EE30FEAF@cs.columbia.edu> <39B5ADED.7B2DC541@agcs.com>
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Ooops.  I meant that I hope the union of options is what is required.  Sorry
for the slip...

Rex Coldren wrote:

> Flemming, Henning,
>
> I agree with both of you.  DTMF support should not be required.  Realistically,
> there will likely be zero implementations that will implement all of the possible
> combinations of events and tones.  So how do we deal with trying to push RFC
> 2833 toward Draft Standard when none of the implementations support the full
> set of capabilities?  Do the IETF rules for deprecating options require that all of
> the interoperable implementations support option sets that are a union or an
> intersection of working options of the common group (forgive my ignorance of
> the rules; RFC 2026 didn't help me much here)?  I am hoping that a simple
> intersecting set is the answer.  Otherwise RFC 2833 will disintegrate.  Thoughts?
> .
> Cheers,
> Rex Coldren
>
> Henning Schulzrinne wrote:
>
> > Flemming Andreasen wrote:
> > >
> > > Rex Coldren wrote:
> > >
> > > > Flemming,
> > > >    Thanks for the heads-up.  The implementation note conflicts with what
> > > > is stated in RFC 2833, by the way.
> > >
> > > I guess that depends on how you interpret the words.
> > >
> > > > RFC 2833 says the following:
> > > >
> > > >    Since all implementations MUST be able to receive events 0 through
> > > >    15, listing these events in the a=fmtp line is OPTIONAL.
> > > >
> > > >    I guess I'm wondering why the implementation notes say that the 0 thru
> > > > 15 numbers MUST be declared when the RFC says declaring them is
> > > > optional.
> > >
> > > I should probably let Henning answer, but the idea is to not require all RFC
> > > 2833 implementations to implement DTMF support. The reason is, that RFC 2833
> > > is a general and very useful mechanism, and requiring all use of it to
> > > include support for DTMF is overly restrictive. The note provides a way out
> > > of that and I believe will be included in RFC 2833 when it progresses to
> > > Draft Standard.
> >
> > It is also backward compatible for now, in that declaring support is
> > currently redundant.
> >
> > Practically speaking, with DTMF, support is also a rather vague term.
> > Support could mean anything short of core dumping. If the other side
> > silently swallows type 0-15, is that support? Presumably, this would
> > only occur in real systems if DTMF doesn't make sense. Maybe it would be
> > helpful is somebody could provide examples of RFC 2833 systems that
> > would have no use for DTMF. If you're building a system that logically
> > should support DTMF, but doesn't, having the protocol police chase after
> > you is probably not your major concern.
> >
> > >
> > > > Also, for my information, do the implementation notes override
> > > > the contents of the RFC?
> > >
> > > They clarify them at least.
> >
> > No, they obviously don't override the RFC. However, RFC 2026 says
> >
> >    A Proposed Standard specification is generally stable, has resolved
> >    known design choices, is believed to be well-understood, has received
> >    significant community review, and appears to enjoy enough community
> >    interest to be considered valuable.  However, further experience
> >    might result in a change or even retraction of the specification
> >    before it advances.
> >
> > Thus, implementors can't completely rely on every word staying the same.
> > The implementation note is meant to tell implementors "indicate DTMF
> > support now explicitly as it will prevent problems later".
> >
> > A typical way of resolving this would be to ask around, at Draft time,
> > whether any known implementation is having problems with that change.
> >
> > --
> > Henning Schulzrinne   http://www.cs.columbia.edu/~hgs




From rem-conf Wed Sep 06 04:39:56 2000 
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Date: Wed, 06 Sep 2000 13:28:40 +0200
From: Clever Ricardo Guareis de Farias <farias@cs.utwente.nl>
Organization: CTIT - Centre for Telematics and Information Technology
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Subject: IDMS 2000: Call for Posters and Demonstrators
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[Our apologies if you receive multiple copies of this Call]

              Call for Posters and Demonstrators

                           IDMS 2000

        7th International Workshop on Interactive
Distributed Multimedia Systems and Telecommunication Services

     CTIT/Univ. of Twente, Enschede, the Netherlands
                 October 17-20, 2000
           http://www.ctit.utwente.nl/Docs/news/idms_2000.htm

          In cooperation with ACM SIGCOMM and SIGMM
   Sponsored by KPN Research, Lucent Technologies and Philips


The goal of the IDMS series of workshops is to bring together
researchers, developers, and practitioners from academia and
industry; and to provide a forum for discussion, presentation,
and exploration of technologies and advances in the broad field of
interactive distributed multimedia systems and telecommunication
services. Topics covered in IDMS workshops range from basic system
technologies such as networking and operating system support
to all kinds of teleservices and distributed multimedia
applications.

IDMS 2000 consists of a full day of tutorials, a three days
technical program, and demonstrations during the workshop. For
the latter, we solicit proposals for posters and demonstrators
associated with the scope of the workshop. Posters and demo's
offer oppertunities to researchers and developers to present
and discuss their work in progress and practical results
through more informal interaction. Posters and demo's will
be selected based on their suitability for the workshop and
their technical feasibility. Posters should be presentable on
1 or 2 A0 sheets; demo proposals should indicate which technical
infrastructure facilities are required.

Selected posters and demo's will be announced on the IDMS
website, together with short descriptions, if provided.

Proposals can be sent to:

Richard Bults, bults@ctit.utwente.nl

Deadline for receipt of proposals: October 2nd, 2000




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From: <jo333_4@lawrence-lca.co.uk>
Subject: Inspired Internet Business Opportunity!
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Date: Wed, 6 Sep 2000 08:33:33 -0700
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Greetings,

Have you ever asked yourself, "If only I could find an easier
way to make a higher income" and "If I had more money, I could
spend more time with my family, and less time at work" and 
"I sure could use more money so I could pay off my bills
once and for all" and "I would love to get involved in a 
business which will generate money while I am at work."



Please, please, print this off and read thoroughly.  Be 
sure you don't miss any of the points outlined.  Then put
it down and read it again. I am sending you a whole lot 
of information in which you might not understand the first
time you read it. If you don't believe this program will 
work for you, send it to 10-20 of your closest friends 
(in which you deeply trust), ask them what they think? 
This really works! Have faith, don't miss this opportunity,
get involved also, it will work for you as it does for us!!!
Read On!!!!!!!



Due to the popularity of this letter on the Internet, A 
Major Nightly News Program in the USA recently dedicated
an entire show to the investigation of the program described 
below to see if it really can make people money. The show 
also investigated whether or not the program was legal. Their 
findings proved that there are absolutely no laws prohibiting 
the participation in the program. This has helped to show 
people that this is a simple, harmless and fun way to make
some extra money at home. The results have been truly 
remarkable. So many people are now participating that
those involved are doing much better than ever before.
Since everyone makes more as more people try it out, it has
been very exciting. You will understand only if you get involved!


The Entire Plan is Here Below.....


****Print This Now for Future Reference****



Read it often to fully understand how simple and easy it works!!!

If you would like to make at least... $50,000 US in less than 
90 Days! If not, forward this to someone who would like to make
this kind of money........ It works (like designed) but only 
for those who follow it to the letter!! Please read this program...
Then read it again!!$$$$$$$$$$$$$$$$$


This is a Legitimate, Legal, Money Making Opportunity!! It does
not require you to come into contact with people or make or 
take any telephone calls. Just follow the instructions and you 
will make money.  This simplified E-mail marketing program works
perfectly 100% EVERY TIME!!!


Hello, my name is Johnathon Rourke, I'm from Rhode Island. The
enclosed information is something I almost let slip through my 
fingers.  Fortunately, sometime later I re-read everything and 
gave some considerable thought and study to it.  Two years ago, 
the Corporation I worked for the past twelve years down-sized 
and my position was eliminated. After unproductive job interviews, 
I decided to open my own business.  Over the past year, I incurred
many unforeseen financial problems. I owed my family, friends 
and creditors over $35,000.  The economy was taking a toll on 
my business and I just couldn't seem to make ends meet. I had 
to refinance and borrow more against my home to support my 
family and struggling business. At that moment something 
significant happened in my life.


I am writing this reference to share the experience in hope that
this could change your life... Forever!!! Financially $$$$!!! 
In mid-December, I received this program in my E-mail.  Six 
months prior to receiving this program I had been sending away
for information on various business opportunities. All of the 
programs I received, in my opinion, were not cost effective, 
they were either too difficult for me to comprehend or the 
initial investment was too much for me to risk to see if they 
worked.


As I was saying in December of 1997 I received this program. 
I didn't send for it, or ask for it, they just got my name 
off a mailing list. Thank goodness for that!!!


After reading it several times, to make sure I was reading 
it correctly.  I couldn't believe my eyes! Here was a 
MONEY MAKING MACHINE I could start immediately without any debt. 
Like most of you I was still a little skeptical and a little 
worried about the legal aspects of it all.  So I checked it out with 
the U.S. Post Office (1-800-725-2161 24-hrs) and they confirmed that 
it is indeed legal! After determining the program was LEGAL 
I decided "WHY NOT!?!?!?!?"


Initially I sent out 10,000 e-mails.  It cost me about $15 for 
my time on-line. The great thing about e-mail is that I don't need
any for printing to send out the program, and because I also 
send the product (reports) by e-mail, my only expense is my 
time. In less than one week, I was starting to receive orders 
for REPORT # 1. By January 13 I had received 26 orders for REPORT # 1 .
Your goal is to "RECEIVE at least 20  ORDERS  FOR REPORT #1 
WITHIN  2  WEEKS,  IF YOU DON'T,  SEND MORE PROGRAMS UNTIL YOU DO". 
My first step in making $50,000 in 90 days was done. By January 30, 
I had received 196 orders for REPORT #2. Your goal is to 
"RECEIVE  AT  LEAST  100  +  ORDERS FOR  REPORT  #2   
WITHIN  2  WEEKS, IF NOT, SEND OUT MORE PROGRAMS UNTIL YOU DO.
ONCE YOU HAVE 100  ORDERS, THE REST IS EASY, RELAX, YOU WILL MAKE 
YOUR $50,000 GOAL".


Well, I had 196 orders for REPORT  # 2.  96 more than I needed.  
So I sat back and relaxed.  By March 1, of my E-mailing of 10,000, 
I had received $58,000 with more coming in every day. 
I paid off ALL my debts and bought a much needed new car!


Please take your time to read this plan, IT WILL CHANGE YOUR
LIFE FOREVER $!!!!!!! Remember, it won't work if you don't try it. 
This program does work, BUT you must follow it EXACTLY! Especially 
the rules of not trying to place your name in a different place. It 
won't work and you'll lose out on a lot of money!  In order for this
program to work, you must meet your goal 0f 20+ orders for REPORT # 1, 
and 100+ orders for REPORT # 2, and you will make $50,000 or more in 
90 days.I AM LIVING PROOF THAT IT WORKS !!!!!! If you choose not to 
participate in this program, I am sorry.  It really is a great 
opportunity with little costor no risk to you.


If you choose to participate, follow the program and you will 
be on your way to financial security. If you are a fellow business
owner and are in financial trouble like I was, or you want to start 
your own business, consider this a sign. I DID $$$


Sincerely,

Johnathan Rourke


**************************



A PERSONAL NOTE FROM THE ORIGINATOR OF THIS PROGRAM: By the time you
have read the enclosed program and reports, you should have concluded 
that such a program, and one that is legal, could not have been
created by an amateur.  Let me tell you a little about myself.


I had a profitable business for 10 years.  Then in 1979 my business
began falling off.  I was doing the same things that were previously
successful for me, but it wasn't working. Finally, I figured it out.
It was not me, it was the economy. Inflation and recession had replaced 
the stable economy that had been with us since 1945. 
I don't have to tellyou what happened to the employment rate.... 
because many of you know fromfirst hand experience.  There were 
more failures and bankruptcies than everbefore..  The middle 
class was vanishing. Those who knew what they were doinginvested 
wisely and moved up. Those who did not, including those who 
never had anything to save or to invest, were moving down 
into the ranks of the poor. As the saying goes, "THE RICH GET 
RICHER AND THE POOR GET POORER."


The traditional methods of making money will never allow you to 
"move up" or "get rich", inflation will see to that. 
You have just received information that can give you Financial-Freedom
for the rest of your life, with "NO RISK" and "JUST A LITTLE
BIT OF EFFORT."You can make more money in the next few months 
than you have ever imagined.I should also point out that I
will not see a penny of this money,nor anyone else who has 
provided a testimonial for this program.



I have retired from the program after sending thousands and 
thousands of programs. Follow the program EXACTLY AS INSTRUCTED. 
Do not change it in any way. It works exceedingly well as it is now. 
Remember to e-mail a copy of this exciting report to everyone you
can think of. One of the people you send this to may send 
out 50,000... and your name will be on everyone of them!


REMEMBER though, ~~~~ the MORE YOU SEND OUT, the more 
potential customersyou will reach.  So my friend, I have 
given you the ideas, information, materials and opportunity 
to become financially independent. IT IS UP TO YOU !!!    NOW DO IT !!!!


BEFORE YOU delete this program from your in box, as I almost did,
take a little time to read it and REALLY THINK ABOUT IT. Get a 
pencil and figure out what could happen when YOU participate.
Figure out the worst possible response and no matter how you 
calculate it, you will still make a lot of money!  You will
definitely get back what you invested. Any doubt you have will
vanish when your first orders come in.

$$$$$ IT WORKS!!! $$$$$



Jody Jacobs Richmond, VA

HERE'S HOW THIS AMAZING PROGRAM WILL
MAKE YOU THOUSANDS OF DOLLARS $$$$$


This method of raising capital REALLY WORKS 100% EVERY TIME. I am 
sure that you could use up to $50,000 or more in the next 90 days. 
Before you say "BULL...", please read this program care-fully.  
This is not a chain letter, but a perfectly legal money making business.


As with all multi-level businesses, we build our business by 
recruiting new partners and selling our products.  Every state in
the USA allows you to recruit new multi-level business partners, 
and we sell and deliver a product for EVERY dollar received. 
YOUR ORDERS COME BY MAIL AND ARE FILLED BY E-MAIL, so you are not 
involved in personal selling. You do it privately in your own home,
store or office. This is the EASIEST marketing plan anywhere! 
It is simply order filling by E-mail. The product is informational and 
instructional material; keys to the secrets for everyone on 
how to open the doors to the magic world of E-COMMERCE, 
the information high-way, the wave of the future!



PLAN SUMMARY:

(1) You order the 4 reports listed below ($5 US each). 
They come to you by e-mail.

(2) Save a copy of this entire letter and put your name 
after Report#1 and move the other names down.

(3) Via the internet, access Yahoo.com or any of the other
major searchengines to locate hundreds of bulk email 
service companies (search for "bulk email") and have them send 
25,000 - 50,000 e-mail for you about $49 +.

(4) Orders will come to you by postal mail - simply email them the
Report they ordered. Let me ask you - isn't this about as easy as
it gets?.


By the way there are over 50 MILLION email addresses with 
millions more joining the Internet each year so don't worry 
about "running out" or "saturation".  People are used to seeing
and hearing the same advertisements every day on radio and TV.
How many times have you received the same pizza flyers on your 
door?  Then one day you are hungry for a pizza and you order 
one. Same thing with this letter.  I received this letter many 
times - then one day I decided it was time to try it.


YOU CAN START TODAY - JUST DO THESE EASY STEPS:


STEP # 1 ORDER THE FOUR REPORTS
Order the four Reports shown on the list below (you can't sell
them if you don't order them).  For each Report send $5.00 US 
CASH, the NAME & NUMBER OF THE REPORT YOU ARE ORDERING, 
YOUR E-MAIL ADDRESS, and YOUR NAME & RETURN ADDRESS 
(in case of a problem) to the person whose name appears on the 
list next to the report. MAKE SURE YOUR RETURN ADDRESS IS ON 
YOUR ENVELOPE IN CASE OF ANY MAIL PROBLEMS!


Within a few days you will receive, by e-mail, each of the 
four reports. Save them on your computer so you can send 
them to the 1,000's of people who will order them from you.



STEP # 2 ADD YOUR MAILING ADDRESS TO THIS LETTER

A. Look below for the listing of the four reports.
B. After you've ordered the four reports delete the name and address
under REPORT # 4. This person has made it through the cycle.
C. Move the name and address under REPORT # 3 down to REPORT # 4.
D. Move the name and address under REPORT # 2 down to REPORT # 3.
E. Move the name and address under REPORT # 1 down to REPORT # 2.
F. Insert your name and address in the REPORT # 1 position


Please make sure you COPY ALL INFORMATION, every name and address,
ACCURATELY.


STEP # 3 SAVE THIS ON COMPUTER

Take this entire letter, including the modified list of names, 
and save it to your computer.  Make NO changes to these instructions.
Now you are ready to use this entire E-mail to send by E-mail to prospects.


Report # 1 will tell you how to download bulk e-mail software 
and e-mail addresses so you can send it out to thousands of people
while you sleep! Remember that 50,000 + new people are joining the 
internet every month.Your cost to participate in this is practically
nothing (surely you can afford $20 US and initial bulk mailing cost).
You obviously already have a computer and an Internet connection 
and E-mail is FREE.

There are two primary methods of building your downline:
METHOD # 1: SENDING BULK E-MAIL, Let's say that you decide to 
start small, just to see how it goes, and we'll assume you and
all those involved email out only 2,000 programs each. Let's 
also assume that the mailing receives a 0.5% response. The response
could be much better. Also, many people will email out hundreds of 
thousands of programs instead of 2,000 (why stop at 2,000?) But 
continuing with this example, you send out only 2,000 programs. 
With a 0.5% response, that is only 10 orders for REPORT # 1. 
Those 10 people respond by sending out 2,000 programs each for 
a total of 20,000. Out of those 0.5%, 100 people respond and 
order REPORT  # 2. Those 100 mail out 2,000 programs each for
a total of 200,000. The 0.5% respond to that is 1,000 orders for...
REPORT  # 3. Those 1,000 send out 2,000 programs each for a 
2,000,000 total. The 0.5% respond to that is 10,000 orders for...
REPORT # 4. That's 10,000 $5 US bills for you. CASH!!! 
Your total income in this example is...$50 + $500 + 
$5000 + $50,000 for a total of...$55,550!!!!


Remember friend, this is assuming 1,990 out of 2,000 people you
email to will do absolutely nothing and trash this program! Dare 
to think for a moment what would happen if everyone, or half sent 
out 100,000 programs instead of 2,000, believe me, many people 
will do just that and more!


METHOD # 2: Placing free Ads on the Internet.  Advertising on
the Internet is very, very inexpensive, as there are HUNDREDS 
of FREE places to advertise. Lets say you decide to start small
just to see how well it works. Assume your goal is to get ONLY 
10 people to participate on your first level (placing a lot of 
FREE ads on the Internet will easily get a LARGER response). 
Also assume that everyone in your organization gets only 10
downline members, just look at how this small number accumulates 
to achieve the STAGGERING results below...

1st Level-- Your first 10 send you $5 = $50
2nd Level-- 10 Members from those 10 ($5X100) =  $500
3rd Level-- 10 Members from those 100 ($5X1000) =  $5,000
4th Level-- 10 Members from those 1,000 ($5X10,000) = $50,000
$$$$$$$ THIS TOTALS = $55,550 $$$$$$$



AMAZING ISN'T IT??  Remember friends, this assumes that the
people who participate only recruit 10 people each. Think for
a moment what would happen if they got 20 people to participate!
Most people get 100's of participants and many will continue to 
work this program, WITH YOUR NAME ON THEM FOR YEARS!! THINK ABOUT IT!
People are going to get details about this plan from you 'or' someone 
else...the question is... Don't you want your name to be on the Emails 
they will send out??
*** DON'T MISS OUT!!!
*** JUST TRY IT ONCE!!!
*** SEE WHAT HAPPENS!!!
*** YOU'LL BE AMAZED!!!


ALWAYS PROVIDE SAME-DAY SERVICE ON ORDERS! This will guarantee 
that the Email THEY send out with your name and address on it 
will be prompt because they can't advertise until they receive 
the REPORT!


GET STARTED TO-DAY: PLACE YOUR ORDER FOR THE FOUR REPORTS...NOW!!
NOTE  ALWAYS SEND $5 CASH (US CURRENCY) FOR EACH REPORT. CHEQUES
ARE NOT ACCEPTED!  Make sure the CASH is concealed by wrapping it
in two sheets of paper. On one of those sheets write...
(A) The Number and Name of the Report you are Ordering.
(B) Your E-mail address, and
(C) Your name and postal address.



REPORT # 1 "The Insider's Guide to Advertising for Free on the Internet".
ORDER FROM...
CAROL ANDERSON
5451 PLEASANT VIEW LANE
FREELAND, WA  98249
USA.


PLEASE NOTE...I and every member below will work for you also. Try us!
REPORT # 2  "The Insider's Guide to Sending Bulk email.
ORDER FROM...
HARRY FERNTHEIL
5626 WOODRIDGE STREET
HUNTSVILLE, AL  35802
USA.


REPORT  # 3 "The Secrets to Multilevel Marketing on the Internet".
ORDER FROM...
JOCA VAN DEN BOS
1191 PUKAKI STREET
ROTORUA, 3201
NEW ZEALAND.


REPORT  # 4  "How to become a Millionaire Utilizing the
Power of Multilevel Marketing and the Internet".
ORDER FROM...
LANCE DENTON
100 WILDWOOD RD.
ROCKPORT, TX  78382
USA.



*******TIPS FOR SUCCESS*******

TREAT THIS AS YOUR BUSINESS!!

Be prompt, professional, follow the directions accurately.  Send 
for the four reports 'IMMEDIATELY' so you have them when the orders
start coming in because...when you receive a $5 order, you MUST send
out the requested product/report. It is required for this to be a 
legal business and they need the reports to send out their letters 
(with your name on them!)



ALWAYS PROVIDE SAME-DAY SERVICE ON THE ORDERS YOU RECEIVE.
BE PATIENT AND PERSISTENT WITH THIS PROGRAM!!
If you follow the instructions exactly-results
WILL FOLLOW. $$$$.


******* YOUR SUCCESS GUIDELINES*******
Follow these guidelines to guarantee your success:
If you don't receive 20 orders for REPORT # 1 within two weeks,
continue advertising or sending e-mails until you do. Then, a couple 
weeks later you should receive at least 100 or more orders for REPORT 
# 2. If you don't, continue advertising or sending e-mails until you do.
Once you have received 100 or more orders for REPORT  # 2, YOU CAN RELAX,
because the system is already working for you, cash will continue to roll 
in!! THIS IS IMPORTANT TO REMEMBER... Every time your name is moved down 
on the list, you are placed in front of a DIFFERENT REPORT. You can keep 
track of your PROGRESS by watching which report people are ordering from 
you. To generate more income, simply send another batch of e-mails or 
continue placing ads and start the whole process again!!



THERE IS NO LIMIT TO THE INCOME YOU WILL GENERATE FROM
THIS BUSINESS!


Before you make your decision as to weather or not to
participate in this program. Please answer one question.
ARE YOU HAPPY WITH YOUR PRESENT INCOME OR JOB?? If the
answer is no, then please look at the following facts
about this super simple MLM program:
1~ No face to face selling, No meetings, No Inventory,
No telephone calls, No big cost to start, Nothing to learn,
No skills needed! (Surely you know how to send an e-mail?).
2~ No equipment to buy- you already have a computer
and Internet connection- so you have everything you
need to fill orders!
3~ You are selling a PRODUCT which does NOT COST YOU
ANYTHING TO PRODUCE OR SHIP! E-mailing copies of
the Reports is FREE!
4~ All of your customers pay you in CASH! This program
will change your LIFE FOREVER! Look at the potential for
you to be able to quit your job, and live the life of LUXURY
that you could only dream about before!



Imagine getting out of debt and buying the car and home of your 
dreams and being able to work a super-high paying leisurely 
business, from home with no overhead!
$$$ FINALLY MAKE SOME DREAMS COME TRUE !! $$$ ACT NOW!!
Take your first step towards achieving financial independence.


Order the  REPORTS and follow the PROGRAM outlined above.
SUCCESS WILL BE YOUR REWARD !!!
Thank you for your time and consideration.

PLEASE NOTE: If you need help with starting a business, registering
a business name, learning how income tax is handled, etc; contact
your local office of the Small Business Administration  
(a Federal Agency) 1-800-827-5722 for free help and answers 
to questions. Also, the Internal Revenue Service offers free
help via the telephone and free seminars about business tax requirements.
Your earnings are highly dependent on your activities and 
advertising. The information contained on this site and in
the report constitutes no guarantees stated or implied. In 
the event that it is determined that this site or report 
constitutes a guarantee of any kind, that guarantee is now void.
The earnings amounts listed on this site and in the report are 
estimates only. If you have any questions of the legality of 
this program, contact the Office of Associate Director for 
Marketing Practices, Federal Trade Commission, Bureau of 
Consumer Protection in Washington, DC. Under Bill 1618 TITLE III.
This is a ONE TIME ONLY e-mail transmission. 
No request for removal is necessary.



From rem-conf Wed Sep 06 10:07:55 2000 
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Date: Wed, 06 Sep 2000 12:58:32 -0400
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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To: coldrenr@agcs.com
CC: Flemming Andreasen <fandreas@cisco.com>, rem-conf <rem-conf@es.net>
Subject: Re: Optional DTMF support in RFC 2833
References: <39A2C95B.290E8189@cisco.com> <39B563EB.E7999097@agcs.com> <39B57367.1FCB5071@cisco.com> <39B58F64.EE30FEAF@cs.columbia.edu> <39B5ADED.7B2DC541@agcs.com>
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Rex Coldren wrote:
> 
> Flemming, Henning,
> 
> I agree with both of you.  DTMF support should not be required.  Realistically,
> there will likely be zero implementations that will implement all of the possible
> combinations of events and tones.  So how do we deal with trying to push RFC
> 2833 toward Draft Standard when none of the implementations support the full
> set of capabilities?  Do the IETF rules for deprecating options require that all of
> the interoperable implementations support option sets that are a union or an
> intersection of working options of the common group (forgive my ignorance of
> the rules; RFC 2026 didn't help me much here)?  I am hoping that a simple
> intersecting set is the answer.  Otherwise RFC 2833 will disintegrate.  Thoughts?
> .

Any one feature (or event, for 2833) will need to be supported by two
independent implementations. It is not necessary that two
implementations support all events. In the case of simple constants such
as the events, it's also not exactly clear that it makes sense to remove
one event that currently may not have two implementations. After all,
somebody could just go right back to IANA and register that number.
(With IANA, there is no two-implementations requirement.)

Thus, I'd suggest that we find a volunteer with active implementation
interest in RFC 2833 to gather implementation reports on features and
events. This list (and the list of implementations) does not have to be
public, in case that's a concern.

After that's done, we can take a look at the problem cases of events
that have zero or one implementations and see whether this is a matter
of "useful, but just not there yet" or "not that useful after all".

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



From rem-conf Wed Sep 06 10:27:39 2000 
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To: coldrenr@agcs.com
cc: Henning Schulzrinne <schulzrinne@cs.columbia.edu>,
        Flemming Andreasen <fandreas@cisco.com>, rem-conf <rem-conf@es.net>
Subject: Re: Optional DTMF support in RFC 2833 
In-Reply-To: Your message of "Tue, 05 Sep 2000 19:37:34 PDT."
             <39B5ADED.7B2DC541@agcs.com> 
Date: Wed, 06 Sep 2000 13:24:46 -0400
From: Colin Perkins <csp@east.isi.edu>
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--> Rex Coldren writes:
>Flemming, Henning,
>
>I agree with both of you.  DTMF support should not be required.  Realistically,
>there will likely be zero implementations that will implement all of the possible
>combinations of events and tones.  So how do we deal with trying to push RFC
>2833 toward Draft Standard when none of the implementations support the full
>set of capabilities?  Do the IETF rules for deprecating options require that all of
>the interoperable implementations support option sets that are a union or an
>intersection of working options of the common group (forgive my ignorance of
>the rules; RFC 2026 didn't help me much here)?  I am hoping that a simple
>intersecting set is the answer.  Otherwise RFC 2833 will disintegrate.  Thoughts?

The rules require that at least two independent interoperable
implementations exist for each feature. There is no requirement
that a particular implementation support every feature, so long
as each feature is supported by some implementations.

Colin



From rem-conf Wed Sep 06 10:29:05 2000 
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To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
cc: coldrenr@agcs.com, Flemming Andreasen <fandreas@cisco.com>,
        rem-conf <rem-conf@es.net>
Subject: Re: Optional DTMF support in RFC 2833 
In-Reply-To: Your message of "Wed, 06 Sep 2000 12:58:32 EDT."
             <39B677B8.80AE4F9B@cs.columbia.edu> 
Date: Wed, 06 Sep 2000 13:26:55 -0400
From: Colin Perkins <csp@east.isi.edu>
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--> Henning Schulzrinne writes:
>Rex Coldren wrote:
>> 
>> Flemming, Henning,
>> 
>> I agree with both of you.  DTMF support should not be required.  Realistically,
>> there will likely be zero implementations that will implement all of the possible
>> combinations of events and tones.  So how do we deal with trying to push RFC
>> 2833 toward Draft Standard when none of the implementations support the full
>> set of capabilities?  Do the IETF rules for deprecating options require that all of
>> the interoperable implementations support option sets that are a union or an
>> intersection of working options of the common group (forgive my ignorance of
>> the rules; RFC 2026 didn't help me much here)?  I am hoping that a simple
>> intersecting set is the answer.  Otherwise RFC 2833 will disintegrate.  Thoughts?
>> .
>
>Any one feature (or event, for 2833) will need to be supported by two
>independent implementations. It is not necessary that two
>implementations support all events. In the case of simple constants such
>as the events, it's also not exactly clear that it makes sense to remove
>one event that currently may not have two implementations. After all,
>somebody could just go right back to IANA and register that number.
>(With IANA, there is no two-implementations requirement.)
>
>Thus, I'd suggest that we find a volunteer with active implementation
>interest in RFC 2833 to gather implementation reports on features and
>events. This list (and the list of implementations) does not have to be
>public, in case that's a concern.
>
>After that's done, we can take a look at the problem cases of events
>that have zero or one implementations and see whether this is a matter
>of "useful, but just not there yet" or "not that useful after all".

Although note that this can't go to draft standard until RTP goes to draft,
which means we need to complete the RTP interop first.....

Colin



From rem-conf Wed Sep 06 20:13:26 2000 
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To: mobile-systems@cs.ucl.ac.uk, rem-conf@es.net, havinga@cs.utwente.nl, 
    oulusoy@cs.bilkent.edu.tr, helal@cise.ufl.edu, anan@master.cpe.ku.ac.th, 
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From: Farez <f.abdulrahman@cs.ucl.ac.uk>
Subject: AMOC 2000 Programme and Call for Participation
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Please accept our apologies if you have received multiple copies of this.

Farez

------------------------------------------------------------
              Preliminary Programme & Call for Participation
                        
                               AMOC 2000
         FIRST ASIAN INTERNATIONAL MOBILE COMPUTING CONFERENCE
            31 October - 3 November, 2000. Penang, Malaysia
                
                Organised by University Malaya, Malaysia
                   In cooperation with ACM SIGMOBILE
              With support from MRCB Multimedia, Malaysia

                    http://www.fsktm.um.edu.my/amoc
            http://www.cs.ucl.ac.uk/staff/F.AbdulRahman/amoc


We are pleased to invite you to AMOC 2000, which will be held in Penang,
Malaysia. Below is the preliminary programme. Registration forms and
instructions are attached below the programme.

Regards,

Alfarez Abdul Rahman <farez@acm.org>
Dr. Mazliza Othman <mazliza@fsktm.um.edu.my>
AMOC Co-chairs

----------------------------------------------------------
Preliminary Programme (please see website for more detail)
----------------------------------------------------------

* 31 October, 2000 - Tutorials only

0900-1215: Half-Day Tutorial 1: 

           Mobile Ad-Hoc Wireless Networks

           Dr. Somprakash Bandyopadhyay
           PriceWaterhouseCoopers, India
           Dr. Krishna Paul, 
           Cognizant Technology Solutions, India

1215-1400: Lunch

1400-1715: Half-Day Tutorial 2: 

           Java based Mobile Computing -- 
           JINI, Mobile Objects and Agents 

           Dr. Omer F. Rana, 
           University of Wales, Cardiff, Wales

................................................

0900-1715: Full Day Tutorial (to be confirmed)

           Client/Server Computing in Wireless and Mobile Environments

           Prof. Abdelsalam Helal,
           University of Florida, USA


==========================================================

* 1 November, 2000 

0900-1215: Registration
           
           Welcome Reception
           
           Guest Speakers
           
1215-1400: Lunch

1400-1530: Session 1: Medium Access

           Energy-efficient TDMA Medium Access Control Protocol Scheduling
           Paul J.M. Havinga, Gerard J.M. Smith	
           Dept. of Computer Science, University of Twente, NETHERLANDS

           Fault Detection and Recovery Algorithm for a Centralised
MediaArbitration Scheme in Wireless Networks
           Anan Phonphoem, Aura Ganz	
           University of Massachusetts, USA

           Adaptive Slot Assignment Algorithm for Improving the MAC Frame
Access Delay of Wireless ATM Channel
           Mohamed Hadi Habaebi, Borhanuddin Mohd. Ali	
           Unviversiti Putra Malaysia, MALAYSIA

1530-1545: Break

1545-1715: Session 2: Asymmetric Channels

           Incorporation of QoS and Mitigated TCP/IP over Satellite Links
           Swee Keong Joo, Tat Chee Wan	
           School of Computer Science, U. of Science MALAYSIA

           Traffic Allocation in LEO Satellite Communication
           Graham McMahon, Stephen Sugden, Reza Septiawan	
           School of IT Bond University, AUSTRALIA

           Mild Aggression: A New Approach for Improving TCP Performance in
Asymmetric Networks
           Vijay T. Raisinghani, Abishek Patil, Sridhar Iyer	
           KR School of Information Technology, Indian Institute of
Technology, INDIA

==========================================================

* 2 November, 2000

0900-1030: Session 3: Broadband and Multimedia

           Design and Characteristics of a Broadband Subcarrier Multiplexed
2.375 GHz Wireless Link
           Randy C. Barstan, Rolando C. Guevarra, Mia G. Antonio, Delfin J
Sabido IX	
           Advanced Science & Technology Institute, Unversity of the
Philippines Technology Park, PHILIPPINES

           An Efficient Scalable Image Coding for Multimedia Applications
           Mohd Al-zoubi, R. K. Subramanian	
           School of Computer Sciences, USM, MALAYSIA

           A Framework for Live Video Delivery over GPRS Networks
           Bing Du, Anthony Maeder, Miles Moody	
           Queensland University of Technology; Cooperative Research Center
for Satellite Systems, AUSTRALIA.

1030-1045: Break

1045-1215: Session 4: Agent Technology

           Towards Mission Tamper-Resistant Mobile Agent-Based Electronic
Commerce
           Xunhua Wang, Yih Huang, David Rine	
           Dept. of Computer Science, George Mason Univesity, USA

           Requirements for Disconnected Mobile Agents within Cellular
Mobile Telecommunication Systems
           Carsten Pils, Jens Hartmann 	
           Informatik 4 (Communications Systems), Ericsson Eurolab
Deutschland GmbH, GERMANY

           Software Agent Framework for Nomadic Computing
           Heikki Helin, Heimo Laamanen, Mikko Laukkanen	
           Cellular Systems Development, FINLAND

1215-1400: Lunch

1400-1530: Open Forum Session

1530-1545: Break

1545-1700: Open Forum Session (continued)

PM: Banquet

==========================================================

* 3 November, 2000

0900-1030: Session 5: Data Acccess, Display and Management

           A Three-tier Architecture for Ubiquitous Data Access
           Abdelsalam Helal, Joachim Hammer, Jinsuo Zhang, Abhinav Khushraj	
           University of Florida, USA

           A Remote WWW Display Environment using a Cellular Phone WWW Service
           Toshiaki Uemukai, Hiroaki Hagino, Takahiro Hara, Masahiko
Tsukamoto, Shojiro Nishio	
           Dept. of Information Systems Engineering; Graduate School of
Engineering, Osaka University, JAPAN

           Increasing Transaction Autonomy Using Adaptive Broadcasting
Schemes in Mobile DBMSs
           IlYoung Chung, Chong-Sun Hwang	
           Dept. of Computer Science & Engineering, Korea University, KOREA

1030-1045: Break

1045-1215: Session 6: User Mobility

           Evaluation of Methods for Transmission of Location-Dependent
Continuous Query Results
           Gokmen Gok, Ozgur Ulusoy	
           Bilkent University, TURKEY

           User Mobility Simulation by Travelling Demand Model in Mobile
Environments
           Joon-Min Gil, Chan Yeol Park, Youn-Hee Han, Chong-Sun Hwang	
           Dept. of Computer Science & Engineering, Korea University, KOREA

           Mr. G: Mobile Routing Method for Group Migration
           Hiroaki Haginao, Takahiro Hara, Masahiko Tsukamoto, Shojiro Nishio	
           Dept. of Information Systems Engineering; Graduate School of
Engineering, Osaka University, JAPAN

1215: Conference Close


------------------------------------------------------------------

AMOC 2000
1st Asian International Mobile Computing Conference
31 October - 3 November, 2000 
Penang, Malaysia

* Registration deadline: 30 September, 2000

============================
Conference Registration Form
============================

Your details
------------

Title: _______ Surname: ____________________________________

Other names: _______________________________________________

Designation: _______________________________________________

Company Name/ Institution: _________________________________

Address: ___________________________________________________

City: ____________________ Postcode: _______________________

State: ___________________ Country: ________________________

Office Telephone: __________________ Fax: __________________

E-mail address: ____________________________________________

Would you like to include your name in the list 
of attendees?                                     _ YES _ NO

Please specify if you have any special needs:

____________________________________________________________

Registration Fee (please tick as appropriate)
---------------------------------------------
* You can pay in Malaysian Ringgit (RM) or US Dollars (USD)

Students:    _ RM 400 or USD 100

Others:      _ RM 600 or USD 150

Note for student registrants:
In order to qualify for the discounted conference fee, please 
ask your Head of Department to certify that you are a student
here.

Head of Department's name: __________________________________


Head of Department's signature: _____________________________


Lunch is provided and a banquet will be held on 2 November.
(You do not need to pay extra for the banquet and lunches. 
The banquet may be cancelled if there is less than a minimum
number of attendees agree to turn up.)

Will you be attending the banquet?                _ YES _ NO

Are you a vegetarian?                             _ YES _ NO



Your signature: _____________________________ Date: ________


Payment Method
--------------

Please make your crossed cheque / bank draft payable to
'BENDAHARI UNIVERSITI MALAYA' 
and mail it with this form to:

Asian International Mobile Computing Conference
Faculty of Computer Science & IT
University of  Malaya
50603 Kuala Lumpur                      Tel: +60-3-7969 6383
Malaysia                                Fax: +60-3-757 9249  
(Attn. Mr Mustaffa Kamal)

http://www.fsktm.um.edu.my/amoc
http://www.cs.ucl.ac.uk/staff/F.AbdulRahman/amoc
Enquiries: amoc-registration@fsktm.um.edu.my

------------------------------------------------------------------

AMOC 2000
1st Asian International Mobile Computing Conference
31 October - 3 November, 2000 
Penang, Malaysia

* Registration deadline: 30 September, 2000

==========================
Tutorial Registration Form
==========================

Your details
------------

Title: _______ Surname: ____________________________________

Other names: _______________________________________________

Designation: _______________________________________________

Company Name/ Institution: _________________________________

Address: ___________________________________________________

City: ____________________ Postcode: _______________________

State: ___________________ Country: ________________________

Office Telephone: __________________ Fax: __________________

E-mail address: ____________________________________________


Registration Fee (please tick as appropriate)
------------------------------------------------
* Price includes lunch and tutorial notes.
* You can pay in Malaysian Ringgit (RM) or US Dollars (USD)

1. Client/Server Computing in Mobile and   
   Wireless Environments (Full Day)                  _ RM 250 or USD 65

2. Mobile Ad Hoc Wireless Networks (Half Day - AM)   _ RM 200 or USD 50

3. Java Based Mobile Computing -- 
   JINI, Mobile Objects and Agents (Half Day - PM)   _ RM 200 or USD 50
   

Are you a vegetarian?                             _ YES _ NO

Would you like to include your name in the list 
of attendees?                                     _ YES _ NO

Please specify if you have any special needs:

____________________________________________________________



Your signature: _____________________________ Date: ________


Payment Method
--------------

Please make your crossed cheque / bank draft payable to
'BENDAHARI UNIVERSITI MALAYA' 
and mail it with this form to:

Asian International Mobile Computing Conference
Faculty of Computer Science & IT
University of  Malaya
50603 Kuala Lumpur                      Tel: +60-3-7969 6383
Malaysia                                Fax: +60-3-757 9249  
(Attn. Mr Mustaffa Kamal)

http://www.fsktm.um.edu.my/amoc
http://www.cs.ucl.ac.uk/staff/F.AbdulRahman/amoc
Enquiries: amoc-registration@fsktm.um.edu.my



------------------------------------------------------------------

AMOC 2000
1st Asian International Mobile Computing Conference
31 October - 3 November, 2000 
Penang, Malaysia

* Booking deadline: 1 October, 2000

=======================================
Room Booking at Penang Parkroyal Resort
=======================================

Your details
------------

Title: _______ Surname: ____________________________________

Other names: _______________________________________________

Designation: _______________________________________________

Company Name/ Institution: _________________________________

Address: ___________________________________________________

City: ____________________ Postcode: _______________________

State: ___________________ Country: ________________________

Office Telephone: __________________ Fax: __________________

E-mail address: ____________________________________________


Booking Details
---------------

Booking is from: _____________ to ______________ (__ nights)

Arrival date: ______________________________________________

Number of rooms: ___________________________________________

_ Single  _ Double

A confirmed booking must be guaranteed either with credit card 
details or one night pre-payment.


Payment Method
--------------

The room rate is RM200 per night. If paying by a cheque/bank 
draft, pelase make it payable to PENANG PARKROYAL RESORT.

_ Cheque/bank draft enclosed

Credit card

_ Visa   _ MasterCard  _ American Express

Credit Card #: ____________________________________________

Expiry date: ______________________________________________


Signature: ________________________________________________

Date: _____________________________________________________

-----------------------------------------------------------
Please mail/fax this form to:

Asian International Mobile Computing Conference
Faculty of Computer Science & IT
University of  Malaya
50603 Kuala Lumpur                      Tel: +60-3-7969 6383
Malaysia                                Fax: +60-3-757 9249  
(Attn. Mr Mustaffa Kamal)

http://www.fsktm.um.edu.my/amoc
http://www.cs.ucl.ac.uk/staff/F.AbdulRahman/amoc
Enquiries: amoc-registration@fsktm.um.edu.my


-------------------------------------------------------------

Hotel booking info (not part of form above):

AMOC 2000 will be held at the Penang Parkroyal Resort in Penang Malaysia.
AMOC participants will have the special rate of RM200 per night. All
cheques / bank draft must be made payable to PENANG PARKROYAL RESORT. Hotel
bookings for Parkroyal Resort must be made through University Malaya
(details are at the bottom of the form). Please do not book through
Parkroyal itself or a travel agent as this may not entitle you to the
special conference rate. 

For more information about the hotel, please see their website
(http://www.sphc.com.au/), or contact them directly using the contact
details below. Please remember to book through the AMOC organisers to get
the special conference rate.

Penang Parkroyal Resort
Batu Ferringhi Beach, Batu Ferringhi 
Penang 11100, Malaysia. 

Phone: (604) 8811133 
Fax: (604) 8812233 
Email: inqppr@sphc.com.my 

There are alternative hotels and other types of accommodation along the
Baru Ferringhi beach where AMOC will be located. For a list of hotels you
can try the HotelStreet website
(http://www.hotelstreet.com.my/hotellist.asp?sCity=Penang) (they also
support online reservations) or consult your travel agent. When using the
website, please search for hotels in Penang and choose those that are
located in Batu Ferringhi or Tanjung Bungah areas. Be sure to enquire their
distance from AMOC's location as some of them may not be in convenient
walking distance.

Participants are adviced to book their accommodations early as AMOC will
coincide with the local school holiday season. This means that hotels are
likely to be fully booked.



+44 7939 610241
www.cs.ucl.ac.uk/staff/F.AbdulRahman
Department of Computer Science, University College London



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To: "Koji Imura" <imura@telecom.mci.mei.co.jp>, "IETF AVT" <rem-conf@es.net>,
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        "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
References: <200008251549.QAA01356@purple.cs.ucl.ac.uk> <00f001c017a9$28852460$a3d2b785@telecom.mci.mei.co.jp>
Subject: Re: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt
Date: Thu, 7 Sep 2000 14:40:29 +0900
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Dear Imura-san,

Thank you very much for the comment.

> I reviewed the draft and had a question.
> In the section 3.1, there is a description about the marker bit and
> time stamp for the MPEG-4 visual.
> According to this description, it is possible to include some VOPs
> in one RTP packet.
> In such a case, the only time stamp for first VOP is included in the
> RTP header, but there is no time information for rest of VOPs in the
> same RTP packet.
> The temporal interval between VOPs is variable in MPEG-4 visual
> So it is difficult to estimate the time stamp of the other VOPs.
>
> If there is a system that presentation/composition time is decided
> by the RTP time stamp, how do the system estimate the time stamp
> of rest of the VOPs?
>
Since MPEG-4 Video stream contains timing information in the VOP header
fields (modulo_time_base and vop_time_increment), the timestamp can be
derived from these fields inside the RTP payload.

It would be better to add a description to clarify this. How about adding
the following description in the timestamp definition in section 3.1?

- When multiple VOPs are carried in the same RTP packet, the timestamp
indicates the earliest of the composition time within the VOPs carried in
the RTP packet. Timestamp information of the rest of VOPs are derived from
the timestamp fields in the VOP header (modulo_time_base and
vop_time_increment).?

Best regards,
Yoshihiro

----
        Yoshihiro Kikuchi

E-mail: kiku@eel.rdc.toshiba.co.jp
Corporate Research and Development Center
TOSHIBA Corporation
TEL: +81 +44 549 2288    FAX: +81 +44 520 1267






From rem-conf Thu Sep 07 00:47:37 2000 
From rem-conf-request@es.net Thu Sep 07 00:47:36 2000
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From: "Guler, Amos" <Amos_Guler@icomverse.com>
To: "'rem-conf@es.net'" <rem-conf@es.net>
Subject: Question: packing several frames in a single RTP packet
Date: Thu, 7 Sep 2000 10:42:41 +0300 
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Hello all,

(I hope I am sending this to the correct mailing list. If not, please accept
my apology and advise me of the correct one)

In the new draft "RTP Profile for Audio and Video Conferences with Minimal
Control"
(http://www.ietf.org/internet-drafts/draft-ietf-avt-profile-new-09.txt) that
is supposed to replace RFC-1890, it says (in section 4.4, Guidelines for
Frame-Based Audio Encodings):

"The receiver can tell the number of frames contained in an RTP packet, if
all the frames have the same length, by dividing the RTP payload length by
the audio frame size which is defined as part of the encoding. This does not
work when carrying frames of different sizes unless the frame sizes are
relatively prime."

My question concerns the last statement: suppose, for example, that the
frame sizes are 5 and 7 octets, which are relatively prime, and the receiver
gets a 35-octet packet. How can he/she tell whether the packet contains five
7-octet frames or seven 5-octet frames?


Amos Guler

==================== 
Amos Guler 
Comverse Network Systems Ltd.
amos_guler@icomverse.com 
http://www.comverse.com 



From rem-conf Thu Sep 07 01:39:28 2000 
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Date: Thu, 07 Sep 2000 10:20:19 +0200
From: Franceschini Guido <Guido.Franceschini@CSELT.IT>
Subject: RE: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt
To: Koji Imura <imura@telecom.mci.mei.co.jp>, IETF AVT <rem-conf@es.net>,
 Colin Perkins <colin@isi.edu>,
 'Yoshihiro Kikuchi' <kiku@eel.rdc.toshiba.co.jp>,
 Varesio Andrea <Andrea.Varesio@CSELT.IT>
Cc: 4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>
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Dear Kikuchi-Sun,

> ----------
> From: 	Yoshihiro Kikuchi[SMTP:kiku@eel.rdc.toshiba.co.jp]
> Sent: 	Thursday, September 07, 2000 7:40 AM
> To: 	Koji Imura; IETF AVT; Colin Perkins
> Cc: 	4on2andIP-sys; Yoshihiro Kikuchi
> Subject: 	Re: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt
> 
> Dear Imura-san,
> 
> Thank you very much for the comment.
> 
> > I reviewed the draft and had a question.
> > In the section 3.1, there is a description about the marker bit and
> > time stamp for the MPEG-4 visual.
> > According to this description, it is possible to include some VOPs
> > in one RTP packet.
> > In such a case, the only time stamp for first VOP is included in the
> > RTP header, but there is no time information for rest of VOPs in the
> > same RTP packet.
> > The temporal interval between VOPs is variable in MPEG-4 visual
> > So it is difficult to estimate the time stamp of the other VOPs.
> >
> > If there is a system that presentation/composition time is decided
> > by the RTP time stamp, how do the system estimate the time stamp
> > of rest of the VOPs?
> >
> Since MPEG-4 Video stream contains timing information in the VOP header
> fields (modulo_time_base and vop_time_increment), the timestamp can be
> derived from these fields inside the RTP payload.
> 
> It would be better to add a description to clarify this. How about adding
> the following description in the timestamp definition in section 3.1?
> 
> - When multiple VOPs are carried in the same RTP packet, the timestamp
> indicates the earliest of the composition time within the VOPs carried in
> the RTP packet. Timestamp information of the rest of VOPs are derived from
> the timestamp fields in the VOP header (modulo_time_base and
> vop_time_increment).?
> 
Are you sure you want to possibly indicate the CTS of the second VOP in a
packet ?
This seems to me quite difficult to manage
I would better do either of the following:
- always indicate the CTS of the first VOP in the packet, even if the second
VOP will be presented first
- disallow concatenation of VOPs whose decoding order differs from the
composition order

I am also doubtful about allowing variable frame rate videos to take
advantage of such concatenation of VOPs in the same RTP packet.
IMHO it would be safer to add some more constraints in the specification
(i.e. prohibit VOP concatenation in case of variable frame rates), instead
of adding the complexity of extracting TS information through cooperation of
different layers.
In other words: VOP concatenation has pros and cons.
Pros: optimizes bandwidth consuption
Cons: deteriorates packet loss resilience; increases implementation
complexity.
I therefore deduce that VOP concatenation is nice but not that wonderful,
and I would suggest to allow VOP concatenation only when this is trivial and
does not require significant complexity in the receivers implementations.

Bye
Guido


> Best regards,
> Yoshihiro
> 
> ----
>         Yoshihiro Kikuchi
> 
> E-mail: kiku@eel.rdc.toshiba.co.jp
> Corporate Research and Development Center
> TOSHIBA Corporation
> TEL: +81 +44 549 2288    FAX: +81 +44 520 1267
> 
> 
> 



From rem-conf Thu Sep 07 02:11:06 2000 
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From: "Y. Matsui" <matsui@drl.mei.co.jp>
To: "Franceschini Guido" <Guido.Franceschini@cselt.it>,
        "Koji Imura" <imura@telecom.mci.mei.co.jp>,
        "IETF AVT" <rem-conf@es.net>, "Colin Perkins" <colin@isi.edu>,
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Cc: "4on2andIP-sys" <4on2andIP-sys@advent.ee.columbia.edu>
References: <85077463E51D6A498C453073E1677940034E9C@exc2k02.cselt.it>
Subject: Re: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt
Date: Thu, 7 Sep 2000 18:06:39 +0900
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Dear Guido, Kikuchi-san and all,

I agree on Guido's opinion that concatenation of VOPs is
allowed only when constant frame rate, that is defined when
the fixed_vop_rate field of the VOL header is set to 1.

Best regards,
Yoshinori Matsui
Matsushita Electric Industrial Co., LTD.

----- Original Message -----
From: Franceschini Guido <Guido.Franceschini@cselt.it>
To: Koji Imura <imura@telecom.mci.mei.co.jp>; IETF AVT <rem-conf@es.net>; Colin Perkins <colin@isi.edu>; 'Yoshihiro Kikuchi'
<kiku@eel.rdc.toshiba.co.jp>; Varesio Andrea <Andrea.Varesio@cselt.it>
Cc: 4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>
Sent: Thursday, September 07, 2000 5:20 PM
Subject: RE: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt


> Dear Kikuchi-Sun,
>
> > ----------
> > From: Yoshihiro Kikuchi[SMTP:kiku@eel.rdc.toshiba.co.jp]
> > Sent: Thursday, September 07, 2000 7:40 AM
> > To: Koji Imura; IETF AVT; Colin Perkins
> > Cc: 4on2andIP-sys; Yoshihiro Kikuchi
> > Subject: Re: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt
> >
> > Dear Imura-san,
> >
> > Thank you very much for the comment.
> >
> > > I reviewed the draft and had a question.
> > > In the section 3.1, there is a description about the marker bit and
> > > time stamp for the MPEG-4 visual.
> > > According to this description, it is possible to include some VOPs
> > > in one RTP packet.
> > > In such a case, the only time stamp for first VOP is included in the
> > > RTP header, but there is no time information for rest of VOPs in the
> > > same RTP packet.
> > > The temporal interval between VOPs is variable in MPEG-4 visual
> > > So it is difficult to estimate the time stamp of the other VOPs.
> > >
> > > If there is a system that presentation/composition time is decided
> > > by the RTP time stamp, how do the system estimate the time stamp
> > > of rest of the VOPs?
> > >
> > Since MPEG-4 Video stream contains timing information in the VOP header
> > fields (modulo_time_base and vop_time_increment), the timestamp can be
> > derived from these fields inside the RTP payload.
> >
> > It would be better to add a description to clarify this. How about adding
> > the following description in the timestamp definition in section 3.1?
> >
> > - When multiple VOPs are carried in the same RTP packet, the timestamp
> > indicates the earliest of the composition time within the VOPs carried in
> > the RTP packet. Timestamp information of the rest of VOPs are derived from
> > the timestamp fields in the VOP header (modulo_time_base and
> > vop_time_increment).?
> >
> Are you sure you want to possibly indicate the CTS of the second VOP in a
> packet ?
> This seems to me quite difficult to manage
> I would better do either of the following:
> - always indicate the CTS of the first VOP in the packet, even if the second
> VOP will be presented first
> - disallow concatenation of VOPs whose decoding order differs from the
> composition order
>
> I am also doubtful about allowing variable frame rate videos to take
> advantage of such concatenation of VOPs in the same RTP packet.
> IMHO it would be safer to add some more constraints in the specification
> (i.e. prohibit VOP concatenation in case of variable frame rates), instead
> of adding the complexity of extracting TS information through cooperation of
> different layers.
> In other words: VOP concatenation has pros and cons.
> Pros: optimizes bandwidth consuption
> Cons: deteriorates packet loss resilience; increases implementation
> complexity.
> I therefore deduce that VOP concatenation is nice but not that wonderful,
> and I would suggest to allow VOP concatenation only when this is trivial and
> does not require significant complexity in the receivers implementations.
>
> Bye
> Guido
>
>
> > Best regards,
> > Yoshihiro
> >
> > ----
> >         Yoshihiro Kikuchi
> >
> > E-mail: kiku@eel.rdc.toshiba.co.jp
> > Corporate Research and Development Center
> > TOSHIBA Corporation
> > TEL: +81 +44 549 2288    FAX: +81 +44 520 1267
> >
> >
> >
>




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Subject: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call
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=20
International SIP 2001 (second edition)
Implementing H.323
MGC 2000 (second edition)
=20
Three conferences organised by Upper Side gathering the most eminent =
specialists in VoIP technologies.
=20
Get more details at:
http://www.upperside.fr/index.html



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<DIV><FONT face=3DArial size=3D2><FONT face=3DArial size=3D2>&nbsp;
<DIV><FONT color=3D#000000 size=3D2>International <STRONG>SIP =
</STRONG>2001 (second=20
edition)</FONT></DIV>
<DIV><FONT color=3D#000000 size=3D2>Implementing =
<STRONG>H.323</FONT></STRONG></DIV>
<DIV><FONT color=3D#000000 size=3D2><STRONG>MGC</STRONG> 2000 (second=20
edition)</FONT></DIV>
<DIV><FONT color=3D#000000 size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#000000 size=3D2>Three conferences organised by Upper =
Side=20
gathering the most eminent specialists in VoIP =
technologies.</FONT></DIV>
<DIV><FONT color=3D#000000 size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>Get more details at:</FONT></DIV>
<DIV></FONT><A=20
href=3D"http://www.upperside.fr/index.html">http://www.upperside.fr/index=
.html</A></FONT></DIV></DIV>
<DIV>&nbsp;</DIV>
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From rem-conf Thu Sep 07 08:20:43 2000 
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From rem-conf Thu Sep 07 09:35:54 2000 
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Date: Thu, 07 Sep 2000 09:21:20 -0700
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From: Florissa Colina <florissa@bmrc.berkeley.edu>
Subject: 9/13 Making Interactive Workspaces Real -- Armando Fox,
  Computer Science Department, Stanford University
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Berkeley Multimedia, Graphics and Interfaces Seminar

Making Interactive Workspaces Real

Wednesday September 13, 2000,=20
1:10-2:30 p.m. PDT
Fujitsu Seminar Room (405 Soda Hall)=20

Armando Fox=20
Computer Science Department, Stanford University

The Stanford Interactive Workspaces project is developing the computing
foundations for bringing reality to ubiquitous computing. A diverse
collection of faculty and students from the areas of graphics,
human-computer interaction (HCI),networking,ubiquitous computing,and
databases, we are looking at new types of human/machine interaction
including multimodal input, heterogeneous device integration from
wall-sized displays to handheld information appliances, and the "invisibl=
e"
software infrastructure required to support them. In the same way that
today=92s standard operating systems make it feasible to write
single-workstation software that uses multiple devices and networked
resources,we are constructing a higher level operating system for the wor=
ld
of ubiquitous computing. We combine research on infrastructure (ways of
flexibly configuring and connecting devices, processes,and communication
links)with research on HCI (ways of interacting with heterogeneous changi=
ng
collections of devices with multiple modalities). I will describe our
progress on the software infrastructure, including applications running i=
n
our deployed iRoom that do in fact integrate a variety of devices.=20

---------------

The seminar is broadcast live on the Internet Mbone and as a Real Network=
s
G2 broadcast.

You can connect to the live RealNetworks broadcast at:
http://bmrc.berkeley.edu/bibs/cs298-5

You can also directly put in this url into the Real Player:=20
rtsp://media2.bmrc.berkeley.edu/encoder/cs298-5.rm

To do so you will need to have the "Real Player G2" installed, which is
available from:
http://www.real.com/products/player/downloadrealplayer.html

To tune into the Internet MBone broadcast, look for the announcement in
your MBone session directory program ('sdr').  If you are not receiving t=
he
announcement you may be able to receive the session by manually configuri=
ng
the client programs ('vic', and 'vat') with the session addresses:

low bit rate
	video: 233.0.25.1/22334
	audio: 233.0.25.2/22446

medium bit rate
	video: 233.0.25.129/22334
	audio: 233.0.25.2/22446

You can get further information about this seminar, and access to replays
of previous seminars at the MIG Seminar web page:

http://bmrc.berkeley.edu/298/index.html

Sponsored by the Berkeley Multimedia Research Center=20




From rem-conf Thu Sep 07 18:29:16 2000 
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To: Andy Valencia <vandys@zendo.com>, "W. Mark Townsley" <townsley@cisco.com>
From: Tmima Koren <tmima@cisco.com>
Subject: 0 byte L2TPHC header and implied PPPMUX
Cc: brucet@cisco.com, dwing@cisco.com,
        Stephen Casner <casner@packetdesign.com>, l2tp@diameter.org,
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I suggest a few minor changes to the L2TPHC draft draft-ietf-l2tpext-l2tphc-02.txt
to improve bandwidth utilization in special cases.


When the L2TPHC header is 1 byte long, actually only 1 bit is significant: the P (Priority) bit.

If all the packets in the tunnel will have the same priority, the L2TPHC byte is redundant.

We could add to the L2TPHC AVP one more byte (byte 9)
We really need only 2 flag bits:

flag 1:  value 1 means: L2TPHC byte will be omitted
            value 0 means: L2TPHC byte is there, get P (Priority) from the L2TPHC byte

flag 2:  value 1 means all packets have implied P=1
            value 0 means all packets have implied P=0

consult flag 2 only when flag 1 is 1


If the packet in the L2TPHC tunnel is always PPPMUX, there is another useful flag:

flag 3: value 1 means 'PPPMUX only'
           value 0 means: any PPP packet


When flags 1 and 3 are on (=1), both L2TPHC header and PPPMUX protocol id can be omitted 
PPP packets that have a different priority or that are not PPPMUX can still be sent in the tunnel using the UDP/L2TP format, same as the L2TP control packets

TCRTP (Tunneled Compressed multiplexed RTP) can benefit from these changes: the tunneling overhead will be reduced by 2 bytes.

Tmima




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References: <007c01c018ad$cfc532b0$013de8c3@Young>
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From rem-conf Thu Sep 07 22:19:43 2000 
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From rem-conf Thu Sep 07 23:07:22 2000 
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        "'Yoshihiro Kikuchi'" <kiku@eel.rdc.toshiba.co.jp>,
        "Varesio Andrea" <Andrea.Varesio@cselt.it>
Cc: "4on2andIP-sys" <4on2andIP-sys@advent.ee.columbia.edu>
References: <85077463E51D6A498C453073E1677940034E9C@exc2k02.cselt.it>
Subject: Re: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt
Date: Fri, 8 Sep 2000 15:00:55 +0900
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Dear Guido, Kikuchi-san, all,

> > - When multiple VOPs are carried in the same RTP packet, the timestamp
> > indicates the earliest of the composition time within the VOPs carried in
> > the RTP packet. Timestamp information of the rest of VOPs are derived from
> > the timestamp fields in the VOP header (modulo_time_base and
> > vop_time_increment).?
> > 
> Are you sure you want to possibly indicate the CTS of the second VOP in a
> packet ?
> This seems to me quite difficult to manage
> I would better do either of the following:
> - always indicate the CTS of the first VOP in the packet, even if the second
> VOP will be presented first
> - disallow concatenation of VOPs whose decoding order differs from the
> composition order

I would prefer the latter one.  I think it should be added to the current
draft-ietf-avt-rtp-mpeg4-es-03 document.

> I am also doubtful about allowing variable frame rate videos to take
> advantage of such concatenation of VOPs in the same RTP packet.
> IMHO it would be safer to add some more constraints in the specification
> (i.e. prohibit VOP concatenation in case of variable frame rates), instead
> of adding the complexity of extracting TS information through cooperation of
> different layers.
> In other words: VOP concatenation has pros and cons.
> Pros: optimizes bandwidth consuption
> Cons: deteriorates packet loss resilience; increases implementation
> complexity.
> I therefore deduce that VOP concatenation is nice but not that wonderful,
> and I would suggest to allow VOP concatenation only when this is trivial and
> does not require significant complexity in the receivers implementations.

How about the following change ?

[Original sentense in 1.1 MPEG-4 Visual RTP Payload Format]
The fragmentation rule recommends not to map more than one VOP in an
RTP packet so that RTP timestamp uniquely indicates the VOP time
framing. On the other hand, MPEG-4 video may generate VOPs of very
small size, in cases with a not coded VOP containing only VOP header or
an arbitrary shaped VOP with a small number. To reduce the overhead for
such cases, the fragmentation rule permits concatenating multiple VOPs in
an RTP packet when the VOP size is small.

[My proposed modification]
MPEG-4 video may generate VOPs of very
small size, in cases with a not coded VOP containing only VOP header or
an arbitrary shaped VOP with a small number. To reduce the overhead for
such cases, the fragmentation rule permits concatenating multiple VOPs in
an RTP packet when the VOP size is small.  However, for the variable
frame rate video, the fragmentation rule recommends not to map more
than one VOP in an RTP packet so that RTP timestamp uniquely indicates
the VOP time framing.

Is this sentense enough for prohibiting the mapping of the multiple
VOPs in a packet when variable frame rate video ?
If there is no any objections on above sentense, I would like to propose
to change the current draft as this.

Best regards,
Yoshinori Matsui
Matsushita Electric Industrial Co., LTD.





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References: <007c01c018ad$cfc532b0$013de8c3@Young> <00c901c0193f$e578b0e0$dcfa0985@katto.comm.waseda.ac.jp>
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call
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From rem-conf Thu Sep 07 23:22:03 2000 
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From: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
To: "Franceschini Guido" <Guido.Franceschini@CSELT.IT>,
        "Koji Imura" <imura@telecom.mci.mei.co.jp>,
        "IETF AVT" <rem-conf@es.net>, "Colin Perkins" <colin@isi.edu>,
        "Varesio Andrea" <Andrea.Varesio@CSELT.IT>
Cc: "4on2andIP-sys" <4on2andIP-sys@advent.ee.columbia.edu>,
        "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
References: <85077463E51D6A498C453073E1677940034E9C@exc2k02.cselt.it>
Subject: Re: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt
Date: Fri, 8 Sep 2000 15:17:57 +0900
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Dear Guido, Matsui-san, Imura-san, all,

> > - When multiple VOPs are carried in the same RTP packet, the timestamp
> > indicates the earliest of the composition time within the VOPs carried
in
> > the RTP packet. Timestamp information of the rest of VOPs are derived
from
> > the timestamp fields in the VOP header (modulo_time_base and
> > vop_time_increment).?
> >
> Are you sure you want to possibly indicate the CTS of the second VOP in a
> packet ?
> This seems to me quite difficult to manage
> I would better do either of the following:
> - always indicate the CTS of the first VOP in the packet, even if the
second
> VOP will be presented first
> - disallow concatenation of VOPs whose decoding order differs from the
> composition order
>
I agree with Guido. I support the latter one becase it is a safer soluiton,
if there is no objection.

> In other words: VOP concatenation has pros and cons.
> Pros: optimizes bandwidth consuption
> Cons: deteriorates packet loss resilience; increases implementation
> complexity.
> I therefore deduce that VOP concatenation is nice but not that wonderful,

Therefore, fragmentation rule (4) in section 3.2 is defined that multiple
VOPs SHOULD NOT be concatenated. The concatenation is an exception rule only
when VOP size is small.

   (4) Two or more VOPs SHOULD be fragmented into different RTP packets so
   that one RTP packet consists of the data bytes associated with a unique
   presentation time (that is indicated in the timestamp field in the RTP
   packet header), with the exception that multiple VOPs MAY be carried
   within one RTP packet if the size of the VOPs is small.

> and I would suggest to allow VOP concatenation only when this is trivial
and
> does not require significant complexity in the receivers implementations.
>
The complexity may be huge if we need to check every RTP packets. However,
because of the marker bit and fragmentation rule definitions, we need check
only RTP packets satisfying the following both two conditions:
- Marker bit equal to one, and
  (Because marker bit in section 3.1 is defined that it is set to one when
multiple VOP is concatenated.)
- The beginning of RTP payload is not a start code.
  (Because the fragmentation rule is defined that a start code is placed
always at the beginning of RTP payload, if exists.)

I would like to propose adding a note to describe this for clarify, so that
implementer can avoid unnecessary operation.

Best regards,
Yoshihiro

----
        Yoshihiro Kikuchi

E-mail: kiku@eel.rdc.toshiba.co.jp
Corporate Research and Development Center
TOSHIBA Corporation
TEL: +81 +44 549 2288    FAX: +81 +44 520 1267





From rem-conf Fri Sep 08 00:10:35 2000 
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To: "Franceschini Guido" <Guido.Franceschini@CSELT.IT>,
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	"IETF AVT" <rem-conf@es.net>,
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Cc: "4on2andIP-sys" <4on2andIP-sys@advent.ee.columbia.edu>
References: <85077463E51D6A498C453073E1677940034E9C@exc2k02.cselt.it>
Subject: Re: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt
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Message-ID: <04a801c0196d$d2d9e980$280f010a@default>
From: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
To: "Young-Kwon LIM" <young@techway.co.kr>,
        "Jiro Katto" <katto@katto.comm.waseda.ac.jp>, <rem-conf@es.net>
Cc: <4on2andIP-sys@advent.ee.columbia.edu>,
        "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
References: <007c01c018ad$cfc532b0$013de8c3@Young> <00c901c0193f$e578b0e0$dcfa0985@katto.comm.waseda.ac.jp> <012701c0195c$e286bae0$f47d86c2@Young>
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call
Date: Fri, 8 Sep 2000 17:21:42 +0900
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Dear Katto-san, Young-Kwon, all,

> > 1.  We suggest to have a room for future extension of RTP header (e.g.
one byte
> > of payload header at the beginning of the payload part) to make the
devices
> > implementing this draft interoperable with the bitstream based on the
generic
> > format under development. For example, it is possible to use the
reserved byte to
> > specify the length of the payload header of generic payload format. Then
the device
> > implementing this draft can play the bitstream of generic payload format
by simply
> > skipping those bytes.
>
> This may be good compromise. But, usage of extension byte in the MPEG-4
> payload header, instead of defining another payload header, seems to
conclude
> that the "generic" payload format is MPEG-4 "specific." Let me confirm
that
> "generic" means MPEG-4 generic, not applied to other codecs nor future
ones.
>
As clearly described in the introduction, the intention of the MPEG-4
Audio/Visual payload format includes that this payload format will be used
in the existing frameworks (such as H.323) in an ordinary way, and can be
handled in an unified way together with those formats defined for non-MPEG-4
codecs. This, of cource, does not exclude the use in the MPEG's 4-on-IP
framework, but we must be care that any changes will not harm the original
intention.

>
> ==> Yes, you are right. The AHG group is working on MPEG-4 over IP.
>

I understand the importance of interoperability, however, isn't it better to
ask IETF experts if the specific proposal of adding a reserved byte in the
RTP header is really a general way also for other frameworks?

I believe this is the intention of the comment.

Kind regards,
Yoshihiro Kikuchi

----
        Yoshihiro Kikuchi

E-mail: kiku@eel.rdc.toshiba.co.jp
Corporate Research and Development Center
TOSHIBA Corporation
TEL: +81 +44 549 2288    FAX: +81 +44 520 1267





From rem-conf Fri Sep 08 01:49:11 2000 
From rem-conf-request@es.net Fri Sep 08 01:49:10 2000
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Message-ID: <000b01c01971$129530e0$390886c2@Young>
Reply-To: "Young-Kwon LIM" <young@techway.co.kr>
From: "Young-Kwon LIM" <young@techway.co.kr>
To: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>,
	"Franceschini Guido" <Guido.Franceschini@cselt.it>,
	"Koji Imura" <imura@telecom.mci.mei.co.jp>,
	"IETF AVT" <rem-conf@es.net>,
	"Colin Perkins" <colin@isi.edu>,
	"Varesio Andrea" <Andrea.Varesio@cselt.it>
Cc: "4on2andIP-sys" <4on2andIP-sys@advent.ee.columbia.edu>,
	"Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
References: <85077463E51D6A498C453073E1677940034E9C@exc2k02.cselt.it> <02f401c0195c$89eefb40$280f010a@default>
Subject: Re: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt
Date: Fri, 8 Sep 2000 17:38:51 +0900
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From rem-conf Fri Sep 08 04:02:07 2000 
From rem-conf-request@es.net Fri Sep 08 04:02:05 2000
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Date: Fri, 08 Sep 2000 12:55:27 +0200
To: "Young-Kwon LIM" <young@techway.co.kr>
From: Stephan Wenger <stewe@cs.tu-berlin.de>
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
  to the last call
Cc: <rem-conf@es.net>, <4on2andIP-sys@advent.ee.columbia.edu>
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Folks,

>1.  We suggest to have a room for future extension of RTP header (e.g. one 
>byte of payload header at the beginning of the payload part) to make the 
>devices implementing this draft interoperable with the bitstream based on 
>the generic format under development. For example, it is possible to use 
>the reserved byte to specify the length of the payload header of generic 
>payload format. Then the device implementing this draft can play the 
>bitstream of generic payload format by simply skipping those bytes.

I'm not sure whether this is a good idea.  We would waste one byte of
payload (from RTP's point of view) for something which is probably not
used by any systems currently under discussion or development.

Regardless how the generic payload format of the future may look like,
systems can be most easily designed to support both payload types.
RTP's PT mechanism has enough numbering space to support both.
Capability announcement/negotiation will be different and have to
be redesigned anyway for anything more capable then what is
currently in the MPEG-4 ES draft.  Annoucing/Negotiating another
PT for the new generic payload spec will not make a difference.

When, in the future, a generic payload type would be available that
includes a super set of functionality of the MPEG-4 ES payload format,
then we could declare the MPEG-4 ES RFC as obsolete, thereby
advising people to move over to the new, generic payload type.

Or, it may also be possible to have two payload specs coexisting,
one for simple audio/video ES packetization only, the other containing
more (the rest of?) functionality of MPEG-4.  This does not sound
like a bad idea to me, because the necessary demands of functionality
both from the network (in terms of QoS) and implementation efforts
(single audio/visual ESs vs. system layer, flexmux, whatever...) are
very different.

Please let the header format as is -- no more changes, otherwise you
can almost certainly forget the deadline you want to meet (mid
November's SG16 meeting).

Stephan






From rem-conf Fri Sep 08 04:43:19 2000 
From rem-conf-request@es.net Fri Sep 08 04:43:18 2000
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Message-ID: <001d01c01988$e6cb3e60$dcfa0985@katto.comm.waseda.ac.jp>
From: "Jiro Katto" <katto@katto.comm.waseda.ac.jp>
To: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>,
        "Young-Kwon LIM" <young@techway.co.kr>, <rem-conf@es.net>
Cc: <4on2andIP-sys@advent.ee.columbia.edu>
References: <007c01c018ad$cfc532b0$013de8c3@Young> <00c901c0193f$e578b0e0$dcfa0985@katto.comm.waseda.ac.jp> <012701c0195c$e286bae0$f47d86c2@Young> <04a801c0196d$d2d9e980$280f010a@default>
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call
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From rem-conf Fri Sep 08 05:41:36 2000 
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From: ryanben4@yahoo.com
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To: ryanben4@yahoo.com
Subject: Does your job pay you CASH?
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PROGRAMS UNTIL YOU DO. ONCE YOU HAVE 100 ORDERS, THE
REST IS EASY, RELAX, YOU WILL MAKE YOUR $50,000 GOAL."

Well, I had 196 orders for REPORT #2. 96 more than I
needed. So I sat back and relaxed. By March 1, of my
e-mailing of 10,000, received $58,000 with more coming
in every day. I paid off ALL my debts and bought a
much needed new car!

Please take your time to read this plan, IT WILL
CHANGE YOUR LIFE FOREVER!!!

Remember, it won=92t work if you don=92t try it. This
program does work, but you must follow it EXACTLY!
Especially the rules of not trying to place your name
in a different place. It won=92t work and you=92ll lose
out on a lot of money! In order for this program to
work, you must meet your goal of 20+ orders for
REPORT#1, and 100+orders for REPORT#2 and you will
make $50,000 or more in 90 days.

I AM LIVING PROOF THAT IT WORKS!!! If you choose not
to participate in this program, I am sorry. It really
is a great opportunity with little cost or risk to
you. If you choose to participate, follow the program
and you will be on your way to financial security. If
you are a fellow business owner and are in financial
trouble like I was, or your want to start your own
business, consider this a sigh. I DID!!!

Sincerely,

Jonathan Rourke

A PERSONAL NOTE FROM THE ORIGINATOR OF THIS PROGRAM:

By the time you have read the enclosed program and
reports, you should have concluded that such a
program, and one that is legal, could not have been
created by an amateur. Let me tell you a little about
myself, I had a profitable business for 10 years. Then
in 1979 my business began falling off. I was doing the
same things that were previously successful for me,
but it wasn=92t working. Finally, I figured it out. It
wasn=92t me, it was the economy. Inflation and recession
had replaced the stable economy that had been with us
since 1945.

I don=92t have to tell you what happened to the
unemployment rate because many of you know from first
hand experience. There were more failures and
bankruptcies than ever before. The middle class was
vanishing. Those who know what they were doing
invested wisely and moved up. Those who did not,
including those who never had anything to save or
invest, were moving down into the ranks of the poor.

As the saying goes, "THE RICH GET RICHER AND THE POOR
GET POORER. The traditional methods of making money
will never allow you to "move up" or "get rich",
inflation will see to that.

You have just received information that can give you
financial freedom for the rest of you life, with "NO
RISK" and "JUST A LITTLE BIT OF EFFORT." You can make
more money in the next few months than you have ever
imagined. I should also point out that I will not see
a penny of this money, nor anyone else who has
provided a testimonial for this program.

I have retired from the program after sending
thousands and thousands of programs. Follow the
program EXACTLY ASINSTRUCTED. Do not change it in any
way. It works exceedingly well as it is now.

Remember to e-mail a copy of this exciting report to
everyone you can think of. One of the people you send
this to may send out 50,000=85 and your name will be on
every one of them!

Remember though, the more you send out, the more
potential customers you will reach. So my friend, I
have given you the ideas, information, materials and
opportunity to become financially independent.

$$$IT WORKS!!!$$$

 HERE=92S HOW THIS AMAZING PROGRAM WILL MAKE
YOU THOUSANDS OF DOLLARS$$$$!!!!

This method of raising capital REALLY WORKS 100% EVERY
TIME.  As with all multi-level business, we build our
business by recruiting new partners and selling our
products. Every state in the USA allows you to recruit
new multi-level business partners, and we sell and
deliver a product for EVERY dollar received.

YOUR ORDERS COME BY MAIL AND ARE FILLED BY E-MAIL, so
you are not involved in personal selling. You do it
privately in your own home, store or office. This is
the EASIEST marketing plan anywhere! It is simply
order filling by e-mail!

The product is informational and instructional
material. Keys to the secrets for everyone on how to
open the doors to the magic world of E-COMMERCE, the
information highway, the wave of the future!!!


PLAN SUMMARY:

  1.You order the 4 reports listed below ($5 each)
They come
     to you by e-mail
  2.Save a copy of this entire letter and put your
name after
     Report #1 and move the other names down.
  3.Via the internet, access Yahoo.com or any of the
other
     major search engines to locate hundreds of bulk
e-mail
     service companies (search for "Bulk e-mail" and
have them
     send 25,000 =96 50,000 e-mails for you.)
  4.Orders will come to you by postal mail- simply
e-mail them
     the Report they ordered.

Let me ask you =96 isn=92t this about as easy as it gets?

 By the way there are over 100 MILLION e-mail
addresses with millions more joining the internet each
year so don=92t worry about "running out" or
"saturation". People are used to seeing and hearing
the same advertisements every day on radio/TV. How
many times have you received the same pizza flyers on
your door? Then one day you are hungry for pizza and
you order one. Same thing with this letter I received
this letter many times =96 then one day I decided it was
time to try it.

 YOU CAN START TODAY =96 JUST DO THESE EASY STEPS:

STEP #1. ORDER THE FOUR REPORTS

Order the four reports shown on the list below (you
can=92t sell them if you don=92t order them). =96 For each
report, send $5.00 CASH, the NAME & NUMBER OF THE
REPORT YOU ARE ORDERING YOUR E-MAIL ADDRESS, and YOUR
NAME & RETURN ADDRESS (in case of a problem) to the
person whose name appears on the list next to the
report.

MAKE SURE YOUR RETURN ADDRESS IS ON YOUR
ENVELOPE IN CASE OF ANY MAIL PROBLEMS!

Within a few days you will receive, by e-mail, each of
the four reports. Save them on your computer so you
can send them to the 1,000=92s of people who will order
them from you.

STEP #2. ADD YOUR MAILING ADDRESS TO THIS LETTER

  a.Look below for the listing of the four reports.
  b.After you=92ve ordered the four reports, delete the
name and
     address under REPORT #4. This person has made it
     through the cycle.
  c.Move the name and address under REPORT #3 down to
     REPORT #4.
  d.Move the name and address under REPORT #2 down to
     REPORT #3.
  e.Move the name and address under REPORT #1 down to
     REPORT #2.
  f.Insert your name/address in the REPORT #1
position.

Please make sure you COPY ALL INFORMATION, every name
and address, ACCURATELY!

STEP #3. Take this entire letter, including the
modified list of names, and save it to your computer.
Make NO changes to these instructions. Now you are
ready to use this entire e-mail to send by e-mail to
prospects.

Report #1 will tell you how to download bulk e-mail
software and e-mail addresses so you can send it out
to thousands of people while you sleep! Remember that
50,000+ new people are joining the internet every
month.

Your cost to participate in this is practically
nothing (surely you can afford $20). You obviously
already have a computer and an Internet connection and
e-mail is FREE!

There are two primary methods of building your
downline:

METHOD #1: SENDING BULK E-MAIL

Let=92s say that you decide to start small, just to see
how it goes, and we=92ll assume you and all those
involved e-mail out only 2,000 programs each. Let=92s
also assume that the mailing receives a 0.5% response.
The response could be much better. Also, many people
will e-mail out hundreds of thousands of programs
instead of 2,000 (Why stop at 2,000?). But continuing
with this example, you send out only 2,000 programs.
With a 0.5% response, that is only 10 orders for
REPORT #1. Those 10 people respond by sending out
2,000 programs each for a total of 20,000. Out of
those 0.5%, 100 people respond and order REPORT #2.
Those 100 mail out 2,000 programs each for a total of
200,000.

The 0.5% response to that is 1,000 orders for REPORT
#3. Those 1,000 send out 2,000 programs each for a
2,000,000 total. The 0.5% response to that is 10,000
orders for REPORT #4. That=92s 10,000 $5 bills for you.
CASH!!! Your total income in this example is $50 +
$500 +$5,00 + $50,000 for a total of $55,550!!!

REMEMBER, THIS IS ASSUMING 1,990 OUT OF THE 2,000
PEOPLE YOU MAIL TO WILL DO ABSOLUTELY NOTHING AND
TRASH THIS PROGRAM! DARE TO THINK FOR A MOMENT THAT
WOULD HAPPEN IF EVERYONE, OR HALF SENT OUT 100,000
PROGRAMS INSTEAD OF 2,000

Believe me, many people will do just that, and more!

METHOD #2 =96 PLACING FREE ADS ON THE INTERNET

Advertising on the internet is very, very inexpensive,
and there are HUNDREDS of FREE places to advertise.
Let=92s say you decide to start small just to see how
well it works. Assume your goal is to get ONLY 10
people to participate on your first level. (Placing a
lot of FREE ads on the Internet will EASILY get a
larger response.)

Also assume that everyone else in YOUR ORGANIZATION
gets ONLY 10 downline members. Look how this small
number accumulates to achieve the STAGGERING results
below:

1st level: your first 10 send you $5=85=85=85=85.$50

2nd level: 10 members from those 10 ($5 x 100)=85=85=85.$500

3rd level: 10 members from those 100 ($5 x
1,000)=85=85$5,000

4th level: 10 members from those 1,000 ($5 x
10,000)=85$50,000

$$$$$$$$$$$$THIS
TOTALS------------$55,550$$$$$$$$$$$$$$

AMAZING ISN=92T IT? Remember, this assumes that the
people who participate only recruit 10 people each.
Think for a moment what would happen if they got 20
people to participate! Most people get 100=92s of
participants and many will continue to work this
program, sending out programs WITH YOUR NAME ON THEM
for years! THINK ABOUT IT!


People are going to get e-mails about this plan from
you or somebody else and many will work this plan- the
question is =96 Don=92t you want your name to be on the
e-mails they send out?

***DON=92T MISS OUT!!!***JUST TRY IT ONCE!!!***

***SEE WHAT HAPPENS!!!***YOU=92LL BE AMAZED!!!***

 ALWAYS PROVIDE SAME-DAY SERVICE ON ALL ORDERS!

This will guarantee that the e-mail THEY send out with
YOUR name and address on it will be prompt because
they can=92t advertise until they receive the report!



GET STARTED TODAY: PLACE YOUR ORDER FOR THE
FOUR REPORTS NOW.

Notes: ALWAYS SEND $5 CASH (U.S. CURRENCY) FOR EACH
REPORT. CHECKS NOT ACCEPTED.

Enclose the $5 with a sheet of paper including:

  a.the number & name of the report you are ordering.
  b.Your e-mail address,
  c.Your name & postal address.

REPORT #1 "The Insider=92s Guide to Advertising for Free
on the Internet"

ORDER REPORT #1 FROM

Ryan Doyle
3327 Boyce Ln.
San Diego, CA  92105

REPORT #2 "The Insider=92s Guide to Sending Bulk E-mail
on
the Internet"

ORDER REPORT #2 FROM

Max Glinsky
3532 Governor Dr.
La Jolla, CA  92122

REPORT #3 "The Secrets to Multilevel Marketing on the
Internet"

ORDER REPORT #3 FROM

Randy Catlett
817 W. Iola Pl.
Broken Arrow, OK 74012

REPORT #4 "How to become a Millionaire utilizing the
Power
of Multilevel Marketing and the Internet"

 ORDER REPORT #4 FROM

Vic Roberts
HC 34 Box 2831
Wasilla, AK 99688


*************TIPS FOR SUCCESS***********************

TREAT THIS AS YOUR BUSINESS! Be prompt, professional,
and follow the directions accurately. Send for the
four reports IMMEDIATELY so you will have them when
the orders start coming in, because: When you receive
a $5 order, you MUST send out the requested
product/report. It is required for this to be a legal
business and they need the reports to send out their
letters (with your name on them!)

ALWAYS PROVIDE SAME-DAY SERVICE ON THE ORDERS YOU
RECEIVE. Be patient and persistent with this program-
If you follow the instructions exactly =96 results WILL
FOLLOW.$$$$

***********YOUR SUCCESS GUIDELINES***************

Follow these guidelines to guarantee your success" If
you don=92t receive 20 orders for REPORT #1 within two
weeks, continue advertising or sending e-mails until
you do. Then, a couple of weeks later you should
receive at least 100 orders for REPORT #2

 If you don=92t continue advertising or sending e-mails
until you do. Once you have received 100 or more
orders for REPORT #2, YOU CAN RELAX, because the
system is already working for you, and the cash will
continue to roll in!

THIS IS IMPORTANT TO REMEMBER: Every time your name is
moved down on the list, you are placed in front of a
DIFFERENT report. You can KEEP TRACK of your PROGRESS
by watching which report people are ordering from you.

To generate more income, simply send another batch of
e-mails or continue placing ads and start the whole
process again! There is no limit to the income you
will generate from this business!

Before you make your decision as to whether or not you
participate in this program. Please answer one
question. ARE YOU HAPPY WITH YOUR PRESENT INCOME OR
JOB? If the answer is no, then please look at the
following facts about this super simple MLM program:

  1.NO face to face selling, NO meetings, NO
inventory, NO
     Telephone calls, NO big cost to start, Nothing to
learn, NO
     skills needed! (Surely you know how to send
e-mail?)
  2.NO equipment to buy =96 you already have a computer
and
     internet connection =96 so you have everything you
need to
     fill orders!
  3.You are selling a product which does NOT COST
     ANYTHING TO PRODUCE OR SHIP! (e-mailing copies
     of the reports is FREE!)
  4.All of your customers pay you in CA$H! This
program will
     change your LIFE FOREVER!!! Look at the potential
for
     you to be able to work a super-high paying
leisurely easy
     business from home!

$$$$$$FINALLY MAKE SOME DREAMS COME TRUE!$$$$$

ACT NOW! Take your first step toward making some extra
cash, or even achieving financial independence. Order
the reports and follow the program outlined
above=97SUCCESS will be your reward.

See you at the top.

PLEASE NOTE: If you need help with starting a
business, registering a business name, learning how
income tax is handled, etc., contact your local office
of the Small Business Administration (a Federal
Agency) 1-800-827-5722 for free help and answers to
questions.

Also, the Internal Revenue Service offers free help
via telephone and free seminars about business tax
requirements. Your earnings are highly dependent on
your activities and advertising. The information
contained on this site and in the report constitutes
no guarantees stated nor implied. In the event that it
is determined that this site or report constitutes a
guarantee of any kind, that guarantee is now void. The
earnings amounts listed on this site and in the report
are estimates only. If you have any questions of the
legality of this program, contact the Office of
Associate Director for Marketing Practices, Federal
Trade Commission, Bureau of Consumer Protection in
Washington, DC.

Under Bill s.1618 TITLE 111 passed by the 105th US
Congress, this letter cannot be considered spam as
long as the sender includes contact information and a
method of removal.

This is a one time e-mail transmission. No request for
removal is necessary.





From rem-conf Fri Sep 08 07:03:33 2000 
From rem-conf-request@es.net Fri Sep 08 07:03:33 2000
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Message-ID: <00f701c01991$54a18e00$27c77885@default>
From: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
To: "Young-Kwon LIM" <young@techway.co.kr>,
        "Franceschini Guido" <Guido.Franceschini@cselt.it>,
        "Koji Imura" <imura@telecom.mci.mei.co.jp>,
        "IETF AVT" <rem-conf@es.net>, "Colin Perkins" <colin@isi.edu>,
        "Varesio Andrea" <Andrea.Varesio@cselt.it>
Cc: "4on2andIP-sys" <4on2andIP-sys@advent.ee.columbia.edu>,
        "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
References: <85077463E51D6A498C453073E1677940034E9C@exc2k02.cselt.it> <02f401c0195c$89eefb40$280f010a@default> <000b01c01971$129530e0$390886c2@Young>
Subject: Re: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt
Date: Fri, 8 Sep 2000 21:35:51 +0900
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Dear Young-Kwon, Guido, Matsui-san, Imura-san, all,

Thank you very much for the comment.

It seems to be better to list all description related the multiple VOPs
concatenation in the "current" draft text posted on Sep. 5 here in this mail
for the better understandings.

1. Introduction
   The fragmentation rule recommends not to map more than one VOP in an RTP
   packet so that RTP timestamp uniquely indicates the VOP time framing. On
   the other hand, MPEG-4 video may generate VOPs of very small size, in
   cases with a not coded VOP containing only VOP header or an arbitrary
   shaped VOP with a small number. To reduce the overhead for such cases,
   the fragmentation rule permits concatenating multiple VOPs in an RTP
   packet when the VOP size is small.

3.1 Use of RTP header fields for MPEG-4 Visual
   Marker (M) bit: The marker bit is set to one to indicate the last RTP
   packet (or only RTP packet) of a VOP. When multiple VOPs are carried in
   the same RTP packet, the marker bit is set to 1.

   Timestamp:
   - When multiple VOPs are carried in the same RTP packet, the timestamp
     indicates the earliest of the composition time within the VOPs carried
     in the RTP packet.

3.2 Fragmentation of MPEG-4 Visual bitstream
   (4) Two or more VOPs SHOULD be fragmented into different RTP packets so
   that one RTP packet consists of the data bytes associated with a unique
   presentation time (that is indicated in the timestamp field in the RTP
   packet header), with the exception that multiple VOPs MAY be carried
   within one RTP packet if the size of the VOPs is small.


I think it is better to add cross-references (see section ..., definition of
....) for each description above in the draft so that every readers of the
draft can understand this set of rule easily.

Young-Kwon wrote:

> I don't think there is additional complexity. MPEG-4 video decoder always
calculate the time difference of current VOP from the last one.  So, when
the decoder push the reconstructed data to composition buffer, it can easily
send a presentation time information to the compositor. It is nothing to the
implementation.

> I'm OK with this because this allows concatenation of VOPs.

Oh, now there are agreements on the current description. As usual, if there
is any other strong disagreements, we should stick with the current text.
(except for editorial changes and clarifications)

Please look not only sentences I posted but a set of rules. It is defined
that multiple VOPs SHOULD NOT be concatenated so as not to force huge amout
of complexity. The concatenation is an exception rule only when VOP size is
small.

>For more clarification I'd like to add "When multiple VOPs are carried in
one RTP payload, the presentation time of the VOPs after the first one may
be calculated by the decoder."
>
Agreed.

> BTW, where is a sentence prohibiting concatenation of VOPs with different
decoding order?

There is no sentence now in the draft but only in mails exchanged.

Following is the list of proposed changes by mails on this issue for
editorial update and clarifications but not technical.

1) Add cross-cross references to each descriptions.
2) Add the following sentence to the timestamp definition
    "Timestamp information of the rest of VOPs are derived from the
timestamp fields in the VOP header (modulo_time_base and
vop_time_increment)."
3) Add the following description as a note to fragmentation rule (4)
    "When multiple VOPs are carried in one RTP payload, the presentation
time of the VOPs after the first one may be calculated by the decoder."
    "This operation is necessary only for RTP packets satisfying the
following both two conditions:
- Marker bit equal to one, and
  (Because marker bit in section 3.1 is defined that it is set to one when
multiple VOP is concatenated.)
- The beginning of RTP payload is not a start code.
  (Because the fragmentation rule is defined that a start code is placed
always at the beginning of RTP payload, if exists.)"


If there is no further objection, only changes reflected to the next version
of I/D will be these descriptions for editorial update and clarifications.


Best regards,
Yoshihiro

----
        Yoshihiro Kikuchi

E-mail: kiku@eel.rdc.toshiba.co.jp
Corporate Research and Development Center
TOSHIBA Corporation
TEL: +81 +44 549 2288    FAX: +81 +44 520 1267





From rem-conf Fri Sep 08 07:06:24 2000 
From rem-conf-request@es.net Fri Sep 08 07:06:23 2000
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To: Young-Kwon LIM <young@techway.co.kr>
cc: rem-conf <rem-conf@es.net>,
        4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
In-Reply-To: Your message of "Thu, 07 Sep 2000 18:19:41 +0900."
             <007c01c018ad$cfc532b0$013de8c3@Young> 
Date: Fri, 08 Sep 2000 10:04:20 -0400
From: Colin Perkins <csp@east.isi.edu>
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--> Young-Kwon LIM writes:
>Dear Collin and AVT working group members,
>
>Yesterday 20 MPEG experts had an AHG meeting at Paris. We discussed the
>technical problems and the solutions in terms of basic interoperabilities
>what we want to achieve form the ongoing work to develop overall framework
>for carriage of MPEG-4 content over IP. Based on the consensus of the AHG
>members, we would like to make two comments to possibly ensure the basic
>interoperabilities between the payload format of this draft and the generic
>payload foramt under development.
>
>1.  We suggest to have a room for future extension of RTP header (e.g. one
>byte of payload header at the beginning of the payload part) to make the
>devices implementing this draft interoperable with the bitstream based on
>the generic format under development. For example, it is possible to use
>the reserved byte to specify the length of the payload header of generic
>payload format. Then the device implementing this draft can play the
>bitstream of generic payload format by simply skipping those bytes.

I agree with the comment made by Stephan Wenger. The RTP payload type field
is intended to differentiate formats, and I don't believe that this byte will 
sufficiently aid future interoperability to make its inclusion worthwhile. 

>2. We request to clarify the usage of LATM. It was recognized during the
>meeting that there seems no reason to use multiplexing feature of LATM for
>the purpose of this draft. In addition, limiting the usage of LATM to carry
>configuation information and concatenation of audio frames is desirable to
>make this draft more compatible to the generic payload format under
>development.

There is indeed no need for the multiplexing features of LATM. Is the gain
of designing an alternative header worth it? We may simply specify that
that feature of LATM should not be used?

Colin



From rem-conf Fri Sep 08 07:40:33 2000 
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Date: Fri, 08 Sep 2000 16:33:42 +0200
From: Franceschini Guido <Guido.Franceschini@CSELT.IT>
Subject: RE: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt
To: Young-Kwon LIM <young@techway.co.kr>,
 Franceschini Guido <guido.franceschini@cselt.it>,
 Koji Imura <imura@telecom.mci.mei.co.jp>, IETF AVT <rem-conf@es.net>,
 Colin Perkins <colin@isi.edu>, Varesio Andrea <Andrea.Varesio@cselt.it>,
 'Yoshihiro Kikuchi' <kiku@eel.rdc.toshiba.co.jp>
Cc: 4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>
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Some further comment below

> ----------
> From: 	Yoshihiro Kikuchi[SMTP:kiku@eel.rdc.toshiba.co.jp]
> Sent: 	Friday, September 08, 2000 2:35 PM
> To: 	Young-Kwon LIM; Franceschini Guido; Koji Imura; IETF AVT; Colin
> Perkins; Varesio Andrea
> Cc: 	4on2andIP-sys; Yoshihiro Kikuchi
> Subject: 	Re: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt
> 
> Dear Young-Kwon, Guido, Matsui-san, Imura-san, all,
> 
> Thank you very much for the comment.
> 
> It seems to be better to list all description related the multiple VOPs
> concatenation in the "current" draft text posted on Sep. 5 here in this
> mail
> for the better understandings.
> 
> 1. Introduction
>    The fragmentation rule recommends not to map more than one VOP in an
> RTP
>    packet so that RTP timestamp uniquely indicates the VOP time framing.
> On
>    the other hand, MPEG-4 video may generate VOPs of very small size, in
>    cases with a not coded VOP containing only VOP header or an arbitrary
>    shaped VOP with a small number. To reduce the overhead for such cases,
>    the fragmentation rule permits concatenating multiple VOPs in an RTP
>    packet when the VOP size is small.
Editorial: I don't think the last statement 'when the VOP size is small' is
necessary/appropriate

> 3.1 Use of RTP header fields for MPEG-4 Visual
>    Marker (M) bit: The marker bit is set to one to indicate the last RTP
>    packet (or only RTP packet) of a VOP. When multiple VOPs are carried in
>    the same RTP packet, the marker bit is set to 1.
> 
>    Timestamp:
>    - When multiple VOPs are carried in the same RTP packet, the timestamp
>      indicates the earliest of the composition time within the VOPs
> carried
>      in the RTP packet.
> 
> 3.2 Fragmentation of MPEG-4 Visual bitstream
>    (4) Two or more VOPs SHOULD be fragmented into different RTP packets so
>    that one RTP packet consists of the data bytes associated with a unique
>    presentation time (that is indicated in the timestamp field in the RTP
>    packet header), with the exception that multiple VOPs MAY be carried
>    within one RTP packet if the size of the VOPs is small.
Editorial: again, what is the meaning of 'small' ? Also, at the beginning, I
would change to 'Different VOPs SHOULD be fragmented into different RTP
packets' 


> I think it is better to add cross-references (see section ..., definition
> of
> ....) for each description above in the draft so that every readers of the
> draft can understand this set of rule easily.
> 
> Young-Kwon wrote:
> 
> > I don't think there is additional complexity. MPEG-4 video decoder
> always
> calculate the time difference of current VOP from the last one.  So, when
> the decoder push the reconstructed data to composition buffer, it can
> easily
> send a presentation time information to the compositor. It is nothing to
> the
> implementation.
> 
> > I'm OK with this because this allows concatenation of VOPs.
> 
> Oh, now there are agreements on the current description. As usual, if
> there
> is any other strong disagreements, we should stick with the current text.
> (except for editorial changes and clarifications)
> 
> Please look not only sentences I posted but a set of rules. It is defined
> that multiple VOPs SHOULD NOT be concatenated so as not to force huge
> amout
> of complexity. The concatenation is an exception rule only when VOP size
> is
> small.
Just to make my point clear: either you quantify 'small', or you can at most
say something like: "concatenation is allowed only when an integral number
of VOPs fit into a single RTP packet"
I would suggest to have some text clearly specifying under which
circumstances VOP concatenation may occur:
Some idea for this text:

Concatenation is allowed only when the following conditions occur:
- the result of concatenting N consecutive VOPs still fits into a single RTP
packet
- the presentation and decoding order of the VOPs is identical
- ...

> >For more clarification I'd like to add "When multiple VOPs are carried in
> one RTP payload, the presentation time of the VOPs after the first one may
> be calculated by the decoder."
> >
> Agreed.
> 
> > BTW, where is a sentence prohibiting concatenation of VOPs with
> different
> decoding order?
> 
> There is no sentence now in the draft but only in mails exchanged.
> 
> Following is the list of proposed changes by mails on this issue for
> editorial update and clarifications but not technical.
> 
> 1) Add cross-cross references to each descriptions.
> 2) Add the following sentence to the timestamp definition
>     "Timestamp information of the rest of VOPs are derived from the
> timestamp fields in the VOP header (modulo_time_base and
> vop_time_increment)."
> 3) Add the following description as a note to fragmentation rule (4)
>     "When multiple VOPs are carried in one RTP payload, the presentation
> time of the VOPs after the first one may be calculated by the decoder."
>     "This operation is necessary only for RTP packets satisfying the
> following both two conditions:
> - Marker bit equal to one, and
>   (Because marker bit in section 3.1 is defined that it is set to one when
> multiple VOP is concatenated.)
> - The beginning of RTP payload is not a start code.
>   (Because the fragmentation rule is defined that a start code is placed
> always at the beginning of RTP payload, if exists.)"
I do not understand this second condition. I would have reversed it "The
beginning of RTP payload corresponds to a start code"

Bye
Guido


> If there is no further objection, only changes reflected to the next
> version
> of I/D will be these descriptions for editorial update and clarifications.
> 
> 
> Best regards,
> Yoshihiro
> 
> ----
>         Yoshihiro Kikuchi
> 
> E-mail: kiku@eel.rdc.toshiba.co.jp
> Corporate Research and Development Center
> TOSHIBA Corporation
> TEL: +81 +44 549 2288    FAX: +81 +44 520 1267
> 
> 



From rem-conf Fri Sep 08 10:33:54 2000 
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From: "Toshiyuki Nomura" <t-nomura@ccm.cl.nec.co.jp>
To: "Young-Kwon LIM" <young@techway.co.kr>, "Colin Perkins" <csp@east.isi.edu>
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Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
Date: Sat, 9 Sep 2000 02:44:04 +0900
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Dear Colin and Young,

> >2. We request to clarify the usage of LATM. It was recognized during the
> >meeting that there seems no reason to use multiplexing feature of LATM
for
> >the purpose of this draft. In addition, limiting the usage of LATM to
carry
> >configuation information and concatenation of audio frames is desirable
to
> >make this draft more compatible to the generic payload format under
> >development.
>
> There is indeed no need for the multiplexing features of LATM. Is the gain
> of designing an alternative header worth it? We may simply specify that
> that feature of LATM should not be used?

I think the content of Comment 1 does not relate to Comment 2.
Anyway, if I get the consensus from MPEG Audio experts,
I'll modify Introduction of the I/D to specify that some features
of LATM should not be used in RTP transmission.

Best Regards,

Toshiyuki Nomura, NEC
---------------------




From rem-conf Fri Sep 08 13:08:42 2000 
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From: Kris Huber <khuber@sorensonlabs.com>
To: Young-Kwon LIM <young@techway.co.kr>, Franceschini Guido
	 <Guido.Franceschini@cselt.it>, Koji Imura <imura@telecom.mci.mei.co.jp>, 
	IETF AVT <rem-conf@es.net>, Colin Perkins <colin@isi.edu>, 
	'Yoshihiro Kikuchi' <kiku@eel.rdc.toshiba.co.jp>, Varesio Andrea
	 <Andrea.Varesio@cselt.it>
Cc: 4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>, 
	garysull@microsoft.com
Subject: RE: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt - ti
	mestamp of short header VOP
Date: Fri, 8 Sep 2000 14:04:15 -0600 
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Dear Young and all,

Unless "MAY" is changed to "SHOULD" in "When the short video header mode is
used, the RTP payload format used MAY be that specified for H.263 in the
relevant RFCs or in other relevant standards,"  I think additional
modifications are necessary to accommodate MPEG-4 short header mode.  This
is because vop_time_increment_resolution, modulo_time_base, and time_code
are not present in a short header elementary stream.  Here is my proposed
modification (new text is indicated by surrounding it by "<<" and ">>").
No deletions are required.

   Timestamp: The timestamp indicates the composition time, or the
   presentation time in a no-compositor decoder. A constant offset, which is
   random, is added for security reasons. The detailed definition of the
   timestamp is as follows:
<<  - For a video object plane with short header, the timestamps (after the
first random timestamp) are equal to the presentation time sequence
associated with the semantics of the temporal_reference field.
Specifically, each timestamp value SHALL be calculated by rounding the
value of a precise clock that advances delta_time with each successive
video object plane with short header.  The time increment SHOULD be
calculated as delta_time = (((temporal_reference + 256 -
(temporal_reference of previous VOP) modulo 256) * 1001/30000) for each
successive video object plane with short header.  The RTP timestamp should
be consistently rounded or truncated to the resolution of the RTP timestamp
field.>>
   - <<Otherwise,>> for a video object plane, it is defined as
vop_time_increment (in units
     of 1/vop_time_increment_resolution seconds) plus the cumulative number
     of whole seconds specified by modulo_time_base and, if present,
     time_code of Group_of_VideoObjectPlane() fields.  
   - In the case of interlaced video, a VOP will consist of lines from two
     fields, and the timestamp will indicate the composition time of the
     first field.
(   - When multiple VOPs are carried in the same RTP packet, the timestamp
     indicates the earliest of the composition time within the VOPs carried
     in the RTP packet.)
Young-Kwon's proposed modification of the above bullet:
   - When multiple VOPs are carried in the same RTP packet, the timestamp
     indicates the earliest of the composition time<<s>> within the VOPs
carried in
     the RTP packet.  Timestamp information of the rest of <<the>> VOPs are
derived from
     the timestamp fields in the VOP header (modulo_time_base and
     vop_time_increment)<<, or from the temporal_reference field
     in the case of short video header>>.
   - If the RTP packet contains only configuration information and/or
     Group_of_VideoObjectPlane() fields, the composition time of the next
     VOP in the coding order is used.
   - If the RTP packet contains only visual_object_sequence_end_code
     information, the composition time of the immediately preceding VOP in
     the coding order is used.

Perhaps someone knows better wording for the first bullet.  I used SHALL in
that paragraph to prevent accumulation of round-off error in the sequence
of time stamps, and SHOULD to allow for other picture clock frequencies
than the one specified in the MPEG-4 visual semantics.  More detail about
this difference in MPEG-4 short header and H.263 semantics follows:

Quoting ISO/IEC 14496-2:1999, clause 6.3.5.2:  "temporal reference:  This
is an 8-bit number which can have 256 possible values.  It is formed by
incrementing its value in the previously transmitted
video_plane_with_short_header() by one plus the number of non-transmitted
pictures (at 30000/1001 Hz) since the previously transmitted picture.  The
arithmetic is performed with only the eight LSBs."

In contrast the TR field in the H.263 spec allows different clocks than
29.97 Hz:  "the value of TR is formed by incrementing its value in the
temporally-previous reference picture header by one plus the number of
skipped or non-reference pictures at the picture clock frequency since the
previously transmitted one.  The interpretation of TR depends upon the
active picture clock frequency. ..."  Negotiation of the picture clock
frequency is done by external means, I believe.

I suspect that fixing the H.263 TR frequency in the MPEG-4
temporal_reference semantics causes problems particularly for use of short-
header on 25 Hz material in Europe.

As I noted in the first sentence, an alternative to describing how to
compute timestamp in short header mode is simply to change MAY to SHOULD or
SHALL regarding use of H.263 payload format for short header mode.

Best regards,
Kris Huber
Sorenson Laboratories, Inc.

-----Original Message-----
From: Young-Kwon LIM [mailto:young@techway.co.kr]
Sent: Friday, September 08, 2000 1:08 AM
To: Franceschini Guido; Koji Imura; IETF AVT; Colin Perkins; 'Yoshihiro
Kikuchi'; Varesio Andrea
Cc: 4on2andIP-sys
Subject: Re: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt


Dear Guido and all,

> > 
> > It would be better to add a description to clarify this. How about
adding
> > the following description in the timestamp definition in section 3.1?
> > 
> > - When multiple VOPs are carried in the same RTP packet, the timestamp
> > indicates the earliest of the composition time within the VOPs carried
in
> > the RTP packet. Timestamp information of the rest of VOPs are derived
from
> > the timestamp fields in the VOP header (modulo_time_base and
> > vop_time_increment).?
> > 
> Are you sure you want to possibly indicate the CTS of the second VOP in a
> packet ?
> This seems to me quite difficult to manage
> I would better do either of the following:
> - always indicate the CTS of the first VOP in the packet, even if the
second
> VOP will be presented first
> - disallow concatenation of VOPs whose decoding order differs from the
> composition order

"modulo_time_base" and "vop_time_increment" are the syntax elements of VOP
header invented for the purpose of carrying relative timing information.
So, I don't see any problem to use those information to calcute CTS. When
the receiver get a RTP packet with multiple VOPs, the CTS of first VOP can
be derived from the RTP time stmap and CTS of successive VOPs can be
calculated by using time stamp and two syntax elements in VOP header.
Calculating timing information of each VOP is nothing new to the decoder.

> 
> I am also doubtful about allowing variable frame rate videos to take
> advantage of such concatenation of VOPs in the same RTP packet.
> IMHO it would be safer to add some more constraints in the specification
> (i.e. prohibit VOP concatenation in case of variable frame rates), instead
> of adding the complexity of extracting TS information through cooperation
of
> different layers.
> In other words: VOP concatenation has pros and cons.
> Pros: optimizes bandwidth consuption
> Cons: deteriorates packet loss resilience; increases implementation
> complexity.
> I therefore deduce that VOP concatenation is nice but not that wonderful,
> and I would suggest to allow VOP concatenation only when this is trivial
and
> does not require significant complexity in the receivers implementations.
> 

I don't think there is additional complexity. MPEG-4 video decoder always
calculate the time difference of current VOP from the last one.  So, when
the decoder push the reconstructed data to composition buffer, it can
easily send a presentation time information to the compositor. It is
nothing to the implementation.

Sincerely,
Young.



From rem-conf Fri Sep 08 14:53:55 2000 
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From: Gary Sullivan <garysull@microsoft.com>
To: 'Kris Huber' <khuber@sorensonlabs.com>, Young-Kwon LIM
	 <young@techway.co.kr>, Franceschini Guido <Guido.Franceschini@cselt.it>, 
	Koji Imura <imura@telecom.mci.mei.co.jp>, IETF AVT <rem-conf@es.net>, 
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	 <kiku@eel.rdc.toshiba.co.jp>, Varesio Andrea <Andrea.Varesio@cselt.it>
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Subject: RE: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt - ti
	 mestamp of short header VOP
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Kris et al,

A bit of H.263 explanation with a bit of a history lesson is
spliced into Kris's quoted message below:


+> -----Original Message-----
+> From: Kris Huber [mailto:khuber@sorensonlabs.com]
+> Sent: Friday, September 08, 2000 1:04 PM
+> To: Young-Kwon LIM; Franceschini Guido; Koji Imura; IETF 
+> AVT; Colin Perkins; 'Yoshihiro Kikuchi'; Varesio Andrea
+> Cc: 4on2andIP-sys; Gary Sullivan
+> Subject: RE: IETF AVT last call on 
+> draft-ietf-avt-rtp-mpeg4-es-03.txt - ti mestamp of short header VOP
+> 
+> 
+> Dear Young and all,
+> 
+> Unless "MAY" is changed to "SHOULD" in "When the short video 
+> header mode is
+> used, the RTP payload format used MAY be that specified for 
+> H.263 in the
+> relevant RFCs or in other relevant standards,"  I think additional
+> modifications are necessary to accommodate MPEG-4 short 
+> header mode.  This
+> is because vop_time_increment_resolution, modulo_time_base, 
+> and time_code
+> are not present in a short header elementary stream.  Here 
+> is my proposed
+> modification (new text is indicated by surrounding it by 
+> "<<" and ">>").
+> No deletions are required.
+> 
+>    Timestamp: The timestamp indicates the composition time, or the
+>    presentation time in a no-compositor decoder. A constant 
+> offset, which is
+>    random, is added for security reasons. The detailed 
+> definition of the
+>    timestamp is as follows:
+> <<  - For a video object plane with short header, the 
+> timestamps (after the
+> first random timestamp) are equal to the presentation time sequence
+> associated with the semantics of the temporal_reference field.
+> Specifically, each timestamp value SHALL be calculated by 
+> rounding the
+> value of a precise clock that advances delta_time with each 
+> successive
+> video object plane with short header.  The time increment SHOULD be
+> calculated as delta_time = (((temporal_reference + 256 -
+> (temporal_reference of previous VOP) modulo 256) * 
+> 1001/30000) for each
+> successive video object plane with short header.  The RTP 
+> timestamp should
+> be consistently rounded or truncated to the resolution of 
+> the RTP timestamp
+> field.>>
+>    - <<Otherwise,>> for a video object plane, it is defined as
+> vop_time_increment (in units
+>      of 1/vop_time_increment_resolution seconds) plus the 
+> cumulative number
+>      of whole seconds specified by modulo_time_base and, if present,
+>      time_code of Group_of_VideoObjectPlane() fields.  
+>    - In the case of interlaced video, a VOP will consist of 
+> lines from two
+>      fields, and the timestamp will indicate the composition 
+> time of the
+>      first field.
+> (   - When multiple VOPs are carried in the same RTP packet, 
+> the timestamp
+>      indicates the earliest of the composition time within 
+> the VOPs carried
+>      in the RTP packet.)
+> Young-Kwon's proposed modification of the above bullet:
+>    - When multiple VOPs are carried in the same RTP packet, 
+> the timestamp
+>      indicates the earliest of the composition time<<s>> 
+> within the VOPs
+> carried in
+>      the RTP packet.  Timestamp information of the rest of 
+> <<the>> VOPs are
+> derived from
+>      the timestamp fields in the VOP header (modulo_time_base and
+>      vop_time_increment)<<, or from the temporal_reference field
+>      in the case of short video header>>.
+>    - If the RTP packet contains only configuration information and/or
+>      Group_of_VideoObjectPlane() fields, the composition 
+> time of the next
+>      VOP in the coding order is used.
+>    - If the RTP packet contains only visual_object_sequence_end_code
+>      information, the composition time of the immediately 
+> preceding VOP in
+>      the coding order is used.
+> 
+> Perhaps someone knows better wording for the first bullet.  
+> I used SHALL in
+> that paragraph to prevent accumulation of round-off error in 
+> the sequence
+> of time stamps, and SHOULD to allow for other picture clock 
+> frequencies
+> than the one specified in the MPEG-4 visual semantics.  More 
+> detail about
+> this difference in MPEG-4 short header and H.263 semantics follows:
+> 
+> Quoting ISO/IEC 14496-2:1999, clause 6.3.5.2:  "temporal 
+> reference:  This
+> is an 8-bit number which can have 256 possible values.  It 
+> is formed by
+> incrementing its value in the previously transmitted
+> video_plane_with_short_header() by one plus the number of 
+> non-transmitted
+> pictures (at 30000/1001 Hz) since the previously transmitted 
+> picture.  The
+> arithmetic is performed with only the eight LSBs."
+> 
+> In contrast the TR field in the H.263 spec allows different 
+> clocks than
+> 29.97 Hz:  "the value of TR is formed by incrementing its 
+> value in the
+> temporally-previous reference picture header by one plus the 
+> number of
+> skipped or non-reference pictures at the picture clock 
+> frequency since the
+> previously transmitted one.  The interpretation of TR 
+> depends upon the
+> active picture clock frequency. ..."  Negotiation of the 
+> picture clock
+> frequency is done by external means, I believe.
+> 
+> I suspect that fixing the H.263 TR frequency in the MPEG-4
+> temporal_reference semantics causes problems particularly 
+> for use of short-
+> header on 25 Hz material in Europe.


MPEG-4's short header mode corresponds with the H.263 "baseline"
profile of operation.  The baseline form only supports one picture
clock frequency -- 30000/1001 Hz.  The extra flexibility of
supporting other picture clock frequencies is an optional
feature that was added in version 2.  MPEG-4 does not support
compatibility for any optional H.263 optional features (even for the
Advanced Prediction mode, which would have been a very easy one
to specify since all the tools are in MPEG-4 in an identical form).
It only supports the baseline form.  The baseline form only has one
picture clock frequency.

So the MPEG-4 form of short header temporal reference corresponds to
the old definition in H.263 version 1.

The H.263 picture clock frequency is specified within the bitstream
syntax.  It is not something negotiated only externally.  The
standard picture clock frequency of 30000/1001 Hz is used unless
the following is indicated:
  1) Bits 6-8 of PTYPE are set to '111' (indicating that PLUSPTYPE
     is present), and
  2) The UFEP parameter of PLUSPTYPE is '001' (indicating that OPPTYPE
     is present), and
  2) Bit 4 of OPPTYPE is set to '1'
In which case a "Custom Picture Clock Frequency Code (CPCFC)" is
present that specifies the picture clock frequency.

MPEG-4 does not allow bits 6-8 of PTYPE to be '111', so
it does not allow its short header mode to use custom picture
clock frequencies.

The reason that 25 Hz was not originally supported in baseline mode
is the famous/infamous "CIF compromise" originating in H.261 which
created a picture size that favored the 625/50 world but used it
with a picture clock frequency that favored the 525/60 world.
This created a single video format for use world-wide so that
videoconferencing systems made in all countries would be able to
communicate with each other seamlessly.

So the intent of the compromise was that there should not be
separate flavors of H.261 or H.263 for the 50 Hz and 60 Hz worlds.
Everyone should be able to communicate equally well with everyone.

Today there is somewhat less concern over requiring a decoder to be
able to rescale and retime video data than there was in the early
'90s when that compromise was struck.  So when H.263 went up for
its second (H.263+) revision completed in Sept. '97, more flexible
picture format support was added.  Myself and Samson Cheung (then
of Compression Labs, Inc.) were the primary contributors to adding
that flexibility in the second version.


+> 
+> As I noted in the first sentence, an alternative to describing how to
+> compute timestamp in short header mode is simply to change 
+> MAY to SHOULD or
+> SHALL regarding use of H.263 payload format for short header mode.


Perhaps "SHALL" is the appropriate thing unless there is some alternative
that also provides interoperability.

It should probably also include something like "(e.g., RFC 2190 or RFC
2429)", as
those are the two packetization formats that have been defined that can
carry
H.263 video.


+> 
+> Best regards,
+> Kris Huber
+> Sorenson Laboratories, Inc.
+> 
+> -----Original Message-----
+> From: Young-Kwon LIM [mailto:young@techway.co.kr]
+> Sent: Friday, September 08, 2000 1:08 AM
+> To: Franceschini Guido; Koji Imura; IETF AVT; Colin Perkins; 
+> 'Yoshihiro
+> Kikuchi'; Varesio Andrea
+> Cc: 4on2andIP-sys
+> Subject: Re: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt
+> 
+> 
+> Dear Guido and all,
+> 
+> > > 
+> > > It would be better to add a description to clarify this. 
+> How about
+> adding
+> > > the following description in the timestamp definition in 
+> section 3.1?
+> > > 
+> > > - When multiple VOPs are carried in the same RTP packet, 
+> the timestamp
+> > > indicates the earliest of the composition time within 
+> the VOPs carried
+> in
+> > > the RTP packet. Timestamp information of the rest of 
+> VOPs are derived
+> from
+> > > the timestamp fields in the VOP header (modulo_time_base and
+> > > vop_time_increment).?
+> > > 
+> > Are you sure you want to possibly indicate the CTS of the 
+> second VOP in a
+> > packet ?
+> > This seems to me quite difficult to manage
+> > I would better do either of the following:
+> > - always indicate the CTS of the first VOP in the packet, 
+> even if the
+> second
+> > VOP will be presented first
+> > - disallow concatenation of VOPs whose decoding order 
+> differs from the
+> > composition order
+> 
+> "modulo_time_base" and "vop_time_increment" are the syntax 
+> elements of VOP
+> header invented for the purpose of carrying relative timing 
+> information.
+> So, I don't see any problem to use those information to 
+> calcute CTS. When
+> the receiver get a RTP packet with multiple VOPs, the CTS of 
+> first VOP can
+> be derived from the RTP time stmap and CTS of successive VOPs can be
+> calculated by using time stamp and two syntax elements in VOP header.
+> Calculating timing information of each VOP is nothing new to 
+> the decoder.
+> 
+> > 
+> > I am also doubtful about allowing variable frame rate 
+> videos to take
+> > advantage of such concatenation of VOPs in the same RTP packet.
+> > IMHO it would be safer to add some more constraints in the 
+> specification
+> > (i.e. prohibit VOP concatenation in case of variable frame 
+> rates), instead
+> > of adding the complexity of extracting TS information 
+> through cooperation
+> of
+> > different layers.
+> > In other words: VOP concatenation has pros and cons.
+> > Pros: optimizes bandwidth consuption
+> > Cons: deteriorates packet loss resilience; increases implementation
+> > complexity.
+> > I therefore deduce that VOP concatenation is nice but not 
+> that wonderful,
+> > and I would suggest to allow VOP concatenation only when 
+> this is trivial
+> and
+> > does not require significant complexity in the receivers 
+> implementations.
+> > 
+> 
+> I don't think there is additional complexity. MPEG-4 video 
+> decoder always
+> calculate the time difference of current VOP from the last 
+> one.  So, when
+> the decoder push the reconstructed data to composition buffer, it can
+> easily send a presentation time information to the compositor. It is
+> nothing to the implementation.
+> 
+> Sincerely,
+> Young.
+> 



From rem-conf Fri Sep 08 22:16:44 2000 
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From: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
To: "Gary Sullivan" <garysull@microsoft.com>,
        "'Kris Huber'" <khuber@sorensonlabs.com>,
        "Young-Kwon LIM" <young@techway.co.kr>,
        "Franceschini Guido" <Guido.Franceschini@cselt.it>,
        "Koji Imura" <imura@telecom.mci.mei.co.jp>,
        "IETF AVT" <rem-conf@es.net>, "Colin Perkins" <colin@isi.edu>,
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Cc: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
References: <6CEF693DEA38994786B8F1B115897A2E220FE8@red-msg-06.redmond.corp.microsoft.com>
Subject: Re: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt - ti mestamp of short header VOP
Date: Sat, 9 Sep 2000 14:06:15 +0900
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Dear Kris, Gary, all,

I hope this message is not too late...

Thank you Kris for your clarification. I will incorporate your proposed
description for clarification in the Internet-Draft.

> Perhaps "SHALL" is the appropriate thing unless there is some alternative
> that also provides interoperability.

I think "SHALL" is too strong because there are definitions related to short
video header mode in other places of the Internet-Draft. The change may
cause interoperability problem with other payload format.

> It should probably also include something like "(e.g., RFC 2190 or RFC
> 2429)", as
> those are the two packetization formats that have been defined that can
> carry
> H.263 video.

Thank you for your suggestion for your editorial update.

Best regards,
Yoshihiro

----- Original Message -----
From: Gary Sullivan <garysull@microsoft.com>
To: 'Kris Huber' <khuber@sorensonlabs.com>; Young-Kwon LIM
<young@techway.co.kr>; Franceschini Guido <Guido.Franceschini@cselt.it>;
Koji Imura <imura@telecom.mci.mei.co.jp>; IETF AVT <rem-conf@es.net>; Colin
Perkins <colin@isi.edu>; 'Yoshihiro Kikuchi' <kiku@eel.rdc.toshiba.co.jp>;
Varesio Andrea <Andrea.Varesio@cselt.it>
Cc: 4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>
Sent: Saturday, September 09, 2000 6:44 AM
Subject: RE: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt - ti
mestamp of short header VOP


>
> Kris et al,
>
> A bit of H.263 explanation with a bit of a history lesson is
> spliced into Kris's quoted message below:
>
>
> +> -----Original Message-----
> +> From: Kris Huber [mailto:khuber@sorensonlabs.com]
> +> Sent: Friday, September 08, 2000 1:04 PM
> +> To: Young-Kwon LIM; Franceschini Guido; Koji Imura; IETF
> +> AVT; Colin Perkins; 'Yoshihiro Kikuchi'; Varesio Andrea
> +> Cc: 4on2andIP-sys; Gary Sullivan
> +> Subject: RE: IETF AVT last call on
> +> draft-ietf-avt-rtp-mpeg4-es-03.txt - ti mestamp of short header VOP
> +>
> +>
> +> Dear Young and all,
> +>
> +> Unless "MAY" is changed to "SHOULD" in "When the short video
> +> header mode is
> +> used, the RTP payload format used MAY be that specified for
> +> H.263 in the
> +> relevant RFCs or in other relevant standards,"  I think additional
> +> modifications are necessary to accommodate MPEG-4 short
> +> header mode.  This
> +> is because vop_time_increment_resolution, modulo_time_base,
> +> and time_code
> +> are not present in a short header elementary stream.  Here
> +> is my proposed
> +> modification (new text is indicated by surrounding it by
> +> "<<" and ">>").
> +> No deletions are required.
> +>
> +>    Timestamp: The timestamp indicates the composition time, or the
> +>    presentation time in a no-compositor decoder. A constant
> +> offset, which is
> +>    random, is added for security reasons. The detailed
> +> definition of the
> +>    timestamp is as follows:
> +> <<  - For a video object plane with short header, the
> +> timestamps (after the
> +> first random timestamp) are equal to the presentation time sequence
> +> associated with the semantics of the temporal_reference field.
> +> Specifically, each timestamp value SHALL be calculated by
> +> rounding the
> +> value of a precise clock that advances delta_time with each
> +> successive
> +> video object plane with short header.  The time increment SHOULD be
> +> calculated as delta_time = (((temporal_reference + 256 -
> +> (temporal_reference of previous VOP) modulo 256) *
> +> 1001/30000) for each
> +> successive video object plane with short header.  The RTP
> +> timestamp should
> +> be consistently rounded or truncated to the resolution of
> +> the RTP timestamp
> +> field.>>
> +>    - <<Otherwise,>> for a video object plane, it is defined as
> +> vop_time_increment (in units
> +>      of 1/vop_time_increment_resolution seconds) plus the
> +> cumulative number
> +>      of whole seconds specified by modulo_time_base and, if present,
> +>      time_code of Group_of_VideoObjectPlane() fields.
> +>    - In the case of interlaced video, a VOP will consist of
> +> lines from two
> +>      fields, and the timestamp will indicate the composition
> +> time of the
> +>      first field.
> +> (   - When multiple VOPs are carried in the same RTP packet,
> +> the timestamp
> +>      indicates the earliest of the composition time within
> +> the VOPs carried
> +>      in the RTP packet.)
> +> Young-Kwon's proposed modification of the above bullet:
> +>    - When multiple VOPs are carried in the same RTP packet,
> +> the timestamp
> +>      indicates the earliest of the composition time<<s>>
> +> within the VOPs
> +> carried in
> +>      the RTP packet.  Timestamp information of the rest of
> +> <<the>> VOPs are
> +> derived from
> +>      the timestamp fields in the VOP header (modulo_time_base and
> +>      vop_time_increment)<<, or from the temporal_reference field
> +>      in the case of short video header>>.
> +>    - If the RTP packet contains only configuration information and/or
> +>      Group_of_VideoObjectPlane() fields, the composition
> +> time of the next
> +>      VOP in the coding order is used.
> +>    - If the RTP packet contains only visual_object_sequence_end_code
> +>      information, the composition time of the immediately
> +> preceding VOP in
> +>      the coding order is used.
> +>
> +> Perhaps someone knows better wording for the first bullet.
> +> I used SHALL in
> +> that paragraph to prevent accumulation of round-off error in
> +> the sequence
> +> of time stamps, and SHOULD to allow for other picture clock
> +> frequencies
> +> than the one specified in the MPEG-4 visual semantics.  More
> +> detail about
> +> this difference in MPEG-4 short header and H.263 semantics follows:
> +>
> +> Quoting ISO/IEC 14496-2:1999, clause 6.3.5.2:  "temporal
> +> reference:  This
> +> is an 8-bit number which can have 256 possible values.  It
> +> is formed by
> +> incrementing its value in the previously transmitted
> +> video_plane_with_short_header() by one plus the number of
> +> non-transmitted
> +> pictures (at 30000/1001 Hz) since the previously transmitted
> +> picture.  The
> +> arithmetic is performed with only the eight LSBs."
> +>
> +> In contrast the TR field in the H.263 spec allows different
> +> clocks than
> +> 29.97 Hz:  "the value of TR is formed by incrementing its
> +> value in the
> +> temporally-previous reference picture header by one plus the
> +> number of
> +> skipped or non-reference pictures at the picture clock
> +> frequency since the
> +> previously transmitted one.  The interpretation of TR
> +> depends upon the
> +> active picture clock frequency. ..."  Negotiation of the
> +> picture clock
> +> frequency is done by external means, I believe.
> +>
> +> I suspect that fixing the H.263 TR frequency in the MPEG-4
> +> temporal_reference semantics causes problems particularly
> +> for use of short-
> +> header on 25 Hz material in Europe.
>
>
> MPEG-4's short header mode corresponds with the H.263 "baseline"
> profile of operation.  The baseline form only supports one picture
> clock frequency -- 30000/1001 Hz.  The extra flexibility of
> supporting other picture clock frequencies is an optional
> feature that was added in version 2.  MPEG-4 does not support
> compatibility for any optional H.263 optional features (even for the
> Advanced Prediction mode, which would have been a very easy one
> to specify since all the tools are in MPEG-4 in an identical form).
> It only supports the baseline form.  The baseline form only has one
> picture clock frequency.
>
> So the MPEG-4 form of short header temporal reference corresponds to
> the old definition in H.263 version 1.
>
> The H.263 picture clock frequency is specified within the bitstream
> syntax.  It is not something negotiated only externally.  The
> standard picture clock frequency of 30000/1001 Hz is used unless
> the following is indicated:
>   1) Bits 6-8 of PTYPE are set to '111' (indicating that PLUSPTYPE
>      is present), and
>   2) The UFEP parameter of PLUSPTYPE is '001' (indicating that OPPTYPE
>      is present), and
>   2) Bit 4 of OPPTYPE is set to '1'
> In which case a "Custom Picture Clock Frequency Code (CPCFC)" is
> present that specifies the picture clock frequency.
>
> MPEG-4 does not allow bits 6-8 of PTYPE to be '111', so
> it does not allow its short header mode to use custom picture
> clock frequencies.
>
> The reason that 25 Hz was not originally supported in baseline mode
> is the famous/infamous "CIF compromise" originating in H.261 which
> created a picture size that favored the 625/50 world but used it
> with a picture clock frequency that favored the 525/60 world.
> This created a single video format for use world-wide so that
> videoconferencing systems made in all countries would be able to
> communicate with each other seamlessly.
>
> So the intent of the compromise was that there should not be
> separate flavors of H.261 or H.263 for the 50 Hz and 60 Hz worlds.
> Everyone should be able to communicate equally well with everyone.
>
> Today there is somewhat less concern over requiring a decoder to be
> able to rescale and retime video data than there was in the early
> '90s when that compromise was struck.  So when H.263 went up for
> its second (H.263+) revision completed in Sept. '97, more flexible
> picture format support was added.  Myself and Samson Cheung (then
> of Compression Labs, Inc.) were the primary contributors to adding
> that flexibility in the second version.
>
>
> +>
> +> As I noted in the first sentence, an alternative to describing how to
> +> compute timestamp in short header mode is simply to change
> +> MAY to SHOULD or
> +> SHALL regarding use of H.263 payload format for short header mode.
>
>
> Perhaps "SHALL" is the appropriate thing unless there is some alternative
> that also provides interoperability.
>
> It should probably also include something like "(e.g., RFC 2190 or RFC
> 2429)", as
> those are the two packetization formats that have been defined that can
> carry
> H.263 video.
>
>
> +>
> +> Best regards,
> +> Kris Huber
> +> Sorenson Laboratories, Inc.
> +>
> +> -----Original Message-----
> +> From: Young-Kwon LIM [mailto:young@techway.co.kr]
> +> Sent: Friday, September 08, 2000 1:08 AM
> +> To: Franceschini Guido; Koji Imura; IETF AVT; Colin Perkins;
> +> 'Yoshihiro
> +> Kikuchi'; Varesio Andrea
> +> Cc: 4on2andIP-sys
> +> Subject: Re: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt
> +>
> +>
> +> Dear Guido and all,
> +>
> +> > >
> +> > > It would be better to add a description to clarify this.
> +> How about
> +> adding
> +> > > the following description in the timestamp definition in
> +> section 3.1?
> +> > >
> +> > > - When multiple VOPs are carried in the same RTP packet,
> +> the timestamp
> +> > > indicates the earliest of the composition time within
> +> the VOPs carried
> +> in
> +> > > the RTP packet. Timestamp information of the rest of
> +> VOPs are derived
> +> from
> +> > > the timestamp fields in the VOP header (modulo_time_base and
> +> > > vop_time_increment).?
> +> > >
> +> > Are you sure you want to possibly indicate the CTS of the
> +> second VOP in a
> +> > packet ?
> +> > This seems to me quite difficult to manage
> +> > I would better do either of the following:
> +> > - always indicate the CTS of the first VOP in the packet,
> +> even if the
> +> second
> +> > VOP will be presented first
> +> > - disallow concatenation of VOPs whose decoding order
> +> differs from the
> +> > composition order
> +>
> +> "modulo_time_base" and "vop_time_increment" are the syntax
> +> elements of VOP
> +> header invented for the purpose of carrying relative timing
> +> information.
> +> So, I don't see any problem to use those information to
> +> calcute CTS. When
> +> the receiver get a RTP packet with multiple VOPs, the CTS of
> +> first VOP can
> +> be derived from the RTP time stmap and CTS of successive VOPs can be
> +> calculated by using time stamp and two syntax elements in VOP header.
> +> Calculating timing information of each VOP is nothing new to
> +> the decoder.
> +>
> +> >
> +> > I am also doubtful about allowing variable frame rate
> +> videos to take
> +> > advantage of such concatenation of VOPs in the same RTP packet.
> +> > IMHO it would be safer to add some more constraints in the
> +> specification
> +> > (i.e. prohibit VOP concatenation in case of variable frame
> +> rates), instead
> +> > of adding the complexity of extracting TS information
> +> through cooperation
> +> of
> +> > different layers.
> +> > In other words: VOP concatenation has pros and cons.
> +> > Pros: optimizes bandwidth consuption
> +> > Cons: deteriorates packet loss resilience; increases implementation
> +> > complexity.
> +> > I therefore deduce that VOP concatenation is nice but not
> +> that wonderful,
> +> > and I would suggest to allow VOP concatenation only when
> +> this is trivial
> +> and
> +> > does not require significant complexity in the receivers
> +> implementations.
> +> >
> +>
> +> I don't think there is additional complexity. MPEG-4 video
> +> decoder always
> +> calculate the time difference of current VOP from the last
> +> one.  So, when
> +> the decoder push the reconstructed data to composition buffer, it can
> +> easily send a presentation time information to the compositor. It is
> +> nothing to the implementation.
> +>
> +> Sincerely,
> +> Young.
> +>
>




From rem-conf Sat Sep 09 04:15:37 2000 
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From: "Jiro Katto" <katto@katto.comm.waseda.ac.jp>
To: "Franceschini Guido" <Guido.Franceschini@CSELT.IT>,
        "'Yoshihiro Kikuchi'" <kiku@eel.rdc.toshiba.co.jp>,
        "IETF AVT" <rem-conf@es.net>
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References: <85077463E51D6A498C453073E1677940034EAC@exc2k02.cselt.it>
Subject: Re: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt
Date: Sat, 9 Sep 2000 20:15:52 +0900
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Dear all,

Current mechanism seems to have two problems:
(1) When M-bit is 1, an RTP handler has to parse all the bits every time
just to
find VOP headers (no mechanisms to indicate the number of VOPs).
(2) Encoder manufacturers may not precisely implement the two fields,
modulo_
time_base and vop_time_increment, similar to MPEG-2 case.

Therefore, I prefer just to say, "Two or more VOPs SHALL be fragmented into
different RTP packets" in the current draft. This is simple and stable.

An alternative, but may not be MPEG-4 specific, is to put the number of VOPs
and time stamps into a payload header using header extension mechanism.
Syntax
will be type, length (= number of VOPs -1), time stamps and padding bytes,
similar to other protocols' option fields. Time stamp can be shrinked to one
byte
by taking differences.

Just a comment. Thanks.

Best regards,
--
Jiro Katto
Waseda University

Guido wrote:
> Some further comment below
>
> > ----------
> > From: Yoshihiro Kikuchi[SMTP:kiku@eel.rdc.toshiba.co.jp]
> > Sent: Friday, September 08, 2000 2:35 PM
> > To: Young-Kwon LIM; Franceschini Guido; Koji Imura; IETF AVT; Colin
> > Perkins; Varesio Andrea
> > Cc: 4on2andIP-sys; Yoshihiro Kikuchi
> > Subject: Re: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt
> >
> > Dear Young-Kwon, Guido, Matsui-san, Imura-san, all,
> >
> > Thank you very much for the comment.
> >
> > It seems to be better to list all description related the multiple VOPs
> > concatenation in the "current" draft text posted on Sep. 5 here in this
> > mail
> > for the better understandings.
> >
> > 1. Introduction
> >    The fragmentation rule recommends not to map more than one VOP in an
> > RTP
> >    packet so that RTP timestamp uniquely indicates the VOP time framing.
> > On
> >    the other hand, MPEG-4 video may generate VOPs of very small size, in
> >    cases with a not coded VOP containing only VOP header or an arbitrary
> >    shaped VOP with a small number. To reduce the overhead for such
cases,
> >    the fragmentation rule permits concatenating multiple VOPs in an RTP
> >    packet when the VOP size is small.
> Editorial: I don't think the last statement 'when the VOP size is small'
is
> necessary/appropriate
>
> > 3.1 Use of RTP header fields for MPEG-4 Visual
> >    Marker (M) bit: The marker bit is set to one to indicate the last RTP
> >    packet (or only RTP packet) of a VOP. When multiple VOPs are carried
in
> >    the same RTP packet, the marker bit is set to 1.
> >
> >    Timestamp:
> >    - When multiple VOPs are carried in the same RTP packet, the
timestamp
> >      indicates the earliest of the composition time within the VOPs
> > carried
> >      in the RTP packet.
> >
> > 3.2 Fragmentation of MPEG-4 Visual bitstream
> >    (4) Two or more VOPs SHOULD be fragmented into different RTP packets
so
> >    that one RTP packet consists of the data bytes associated with a
unique
> >    presentation time (that is indicated in the timestamp field in the
RTP
> >    packet header), with the exception that multiple VOPs MAY be carried
> >    within one RTP packet if the size of the VOPs is small.
> Editorial: again, what is the meaning of 'small' ? Also, at the beginning,
I
> would change to 'Different VOPs SHOULD be fragmented into different RTP
> packets'
>
>
> > I think it is better to add cross-references (see section ...,
definition
> > of
> > ....) for each description above in the draft so that every readers of
the
> > draft can understand this set of rule easily.
> >
> > Young-Kwon wrote:
> >
> > > I don't think there is additional complexity. MPEG-4 video decoder
> > always
> > calculate the time difference of current VOP from the last one.  So,
when
> > the decoder push the reconstructed data to composition buffer, it can
> > easily
> > send a presentation time information to the compositor. It is nothing to
> > the
> > implementation.
> >
> > > I'm OK with this because this allows concatenation of VOPs.
> >
> > Oh, now there are agreements on the current description. As usual, if
> > there
> > is any other strong disagreements, we should stick with the current
text.
> > (except for editorial changes and clarifications)
> >
> > Please look not only sentences I posted but a set of rules. It is
defined
> > that multiple VOPs SHOULD NOT be concatenated so as not to force huge
> > amout
> > of complexity. The concatenation is an exception rule only when VOP size
> > is
> > small.
> Just to make my point clear: either you quantify 'small', or you can at
most
> say something like: "concatenation is allowed only when an integral number
> of VOPs fit into a single RTP packet"
> I would suggest to have some text clearly specifying under which
> circumstances VOP concatenation may occur:
> Some idea for this text:
>
> Concatenation is allowed only when the following conditions occur:
> - the result of concatenting N consecutive VOPs still fits into a single
RTP
> packet
> - the presentation and decoding order of the VOPs is identical
> - ...
>
> > >For more clarification I'd like to add "When multiple VOPs are carried
in
> > one RTP payload, the presentation time of the VOPs after the first one
may
> > be calculated by the decoder."
> > >
> > Agreed.
> >
> > > BTW, where is a sentence prohibiting concatenation of VOPs with
> > different
> > decoding order?
> >
> > There is no sentence now in the draft but only in mails exchanged.
> >
> > Following is the list of proposed changes by mails on this issue for
> > editorial update and clarifications but not technical.
> >
> > 1) Add cross-cross references to each descriptions.
> > 2) Add the following sentence to the timestamp definition
> >     "Timestamp information of the rest of VOPs are derived from the
> > timestamp fields in the VOP header (modulo_time_base and
> > vop_time_increment)."
> > 3) Add the following description as a note to fragmentation rule (4)
> >     "When multiple VOPs are carried in one RTP payload, the presentation
> > time of the VOPs after the first one may be calculated by the decoder."
> >     "This operation is necessary only for RTP packets satisfying the
> > following both two conditions:
> > - Marker bit equal to one, and
> >   (Because marker bit in section 3.1 is defined that it is set to one
when
> > multiple VOP is concatenated.)
> > - The beginning of RTP payload is not a start code.
> >   (Because the fragmentation rule is defined that a start code is placed
> > always at the beginning of RTP payload, if exists.)"
> I do not understand this second condition. I would have reversed it "The
> beginning of RTP payload corresponds to a start code"
>
> Bye
> Guido
>
>
> > If there is no further objection, only changes reflected to the next
> > version
> > of I/D will be these descriptions for editorial update and
clarifications.
> >
> >
> > Best regards,
> > Yoshihiro
> >
> > ----
> >         Yoshihiro Kikuchi
> >
> > E-mail: kiku@eel.rdc.toshiba.co.jp
> > Corporate Research and Development Center
> > TOSHIBA Corporation
> > TEL: +81 +44 549 2288    FAX: +81 +44 520 1267
> >
> >
>




From rem-conf Sat Sep 09 04:36:29 2000 
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Date: Sat, 09 Sep 2000 13:32:24 +0200
From: Chiariglione Leonardo <Leonardo.Chiariglione@CSELT.IT>
Subject: RE: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt - ti
	mestamp of short header VOP
To: 'Gary Sullivan' <garysull@microsoft.com>,
 'Kris Huber' <khuber@sorensonlabs.com>, Young-Kwon LIM <young@techway.co.kr>,
 Franceschini Guido <Guido.Franceschini@cselt.it>,
 Koji Imura <imura@telecom.mci.mei.co.jp>, IETF AVT <rem-conf@es.net>,
 Colin Perkins <colin@isi.edu>,
 'Yoshihiro Kikuchi' <kiku@eel.rdc.toshiba.co.jp>,
 Varesio Andrea <Andrea.Varesio@cselt.it>
Cc: 4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>
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>Today there is somewhat less concern over requiring a decoder to be
>able to rescale and retime video data than there was in the early
'>90s when that compromise was struck.

Actually the "compromise" was reached at the Specialists Group on =
Visual
Telephony meeting of Torino in September 1985. At that meeting I was =
the
only one opposing CIF because I thought accommodating different video =
format
in the coded domain should have been a matter for the decoder to =
handle. But
the Specialists Group at that time was still dominated by people =
focusing on
the "encoder" instead of the "decoder". My position of that time was
belatedly vindicated in 1990 when MPEG adopted SIF and later the =
Constrained
Parameter Set, the obvious thing to do when you work on the decoder. I =
have
to say that I did not play any role in defining CIF or CPS.
Leonardo


> -----Original Message-----
> From: Gary Sullivan [mailto:garysull@microsoft.com]
> Sent: venerd=EC 8 settembre 2000 23.44
> To: 'Kris Huber'; Young-Kwon LIM; Franceschini Guido; Koji Imura; =
IETF
> AVT; Colin Perkins; 'Yoshihiro Kikuchi'; Varesio Andrea
> Cc: 4on2andIP-sys
> Subject: RE: IETF AVT last call on=20
> draft-ietf-avt-rtp-mpeg4-es-03.txt -
> ti mestamp of short header VOP
>=20
>=20
>=20
> Kris et al,
>=20
> A bit of H.263 explanation with a bit of a history lesson is
> spliced into Kris's quoted message below:
>=20
>=20
> +> -----Original Message-----
> +> From: Kris Huber [mailto:khuber@sorensonlabs.com]
> +> Sent: Friday, September 08, 2000 1:04 PM
> +> To: Young-Kwon LIM; Franceschini Guido; Koji Imura; IETF=20
> +> AVT; Colin Perkins; 'Yoshihiro Kikuchi'; Varesio Andrea
> +> Cc: 4on2andIP-sys; Gary Sullivan
> +> Subject: RE: IETF AVT last call on=20
> +> draft-ietf-avt-rtp-mpeg4-es-03.txt - ti mestamp of short header =
VOP
> +>=20
> +>=20
> +> Dear Young and all,
> +>=20
> +> Unless "MAY" is changed to "SHOULD" in "When the short video=20
> +> header mode is
> +> used, the RTP payload format used MAY be that specified for=20
> +> H.263 in the
> +> relevant RFCs or in other relevant standards,"  I think additional
> +> modifications are necessary to accommodate MPEG-4 short=20
> +> header mode.  This
> +> is because vop_time_increment_resolution, modulo_time_base,=20
> +> and time_code
> +> are not present in a short header elementary stream.  Here=20
> +> is my proposed
> +> modification (new text is indicated by surrounding it by=20
> +> "<<" and ">>").
> +> No deletions are required.
> +>=20
> +>    Timestamp: The timestamp indicates the composition time, or the
> +>    presentation time in a no-compositor decoder. A constant=20
> +> offset, which is
> +>    random, is added for security reasons. The detailed=20
> +> definition of the
> +>    timestamp is as follows:
> +> <<  - For a video object plane with short header, the=20
> +> timestamps (after the
> +> first random timestamp) are equal to the presentation time =
sequence
> +> associated with the semantics of the temporal_reference field.
> +> Specifically, each timestamp value SHALL be calculated by=20
> +> rounding the
> +> value of a precise clock that advances delta_time with each=20
> +> successive
> +> video object plane with short header.  The time increment SHOULD =
be
> +> calculated as delta_time =3D (((temporal_reference + 256 -
> +> (temporal_reference of previous VOP) modulo 256) *=20
> +> 1001/30000) for each
> +> successive video object plane with short header.  The RTP=20
> +> timestamp should
> +> be consistently rounded or truncated to the resolution of=20
> +> the RTP timestamp
> +> field.>>
> +>    - <<Otherwise,>> for a video object plane, it is defined as
> +> vop_time_increment (in units
> +>      of 1/vop_time_increment_resolution seconds) plus the=20
> +> cumulative number
> +>      of whole seconds specified by modulo_time_base and,=20
> if present,
> +>      time_code of Group_of_VideoObjectPlane() fields. =20
> +>    - In the case of interlaced video, a VOP will consist of=20
> +> lines from two
> +>      fields, and the timestamp will indicate the composition=20
> +> time of the
> +>      first field.
> +> (   - When multiple VOPs are carried in the same RTP packet,=20
> +> the timestamp
> +>      indicates the earliest of the composition time within=20
> +> the VOPs carried
> +>      in the RTP packet.)
> +> Young-Kwon's proposed modification of the above bullet:
> +>    - When multiple VOPs are carried in the same RTP packet,=20
> +> the timestamp
> +>      indicates the earliest of the composition time<<s>>=20
> +> within the VOPs
> +> carried in
> +>      the RTP packet.  Timestamp information of the rest of=20
> +> <<the>> VOPs are
> +> derived from
> +>      the timestamp fields in the VOP header (modulo_time_base and
> +>      vop_time_increment)<<, or from the temporal_reference field
> +>      in the case of short video header>>.
> +>    - If the RTP packet contains only configuration=20
> information and/or
> +>      Group_of_VideoObjectPlane() fields, the composition=20
> +> time of the next
> +>      VOP in the coding order is used.
> +>    - If the RTP packet contains only=20
> visual_object_sequence_end_code
> +>      information, the composition time of the immediately=20
> +> preceding VOP in
> +>      the coding order is used.
> +>=20
> +> Perhaps someone knows better wording for the first bullet. =20
> +> I used SHALL in
> +> that paragraph to prevent accumulation of round-off error in=20
> +> the sequence
> +> of time stamps, and SHOULD to allow for other picture clock=20
> +> frequencies
> +> than the one specified in the MPEG-4 visual semantics.  More=20
> +> detail about
> +> this difference in MPEG-4 short header and H.263 semantics =
follows:
> +>=20
> +> Quoting ISO/IEC 14496-2:1999, clause 6.3.5.2:  "temporal=20
> +> reference:  This
> +> is an 8-bit number which can have 256 possible values.  It=20
> +> is formed by
> +> incrementing its value in the previously transmitted
> +> video_plane_with_short_header() by one plus the number of=20
> +> non-transmitted
> +> pictures (at 30000/1001 Hz) since the previously transmitted=20
> +> picture.  The
> +> arithmetic is performed with only the eight LSBs."
> +>=20
> +> In contrast the TR field in the H.263 spec allows different=20
> +> clocks than
> +> 29.97 Hz:  "the value of TR is formed by incrementing its=20
> +> value in the
> +> temporally-previous reference picture header by one plus the=20
> +> number of
> +> skipped or non-reference pictures at the picture clock=20
> +> frequency since the
> +> previously transmitted one.  The interpretation of TR=20
> +> depends upon the
> +> active picture clock frequency. ..."  Negotiation of the=20
> +> picture clock
> +> frequency is done by external means, I believe.
> +>=20
> +> I suspect that fixing the H.263 TR frequency in the MPEG-4
> +> temporal_reference semantics causes problems particularly=20
> +> for use of short-
> +> header on 25 Hz material in Europe.
>=20
>=20
> MPEG-4's short header mode corresponds with the H.263 "baseline"
> profile of operation.  The baseline form only supports one picture
> clock frequency -- 30000/1001 Hz.  The extra flexibility of
> supporting other picture clock frequencies is an optional
> feature that was added in version 2.  MPEG-4 does not support
> compatibility for any optional H.263 optional features (even for the
> Advanced Prediction mode, which would have been a very easy one
> to specify since all the tools are in MPEG-4 in an identical form).
> It only supports the baseline form.  The baseline form only has one
> picture clock frequency.
>=20
> So the MPEG-4 form of short header temporal reference corresponds to
> the old definition in H.263 version 1.
>=20
> The H.263 picture clock frequency is specified within the bitstream
> syntax.  It is not something negotiated only externally.  The
> standard picture clock frequency of 30000/1001 Hz is used unless
> the following is indicated:
>   1) Bits 6-8 of PTYPE are set to '111' (indicating that PLUSPTYPE
>      is present), and
>   2) The UFEP parameter of PLUSPTYPE is '001' (indicating that =
OPPTYPE
>      is present), and
>   2) Bit 4 of OPPTYPE is set to '1'
> In which case a "Custom Picture Clock Frequency Code (CPCFC)" is
> present that specifies the picture clock frequency.
>=20
> MPEG-4 does not allow bits 6-8 of PTYPE to be '111', so
> it does not allow its short header mode to use custom picture
> clock frequencies.
>=20
> The reason that 25 Hz was not originally supported in baseline mode
> is the famous/infamous "CIF compromise" originating in H.261 which
> created a picture size that favored the 625/50 world but used it
> with a picture clock frequency that favored the 525/60 world.
> This created a single video format for use world-wide so that
> videoconferencing systems made in all countries would be able to
> communicate with each other seamlessly.
>=20
> So the intent of the compromise was that there should not be
> separate flavors of H.261 or H.263 for the 50 Hz and 60 Hz worlds.
> Everyone should be able to communicate equally well with everyone.
>=20
> Today there is somewhat less concern over requiring a decoder to be
> able to rescale and retime video data than there was in the early
> '90s when that compromise was struck.  So when H.263 went up for
> its second (H.263+) revision completed in Sept. '97, more flexible
> picture format support was added.  Myself and Samson Cheung (then
> of Compression Labs, Inc.) were the primary contributors to adding
> that flexibility in the second version.
>=20
>=20
> +>=20
> +> As I noted in the first sentence, an alternative to=20
> describing how to
> +> compute timestamp in short header mode is simply to change=20
> +> MAY to SHOULD or
> +> SHALL regarding use of H.263 payload format for short header mode.
>=20
>=20
> Perhaps "SHALL" is the appropriate thing unless there is some=20
> alternative
> that also provides interoperability.
>=20
> It should probably also include something like "(e.g., RFC 2190 or =
RFC
> 2429)", as
> those are the two packetization formats that have been=20
> defined that can
> carry
> H.263 video.
>=20
>=20
> +>=20
> +> Best regards,
> +> Kris Huber
> +> Sorenson Laboratories, Inc.
> +>=20
> +> -----Original Message-----
> +> From: Young-Kwon LIM [mailto:young@techway.co.kr]
> +> Sent: Friday, September 08, 2000 1:08 AM
> +> To: Franceschini Guido; Koji Imura; IETF AVT; Colin Perkins;=20
> +> 'Yoshihiro
> +> Kikuchi'; Varesio Andrea
> +> Cc: 4on2andIP-sys
> +> Subject: Re: IETF AVT last call on=20
> draft-ietf-avt-rtp-mpeg4-es-03.txt
> +>=20
> +>=20
> +> Dear Guido and all,
> +>=20
> +> > >=20
> +> > > It would be better to add a description to clarify this.=20
> +> How about
> +> adding
> +> > > the following description in the timestamp definition in=20
> +> section 3.1?
> +> > >=20
> +> > > - When multiple VOPs are carried in the same RTP packet,=20
> +> the timestamp
> +> > > indicates the earliest of the composition time within=20
> +> the VOPs carried
> +> in
> +> > > the RTP packet. Timestamp information of the rest of=20
> +> VOPs are derived
> +> from
> +> > > the timestamp fields in the VOP header (modulo_time_base and
> +> > > vop_time_increment).?
> +> > >=20
> +> > Are you sure you want to possibly indicate the CTS of the=20
> +> second VOP in a
> +> > packet ?
> +> > This seems to me quite difficult to manage
> +> > I would better do either of the following:
> +> > - always indicate the CTS of the first VOP in the packet,=20
> +> even if the
> +> second
> +> > VOP will be presented first
> +> > - disallow concatenation of VOPs whose decoding order=20
> +> differs from the
> +> > composition order
> +>=20
> +> "modulo_time_base" and "vop_time_increment" are the syntax=20
> +> elements of VOP
> +> header invented for the purpose of carrying relative timing=20
> +> information.
> +> So, I don't see any problem to use those information to=20
> +> calcute CTS. When
> +> the receiver get a RTP packet with multiple VOPs, the CTS of=20
> +> first VOP can
> +> be derived from the RTP time stmap and CTS of successive=20
> VOPs can be
> +> calculated by using time stamp and two syntax elements in=20
> VOP header.
> +> Calculating timing information of each VOP is nothing new to=20
> +> the decoder.
> +>=20
> +> >=20
> +> > I am also doubtful about allowing variable frame rate=20
> +> videos to take
> +> > advantage of such concatenation of VOPs in the same RTP packet.
> +> > IMHO it would be safer to add some more constraints in the=20
> +> specification
> +> > (i.e. prohibit VOP concatenation in case of variable frame=20
> +> rates), instead
> +> > of adding the complexity of extracting TS information=20
> +> through cooperation
> +> of
> +> > different layers.
> +> > In other words: VOP concatenation has pros and cons.
> +> > Pros: optimizes bandwidth consuption
> +> > Cons: deteriorates packet loss resilience; increases=20
> implementation
> +> > complexity.
> +> > I therefore deduce that VOP concatenation is nice but not=20
> +> that wonderful,
> +> > and I would suggest to allow VOP concatenation only when=20
> +> this is trivial
> +> and
> +> > does not require significant complexity in the receivers=20
> +> implementations.
> +> >=20
> +>=20
> +> I don't think there is additional complexity. MPEG-4 video=20
> +> decoder always
> +> calculate the time difference of current VOP from the last=20
> +> one.  So, when
> +> the decoder push the reconstructed data to composition=20
> buffer, it can
> +> easily send a presentation time information to the=20
> compositor. It is
> +> nothing to the implementation.
> +>=20
> +> Sincerely,
> +> Young.
> +>=20
>=20



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=================================== 2000.09.8 s:0001==
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
To: "'Guler, Amos'" <Amos_Guler@icomverse.com>,
        "'rem-conf@es.net'"
	 <rem-conf@es.net>
Subject: RE: Question: packing several frames in a single RTP packet
Date: Sun, 10 Sep 2000 02:58:32 -0400
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This may be of theoretical interest only. In practice, I think that all
frame based codecs are either (1) the same frame length, (2) indicate within
the frame itself what size or type it is, when the lengths are not the same.
Its only for frames that have variable length AND don't indicate their
length that you  need worry about this. Does anyone know of any frame codecs
which are a problem?

-Jonathan R.


Amos wrote:
> -----Original Message-----
> Hello all,
> 
> (I hope I am sending this to the correct mailing list. If 
> not, please accept
> my apology and advise me of the correct one)
> 
> In the new draft "RTP Profile for Audio and Video Conferences 
> with Minimal
> Control"
> (http://www.ietf.org/internet-drafts/draft-ietf-avt-profile-ne
> w-09.txt) that
> is supposed to replace RFC-1890, it says (in section 4.4, 
> Guidelines for
> Frame-Based Audio Encodings):
> 
> "The receiver can tell the number of frames contained in an 
> RTP packet, if
> all the frames have the same length, by dividing the RTP 
> payload length by
> the audio frame size which is defined as part of the 
> encoding. This does not
> work when carrying frames of different sizes unless the frame 
> sizes are
> relatively prime."
> 
> My question concerns the last statement: suppose, for 
> example, that the
> frame sizes are 5 and 7 octets, which are relatively prime, 
> and the receiver
> gets a 35-octet packet. How can he/she tell whether the 
> packet contains five
> 7-octet frames or seven 5-octet frames?
> 
> 
> Amos Guler
> 
> ==================== 
> Amos Guler 
> Comverse Network Systems Ltd.
> amos_guler@icomverse.com 
> http://www.comverse.com 
> 
> 
> 



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From rem-conf Sun Sep 10 02:56:09 2000 
From rem-conf-request@es.net Sun Sep 10 02:56:08 2000
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From: Orion Hodson <O.Hodson@cs.ucl.ac.uk>
X-Organisation: University College London, CS Dept.
X-Phone: +44 (0)20 7679 3704
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
cc: "'Guler, Amos'" <Amos_Guler@icomverse.com>, 
    "'rem-conf@es.net'" <rem-conf@es.net>
Subject: Re: Question: packing several frames in a single RTP packet
In-reply-to: Your message of "Sun, 10 Sep 2000 02:58:32 EDT." <B65B4F8437968F488A01A940B21982BF21FC33@DYN-EXCH-001.dynamicsoft.com>
Date: Sun, 10 Sep 2000 10:50:22 +0100
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<B65B4F8437968F488A01A940B21982BF21FC33@DYN-EXCH-001.dynamicsoft.com>Jonathan R
osenberg writes:
> This may be of theoretical interest only. In practice, I think that all
> frame based codecs are either (1) the same frame length, (2) indicate within
> the frame itself what size or type it is, when the lengths are not the same.
> Its only for frames that have variable length AND don't indicate their
> length that you  need worry about this. Does anyone know of any frame codecs
> which are a problem?
 
VDVI is variable length and has no size indicator.  Since it's a very
low cost codec this isn't a major deal - it's 160 16-entry table
searches per frame.  It's slightly unusual because it is a sample
based codec cast that's effectively cast into a frame based format for
unambiguous decoding.  I can't think of any other's off hand.

- Orion.



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To: jan.vandermeer@philips.com
cc: stewe@cs.tu-berlin.de, rem-conf@es.net,
        4on2andIP-sys@advent.ee.columbia.edu
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
In-Reply-To: Your message of "Mon, 11 Sep 2000 13:03:46 +0200."
             <0056890013989833000002L932*@MHS> 
Date: Mon, 11 Sep 2000 11:22:27 -0400
From: Colin Perkins <csp@east.isi.edu>
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--> jan.vandermeer@philips.com writes:
>Stephan, Colin,
>
>Stephan wrote:
>
>Regardless how the generic payload format of the future may look like,
>systems can be most easily designed to support both payload types.
>
>True. But the issue is different. Mobile devices that are build to support
>the current MPEG-4 ES draft will not be capable to decode RTP payloads with
>the additional info flagged by the first byte in the payload. (Note that the
>additional info would be needed by other applications that include usage of
>the same stream)
>
>In the past of MPEG, the use of "reserved fields" proved extremely powerful.
>To not allow them here would be a failure in my opinion. Of course it would
>be best to avoid unnecessary overhead. Therefore it would be most attractive
>if the inclusion of the additional field is indicated, for example by a flag
>or by a parameter that goes with the payload type. I understand that in the
>RTP header no such flag is available, but that the use of a parameter
>associated with the payload type would be possible. Would that be a good way
>to solve this ?

In RTP, that parameter is the payload type field. 

I really don't understand why this additional field will add value. It 
allows for systems to skip data which they don't understand, how is this
different from using the payload type for the same effect?

Colin



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To: <csp@east.isi.edu>, <stewe@cs.tu-berlin.de>
Cc: <rem-conf@es.net>, <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the
         last call
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Stephan, Colin,

Stephan wrote:

Regardless how the generic payload format of the future may look like,
systems can be most easily designed to support both payload types.

True. But the issue is different. Mobile devices that are build to supp=
ort
the current MPEG-4 ES draft will not be capable to decode RTP payloads =
with
the additional info flagged by the first byte in the payload. (Note tha=
t the
additional info would be needed by other applications that include usag=
e of
the same stream)

In the past of MPEG, the use of "reserved fields" proved extremely powe=
rful.
To not allow them here would be a failure in my opinion. Of course it w=
ould
be best to avoid unnecessary overhead. Therefore it would be most attra=
ctive
if the inclusion of the additional field is indicated, for example by a=
 flag
or by a parameter that goes with the payload type. I understand that in=
 the
RTP header no such flag is available, but that the use of a parameter
associated with the payload type would be possible. Would that be a goo=
d way
to solve this ?

Regards,

Jan

************************
Jan van der Meer
Philips - Digital Networks
Building OAN-4
P.O. Box 80002
5600 JB Eindhoven
The Netherlands
Phone +31 402 735 774
Fax +31 402 735 545
Email jan.vandermeer@philips.com
************************


-----Original Message-----
From:	stewe@cs.tu-berlin.de [mailto:stewe@cs.tu-berlin.de]
Sent:	vrijdag, 08 september, 2000 13:05
To:	young@techway.co.kr
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response=
 to
the last call


Folks,

>1.  We suggest to have a room for future extension of RTP header (e.g.=
 one
>byte of payload header at the beginning of the payload part) to make t=
he
>devices implementing this draft interoperable with the bitstream based=
 on
>the generic format under development. For example, it is possible to u=
se
>the reserved byte to specify the length of the payload header of gener=
ic
>payload format. Then the device implementing this draft can play the
>bitstream of generic payload format by simply skipping those bytes.

I'm not sure whether this is a good idea.  We would waste one byte of
payload (from RTP's point of view) for something which is probably not
used by any systems currently under discussion or development.

Regardless how the generic payload format of the future may look like,
systems can be most easily designed to support both payload types.
RTP's PT mechanism has enough numbering space to support both.
Capability announcement/negotiation will be different and have to
be redesigned anyway for anything more capable then what is
currently in the MPEG-4 ES draft.  Annoucing/Negotiating another
PT for the new generic payload spec will not make a difference.

When, in the future, a generic payload type would be available that
includes a super set of functionality of the MPEG-4 ES payload format,
then we could declare the MPEG-4 ES RFC as obsolete, thereby
advising people to move over to the new, generic payload type.

Or, it may also be possible to have two payload specs coexisting,
one for simple audio/video ES packetization only, the other containing
more (the rest of?) functionality of MPEG-4.  This does not sound
like a bad idea to me, because the necessary demands of functionality
both from the network (in terms of QoS) and implementation efforts
(single audio/visual ESs vs. system layer, flexmux, whatever...) are
very different.

Please let the header format as is -- no more changes, otherwise you
can almost certainly forget the deadline you want to meet (mid
November's SG16 meeting).

Stephan


=



From rem-conf Mon Sep 11 14:34:52 2000 
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The IESG has received a request from the Audio/Video Transport Working
Group to consider RTP Payload Format for ITU-T Recommendation G.722.1
<draft-ietf-avt-rtp-g7221-01.txt> as a Proposed Standard.

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send any comments to the
iesg@ietf.org or ietf@ietf.org mailing lists by September 25, 2000.

Files can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-g7221-01.txt





From rem-conf Mon Sep 11 14:35:20 2000 
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From: Zvi Lifshitz <zvil@csi.com>
To: "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>,
        "csp@east.isi.edu" <csp@east.isi.edu>,
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Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call
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Jan, all,

The simplest solution IMO is, as I proposed in the previous message, to use 
the first byte of the payload in the Generic format to indicate the length 
of the following extra header. No change is needed to the ES format.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
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Streaming Media Europe Earls Court Centre, Booth # 628, London, October 
10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Monday, September 11, 2000 1:04 PM
To:	csp@east.isi.edu; stewe@cs.tu-berlin.de
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to 
the last call


Stephan, Colin,

Stephan wrote:

Regardless how the generic payload format of the future may look like,
systems can be most easily designed to support both payload types.

True. But the issue is different. Mobile devices that are build to support
the current MPEG-4 ES draft will not be capable to decode RTP payloads with
the additional info flagged by the first byte in the payload. (Note that 
the
additional info would be needed by other applications that include usage of
the same stream)

In the past of MPEG, the use of "reserved fields" proved extremely 
powerful.
To not allow them here would be a failure in my opinion. Of course it would
be best to avoid unnecessary overhead. Therefore it would be most 
attractive
if the inclusion of the additional field is indicated, for example by a 
flag
or by a parameter that goes with the payload type. I understand that in the
RTP header no such flag is available, but that the use of a parameter
associated with the payload type would be possible. Would that be a good 
way
to solve this ?

Regards,

Jan

************************
Jan van der Meer
Philips - Digital Networks
Building OAN-4
P.O. Box 80002
5600 JB Eindhoven
The Netherlands
Phone +31 402 735 774
Fax +31 402 735 545
Email jan.vandermeer@philips.com
************************


-----Original Message-----
From:	stewe@cs.tu-berlin.de [mailto:stewe@cs.tu-berlin.de]
Sent:	vrijdag, 08 september, 2000 13:05
To:	young@techway.co.kr
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to
the last call


Folks,

>1.  We suggest to have a room for future extension of RTP header (e.g. one
>byte of payload header at the beginning of the payload part) to make the
>devices implementing this draft interoperable with the bitstream based on
>the generic format under development. For example, it is possible to use
>the reserved byte to specify the length of the payload header of generic
>payload format. Then the device implementing this draft can play the
>bitstream of generic payload format by simply skipping those bytes.

I'm not sure whether this is a good idea.  We would waste one byte of
payload (from RTP's point of view) for something which is probably not
used by any systems currently under discussion or development.

Regardless how the generic payload format of the future may look like,
systems can be most easily designed to support both payload types.
RTP's PT mechanism has enough numbering space to support both.
Capability announcement/negotiation will be different and have to
be redesigned anyway for anything more capable then what is
currently in the MPEG-4 ES draft.  Annoucing/Negotiating another
PT for the new generic payload spec will not make a difference.

When, in the future, a generic payload type would be available that
includes a super set of functionality of the MPEG-4 ES payload format,
then we could declare the MPEG-4 ES RFC as obsolete, thereby
advising people to move over to the new, generic payload type.

Or, it may also be possible to have two payload specs coexisting,
one for simple audio/video ES packetization only, the other containing
more (the rest of?) functionality of MPEG-4.  This does not sound
like a bad idea to me, because the necessary demands of functionality
both from the network (in terms of QoS) and implementation efforts
(single audio/visual ESs vs. system layer, flexmux, whatever...) are
very different.

Please let the header format as is -- no more changes, otherwise you
can almost certainly forget the deadline you want to meet (mid
November's SG16 meeting).

Stephan




From rem-conf Mon Sep 11 14:36:01 2000 
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From: Zvi Lifshitz <zvil@csi.com>
To: "'Colin Perkins'" <csp@east.isi.edu>,
        "jan.vandermeer@philips.com"
	 <jan.vandermeer@philips.com>
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Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
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Indeed.

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm ==================================================================
> Come see us at:
IBC Amsterdam RAI, Booth # 2.251,  September 8th-12, 2000
Streaming Media Europe Earls Court Centre, Booth # 628, London, October 10th-12th, 2000


-----Original Message-----
From:	Colin Perkins [SMTP:csp@east.isi.edu]
Sent:	Monday, September 11, 2000 5:22 PM
To:	jan.vandermeer@philips.com
Cc:	stewe@cs.tu-berlin.de; rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 

--> jan.vandermeer@philips.com writes:
>Stephan, Colin,
>
>Stephan wrote:
>
>Regardless how the generic payload format of the future may look like,
>systems can be most easily designed to support both payload types.
>
>True. But the issue is different. Mobile devices that are build to support
>the current MPEG-4 ES draft will not be capable to decode RTP payloads with
>the additional info flagged by the first byte in the payload. (Note that the
>additional info would be needed by other applications that include usage of
>the same stream)
>
>In the past of MPEG, the use of "reserved fields" proved extremely powerful.
>To not allow them here would be a failure in my opinion. Of course it would
>be best to avoid unnecessary overhead. Therefore it would be most attractive
>if the inclusion of the additional field is indicated, for example by a flag
>or by a parameter that goes with the payload type. I understand that in the
>RTP header no such flag is available, but that the use of a parameter
>associated with the payload type would be possible. Would that be a good way
>to solve this ?

In RTP, that parameter is the payload type field. 

I really don't understand why this additional field will add value. It 
allows for systems to skip data which they don't understand, how is this
different from using the payload type for the same effect?

Colin




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From rem-conf Mon Sep 11 14:37:05 2000 
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Subject: Re: Question: packing several frames in a single RTP packet
To: Orion Hodson <O.Hodson@cs.ucl.ac.uk>
Cc: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
	"'Guler, Amos'" <Amos_Guler@icomverse.com>,
	"'rem-conf@es.net'" <rem-conf@es.net>
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Later drafts of RFC 1890 resolve ambiguities even for variable frame length
audio codecs without size indicators (which are not sample-based).  For
example, take G.729A/B.  It has two types of frames, a 10 byte active, and
a 2 byte Annex B comfort noise.  The latest draft profile specifies that
the payload must consist of zero or more active frames followed by zero or
one comfort noise frame.

Vince Virgilio





Orion Hodson <O.Hodson@cs.ucl.ac.uk>@cs.ucl.ac.uk on 09/10/2000 04:50:22 AM

Sent by:  O.Hodson@cs.ucl.ac.uk


To:   Jonathan Rosenberg <jdrosen@dynamicsoft.com>
cc:   "'Guler, Amos'" <Amos_Guler@icomverse.com>, "'rem-conf@es.net'"
      <rem-conf@es.net>
Subject:  Re: Question: packing several frames in a single RTP packet


<B65B4F8437968F488A01A940B21982BF21FC33@DYN-EXCH-001.dynamicsoft.com>Jonathan

R
osenberg writes:
> This may be of theoretical interest only. In practice, I think that all
> frame based codecs are either (1) the same frame length, (2) indicate
within
> the frame itself what size or type it is, when the lengths are not the
same.
> Its only for frames that have variable length AND don't indicate their
> length that you  need worry about this. Does anyone know of any frame
codecs
> which are a problem?

VDVI is variable length and has no size indicator.  Since it's a very
low cost codec this isn't a major deal - it's 160 16-entry table
searches per frame.  It's slightly unusual because it is a sample
based codec cast that's effectively cast into a frame based format for
unambiguous decoding.  I can't think of any other's off hand.

- Orion.







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From: Herpel Carsten <HerpelC@thmulti.com>
To: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to
	 the last call 
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Dear all, 

Jan:
>True. But the issue is different. Mobile devices that are build to support
>the current MPEG-4 ES draft will not be capable to decode RTP payloads with
>the additional info flagged by the first byte in the payload. (Note that
the
>additional info would be needed by other applications that include usage of
>the same stream)
>
>In the past of MPEG, the use of "reserved fields" proved extremely
powerful.
>To not allow them here would be a failure in my opinion. Of course it would
>be best to avoid unnecessary overhead. Therefore it would be most
attractive
>if the inclusion of the additional field is indicated, for example by a
flag
>or by a parameter that goes with the payload type. I understand that in the
>RTP header no such flag is available, but that the use of a parameter
>associated with the payload type would be possible. Would that be a good
way
>to solve this ?

Colin:
>In RTP, that parameter is the payload type field. 
>
>I really don't understand why this additional field will add value. It 
>allows for systems to skip data which they don't understand, how is this
>different from using the payload type for the same effect?

What happens here is a clash of worlds.
The two of you have different pre-conditions for their proposals, if I am
not mistaken.

One assumption is that huge numbers of devices will be out there that use
the ES payload format. Assuming that these devices cannot be upgraded to a
newer payload format!
Then applications come up with more functionality, using MPEG-4 Systems. New
devices will exist that know how to use this functionality. Old devices
should at least be able to understand the (unmodified!) MPEG-4 Video over
RTP stream. Since the old devices are not reprogrammable, they cannot be
upgraded to another payload format. So the old format needs to be by design
compatible to any new format.

On the other hand is the assumption that when a new payload format gets
defined, all the existing devices will just be reprogrammed to read the new
format. And possibly the assumption that the changes in payload format will
be bigger than what can be captured by the proposal of just adding a single
field to be skipped. 

Is the first assumption so completely wrong that we can afford to go the
second way that is apparently being advocated by a majority of the speakers
in this thread?

Regards,
Carsten



From rem-conf Mon Sep 11 14:37:52 2000 
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[Herpel Carsten:]

One assumption is that huge numbers of devices will be out there that use
the ES payload format. Assuming that these devices cannot be upgraded to a
newer payload format!
Then applications come up with more functionality, using MPEG-4 Systems. 
New
devices will exist that know how to use this functionality. Old devices
should at least be able to understand the (unmodified!) MPEG-4 Video over
RTP stream. Since the old devices are not reprogrammable, they cannot be
upgraded to another payload format. So the old format needs to be by design
compatible to any new format.

[Reply:]

How would it help? The new format will come with a new payload type, and if 
the old devices don't know this payload type they will not read it because 
they don't know what it is. In other words they will not know they know how 
to read it...

So they will need to be reprogrammed to identify the new payload type. 2 
extra lines of code and they will know how to read it even if it's not 
identical to the old one.

And BTW I'm not sure the assumption is real. The two formats are there more 
or less at the same time. One is a little bit advanced in the procedure, 
but I don't believe this will create the "huge numbers" of devices that 
know one and not the other.

Regards,


z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm 
==================================================================
> Come see us at:
IBC Amsterdam RAI, Booth # 2.251,  September 8th-12, 2000
Streaming Media Europe Earls Court Centre, Booth # 628, London, October 
10th-12th, 2000






From rem-conf Mon Sep 11 14:38:16 2000 
From rem-conf-request@es.net Mon Sep 11 14:38:16 2000
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From: Orion Hodson <O.Hodson@cs.ucl.ac.uk>
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X-Phone: +44 (0)20 7679 3704
To: Vincent_Virgilio@nmss.com
cc: "'rem-conf@es.net'" <rem-conf@es.net>
Subject: Re: Question: packing several frames in a single RTP packet
In-reply-to: Your message of "Mon, 11 Sep 2000 11:33:56 CDT." <OF2301DF41.50469773-ON86256957.00559D8B@nmss.com>
Date: Mon, 11 Sep 2000 18:58:50 +0100
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<OF2301DF41.50469773-ON86256957.00559D8B@nmss.com>Vincent_Virgilio@nmss.com wri
tes:
> 
> Later drafts of RFC 1890 resolve ambiguities even for variable frame length
> audio codecs without size indicators (which are not sample-based).  For
> example, take G.729A/B.  It has two types of frames, a 10 byte active, and
> a 2 byte Annex B comfort noise.  The latest draft profile specifies that
> the payload must consist of zero or more active frames followed by zero or
> one comfort noise frame.

The current draft (draft-ietf-avt-profile-new-09.txt) has the wording
about relatively prime payload sizes.  I believe the wording about
mixing CN and coded frames is specific to G.729, and the combination
of speech and CN frames.

G.729 and it's annexes cover several different bit-rates and
correspondingly different frame sizes.  Different payload identifiers
are used for the different bit-rate representations and therefore
different speech frame sizes cannot be packetized within packets the
same packet.  This solution is an implicit realization of Amos's point
and is one way of solving the problem should it occur with other
codecs.

- orion.





From rem-conf Mon Sep 11 14:38:34 2000 
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Date: Mon, 11 Sep 2000 20:57:18 +0200
To: Herpel Carsten <HerpelC@thmulti.com>
From: Stephan Wenger <stewe@cs.tu-berlin.de>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
  to the last call 
Cc: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
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Folks

>What happens here is a clash of worlds.

Clearly, it is!

But I don't think that Carsten came to the real point here.

IMHO, the main point is that MPEG-folk seem to think that
the media stream (RTP/payload/MPEG) should be self-contained
and  independent from the higher protocol layers that perform the
capability exchange/announcement (and other things such
as call establishment etc.).
IETF/AVT folk (including me), however, make the implicit
assumption that media cannot be seen independently from
higher protocols.

The latter, however, is not an assumption -- its the plain truth,
especially after AVT made it the general policy to no more
assign static payload types for media coding.

This means, that RTP systems rely (and cannot live without)
some external form of capability exchange/announcement, which,
as a minimum, dynamically (per call) associates a freely chosen
number with a certain payload (e.g. MPEG-4 ES or, in the future,
MPEG-4 generic/SL/whatever).  The information which payload is
to be transmitted is in H.245, for example, coded in ASN.1.  For SIP
systems, the relevant information is transmitted in ASCII form,
where the payload types (such as MPEG-4-ES) has typically
a MIME registration.

That is why Colin and me and others insist that there is no need to
duplicate this type of information in the media stream.  Every
system that uses RTP in a standard-compliant way has already
the functionality build in that is necessary to distinguish between
different payload types.

>One assumption is that huge numbers of devices will be out there that use
>the ES payload format. Assuming that these devices cannot be upgraded to a
>newer payload format!
>Then applications come up with more functionality, using MPEG-4 Systems. New
>devices will exist that know how to use this functionality. Old devices
>should at least be able to understand the (unmodified!) MPEG-4 Video over
>RTP stream. Since the old devices are not reprogrammable, they cannot be
>upgraded to another payload format. So the old format needs to be by design
>compatible to any new format.

Very simple to achieve.  Have two payload types, MPEG-4-SL and MPEG-4-ES.
Try to negotiate MPEG-4-SL.  If the other system rejects, use MPEG-4-ES.
This is how things are done, at least in H.323, all the time.

Stephan




From rem-conf Mon Sep 11 19:33:47 2000 
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From rem-conf Mon Sep 11 23:37:27 2000 
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From: "Sales/Marketing" <info@luxurycarts.com>
To: <rem-conf@es.net>
Subject: luxuryCarts,com - Online Source for Custom Golf Carts
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Please visit www.luxuryCarts.com for the premier online source for custom golf carts and kit carts available.

If this email was received at the wrong party, please type "remove" in the subject line and hit send.

Regards,

The Marketing Staff at luxuryCarts.com



From rem-conf Tue Sep 12 03:17:30 2000 
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Message-ID: <001201c019fc$d1f066e0$49d78796@rsk>
From: "Riyad Kalla" <rsk@u.arizona.edu>
To: "Dave Dunstan" <dave@webuildthe.com>,
	"'Luis Montes'" <LMontes@computer-guidance.com>,
	"'rusty'" <rustyc@inficad.com>,
	"'Mark Weaver'" <mdw1982@epix.net>
Cc: "'Li Bing'" <bing.li@asu.edu>,
	"'Zigang Zhu'" <eagle@cloudnet.com.cn>,
	"'Zhu Zigang'" <sales@cncard.com>,
	"'ZhenweiShen'" <ZhenweiShen@acersoftech.com.cn>,
	"'Zheng. Tom. Yuan'" <Tom.Yuan@asu.edu>,
	"'Yingqi Kong'" <yingqik@asu.edu>,
	"'Xiaoying Bai'" <xiaoying.bai@asu.edu>,
	"'WINSOCK-2'" <WINSOCK-2@mailbag.cps.intel.com>,
	"'Webdesigners'" <webdesigners@egroups.com>,
	"'VOA XinWen News Service'" <xinwen-l@lists.VOA.GOV>,
	"'Techeng Shen'" <techeng@asu.edu>,
	"'SERVLET-INTEREST'" <SERVLET-INTEREST@JAVA.SUN.COM>,
	"'RTLINUX'" <rtl@rtlinux.org>,
	"'Rtl-Announce'" <rtl-announce@rtlinux.org>,
	"'Rtl'" <rtl@www.rtlinux.org>,
	"'Rsvp'" <rsvp@ISI.EDU>,
	"'Rem-Conf'" <rem-conf@es.net>,
	"'Redhat-Devel-List'" <redhat-devel-list@redhat.com>,
	"'Quyouli'" <quyouli@frdc-fujitsu.com.cn>,
	"'Pan Guobin'" <Guobin.Pan@asu.edu>,
	"'MySQL'" <mysql@lists.mysql.com>,
	"'Msql-Jdbc'" <msql-jdbc@list.imaginary.com>,
	"'Lyx-Users'" <lyx-users@lists.lyx.org>,
	"'Lyx-Devel'" <lyx-devel@lyx.org>,
	"'Lyx Mailing List'" <lyx-devel@lists.lyx.org>,
	"'Linux-Atm'" <linux-atm@lrc.epfl.ch>,
	"'JDCMember'" <JDCMember@sun.com>,
	"'Javamug'" <javamug@utdallas.edu>,
	"'JavaLearners'" <JavaLearners@egroups.com>,
	"'Java_Help'" <java_help@egroups.com>,
	"'Java'" <java@phxjug.org>,
	"'Java'" <java@egroups.com>,
	"'Java'" <java@jfind.com>,
	"'J2EE-INTEREST'" <J2EE-INTEREST@JAVA.SUN.COM>,
	"'Ietf'" <ietf@ietf.org>,
	"'IDL-USERS'" <IDL-USERS@JAVA.SUN.COM>,
	"'Huang Ning'" <ninghuang@lucent.com>,
	"'Gnome-List'" <gnome-list@gnome.org>,
	"'Fariaz Karim'" <karim@asu.edu>,
	"'Dmiller'" <donald.miller@asu.edu>,
	"'CORBA-patterns'" <CORBA-patterns@cs.uiuc.edu>,
	"'Corba-Dev'" <corba-dev@randomwalk.com>,
	"'Confctrl'" <confctrl@ISI.EDU>,
	"'Cgi-List'" <cgi-list@jann.com>,
	"'Blinux-Newbie'" <blinux-newbie@braille.uwo.ca>,
	"'Blinux-List'" <blinux-list@redhat.com>,
	"'BJ CN \) Jun_Fan Hu \(MED'" <Jun_Fan.Hu@china.med.ge.com>,
	"'Baisu Huang'" <baisu.huang@asu.edu>,
	"'ASULUG'" <ASULUG@asu.edu>,
	"'Agoracgi'" <agoracgi@egroups.com>,
	"'Advanced-Java'" <advanced-java@xcf.berkeley.edu>
References: <CA129E889266D411924000E018C00201024186@dnai-216-15-54-199.cust.dnai.com>
Subject: Re: [phxjug-java] Re: Two tools to remove the virus LIFE_STAGES.TXT.SHS
Date: Fri, 8 Sep 2000 18:25:20 -0700
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Don't say that, its too intelligent.

Lets get back to talking about how other OSs cause cancer and kill farm
animals

<turns around and sees dog drop dead>

NOOOOOO!!! I'LL GET YOU WINDOWS!!!

Have a good weekend everyone!

-Riyad

----- Original Message -----
From: "Dave Dunstan" <dave@webuildthe.com>
To: "'Luis Montes'" <LMontes@computer-guidance.com>; "'rusty'"
<rustyc@inficad.com>; "'Mark Weaver'" <mdw1982@epix.net>
Cc: "'Li Bing'" <bing.li@asu.edu>; "'Zigang Zhu'" <eagle@cloudnet.com.cn>;
"'Zhu Zigang'" <sales@cncard.com>; "'ZhenweiShen'"
<ZhenweiShen@acersoftech.com.cn>; "'Zheng. Tom. Yuan'" <Tom.Yuan@asu.edu>;
"'Yingqi Kong'" <yingqik@asu.edu>; "'Xiaoying Bai'" <xiaoying.bai@asu.edu>;
"'WINSOCK-2'" <WINSOCK-2@mailbag.cps.intel.com>; "'Webdesigners'"
<webdesigners@egroups.com>; "'VOA XinWen News Service'"
<xinwen-l@lists.VOA.GOV>; "'Techeng Shen'" <techeng@asu.edu>;
"'SERVLET-INTEREST'" <SERVLET-INTEREST@JAVA.SUN.COM>; "'RTLINUX'"
<rtl@rtlinux.org>; "'Rtl-Announce'" <rtl-announce@rtlinux.org>; "'Rtl'"
<rtl@www.rtlinux.org>; "'Rsvp'" <rsvp@ISI.EDU>; "'Rem-Conf'"
<rem-conf@es.net>; "'Redhat-Devel-List'" <redhat-devel-list@redhat.com>;
"'Quyouli'" <quyouli@frdc-fujitsu.com.cn>; "'Pan Guobin'"
<Guobin.Pan@asu.edu>; "'MySQL'" <mysql@lists.mysql.com>; "'Msql-Jdbc'"
<msql-jdbc@list.imaginary.com>; "'Lyx-Users'" <lyx-users@lists.lyx.org>;
"'Lyx-Devel'" <lyx-devel@lyx.org>; "'Lyx Mailing List'"
<lyx-devel@lists.lyx.org>; "'Linux-Atm'" <linux-atm@lrc.epfl.ch>;
"'JDCMember'" <JDCMember@sun.com>; "'Javamug'" <javamug@utdallas.edu>;
"'JavaLearners'" <JavaLearners@egroups.com>; "'Java_Help'"
<java_help@egroups.com>; "'Java'" <java@phxjug.org>; "'Java'"
<java@egroups.com>; "'Java'" <java@jfind.com>; "'J2EE-INTEREST'"
<J2EE-INTEREST@JAVA.SUN.COM>; "'Ietf'" <ietf@ietf.org>; "'IDL-USERS'"
<IDL-USERS@JAVA.SUN.COM>; "'Huang Ning'" <ninghuang@lucent.com>;
"'Gnome-List'" <gnome-list@gnome.org>; "'Fariaz Karim'" <karim@asu.edu>;
"'Dmiller'" <donald.miller@asu.edu>; "'CORBA-patterns'"
<CORBA-patterns@cs.uiuc.edu>; "'Corba-Dev'" <corba-dev@randomwalk.com>;
"'Confctrl'" <confctrl@ISI.EDU>; "'Cgi-List'" <cgi-list@jann.com>;
"'Blinux-Newbie'" <blinux-newbie@braille.uwo.ca>; "'Blinux-List'"
<blinux-list@redhat.com>; "'BJ CN ) Jun_Fan Hu (MED'"
<Jun_Fan.Hu@china.med.ge.com>; "'Baisu Huang'" <baisu.huang@asu.edu>;
"'ASULUG'" <ASULUG@asu.edu>; "'Agoracgi'" <agoracgi@egroups.com>;
"'Advanced-Java'" <advanced-java@xcf.berkeley.edu>
Sent: Friday, September 08, 2000 5:17 PM
Subject: RE: [phxjug-java] Re: Two tools to remove the virus
LIFE_STAGES.TXT.SHS


> ok zealots. use the right tool for the job and feed your families.
>
> Dave Dunstan | dave@webuildthe.com
> Mission Control Consulting LLC
> 1383a 9th Ave, San Francisco CA 94122
> ph: 415.504.8177 | fx: 415.504.8172
>
>
> > -----Original Message-----
> > From: Luis Montes [mailto:LMontes@computer-guidance.com]
> > Sent: Friday, September 08, 2000 5:23 PM
> > To: rusty; Mark Weaver
> > Cc: Li Bing; Zigang Zhu; Zhu Zigang; ZhenweiShen; Zheng. Tom. Yuan;
> > Yingqi Kong; Xiaoying Bai; WINSOCK-2; Webdesigners; VOA XinWen News
> > Service; Techeng Shen; SERVLET-INTEREST; RTLINUX; Rtl-Announce; Rtl;
> > Rsvp; Rem-Conf; Redhat-Devel-List; Quyouli; Pan Guobin; MySQL;
> > Msql-Jdbc; Lyx-Users; Lyx-Devel; Lyx Mailing List; Linux-Atm;
> > JDCMember;
> > Javamug; JavaLearners; Java_Help; Java; Java; Java;
> > J2EE-INTEREST; Ietf;
> > IDL-USERS; Huang Ning; Gnome-List; Fariaz Karim; Dmiller;
> > CORBA-patterns; Corba-Dev; Confctrl; Cgi-List; Blinux-Newbie;
> > Blinux-List; BJ CN ) Jun_Fan Hu (MED; Baisu Huang; ASULUG; Agoracgi;
> > Advanced-Java
> > Subject: RE: [phxjug-java] Re: Two tools to remove the virus
> > LIFE_STAGES.TXT.SHS
> >
> >
> > Amen Brother
> >
> > > -----Original Message-----
> > > From: rusty [SMTP:rustyc@inficad.com]
> > > Sent: Friday, September 08, 2000 4:59 PM
> > > To: Mark Weaver
> > > Cc: Li Bing; Zigang Zhu; Zhu Zigang; ZhenweiShen; Zheng. Tom. Yuan;
> > > Yingqi Kong; Xiaoying Bai; WINSOCK-2; Webdesigners; VOA XinWen News
> > > Service; Techeng Shen; SERVLET-INTEREST; RTLINUX;
> > Rtl-Announce; Rtl; Rsvp;
> > > Rem-Conf; Redhat-Devel-List; Quyouli; Pan Guobin; MySQL; Msql-Jdbc;
> > > Lyx-Users; Lyx-Devel; Lyx Mailing List; Linux-Atm;
> > JDCMember; Javamug;
> > > JavaLearners; Java_Help; Java; Java; Java; J2EE-INTEREST;
> > Ietf; IDL-USERS;
> > > Huang Ning; Gnome-List; Fariaz Karim; Dmiller;
> > CORBA-patterns; Corba-Dev;
> > > Confctrl; Cgi-List; Blinux-Newbie; Blinux-List; BJ CN )
> > Jun_Fan Hu (MED;
> > > Baisu Huang; ASULUG; Agoracgi; Advanced-Java
> > > Subject: Re: [phxjug-java] Re: Two tools to remove the virus
> > > LIFE_STAGES.TXT.SHS
> > >
> > > Mark Weaver wrote:
> > > >
> > > > What the HELL is this? A virus warning? This is exactly
> > why I don't use
> > > > Windows anymore. Thank God for Linux. You all can have
> > your Windows and
> > > > your virii.
> > > >
> > > I'd like to put it a bit more strongly - Anybody who uses microsoft
> > > tools
> > > deserves everything that happens to them, given the track history of
> > > viruses on that 'platform'.  There is no longer any excuse.
> > >
> > > And any IT 'professional' who chooses Microsoft tools for their
> > > company
> > > should pay for the damage caused by use (and breakins using) those
> > > tools.
> > >
> > > Or perhaps just get fired.
> > >
> > > rusty
> > >
> > > --
> > > To unsubscribe, send a message to java-request@phxjug.org with
> > > "unsubscribe" in the *subject line*. If you have problems, send a
> > > message to 'listmaster@phxjug.org'.
> >
> > --
> > ---------------------------------------------------------------------
> > Please check "http://www.mysql.com/php/manual.php" before
> > posting. To request this thread, e-mail
> > mysql-thread50124@lists.mysql.com
> >
> > To unsubscribe, send a message to:
> >     <mysql-unsubscribe-dave=webuildthe.com@lists.mysql.com>
> >
> > If you have a broken mail client that cannot send a message
> > to the above address(Microsoft Outlook), you can use
> http://lists.mysql.com/php/unsubscribe.php
>
> --
> To unsubscribe, send a message to java-request@phxjug.org with
> "unsubscribe" in the *subject line*. If you have problems, send a
> message to 'listmaster@phxjug.org'.
>


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From rem-conf Tue Sep 12 03:43:44 2000 
From rem-conf-request@es.net Tue Sep 12 03:43:42 2000
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To: <HerpelC@thmulti.com>, <stewe@cs.tu-berlin.de>
Cc: <rem-conf@es.net>, <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the
         last call
Message-ID: <0056890014020160000002L902*@MHS>
Date: Tue, 12 Sep 2000 12:43:44 +0200
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Stephan, Carsten, Colin, Zvi and others,

It surely is a clash of worlds. And I think we did not come to full
understanding yet. I think I understand the point made by Stephan, but =
I
also think that Carsten did make the real point. Let me explain below w=
ith
an example.

A service provides several multicast sessions to mobile devices using t=
he
simple MPEG-4 ES payload format. This is done during a sport event such=
 as
the "Tour de France" providing the performance of a number of favorite
racers. At the same time there is a broadcast coverage of the event,
providing a more global overview. Now, when I am "on the road" I watch =
my
own favorite racer on my mobile device. When at home, I watch the broad=
cast
coverage on my TV. This TV has the capability to also receive mobile
services. Over the normal broadcast channel an MPEG-4 application is
delivered that allows me to display the multicast streaming of my favor=
ite
racer on TV, as an overlay over the broadcast video. For this purpose, =
some
information is needed that is contained in an SL header. In order to be=

bandwidth efficient on the mobile network, the same multicast session i=
s
used as for mobile devices.

Now follows an essential assumption: the mobile devices are not_upgrada=
ble;
these dedicated devices are not capable to even process a single byte t=
o
upgrade their functionality. If this assumption is true, they need the
capability to accept the additional bytes from day one. Otherwise two
separate multicast sessions are needed.

In my understanding there are no higher level protocols involved. And t=
he
essential question is whether these mobile devices are indeed not
upgradable. It is my guess that upgradability is not a requirement for =
such
devices, but only a feature that cannot be relied upon.

I hope this clarifies.

Kind regards,

Jan

-----Original Message-----
From:	stewe@cs.tu-berlin.de [mailto:stewe@cs.tu-berlin.de]
Sent:	maandag, 11 september, 2000 21:05
To:	HerpelC@thmulti.com
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response=
 to
the last call


Folks

>What happens here is a clash of worlds.

Clearly, it is!

But I don't think that Carsten came to the real point here.

IMHO, the main point is that MPEG-folk seem to think that
the media stream (RTP/payload/MPEG) should be self-contained
and  independent from the higher protocol layers that perform the
capability exchange/announcement (and other things such
as call establishment etc.).
IETF/AVT folk (including me), however, make the implicit
assumption that media cannot be seen independently from
higher protocols.

The latter, however, is not an assumption -- its the plain truth,
especially after AVT made it the general policy to no more
assign static payload types for media coding.

This means, that RTP systems rely (and cannot live without)
some external form of capability exchange/announcement, which,
as a minimum, dynamically (per call) associates a freely chosen
number with a certain payload (e.g. MPEG-4 ES or, in the future,
MPEG-4 generic/SL/whatever).  The information which payload is
to be transmitted is in H.245, for example, coded in ASN.1.  For SIP
systems, the relevant information is transmitted in ASCII form,
where the payload types (such as MPEG-4-ES) has typically
a MIME registration.

That is why Colin and me and others insist that there is no need to
duplicate this type of information in the media stream.  Every
system that uses RTP in a standard-compliant way has already
the functionality build in that is necessary to distinguish between
different payload types.

>One assumption is that huge numbers of devices will be out there that =
use
>the ES payload format. Assuming that these devices cannot be upgraded =
to a
>newer payload format!
>Then applications come up with more functionality, using MPEG-4 System=
s.
New
>devices will exist that know how to use this functionality. Old device=
s
>should at least be able to understand the (unmodified!) MPEG-4 Video o=
ver
>RTP stream. Since the old devices are not reprogrammable, they cannot =
be
>upgraded to another payload format. So the old format needs to be by d=
esign
>compatible to any new format.

Very simple to achieve.  Have two payload types, MPEG-4-SL and MPEG-4-E=
S.
Try to negotiate MPEG-4-SL.  If the other system rejects, use MPEG-4-ES=
.
This is how things are done, at least in H.323, all the time.

Stephan

=



From rem-conf Tue Sep 12 04:02:03 2000 
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From: "Eric Horlait" <eric.horlait@lip6.fr>
To: "Rem-Conf@Es. Net" <rem-conf@es.net>
Subject: Mobile Agents for Telecommunication Applications, Paris, France
Date: Tue, 12 Sep 2000 12:59:42 +0200
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==============================================================(360628)
 [My apologies if you receive this more than once]
==============================================================
CALL FOR PARTICIPATION MATA'2000 - Paris, France NEXT WEEK
Second International Workshop on Mobile Agents for Telecommunication
Applications
========================================================================
=
September 19-20, 2000
Paris, France

http://netconf.lip6.fr/mata00/

MATA '2000 presents a unique opportunity for researchers, software and
Application developers, and computer network technologists to discuss
new
developments on the mobile agent technology and applications.

For complete program information, registration fees, or online
registration form please visit

http://netconf.lip6.fr/mata00/






From rem-conf Tue Sep 12 04:29:34 2000 
From rem-conf-request@es.net Tue Sep 12 04:29:33 2000
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Message-ID: <D33D3145480ED4119EEB000629398551908155@optmail.optibase.co.il>
From: Zvi Lifshitz <ZviL@optibase.com>
To: jan.vandermeer@philips.com, HerpelC@thmulti.com, stewe@cs.tu-berlin.de
Cc: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to
	 the last call
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Dear Jan,

You clarified the scenario, but not how it relates to the discussion.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com <mailto:zvil@csi.com> 			Fax
+972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
IBC Amsterdam RAI, Booth # 2.251,  September 8th-12, 2000
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Tuesday, September 12, 2000 12:44 PM
To:	HerpelC@thmulti.com; stewe@cs.tu-berlin.de
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to the last call

 
Stephan, Carsten, Colin, Zvi and others,

It surely is a clash of worlds. And I think we did not come to full
understanding yet. I think I understand the point made by Stephan, but I
also think that Carsten did make the real point. Let me explain below with
an example.

A service provides several multicast sessions to mobile devices using the
simple MPEG-4 ES payload format. This is done during a sport event such as
the "Tour de France" providing the performance of a number of favorite
racers. At the same time there is a broadcast coverage of the event,
providing a more global overview. Now, when I am "on the road" I watch my
own favorite racer on my mobile device. When at home, I watch the broadcast
coverage on my TV. This TV has the capability to also receive mobile
services. Over the normal broadcast channel an MPEG-4 application is
delivered that allows me to display the multicast streaming of my favorite
racer on TV, as an overlay over the broadcast video. For this purpose, some
information is needed that is contained in an SL header. In order to be
bandwidth efficient on the mobile network, the same multicast session is
used as for mobile devices.

Now follows an essential assumption: the mobile devices are not_upgradable;
these dedicated devices are not capable to even process a single byte to
upgrade their functionality. If this assumption is true, they need the
capability to accept the additional bytes from day one. Otherwise two
separate multicast sessions are needed.

In my understanding there are no higher level protocols involved. And the
essential question is whether these mobile devices are indeed not
upgradable. It is my guess that upgradability is not a requirement for such
devices, but only a feature that cannot be relied upon.

I hope this clarifies.

Kind regards,

Jan



From rem-conf Tue Sep 12 06:54:25 2000 
From rem-conf-request@es.net Tue Sep 12 06:54:24 2000
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To: <ZviL@optibase.com>
Cc: <rem-conf@es.net>, <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the
         last call
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Date: Tue, 12 Sep 2000 15:50:36 +0200
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Dear Zvi,

I thought I did. What I try to say is: to have this type of functionali=
ty in
future, you need to allow for inclusion of the additional bytes in the
MPEG-2 ES payload now.

Regards,

Jan

-----Original Message-----
From:	ZviL@optibase.com [mailto:ZviL@optibase.com]
Sent:	dinsdag, 12 september, 2000 13:23
To:	stewe@cs.tu-berlin.de; HerpelC@thmulti.com; Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response=
 to
the last call


Dear Jan,

You clarified the scenario, but not how it relates to the discussion.

Regards,

z
soon the whole world will STREAM
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com <mailto:zvil@csi.com> 			Fax
+972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> Come see us at:
IBC Amsterdam RAI, Booth # 2.251,  September 8th-12, 2000
Streaming Media Europe Earls Court Centre, Booth # 628, London, October=

10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Tuesday, September 12, 2000 12:44 PM
To:	HerpelC@thmulti.com; stewe@cs.tu-berlin.de
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to the last call


Stephan, Carsten, Colin, Zvi and others,

It surely is a clash of worlds. And I think we did not come to full
understanding yet. I think I understand the point made by Stephan, but =
I
also think that Carsten did make the real point. Let me explain below w=
ith
an example.

A service provides several multicast sessions to mobile devices using t=
he
simple MPEG-4 ES payload format. This is done during a sport event such=
 as
the "Tour de France" providing the performance of a number of favorite
racers. At the same time there is a broadcast coverage of the event,
providing a more global overview. Now, when I am "on the road" I watch =
my
own favorite racer on my mobile device. When at home, I watch the broad=
cast
coverage on my TV. This TV has the capability to also receive mobile
services. Over the normal broadcast channel an MPEG-4 application is
delivered that allows me to display the multicast streaming of my favor=
ite
racer on TV, as an overlay over the broadcast video. For this purpose, =
some
information is needed that is contained in an SL header. In order to be=

bandwidth efficient on the mobile network, the same multicast session i=
s
used as for mobile devices.

Now follows an essential assumption: the mobile devices are not_upgrada=
ble;
these dedicated devices are not capable to even process a single byte t=
o
upgrade their functionality. If this assumption is true, they need the
capability to accept the additional bytes from day one. Otherwise two
separate multicast sessions are needed.

In my understanding there are no higher level protocols involved. And t=
he
essential question is whether these mobile devices are indeed not
upgradable. It is my guess that upgradability is not a requirement for =
such
devices, but only a feature that cannot be relied upon.

I hope this clarifies.

Kind regards,

Jan

=



From rem-conf Tue Sep 12 07:08:47 2000 
From rem-conf-request@es.net Tue Sep 12 07:08:47 2000
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Message-ID: <D33D3145480ED4119EEB000629398551908161@optmail.optibase.co.il>
From: Zvi Lifshitz <ZviL@optibase.com>
To: jan.vandermeer@philips.com, Zvi Lifshitz <ZviL@optibase.com>
Cc: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to
	 the last call
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It will not help, because when you get the new payload in the future the old
devices don't recognize the payload type and will not handle it.

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com <mailto:zvil@csi.com> 			Fax
+972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
IBC Amsterdam RAI, Booth # 2.251,  September 8th-12, 2000
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Tuesday, September 12, 2000 3:51 PM
To:	ZviL@optibase.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to the last call

Dear Zvi,

I thought I did. What I try to say is: to have this type of functionality in
future, you need to allow for inclusion of the additional bytes in the
MPEG-2 ES payload now.

Regards,

Jan

-----Original Message-----
From:	ZviL@optibase.com [mailto:ZviL@optibase.com]
Sent:	dinsdag, 12 september, 2000 13:23
To:	stewe@cs.tu-berlin.de; HerpelC@thmulti.com; Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to
the last call


Dear Jan,

You clarified the scenario, but not how it relates to the discussion.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com <mailto:zvil@csi.com> 			Fax
+972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
IBC Amsterdam RAI, Booth # 2.251,  September 8th-12, 2000
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Tuesday, September 12, 2000 12:44 PM
To:	HerpelC@thmulti.com; stewe@cs.tu-berlin.de
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to the last call


Stephan, Carsten, Colin, Zvi and others,

It surely is a clash of worlds. And I think we did not come to full
understanding yet. I think I understand the point made by Stephan, but I
also think that Carsten did make the real point. Let me explain below with
an example.

A service provides several multicast sessions to mobile devices using the
simple MPEG-4 ES payload format. This is done during a sport event such as
the "Tour de France" providing the performance of a number of favorite
racers. At the same time there is a broadcast coverage of the event,
providing a more global overview. Now, when I am "on the road" I watch my
own favorite racer on my mobile device. When at home, I watch the broadcast
coverage on my TV. This TV has the capability to also receive mobile
services. Over the normal broadcast channel an MPEG-4 application is
delivered that allows me to display the multicast streaming of my favorite
racer on TV, as an overlay over the broadcast video. For this purpose, some
information is needed that is contained in an SL header. In order to be
bandwidth efficient on the mobile network, the same multicast session is
used as for mobile devices.

Now follows an essential assumption: the mobile devices are not_upgradable;
these dedicated devices are not capable to even process a single byte to
upgrade their functionality. If this assumption is true, they need the
capability to accept the additional bytes from day one. Otherwise two
separate multicast sessions are needed.

In my understanding there are no higher level protocols involved. And the
essential question is whether these mobile devices are indeed not
upgradable. It is my guess that upgradability is not a requirement for such
devices, but only a feature that cannot be relied upon.

I hope this clarifies.

Kind regards,

Jan




From rem-conf Tue Sep 12 07:19:32 2000 
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To: jan.vandermeer@philips.com
cc: HerpelC <HerpelC@thmulti.com>, stewe <stewe@cs.tu-berlin.de>,
        rem-conf <rem-conf@es.net>,
        4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
In-Reply-To: Your message of "Tue, 12 Sep 2000 12:43:44 +0200."
             <0056890014020160000002L902*@MHS> 
Date: Tue, 12 Sep 2000 10:18:12 -0400
From: Colin Perkins <csp@east.isi.edu>
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I don't understand why you don't simply send the composition information
along with the broadcast signal? Sending it with the mobile signal (where
it is useless unless the broadcast signal is also received) seems to be a
waste of mobile capacity.

Colin


--> jan.vandermeer@philips.com writes:
>Stephan, Carsten, Colin, Zvi and others,
>
>It surely is a clash of worlds. And I think we did not come to full
>understanding yet. I think I understand the point made by Stephan, but I
>also think that Carsten did make the real point. Let me explain below with
>an example.
>
>A service provides several multicast sessions to mobile devices using the
>simple MPEG-4 ES payload format. This is done during a sport event such as
>the "Tour de France" providing the performance of a number of favorite
>racers. At the same time there is a broadcast coverage of the event,
>providing a more global overview. Now, when I am "on the road" I watch my
>own favorite racer on my mobile device. When at home, I watch the broadcast
>coverage on my TV. This TV has the capability to also receive mobile
>services. Over the normal broadcast channel an MPEG-4 application is
>delivered that allows me to display the multicast streaming of my favorite
>racer on TV, as an overlay over the broadcast video. For this purpose, some
>information is needed that is contained in an SL header. In order to be
>bandwidth efficient on the mobile network, the same multicast session is
>used as for mobile devices.
>
>Now follows an essential assumption: the mobile devices are not_upgradable;
>these dedicated devices are not capable to even process a single byte to
>upgrade their functionality. If this assumption is true, they need the
>capability to accept the additional bytes from day one. Otherwise two
>separate multicast sessions are needed.
>
>In my understanding there are no higher level protocols involved. And the
>essential question is whether these mobile devices are indeed not
>upgradable. It is my guess that upgradability is not a requirement for such
>devices, but only a feature that cannot be relied upon.
>
>I hope this clarifies.
>
>Kind regards,
>
>Jan
>
>-----Original Message-----
>From:	stewe@cs.tu-berlin.de [mailto:stewe@cs.tu-berlin.de]
>Sent:	maandag, 11 september, 2000 21:05
>To:	HerpelC@thmulti.com
>Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
>Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to
>the last call
>
>
>Folks
>
>>What happens here is a clash of worlds.
>
>Clearly, it is!
>
>But I don't think that Carsten came to the real point here.
>
>IMHO, the main point is that MPEG-folk seem to think that
>the media stream (RTP/payload/MPEG) should be self-contained
>and  independent from the higher protocol layers that perform the
>capability exchange/announcement (and other things such
>as call establishment etc.).
>IETF/AVT folk (including me), however, make the implicit
>assumption that media cannot be seen independently from
>higher protocols.
>
>The latter, however, is not an assumption -- its the plain truth,
>especially after AVT made it the general policy to no more
>assign static payload types for media coding.
>
>This means, that RTP systems rely (and cannot live without)
>some external form of capability exchange/announcement, which,
>as a minimum, dynamically (per call) associates a freely chosen
>number with a certain payload (e.g. MPEG-4 ES or, in the future,
>MPEG-4 generic/SL/whatever).  The information which payload is
>to be transmitted is in H.245, for example, coded in ASN.1.  For SIP
>systems, the relevant information is transmitted in ASCII form,
>where the payload types (such as MPEG-4-ES) has typically
>a MIME registration.
>
>That is why Colin and me and others insist that there is no need to
>duplicate this type of information in the media stream.  Every
>system that uses RTP in a standard-compliant way has already
>the functionality build in that is necessary to distinguish between
>different payload types.
>
>>One assumption is that huge numbers of devices will be out there that use
>>the ES payload format. Assuming that these devices cannot be upgraded to a
>>newer payload format!
>>Then applications come up with more functionality, using MPEG-4 Systems.
>New
>>devices will exist that know how to use this functionality. Old devices
>>should at least be able to understand the (unmodified!) MPEG-4 Video over
>>RTP stream. Since the old devices are not reprogrammable, they cannot be
>>upgraded to another payload format. So the old format needs to be by design
>>compatible to any new format.
>
>Very simple to achieve.  Have two payload types, MPEG-4-SL and MPEG-4-ES.
>Try to negotiate MPEG-4-SL.  If the other system rejects, use MPEG-4-ES.
>This is how things are done, at least in H.323, all the time.
>
>Stephan
>



From rem-conf Tue Sep 12 07:20:41 2000 
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To: <ZviL@optibase.com>
Cc: <rem-conf@es.net>, <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the
         last call
Message-ID: <0056890014029173000002L932*@MHS>
Date: Tue, 12 Sep 2000 16:20:59 +0200
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Zvi,

It is not another payload type; it's the MPEG-4 ES one, but with some o=
ther
bytes which are not understood by the mobile device but which are allow=
ed to
be there.

Regards,

Jan

-----Original Message-----
From:	ZviL@optibase.com [mailto:ZviL@optibase.com]
Sent:	dinsdag, 12 september, 2000 16:09
To:	ZviL@optibase.com; Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response=
 to
the last call


It will not help, because when you get the new payload in the future th=
e old
devices don't recognize the payload type and will not handle it.

z
soon the whole world will STREAM
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com <mailto:zvil@csi.com> 			Fax
+972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> Come see us at:
IBC Amsterdam RAI, Booth # 2.251,  September 8th-12, 2000
Streaming Media Europe Earls Court Centre, Booth # 628, London, October=

10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Tuesday, September 12, 2000 3:51 PM
To:	ZviL@optibase.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to the last call

Dear Zvi,

I thought I did. What I try to say is: to have this type of functionali=
ty in
future, you need to allow for inclusion of the additional bytes in the
MPEG-2 ES payload now.

Regards,

Jan

-----Original Message-----
From:	ZviL@optibase.com [mailto:ZviL@optibase.com]
Sent:	dinsdag, 12 september, 2000 13:23
To:	stewe@cs.tu-berlin.de; HerpelC@thmulti.com; Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to
the last call


Dear Jan,

You clarified the scenario, but not how it relates to the discussion.

Regards,

z
soon the whole world will STREAM
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com <mailto:zvil@csi.com> 			Fax
+972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> Come see us at:
IBC Amsterdam RAI, Booth # 2.251,  September 8th-12, 2000
Streaming Media Europe Earls Court Centre, Booth # 628, London, October=

10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Tuesday, September 12, 2000 12:44 PM
To:	HerpelC@thmulti.com; stewe@cs.tu-berlin.de
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to the last call


Stephan, Carsten, Colin, Zvi and others,

It surely is a clash of worlds. And I think we did not come to full
understanding yet. I think I understand the point made by Stephan, but =
I
also think that Carsten did make the real point. Let me explain below w=
ith
an example.

A service provides several multicast sessions to mobile devices using t=
he
simple MPEG-4 ES payload format. This is done during a sport event such=
 as
the "Tour de France" providing the performance of a number of favorite
racers. At the same time there is a broadcast coverage of the event,
providing a more global overview. Now, when I am "on the road" I watch =
my
own favorite racer on my mobile device. When at home, I watch the broad=
cast
coverage on my TV. This TV has the capability to also receive mobile
services. Over the normal broadcast channel an MPEG-4 application is
delivered that allows me to display the multicast streaming of my favor=
ite
racer on TV, as an overlay over the broadcast video. For this purpose, =
some
information is needed that is contained in an SL header. In order to be=

bandwidth efficient on the mobile network, the same multicast session i=
s
used as for mobile devices.

Now follows an essential assumption: the mobile devices are not_upgrada=
ble;
these dedicated devices are not capable to even process a single byte t=
o
upgrade their functionality. If this assumption is true, they need the
capability to accept the additional bytes from day one. Otherwise two
separate multicast sessions are needed.

In my understanding there are no higher level protocols involved. And t=
he
essential question is whether these mobile devices are indeed not
upgradable. It is my guess that upgradability is not a requirement for =
such
devices, but only a feature that cannot be relied upon.

I hope this clarifies.

Kind regards,

Jan

=



From rem-conf Tue Sep 12 07:27:32 2000 
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To: <csp@east.isi.edu>
Cc: <HerpelC@thmulti.com>, <stewe@cs.tu-berlin.de>, <rem-conf@es.net>,
        <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the
         last call
Message-ID: <0056890014029423000002L932*@MHS>
Date: Tue, 12 Sep 2000 16:27:22 +0200
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Colin,

The composition info is sent with the broadcast; with the mobile servic=
e
only the information needed to allow for the composition is sent.

Regards,

Jan

-----Original Message-----
From:	csp@east.isi.edu [mailto:csp@east.isi.edu]
Sent:	dinsdag, 12 september, 2000 16:22
To:	Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net;
stewe@cs.tu-berlin.de; HerpelC@thmulti.com
Subject:	Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response=
 to
the last call


I don't understand why you don't simply send the composition informatio=
n
along with the broadcast signal? Sending it with the mobile signal (whe=
re
it is useless unless the broadcast signal is also received) seems to be=
 a
waste of mobile capacity.

Colin


--> jan.vandermeer@philips.com writes:
>Stephan, Carsten, Colin, Zvi and others,
>
>It surely is a clash of worlds. And I think we did not come to full
>understanding yet. I think I understand the point made by Stephan, but=
 I
>also think that Carsten did make the real point. Let me explain below =
with
>an example.
>
>A service provides several multicast sessions to mobile devices using =
the
>simple MPEG-4 ES payload format. This is done during a sport event suc=
h as
>the "Tour de France" providing the performance of a number of favorite=

>racers. At the same time there is a broadcast coverage of the event,
>providing a more global overview. Now, when I am "on the road" I watch=
 my
>own favorite racer on my mobile device. When at home, I watch the broa=
dcast
>coverage on my TV. This TV has the capability to also receive mobile
>services. Over the normal broadcast channel an MPEG-4 application is
>delivered that allows me to display the multicast streaming of my favo=
rite
>racer on TV, as an overlay over the broadcast video. For this purpose,=
 some
>information is needed that is contained in an SL header. In order to b=
e
>bandwidth efficient on the mobile network, the same multicast session =
is
>used as for mobile devices.
>
>Now follows an essential assumption: the mobile devices are not_upgrad=
able;
>these dedicated devices are not capable to even process a single byte =
to
>upgrade their functionality. If this assumption is true, they need the=

>capability to accept the additional bytes from day one. Otherwise two
>separate multicast sessions are needed.
>
>In my understanding there are no higher level protocols involved. And =
the
>essential question is whether these mobile devices are indeed not
>upgradable. It is my guess that upgradability is not a requirement for=
 such
>devices, but only a feature that cannot be relied upon.
>
>I hope this clarifies.
>
>Kind regards,
>
>Jan
>
>-----Original Message-----
>From:	stewe@cs.tu-berlin.de [mailto:stewe@cs.tu-berlin.de]
>Sent:	maandag, 11 september, 2000 21:05
>To:	HerpelC@thmulti.com
>Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
>Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in respons=
e to
>the last call
>
>
>Folks
>
>>What happens here is a clash of worlds.
>
>Clearly, it is!
>
>But I don't think that Carsten came to the real point here.
>
>IMHO, the main point is that MPEG-folk seem to think that
>the media stream (RTP/payload/MPEG) should be self-contained
>and  independent from the higher protocol layers that perform the
>capability exchange/announcement (and other things such
>as call establishment etc.).
>IETF/AVT folk (including me), however, make the implicit
>assumption that media cannot be seen independently from
>higher protocols.
>
>The latter, however, is not an assumption -- its the plain truth,
>especially after AVT made it the general policy to no more
>assign static payload types for media coding.
>
>This means, that RTP systems rely (and cannot live without)
>some external form of capability exchange/announcement, which,
>as a minimum, dynamically (per call) associates a freely chosen
>number with a certain payload (e.g. MPEG-4 ES or, in the future,
>MPEG-4 generic/SL/whatever).  The information which payload is
>to be transmitted is in H.245, for example, coded in ASN.1.  For SIP
>systems, the relevant information is transmitted in ASCII form,
>where the payload types (such as MPEG-4-ES) has typically
>a MIME registration.
>
>That is why Colin and me and others insist that there is no need to
>duplicate this type of information in the media stream.  Every
>system that uses RTP in a standard-compliant way has already
>the functionality build in that is necessary to distinguish between
>different payload types.
>
>>One assumption is that huge numbers of devices will be out there that=
 use
>>the ES payload format. Assuming that these devices cannot be upgraded=
 to a
>>newer payload format!
>>Then applications come up with more functionality, using MPEG-4 Syste=
ms.
>New
>>devices will exist that know how to use this functionality. Old devic=
es
>>should at least be able to understand the (unmodified!) MPEG-4 Video =
over
>>RTP stream. Since the old devices are not reprogrammable, they cannot=
 be
>>upgraded to another payload format. So the old format needs to be by
design
>>compatible to any new format.
>
>Very simple to achieve.  Have two payload types, MPEG-4-SL and MPEG-4-=
ES.
>Try to negotiate MPEG-4-SL.  If the other system rejects, use MPEG-4-E=
S.
>This is how things are done, at least in H.323, all the time.
>
>Stephan
>

=



From rem-conf Tue Sep 12 07:28:23 2000 
From rem-conf-request@es.net Tue Sep 12 07:28:23 2000
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Message-ID: <D33D3145480ED4119EEB000629398551908164@optmail.optibase.co.il>
From: Zvi Lifshitz <ZviL@optibase.com>
To: jan.vandermeer@philips.com
Cc: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to
	 the last call
Date: Tue, 12 Sep 2000 17:28:53 +0200
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The ES payload format is not configurable and will never have extra SL
header fields, so what is the future functionality of this format you have
in mind?

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Tuesday, September 12, 2000 4:21 PM
To:	ZviL@optibase.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to the last call

 
Zvi,

It is not another payload type; it's the MPEG-4 ES one, but with some other
bytes which are not understood by the mobile device but which are allowed to
be there.

Regards,

Jan

-----Original Message-----
From:	ZviL@optibase.com [mailto:ZviL@optibase.com]
Sent:	dinsdag, 12 september, 2000 16:09
To:	ZviL@optibase.com; Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to
the last call


It will not help, because when you get the new payload in the future the old
devices don't recognize the payload type and will not handle it.

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com <mailto:zvil@csi.com> 			Fax
+972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
IBC Amsterdam RAI, Booth # 2.251,  September 8th-12, 2000
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Tuesday, September 12, 2000 3:51 PM
To:	ZviL@optibase.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to the last call

Dear Zvi,

I thought I did. What I try to say is: to have this type of functionality in
future, you need to allow for inclusion of the additional bytes in the
MPEG-2 ES payload now.

Regards,

Jan

-----Original Message-----
From:	ZviL@optibase.com [mailto:ZviL@optibase.com]
Sent:	dinsdag, 12 september, 2000 13:23
To:	stewe@cs.tu-berlin.de; HerpelC@thmulti.com; Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to
the last call


Dear Jan,

You clarified the scenario, but not how it relates to the discussion.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com <mailto:zvil@csi.com> 			Fax
+972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
IBC Amsterdam RAI, Booth # 2.251,  September 8th-12, 2000
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Tuesday, September 12, 2000 12:44 PM
To:	HerpelC@thmulti.com; stewe@cs.tu-berlin.de
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to the last call


Stephan, Carsten, Colin, Zvi and others,

It surely is a clash of worlds. And I think we did not come to full
understanding yet. I think I understand the point made by Stephan, but I
also think that Carsten did make the real point. Let me explain below with
an example.

A service provides several multicast sessions to mobile devices using the
simple MPEG-4 ES payload format. This is done during a sport event such as
the "Tour de France" providing the performance of a number of favorite
racers. At the same time there is a broadcast coverage of the event,
providing a more global overview. Now, when I am "on the road" I watch my
own favorite racer on my mobile device. When at home, I watch the broadcast
coverage on my TV. This TV has the capability to also receive mobile
services. Over the normal broadcast channel an MPEG-4 application is
delivered that allows me to display the multicast streaming of my favorite
racer on TV, as an overlay over the broadcast video. For this purpose, some
information is needed that is contained in an SL header. In order to be
bandwidth efficient on the mobile network, the same multicast session is
used as for mobile devices.

Now follows an essential assumption: the mobile devices are not_upgradable;
these dedicated devices are not capable to even process a single byte to
upgrade their functionality. If this assumption is true, they need the
capability to accept the additional bytes from day one. Otherwise two
separate multicast sessions are needed.

In my understanding there are no higher level protocols involved. And the
essential question is whether these mobile devices are indeed not
upgradable. It is my guess that upgradability is not a requirement for such
devices, but only a feature that cannot be relied upon.

I hope this clarifies.

Kind regards,

Jan



From rem-conf Tue Sep 12 07:44:37 2000 
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Cc: <rem-conf@es.net>, <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the
         last call
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Zvi,

I would like to allow for some optional bytes that proceed the ES paylo=
ad,
e.g. starting with a byte signaling the number of optional bytes. But t=
hat
would burn at least always one byte signaling no more bytes. Therefore =
it
would be better to find another signaling method, e.g. two payload type=
s of
MPEG-4 ES, one with and one without the additional bytes. Does not look=

elegant of course and therefore I am still hoping for a better solution=
.


Jan

-----Original Message-----
From:	ZviL@optibase.com [mailto:ZviL@optibase.com]
Sent:	dinsdag, 12 september, 2000 16:31
To:	Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response=
 to
the last call


The ES payload format is not configurable and will never have extra SL
header fields, so what is the future functionality of this format you h=
ave
in mind?

Regards,

z
soon the whole world will STREAM
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October=

10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Tuesday, September 12, 2000 4:21 PM
To:	ZviL@optibase.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to the last call


Zvi,

It is not another payload type; it's the MPEG-4 ES one, but with some o=
ther
bytes which are not understood by the mobile device but which are allow=
ed to
be there.

Regards,

Jan

-----Original Message-----
From:	ZviL@optibase.com [mailto:ZviL@optibase.com]
Sent:	dinsdag, 12 september, 2000 16:09
To:	ZviL@optibase.com; Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to
the last call


It will not help, because when you get the new payload in the future th=
e old
devices don't recognize the payload type and will not handle it.

z
soon the whole world will STREAM
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com <mailto:zvil@csi.com> 			Fax
+972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> Come see us at:
IBC Amsterdam RAI, Booth # 2.251,  September 8th-12, 2000
Streaming Media Europe Earls Court Centre, Booth # 628, London, October=

10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Tuesday, September 12, 2000 3:51 PM
To:	ZviL@optibase.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to the last call

Dear Zvi,

I thought I did. What I try to say is: to have this type of functionali=
ty in
future, you need to allow for inclusion of the additional bytes in the
MPEG-2 ES payload now.

Regards,

Jan

-----Original Message-----
From:	ZviL@optibase.com [mailto:ZviL@optibase.com]
Sent:	dinsdag, 12 september, 2000 13:23
To:	stewe@cs.tu-berlin.de; HerpelC@thmulti.com; Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to
the last call


Dear Jan,

You clarified the scenario, but not how it relates to the discussion.

Regards,

z
soon the whole world will STREAM
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com <mailto:zvil@csi.com> 			Fax
+972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> Come see us at:
IBC Amsterdam RAI, Booth # 2.251,  September 8th-12, 2000
Streaming Media Europe Earls Court Centre, Booth # 628, London, October=

10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Tuesday, September 12, 2000 12:44 PM
To:	HerpelC@thmulti.com; stewe@cs.tu-berlin.de
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to the last call


Stephan, Carsten, Colin, Zvi and others,

It surely is a clash of worlds. And I think we did not come to full
understanding yet. I think I understand the point made by Stephan, but =
I
also think that Carsten did make the real point. Let me explain below w=
ith
an example.

A service provides several multicast sessions to mobile devices using t=
he
simple MPEG-4 ES payload format. This is done during a sport event such=
 as
the "Tour de France" providing the performance of a number of favorite
racers. At the same time there is a broadcast coverage of the event,
providing a more global overview. Now, when I am "on the road" I watch =
my
own favorite racer on my mobile device. When at home, I watch the broad=
cast
coverage on my TV. This TV has the capability to also receive mobile
services. Over the normal broadcast channel an MPEG-4 application is
delivered that allows me to display the multicast streaming of my favor=
ite
racer on TV, as an overlay over the broadcast video. For this purpose, =
some
information is needed that is contained in an SL header. In order to be=

bandwidth efficient on the mobile network, the same multicast session i=
s
used as for mobile devices.

Now follows an essential assumption: the mobile devices are not_upgrada=
ble;
these dedicated devices are not capable to even process a single byte t=
o
upgrade their functionality. If this assumption is true, they need the
capability to accept the additional bytes from day one. Otherwise two
separate multicast sessions are needed.

In my understanding there are no higher level protocols involved. And t=
he
essential question is whether these mobile devices are indeed not
upgradable. It is my guess that upgradability is not a requirement for =
such
devices, but only a feature that cannot be relied upon.

I hope this clarifies.

Kind regards,

Jan

=



From rem-conf Tue Sep 12 08:01:21 2000 
From rem-conf-request@es.net Tue Sep 12 08:01:20 2000
Received: from list by mail1.es.net with local (Exim 1.81 #2)
	id 13YrWf-0000DO-00; Tue, 12 Sep 2000 07:59:09 -0700
Received: from optmail.optibase.co.il [199.203.75.17] 
	by mail1.es.net with esmtp (Exim 1.81 #2)
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Received: by optmail.optibase.co.il with Internet Mail Service (5.5.2650.21)
	id <S48NLC6L>; Tue, 12 Sep 2000 18:00:04 +0200
Message-ID: <D33D3145480ED4119EEB000629398551908166@optmail.optibase.co.il>
From: Zvi Lifshitz <ZviL@optibase.com>
To: jan.vandermeer@philips.com
Cc: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to
	 the last call
Date: Tue, 12 Sep 2000 18:00:03 +0200
MIME-Version: 1.0
X-Mailer: Internet Mail Service (5.5.2650.21)
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Dear Jan,

I think there is a confusion. The ES format has been proceeded as an
Internet Draft and there is no more time for comments. I don't think it's
possible to make this format extendable, because, first, I'm not sure this
is at all possible in IETF. Even if it is, text on extendibility, both
syntactically and semantically must be included in the draft, and it is not.
So the only thing we can do is proposing that in the Generic format the
extra SL header that is contained in the payload will be preceded by a
length byte, so it will be readable by systems that don't use MPEG-4
Systems.

In any case, the request to have an extra byte in the ES format makes,
IMNSHO, no sense.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Tuesday, September 12, 2000 4:45 PM
To:	ZviL@optibase.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to the last call

 
Zvi,

I would like to allow for some optional bytes that proceed the ES payload,
e.g. starting with a byte signaling the number of optional bytes. But that
would burn at least always one byte signaling no more bytes. Therefore it
would be better to find another signaling method, e.g. two payload types of
MPEG-4 ES, one with and one without the additional bytes. Does not look
elegant of course and therefore I am still hoping for a better solution.


Jan

-----Original Message-----
From:	ZviL@optibase.com [mailto:ZviL@optibase.com]
Sent:	dinsdag, 12 september, 2000 16:31
To:	Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to
the last call


The ES payload format is not configurable and will never have extra SL
header fields, so what is the future functionality of this format you have
in mind?

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Tuesday, September 12, 2000 4:21 PM
To:	ZviL@optibase.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to the last call


Zvi,

It is not another payload type; it's the MPEG-4 ES one, but with some other
bytes which are not understood by the mobile device but which are allowed to
be there.

Regards,

Jan

-----Original Message-----
From:	ZviL@optibase.com [mailto:ZviL@optibase.com]
Sent:	dinsdag, 12 september, 2000 16:09
To:	ZviL@optibase.com; Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to
the last call


It will not help, because when you get the new payload in the future the old
devices don't recognize the payload type and will not handle it.

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com <mailto:zvil@csi.com> 			Fax
+972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
IBC Amsterdam RAI, Booth # 2.251,  September 8th-12, 2000
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Tuesday, September 12, 2000 3:51 PM
To:	ZviL@optibase.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to the last call

Dear Zvi,

I thought I did. What I try to say is: to have this type of functionality in
future, you need to allow for inclusion of the additional bytes in the
MPEG-2 ES payload now.

Regards,

Jan

-----Original Message-----
From:	ZviL@optibase.com [mailto:ZviL@optibase.com]
Sent:	dinsdag, 12 september, 2000 13:23
To:	stewe@cs.tu-berlin.de; HerpelC@thmulti.com; Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to
the last call


Dear Jan,

You clarified the scenario, but not how it relates to the discussion.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com <mailto:zvil@csi.com> 			Fax
+972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
IBC Amsterdam RAI, Booth # 2.251,  September 8th-12, 2000
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Tuesday, September 12, 2000 12:44 PM
To:	HerpelC@thmulti.com; stewe@cs.tu-berlin.de
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to the last call


Stephan, Carsten, Colin, Zvi and others,

It surely is a clash of worlds. And I think we did not come to full
understanding yet. I think I understand the point made by Stephan, but I
also think that Carsten did make the real point. Let me explain below with
an example.

A service provides several multicast sessions to mobile devices using the
simple MPEG-4 ES payload format. This is done during a sport event such as
the "Tour de France" providing the performance of a number of favorite
racers. At the same time there is a broadcast coverage of the event,
providing a more global overview. Now, when I am "on the road" I watch my
own favorite racer on my mobile device. When at home, I watch the broadcast
coverage on my TV. This TV has the capability to also receive mobile
services. Over the normal broadcast channel an MPEG-4 application is
delivered that allows me to display the multicast streaming of my favorite
racer on TV, as an overlay over the broadcast video. For this purpose, some
information is needed that is contained in an SL header. In order to be
bandwidth efficient on the mobile network, the same multicast session is
used as for mobile devices.

Now follows an essential assumption: the mobile devices are not_upgradable;
these dedicated devices are not capable to even process a single byte to
upgrade their functionality. If this assumption is true, they need the
capability to accept the additional bytes from day one. Otherwise two
separate multicast sessions are needed.

In my understanding there are no higher level protocols involved. And the
essential question is whether these mobile devices are indeed not
upgradable. It is my guess that upgradability is not a requirement for such
devices, but only a feature that cannot be relied upon.

I hope this clarifies.

Kind regards,

Jan



From rem-conf Tue Sep 12 08:16:02 2000 
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To: <ZviL@optibase.com>
Cc: <rem-conf@es.net>, <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the
         last call
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Date: Tue, 12 Sep 2000 17:15:18 +0200
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Zvi,

I don't think there is any confusion. If no such bytes are allowed, the=

scenario I gave will not be possible. Which in my view would be a mista=
ke.
>From MPEG point of view a serious mistake.

To allow for extensibility in the "generic format" will solve problems,=
 but
not this one

Regards,

Jan


-----Original Message-----
From:	ZviL@optibase.com [mailto:ZviL@optibase.com]
Sent:	dinsdag, 12 september, 2000 17:04
To:	Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response=
 to
the last call


Dear Jan,

I think there is a confusion. The ES format has been proceeded as an
Internet Draft and there is no more time for comments. I don't think it=
's
possible to make this format extendable, because, first, I'm not sure t=
his
is at all possible in IETF. Even if it is, text on extendibility, both
syntactically and semantically must be included in the draft, and it is=
 not.
So the only thing we can do is proposing that in the Generic format the=

extra SL header that is contained in the payload will be preceded by a
length byte, so it will be readable by systems that don't use MPEG-4
Systems.

In any case, the request to have an extra byte in the ES format makes,
IMNSHO, no sense.

Regards,

z
soon the whole world will STREAM
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

=



From rem-conf Tue Sep 12 08:27:48 2000 
From rem-conf-request@es.net Tue Sep 12 08:27:47 2000
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From: Zvi Lifshitz <ZviL@optibase.com>
To: jan.vandermeer@philips.com
Cc: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to
	 the last call
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The scenario you gave is not possible, for the reasons I previously
explained.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Tuesday, September 12, 2000 5:15 PM
To:	ZviL@optibase.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to the last call

 
Zvi,

I don't think there is any confusion. If no such bytes are allowed, the
scenario I gave will not be possible. Which in my view would be a mistake.
>From MPEG point of view a serious mistake.

To allow for extensibility in the "generic format" will solve problems, but
not this one

Regards,

Jan




From rem-conf Tue Sep 12 08:39:22 2000 
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Cc: <rem-conf@es.net>, <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the
         last call
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Which is not a mistake ????

Regards,

Jan

-----Original Message-----
From:	ZviL@optibase.com [mailto:ZviL@optibase.com]
Sent:	dinsdag, 12 september, 2000 17:33
To:	Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response=
 to
the last call


The scenario you gave is not possible, for the reasons I previously
explained.

Regards,

z
soon the whole world will STREAM
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October=

10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Tuesday, September 12, 2000 5:15 PM
To:	ZviL@optibase.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to the last call


Zvi,

I don't think there is any confusion. If no such bytes are allowed, the=

scenario I gave will not be possible. Which in my view would be a mista=
ke.
>From MPEG point of view a serious mistake.

To allow for extensibility in the "generic format" will solve problems,=
 but
not this one

Regards,

Jan

=



From rem-conf Tue Sep 12 08:58:30 2000 
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Date: Tue, 12 Sep 2000 11:58:45 -0400
From: "Narasimhan, Sam (SD-EX)" <SNarasimhan@gi.com>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to	the
 last call
To: "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>,
 ZviL@optibase.com
Cc: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
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Zvi,

 Let me see if I can capture what Jan is proposing as a change
to Kikuchi-San's draft. For the same payload type (MPEG4 Video ES), 
add an adaptation header length followed by the length bytes before
MPEG-4 visual stream payload.


   0                   1                   2                   3
   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |V=2|P|X|  CC   |M|     PT      |       sequence number         | RTP
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                           timestamp                           | Header
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |           synchronization source (SSRC) identifier            |
   +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
   |            contributing source (CSRC) identifiers             |
   |                             ....                              |
   +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
   |adaptation length +--length--+                                 |

   |                                                               |  RTP
   |                                                               |
   |                                                               |
   |       MPEG-4 Visual stream (byte aligned)                     | Payload
   |                                                               |
   |                               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                               :...OPTIONAL RTP padding        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

        Figure 1 - An RTP packet for MPEG-4 Visual stream

 If this draft were to say that receivers should parse over the
adaptation bytes and not process them (as this draft does not
define these), then such receivers will be able to process payload
>from future draft (from MPEG) which included meaningful adaptation
field data as these receivers will be able to 'parse over' the
adaptation and get the ES payload. If they cannot 'parse over' 
such field, then they will crash. Parsing over such data is not 
very complex and is easy to implement. 

 If Kikuchi-San's draft can make this change, then it may become
fully compatible with what the MPEG AHG may define as 'generic'
format. It is always easy to define 'hundred's' of payload types;
however service providers want to reach as many receivers as
they can and this change will make receivers that implement Kikuchi-San's
proposal receive properly generated data in the future that may
include adaptation header.

 If this change cannot be made for some process reason's, then we need
to accept the fact that this draft will not be compatible with what MPEG
will
define for a generic format as there is no place to even carry DTS in this
draft.
 
Best Regards
Sam Narasimhan
Motorola

-----Original Message-----
From: jan.vandermeer@philips.com [mailto:jan.vandermeer@philips.com]
Sent: Tuesday, September 12, 2000 8:15 AM
To: ZviL@optibase.com
Cc: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
to the last call


Zvi,

I don't think there is any confusion. If no such bytes are allowed, the
scenario I gave will not be possible. Which in my view would be a mistake.
>From MPEG point of view a serious mistake.

To allow for extensibility in the "generic format" will solve problems, but
not this one

Regards,

Jan


-----Original Message-----
From:	ZviL@optibase.com [mailto:ZviL@optibase.com]
Sent:	dinsdag, 12 september, 2000 17:04
To:	Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to
the last call


Dear Jan,

I think there is a confusion. The ES format has been proceeded as an
Internet Draft and there is no more time for comments. I don't think it's
possible to make this format extendable, because, first, I'm not sure this
is at all possible in IETF. Even if it is, text on extendibility, both
syntactically and semantically must be included in the draft, and it is not.
So the only thing we can do is proposing that in the Generic format the
extra SL header that is contained in the payload will be preceded by a
length byte, so it will be readable by systems that don't use MPEG-4
Systems.

In any case, the request to have an extra byte in the ES format makes,
IMNSHO, no sense.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================



From rem-conf Tue Sep 12 09:03:13 2000 
From rem-conf-request@es.net Tue Sep 12 09:03:12 2000
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From: Zvi Lifshitz <ZviL@optibase.com>
To: jan.vandermeer@philips.com
Cc: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to
	 the last call
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Mistake is when you can do something and you do it wrong. Here we could not
do anything, except asking to reject the Kikuci-san proposal and start
another one. Is that what you wanted?

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Tuesday, September 12, 2000 5:39 PM
To:	ZviL@optibase.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to the last call

 
Which is not a mistake ????

Regards,

Jan

-----Original Message-----
From:	ZviL@optibase.com [mailto:ZviL@optibase.com]
Sent:	dinsdag, 12 september, 2000 17:33
To:	Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to
the last call


The scenario you gave is not possible, for the reasons I previously
explained.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Tuesday, September 12, 2000 5:15 PM
To:	ZviL@optibase.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to the last call


Zvi,

I don't think there is any confusion. If no such bytes are allowed, the
scenario I gave will not be possible. Which in my view would be a mistake.
>From MPEG point of view a serious mistake.

To allow for extensibility in the "generic format" will solve problems, but
not this one

Regards,

Jan



From rem-conf Tue Sep 12 09:07:41 2000 
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To: "Narasimhan, Sam (SD-EX)" <SNarasimhan@gi.com>
cc: "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>,
        ZviL@optibase.com, rem-conf@es.net,
        4on2andIP-sys@advent.ee.columbia.edu
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
In-Reply-To: Your message of "Tue, 12 Sep 2000 11:58:45 EDT."
             <973597126BDDD11197AA00805FA7EBC9034239A2@ntas0026.gi.com> 
Date: Tue, 12 Sep 2000 12:06:54 -0400
From: Colin Perkins <csp@east.isi.edu>
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Sam,

There is no process problem, except that delaying the Kikuchi-san draft
will prevent MPEG-4 ES being used with H.323 systems. 

My main issue is technical: I think that the addition of this extra field
is a mistake, and that the correct approach is to use the payload type to
indicate that the media data transported is in a different format. I have
yet to hear a convincing argument otherwise.

Colin

--> "Narasimhan, Sam (SD-EX)" writes:
>Zvi,
>
> Let me see if I can capture what Jan is proposing as a change
>to Kikuchi-San's draft. For the same payload type (MPEG4 Video ES), 
>add an adaptation header length followed by the length bytes before
>MPEG-4 visual stream payload.
>
>
>   0                   1                   2                   3
>   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
>   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>   |V=2|P|X|  CC   |M|     PT      |       sequence number         | RTP
>   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>   |                           timestamp                           | Header
>   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>   |           synchronization source (SSRC) identifier            |
>   +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
>   |            contributing source (CSRC) identifiers             |
>   |                             ....                              |
>   +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
>   |adaptation length +--length--+                                 |
>
>   |                                                               |  RTP
>   |                                                               |
>   |                                                               |
>   |       MPEG-4 Visual stream (byte aligned)                     | Payload
>   |                                                               |
>   |                               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>   |                               :...OPTIONAL RTP padding        |
>   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>
>        Figure 1 - An RTP packet for MPEG-4 Visual stream
>
> If this draft were to say that receivers should parse over the
>adaptation bytes and not process them (as this draft does not
>define these), then such receivers will be able to process payload
>from future draft (from MPEG) which included meaningful adaptation
>field data as these receivers will be able to 'parse over' the
>adaptation and get the ES payload. If they cannot 'parse over' 
>such field, then they will crash. Parsing over such data is not 
>very complex and is easy to implement. 
>
> If Kikuchi-San's draft can make this change, then it may become
>fully compatible with what the MPEG AHG may define as 'generic'
>format. It is always easy to define 'hundred's' of payload types;
>however service providers want to reach as many receivers as
>they can and this change will make receivers that implement Kikuchi-San's
>proposal receive properly generated data in the future that may
>include adaptation header.
>
> If this change cannot be made for some process reason's, then we need
>to accept the fact that this draft will not be compatible with what MPEG
>will
>define for a generic format as there is no place to even carry DTS in this
>draft.
> 
>Best Regards
>Sam Narasimhan
>Motorola
>
>-----Original Message-----
>From: jan.vandermeer@philips.com [mailto:jan.vandermeer@philips.com]
>Sent: Tuesday, September 12, 2000 8:15 AM
>To: ZviL@optibase.com
>Cc: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
>Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
>to the last call
>
>
>Zvi,
>
>I don't think there is any confusion. If no such bytes are allowed, the
>scenario I gave will not be possible. Which in my view would be a mistake.
>>From MPEG point of view a serious mistake.
>
>To allow for extensibility in the "generic format" will solve problems, but
>not this one
>
>Regards,
>
>Jan
>
>
>-----Original Message-----
>From:	ZviL@optibase.com [mailto:ZviL@optibase.com]
>Sent:	dinsdag, 12 september, 2000 17:04
>To:	Jan vanderMeer
>Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
>Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
>response to
>the last call
>
>
>Dear Jan,
>
>I think there is a confusion. The ES format has been proceeded as an
>Internet Draft and there is no more time for comments. I don't think it's
>possible to make this format extendable, because, first, I'm not sure this
>is at all possible in IETF. Even if it is, text on extendibility, both
>syntactically and semantically must be included in the draft, and it is not.
>So the only thing we can do is proposing that in the Generic format the
>extra SL header that is contained in the payload will be preceded by a
>length byte, so it will be readable by systems that don't use MPEG-4
>Systems.
>
>In any case, the request to have an extra byte in the ES format makes,
>IMNSHO, no sense.
>
>Regards,
>
>z
>soon the whole world will STREAM
>==================================================================
>Zvi Lifshitz				Phone +972(2)679-4788
>zvil@optibase.com			Fax +972(2)679-4789
>Optibase Ltd.				Mobile: +972(53)461-787
>http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
>Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
>==================================================================



From rem-conf Tue Sep 12 09:18:05 2000 
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Date: Tue, 12 Sep 2000 12:17:54 -0400
From: "Luthra, Ajay (SD-EX)" <ALuthra@gi.com>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to	the
 last call
To: 'Zvi Lifshitz' <ZviL@optibase.com>, jan.vandermeer@philips.com
Cc: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
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Dear Zvi,
  Now, I am confused by your statement.  As we discussed in Beijing, the ES
format draft being considered by IETF is proposed by individuals and we can
not and should not control what they input to IETF. That is IETF's business.
What we had agreed to do in Beijing was to come up with WG-11's
recommendation and contribution on how to carry ES (as well as SL and other
streams types) in RTP and to let IETF know that that contribution is coming
(by Oct 2000) and make them aware that WG-11 contribution may or may not
align with what is being considered by IETF.  As I understand, WG-11 has
done that.

  So, as agreed in Beijing, compatibility with what individual contributions
IETF is considering is not a requirement (desired but not required).
Therefore, if we think we need additional byte then let us put it in WG-11's
contribution to IETF on how to carry ES. The discussion should be on whether
we need additional bytes or not, and not whether it is compatible with what
was proposed to IETF by some other authors. I know it is a mess and is
partly (major part) due to our fault in WG-11 and partly (minor part) due to
IETF process. In Beijing, WG-11 had asked people to not to submit individual
contributions and/or withdraw their inputs until this matter is fully
discussed in WG-11. The decision to freeze or withdraw is not WG-11's. There
is nothing we can do about it at this stage. It is water under the bridge. 

  So, let us just discuss whether the additional bytes, as proposed by Jan,
make sense or not. In that regard, I think it makes good sense. It gives you
lot of flexibility that will be needed in future. We are still reaping the
benefits of the flexibility of MPEG-2 systems. I really do not understand
the reason for not accepting it.

With regards,
Ajay


-----Original Message-----
From: Zvi Lifshitz [mailto:ZviL@optibase.com]
Sent: Tuesday, September 12, 2000 9:00 AM
To: jan.vandermeer@philips.com
Cc: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
to the last call


Dear Jan,

I think there is a confusion. The ES format has been proceeded as an
Internet Draft and there is no more time for comments. I don't think it's
possible to make this format extendable, because, first, I'm not sure this
is at all possible in IETF. Even if it is, text on extendibility, both
syntactically and semantically must be included in the draft, and it is not.
So the only thing we can do is proposing that in the Generic format the
extra SL header that is contained in the payload will be preceded by a
length byte, so it will be readable by systems that don't use MPEG-4
Systems.

In any case, the request to have an extra byte in the ES format makes,
IMNSHO, no sense.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Tuesday, September 12, 2000 4:45 PM
To:	ZviL@optibase.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to the last call

 
Zvi,

I would like to allow for some optional bytes that proceed the ES payload,
e.g. starting with a byte signaling the number of optional bytes. But that
would burn at least always one byte signaling no more bytes. Therefore it
would be better to find another signaling method, e.g. two payload types of
MPEG-4 ES, one with and one without the additional bytes. Does not look
elegant of course and therefore I am still hoping for a better solution.


Jan

-----Original Message-----
From:	ZviL@optibase.com [mailto:ZviL@optibase.com]
Sent:	dinsdag, 12 september, 2000 16:31
To:	Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to
the last call


The ES payload format is not configurable and will never have extra SL
header fields, so what is the future functionality of this format you have
in mind?

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Tuesday, September 12, 2000 4:21 PM
To:	ZviL@optibase.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to the last call


Zvi,

It is not another payload type; it's the MPEG-4 ES one, but with some other
bytes which are not understood by the mobile device but which are allowed to
be there.

Regards,

Jan

-----Original Message-----
From:	ZviL@optibase.com [mailto:ZviL@optibase.com]
Sent:	dinsdag, 12 september, 2000 16:09
To:	ZviL@optibase.com; Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to
the last call


It will not help, because when you get the new payload in the future the old
devices don't recognize the payload type and will not handle it.

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com <mailto:zvil@csi.com> 			Fax
+972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
IBC Amsterdam RAI, Booth # 2.251,  September 8th-12, 2000
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Tuesday, September 12, 2000 3:51 PM
To:	ZviL@optibase.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to the last call

Dear Zvi,

I thought I did. What I try to say is: to have this type of functionality in
future, you need to allow for inclusion of the additional bytes in the
MPEG-2 ES payload now.

Regards,

Jan

-----Original Message-----
From:	ZviL@optibase.com [mailto:ZviL@optibase.com]
Sent:	dinsdag, 12 september, 2000 13:23
To:	stewe@cs.tu-berlin.de; HerpelC@thmulti.com; Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to
the last call


Dear Jan,

You clarified the scenario, but not how it relates to the discussion.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com <mailto:zvil@csi.com> 			Fax
+972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
IBC Amsterdam RAI, Booth # 2.251,  September 8th-12, 2000
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Tuesday, September 12, 2000 12:44 PM
To:	HerpelC@thmulti.com; stewe@cs.tu-berlin.de
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to the last call


Stephan, Carsten, Colin, Zvi and others,

It surely is a clash of worlds. And I think we did not come to full
understanding yet. I think I understand the point made by Stephan, but I
also think that Carsten did make the real point. Let me explain below with
an example.

A service provides several multicast sessions to mobile devices using the
simple MPEG-4 ES payload format. This is done during a sport event such as
the "Tour de France" providing the performance of a number of favorite
racers. At the same time there is a broadcast coverage of the event,
providing a more global overview. Now, when I am "on the road" I watch my
own favorite racer on my mobile device. When at home, I watch the broadcast
coverage on my TV. This TV has the capability to also receive mobile
services. Over the normal broadcast channel an MPEG-4 application is
delivered that allows me to display the multicast streaming of my favorite
racer on TV, as an overlay over the broadcast video. For this purpose, some
information is needed that is contained in an SL header. In order to be
bandwidth efficient on the mobile network, the same multicast session is
used as for mobile devices.

Now follows an essential assumption: the mobile devices are not_upgradable;
these dedicated devices are not capable to even process a single byte to
upgrade their functionality. If this assumption is true, they need the
capability to accept the additional bytes from day one. Otherwise two
separate multicast sessions are needed.

In my understanding there are no higher level protocols involved. And the
essential question is whether these mobile devices are indeed not
upgradable. It is my guess that upgradability is not a requirement for such
devices, but only a feature that cannot be relied upon.

I hope this clarifies.

Kind regards,

Jan



From rem-conf Tue Sep 12 09:18:07 2000 
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From: Zvi Lifshitz <ZviL@optibase.com>
To: "Narasimhan, Sam (SD-EX)" <SNarasimhan@gi.com>, 
	"'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>
Cc: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to
		the last call
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[Narasimhan, Sam (SD-EX):]

If this draft were to say that receivers should parse over the
adaptation bytes and not process them (as this draft does not
define these), then such receivers will be able to process payload
>from future draft (from MPEG)

[Reply:]

They will not, because they don't know the payload type of the future draft
therefore they will not know they can process payloads from the future draft
even though they can.

If they know, they also know that the future draft has extra header info. So
work on the future draft, not on the current draft.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000





From rem-conf Tue Sep 12 09:18:08 2000 
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Date: Tue, 12 Sep 2000 12:19:05 -0400
From: "Narasimhan, Sam (SD-EX)" <SNarasimhan@gi.com>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to	the
 last call
To: 'Colin Perkins' <csp@east.isi.edu>,
 "Narasimhan, Sam (SD-EX)" <SNarasimhan@GI.COM>
Cc: "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>,
 ZviL@optibase.com, rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
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Colin,

 If the objective of receivers that implement this draft is to
only do one thing and process only one payload type, then there
is no change required. Those that are in the CE business want
to build receivers that can be fielded in many application
areas and will build receivers that will understand different
payload types and formats they are carried in. However, content
providers will not want to generate 2 separate streams (with 2
payload types) in order to reach 2 different receivers (one that
implemented Kikuchi-San's proposal and one that implemented
what Jan is proposing). They will probably use the more generic
format and reach those receivers that process these adaptation
headers.

 For Example, in US digital cable the receivers span a 5 year
deployment window and content providers create a single content 
that is decodable by ALL these different models. 

Best Regards
Sam Narasimhan

-----Original Message-----
From: Colin Perkins [mailto:csp@east.isi.edu]
Sent: Tuesday, September 12, 2000 9:07 AM
To: Narasimhan, Sam (SD-EX)
Cc: 'jan.vandermeer@philips.com'; ZviL@optibase.com; rem-conf@es.net;
4on2andIP-sys@advent.ee.columbia.edu
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
to the last call


Sam,

There is no process problem, except that delaying the Kikuchi-san draft
will prevent MPEG-4 ES being used with H.323 systems. 

My main issue is technical: I think that the addition of this extra field
is a mistake, and that the correct approach is to use the payload type to
indicate that the media data transported is in a different format. I have
yet to hear a convincing argument otherwise.

Colin

--> "Narasimhan, Sam (SD-EX)" writes:
>Zvi,
>
> Let me see if I can capture what Jan is proposing as a change
>to Kikuchi-San's draft. For the same payload type (MPEG4 Video ES), 
>add an adaptation header length followed by the length bytes before
>MPEG-4 visual stream payload.
>
>
>   0                   1                   2                   3
>   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
>   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>   |V=2|P|X|  CC   |M|     PT      |       sequence number         | RTP
>   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>   |                           timestamp                           | Header
>   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>   |           synchronization source (SSRC) identifier            |
>   +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
>   |            contributing source (CSRC) identifiers             |
>   |                             ....                              |
>   +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
>   |adaptation length +--length--+                                 |
>
>   |                                                               |  RTP
>   |                                                               |
>   |                                                               |
>   |       MPEG-4 Visual stream (byte aligned)                     |
Payload
>   |                                                               |
>   |                               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>   |                               :...OPTIONAL RTP padding        |
>   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>
>        Figure 1 - An RTP packet for MPEG-4 Visual stream
>
> If this draft were to say that receivers should parse over the
>adaptation bytes and not process them (as this draft does not
>define these), then such receivers will be able to process payload
>from future draft (from MPEG) which included meaningful adaptation
>field data as these receivers will be able to 'parse over' the
>adaptation and get the ES payload. If they cannot 'parse over' 
>such field, then they will crash. Parsing over such data is not 
>very complex and is easy to implement. 
>
> If Kikuchi-San's draft can make this change, then it may become
>fully compatible with what the MPEG AHG may define as 'generic'
>format. It is always easy to define 'hundred's' of payload types;
>however service providers want to reach as many receivers as
>they can and this change will make receivers that implement Kikuchi-San's
>proposal receive properly generated data in the future that may
>include adaptation header.
>
> If this change cannot be made for some process reason's, then we need
>to accept the fact that this draft will not be compatible with what MPEG
>will
>define for a generic format as there is no place to even carry DTS in this
>draft.
> 
>Best Regards
>Sam Narasimhan
>Motorola
>
>-----Original Message-----
>From: jan.vandermeer@philips.com [mailto:jan.vandermeer@philips.com]
>Sent: Tuesday, September 12, 2000 8:15 AM
>To: ZviL@optibase.com
>Cc: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
>Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
>to the last call
>
>
>Zvi,
>
>I don't think there is any confusion. If no such bytes are allowed, the
>scenario I gave will not be possible. Which in my view would be a mistake.
>>From MPEG point of view a serious mistake.
>
>To allow for extensibility in the "generic format" will solve problems, but
>not this one
>
>Regards,
>
>Jan
>
>
>-----Original Message-----
>From:	ZviL@optibase.com [mailto:ZviL@optibase.com]
>Sent:	dinsdag, 12 september, 2000 17:04
>To:	Jan vanderMeer
>Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
>Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
>response to
>the last call
>
>
>Dear Jan,
>
>I think there is a confusion. The ES format has been proceeded as an
>Internet Draft and there is no more time for comments. I don't think it's
>possible to make this format extendable, because, first, I'm not sure this
>is at all possible in IETF. Even if it is, text on extendibility, both
>syntactically and semantically must be included in the draft, and it is
not.
>So the only thing we can do is proposing that in the Generic format the
>extra SL header that is contained in the payload will be preceded by a
>length byte, so it will be readable by systems that don't use MPEG-4
>Systems.
>
>In any case, the request to have an extra byte in the ES format makes,
>IMNSHO, no sense.
>
>Regards,
>
>z
>soon the whole world will STREAM
>==================================================================
>Zvi Lifshitz				Phone +972(2)679-4788
>zvil@optibase.com			Fax +972(2)679-4789
>Optibase Ltd.				Mobile: +972(53)461-787
>http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
>Send SMS from:
http://isend.cellcom.co.il/English/Login.htm
>==================================================================



From rem-conf Tue Sep 12 09:19:32 2000 
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Date: Tue, 12 Sep 2000 12:20:37 -0400
From: "Narasimhan, Sam (SD-EX)" <SNarasimhan@gi.com>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to	the
 last call
To: 'Zvi Lifshitz' <ZviL@optibase.com>,
 "Narasimhan, Sam (SD-EX)" <SNarasimhan@GI.COM>,
 "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>
Cc: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
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Zvi,

 I am assuming that the payload type from the future draft
could be set to the same value as this draft, if this
draft can accommodate the change.

Regards
Sam Narasimhan
Motorola

-----Original Message-----
From: Zvi Lifshitz [mailto:ZviL@optibase.com]
Sent: Tuesday, September 12, 2000 10:18 AM
To: Narasimhan, Sam (SD-EX); 'jan.vandermeer@philips.com'
Cc: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
to the last call


[Narasimhan, Sam (SD-EX):]

If this draft were to say that receivers should parse over the
adaptation bytes and not process them (as this draft does not
define these), then such receivers will be able to process payload
>from future draft (from MPEG)

[Reply:]

They will not, because they don't know the payload type of the future draft
therefore they will not know they can process payloads from the future draft
even though they can.

If they know, they also know that the future draft has extra header info. So
work on the future draft, not on the current draft.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000




From rem-conf Tue Sep 12 09:24:35 2000 
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        4on2andIP-sys@advent.ee.columbia.edu
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
In-Reply-To: Your message of "Tue, 12 Sep 2000 12:19:05 EDT."
             <973597126BDDD11197AA00805FA7EBC9034239A3@ntas0026.gi.com> 
Date: Tue, 12 Sep 2000 12:24:02 -0400
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--> "Narasimhan, Sam (SD-EX)" writes:
>Colin,
>
> If the objective of receivers that implement this draft is to
>only do one thing and process only one payload type, then there
>is no change required. Those that are in the CE business want
>to build receivers that can be fielded in many application
>areas and will build receivers that will understand different
>payload types and formats they are carried in. However, content
>providers will not want to generate 2 separate streams (with 2
>payload types) in order to reach 2 different receivers (one that
>implemented Kikuchi-San's proposal and one that implemented
>what Jan is proposing). They will probably use the more generic
>format and reach those receivers that process these adaptation
>headers.
>
> For Example, in US digital cable the receivers span a 5 year
>deployment window and content providers create a single content 
>that is decodable by ALL these different models. 

I see nothing in Jan's proposal which makes this easier. 

Colin



From rem-conf Tue Sep 12 09:24:41 2000 
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cc: "'Zvi Lifshitz'" <ZviL@optibase.com>, jan.vandermeer@philips.com,
        rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
In-Reply-To: Your message of "Tue, 12 Sep 2000 12:17:54 EDT."
             <973597126BDDD11197AA00805FA7EBC903421643@ntas0026.gi.com> 
Date: Tue, 12 Sep 2000 12:24:05 -0400
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--> "Luthra, Ajay (SD-EX)" writes:
>Dear Zvi,
>  Now, I am confused by your statement.  As we discussed in Beijing, the ES
>format draft being considered by IETF is proposed by individuals and we can
>not and should not control what they input to IETF. That is IETF's business.
>What we had agreed to do in Beijing was to come up with WG-11's
>recommendation and contribution on how to carry ES (as well as SL and other
>streams types) in RTP and to let IETF know that that contribution is coming
>(by Oct 2000) and make them aware that WG-11 contribution may or may not
>align with what is being considered by IETF.  As I understand, WG-11 has
>done that.
>
>  So, as agreed in Beijing, compatibility with what individual contributions
>IETF is considering is not a requirement (desired but not required).
>Therefore, if we think we need additional byte then let us put it in WG-11's
>contribution to IETF on how to carry ES. The discussion should be on whether
>we need additional bytes or not, and not whether it is compatible with what
>was proposed to IETF by some other authors. I know it is a mess and is
>partly (major part) due to our fault in WG-11 and partly (minor part) due to
>IETF process. In Beijing, WG-11 had asked people to not to submit individual
>contributions and/or withdraw their inputs until this matter is fully
>discussed in WG-11. The decision to freeze or withdraw is not WG-11's. There
>is nothing we can do about it at this stage. It is water under the bridge. 
>
>  So, let us just discuss whether the additional bytes, as proposed by Jan,
>make sense or not. In that regard, I think it makes good sense. It gives you
>lot of flexibility that will be needed in future. We are still reaping the
>benefits of the flexibility of MPEG-2 systems. I really do not understand
>the reason for not accepting it.

Because we have a mechanism for extensibility in RTP via the payload type.
This extra byte provides no advantage - all the examples which have been 
proposed can be done without using this field - and is incompatible with 
existing practise.

Colin



From rem-conf Tue Sep 12 09:38:14 2000 
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From: Florissa Colina <florissa@bmrc.berkeley.edu>
Subject: 9/13 Making Interactive Workspaces Real -- Armando Fox,
  Computer Science Department, Stanford University
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Berkeley Multimedia, Graphics and Interfaces Seminar

Making Interactive Workspaces Real

Wednesday September 13, 2000,=20
1:10-2:30 p.m. PDT
Fujitsu Seminar Room (405 Soda Hall)=20

Armando Fox=20
Computer Science Department, Stanford University

The Stanford Interactive Workspaces project is developing the computing
foundations for bringing reality to ubiquitous computing. A diverse
collection of faculty and students from the areas of graphics,
human-computer interaction (HCI),networking,ubiquitous computing,and
databases, we are looking at new types of human/machine interaction
including multimodal input, heterogeneous device integration from
wall-sized displays to handheld information appliances, and the "invisibl=
e"
software infrastructure required to support them. In the same way that
today=92s standard operating systems make it feasible to write
single-workstation software that uses multiple devices and networked
resources,we are constructing a higher level operating system for the wor=
ld
of ubiquitous computing. We combine research on infrastructure (ways of
flexibly configuring and connecting devices, processes,and communication
links)with research on HCI (ways of interacting with heterogeneous changi=
ng
collections of devices with multiple modalities). I will describe our
progress on the software infrastructure, including applications running i=
n
our deployed iRoom that do in fact integrate a variety of devices.=20

---------------

The seminar is broadcast live on the Internet Mbone and as a Real Network=
s
G2 broadcast.

You can connect to the live RealNetworks broadcast at:
http://bmrc.berkeley.edu/bibs/cs298-5

You can also directly put in this url into the Real Player:=20
rtsp://media2.bmrc.berkeley.edu/encoder/cs298-5.rm

To do so you will need to have the "Real Player G2" installed, which is
available from:
http://www.real.com/products/player/downloadrealplayer.html

To tune into the Internet MBone broadcast, look for the announcement in
your MBone session directory program ('sdr').  If you are not receiving t=
he
announcement you may be able to receive the session by manually configuri=
ng
the client programs ('vic', and 'vat') with the session addresses:

low bit rate
	video: 233.0.25.1/22334
	audio: 233.0.25.2/22446

medium bit rate
	video: 233.0.25.129/22334
	audio: 233.0.25.2/22446

You can get further information about this seminar, and access to replays
of previous seminars at the MIG Seminar web page:

http://media2.bmrc.berkeley.edu/bibs/archive.cfm

Sponsored by the Berkeley Multimedia Research Center=20




From rem-conf Tue Sep 12 10:22:23 2000 
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To: "Narasimhan, Sam (SD-EX)" <SNarasimhan@gi.com>
cc: "'Zvi Lifshitz'" <ZviL@optibase.com>,
        "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>,
        rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
In-Reply-To: Your message of "Tue, 12 Sep 2000 12:20:37 EDT."
             <973597126BDDD11197AA00805FA7EBC9034239A4@ntas0026.gi.com> 
Date: Tue, 12 Sep 2000 13:19:53 -0400
From: Colin Perkins <csp@east.isi.edu>
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--> "Narasimhan, Sam (SD-EX)" writes:
> I am assuming that the payload type from the future draft
>could be set to the same value as this draft, if this
>draft can accommodate the change.

The payload type will have to be negotiated out of band anyway, there will
be no more static payload type assignments. Since there must be negotiation
having this extra byte in the payload format makes no sense.

Colin



From rem-conf Tue Sep 12 10:36:29 2000 
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To: "Luthra, Ajay (SD-EX)" <ALuthra@gi.com>
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        "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>,
        rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
In-Reply-To: Your message of "Tue, 12 Sep 2000 13:36:14 EDT."
             <973597126BDDD11197AA00805FA7EBC903421646@ntas0026.gi.com> 
Date: Tue, 12 Sep 2000 13:34:49 -0400
From: Colin Perkins <csp@east.isi.edu>
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--> "Luthra, Ajay (SD-EX)" writes:
>Will it work this way in Multicast environment also?

Yes.

Colin



From rem-conf Tue Sep 12 10:36:29 2000 
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From: "Luthra, Ajay (SD-EX)" <ALuthra@gi.com>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to	the
 last call
To: 'Colin Perkins' <csp@east.isi.edu>
Cc: 'Zvi Lifshitz' <ZviL@optibase.com>,
 "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>, rem-conf@es.net,
 4on2andIP-sys@advent.ee.columbia.edu
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Will it work this way in Multicast environment also?

Ajay

-----Original Message-----
From: Colin Perkins [mailto:csp@east.isi.edu]
Sent: Tuesday, September 12, 2000 10:20 AM
To: Narasimhan, Sam (SD-EX)
Cc: 'Zvi Lifshitz'; 'jan.vandermeer@philips.com'; rem-conf@es.net;
4on2andIP-sys@advent.ee.columbia.edu
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
to the last call


--> "Narasimhan, Sam (SD-EX)" writes:
> I am assuming that the payload type from the future draft
>could be set to the same value as this draft, if this
>draft can accommodate the change.

The payload type will have to be negotiated out of band anyway, there will
be no more static payload type assignments. Since there must be negotiation
having this extra byte in the payload format makes no sense.

Colin



From rem-conf Tue Sep 12 11:20:22 2000 
From rem-conf-request@es.net Tue Sep 12 11:20:21 2000
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	id 13Yuca-0001eN-00; Tue, 12 Sep 2000 11:17:29 -0700
From:     Suntan - Wilfred <info@suntan.com.hk>
To:        <rem-conf@es.net>
Message-Id: <419.436782.09869606info@suntan.com.hk>
Subject: Do you need Capacitors ?
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Dear Sir / Madam,

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factory 
in Chinese Mainland - Polyester, Metallized, Polypropylene, Polystyrene. And also handle all kinds of 
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***********************************************************
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Unit A & B, 12/F., Everest Industrial Centre, 396 Kwun Tong Road,
Kwun Tong Road, Kowloon, Hong Kong.
Tel       : (852) 2343 0268
Fax      : (852) 2330 9304
Web    : http://www.suntan.com.hk    E-mail : info@suntan.com.hk
*********************************************************** 




From rem-conf Tue Sep 12 11:52:13 2000 
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From: Kris Huber <khuber@sorensonlabs.com>
To: Colin Perkins <csp@east.isi.edu>
Cc: "Luthra, Ajay (SD-EX)" <ALuthra@gi.com>, 'Zvi Lifshitz'
	 <ZviL@optibase.com>, "'jan.vandermeer@philips.com'"
	 <jan.vandermeer@philips.com>, rem-conf@es.net, 
	4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to
	 the last call 
Date: Tue, 12 Sep 2000 12:48:46 -0600
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Hello Colin,

I'm a bit confused.  It seems to me that one or more of these assumptions
must be wrong:
- that multicast involves one-way communication from one sender to many
receivers
- 'dynamic negotiation' (of payload type in this case) requires two-way
communication.

Perhaps the precise technical definition of 'negotiate' in this case is what
I lack.  The definition of the word as generally used seems to require
two-way communication (A talks to B, B replies to A, A replies back, etc.,
until they converge to agreement).

Thanks for any clarification,
Kris

-----Original Message-----
From: Colin Perkins [mailto:csp@east.isi.edu]
Sent: Tuesday, September 12, 2000 11:35 AM
To: Luthra, Ajay (SD-EX)
Cc: 'Zvi Lifshitz'; 'jan.vandermeer@philips.com'; rem-conf@es.net;
4on2andIP-sys@advent.ee.columbia.edu
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
to the last call 


--> "Luthra, Ajay (SD-EX)" writes:
>Will it work this way in Multicast environment also?

Yes.

Colin



From rem-conf Tue Sep 12 12:13:40 2000 
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To: Kris Huber <khuber@sorensonlabs.com>
cc: "Luthra, Ajay (SD-EX)" <ALuthra@gi.com>,
        "'Zvi Lifshitz'" <ZviL@optibase.com>,
        "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>,
        rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
In-Reply-To: Your message of "Tue, 12 Sep 2000 12:48:46 MDT."
             <6E031E06378BD311AEF20090273CE1BA7478EA@el-postino.s-vision.com> 
Date: Tue, 12 Sep 2000 15:11:14 -0400
From: Colin Perkins <csp@east.isi.edu>
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--> Kris Huber writes:
>Hello Colin,
>
>I'm a bit confused.  It seems to me that one or more of these assumptions
>must be wrong:
>- that multicast involves one-way communication from one sender to many
>  receivers
>- 'dynamic negotiation' (of payload type in this case) requires two-way
>  communication.
>
>Perhaps the precise technical definition of 'negotiate' in this case is what
>I lack.  The definition of the word as generally used seems to require
>two-way communication (A talks to B, B replies to A, A replies back, etc.,
>until they converge to agreement).
>
>Thanks for any clarification,

In the typical "mbone conference" style of multicast, the sender advertises
that a named payload format is using a particular payload type. There's no
negotiation (and none needed, since payload type numbers are local to a 
session).

Colin



From rem-conf Tue Sep 12 12:36:06 2000 
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Date: Tue, 12 Sep 2000 15:36:18 -0400
From: "Narasimhan, Sam (SD-EX)" <SNarasimhan@gi.com>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to	the
 last call
To: 'Colin Perkins' <csp@east.isi.edu>, Kris Huber <khuber@sorensonlabs.com>
Cc: "Luthra, Ajay (SD-EX)" <ALuthra@GI.COM>,
 'Zvi Lifshitz' <ZviL@optibase.com>,
 "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>, rem-conf@es.net,
 4on2andIP-sys@advent.ee.columbia.edu
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Colin,

 In this mode, if the sender wants to reach both the
'generic format receiver' and 'mpeg4-es-03.txt receiver', 
then he has to use the same payload type. If payload format 
for the generic format differs from mpeg4-es-03.txt draft, 
then the sender has to use 2 different payload types, which 
does not seem to be possible in multicast.

 If mpeg4-es-03.txt included the header with no processing
requirement, then the 'generic format' draft could use the
same payload type as this draft. The generic format would
define the content of adaptation fields that future receivers
that support this will understand. The sender can author
content that will play on both the receivers (slightly
different from each other) using the same payload type.
This is what Jan has been suggesting. This is similar to
many 'scalable' modes where same content can play on
low end and high end players. Low end players play the base
layer and parse over or ignore the enhancement layers while
high end players enhance the decoded presentation with the
enhancement layers. The adaptation field gives the capability
to include such enhancements later without breaking the
base layer (at little extra complexity).

Best Regards
Sam Narasimhan
Motorola 

-----Original Message-----
From: Colin Perkins [mailto:csp@east.isi.edu]
Sent: Tuesday, September 12, 2000 12:11 PM
To: Kris Huber
Cc: Luthra, Ajay (SD-EX); 'Zvi Lifshitz'; 'jan.vandermeer@philips.com';
rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
to the last call


--> Kris Huber writes:
>Hello Colin,
>
>I'm a bit confused.  It seems to me that one or more of these assumptions
>must be wrong:
>- that multicast involves one-way communication from one sender to many
>  receivers
>- 'dynamic negotiation' (of payload type in this case) requires two-way
>  communication.
>
>Perhaps the precise technical definition of 'negotiate' in this case is
what
>I lack.  The definition of the word as generally used seems to require
>two-way communication (A talks to B, B replies to A, A replies back, etc.,
>until they converge to agreement).
>
>Thanks for any clarification,

In the typical "mbone conference" style of multicast, the sender advertises
that a named payload format is using a particular payload type. There's no
negotiation (and none needed, since payload type numbers are local to a 
session).

Colin



From rem-conf Tue Sep 12 12:44:08 2000 
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To: "Narasimhan, Sam (SD-EX)" <SNarasimhan@gi.com>
cc: Kris Huber <khuber@sorensonlabs.com>,
        "Luthra, Ajay (SD-EX)" <ALuthra@gi.com>,
        "'Zvi Lifshitz'" <ZviL@optibase.com>,
        "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>,
        rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
In-Reply-To: Your message of "Tue, 12 Sep 2000 15:36:18 EDT."
             <973597126BDDD11197AA00805FA7EBC9034239A7@ntas0026.gi.com> 
Date: Tue, 12 Sep 2000 15:43:18 -0400
From: Colin Perkins <csp@east.isi.edu>
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Sam,

--> "Narasimhan, Sam (SD-EX)" writes:
> In this mode, if the sender wants to reach both the
>'generic format receiver' and 'mpeg4-es-03.txt receiver', 
>then he has to use the same payload type. If payload format 
>for the generic format differs from mpeg4-es-03.txt draft, 
>then the sender has to use 2 different payload types, which 
>does not seem to be possible in multicast.

Why not? 

> If mpeg4-es-03.txt included the header with no processing
>requirement, then the 'generic format' draft could use the
>same payload type as this draft. The generic format would
>define the content of adaptation fields that future receivers
>that support this will understand. The sender can author
>content that will play on both the receivers (slightly
>different from each other) using the same payload type.
>This is what Jan has been suggesting. This is similar to
>many 'scalable' modes where same content can play on
>low end and high end players. Low end players play the base
>layer and parse over or ignore the enhancement layers while
>high end players enhance the decoded presentation with the
>enhancement layers. The adaptation field gives the capability
>to include such enhancements later without breaking the
>base layer (at little extra complexity).

How are the enhancements signalled? 

Colin



From rem-conf Tue Sep 12 13:33:45 2000 
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Date: Tue, 12 Sep 2000 13:31:12 -0700 (PDT)
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To: rem-conf@es.net
Subject: Sigcomm 2000 - talks available on multimedia jukebox
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		     ACM SIGCOMM 2000 Conference
		    Multicast jukebox announcement
		http://www.acm.org/sigcomm/sigcomm2000

SIGCOMM 2000 is the annual conference of the Special Interest Group on
Data Communication (SIGCOMM), a single-track, highly selective
conference with a technical program of 26 papers, tutorials by noted
instructors on the two days prior, and a work-in-progress poster
session and a social session on outrageous opinions. 

Multicast jukebox on the Internet

There is a multimedia jukebox available at Georgia Institute of
Technology, where Sigcomm 2000 (and Sigcomm 99) talks can be selected
for multicast. Instructions and further information is available at:

      http://imj.gatech.edu/






From rem-conf Tue Sep 12 14:01:04 2000 
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From: David.Leon@nokia.com
To: SNarasimhan@gi.com, ZviL@optibase.com, jan.vandermeer@philips.com
Cc: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to
		the last call
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> 
> Zvi,
> 
>  I am assuming that the payload type from the future draft
> could be set to the same value as this draft, if this
> draft can accommodate the change.
> 
> Regards
> Sam Narasimhan
> Motorola
> 

If guess that what you assume is rather that the current and future drafts
will have the same  NAME and that the future draft will be backward
compatible to the current draft. Is that it?

Regards, David. 



From rem-conf Tue Sep 12 14:34:35 2000 
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Date: Tue, 12 Sep 2000 17:34:18 -0400
From: "Narasimhan, Sam (SD-EX)" <SNarasimhan@gi.com>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to	the
 last call
To: "'David.Leon@nokia.com'" <David.Leon@nokia.com>, SNarasimhan@GI.COM,
 ZviL@optibase.com, jan.vandermeer@philips.com
Cc: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
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David,

 I am assuming that the 2 drafts will use the same
'payload type' and the header fields before the
ES so that the server can send the same RTP packets
which can be assembled by either of the receivers.
I am not suggesting the same NAME for the drafts.

Best Regards
Sam Narasimhan
Motorola

-----Original Message-----
From: David.Leon@nokia.com [mailto:David.Leon@nokia.com]
Sent: Tuesday, September 12, 2000 1:53 PM
To: SNarasimhan@GI.COM; ZviL@optibase.com; jan.vandermeer@philips.com
Cc: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
to the last call



> 
> Zvi,
> 
>  I am assuming that the payload type from the future draft
> could be set to the same value as this draft, if this
> draft can accommodate the change.
> 
> Regards
> Sam Narasimhan
> Motorola
> 

If guess that what you assume is rather that the current and future drafts
will have the same  NAME and that the future draft will be backward
compatible to the current draft. Is that it?

Regards, David. 



From rem-conf Tue Sep 12 14:38:43 2000 
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Date: Tue, 12 Sep 2000 17:38:41 -0400
From: "Narasimhan, Sam (SD-EX)" <SNarasimhan@gi.com>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to	the
 last call
To: 'Colin Perkins' <csp@east.isi.edu>,
 "Narasimhan, Sam (SD-EX)" <SNarasimhan@GI.COM>
Cc: Kris Huber <khuber@sorensonlabs.com>,
 "Luthra, Ajay (SD-EX)" <ALuthra@GI.COM>, 'Zvi Lifshitz' <ZviL@optibase.com>,
 "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>, rem-conf@es.net,
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Colin,

 See my comments.

Best Regards
Sam Narasimhan
Motorola

-----Original Message-----
From: Colin Perkins [mailto:csp@east.isi.edu]
Sent: Tuesday, September 12, 2000 12:43 PM
To: Narasimhan, Sam (SD-EX)
Cc: Kris Huber; Luthra, Ajay (SD-EX); 'Zvi Lifshitz';
'jan.vandermeer@philips.com'; rem-conf@es.net;
4on2andIP-sys@advent.ee.columbia.edu
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
to the last call


Sam,

--> "Narasimhan, Sam (SD-EX)" writes:
> In this mode, if the sender wants to reach both the
>'generic format receiver' and 'mpeg4-es-03.txt receiver', 
>then he has to use the same payload type. If payload format 
>for the generic format differs from mpeg4-es-03.txt draft, 
>then the sender has to use 2 different payload types, which 
>does not seem to be possible in multicast.

Why not? 

Sam - If the payload type is different and the payload format
is different, then the sender has to send 2 different RTP
packets. This can be avoided if the draft included the
header fields.

> If mpeg4-es-03.txt included the header with no processing
>requirement, then the 'generic format' draft could use the
>same payload type as this draft. The generic format would
>define the content of adaptation fields that future receivers
>that support this will understand. The sender can author
>content that will play on both the receivers (slightly
>different from each other) using the same payload type.
>This is what Jan has been suggesting. This is similar to
>many 'scalable' modes where same content can play on
>low end and high end players. Low end players play the base
>layer and parse over or ignore the enhancement layers while
>high end players enhance the decoded presentation with the
>enhancement layers. The adaptation field gives the capability
>to include such enhancements later without breaking the
>base layer (at little extra complexity).

How are the enhancements signalled?

Sam - they do not have to be signaled. The receiver which
understands enhancements will parse the information present
in the stream. The receiver which does not support enhancement
will 'parse over' the enhancement information. 

Colin



From rem-conf Tue Sep 12 14:40:20 2000 
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To: "Narasimhan, Sam (SD-EX)" <SNarasimhan@gi.com>
cc: "'David.Leon@nokia.com'" <David.Leon@nokia.com>, ZviL@optibase.com,
        jan.vandermeer@philips.com, rem-conf@es.net,
        4on2andIP-sys@advent.ee.columbia.edu
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
In-Reply-To: Your message of "Tue, 12 Sep 2000 17:34:18 EDT."
             <973597126BDDD11197AA00805FA7EBC9034239AB@ntas0026.gi.com> 
Date: Tue, 12 Sep 2000 17:39:37 -0400
From: Colin Perkins <csp@east.isi.edu>
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This makes no sense. The payload type is an ephemeral identifier used only
during a particular session. It is bound to a fixed name during session
setup.

Colin



--> "Narasimhan, Sam (SD-EX)" writes:
>David,
>
> I am assuming that the 2 drafts will use the same
>'payload type' and the header fields before the
>ES so that the server can send the same RTP packets
>which can be assembled by either of the receivers.
>I am not suggesting the same NAME for the drafts.
>
>Best Regards
>Sam Narasimhan
>Motorola
>
>-----Original Message-----
>From: David.Leon@nokia.com [mailto:David.Leon@nokia.com]
>Sent: Tuesday, September 12, 2000 1:53 PM
>To: SNarasimhan@GI.COM; ZviL@optibase.com; jan.vandermeer@philips.com
>Cc: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
>Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
>to the last call
>
>
>
>> 
>> Zvi,
>> 
>>  I am assuming that the payload type from the future draft
>> could be set to the same value as this draft, if this
>> draft can accommodate the change.
>> 
>> Regards
>> Sam Narasimhan
>> Motorola
>> 
>
>If guess that what you assume is rather that the current and future drafts
>will have the same  NAME and that the future draft will be backward
>compatible to the current draft. Is that it?
>
>Regards, David. 



From rem-conf Tue Sep 12 14:42:53 2000 
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To: "Narasimhan, Sam (SD-EX)" <SNarasimhan@gi.com>
cc: Kris Huber <khuber@sorensonlabs.com>,
        "Luthra, Ajay (SD-EX)" <ALuthra@gi.com>,
        "'Zvi Lifshitz'" <ZviL@optibase.com>,
        "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>,
        rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
In-Reply-To: Your message of "Tue, 12 Sep 2000 17:38:41 EDT."
             <973597126BDDD11197AA00805FA7EBC9034239AC@ntas0026.gi.com> 
Date: Tue, 12 Sep 2000 17:42:18 -0400
From: Colin Perkins <csp@east.isi.edu>
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--> "Narasimhan, Sam (SD-EX)" writes:
>Colin,
>
> See my comments.
>
>Best Regards
>Sam Narasimhan
>Motorola
>
>-----Original Message-----
>From: Colin Perkins [mailto:csp@east.isi.edu]
>Sent: Tuesday, September 12, 2000 12:43 PM
>To: Narasimhan, Sam (SD-EX)
>Cc: Kris Huber; Luthra, Ajay (SD-EX); 'Zvi Lifshitz';
>'jan.vandermeer@philips.com'; rem-conf@es.net;
>4on2andIP-sys@advent.ee.columbia.edu
>Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
>to the last call
>
>
>Sam,
>
>--> "Narasimhan, Sam (SD-EX)" writes:
>> In this mode, if the sender wants to reach both the
>>'generic format receiver' and 'mpeg4-es-03.txt receiver', 
>>then he has to use the same payload type. If payload format 
>>for the generic format differs from mpeg4-es-03.txt draft, 
>>then the sender has to use 2 different payload types, which 
>>does not seem to be possible in multicast.
>
>Why not? 
>
>Sam - If the payload type is different and the payload format
>is different, then the sender has to send 2 different RTP
>packets. This can be avoided if the draft included the
>header fields.

Since they include different information, and since one is not
understandable by clients of the other, this would seem to be
a good thing!

>> If mpeg4-es-03.txt included the header with no processing
>>requirement, then the 'generic format' draft could use the
>>same payload type as this draft. The generic format would
>>define the content of adaptation fields that future receivers
>>that support this will understand. The sender can author
>>content that will play on both the receivers (slightly
>>different from each other) using the same payload type.
>>This is what Jan has been suggesting. This is similar to
>>many 'scalable' modes where same content can play on
>>low end and high end players. Low end players play the base
>>layer and parse over or ignore the enhancement layers while
>>high end players enhance the decoded presentation with the
>>enhancement layers. The adaptation field gives the capability
>>to include such enhancements later without breaking the
>>base layer (at little extra complexity).
>
>How are the enhancements signalled?
>
>Sam - they do not have to be signaled. The receiver which
>understands enhancements will parse the information present
>in the stream. The receiver which does not support enhancement
>will 'parse over' the enhancement information. 

Of course they have to be signalled - else how do receivers know 
what sort of enhancement is present?

Colin



From rem-conf Tue Sep 12 14:48:49 2000 
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Date: Tue, 12 Sep 2000 17:48:52 -0400
From: "Luthra, Ajay (SD-EX)" <ALuthra@gi.com>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to	the
 last call
To: 'Zvi Lifshitz' <ZviL@optibase.com>, jan.vandermeer@philips.com
Cc: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
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Yes, if WG-11 does not think Kikuchi-san's proposal meets the needs (I am
not saying whether it does or does not - we are currently debating it here).
As I understood, conclusion in Beijing was to start WG-11's proposals on how
to carry different MPEG-4 streams (ES, SL etc) on RTP. If that aligns with
Kikuchi-san's proposal then fine otherwise have "official" proposals from
WG-11 according to what its experts collectively think are the best ways to
carry MPEG-4 streams. As I mentioned before, alignment with Kikuchi-san's
proposal was desired but not required. 

Ajay

-----Original Message-----
From: Zvi Lifshitz [mailto:ZviL@optibase.com]
Sent: Tuesday, September 12, 2000 10:03 AM
To: jan.vandermeer@philips.com
Cc: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
to the last call


Mistake is when you can do something and you do it wrong. Here we could not
do anything, except asking to reject the Kikuci-san proposal and start
another one. Is that what you wanted?

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Tuesday, September 12, 2000 5:39 PM
To:	ZviL@optibase.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to the last call

 
Which is not a mistake ????

Regards,

Jan

-----Original Message-----
From:	ZviL@optibase.com [mailto:ZviL@optibase.com]
Sent:	dinsdag, 12 september, 2000 17:33
To:	Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to
the last call


The scenario you gave is not possible, for the reasons I previously
explained.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Tuesday, September 12, 2000 5:15 PM
To:	ZviL@optibase.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to the last call


Zvi,

I don't think there is any confusion. If no such bytes are allowed, the
scenario I gave will not be possible. Which in my view would be a mistake.
>From MPEG point of view a serious mistake.

To allow for extensibility in the "generic format" will solve problems, but
not this one

Regards,

Jan



From rem-conf Tue Sep 12 15:31:29 2000 
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> 
> 
> David,
> 
>  I am assuming that the 2 drafts will use the same
> 'payload type' and the header fields before the
> ES so that the server can send the same RTP packets
> which can be assembled by either of the receivers.
> I am not suggesting the same NAME for the drafts.

As it has been said, the payload type is not static. The name is fixed but
if you assume that the two formats are using different names, how could a
receiver which has implemented the current payload format even know the name
of a future (and thus non assigned) payload format? 

David.


 



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From: "Lawrence A. Rowe" <Rowe@bmrc.berkeley.edu>
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Hi -

I'll be giving an all day tutorial on Distance/Asynchronous Learning as
part of the ACM Multimedia 2000 conference in LA on tuesday October
31st.  Please pass on the enclosed announcement and details to anyone
who you think might be interested in attending either the tutorial or
the conference.  More details on both is available at
	http://www.acm.org/sigs/sigmm/MM2000/
You can attend just the tutorial if you are not interested in the
general conference.

If you're interested in establishing a distance/asynchronous learning
program or you want to learn about current technology for webcasting
including both room setup and operations, this tutorial might prove
interesting.

If you have any questions, please contact me.
	Larry
-- 
Professor Lawrence A. Rowe          Internet:  Rowe@BMRC.Berkeley.EDU
Computer Science Division - EECS       Phone: 510-642-5117
University of California, Berkeley       Fax: 510-642-5615
Berkeley, CA 94720-1776            URL: http://bmrc.berkeley.edu/~larry

----------------------------------------

                Distance/Asynchronous Learning Tutorial
                 Pedagogy, Technology, and Operations

                          Lawrence A. Rowe
                 Berkeley Multiedia Research Center
                 University of California, Berkeley 
                  http://bmrc.berkeley.edu/~larry

Distance and asynchronous learning are important topics in both industry
and academia.  Many educators and technologists developing distance
learning programs are unfamiliar with previous research and experimental
results in pedagogy, technology, and operations related to distributed
collaboration and distance learning.  Distance learning began over a
hundred years ago with correspondence courses. The introduction of
television and video conferencing technology created new opportunities
for delivering education.  More recently, the development of desktop
distributed collaboration tools (e.g., CuSeeMe, the H.32x standards,
Webx, the Mbone tools, and NetMeeting) and webcasting technology (e.g.,
Quicktime Streaming, Real Networks G2, and Microsoft Windows Media) that
operate over the Internet provides more alternatives for educational
delivery. Many universities and companies are using these technologies
to deliver remote instruction.  Many questions remain about educational
effectiveness and costs.

This tutorial will present a broad introduction and survey of many
topics in distance learning ranging from pedagogy to technology
including both studio classroom design and operational issues.  The
first topic covered will be past results from deploying distance and
asynchronous learning technologies.  Learn why "sage on the stage" is a
discredited approach from a pedagogical perspective and why "tutored
video tape" has proven successful in many experiments. In addition,
learn about successful synchronous distance learning strategies that
work with either television or video conferencing technologies.

The second topic covered will be room design and cost.  The basic
equipment needed to produce video conferences and webcasts will be
described and solutions to common problems will be discussed. The
development of a well-designed studio classroom with cameras,
microphones, multimedia presentation equipment (e.g., projectors,
document cameras, VCR's, etc.) is difficult and can be very costly. 
Different classroom designs will be described and compared.

The third topic covered will be the basic technology for distance
learning. Different technologies will be described and compared
including support for audio/video interaction, whiteboard and multimedia
presentations, and other collaboration tools (e.g., chat rooms, media
spaces, etc.). Specific examples covered will include H.32x standards,
the Internet Mbone tools, and streaming media technologies. Tools for
producing live collaborations and for producing rich multimedia material
for on-demand replay will be discussed including emerging multimedia
standards (e.g., SMIL) and lecture 
browser technologies.

Finally, operational issues of producing distance and asynchronous
learning classes will be discussed including program guides, equipment
and labor costs, and operational procedures.

The tutorial will focus on teaching you what you need to know to develop
a distance and asynchronous learning program and what the major research
issues that remain to be solved.

Date/Location/Registration:

The tutorial will be all day tuesday October 31, 2000 at the Marina
Beach Marriott in Marina Del Rey in Los Angeles.  The tutorial is being
held in conjunction with the annual ACM Multimedia Conference.  Cost for
the tutorial is $500 ($400 for ACM SIGMM/SIGCOMM/SIGGRAPH members).  You
can attend the tutorial without attending the conference.  For more
information, including registration see:
             http://www.acm.org/sigs/sigmm/MM2000/	


Background/Prerequisite:  

Interest in developing a distance or asynchronous learning programming
or establishing a research group to explore problems related to 
distributed collaboration. 

Instructor Biography:

Professor Rowe is a Professor of Electrical Engineering and Computer
Science at the University of California at Berkeley.  He received a BA
in Mathematics and a PhD in Information and Computer Science from the
University of California at Irvine in 1970 and 1976, respectively.  He
is also the founding director of the Berkeley Multimedia Research Center
which is an interdisciplinary research group working on applications of
multimedia technology to business, education, research, and society. 

Professor Rowe's current research interests are multimedia applications
and databases, video conferencing, hypermedia courseware, and video
compression. He heads the research group that produces the regularly
scheduled Berkeley Multimedia, Interfaces, and Graphics Seminar
http://www.bmrc.berkeley.edu/mig) webcast world-wide on the Internet. 
Presently, this technology is being used to offer more live and
on-demand courses through the Berkeley Internet Broadcasting System
(BIBS) - http://www.bmrc.berkeley.edu/bibs. This semester the BIBS
system is being used to webcast over 10 classes which produces over 35
hours a week of new video material.

Professor Rowe is an ACM Fellow. He has published over ninety papers on
multimedia systems and applications, programming systems, and database
systems. He is currently Chair of the ACM Special Interest Group on
Multimedia. And, he is a member of the editorial board of the ACM
Multimedia Systems Journal. He has co-authored papers that have won best
paper awards at OOPSLA '91, ACM SIGMOD '96, and ACM Multimedia '98. And,
he has organized and chaired several conferences and served on numerous
program committees.



From rem-conf Tue Sep 12 16:58:55 2000 
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Date: Tue, 12 Sep 2000 19:58:47 -0400
From: "Luthra, Ajay (SD-EX)" <ALuthra@gi.com>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to	the
 last call
To: 'Colin Perkins' <csp@east.isi.edu>
Cc: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
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Collin,
   But you said in your previous email, " ... The payload type will have to
be negotiated out of band anyway .. ". And in another email you said, "Yes"
to the question - does one need to work this way in multicast environment
also. And now you are saying it does not need to be negotiated. My
understanding of this subject is not yet excellent. So, I do not understand
these three emails from you which seem to contradict each other. 

Does the payload type need to be negotiated or not?

With regards,
Ajay

-----Original Message-----
From: Colin Perkins [mailto:csp@east.isi.edu]
Sent: Tuesday, September 12, 2000 12:11 PM
To: Kris Huber
Cc: Luthra, Ajay (SD-EX); 'Zvi Lifshitz'; 'jan.vandermeer@philips.com';
rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
to the last call


--> Kris Huber writes:
>Hello Colin,
>
>I'm a bit confused.  It seems to me that one or more of these assumptions
>must be wrong:
>- that multicast involves one-way communication from one sender to many
>  receivers
>- 'dynamic negotiation' (of payload type in this case) requires two-way
>  communication.
>
>Perhaps the precise technical definition of 'negotiate' in this case is
what
>I lack.  The definition of the word as generally used seems to require
>two-way communication (A talks to B, B replies to A, A replies back, etc.,
>until they converge to agreement).
>
>Thanks for any clarification,

In the typical "mbone conference" style of multicast, the sender advertises
that a named payload format is using a particular payload type. There's no
negotiation (and none needed, since payload type numbers are local to a 
session).

Colin



From rem-conf Tue Sep 12 20:30:51 2000 
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Colin, all,

I think it is important to keep in mind that often MPEG uses in-stream
signaling. For example, after the length byte, the adaptation field may=

start with an identifier that specifies the sort of enhancement. Based =
on
this approach, the following would be needed :

1) Define an MPEG-4 ES payload format that allows for an adaptation fie=
ld
without defining the meaning of the adaptation field.

2) Define the meaning of the adaptation field in the "generic MPEG-4
specification". For carriage of MPEG-4 ES and MPEG-4 SL streams the "ge=
neric
specification" defines the use of exactly_the_same payload type as defi=
ned
in 1). The only difference is the defined use of the adaptation field.
Whether or not an SL header is carried is signaled in the adaptation fi=
eld.

3) Next to the payload types for MPEG-4 ES and SL streams, the "generic=

specification" may also define payloads for other streams, such as Flex=
Mux
streams. For these, other payload types will be needed.

Now the initial services start using the MPEG-4 ES payload format; thou=
gh
the adaptation field is not used, this field is allowed to occur. Futur=
e
services may use exactly the same payload type, but with the adaptation=

field as defined by the "generic specification".

It may also be important to understand that the presence of the adaptat=
ion
field is intended to be transparent from RTP point of view and for the
session in general. The adaptation field serves MPEG-4 system purposes =
which
are fully orthogonal to RTP and the streaming session.

Kind regards,

Jan

-----Original Message-----
From:	csp@east.isi.edu [mailto:csp@east.isi.edu]
Sent:	dinsdag, 12 september, 2000 23:45
To:	SNarasimhan@gi.com
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net; Jan vanderMe=
er;
ZviL@optibase.com; ALuthra@gi.com; khuber@sorensonlabs.com
Subject:	Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response=
 to
the last call


--> "Narasimhan, Sam (SD-EX)" writes:
>Colin,
>
> See my comments.
>
>Best Regards
>Sam Narasimhan
>Motorola
>
>-----Original Message-----
>From: Colin Perkins [mailto:csp@east.isi.edu]
>Sent: Tuesday, September 12, 2000 12:43 PM
>To: Narasimhan, Sam (SD-EX)
>Cc: Kris Huber; Luthra, Ajay (SD-EX); 'Zvi Lifshitz';
>'jan.vandermeer@philips.com'; rem-conf@es.net;
>4on2andIP-sys@advent.ee.columbia.edu
>Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in respons=
e
>to the last call
>
>
>Sam,
>
>--> "Narasimhan, Sam (SD-EX)" writes:
>> In this mode, if the sender wants to reach both the
>>'generic format receiver' and 'mpeg4-es-03.txt receiver',
>>then he has to use the same payload type. If payload format
>>for the generic format differs from mpeg4-es-03.txt draft,
>>then the sender has to use 2 different payload types, which
>>does not seem to be possible in multicast.
>
>Why not?
>
>Sam - If the payload type is different and the payload format
>is different, then the sender has to send 2 different RTP
>packets. This can be avoided if the draft included the
>header fields.

Since they include different information, and since one is not
understandable by clients of the other, this would seem to be
a good thing!

>> If mpeg4-es-03.txt included the header with no processing
>>requirement, then the 'generic format' draft could use the
>>same payload type as this draft. The generic format would
>>define the content of adaptation fields that future receivers
>>that support this will understand. The sender can author
>>content that will play on both the receivers (slightly
>>different from each other) using the same payload type.
>>This is what Jan has been suggesting. This is similar to
>>many 'scalable' modes where same content can play on
>>low end and high end players. Low end players play the base
>>layer and parse over or ignore the enhancement layers while
>>high end players enhance the decoded presentation with the
>>enhancement layers. The adaptation field gives the capability
>>to include such enhancements later without breaking the
>>base layer (at little extra complexity).
>
>How are the enhancements signalled?
>
>Sam - they do not have to be signaled. The receiver which
>understands enhancements will parse the information present
>in the stream. The receiver which does not support enhancement
>will 'parse over' the enhancement information.

Of course they have to be signalled - else how do receivers know
what sort of enhancement is present?

Colin

=



From rem-conf Tue Sep 12 21:13:36 2000 
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From: Stephan Wenger <stewe@cs.tu-berlin.de>
Subject: In band signalling of PT
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Folks,

the recent discussion about the MPEG-4 ES payload brought up an
issue that, IMHO, may be a shortcoming of current RTP and could
easily be fixed.  This shortcoming is RTP's need for the out-of-band
announcement/negotiation of parameters, including the PT.

I wonder whether it may make sense to signal such information in-band
as well, e.g. through sender reports or through special media packets.
That would allow receivers to "tune" to RTP broadcasts with only
IP/Port information (which could, for example, be owned by a TV station).

Questions:

a) Did I miss something in the current spec?  Is this already specified?
b) Does such a scenario make sense?  Specifically, how do people think
   to include session announcements in the media packetstream (or in
   RTCP)?  I recall that some time ago there was some discussion
   about architectural cleanness (it does / doesn't make sense to include
   session info in the media stream).  What is the status of that?
c) Would such an enhancement have any impact on the current MPEG
   discussion? (IMHO not, but I may have missed something)

Stephan




From rem-conf Tue Sep 12 23:54:46 2000 
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To: "'Luthra, Ajay (SD-EX)'" <ALuthra@gi.com>,
        "jan.vandermeer@philips.com"
	 <jan.vandermeer@philips.com>
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Dear Ajay,

In Paris we came the conclusion that MPEG can support both the ES format 
proposed by Kikuci-san, and the Generic format (Civanlar et al), since with 
a proper use of SLConfig the first becomes a subset of the second (at least 
for video). Therefore we didn't find a reason to delay the process of the 
ES draft. All seemed pretty nice, until this issue of the extra byte came 
up and created an awful mess. We can still go forward, if everybody 
understands that the extension mechanism is dealt by the Generic format, 
and is not necessary and can serve nothing in the ES draft.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm 
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October 
10th-12th, 2000


-----Original Message-----
From:	Luthra, Ajay (SD-EX) [SMTP:ALuthra@gi.com]
Sent:	Tuesday, September 12, 2000 6:18 PM
To:	'Zvi Lifshitz'; jan.vandermeer@philips.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response 
to	the last call


Dear Zvi,
  Now, I am confused by your statement.  As we discussed in Beijing, the ES
format draft being considered by IETF is proposed by individuals and we can
not and should not control what they input to IETF. That is IETF's 
business.
What we had agreed to do in Beijing was to come up with WG-11's
recommendation and contribution on how to carry ES (as well as SL and other
streams types) in RTP and to let IETF know that that contribution is coming
(by Oct 2000) and make them aware that WG-11 contribution may or may not
align with what is being considered by IETF.  As I understand, WG-11 has
done that.

  So, as agreed in Beijing, compatibility with what individual 
contributions
IETF is considering is not a requirement (desired but not required).
Therefore, if we think we need additional byte then let us put it in 
WG-11's
contribution to IETF on how to carry ES. The discussion should be on 
whether
we need additional bytes or not, and not whether it is compatible with what
was proposed to IETF by some other authors. I know it is a mess and is
partly (major part) due to our fault in WG-11 and partly (minor part) due 
to
IETF process. In Beijing, WG-11 had asked people to not to submit 
individual
contributions and/or withdraw their inputs until this matter is fully
discussed in WG-11. The decision to freeze or withdraw is not WG-11's. 
There
is nothing we can do about it at this stage. It is water under the bridge.

  So, let us just discuss whether the additional bytes, as proposed by Jan,
make sense or not. In that regard, I think it makes good sense. It gives 
you
lot of flexibility that will be needed in future. We are still reaping the
benefits of the flexibility of MPEG-2 systems. I really do not understand
the reason for not accepting it.

With regards,
Ajay




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Sam,

It seems you want the current draft to be modified in the future. Then:

1. I don't believe this is possible.
2. I don't think this is necessary.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
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-----Original Message-----
From:	Narasimhan, Sam (SD-EX) [SMTP:SNarasimhan@gi.com]
Sent:	Tuesday, September 12, 2000 6:21 PM
To:	'Zvi Lifshitz'; Narasimhan, Sam (SD-EX); 'jan.vandermeer@philips.com'
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to	the last call

 
Zvi,

 I am assuming that the payload type from the future draft
could be set to the same value as this draft, if this
draft can accommodate the change.

Regards
Sam Narasimhan
Motorola

-----Original Message-----
From: Zvi Lifshitz [mailto:ZviL@optibase.com]
Sent: Tuesday, September 12, 2000 10:18 AM
To: Narasimhan, Sam (SD-EX); 'jan.vandermeer@philips.com'
Cc: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
to the last call


[Narasimhan, Sam (SD-EX):]

If this draft were to say that receivers should parse over the
adaptation bytes and not process them (as this draft does not
define these), then such receivers will be able to process payload
>from future draft (from MPEG)

[Reply:]

They will not, because they don't know the payload type of the future draft
therefore they will not know they can process payloads from the future draft
even though they can.

If they know, they also know that the future draft has extra header info. So
work on the future draft, not on the current draft.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000




From rem-conf Tue Sep 12 23:57:19 2000 
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From: Zvi Lifshitz <zvil@csi.com>
To: "'Colin Perkins'" <csp@east.isi.edu>,
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	 <ALuthra@gi.com>
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Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
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I don't think Colin needs my support, but I will nevertheless give it out of frustration. This is a simple truth.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
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> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October 10th-12th, 2000


-----Original Message-----
From:	Colin Perkins [SMTP:csp@east.isi.edu]
Sent:	Tuesday, September 12, 2000 6:24 PM
To:	Luthra, Ajay (SD-EX)
Cc:	'Zvi Lifshitz'; jan.vandermeer@philips.com; rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 

--> "Luthra, Ajay (SD-EX)" writes:
>Dear Zvi,
>  Now, I am confused by your statement.  As we discussed in Beijing, the ES
>format draft being considered by IETF is proposed by individuals and we can
>not and should not control what they input to IETF. That is IETF's business.
>What we had agreed to do in Beijing was to come up with WG-11's
>recommendation and contribution on how to carry ES (as well as SL and other
>streams types) in RTP and to let IETF know that that contribution is coming
>(by Oct 2000) and make them aware that WG-11 contribution may or may not
>align with what is being considered by IETF.  As I understand, WG-11 has
>done that.
>
>  So, as agreed in Beijing, compatibility with what individual contributions
>IETF is considering is not a requirement (desired but not required).
>Therefore, if we think we need additional byte then let us put it in WG-11's
>contribution to IETF on how to carry ES. The discussion should be on whether
>we need additional bytes or not, and not whether it is compatible with what
>was proposed to IETF by some other authors. I know it is a mess and is
>partly (major part) due to our fault in WG-11 and partly (minor part) due to
>IETF process. In Beijing, WG-11 had asked people to not to submit individual
>contributions and/or withdraw their inputs until this matter is fully
>discussed in WG-11. The decision to freeze or withdraw is not WG-11's. There
>is nothing we can do about it at this stage. It is water under the bridge. 
>
>  So, let us just discuss whether the additional bytes, as proposed by Jan,
>make sense or not. In that regard, I think it makes good sense. It gives you
>lot of flexibility that will be needed in future. We are still reaping the
>benefits of the flexibility of MPEG-2 systems. I really do not understand
>the reason for not accepting it.

Because we have a mechanism for extensibility in RTP via the payload type.
This extra byte provides no advantage - all the examples which have been 
proposed can be done without using this field - and is incompatible with 
existing practise.

Colin




From rem-conf Tue Sep 12 23:59:14 2000 
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To: "'Colin Perkins'" <csp@east.isi.edu>,
        "Narasimhan, Sam (SD-EX)"
	 <SNarasimhan@gi.com>
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Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
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[Colin Perkins:]

How are the enhancements signalled? 

[Reply:]

How indeed?

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
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> Come see us at:
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From rem-conf Wed Sep 13 00:01:33 2000 
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To: "'David.Leon@nokia.com'" <David.Leon@nokia.com>,
        "SNarasimhan@gi.com"
	 <SNarasimhan@gi.com>,
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 =?us-ascii?Q?mpeg4=2Des=2D03_in_response_to=09the_last_call?=
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If it has the same name then it is the same format. See my previous comment.

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
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-----Original Message-----
From:	David.Leon@nokia.com [SMTP:David.Leon@nokia.com]
Sent:	Tuesday, September 12, 2000 10:53 PM
To:	SNarasimhan@gi.com; ZviL@optibase.com; jan.vandermeer@philips.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to	the last call

 

> 
> Zvi,
> 
>  I am assuming that the payload type from the future draft
> could be set to the same value as this draft, if this
> draft can accommodate the change.
> 
> Regards
> Sam Narasimhan
> Motorola
> 

If guess that what you assume is rather that the current and future drafts
will have the same  NAME and that the future draft will be backward
compatible to the current draft. Is that it?

Regards, David. 




From rem-conf Wed Sep 13 00:24:34 2000 
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Date: Wed, 13 Sep 2000 09:22:17 +0200
From: Franceschini Guido <Guido.Franceschini@CSELT.IT>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to	the
 last call
To: "'Narasimhan, Sam (SD-EX)'" <SNarasimhan@gi.com>,
 "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>,
 'zvi' <zvil@csi.com>
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Hi all,
my own considerations on this thread

In RTP and IETF procedure terms, I understand this thread on the extra byte
as:
1) modify the current ES draft to accomodate a payload header starting with
a byte indicating its length. No specification of what the payload header
carries
2) allow the modified ES draft to enter the RFC standard track and be widely
implemented
3) propose a new draft for 'generic' ES or SL mapping, which is identical to
the modified ES draft but also defines what the payload header carries
4) the 'generic' draft becomes standard and _obsoletes_ the ES draft.
Devices implementing the ES draft will skip the payload header information
but still understand the media content.

Well, I think we have a technical problem here, which has nothing to do with
usual practices / procedures / timelines.
The problem that I see is that:
- the ES draft defines 2 Payload Formats: 1 for Audio, 1 for Video
- the 'generic' draft will define 1 Payload Format only, and the type of
media (Audio, Video, BIFS ...) being carried will be specified using MPEG-4
Object Descriptors.

What we achieved at the Paris meeting (before discussing this extra byte
feature) has been to allow MPEG-4 Systems terminals implementing the
'generic' draft to use the ES draft as well withouth any need to duplicate
implementations (i.e.: the ES draft is a subset of the generic draft in the
Data Plane).
Now, the 'extra byte' proposal would allow devices implementing the ES draft
to be compatible in the Data Plane with the 'generic' format, too.

Compatibility in the Control Plane (i.e. Payload Type
negotiation/announcement) was not considered in Paris.
I believe that devices implementing the 'generic' draft would have no
difficulties in understanding 1 more SDP keyword or MIME type as introduced
by the ES draft.
But the reverse is more difficult and tricky. As I outlined above, the
'generic' draft would not specify by itself the type of media content it is
carrying, and would rely on the Object Descriptor to do that. If the goal is
to allow devices implementing the ES draft to be compatible with future
content delivered by means of the 'generic' draft, than we need to specify
NOW how to duplicate the information in the OD so that the media type is
made explicit for those devices. And then we will have to accept that
devices implementing MPEG-4 Systems get the indication of the media type
twice :-(

Did I make the point clear ?

Guido

> ----------
> From: 	Zvi Lifshitz[SMTP:zvil@csi.com]
> Sent: 	Wednesday, September 13, 2000 9:51 AM
> To: 	'Narasimhan, Sam (SD-EX)'; 'jan.vandermeer@philips.com'
> Cc: 	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
> Subject: 	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
> response to	the last call
> 
> Sam,
> 
> It seems you want the current draft to be modified in the future. Then:
> 
> 1. I don't believe this is possible.
> 2. I don't think this is necessary.
> 
> Regards,
> 
> z
> soon the whole world will STREAM
> ==================================================================
> Zvi Lifshitz				Phone +972(2)679-4788
> zvil@optibase.com			Fax +972(2)679-4789
> Optibase Ltd.				Mobile: +972(53)461-787
> http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
> Send SMS from:
> http://isend.cellcom.co.il/English/Login.htm
> ==================================================================
> > Come see us at:
> Streaming Media Europe Earls Court Centre, Booth # 628, London, October
> 10th-12th, 2000
> 
> 
> -----Original Message-----
> From:	Narasimhan, Sam (SD-EX) [SMTP:SNarasimhan@gi.com]
> Sent:	Tuesday, September 12, 2000 6:21 PM
> To:	'Zvi Lifshitz'; Narasimhan, Sam (SD-EX);
> 'jan.vandermeer@philips.com'
> Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
> Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
> response to	the last call
> 
>  
> Zvi,
> 
>  I am assuming that the payload type from the future draft
> could be set to the same value as this draft, if this
> draft can accommodate the change.
> 
> Regards
> Sam Narasimhan
> Motorola
> 
> -----Original Message-----
> From: Zvi Lifshitz [mailto:ZviL@optibase.com]
> Sent: Tuesday, September 12, 2000 10:18 AM
> To: Narasimhan, Sam (SD-EX); 'jan.vandermeer@philips.com'
> Cc: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
> Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
> to the last call
> 
> 
> [Narasimhan, Sam (SD-EX):]
> 
> If this draft were to say that receivers should parse over the
> adaptation bytes and not process them (as this draft does not
> define these), then such receivers will be able to process payload
> from future draft (from MPEG)
> 
> [Reply:]
> 
> They will not, because they don't know the payload type of the future
> draft
> therefore they will not know they can process payloads from the future
> draft
> even though they can.
> 
> If they know, they also know that the future draft has extra header info.
> So
> work on the future draft, not on the current draft.
> 
> Regards,
> 
> z
> soon the whole world will STREAM
> ==================================================================
> Zvi Lifshitz				Phone +972(2)679-4788
> zvil@optibase.com			Fax +972(2)679-4789
> Optibase Ltd.				Mobile: +972(53)461-787
> http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
> Send SMS from:
> http://isend.cellcom.co.il/English/Login.htm
> ==================================================================
> > Come see us at:
> Streaming Media Europe Earls Court Centre, Booth # 628, London, October
> 10th-12th, 2000
> 



From rem-conf Wed Sep 13 00:30:23 2000 
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        "'David.Leon@nokia.com'" <David.Leon@nokia.com>,
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So how would old devices recognize the new name?

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
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> Come see us at:
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-----Original Message-----
From:	Narasimhan, Sam (SD-EX) [SMTP:SNarasimhan@gi.com]
Sent:	Tuesday, September 12, 2000 11:34 PM
To:	'David.Leon@nokia.com'; SNarasimhan@gi.com; ZviL@optibase.com; jan.vandermeer@philips.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to	the last call

 
David,

 I am assuming that the 2 drafts will use the same
'payload type' and the header fields before the
ES so that the server can send the same RTP packets
which can be assembled by either of the receivers.
I am not suggesting the same NAME for the drafts.

Best Regards
Sam Narasimhan
Motorola




From rem-conf Wed Sep 13 00:32:24 2000 
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From: Zvi Lifshitz <zvil@csi.com>
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        "'Colin Perkins'"
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Subject: =?us-ascii?Q?RE=3A_AHG_comments_on_draft=2Dietf=2Davt=2Drtp=2D?=
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[Narasimhan, Sam (SD-EX):]

The receiver which
understands enhancements will parse the information present
in the stream.

[Reply:]

How will it know that there are enhancements?

z




From rem-conf Wed Sep 13 00:32:37 2000 
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See my previous reply to you.

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
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> Come see us at:
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-----Original Message-----
From:	Luthra, Ajay (SD-EX) [SMTP:ALuthra@gi.com]
Sent:	Tuesday, September 12, 2000 11:49 PM
To:	'Zvi Lifshitz'; jan.vandermeer@philips.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to	the last call

 
Yes, if WG-11 does not think Kikuchi-san's proposal meets the needs (I am
not saying whether it does or does not - we are currently debating it here).
As I understood, conclusion in Beijing was to start WG-11's proposals on how
to carry different MPEG-4 streams (ES, SL etc) on RTP. If that aligns with
Kikuchi-san's proposal then fine otherwise have "official" proposals from
WG-11 according to what its experts collectively think are the best ways to
carry MPEG-4 streams. As I mentioned before, alignment with Kikuchi-san's
proposal was desired but not required. 

Ajay




From rem-conf Wed Sep 13 00:33:36 2000 
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;-)

z


-----Original Message-----
From:	David.Leon@nokia.com [SMTP:David.Leon@nokia.com]
Sent:	Wednesday, September 13, 2000 12:26 AM
To:	SNarasimhan@gi.com; David.Leon@nokia.com; ZviL@optibase.com; jan.vandermeer@philips.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to	the last call

 


> 
> 
> David,
> 
>  I am assuming that the 2 drafts will use the same
> 'payload type' and the header fields before the
> ES so that the server can send the same RTP packets
> which can be assembled by either of the receivers.
> I am not suggesting the same NAME for the drafts.

As it has been said, the payload type is not static. The name is fixed but
if you assume that the two formats are using different names, how could a
receiver which has implemented the current payload format even know the name
of a future (and thus non assigned) payload format? 

David.


 




From rem-conf Wed Sep 13 00:37:16 2000 
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Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call
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Jan,

Your nice proposal can be incorporated in the Generic format. I hope we managed to explain why it is useless with the ES format.

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
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> Come see us at:
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-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Wednesday, September 13, 2000 5:28 AM
To:	csp@east.isi.edu
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call

 
Colin, all,

I think it is important to keep in mind that often MPEG uses in-stream
signaling. For example, after the length byte, the adaptation field may
start with an identifier that specifies the sort of enhancement. Based on
this approach, the following would be needed :

1) Define an MPEG-4 ES payload format that allows for an adaptation field
without defining the meaning of the adaptation field.

2) Define the meaning of the adaptation field in the "generic MPEG-4
specification". For carriage of MPEG-4 ES and MPEG-4 SL streams the "generic
specification" defines the use of exactly_the_same payload type as defined
in 1). The only difference is the defined use of the adaptation field.
Whether or not an SL header is carried is signaled in the adaptation field.

3) Next to the payload types for MPEG-4 ES and SL streams, the "generic
specification" may also define payloads for other streams, such as FlexMux
streams. For these, other payload types will be needed.

Now the initial services start using the MPEG-4 ES payload format; though
the adaptation field is not used, this field is allowed to occur. Future
services may use exactly the same payload type, but with the adaptation
field as defined by the "generic specification".

It may also be important to understand that the presence of the adaptation
field is intended to be transparent from RTP point of view and for the
session in general. The adaptation field serves MPEG-4 system purposes which
are fully orthogonal to RTP and the streaming session.

Kind regards,

Jan




From rem-conf Wed Sep 13 00:43:38 2000 
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Hi Guido,

You didn't make the point clearer, just showed how messy we got with this 
idea of extending the ES format.

Therefore I propose to kill this discussion about modifications to the ES 
format, and concentrating how to improve the Generic format, if necessary.

Ciao,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
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==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October 
10th-12th, 2000


-----Original Message-----
From:	Franceschini Guido [SMTP:Guido.Franceschini@CSELT.IT]
Sent:	Wednesday, September 13, 2000 9:22 AM
To:	'Narasimhan, Sam (SD-EX)'; 'jan.vandermeer@philips.com'; 'zvi'
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response 
to	the last call


Hi all,
my own considerations on this thread

In RTP and IETF procedure terms, I understand this thread on the extra byte
as:
1) modify the current ES draft to accomodate a payload header starting with
a byte indicating its length. No specification of what the payload header
carries
2) allow the modified ES draft to enter the RFC standard track and be 
widely
implemented
3) propose a new draft for 'generic' ES or SL mapping, which is identical 
to
the modified ES draft but also defines what the payload header carries
4) the 'generic' draft becomes standard and _obsoletes_ the ES draft.
Devices implementing the ES draft will skip the payload header information
but still understand the media content.

Well, I think we have a technical problem here, which has nothing to do 
with
usual practices / procedures / timelines.
The problem that I see is that:
- the ES draft defines 2 Payload Formats: 1 for Audio, 1 for Video
- the 'generic' draft will define 1 Payload Format only, and the type of
media (Audio, Video, BIFS ...) being carried will be specified using MPEG-4
Object Descriptors.

What we achieved at the Paris meeting (before discussing this extra byte
feature) has been to allow MPEG-4 Systems terminals implementing the
'generic' draft to use the ES draft as well withouth any need to duplicate
implementations (i.e.: the ES draft is a subset of the generic draft in the
Data Plane).
Now, the 'extra byte' proposal would allow devices implementing the ES 
draft
to be compatible in the Data Plane with the 'generic' format, too.

Compatibility in the Control Plane (i.e. Payload Type
negotiation/announcement) was not considered in Paris.
I believe that devices implementing the 'generic' draft would have no
difficulties in understanding 1 more SDP keyword or MIME type as introduced
by the ES draft.
But the reverse is more difficult and tricky. As I outlined above, the
'generic' draft would not specify by itself the type of media content it is
carrying, and would rely on the Object Descriptor to do that. If the goal 
is
to allow devices implementing the ES draft to be compatible with future
content delivered by means of the 'generic' draft, than we need to specify
NOW how to duplicate the information in the OD so that the media type is
made explicit for those devices. And then we will have to accept that
devices implementing MPEG-4 Systems get the indication of the media type
twice :-(

Did I make the point clear ?

Guido

> ----------
> From: 	Zvi Lifshitz[SMTP:zvil@csi.com]
> Sent: 	Wednesday, September 13, 2000 9:51 AM
> To: 	'Narasimhan, Sam (SD-EX)'; 'jan.vandermeer@philips.com'
> Cc: 	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
> Subject: 	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
> response to	the last call
>
> Sam,
>
> It seems you want the current draft to be modified in the future. Then:
>
> 1. I don't believe this is possible.
> 2. I don't think this is necessary.
>
> Regards,
>
> z
> soon the whole world will STREAM
> ==================================================================
> Zvi Lifshitz				Phone +972(2)679-4788
> zvil@optibase.com			Fax +972(2)679-4789
> Optibase Ltd.				Mobile: +972(53)461-787
> http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
> Send SMS from:
> http://isend.cellcom.co.il/English/Login.htm
> ==================================================================
> > Come see us at:
> Streaming Media Europe Earls Court Centre, Booth # 628, London, October
> 10th-12th, 2000
>
>
> -----Original Message-----
> From:	Narasimhan, Sam (SD-EX) [SMTP:SNarasimhan@gi.com]
> Sent:	Tuesday, September 12, 2000 6:21 PM
> To:	'Zvi Lifshitz'; Narasimhan, Sam (SD-EX);
> 'jan.vandermeer@philips.com'
> Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
> Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
> response to	the last call
>
>
> Zvi,
>
>  I am assuming that the payload type from the future draft
> could be set to the same value as this draft, if this
> draft can accommodate the change.
>
> Regards
> Sam Narasimhan
> Motorola
>
> -----Original Message-----
> From: Zvi Lifshitz [mailto:ZviL@optibase.com]
> Sent: Tuesday, September 12, 2000 10:18 AM
> To: Narasimhan, Sam (SD-EX); 'jan.vandermeer@philips.com'
> Cc: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
> Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
> to the last call
>
>
> [Narasimhan, Sam (SD-EX):]
>
> If this draft were to say that receivers should parse over the
> adaptation bytes and not process them (as this draft does not
> define these), then such receivers will be able to process payload
> from future draft (from MPEG)
>
> [Reply:]
>
> They will not, because they don't know the payload type of the future
> draft
> therefore they will not know they can process payloads from the future
> draft
> even though they can.
>
> If they know, they also know that the future draft has extra header info.
> So
> work on the future draft, not on the current draft.
>
> Regards,
>
> z
> soon the whole world will STREAM
> ==================================================================
> Zvi Lifshitz				Phone +972(2)679-4788
> zvil@optibase.com			Fax +972(2)679-4789
> Optibase Ltd.				Mobile: +972(53)461-787
> http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
> Send SMS from:
> http://isend.cellcom.co.il/English/Login.htm
> ==================================================================
> > Come see us at:
> Streaming Media Europe Earls Court Centre, Booth # 628, London, October
> 10th-12th, 2000
>




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Subject: Re: In band signalling of PT
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stewe@cs.tu-berlin.de wrote:

> Folks,
>
> the recent discussion about the MPEG-4 ES payload brought up an
> issue that, IMHO, may be a shortcoming of current RTP and could
> easily be fixed.  This shortcoming is RTP's need for the out-of-band
> announcement/negotiation of parameters, including the PT.
>
> I wonder whether it may make sense to signal such information in-band=

> as well, e.g. through sender reports or through special media packets=
.
> That would allow receivers to "tune" to RTP broadcasts with only
> IP/Port information (which could, for example, be owned by a TV stati=
on).
>
> Questions:
>
> a) Did I miss something in the current spec?  Is this already specifi=
ed?

you can do that out of band using SAP with the config info in the
SAP packets

>
> b) Does such a scenario make sense?

Oh YES !


> Specifically, how do people think
>    to include session announcements in the media packetstream (or in
>    RTCP)?  I recall that some time ago there was some discussion
>    about architectural cleanness (it does / doesn't make sense to inc=
lude
>    session info in the media stream).  What is the status of that?

I think config info in the media stream is not a very good idea,
design-wise, but it can be made to work, of course

>
> c) Would such an enhancement have any impact on the current MPEG
>    discussion? (IMHO not, but I may have missed something)
>

if I recap the MPEG-4 discussion the problem is that the data stream sy=
ntax
can (video) or cannot  (audio, other system stuff, ODs, BIFS) include c=
onfig
info,
therefore we have an homogeneity problem for system/transport layers th=
at
should be
media unaware

>
> Stephan

--
Philippe Gentric
Software Architect
PHILIPS DIGITAL NETWORKS
Broadcast & Internet Delivery Solutions
Laboratoires d'Electronique Philips
B.P. 15 - 22, Av. Descartes
94453 Limeil-Brevannes Cedex France
Phone  :   33 (0)1 45 10 68 12
Fax    :   33 (0)1 45 10 69 60
philippe.gentric@philips.com


=



From rem-conf Wed Sep 13 04:09:38 2000 
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From: Herpel Carsten <HerpelC@thmulti.com>
To: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to
		the last call
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[Zvi]

Therefore I propose to kill this discussion about modifications to the ES 
format, and concentrating how to improve the Generic format, if necessary.

[Carsten]

Amen! (But I do think Guido made a good summary of the problem, though ;-)


May I mention again that for the Audio portion of the ES payload format the
extra byte does not buy anything anyway, since here we have the situation
that the ES format uses LATM while the generic one as it stands today does
not. 

We said in Paris that we want to correct MPEG-4 Systems in order to allow
multiple AUs per SL. But I don't recall we said that we want to put LATM
inside SL packets. So incompatibility between payload formats is guaranteed,
no matter whether we have the Jan-byte ;-) or not. (Disclaimer: This
observation is meant to trigger discussion about the _generic_ format ;-)

Regards,
Carsten



From rem-conf Wed Sep 13 08:38:25 2000 
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From: Zvi Lifshitz <zvil@csi.com>
To: "'Narasimhan, Sam (SD-EX)'" <SNarasimhan@gi.com>,
        "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>
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Sam, what you want either will not work or is not necessary.  I'm sorry we didn't manage to convince you. I've run out of words...

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm ==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October 10th-12th, 2000


-----Original Message-----
From:	Narasimhan, Sam (SD-EX) [SMTP:SNarasimhan@gi.com]
Sent:	Wednesday, September 13, 2000 5:28 PM
To:	'Zvi Lifshitz'; 'Narasimhan, Sam (SD-EX)'; 'jan.vandermeer@philips.com'
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to	the last call

 
Zvi,

1.  I thought all the comments were to make changes to 
    Kikuchi-San's draft NOW - not later. If it just added
    the adaptation length followed by length bytes before
    the ES payload (in the RTP payload), then content
    authors could author content that can play on receivers
    that implemented Kikuchi-San's draft as well as the
    proposal that will come out of MPEG. I am assuming
    that the MPEG proposal will use this adaptation
    bytes to signal type of content. If so, this will
    make Kikuchi-San's draft a sub-set of what MPEG will
    propose (preferable was the term you used). The only way
    this can happen is if Kikuchi-San's draft can be modified
    NOW.
2.  If Kikuchi-San's draft can be modified, it does NOT have to
    specify the content of adaptation header payload. MPEG will
    do it. What all it has to say is use the 'adaptation length'
    field to offset where you pick up the ES payload.
3.  In MPEG-2, the byte that follows the adaptation length 
    signals what is in the rest of the adaptation header such
    as PCR, OPCR, splicing etc. Those that cannot process the
    adaptation header just use the adaptation length to offset to
    the next payload start (such as PES header or ES). This
    is what I am assuming MPEG will do for 'generic' mapping. The
    byte after adaptation header may signal presence of DTS,
    SL header, flexmux header, scalable data, buffer etc.
4.  If Kikuchi-San's draft progresses 'as is' and MPEG uses the
    adaptation header, then content authors that used the MPEG
    coding can only play it on receivers that implemented the
    re-assembly of MPEG 'generic' mapping and not on those that
    'ONLY' implemented Kikuchi-San's draft (if it was not modified).
    CE vendors can choose to implement receivers that can support
    multiple payload formats. However, if Kikuchi-San's draft can
    be modified NOW, we may see a single payload format for MPEG-4
    and I believe that is a plus and thus NECESSARY.

Best Regards
Sam Narasimhan 
Motorola

-----Original Message-----
From: Zvi Lifshitz [mailto:zvil@csi.com]
Sent: Wednesday, September 13, 2000 12:52 AM
To: 'Narasimhan, Sam (SD-EX)'; 'jan.vandermeer@philips.com'
Cc: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
to the last call


Sam,

It seems you want the current draft to be modified in the future. Then:

1. I don't believe this is possible.
2. I don't think this is necessary.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000


-----Original Message-----
From:	Narasimhan, Sam (SD-EX) [SMTP:SNarasimhan@gi.com]
Sent:	Tuesday, September 12, 2000 6:21 PM
To:	'Zvi Lifshitz'; Narasimhan, Sam (SD-EX);
'jan.vandermeer@philips.com'
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to	the last call

 
Zvi,

 I am assuming that the payload type from the future draft
could be set to the same value as this draft, if this
draft can accommodate the change.

Regards
Sam Narasimhan
Motorola

-----Original Message-----
From: Zvi Lifshitz [mailto:ZviL@optibase.com]
Sent: Tuesday, September 12, 2000 10:18 AM
To: Narasimhan, Sam (SD-EX); 'jan.vandermeer@philips.com'
Cc: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
to the last call


[Narasimhan, Sam (SD-EX):]

If this draft were to say that receivers should parse over the
adaptation bytes and not process them (as this draft does not
define these), then such receivers will be able to process payload
>from future draft (from MPEG)

[Reply:]

They will not, because they don't know the payload type of the future draft
therefore they will not know they can process payloads from the future draft
even though they can.

If they know, they also know that the future draft has extra header info. So
work on the future draft, not on the current draft.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000




From rem-conf Wed Sep 13 08:38:25 2000 
From rem-conf-request@es.net Wed Sep 13 08:38:25 2000
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Date: Wed, 13 Sep 2000 11:28:26 -0400
From: "Narasimhan, Sam (SD-EX)" <SNarasimhan@gi.com>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to	the
 last call
To: 'Zvi Lifshitz' <zvil@csi.com>,
 "'Narasimhan, Sam (SD-EX)'" <SNarasimhan@GI.COM>,
 "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>
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Zvi,

1.  I thought all the comments were to make changes to 
    Kikuchi-San's draft NOW - not later. If it just added
    the adaptation length followed by length bytes before
    the ES payload (in the RTP payload), then content
    authors could author content that can play on receivers
    that implemented Kikuchi-San's draft as well as the
    proposal that will come out of MPEG. I am assuming
    that the MPEG proposal will use this adaptation
    bytes to signal type of content. If so, this will
    make Kikuchi-San's draft a sub-set of what MPEG will
    propose (preferable was the term you used). The only way
    this can happen is if Kikuchi-San's draft can be modified
    NOW.
2.  If Kikuchi-San's draft can be modified, it does NOT have to
    specify the content of adaptation header payload. MPEG will
    do it. What all it has to say is use the 'adaptation length'
    field to offset where you pick up the ES payload.
3.  In MPEG-2, the byte that follows the adaptation length 
    signals what is in the rest of the adaptation header such
    as PCR, OPCR, splicing etc. Those that cannot process the
    adaptation header just use the adaptation length to offset to
    the next payload start (such as PES header or ES). This
    is what I am assuming MPEG will do for 'generic' mapping. The
    byte after adaptation header may signal presence of DTS,
    SL header, flexmux header, scalable data, buffer etc.
4.  If Kikuchi-San's draft progresses 'as is' and MPEG uses the
    adaptation header, then content authors that used the MPEG
    coding can only play it on receivers that implemented the
    re-assembly of MPEG 'generic' mapping and not on those that
    'ONLY' implemented Kikuchi-San's draft (if it was not modified).
    CE vendors can choose to implement receivers that can support
    multiple payload formats. However, if Kikuchi-San's draft can
    be modified NOW, we may see a single payload format for MPEG-4
    and I believe that is a plus and thus NECESSARY.

Best Regards
Sam Narasimhan 
Motorola

-----Original Message-----
From: Zvi Lifshitz [mailto:zvil@csi.com]
Sent: Wednesday, September 13, 2000 12:52 AM
To: 'Narasimhan, Sam (SD-EX)'; 'jan.vandermeer@philips.com'
Cc: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
to the last call


Sam,

It seems you want the current draft to be modified in the future. Then:

1. I don't believe this is possible.
2. I don't think this is necessary.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000


-----Original Message-----
From:	Narasimhan, Sam (SD-EX) [SMTP:SNarasimhan@gi.com]
Sent:	Tuesday, September 12, 2000 6:21 PM
To:	'Zvi Lifshitz'; Narasimhan, Sam (SD-EX);
'jan.vandermeer@philips.com'
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to	the last call

 
Zvi,

 I am assuming that the payload type from the future draft
could be set to the same value as this draft, if this
draft can accommodate the change.

Regards
Sam Narasimhan
Motorola

-----Original Message-----
From: Zvi Lifshitz [mailto:ZviL@optibase.com]
Sent: Tuesday, September 12, 2000 10:18 AM
To: Narasimhan, Sam (SD-EX); 'jan.vandermeer@philips.com'
Cc: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
to the last call


[Narasimhan, Sam (SD-EX):]

If this draft were to say that receivers should parse over the
adaptation bytes and not process them (as this draft does not
define these), then such receivers will be able to process payload
>from future draft (from MPEG)

[Reply:]

They will not, because they don't know the payload type of the future draft
therefore they will not know they can process payloads from the future draft
even though they can.

If they know, they also know that the future draft has extra header info. So
work on the future draft, not on the current draft.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000



From rem-conf Wed Sep 13 08:50:58 2000 
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From: "Narasimhan, Sam (SD-EX)" <SNarasimhan@gi.com>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to	the
 last call
To: 'Zvi Lifshitz' <zvil@csi.com>,
 "'Narasimhan, Sam (SD-EX)'" <SNarasimhan@GI.COM>,
 "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>
Cc: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
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 I am sorry you see it this way. I do not agree with you; however
I have also run out of words and will save it for the MPEG
work.

Regards
Sam Narasimhan
Motorola

-----Original Message-----
From: Zvi Lifshitz [mailto:zvil@csi.com]
Sent: Wednesday, September 13, 2000 9:33 AM
To: 'Narasimhan, Sam (SD-EX)'; 'jan.vandermeer@philips.com'
Cc: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
to the last call


Sam, what you want either will not work or is not necessary.  I'm sorry we
didn't manage to convince you. I've run out of words...

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000


-----Original Message-----
From:	Narasimhan, Sam (SD-EX) [SMTP:SNarasimhan@gi.com]
Sent:	Wednesday, September 13, 2000 5:28 PM
To:	'Zvi Lifshitz'; 'Narasimhan, Sam (SD-EX)';
'jan.vandermeer@philips.com'
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to	the last call

 
Zvi,

1.  I thought all the comments were to make changes to 
    Kikuchi-San's draft NOW - not later. If it just added
    the adaptation length followed by length bytes before
    the ES payload (in the RTP payload), then content
    authors could author content that can play on receivers
    that implemented Kikuchi-San's draft as well as the
    proposal that will come out of MPEG. I am assuming
    that the MPEG proposal will use this adaptation
    bytes to signal type of content. If so, this will
    make Kikuchi-San's draft a sub-set of what MPEG will
    propose (preferable was the term you used). The only way
    this can happen is if Kikuchi-San's draft can be modified
    NOW.
2.  If Kikuchi-San's draft can be modified, it does NOT have to
    specify the content of adaptation header payload. MPEG will
    do it. What all it has to say is use the 'adaptation length'
    field to offset where you pick up the ES payload.
3.  In MPEG-2, the byte that follows the adaptation length 
    signals what is in the rest of the adaptation header such
    as PCR, OPCR, splicing etc. Those that cannot process the
    adaptation header just use the adaptation length to offset to
    the next payload start (such as PES header or ES). This
    is what I am assuming MPEG will do for 'generic' mapping. The
    byte after adaptation header may signal presence of DTS,
    SL header, flexmux header, scalable data, buffer etc.
4.  If Kikuchi-San's draft progresses 'as is' and MPEG uses the
    adaptation header, then content authors that used the MPEG
    coding can only play it on receivers that implemented the
    re-assembly of MPEG 'generic' mapping and not on those that
    'ONLY' implemented Kikuchi-San's draft (if it was not modified).
    CE vendors can choose to implement receivers that can support
    multiple payload formats. However, if Kikuchi-San's draft can
    be modified NOW, we may see a single payload format for MPEG-4
    and I believe that is a plus and thus NECESSARY.

Best Regards
Sam Narasimhan 
Motorola

-----Original Message-----
From: Zvi Lifshitz [mailto:zvil@csi.com]
Sent: Wednesday, September 13, 2000 12:52 AM
To: 'Narasimhan, Sam (SD-EX)'; 'jan.vandermeer@philips.com'
Cc: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
to the last call


Sam,

It seems you want the current draft to be modified in the future. Then:

1. I don't believe this is possible.
2. I don't think this is necessary.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000


-----Original Message-----
From:	Narasimhan, Sam (SD-EX) [SMTP:SNarasimhan@gi.com]
Sent:	Tuesday, September 12, 2000 6:21 PM
To:	'Zvi Lifshitz'; Narasimhan, Sam (SD-EX);
'jan.vandermeer@philips.com'
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to	the last call

 
Zvi,

 I am assuming that the payload type from the future draft
could be set to the same value as this draft, if this
draft can accommodate the change.

Regards
Sam Narasimhan
Motorola

-----Original Message-----
From: Zvi Lifshitz [mailto:ZviL@optibase.com]
Sent: Tuesday, September 12, 2000 10:18 AM
To: Narasimhan, Sam (SD-EX); 'jan.vandermeer@philips.com'
Cc: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
to the last call


[Narasimhan, Sam (SD-EX):]

If this draft were to say that receivers should parse over the
adaptation bytes and not process them (as this draft does not
define these), then such receivers will be able to process payload
>from future draft (from MPEG)

[Reply:]

They will not, because they don't know the payload type of the future draft
therefore they will not know they can process payloads from the future draft
even though they can.

If they know, they also know that the future draft has extra header info. So
work on the future draft, not on the current draft.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000



From rem-conf Wed Sep 13 09:04:51 2000 
From rem-conf-request@es.net Wed Sep 13 09:04:51 2000
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From: CURET Dominique FTRD/DMI/REN <dominique.curet@rd.francetelecom.fr>
To: 4on2andIP-sys@advent.ee.columbia.edu, rem-conf@es.net
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Dear all, dear Young-Kwon,

we are supposed to meet on Saturday(21/10) and Sunday (22/10)
before the MPEG meeting in La Baule.

Can we say something like:
     Saturday from 14 to 2000
     Sunday from 0900 to 1800
Or only the Sunday from 0900 to 1800

comments

regards
_________________________________________
> Dominique Curet
> France T=E9l=E9com R&D - DMI/PIA
> % + 33 (0)2 99 12 40 05   Fax : + 33 (0)2 99 12 40 98
> )  CCETT       B.P. 59            4, rue du Clos Courtel=20
> 35512           Cesson-Sevigne   Cedex   09     FRANCE
> mailto:dominique.curet@rd.francetelecom.fr
_________________________________________





From rem-conf Wed Sep 13 09:36:52 2000 
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From: Kris Huber <khuber@sorensonlabs.com>
To: Zvi Lifshitz <zvil@csi.com>, Colin Perkins <csp@east.isi.edu>
Cc: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu, 
	"'Narasimhan, Sam (SD-EX)'" <SNarasimhan@gi.com>, 
	"'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to
		the last call
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Hello Zvi, Colin [last comment before letting this discussion transition to
generic format]:

I think some of the discussion has resulted from misunderstanding of the
semantics of the payload type field apparently involving a change (by IETF?)
>from static to dynamic binding of an external "payload name" to this field.
In any case, the desire has been to see whether or not this format can
somehow be a subset of the soon-to-be more generic payload format.  Leaving
out lack of LATM in the draft generic format, the answer appears to still be
"no", unless the payload name for the future format is assigned now and a
subset of its semantics, including how to read its adaptation field, are
included in draft-ietf-avt-rtp-mpeg4-es-03.  But this would be
unprecedented, I suspect (possible?  I don't know).  The other option is if
another static payload type is assigned and a future payload type of a
different name (MPEG SL format) uses the same payload type.

Why, some have asked?  To avoid the redundancy that we video compression
engineers hate so much!  How would this redundancy result?  Information
broadcast services begin initially using the Kichuchi-san payload format,
then have to make the decision later on between continuing to use it, and
either doubling their network bandwidth to do the new broadcasting services
or provide services support the new format (since the video ES data will
tend to dominate the bandwidth).  Allowing early-adopter non-upgradeable
devices to cease to function is another possibility, but that also has a
high cost if the number of such devices is large.  You might wonder about an
approach in which a new enhancement payload type is used rather than a
single scalable payload type (e.g., generic format to add the enhancement
information), and perhaps this is considered the "right way".  The only
problem I see with this that the information goes in separate packets and
therefore additional buffer memory and synchronization functionality are
required.  Let me clarify, for point-to-point usage there is no efficiency
problem, it is only for the broadcast (multipoint) case.

It seems that static payload typing, or early assignment of payload name +
"Jan-byte" + other modifications would give the proposed interoperability
functionality.  Out of curiosity let me ask (Colin?), how is "there will be
no more static payload assignments" formally stated, with SHALL NOT or
SHOULD NOT?  Or is it a policy existing outside of an RFC?  How many devices
would be obsoleted by a new static payload type?

Best regards,
Kris

-----Original Message-----
From: Zvi Lifshitz [mailto:zvil@csi.com]
Sent: Wednesday, September 13, 2000 10:33 AM
To: 'Narasimhan, Sam (SD-EX)'; 'jan.vandermeer@philips.com'
Cc: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
to the last call


Sam, what you want either will not work or is not necessary.  I'm sorry we
didn't manage to convince you. I've run out of words...

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000


-----Original Message-----
From:	Narasimhan, Sam (SD-EX) [SMTP:SNarasimhan@gi.com]
Sent:	Wednesday, September 13, 2000 5:28 PM
To:	'Zvi Lifshitz'; 'Narasimhan, Sam (SD-EX)';
'jan.vandermeer@philips.com'
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to	the last call

 
Zvi,

1.  I thought all the comments were to make changes to 
    Kikuchi-San's draft NOW - not later. If it just added
    the adaptation length followed by length bytes before
    the ES payload (in the RTP payload), then content
    authors could author content that can play on receivers
    that implemented Kikuchi-San's draft as well as the
    proposal that will come out of MPEG. I am assuming
    that the MPEG proposal will use this adaptation
    bytes to signal type of content. If so, this will
    make Kikuchi-San's draft a sub-set of what MPEG will
    propose (preferable was the term you used). The only way
    this can happen is if Kikuchi-San's draft can be modified
    NOW.
2.  If Kikuchi-San's draft can be modified, it does NOT have to
    specify the content of adaptation header payload. MPEG will
    do it. What all it has to say is use the 'adaptation length'
    field to offset where you pick up the ES payload.
3.  In MPEG-2, the byte that follows the adaptation length 
    signals what is in the rest of the adaptation header such
    as PCR, OPCR, splicing etc. Those that cannot process the
    adaptation header just use the adaptation length to offset to
    the next payload start (such as PES header or ES). This
    is what I am assuming MPEG will do for 'generic' mapping. The
    byte after adaptation header may signal presence of DTS,
    SL header, flexmux header, scalable data, buffer etc.
4.  If Kikuchi-San's draft progresses 'as is' and MPEG uses the
    adaptation header, then content authors that used the MPEG
    coding can only play it on receivers that implemented the
    re-assembly of MPEG 'generic' mapping and not on those that
    'ONLY' implemented Kikuchi-San's draft (if it was not modified).
    CE vendors can choose to implement receivers that can support
    multiple payload formats. However, if Kikuchi-San's draft can
    be modified NOW, we may see a single payload format for MPEG-4
    and I believe that is a plus and thus NECESSARY.

Best Regards
Sam Narasimhan 
Motorola

-----Original Message-----
From: Zvi Lifshitz [mailto:zvil@csi.com]
Sent: Wednesday, September 13, 2000 12:52 AM
To: 'Narasimhan, Sam (SD-EX)'; 'jan.vandermeer@philips.com'
Cc: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
to the last call


Sam,

It seems you want the current draft to be modified in the future. Then:

1. I don't believe this is possible.
2. I don't think this is necessary.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000


-----Original Message-----
From:	Narasimhan, Sam (SD-EX) [SMTP:SNarasimhan@gi.com]
Sent:	Tuesday, September 12, 2000 6:21 PM
To:	'Zvi Lifshitz'; Narasimhan, Sam (SD-EX);
'jan.vandermeer@philips.com'
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to	the last call

 
Zvi,

 I am assuming that the payload type from the future draft
could be set to the same value as this draft, if this
draft can accommodate the change.

Regards
Sam Narasimhan
Motorola

-----Original Message-----
From: Zvi Lifshitz [mailto:ZviL@optibase.com]
Sent: Tuesday, September 12, 2000 10:18 AM
To: Narasimhan, Sam (SD-EX); 'jan.vandermeer@philips.com'
Cc: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
to the last call


[Narasimhan, Sam (SD-EX):]

If this draft were to say that receivers should parse over the
adaptation bytes and not process them (as this draft does not
define these), then such receivers will be able to process payload
>from future draft (from MPEG)

[Reply:]

They will not, because they don't know the payload type of the future draft
therefore they will not know they can process payloads from the future draft
even though they can.

If they know, they also know that the future draft has extra header info. So
work on the future draft, not on the current draft.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000



From rem-conf Wed Sep 13 09:50:19 2000 
From rem-conf-request@es.net Wed Sep 13 09:50:19 2000
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From: Zvi Lifshitz <zvil@csi.com>
To: "'Kris Huber'" <khuber@sorensonlabs.com>,
        Colin Perkins
	 <csp@east.isi.edu>
Cc: "rem-conf@es.net" <rem-conf@es.net>,
        "4on2andIP-sys@advent.ee.columbia.edu" <4on2andIP-sys@advent.ee.columbia.edu>,
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 =?us-ascii?Q?mpeg4=2Des=2D03_in_response_to=09the_last_call?=
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Kris, all,

I'm not so concerned about future upgradablity.  The ES format is there 
now, but also we have a pretty good idea about how the Generic format is 
going to look like. As an implementer of MPEG-4 server technology, that's 
enough for me, and the official status of the proposals is not so 
important. I'm sure there are others who have the same approach, and 
together we'll build state of the art systems. This is assuming we use our 
time to develop, and not spend it on useless discussions....

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm 
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October 
10th-12th, 2000


-----Original Message-----
From:	Kris Huber [SMTP:khuber@sorensonlabs.com]
Sent:	Wednesday, September 13, 2000 6:30 PM
To:	Zvi Lifshitz; Colin Perkins
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu; 'Narasimhan, Sam 
(SD-EX)'; 'jan.vandermeer@philips.com'
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response 
to	the last call


Hello Zvi, Colin [last comment before letting this discussion transition to
generic format]:

I think some of the discussion has resulted from misunderstanding of the
semantics of the payload type field apparently involving a change (by 
IETF?)
>from static to dynamic binding of an external "payload name" to this field.
In any case, the desire has been to see whether or not this format can
somehow be a subset of the soon-to-be more generic payload format.  Leaving
out lack of LATM in the draft generic format, the answer appears to still 
be
"no", unless the payload name for the future format is assigned now and a
subset of its semantics, including how to read its adaptation field, are
included in draft-ietf-avt-rtp-mpeg4-es-03.  But this would be
unprecedented, I suspect (possible?  I don't know).  The other option is if
another static payload type is assigned and a future payload type of a
different name (MPEG SL format) uses the same payload type.

Why, some have asked?  To avoid the redundancy that we video compression
engineers hate so much!  How would this redundancy result?  Information
broadcast services begin initially using the Kichuchi-san payload format,
then have to make the decision later on between continuing to use it, and
either doubling their network bandwidth to do the new broadcasting services
or provide services support the new format (since the video ES data will
tend to dominate the bandwidth).  Allowing early-adopter non-upgradeable
devices to cease to function is another possibility, but that also has a
high cost if the number of such devices is large.  You might wonder about 
an
approach in which a new enhancement payload type is used rather than a
single scalable payload type (e.g., generic format to add the enhancement
information), and perhaps this is considered the "right way".  The only
problem I see with this that the information goes in separate packets and
therefore additional buffer memory and synchronization functionality are
required.  Let me clarify, for point-to-point usage there is no efficiency
problem, it is only for the broadcast (multipoint) case.

It seems that static payload typing, or early assignment of payload name +
"Jan-byte" + other modifications would give the proposed interoperability
functionality.  Out of curiosity let me ask (Colin?), how is "there will be
no more static payload assignments" formally stated, with SHALL NOT or
SHOULD NOT?  Or is it a policy existing outside of an RFC?  How many 
devices
would be obsoleted by a new static payload type?

Best regards,
Kris




From rem-conf Wed Sep 13 09:52:10 2000 
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To: <SNarasimhan@gi.com>, <zvil@csi.com>, <Guido.Franceschini@CSELT.IT>
Cc: <rem-conf@es.net>, <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the
         last call
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Guido,

Thanks. You made your point very clear but not clear enough for all. Ne=
ither
did I. We seem to live in too different worlds to really understand eac=
h
other. Sad.

Regards,

Jan

-----Original Message-----
From:	Guido.Franceschini@CSELT.IT [mailto:Guido.Franceschini@CSELT.IT]
Sent:	woensdag, 13 september, 2000 09:26
To:	zvil@csi.com; Jan vanderMeer; SNarasimhan@gi.com
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response=
 to
the last call


Hi all,
my own considerations on this thread

In RTP and IETF procedure terms, I understand this thread on the extra =
byte
as:
1) modify the current ES draft to accomodate a payload header starting =
with
a byte indicating its length. No specification of what the payload head=
er
carries
2) allow the modified ES draft to enter the RFC standard track and be w=
idely
implemented
3) propose a new draft for 'generic' ES or SL mapping, which is identic=
al to
the modified ES draft but also defines what the payload header carries
4) the 'generic' draft becomes standard and _obsoletes_ the ES draft.
Devices implementing the ES draft will skip the payload header informat=
ion
but still understand the media content.

Well, I think we have a technical problem here, which has nothing to do=
 with
usual practices / procedures / timelines.
The problem that I see is that:
- the ES draft defines 2 Payload Formats: 1 for Audio, 1 for Video
- the 'generic' draft will define 1 Payload Format only, and the type o=
f
media (Audio, Video, BIFS ...) being carried will be specified using MP=
EG-4
Object Descriptors.

What we achieved at the Paris meeting (before discussing this extra byt=
e
feature) has been to allow MPEG-4 Systems terminals implementing the
'generic' draft to use the ES draft as well withouth any need to duplic=
ate
implementations (i.e.: the ES draft is a subset of the generic draft in=
 the
Data Plane).
Now, the 'extra byte' proposal would allow devices implementing the ES =
draft
to be compatible in the Data Plane with the 'generic' format, too.

Compatibility in the Control Plane (i.e. Payload Type
negotiation/announcement) was not considered in Paris.
I believe that devices implementing the 'generic' draft would have no
difficulties in understanding 1 more SDP keyword or MIME type as introd=
uced
by the ES draft.
But the reverse is more difficult and tricky. As I outlined above, the
'generic' draft would not specify by itself the type of media content i=
t is
carrying, and would rely on the Object Descriptor to do that. If the go=
al is
to allow devices implementing the ES draft to be compatible with future=

content delivered by means of the 'generic' draft, than we need to spec=
ify
NOW how to duplicate the information in the OD so that the media type i=
s
made explicit for those devices. And then we will have to accept that
devices implementing MPEG-4 Systems get the indication of the media typ=
e
twice :-(

Did I make the point clear ?

Guido

> ----------
> From: 	Zvi Lifshitz[SMTP:zvil@csi.com]
> Sent: 	Wednesday, September 13, 2000 9:51 AM
> To: 	'Narasimhan, Sam (SD-EX)'; 'jan.vandermeer@philips.com'
> Cc: 	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
> Subject: 	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
> response to	the last call
>
> Sam,
>
> It seems you want the current draft to be modified in the future. The=
n:
>
> 1. I don't believe this is possible.
> 2. I don't think this is necessary.
>
> Regards,
>
> z
> soon the whole world will STREAM
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> Zvi Lifshitz				Phone +972(2)679-4788
> zvil@optibase.com			Fax +972(2)679-4789
> Optibase Ltd.				Mobile: +972(53)461-787
> http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
> Send SMS from:
> http://isend.cellcom.co.il/English/Login.htm
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> > Come see us at:
> Streaming Media Europe Earls Court Centre, Booth # 628, London, Octob=
er
> 10th-12th, 2000
>
>
> -----Original Message-----
> From:	Narasimhan, Sam (SD-EX) [SMTP:SNarasimhan@gi.com]
> Sent:	Tuesday, September 12, 2000 6:21 PM
> To:	'Zvi Lifshitz'; Narasimhan, Sam (SD-EX);
> 'jan.vandermeer@philips.com'
> Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
> Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
> response to	the last call
>
>
> Zvi,
>
>  I am assuming that the payload type from the future draft
> could be set to the same value as this draft, if this
> draft can accommodate the change.
>
> Regards
> Sam Narasimhan
> Motorola
>
> -----Original Message-----
> From: Zvi Lifshitz [mailto:ZviL@optibase.com]
> Sent: Tuesday, September 12, 2000 10:18 AM
> To: Narasimhan, Sam (SD-EX); 'jan.vandermeer@philips.com'
> Cc: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
> Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in respon=
se
> to the last call
>
>
> [Narasimhan, Sam (SD-EX):]
>
> If this draft were to say that receivers should parse over the
> adaptation bytes and not process them (as this draft does not
> define these), then such receivers will be able to process payload
> from future draft (from MPEG)
>
> [Reply:]
>
> They will not, because they don't know the payload type of the future=

> draft
> therefore they will not know they can process payloads from the futur=
e
> draft
> even though they can.
>
> If they know, they also know that the future draft has extra header i=
nfo.
> So
> work on the future draft, not on the current draft.
>
> Regards,
>
> z
> soon the whole world will STREAM
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> Zvi Lifshitz				Phone +972(2)679-4788
> zvil@optibase.com			Fax +972(2)679-4789
> Optibase Ltd.				Mobile: +972(53)461-787
> http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
> Send SMS from:
> http://isend.cellcom.co.il/English/Login.htm
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> > Come see us at:
> Streaming Media Europe Earls Court Centre, Booth # 628, London, Octob=
er
> 10th-12th, 2000
>

=



From rem-conf Wed Sep 13 09:52:27 2000 
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To: <csp@east.isi.edu>, <zvil@csi.com>
Cc: <rem-conf@es.net>, <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the
         last call
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Date: Wed, 13 Sep 2000 18:50:35 +0200
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Dear Zvi,

My conclusion is that I failed to let you understand the issue. And you=
 seem
to suggest that you are not the only one. Your suggestion to kill this
discussion ignores the problem addressed. I agree that putting our head=
s
into the sand is a way to go forward. But in my opinion we should take =
our
responsibility to define a future proof solution.

Best regards,

Jan

soon there will be more in this world than plain streaming ...

-----Original Message-----
From:	zvil@csi.com [mailto:zvil@csi.com]
Sent:	woensdag, 13 september, 2000 09:38
To:	csp@east.isi.edu; Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response=
 to
the last call


Jan,

Your nice proposal can be incorporated in the Generic format. I hope we=

managed to explain why it is useless with the ES format.

z
soon the whole world will STREAM
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October=

10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Wednesday, September 13, 2000 5:28 AM
To:	csp@east.isi.edu
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response=
 to
the last call


Colin, all,

I think it is important to keep in mind that often MPEG uses in-stream
signaling. For example, after the length byte, the adaptation field may=

start with an identifier that specifies the sort of enhancement. Based =
on
this approach, the following would be needed :

1) Define an MPEG-4 ES payload format that allows for an adaptation fie=
ld
without defining the meaning of the adaptation field.

2) Define the meaning of the adaptation field in the "generic MPEG-4
specification". For carriage of MPEG-4 ES and MPEG-4 SL streams the "ge=
neric
specification" defines the use of exactly_the_same payload type as defi=
ned
in 1). The only difference is the defined use of the adaptation field.
Whether or not an SL header is carried is signaled in the adaptation fi=
eld.

3) Next to the payload types for MPEG-4 ES and SL streams, the "generic=

specification" may also define payloads for other streams, such as Flex=
Mux
streams. For these, other payload types will be needed.

Now the initial services start using the MPEG-4 ES payload format; thou=
gh
the adaptation field is not used, this field is allowed to occur. Futur=
e
services may use exactly the same payload type, but with the adaptation=

field as defined by the "generic specification".

It may also be important to understand that the presence of the adaptat=
ion
field is intended to be transparent from RTP point of view and for the
session in general. The adaptation field serves MPEG-4 system purposes =
which
are fully orthogonal to RTP and the streaming session.

Kind regards,

Jan

=



From rem-conf Wed Sep 13 10:15:43 2000 
From rem-conf-request@es.net Wed Sep 13 10:15:42 2000
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Message-ID: <01C01DBE.E3B8BA60.zvil@csi.com>
From: Zvi Lifshitz <zvil@csi.com>
To: "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>,
        "csp@east.isi.edu" <csp@east.isi.edu>
Cc: "rem-conf@es.net" <rem-conf@es.net>,
        "4on2andIP-sys@advent.ee.columbia.edu" <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call
Date: Wed, 13 Sep 2000 20:12:05 +0200
Organization: Optibase Ltd.
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Hi Jan,

I have the same conclusion. Maybe I would have had another conclusion if I 
saw answers to the questions Collin and me and other asked. I'll try to 
summarize it for last time:

1. Applications that know only the ES format will not recognize a future 
format because they will not identify the format name.
2. Therefore having extension mechanism in the ES format without fully 
describing the extensions makes no sense.
3. Defining the extensions in the ES format amounts to replacing it by the 
Generic format. Practically in time-to-market terms, that means suppressing 
the ES format. What is the gain, if the Generic format is going to be a 
superset of the ES format?

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm 
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October 
10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Wednesday, September 13, 2000 6:51 PM
To:	csp@east.isi.edu; zvil@csi.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to 
the last call


Dear Zvi,

My conclusion is that I failed to let you understand the issue. And you 
seem
to suggest that you are not the only one. Your suggestion to kill this
discussion ignores the problem addressed. I agree that putting our heads
into the sand is a way to go forward. But in my opinion we should take our
responsibility to define a future proof solution.

Best regards,

Jan

soon there will be more in this world than plain streaming ...




From rem-conf Wed Sep 13 10:19:43 2000 
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From: "Allison, Art" <AAllison@nab.org>
To: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to
	 the last call
Date: Wed, 13 Sep 2000 13:16:40 -0400
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>From this seat at the NAB it seems to me that unless the business models
change and a serious business justification exists, one should not expect
broadcasters (at least the small set of US broadcasters) to change methods
to one that obsolete previous deployments once a method is started. On the
other hand, as this is MPEG4 over IP, and IP will be a clean layer so that a
broadcaster need not worry about the content of the IP, only those
broadcasters that think consumers will complain to the station may care.  If
the problem is only on 'PCs', those users are less likely to complain.  One
could reason that manufacturers of 'TVs' would be unlikely to add support
for M4/IP for similar reasons.  If it doesn't work for broadcasters in the
US, they will find another way to provide the desired functionality.

Disclosure: This view may be biased by my assessment of MPEG-4's timing
capabilities in addition to the issues under discussion. <old dead issue>.
Art
::{)
-----Original Message-----
From: Kris Huber [mailto:khuber@sorensonlabs.com]
Sent: Wednesday, September 13, 2000 12:30 PM
To: Zvi Lifshitz; Colin Perkins
Cc: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu; 'Narasimhan,
Sam (SD-EX)'; 'jan.vandermeer@philips.com'
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
to the last call


Hello Zvi, Colin [last comment before letting this discussion transition to
generic format]:
<snip>

... Let me clarify, for point-to-point usage there is no efficiency
problem, it is only for the broadcast (multipoint) case. ...

<snip>

Best regards,
Kris

<snip>



From rem-conf Wed Sep 13 10:26:27 2000 
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To: "Luthra, Ajay (SD-EX)" <ALuthra@gi.com>
cc: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
In-Reply-To: Your message of "Tue, 12 Sep 2000 19:58:47 EDT."
             <973597126BDDD11197AA00805FA7EBC90342164D@ntas0026.gi.com> 
Date: Wed, 13 Sep 2000 13:23:24 -0400
From: Colin Perkins <csp@east.isi.edu>
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--> "Luthra, Ajay (SD-EX)" writes:
>Collin,
>   But you said in your previous email, " ... The payload type will have to
>be negotiated out of band anyway .. ". And in another email you said, "Yes"
>to the question - does one need to work this way in multicast environment
>also. And now you are saying it does not need to be negotiated. My
>understanding of this subject is not yet excellent. So, I do not understand
>these three emails from you which seem to contradict each other. 
>
>Does the payload type need to be negotiated or not?

The payload type is dynamically assigned, and has to be communicated
between the sender and the receiver(s) somehow. RTP does not specify
how that communication is done. In the unicast case it is typical to
negotiate, in the one-to-many multicast case the sender tells the
receivers (i.e. the receiver has no choice, so it can't really be 
called "negotiation"). But there is always out of band signalling.

Colin



From rem-conf Wed Sep 13 10:26:34 2000 
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To: <4on2andIP-sys@advent.ee.columbia.edu>
Cc: <rem-conf@es.net>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the
         last call
Message-ID: <0056890014059984000002L942*@MHS>
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Dear all,

I would like to put the problem in a little bit wider perspective. Seve=
ral
people perfectly understand the problem while some (?) others don't yet=

understand the problem, while suggesting that we are wasting our time w=
ith
useless discussions. As Ajay stated correctly, it is the responsibility=
 of
IETF to decide how to proceed with the current MPEG-4 ES proposal. The =
only
thing we can say right now is that several (perhaps even well respected=
)
MPEG members have substantial comments on the current proposal. As a
consequence it is likely that MPEG (members) will submit another propos=
al
for carriage of MPEG-4 ES to IETF. Though this proposal is expected to =
have
a high level of overlap with the current one, it is slightly different.=
 IETF
may wish to take this into account when coming to a decision.

Kind regards,

Jan

-----Original Message-----
From:	zvil@csi.com [mailto:zvil@csi.com]
Sent:	woensdag, 13 september, 2000 18:52
To:	csp@east.isi.edu; khuber@sorensonlabs.com
Cc:	Jan vanderMeer; SNarasimhan@gi.com;
4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response=
 to
the last call


Kris, all,

I'm not so concerned about future upgradablity.  The ES format is there=

now, but also we have a pretty good idea about how the Generic format i=
s
going to look like. As an implementer of MPEG-4 server technology, that=
's
enough for me, and the official status of the proposals is not so
important. I'm sure there are others who have the same approach, and
together we'll build state of the art systems. This is assuming we use =
our
time to develop, and not spend it on useless discussions....

Regards,

z
soon the whole world will STREAM
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October=

10th-12th, 2000


-----Original Message-----
From:	Kris Huber [SMTP:khuber@sorensonlabs.com]
Sent:	Wednesday, September 13, 2000 6:30 PM
To:	Zvi Lifshitz; Colin Perkins
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu; 'Narasimhan,=
 Sam
(SD-EX)'; 'jan.vandermeer@philips.com'
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response=

to	the last call


Hello Zvi, Colin [last comment before letting this discussion transitio=
n to
generic format]:

I think some of the discussion has resulted from misunderstanding of th=
e
semantics of the payload type field apparently involving a change (by
IETF?)
>from static to dynamic binding of an external "payload name" to this fi=
eld.
In any case, the desire has been to see whether or not this format can
somehow be a subset of the soon-to-be more generic payload format.  Lea=
ving
out lack of LATM in the draft generic format, the answer appears to sti=
ll
be
"no", unless the payload name for the future format is assigned now and=
 a
subset of its semantics, including how to read its adaptation field, ar=
e
included in draft-ietf-avt-rtp-mpeg4-es-03.  But this would be
unprecedented, I suspect (possible?  I don't know).  The other option i=
s if
another static payload type is assigned and a future payload type of a
different name (MPEG SL format) uses the same payload type.

Why, some have asked?  To avoid the redundancy that we video compressio=
n
engineers hate so much!  How would this redundancy result?  Information=

broadcast services begin initially using the Kichuchi-san payload forma=
t,
then have to make the decision later on between continuing to use it, a=
nd
either doubling their network bandwidth to do the new broadcasting serv=
ices
or provide services support the new format (since the video ES data wil=
l
tend to dominate the bandwidth).  Allowing early-adopter non-upgradeabl=
e
devices to cease to function is another possibility, but that also has =
a
high cost if the number of such devices is large.  You might wonder abo=
ut
an
approach in which a new enhancement payload type is used rather than a
single scalable payload type (e.g., generic format to add the enhanceme=
nt
information), and perhaps this is considered the "right way".  The only=

problem I see with this that the information goes in separate packets a=
nd
therefore additional buffer memory and synchronization functionality ar=
e
required.  Let me clarify, for point-to-point usage there is no efficie=
ncy
problem, it is only for the broadcast (multipoint) case.

It seems that static payload typing, or early assignment of payload nam=
e +
"Jan-byte" + other modifications would give the proposed interoperabili=
ty
functionality.  Out of curiosity let me ask (Colin?), how is "there wil=
l be
no more static payload assignments" formally stated, with SHALL NOT or
SHOULD NOT?  Or is it a policy existing outside of an RFC?  How many
devices
would be obsoleted by a new static payload type?

Best regards,
Kris

=



From rem-conf Wed Sep 13 10:28:10 2000 
From rem-conf-request@es.net Wed Sep 13 10:28:10 2000
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From: "Allison, Art" <AAllison@nab.org>
To: 4on2andIP-sys@advent.ee.columbia.edu, rem-conf@es.net
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to
	 the last call
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Here! here!

Art
::{)
-----Original Message-----
From: jan.vandermeer@philips.com [mailto:jan.vandermeer@philips.com]
Sent: Wednesday, September 13, 2000 12:51 PM
To: csp@east.isi.edu; zvil@csi.com
Cc: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
to the last call


Dear Zvi,

My conclusion is that I failed to let you understand the issue. And you seem
to suggest that you are not the only one. Your suggestion to kill this
discussion ignores the problem addressed. I agree that putting our heads
into the sand is a way to go forward. But in my opinion we should take our
responsibility to define a future proof solution.

Best regards,

Jan

soon there will be more in this world than plain streaming ...

-----Original Message-----
From:	zvil@csi.com [mailto:zvil@csi.com]
Sent:	woensdag, 13 september, 2000 09:38
To:	csp@east.isi.edu; Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to
the last call


Jan,

Your nice proposal can be incorporated in the Generic format. I hope we
managed to explain why it is useless with the ES format.

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Wednesday, September 13, 2000 5:28 AM
To:	csp@east.isi.edu
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in
response to
the last call


Colin, all,

I think it is important to keep in mind that often MPEG uses in-stream
signaling. For example, after the length byte, the adaptation field may
start with an identifier that specifies the sort of enhancement. Based on
this approach, the following would be needed :

1) Define an MPEG-4 ES payload format that allows for an adaptation field
without defining the meaning of the adaptation field.

2) Define the meaning of the adaptation field in the "generic MPEG-4
specification". For carriage of MPEG-4 ES and MPEG-4 SL streams the "generic
specification" defines the use of exactly_the_same payload type as defined
in 1). The only difference is the defined use of the adaptation field.
Whether or not an SL header is carried is signaled in the adaptation field.

3) Next to the payload types for MPEG-4 ES and SL streams, the "generic
specification" may also define payloads for other streams, such as FlexMux
streams. For these, other payload types will be needed.

Now the initial services start using the MPEG-4 ES payload format; though
the adaptation field is not used, this field is allowed to occur. Future
services may use exactly the same payload type, but with the adaptation
field as defined by the "generic specification".

It may also be important to understand that the presence of the adaptation
field is intended to be transparent from RTP point of view and for the
session in general. The adaptation field serves MPEG-4 system purposes which
are fully orthogonal to RTP and the streaming session.

Kind regards,

Jan



From rem-conf Wed Sep 13 10:54:11 2000 
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To: <csp@east.isi.edu>, <zvil@csi.com>
Cc: <rem-conf@es.net>, <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the
         last call
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Zvi,

Your first assumption is incorrect. The name does not change when inclu=
ding
the adaptation field. But rather than explaining once more by email, I =
will
try to reach you by phone. A few minutes ago I failed to reach you on b=
oth
phone numbers you gave, so I will give it another try by tomorrow, assu=
ming
that you will be in your office by than. If needed, please let me know =
how I
can reach you otherwise.

Best regards,

Jan

-----Original Message-----
From:	zvil@csi.com [mailto:zvil@csi.com]
Sent:	woensdag, 13 september, 2000 19:15
To:	csp@east.isi.edu; Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response=
 to
the last call


Hi Jan,

I have the same conclusion. Maybe I would have had another conclusion i=
f I
saw answers to the questions Collin and me and other asked. I'll try to=

summarize it for last time:

1. Applications that know only the ES format will not recognize a futur=
e
format because they will not identify the format name.
2. Therefore having extension mechanism in the ES format without fully
describing the extensions makes no sense.
3. Defining the extensions in the ES format amounts to replacing it by =
the
Generic format. Practically in time-to-market terms, that means suppres=
sing
the ES format. What is the gain, if the Generic format is going to be a=

superset of the ES format?

Regards,

z
soon the whole world will STREAM
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October=

10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Wednesday, September 13, 2000 6:51 PM
To:	csp@east.isi.edu; zvil@csi.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response=
 to
the last call


Dear Zvi,

My conclusion is that I failed to let you understand the issue. And you=

seem
to suggest that you are not the only one. Your suggestion to kill this
discussion ignores the problem addressed. I agree that putting our head=
s
into the sand is a way to go forward. But in my opinion we should take =
our
responsibility to define a future proof solution.

Best regards,

Jan

soon there will be more in this world than plain streaming ...

=



From rem-conf Wed Sep 13 12:23:09 2000 
From rem-conf-request@es.net Wed Sep 13 12:23:08 2000
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Message-ID: <01C01DD0.94631B60.zvil@csi.com>
From: Zvi Lifshitz <zvil@csi.com>
To: "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>,
        "csp@east.isi.edu" <csp@east.isi.edu>
Cc: "rem-conf@es.net" <rem-conf@es.net>,
        "4on2andIP-sys@advent.ee.columbia.edu" <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call
Date: Wed, 13 Sep 2000 22:18:43 +0200
Organization: Optibase Ltd.
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Sorry I'm not available for MPEG discussions 24 hours a day ;-) Try 
tomorrow, why not? I'm curious to hear your explanation to your first 
sentence below.. If you manage to make it through my hard head there will 
be a big apologize on the reflector ;-))

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm 
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October 
10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Wednesday, September 13, 2000 7:51 PM
To:	csp@east.isi.edu; zvil@csi.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to 
the last call


Zvi,

Your first assumption is incorrect. The name does not change when including
the adaptation field. But rather than explaining once more by email, I will
try to reach you by phone. A few minutes ago I failed to reach you on both
phone numbers you gave, so I will give it another try by tomorrow, assuming
that you will be in your office by than. If needed, please let me know how 
I
can reach you otherwise.

Best regards,

Jan

-----Original Message-----
From:	zvil@csi.com [mailto:zvil@csi.com]
Sent:	woensdag, 13 september, 2000 19:15
To:	csp@east.isi.edu; Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to
the last call


Hi Jan,

I have the same conclusion. Maybe I would have had another conclusion if I
saw answers to the questions Collin and me and other asked. I'll try to
summarize it for last time:

1. Applications that know only the ES format will not recognize a future
format because they will not identify the format name.
2. Therefore having extension mechanism in the ES format without fully
describing the extensions makes no sense.
3. Defining the extensions in the ES format amounts to replacing it by the
Generic format. Practically in time-to-market terms, that means suppressing
the ES format. What is the gain, if the Generic format is going to be a
superset of the ES format?

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Wednesday, September 13, 2000 6:51 PM
To:	csp@east.isi.edu; zvil@csi.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to
the last call


Dear Zvi,

My conclusion is that I failed to let you understand the issue. And you
seem
to suggest that you are not the only one. Your suggestion to kill this
discussion ignores the problem addressed. I agree that putting our heads
into the sand is a way to go forward. But in my opinion we should take our
responsibility to define a future proof solution.

Best regards,

Jan

soon there will be more in this world than plain streaming ...




From rem-conf Wed Sep 13 13:42:48 2000 
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To: Stephan Wenger <stewe@cs.tu-berlin.de>
cc: rem-conf@es.net
Subject: Re: In band signalling of PT 
In-Reply-To: Your message of "Tue, 12 Sep 2000 23:07:13 +0200."
             <4.3.2.7.2.20000912225736.02bf17c0@mail.cs.tu-berlin.de> 
Date: Wed, 13 Sep 2000 16:39:10 -0400
From: Colin Perkins <csp@east.isi.edu>
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--> Stephan Wenger writes:
>Folks,
>
>the recent discussion about the MPEG-4 ES payload brought up an
>issue that, IMHO, may be a shortcoming of current RTP and could
>easily be fixed.  This shortcoming is RTP's need for the out-of-band
>announcement/negotiation of parameters, including the PT.
>
>I wonder whether it may make sense to signal such information in-band
>as well, e.g. through sender reports or through special media packets.
>That would allow receivers to "tune" to RTP broadcasts with only
>IP/Port information (which could, for example, be owned by a TV station).
>
>Questions:
>
>a) Did I miss something in the current spec?  Is this already specified?
>b) Does such a scenario make sense?  Specifically, how do people think
>   to include session announcements in the media packetstream (or in
>   RTCP)?  I recall that some time ago there was some discussion
>   about architectural cleanness (it does / doesn't make sense to include
>   session info in the media stream).  What is the status of that?

Does it help? Since you need out of band information to know where to find
the media stream, you may as well include information about the format of
the media.

Colin



From rem-conf Wed Sep 13 14:30:49 2000 
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From: John Lazzaro <lazzaro@CS.Berkeley.EDU>
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[Colin Perkins writes ...]

> Does it help? Since you need out of band information to know where to find
> the media stream, you may as well include information about the format of
> the media.

I think it could really help move the conversation forward if one of
the pro-inband folks explicitly describes the out-of-band system they
have in mind for finding the initial media stream, and explain why this
out-of-band system can't be based on IETF protocols known to be able
to handle the tasks that the pro-inband folks are requesting RTP changes
to handle. There's an implicit sense of "implementing those IETF protocols
on a dumb MPEG-4 terminal is too complex/costly" -- once you make that
implicit statement explicit and supported, there's an opportunity to   
move forward with the discussion ...

-------------------------------------------------------------------------
John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
-------------------------------------------------------------------------




From rem-conf Wed Sep 13 16:42:30 2000 
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To: jan.vandermeer@philips.com
cc: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
In-Reply-To: Your message of "Wed, 13 Sep 2000 05:27:55 +0200."
             <0056890014033793000002L932*@MHS> 
Date: Wed, 13 Sep 2000 19:37:35 -0400
From: Colin Perkins <csp@east.isi.edu>
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--> jan.vandermeer@philips.com writes:
>Colin, all,
>
>I think it is important to keep in mind that often MPEG uses in-stream
>signaling. For example, after the length byte, the adaptation field may
>start with an identifier that specifies the sort of enhancement. Based on
>this approach, the following would be needed :
>
>1) Define an MPEG-4 ES payload format that allows for an adaptation field
>without defining the meaning of the adaptation field.
>
>2) Define the meaning of the adaptation field in the "generic MPEG-4
>specification". For carriage of MPEG-4 ES and MPEG-4 SL streams the "generic
>specification" defines the use of exactly_the_same payload type as defined
>in 1). The only difference is the defined use of the adaptation field.
>Whether or not an SL header is carried is signaled in the adaptation field.

I do not believe that this is possible. The differences between the ES
format and the "generic" format are too great - one assumes the use of
MPEG Systems, one requires that MPEG systems is NOT used (and that LATM
is used instead, in the case of audio). 

Simply adding an extra byte for an extension header will not solve this
fundamental semantic difference. 

Colin



From rem-conf Wed Sep 13 16:48:31 2000 
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To: Franceschini Guido <Guido.Franceschini@cselt.it>
cc: "'Narasimhan, Sam (SD-EX)'" <SNarasimhan@gi.com>,
        "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>,
        "'zvi'" <zvil@csi.com>, rem-conf@es.net,
        4on2andIP-sys@advent.ee.columbia.edu
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
In-Reply-To: Your message of "Wed, 13 Sep 2000 09:22:17 +0200."
             <85077463E51D6A498C453073E1677940034EBF@exc2k02.cselt.it> 
Date: Wed, 13 Sep 2000 19:45:24 -0400
From: Colin Perkins <csp@east.isi.edu>
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--> Franceschini Guido writes:
>Hi all,
>my own considerations on this thread
>
>In RTP and IETF procedure terms, I understand this thread on the extra byte
>as:
>1) modify the current ES draft to accomodate a payload header starting with
>a byte indicating its length. No specification of what the payload header
>carries
>2) allow the modified ES draft to enter the RFC standard track and be widely
>implemented
>3) propose a new draft for 'generic' ES or SL mapping, which is identical to
>the modified ES draft but also defines what the payload header carries
>4) the 'generic' draft becomes standard and _obsoletes_ the ES draft.
>Devices implementing the ES draft will skip the payload header information
>but still understand the media content.
>
>Well, I think we have a technical problem here, which has nothing to do with
>usual practices / procedures / timelines.
>The problem that I see is that:
>- the ES draft defines 2 Payload Formats: 1 for Audio, 1 for Video
>- the 'generic' draft will define 1 Payload Format only, and the type of
>media (Audio, Video, BIFS ...) being carried will be specified using MPEG-4
>Object Descriptors.
>
>What we achieved at the Paris meeting (before discussing this extra byte
>feature) has been to allow MPEG-4 Systems terminals implementing the
>'generic' draft to use the ES draft as well withouth any need to duplicate
>implementations (i.e.: the ES draft is a subset of the generic draft in the
>Data Plane).
>Now, the 'extra byte' proposal would allow devices implementing the ES draft
>to be compatible in the Data Plane with the 'generic' format, too.
>
>Compatibility in the Control Plane (i.e. Payload Type
>negotiation/announcement) was not considered in Paris.
>I believe that devices implementing the 'generic' draft would have no
>difficulties in understanding 1 more SDP keyword or MIME type as introduced
>by the ES draft.
>But the reverse is more difficult and tricky. As I outlined above, the
>'generic' draft would not specify by itself the type of media content it is
>carrying, and would rely on the Object Descriptor to do that. If the goal is
>to allow devices implementing the ES draft to be compatible with future
>content delivered by means of the 'generic' draft, than we need to specify
>NOW how to duplicate the information in the OD so that the media type is
>made explicit for those devices. And then we will have to accept that
>devices implementing MPEG-4 Systems get the indication of the media type
>twice :-(

And get the necessary changes to H.323 made in time for the next ITU
meeting in November, where the ES format is to be included.

Colin



From rem-conf Wed Sep 13 17:16:21 2000 
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To: Kris Huber <khuber@sorensonlabs.com>
cc: Zvi Lifshitz <zvil@csi.com>, rem-conf@es.net,
        4on2andIP-sys@advent.ee.columbia.edu,
        "'Narasimhan, Sam (SD-EX)'" <SNarasimhan@gi.com>,
        "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
In-Reply-To: Your message of "Wed, 13 Sep 2000 10:30:03 MDT."
             <6E031E06378BD311AEF20090273CE1BA798468@el-postino.s-vision.com> 
Date: Wed, 13 Sep 2000 20:13:10 -0400
From: Colin Perkins <csp@east.isi.edu>
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--> Kris Huber writes:
>Hello Zvi, Colin [last comment before letting this discussion transition to
>generic format]:
>
>I think some of the discussion has resulted from misunderstanding of the
>semantics of the payload type field apparently involving a change (by IETF?)
>from static to dynamic binding of an external "payload name" to this field.

The decision to refuse further static payload type assignments was made at
the Munich IETF meeting, in August 1997. 

>In any case, the desire has been to see whether or not this format can
>somehow be a subset of the soon-to-be more generic payload format.  Leaving
>out lack of LATM in the draft generic format, the answer appears to still be
>"no", unless the payload name for the future format is assigned now and a
>subset of its semantics, including how to read its adaptation field, are
>included in draft-ietf-avt-rtp-mpeg4-es-03.  But this would be
>unprecedented, I suspect (possible?  I don't know).  The other option is if
>another static payload type is assigned and a future payload type of a
>different name (MPEG SL format) uses the same payload type.

This is one problem, certainly. The LATM problem is also major, and seems
to be ignored in much of this discussion.

>Why, some have asked?  To avoid the redundancy that we video compression
>engineers hate so much!  How would this redundancy result?  Information
>broadcast services begin initially using the Kichuchi-san payload format,
>then have to make the decision later on between continuing to use it, and
>either doubling their network bandwidth to do the new broadcasting services
>or provide services support the new format (since the video ES data will
>tend to dominate the bandwidth).  Allowing early-adopter non-upgradeable
>devices to cease to function is another possibility, but that also has a
>high cost if the number of such devices is large.  You might wonder about an
>approach in which a new enhancement payload type is used rather than a
>single scalable payload type (e.g., generic format to add the enhancement
>information), and perhaps this is considered the "right way".  The only
>problem I see with this that the information goes in separate packets and
>therefore additional buffer memory and synchronization functionality are
>required.  Let me clarify, for point-to-point usage there is no efficiency
>problem, it is only for the broadcast (multipoint) case.

I would consider the correct solution would be to send the extra information
as an additional stream. I don't believe the overheads are significant.

>It seems that static payload typing, or early assignment of payload name +
>"Jan-byte" + other modifications would give the proposed interoperability
>functionality.  Out of curiosity let me ask (Colin?), how is "there will be
>no more static payload assignments" formally stated, with SHALL NOT or
>SHOULD NOT?  Or is it a policy existing outside of an RFC?  How many devices
>would be obsoleted by a new static payload type?

The policy was made in the AVT working group, to prevent exhaustion of the
limited payload type space. 

Colin



From rem-conf Wed Sep 13 17:29:17 2000 
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To: jan.vandermeer@philips.com
cc: 4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>,
        rem-conf <rem-conf@es.net>
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
In-Reply-To: Your message of "Wed, 13 Sep 2000 19:24:20 +0200."
             <0056890014059984000002L942*@MHS> 
Date: Wed, 13 Sep 2000 20:24:51 -0400
From: Colin Perkins <csp@east.isi.edu>
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--> jan.vandermeer@philips.com writes:
>I would like to put the problem in a little bit wider perspective. Several
>people perfectly understand the problem while some (?) others don't yet
>understand the problem, while suggesting that we are wasting our time with
>useless discussions. As Ajay stated correctly, it is the responsibility of
>IETF to decide how to proceed with the current MPEG-4 ES proposal. The only
>thing we can say right now is that several (perhaps even well respected)
>MPEG members have substantial comments on the current proposal. As a
>consequence it is likely that MPEG (members) will submit another proposal
>for carriage of MPEG-4 ES to IETF. Though this proposal is expected to have
>a high level of overlap with the current one, it is slightly different. IETF
>may wish to take this into account when coming to a decision.

We look forward to your proposal, however note that due to interactions
with the ITU (who which to use MPEG-4 ES as part of the next version of
H.323) you need to produce this as an internet draft within the next few
days or else we will be unable to consider it.

As an alternative, you may wish to propose explicit wording changes to
the Kikuchi-san proposal to bring it into line with your ideas. This 
discussion may reach closure faster if we have a more concrete design.

Colin



From rem-conf Wed Sep 13 17:39:45 2000 
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To: "Narasimhan, Sam (SD-EX)" <SNarasimhan@gi.com>
cc: "'Zvi Lifshitz'" <zvil@csi.com>,
        "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>,
        rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
In-Reply-To: Your message of "Wed, 13 Sep 2000 11:28:26 EDT."
             <973597126BDDD11197AA00805FA7EBC9034239AE@ntas0026.gi.com> 
Date: Wed, 13 Sep 2000 20:36:35 -0400
From: Colin Perkins <csp@east.isi.edu>
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--> "Narasimhan, Sam (SD-EX)" writes:
>Zvi,
>
>1.  I thought all the comments were to make changes to 
>    Kikuchi-San's draft NOW - not later. If it just added
>    the adaptation length followed by length bytes before
>    the ES payload (in the RTP payload), then content
>    authors could author content that can play on receivers
>    that implemented Kikuchi-San's draft as well as the
>    proposal that will come out of MPEG. I am assuming
>    that the MPEG proposal will use this adaptation
>    bytes to signal type of content. If so, this will
>    make Kikuchi-San's draft a sub-set of what MPEG will
>    propose (preferable was the term you used). The only way
>    this can happen is if Kikuchi-San's draft can be modified
>    NOW.

The Kikuchi-san draft is needed for inclusion in the H.323 standard, 
under revision in the ITU. This is why we are very short on time for
modifications.

>2.  If Kikuchi-San's draft can be modified, it does NOT have to
>    specify the content of adaptation header payload. MPEG will
>    do it. What all it has to say is use the 'adaptation length'
>    field to offset where you pick up the ES payload.

This assumes that the ES payload is either LATM audio or video in the
Combined Configuration/Elementary stream mode. Can you really produce
a generic format which always includes ES data in that format? If not,
you will break the old receivers.

>3.  In MPEG-2, the byte that follows the adaptation length 
>    signals what is in the rest of the adaptation header such
>    as PCR, OPCR, splicing etc. Those that cannot process the
>    adaptation header just use the adaptation length to offset to
>    the next payload start (such as PES header or ES). This
>    is what I am assuming MPEG will do for 'generic' mapping. The
>    byte after adaptation header may signal presence of DTS,
>    SL header, flexmux header, scalable data, buffer etc.
>4.  If Kikuchi-San's draft progresses 'as is' and MPEG uses the
>    adaptation header, then content authors that used the MPEG
>    coding can only play it on receivers that implemented the
>    re-assembly of MPEG 'generic' mapping and not on those that
>    'ONLY' implemented Kikuchi-San's draft (if it was not modified).
>    CE vendors can choose to implement receivers that can support
>    multiple payload formats. However, if Kikuchi-San's draft can
>    be modified NOW, we may see a single payload format for MPEG-4
>    and I believe that is a plus and thus NECESSARY.

Colin



From rem-conf Wed Sep 13 17:42:34 2000 
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From: Scott Bradner <sob@harvard.edu>
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To: csp@east.isi.edu, jan.vandermeer@philips.com
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call
Cc: 4on2andIP-sys@advent.ee.columbia.edu, rem-conf@es.net
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> As an alternative, you may wish to propose explicit wording changes to
> the Kikuchi-san proposal to bring it into line with your ideas. This
> discussion may reach closure faster if we have a more concrete design.

I would strongly suggest that this is the better path to take

Scott



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Posted-Date: Thu, 14 Sep 2000 04:31:32 +0100 (WET DST)
From: Jerome Proffit <jeromeproffit@caramail.com>
To: rem-conf@es.net
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Subject: Questions about RTP payload for DTMF digit (RFC 2833) and for Redundant Audio data (RFC 2198)
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Hi everybody,

I have some questions about RTP payload for DTMF digit (RFC 
2833) and for Redundant Audio data (RFC 2198).

1) If there is only one event, does one have to send a 
block header.

2) When one detects a new event, one sends the absolute 
timestamp of the begining of the event in the RTP header. 
But does one have to increment the Timestamp field in the 
RTP header for the following packets concerning the same 
event? or does one still send the same Timestamp (begining 
of the event)?

3) if there is a big gap between two events, does one need 
to send a RTP packet? if yes what is the format of it?

Thanks a lot,

Jerome Proffit
______________________________________________________
Bo=EEte aux lettres - Caramail - http://www.caramail.com


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From rem-conf Wed Sep 13 22:56:00 2000 
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Date: Wed, 13 Sep 2000 22:40:26 -0700
To: rem-conf@es.net
From: Ross Finlayson <finlayson@live.com>
Subject: Re: In band signalling of PT 
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Another point to keep in mind here is that - in the general case - a 
multimedia session can consist of *more than one* RTP stream (e.g., one RTP 
stream for video; another for audio; another for subtitles).  Even if each 
individual stream were - somehow - to have in-band information about its 
payload format, etc., you'd still need a way of describing which individual 
RTP streams make up an entire session.

Although the MPEG world has often assumed that multiple media are 
multiplexed together in a single stream, RTP has more flexibility than this.

         Ross.




From rem-conf Thu Sep 14 00:54:57 2000 
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From: Zvi Lifshitz <zvil@csi.com>
To: "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>,
        "csp@east.isi.edu" <csp@east.isi.edu>
Cc: "rem-conf@es.net" <rem-conf@es.net>,
        "4on2andIP-sys@advent.ee.columbia.edu" <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call
Date: Thu, 14 Sep 2000 10:48:41 +0200
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[jan.vandermeer@philips.com:]

2) Define the meaning of the adaptation field in the "generic MPEG-4
specification". For carriage of MPEG-4 ES and MPEG-4 SL streams the 
"generic
specification" defines the use of exactly_the_same payload type as defined
in 1). The only difference is the defined use of the adaptation field.
Whether or not an SL header is carried is signaled in the adaptation field.

[Reply:]

If I understand you right, you seem to suggest what I asked you about few 
emails ago. To define one payload format, with an empty extension, and 
later to define the content of the extension. Same payload format, same 
name. Well, if this is your meaning at least I can understand what you want 
to do technically. Theoretically this can work, as it solves the problem of 
old devices having to recognize new name.

[Over night I thought about another scenario which solves the 
old-device-new-name issue, and planned to ask you (and Sam) if this was 
your scenario. This is when streams are broadcast but session description 
is interactively negotiated point-to-point. Then, assuming a stream is 
broadcast in the "new" format and a client starts negotiation to locate 
this stream, the server asks the client what formats it recognizes. If the 
client says the new format is OK, the server tells it the truth that this 
is in the new format. But if the client says it knows only the "old" 
format, the server bluffs and tells this is the old format].

Still, I don't believe this can work. I think also Colin said that. First, 
I believe this is against the IETF practice. Then, the Generic format 
differs from the ES format in much more than just the extension content. 
The ES format uses LATM. The Generic format is not only for audio/video 
elementary streams. The ES format defines fragmentation rules, which in the 
Generic format are configured by the SLConfig. The evolution that one needs 
to reach the other is Darwinian and might take 2 billion years... The main 
question is, assuming old devices can peacefully skip the extension field, 
how do we guarantee that they correctly handle the content beyond?

And, again, I don't see why we need it. If a stream that is delivered with 
the ES format can be played and synchronized, why would you ever want to 
put something in its adaptation field? What kind of extra SL info do you 
think can enhance it? The only thing I can think of is combining it with 
other streams that require the Generic format, but this is possible without 
touching the stream.

----
at this point I'm getting a phone call from Jan :-)............
---

(20 minutes later).. OK, we clarified between us few more things. At least 
we better understand now the disagreement... I'll send my message is as and 
let Jan continue...

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm 
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October 
10th-12th, 2000






From rem-conf Thu Sep 14 01:50:35 2000 
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Date: Thu, 14 Sep 2000 01:40:08 -0700
To: Andy Valencia <vandys@zendo.com>, "W. Mark Townsley" <townsley@cisco.com>
From: Tmima Koren <tmima@cisco.com>
Subject: Re: 0 byte L2TPHC header and implied PPPMUX
Cc: brucet@cisco.com, dwing@cisco.com,
        Stephen Casner <casner@packetdesign.com>, gurdeep@microsoft.com,
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Hi
Since there was no feedback from the lists about this issue, can we update the draft with the suggested changes?
(3 flags in byte 9 of L2TPHC AVP)
Thanks,
Tmima


At 06:03 PM 9/7/00 -0700, Tmima Koren wrote:

>I suggest a few minor changes to the L2TPHC draft draft-ietf-l2tpext-l2tphc-02.txt
>to improve bandwidth utilization in special cases.
>
>
>When the L2TPHC header is 1 byte long, actually only 1 bit is significant: the P (Priority) bit.
>
>If all the packets in the tunnel will have the same priority, the L2TPHC byte is redundant.
>
>We could add to the L2TPHC AVP one more byte (byte 9)
>We really need only 2 flag bits:
>
>flag 1:  value 1 means: L2TPHC byte will be omitted
>             value 0 means: L2TPHC byte is there, get P (Priority) from the L2TPHC byte
>
>flag 2:  value 1 means all packets have implied P=1
>             value 0 means all packets have implied P=0
>
>consult flag 2 only when flag 1 is 1
>
>
>If the packet in the L2TPHC tunnel is always PPPMUX, there is another useful flag:
>
>flag 3: value 1 means 'PPPMUX only'
>            value 0 means: any PPP packet
>
>
>When flags 1 and 3 are on (=1), both L2TPHC header and PPPMUX protocol id can be omitted 
>PPP packets that have a different priority or that are not PPPMUX can still be sent in the tunnel using the UDP/L2TP format, same as the L2TP control packets
>
>TCRTP (Tunneled Compressed multiplexed RTP) can benefit from these changes: the tunneling overhead will be reduced by 2 bytes.
>
>Tmima
>




From rem-conf Thu Sep 14 02:04:11 2000 
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To: "CURET Dominique FTRD/DMI/REN" <dominique.curet@rd.francetelecom.fr>,
        <4on2andIP-sys@advent.ee.columbia.edu>, <rem-conf@es.net>
Subject: RE: Pre-MPEG meeting for the '4onIP' AHG
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Dear D., dear all,

for me both days are o.k..

Paul Christ

> -----Original Message-----
> From: CURET Dominique FTRD/DMI/REN
> [mailto:dominique.curet@rd.francetelecom.fr]
> Sent: Mittwoch, 13. September 2000 18:01
> To: 4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
> Subject: Pre-MPEG meeting for the '4onIP' AHG
>
>
> Dear all, dear Young-Kwon,
>
> we are supposed to meet on Saturday(21/10) and Sunday (22/10)
> before the MPEG meeting in La Baule.
>
> Can we say something like:
>      Saturday from 14 to 2000
>      Sunday from 0900 to 1800
> Or only the Sunday from 0900 to 1800
>
> comments
>
> regards
> _________________________________________
> > Dominique Curet
> > France T=E9l=E9com R&D - DMI/PIA
> > % + 33 (0)2 99 12 40 05   Fax : + 33 (0)2 99 12 40 98
> > )  CCETT       B.P. 59            4, rue du Clos Courtel
> > 35512           Cesson-Sevigne   Cedex   09     FRANCE
> > mailto:dominique.curet@rd.francetelecom.fr
> _________________________________________
>
>
>
>




From rem-conf Thu Sep 14 02:34:41 2000 
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CC: 
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	, 4on2andIP-sys@advent.ee.columbia.edu, rem-conf@es.net
Subject: Re: Pre-MPEG meeting for the '4onIP' AHG
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Dominique,

I will attend as well

Pierre

Frederic Pabot a =E9crit :

> Dear Dominique, all,
>
> We would also be happy to participate to this 4onIP AHG meeting.
> Please let me know how to procede.
>
> Best regards,
> F. Pabot
>
> CURET Dominique FTRD/DMI/REN a =E9crit :
> >
> > Dear all, dear Young-Kwon,
> >
> > we are supposed to meet on Saturday(21/10) and Sunday (22/10)
> > before the MPEG meeting in La Baule.
> >
> > Can we say something like:
> >      Saturday from 14 to 2000
> >      Sunday from 0900 to 1800
> > Or only the Sunday from 0900 to 1800
> >
> > comments
> >
> > regards
> > _________________________________________
> > > Dominique Curet
> > > France T=E9l=E9com R&D - DMI/PIA
> > > % + 33 (0)2 99 12 40 05   Fax : + 33 (0)2 99 12 40 98
> > > )  CCETT       B.P. 59            4, rue du Clos Courtel
> > > 35512           Cesson-Sevigne   Cedex   09     FRANCE
> > > mailto:dominique.curet@rd.francetelecom.fr
> > _________________________________________
>
> --
> Frederic PABOT
> SYCOMORE A=E9rospatiale Matra, Division Multimedia, FRANCE
> ENS-Lyon L.H.P.C.           voice : +33 4 72 72 82 30
> 46 allee d'Italie           fax   : +33 4 72 72 04 67
> F-69364  Lyon Cedex 07      mailto:fpabot@matra-ms2i.fr




From rem-conf Thu Sep 14 03:35:01 2000 
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Subject: Kikuchi-san proposal as subset of generic MPEG-4 formats
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Dear all,

As promised in a previous email, I would like to review the problems
associated with making the proposal of Kikuchi-san a subset of the set =
of
generic formats that hopefully will be produced in La Baule. My assumpt=
ion
is that there will be "generic formats" for several MPEG-4 streams, suc=
h as:
- MPEG-4 audio elementary streams (allowing for carriage of the remaind=
er of
an SL header)
- MPEG-4 video elementary streams (allowing for carriage of the remaind=
er of
an SL header)
- MPEG-4 system streams
For now, I would like to constrain the discussion to the first two, as =
these
two are also addressed in the proposal of Kikuchi-san. In the following=

review I make a separation between video specific issues and audio spec=
ific
issues. The general issue of the additional bytes is discussed extensiv=
ely
and I don't think there is a need to recap that here. The only clarific=
ation
that may be useful in this context is that the additional bytes can be =
used
to carry the remainder of an SL header in cases where needed, thereby
providing the same functionality as in the proposal of Civanlar et al.

Video specific issues

My proposal would be to use for the generic MPEG-4 video ES format exac=
tly
the same payload type and names as Kikuchi-san. I also suggest to use t=
he
same fragmentation rules. So from my perspective no need to change anyt=
hing
specific to that.

Audio specific issues

Also for the generic MPEG-4 audio ES format I suggest to use the same
payload type, names and fragmentation rules as Kikuchi-san. With audio =
we
have the "LATM problem". However, with some pragmatism we may be able t=
o
solve this. In Paris we agreed to allow for more than one access unit w=
ithin
one SL packet. We could slightly modify this by allowing LATM as multip=
lex
structure within SL. (Note that the FlexMux provides a multiplex struct=
ure
on top of SL.) To indicate usage of LATM within an SL packet a currentl=
y
reserved bit in the SL header could be assigned. If we agree on his, th=
en
the important question is whether we need also define carriage of MPEG-=
4
audio ES over IP without LATM. Perhaps not. LATM adds only little overh=
ead
in case of a single large AU and for a small AU it is likely that LATM =
is
applied anyhow. A final remark on this issue: LATM is also used for car=
riage
of MPEG-4 audio ES over MPEG-2, pretty much consistent with usage of LA=
TM
within SL and for transport over IP.

Along these lines I will propose specific wording changes to the propos=
al of
Kikuchi-san. Please comment if you disagree with the approach described=
. I
hope that my proposal for usage of LATM is not too big of a shock. In m=
y
view it is an acceptable price to pay for compatibility with the propos=
al
>from Kikuchi-san. And I believe such compatibility is important for the=

success of MPEG-4 on the market place.

Best regards,

Jan


=



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From: Zvi Lifshitz <zvil@csi.com>
To: "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>,
        "4on2andIP-sys@advent.ee.columbia.edu" <4on2andIP-sys@advent.ee.columbia.edu>
Cc: "rem-conf@es.net" <rem-conf@es.net>
Subject: RE: Kikuchi-san proposal as subset of generic MPEG-4 formats
Date: Thu, 14 Sep 2000 15:11:58 +0200
Organization: Optibase Ltd.
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Dear Jan, all,

Even after we talked I wasn't convinced to agree to your approach. The main 
thing is that you are talking about upgrading the Kikuchi-san proposal, 
while I see no reason to do it. Specifically, I cannot think if any in-band 
info I would like to add in the future to a stream that is delivered by the 
ES format. Theoretically crazy ideas can always come up, but practically 
the chances for this are not more than for any content carried over RTP. 
And other payload formats didn't care about such extensions. It seems that 
all the fancy enhancements that you, and Sam, and actually also me and most 
people in MPEG want to have will be done through addition of extra streams 
on top of the audio/video streams for which the Kikuchi-san proposal works 
well.

Therefore I see no reason to change the Kikuchi-san proposal, as long as 
future applications will be able to smoothly integrate streams using this 
format and the generic format, which is indeed seems the situation.

Some other comments regarding your proposal:

1. The Generic payload format is for every MPEG-4 stream, including audio. 
You seem to restrict the use of this format by requiring that whenever 
audio is transmitted, the format cannot be used natively and LATM must be 
used. I don't see a point and this restriction. LATM is for systems that 
don't use MPEG-4 Systems. If an MPEG-4 server doesn't care about 
compatibility with other systems why should we force it? Same argument for 
video fragmentation.

2. Just like we made the ES proposal for video a subset of the Generic 
format from MPEG-4/SL point of view, we can do that for audio, by allowing 
(NOT requiring) LATM as an MPEG-4 stream. You want do indicate that in a 
bit in the SL header, which I find no reason for. We need this indication 
just once per stream. It can be in the SLConfig, but since LATM is only for 
audio, it's a kind of format that should be indicated in the 
DecoderConfigDescriptor. Maybe in objectTypeIndication.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm 
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October 
10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Thursday, September 14, 2000 12:31 PM
To:	4on2andIP-sys@advent.ee.columbia.edu
Cc:	rem-conf@es.net
Subject:	Kikuchi-san proposal as subset of generic MPEG-4 formats


Dear all,

As promised in a previous email, I would like to review the problems
associated with making the proposal of Kikuchi-san a subset of the set of
generic formats that hopefully will be produced in La Baule. My assumption
is that there will be "generic formats" for several MPEG-4 streams, such 
as:
- MPEG-4 audio elementary streams (allowing for carriage of the remainder 
of
an SL header)
- MPEG-4 video elementary streams (allowing for carriage of the remainder 
of
an SL header)
- MPEG-4 system streams
For now, I would like to constrain the discussion to the first two, as 
these
two are also addressed in the proposal of Kikuchi-san. In the following
review I make a separation between video specific issues and audio specific
issues. The general issue of the additional bytes is discussed extensively
and I don't think there is a need to recap that here. The only 
clarification
that may be useful in this context is that the additional bytes can be used
to carry the remainder of an SL header in cases where needed, thereby
providing the same functionality as in the proposal of Civanlar et al.

Video specific issues

My proposal would be to use for the generic MPEG-4 video ES format exactly
the same payload type and names as Kikuchi-san. I also suggest to use the
same fragmentation rules. So from my perspective no need to change anything
specific to that.

Audio specific issues

Also for the generic MPEG-4 audio ES format I suggest to use the same
payload type, names and fragmentation rules as Kikuchi-san. With audio we
have the "LATM problem". However, with some pragmatism we may be able to
solve this. In Paris we agreed to allow for more than one access unit 
within
one SL packet. We could slightly modify this by allowing LATM as multiplex
structure within SL. (Note that the FlexMux provides a multiplex structure
on top of SL.) To indicate usage of LATM within an SL packet a currently
reserved bit in the SL header could be assigned. If we agree on his, then
the important question is whether we need also define carriage of MPEG-4
audio ES over IP without LATM. Perhaps not. LATM adds only little overhead
in case of a single large AU and for a small AU it is likely that LATM is
applied anyhow. A final remark on this issue: LATM is also used for 
carriage
of MPEG-4 audio ES over MPEG-2, pretty much consistent with usage of LATM
within SL and for transport over IP.

Along these lines I will propose specific wording changes to the proposal 
of
Kikuchi-san. Please comment if you disagree with the approach described. I
hope that my proposal for usage of LATM is not too big of a shock. In my
view it is an acceptable price to pay for compatibility with the proposal
>from Kikuchi-san. And I believe such compatibility is important for the
success of MPEG-4 on the market place.

Best regards,

Jan




From rem-conf Thu Sep 14 05:27:08 2000 
From rem-conf-request@es.net Thu Sep 14 05:27:07 2000
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Date: Thu, 14 Sep 2000 14:24:55 +0200
From: Franceschini Guido <Guido.Franceschini@CSELT.IT>
Subject: RE: Kikuchi-san proposal as subset of generic MPEG-4 formats
To: 4on2andIP-sys@advent.ee.columbia.edu,
 "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>
Cc: rem-conf@es.net
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Dear Jan,

I am not really happy to signal LATM usage (which is an audio specific tool)
at the SL level (which is generic). I would prefer to expand the audio
specific configuration parameters.

Also, I think that while compatibility at the level of RTP packets is
reachable (e.g. by means of the extra byte), the real technical difficulty
resides in the control plane. Once somebody finds a solution for the control
plane issues, then maybe we will find more than one possible approach for
the data plane.
The usage of one extra byte (thus a payload header) in all RTP packets is a
possible technical solution, but also the usage of an indicator at config
time telling whether such extra byte (thus a payload header) is present or
not in the RTP packets could be considered (this way you would not force the
extra byte all the time). And maybe there are more solutions.
To conclude: I would not assert that the ONLY way to fulfill your
requirement is to force one extra byte all the time. I understand and agree
with the requirement, but personally don't like the proposed solution. I
would appreciate some more investigation in the control plane aspects.

Also, let's remember that the SDP and the MPEG-4 Object Descriptors serve
two very similar purposes. In IETF current practice, the 'negotiation' is
mostly done using SDP, while when using MPEG-4 Systems it is done through
ODs.
A solution aiming at serving with the same bits of information devices
compliant to the IETF current practice as well as devices compliant to
MPEG-4 Systems MUST consider the overlap between SDP and ODs.

IMHO, we can start this work, but we cannot rush it.
And this work would involve the MMUSIC group.

Regards
Guido



> ----------
> From: 	jan.vandermeer@philips.com[SMTP:jan.vandermeer@philips.com]
> Sent: 	Thursday, September 14, 2000 12:30 PM
> To: 	4on2andIP-sys@advent.ee.columbia.edu
> Cc: 	rem-conf@es.net
> Subject: 	Kikuchi-san proposal as subset of generic MPEG-4 formats
> 
> Dear all,
> 
> As promised in a previous email, I would like to review the problems
> associated with making the proposal of Kikuchi-san a subset of the set of
> generic formats that hopefully will be produced in La Baule. My assumption
> is that there will be "generic formats" for several MPEG-4 streams, such
> as:
> - MPEG-4 audio elementary streams (allowing for carriage of the remainder
> of
> an SL header)
> - MPEG-4 video elementary streams (allowing for carriage of the remainder
> of
> an SL header)
> - MPEG-4 system streams
> For now, I would like to constrain the discussion to the first two, as
> these
> two are also addressed in the proposal of Kikuchi-san. In the following
> review I make a separation between video specific issues and audio
> specific
> issues. The general issue of the additional bytes is discussed extensively
> and I don't think there is a need to recap that here. The only
> clarification
> that may be useful in this context is that the additional bytes can be
> used
> to carry the remainder of an SL header in cases where needed, thereby
> providing the same functionality as in the proposal of Civanlar et al.
> 
> Video specific issues
> 
> My proposal would be to use for the generic MPEG-4 video ES format exactly
> the same payload type and names as Kikuchi-san. I also suggest to use the
> same fragmentation rules. So from my perspective no need to change
> anything
> specific to that.
> 
> Audio specific issues
> 
> Also for the generic MPEG-4 audio ES format I suggest to use the same
> payload type, names and fragmentation rules as Kikuchi-san. With audio we
> have the "LATM problem". However, with some pragmatism we may be able to
> solve this. In Paris we agreed to allow for more than one access unit
> within
> one SL packet. We could slightly modify this by allowing LATM as multiplex
> structure within SL. (Note that the FlexMux provides a multiplex structure
> on top of SL.) To indicate usage of LATM within an SL packet a currently
> reserved bit in the SL header could be assigned. If we agree on his, then
> the important question is whether we need also define carriage of MPEG-4
> audio ES over IP without LATM. Perhaps not. LATM adds only little overhead
> in case of a single large AU and for a small AU it is likely that LATM is
> applied anyhow. A final remark on this issue: LATM is also used for
> carriage
> of MPEG-4 audio ES over MPEG-2, pretty much consistent with usage of LATM
> within SL and for transport over IP.
> 
> Along these lines I will propose specific wording changes to the proposal
> of
> Kikuchi-san. Please comment if you disagree with the approach described. I
> hope that my proposal for usage of LATM is not too big of a shock. In my
> view it is an acceptable price to pay for compatibility with the proposal
> from Kikuchi-san. And I believe such compatibility is important for the
> success of MPEG-4 on the market place.
> 
> Best regards,
> 
> Jan
> 
> 



From rem-conf Thu Sep 14 05:43:14 2000 
From rem-conf-request@es.net Thu Sep 14 05:43:13 2000
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Cc: <rem-conf@es.net>
Subject: Kikuchi-san proposal as subset of generic MPEG-4 formats
Message-ID: <0056890014073139000002L992*@MHS>
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Dear all,

As promised in a previous email, I would like to review the problems
associated with making the proposal of Kikuchi-san a subset of the set =
of
generic formats that hopefully will be produced in La Baule. My assumpt=
ion
is that there will be "generic formats" for several MPEG-4 streams, suc=
h as:
- MPEG-4 audio elementary streams (allowing for carriage of the remaind=
er of
an SL header)
- MPEG-4 video elementary streams (allowing for carriage of the remaind=
er of
an SL header)
- MPEG-4 system streams
For now, I would like to constrain the discussion to the first two, as =
these
two are also addressed in the proposal of Kikuchi-san. In the following=

review I make a separation between video specific issues and audio spec=
ific
issues. The general issue of the additional bytes is discussed extensiv=
ely
and I don't think there is a need to recap that here. The only clarific=
ation
that may be useful in this context is that the additional bytes can be =
used
to carry the remainder of an SL header in cases where needed, thereby
providing the same functionality as in the proposal of Civanlar et al.

Video specific issues

My proposal would be to use for the generic MPEG-4 video ES format exac=
tly
the same payload type and names as Kikuchi-san. I also suggest to use t=
he
same fragmentation rules. So from my perspective no need to change anyt=
hing
specific to that.

Audio specific issues

Also for the generic MPEG-4 audio ES format I suggest to use the same
payload type, names and fragmentation rules as Kikuchi-san. With audio =
we
have the "LATM problem". However, with some pragmatism we may be able t=
o
solve this. In Paris we agreed to allow for more than one access unit w=
ithin
one SL packet. We could slightly modify this by allowing LATM as multip=
lex
structure within SL. (Note that the FlexMux provides a multiplex struct=
ure
on top of SL.) To indicate usage of LATM within an SL packet a currentl=
y
reserved bit in the SL header could be assigned. If we agree on his, th=
en
the important question is whether we need also define carriage of MPEG-=
4
audio ES over IP without LATM. Perhaps not. LATM adds only little overh=
ead
in case of a single large AU and for a small AU it is likely that LATM =
is
applied anyhow. A final remark on this issue: LATM is also used for car=
riage
of MPEG-4 audio ES over MPEG-2, pretty much consistent with usage of LA=
TM
within SL and for transport over IP.

Along these lines I will propose specific wording changes to the propos=
al of
Kikuchi-san. Please comment if you disagree with the approach described=
. I
hope that my proposal for usage of LATM is not too big of a shock. In m=
y
view it is an acceptable price to pay for compatibility with the propos=
al
>from Kikuchi-san. And I believe such compatibility is important for the=

success of MPEG-4 on the market place.

Best regards,

Jan


=



From rem-conf Thu Sep 14 05:43:14 2000 
From rem-conf-request@es.net Thu Sep 14 05:43:13 2000
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Subject: Kikuchi-san proposal as subset of generic MPEG-4 formats
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Dear all,

As promised in a previous email, I would like to review the problems
associated with making the proposal of Kikuchi-san a subset of the set =
of
generic formats that hopefully will be produced in La Baule. My assumpt=
ion
is that there will be "generic formats" for several MPEG-4 streams, suc=
h as:
- MPEG-4 audio elementary streams (allowing for carriage of the remaind=
er of
an SL header)
- MPEG-4 video elementary streams (allowing for carriage of the remaind=
er of
an SL header)
- MPEG-4 system streams
For now, I would like to constrain the discussion to the first two, as =
these
two are also addressed in the proposal of Kikuchi-san. In the following=

review I make a separation between video specific issues and audio spec=
ific
issues. The general issue of the additional bytes is discussed extensiv=
ely
and I don't think there is a need to recap that here. The only clarific=
ation
that may be useful in this context is that the additional bytes can be =
used
to carry the remainder of an SL header in cases where needed, thereby
providing the same functionality as in the proposal of Civanlar et al.

Video specific issues

My proposal would be to use for the generic MPEG-4 video ES format exac=
tly
the same payload type and names as Kikuchi-san. I also suggest to use t=
he
same fragmentation rules. So from my perspective no need to change anyt=
hing
specific to that.

Audio specific issues

Also for the generic MPEG-4 audio ES format I suggest to use the same
payload type, names and fragmentation rules as Kikuchi-san. With audio =
we
have the "LATM problem". However, with some pragmatism we may be able t=
o
solve this. In Paris we agreed to allow for more than one access unit w=
ithin
one SL packet. We could slightly modify this by allowing LATM as multip=
lex
structure within SL. (Note that the FlexMux provides a multiplex struct=
ure
on top of SL.) To indicate usage of LATM within an SL packet a currentl=
y
reserved bit in the SL header could be assigned. If we agree on his, th=
en
the important question is whether we need also define carriage of MPEG-=
4
audio ES over IP without LATM. Perhaps not. LATM adds only little overh=
ead
in case of a single large AU and for a small AU it is likely that LATM =
is
applied anyhow. A final remark on this issue: LATM is also used for car=
riage
of MPEG-4 audio ES over MPEG-2, pretty much consistent with usage of LA=
TM
within SL and for transport over IP.

Along these lines I will propose specific wording changes to the propos=
al of
Kikuchi-san. Please comment if you disagree with the approach described=
. I
hope that my proposal for usage of LATM is not too big of a shock. In m=
y
view it is an acceptable price to pay for compatibility with the propos=
al
>from Kikuchi-san. And I believe such compatibility is important for the=

success of MPEG-4 on the market place.

Best regards,

Jan


=



From rem-conf Thu Sep 14 05:52:03 2000 
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Colin, all,

I am happy to take the challenge. My objective will be to produce a mod=
ified
Kikuchi proposal that can be agreed upon as a subset of the generic for=
mat.
I will propose explicit wording changes, but first I would like to revi=
ew
the problems to be solved. I will come back on both in separate emails
today.

Regards,

Jan

-----Original Message-----
From:	csp@east.isi.edu [mailto:csp@east.isi.edu]
Sent:	donderdag, 14 september, 2000 02:27
To:	Jan vanderMeer
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response=
 to
the last call


--> jan.vandermeer@philips.com writes:
>I would like to put the problem in a little bit wider perspective. Sev=
eral
>people perfectly understand the problem while some (?) others don't ye=
t
>understand the problem, while suggesting that we are wasting our time =
with
>useless discussions. As Ajay stated correctly, it is the responsibilit=
y of
>IETF to decide how to proceed with the current MPEG-4 ES proposal. The=
 only
>thing we can say right now is that several (perhaps even well respecte=
d)
>MPEG members have substantial comments on the current proposal. As a
>consequence it is likely that MPEG (members) will submit another propo=
sal
>for carriage of MPEG-4 ES to IETF. Though this proposal is expected to=
 have
>a high level of overlap with the current one, it is slightly different=
.
IETF
>may wish to take this into account when coming to a decision.

We look forward to your proposal, however note that due to interactions=

with the ITU (who which to use MPEG-4 ES as part of the next version of=

H.323) you need to produce this as an internet draft within the next fe=
w
days or else we will be unable to consider it.

As an alternative, you may wish to propose explicit wording changes to
the Kikuchi-san proposal to bring it into line with your ideas. This
discussion may reach closure faster if we have a more concrete design.

Colin

=



From rem-conf Thu Sep 14 07:04:18 2000 
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Cc: <rem-conf@es.net>, <4on2andIP-sys@advent.ee.columbia.edu>
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Hello Zvi,

I am glad we came to technical understanding. In your email you ask man=
y
questions, some of which I think are answered in the email I sent meanw=
hile
on Kikuchi-san's proposal as a subset of the generic format. Is that ri=
ght ?
Perhaps we could continue our discussion within the "generic context".

Kind regards,

Jan

-----Original Message-----
From:	zvil@csi.com [mailto:zvil@csi.com]
Sent:	donderdag, 14 september, 2000 09:52
To:	csp@east.isi.edu; Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response=
 to
the last call


[jan.vandermeer@philips.com:]

2) Define the meaning of the adaptation field in the "generic MPEG-4
specification". For carriage of MPEG-4 ES and MPEG-4 SL streams the
"generic
specification" defines the use of exactly_the_same payload type as defi=
ned
in 1). The only difference is the defined use of the adaptation field.
Whether or not an SL header is carried is signaled in the adaptation fi=
eld.

[Reply:]

If I understand you right, you seem to suggest what I asked you about f=
ew
emails ago. To define one payload format, with an empty extension, and
later to define the content of the extension. Same payload format, same=

name. Well, if this is your meaning at least I can understand what you =
want
to do technically. Theoretically this can work, as it solves the proble=
m of
old devices having to recognize new name.

[Over night I thought about another scenario which solves the
old-device-new-name issue, and planned to ask you (and Sam) if this was=

your scenario. This is when streams are broadcast but session descripti=
on
is interactively negotiated point-to-point. Then, assuming a stream is
broadcast in the "new" format and a client starts negotiation to locate=

this stream, the server asks the client what formats it recognizes. If =
the
client says the new format is OK, the server tells it the truth that th=
is
is in the new format. But if the client says it knows only the "old"
format, the server bluffs and tells this is the old format].

Still, I don't believe this can work. I think also Colin said that. Fir=
st,
I believe this is against the IETF practice. Then, the Generic format
differs from the ES format in much more than just the extension content=
.
The ES format uses LATM. The Generic format is not only for audio/video=

elementary streams. The ES format defines fragmentation rules, which in=
 the
Generic format are configured by the SLConfig. The evolution that one n=
eeds
to reach the other is Darwinian and might take 2 billion years... The m=
ain
question is, assuming old devices can peacefully skip the extension fie=
ld,
how do we guarantee that they correctly handle the content beyond?

And, again, I don't see why we need it. If a stream that is delivered w=
ith
the ES format can be played and synchronized, why would you ever want t=
o
put something in its adaptation field? What kind of extra SL info do yo=
u
think can enhance it? The only thing I can think of is combining it wit=
h
other streams that require the Generic format, but this is possible wit=
hout
touching the stream.

----
at this point I'm getting a phone call from Jan :-)............
---

(20 minutes later).. OK, we clarified between us few more things. At le=
ast
we better understand now the disagreement... I'll send my message is as=
 and
let Jan continue...

Regards,

z
soon the whole world will STREAM
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October=

10th-12th, 2000


=



From rem-conf Thu Sep 14 07:44:02 2000 
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Cc: <rem-conf@es.net>
Subject: RE: Kikuchi-san proposal as subset of generic MPEG-4 formats
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Zvil,

I fully agree that signaling at OD level is a better solution. With res=
pect
to migration towards other solutions, I think that it is better to have=
 one
simple and straightforward solution for carriage of MPEG-4 A/V ES over =
IP
then the situation in which "you have many standards to choose from".
Specifically in case of thin clients.

With respect to audio, the reason of my suggestion to always use LATM i=
s to
achieve a single solution for audio. I believe that this is often usefu=
l,
while the pain in terms of overhead and processing is very acceptable.

Regards,

Jan

-----Original Message-----
From:	zvil@csi.com [mailto:zvil@csi.com]
Sent:	donderdag, 14 september, 2000 14:15
To:	4on2andIP-sys@advent.ee.columbia.edu; Jan vanderMeer
Cc:	rem-conf@es.net
Subject:	RE: Kikuchi-san proposal as subset of generic MPEG-4 formats


Dear Jan, all,

Even after we talked I wasn't convinced to agree to your approach. The =
main
thing is that you are talking about upgrading the Kikuchi-san proposal,=

while I see no reason to do it. Specifically, I cannot think if any in-=
band
info I would like to add in the future to a stream that is delivered by=
 the
ES format. Theoretically crazy ideas can always come up, but practicall=
y
the chances for this are not more than for any content carried over RTP=
.
And other payload formats didn't care about such extensions. It seems t=
hat
all the fancy enhancements that you, and Sam, and actually also me and =
most
people in MPEG want to have will be done through addition of extra stre=
ams
on top of the audio/video streams for which the Kikuchi-san proposal wo=
rks
well.

Therefore I see no reason to change the Kikuchi-san proposal, as long a=
s
future applications will be able to smoothly integrate streams using th=
is
format and the generic format, which is indeed seems the situation.

Some other comments regarding your proposal:

1. The Generic payload format is for every MPEG-4 stream, including aud=
io.
You seem to restrict the use of this format by requiring that whenever
audio is transmitted, the format cannot be used natively and LATM must =
be
used. I don't see a point and this restriction. LATM is for systems tha=
t
don't use MPEG-4 Systems. If an MPEG-4 server doesn't care about
compatibility with other systems why should we force it? Same argument =
for
video fragmentation.

2. Just like we made the ES proposal for video a subset of the Generic
format from MPEG-4/SL point of view, we can do that for audio, by allow=
ing
(NOT requiring) LATM as an MPEG-4 stream. You want do indicate that in =
a
bit in the SL header, which I find no reason for. We need this indicati=
on
just once per stream. It can be in the SLConfig, but since LATM is only=
 for
audio, it's a kind of format that should be indicated in the
DecoderConfigDescriptor. Maybe in objectTypeIndication.

Regards,

z
soon the whole world will STREAM
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October=

10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Thursday, September 14, 2000 12:31 PM
To:	4on2andIP-sys@advent.ee.columbia.edu
Cc:	rem-conf@es.net
Subject:	Kikuchi-san proposal as subset of generic MPEG-4 formats


Dear all,

As promised in a previous email, I would like to review the problems
associated with making the proposal of Kikuchi-san a subset of the set =
of
generic formats that hopefully will be produced in La Baule. My assumpt=
ion
is that there will be "generic formats" for several MPEG-4 streams, suc=
h
as:
- MPEG-4 audio elementary streams (allowing for carriage of the remaind=
er
of
an SL header)
- MPEG-4 video elementary streams (allowing for carriage of the remaind=
er
of
an SL header)
- MPEG-4 system streams
For now, I would like to constrain the discussion to the first two, as
these
two are also addressed in the proposal of Kikuchi-san. In the following=

review I make a separation between video specific issues and audio spec=
ific
issues. The general issue of the additional bytes is discussed extensiv=
ely
and I don't think there is a need to recap that here. The only
clarification
that may be useful in this context is that the additional bytes can be =
used
to carry the remainder of an SL header in cases where needed, thereby
providing the same functionality as in the proposal of Civanlar et al.

Video specific issues

My proposal would be to use for the generic MPEG-4 video ES format exac=
tly
the same payload type and names as Kikuchi-san. I also suggest to use t=
he
same fragmentation rules. So from my perspective no need to change anyt=
hing
specific to that.

Audio specific issues

Also for the generic MPEG-4 audio ES format I suggest to use the same
payload type, names and fragmentation rules as Kikuchi-san. With audio =
we
have the "LATM problem". However, with some pragmatism we may be able t=
o
solve this. In Paris we agreed to allow for more than one access unit
within
one SL packet. We could slightly modify this by allowing LATM as multip=
lex
structure within SL. (Note that the FlexMux provides a multiplex struct=
ure
on top of SL.) To indicate usage of LATM within an SL packet a currentl=
y
reserved bit in the SL header could be assigned. If we agree on his, th=
en
the important question is whether we need also define carriage of MPEG-=
4
audio ES over IP without LATM. Perhaps not. LATM adds only little overh=
ead
in case of a single large AU and for a small AU it is likely that LATM =
is
applied anyhow. A final remark on this issue: LATM is also used for
carriage
of MPEG-4 audio ES over MPEG-2, pretty much consistent with usage of LA=
TM
within SL and for transport over IP.

Along these lines I will propose specific wording changes to the propos=
al
of
Kikuchi-san. Please comment if you disagree with the approach described=
. I
hope that my proposal for usage of LATM is not too big of a shock. In m=
y
view it is an acceptable price to pay for compatibility with the propos=
al
>from Kikuchi-san. And I believe such compatibility is important for the=

success of MPEG-4 on the market place.

Best regards,

Jan

=



From rem-conf Thu Sep 14 08:01:27 2000 
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From: Zvi Lifshitz <zvil@csi.com>
To: "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>,
        "4on2andIP-sys@advent.ee.columbia.edu" <4on2andIP-sys@advent.ee.columbia.edu>
Cc: "rem-conf@es.net" <rem-conf@es.net>
Subject: RE: Kikuchi-san proposal as subset of generic MPEG-4 formats
Date: Thu, 14 Sep 2000 17:59:24 +0200
Organization: Optibase Ltd.
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[jan.vandermeer@philips.com:]

I fully agree that signaling at OD level is a better solution. With respect
to migration towards other solutions, I think that it is better to have one
simple and straightforward solution for carriage of MPEG-4 A/V ES over IP

[Reply:]

But you want to further complicate the solution that we already have, simple or not..

[Original message:]

then the situation in which "you have many standards to choose from".

[Reply:]

There is something in between - the standard that best fits to every case.

[Original message:]

With respect to audio, the reason of my suggestion to always use LATM is to
achieve a single solution for audio.

[Reply:]

And what is the reason to achieve a single solution for audio?

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm ==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October 10th-12th, 2000






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To: <4on2andIP-sys@advent.ee.columbia.edu>, <Guido.Franceschini@CSELT.IT>
Cc: <rem-conf@es.net>
Subject: RE: Kikuchi-san proposal as subset of generic MPEG-4 formats
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Dear Guido,

As also said in my reply to Zvi, I have no problems to remove the signa=
ling
>from SL level. Zvi and you are right.

With respect to a solution in the control plane, that is fine too, as l=
ong
as it is required that clients MUST support both cases (with and withou=
t the
additional bytes). I agree with your analysis that such solution may be=
 more
elegant; however, taking more time for study is only OK as long as we t=
ake
into account that the ITU needs their "compatible solution" really fast=
. If
the "control plane" solution can not be achieved quickly, I propose to =
focus
on the solution I suggested.

Perhaps for me the best thing to do is wait whether there are some more=

reactions, possibly also from the US. Tomorrow I have a 12 hr+ flight b=
ack
>from the Far East and than I may be able to prepare a more detailed
solution. To be distributed after late tomorrow, European time.

Kind regards,

Jan

-----Original Message-----
From:	Guido.Franceschini@CSELT.IT [mailto:Guido.Franceschini@CSELT.IT]
Sent:	donderdag, 14 september, 2000 14:29
To:	Jan vanderMeer; 4on2andIP-sys@advent.ee.columbia.edu
Cc:	rem-conf@es.net
Subject:	RE: Kikuchi-san proposal as subset of generic MPEG-4 formats


Dear Jan,

I am not really happy to signal LATM usage (which is an audio specific =
tool)
at the SL level (which is generic). I would prefer to expand the audio
specific configuration parameters.

Also, I think that while compatibility at the level of RTP packets is
reachable (e.g. by means of the extra byte), the real technical difficu=
lty
resides in the control plane. Once somebody finds a solution for the co=
ntrol
plane issues, then maybe we will find more than one possible approach f=
or
the data plane.
The usage of one extra byte (thus a payload header) in all RTP packets =
is a
possible technical solution, but also the usage of an indicator at conf=
ig
time telling whether such extra byte (thus a payload header) is present=
 or
not in the RTP packets could be considered (this way you would not forc=
e the
extra byte all the time). And maybe there are more solutions.
To conclude: I would not assert that the ONLY way to fulfill your
requirement is to force one extra byte all the time. I understand and a=
gree
with the requirement, but personally don't like the proposed solution. =
I
would appreciate some more investigation in the control plane aspects.

Also, let's remember that the SDP and the MPEG-4 Object Descriptors ser=
ve
two very similar purposes. In IETF current practice, the 'negotiation' =
is
mostly done using SDP, while when using MPEG-4 Systems it is done throu=
gh
ODs.
A solution aiming at serving with the same bits of information devices
compliant to the IETF current practice as well as devices compliant to
MPEG-4 Systems MUST consider the overlap between SDP and ODs.

IMHO, we can start this work, but we cannot rush it.
And this work would involve the MMUSIC group.

Regards
Guido



> ----------
> From: 	jan.vandermeer@philips.com[SMTP:jan.vandermeer@philips.com]
> Sent: 	Thursday, September 14, 2000 12:30 PM
> To: 	4on2andIP-sys@advent.ee.columbia.edu
> Cc: 	rem-conf@es.net
> Subject: 	Kikuchi-san proposal as subset of generic MPEG-4 formats
>
> Dear all,
>
> As promised in a previous email, I would like to review the problems
> associated with making the proposal of Kikuchi-san a subset of the se=
t of
> generic formats that hopefully will be produced in La Baule. My assum=
ption
> is that there will be "generic formats" for several MPEG-4 streams, s=
uch
> as:
> - MPEG-4 audio elementary streams (allowing for carriage of the remai=
nder
> of
> an SL header)
> - MPEG-4 video elementary streams (allowing for carriage of the remai=
nder
> of
> an SL header)
> - MPEG-4 system streams
> For now, I would like to constrain the discussion to the first two, a=
s
> these
> two are also addressed in the proposal of Kikuchi-san. In the followi=
ng
> review I make a separation between video specific issues and audio
> specific
> issues. The general issue of the additional bytes is discussed extens=
ively
> and I don't think there is a need to recap that here. The only
> clarification
> that may be useful in this context is that the additional bytes can b=
e
> used
> to carry the remainder of an SL header in cases where needed, thereby=

> providing the same functionality as in the proposal of Civanlar et al=
.
>
> Video specific issues
>
> My proposal would be to use for the generic MPEG-4 video ES format ex=
actly
> the same payload type and names as Kikuchi-san. I also suggest to use=
 the
> same fragmentation rules. So from my perspective no need to change
> anything
> specific to that.
>
> Audio specific issues
>
> Also for the generic MPEG-4 audio ES format I suggest to use the same=

> payload type, names and fragmentation rules as Kikuchi-san. With audi=
o we
> have the "LATM problem". However, with some pragmatism we may be able=
 to
> solve this. In Paris we agreed to allow for more than one access unit=

> within
> one SL packet. We could slightly modify this by allowing LATM as mult=
iplex
> structure within SL. (Note that the FlexMux provides a multiplex stru=
cture
> on top of SL.) To indicate usage of LATM within an SL packet a curren=
tly
> reserved bit in the SL header could be assigned. If we agree on his, =
then
> the important question is whether we need also define carriage of MPE=
G-4
> audio ES over IP without LATM. Perhaps not. LATM adds only little ove=
rhead
> in case of a single large AU and for a small AU it is likely that LAT=
M is
> applied anyhow. A final remark on this issue: LATM is also used for
> carriage
> of MPEG-4 audio ES over MPEG-2, pretty much consistent with usage of =
LATM
> within SL and for transport over IP.
>
> Along these lines I will propose specific wording changes to the prop=
osal
> of
> Kikuchi-san. Please comment if you disagree with the approach describ=
ed. I
> hope that my proposal for usage of LATM is not too big of a shock. In=
 my
> view it is an acceptable price to pay for compatibility with the prop=
osal
> from Kikuchi-san. And I believe such compatibility is important for t=
he
> success of MPEG-4 on the market place.
>
> Best regards,
>
> Jan
>
>

=



From rem-conf Thu Sep 14 08:37:06 2000 
From rem-conf-request@es.net Thu Sep 14 08:37:05 2000
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From: Zvi Lifshitz <zvil@csi.com>
To: "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>,
        "csp@east.isi.edu" <csp@east.isi.edu>
Cc: "rem-conf@es.net" <rem-conf@es.net>,
        "4on2andIP-sys@advent.ee.columbia.edu" <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call
Date: Thu, 14 Sep 2000 18:30:17 +0200
Organization: Optibase Ltd.
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Jan,

I would welcome continuing the discussion within the "generic" and, 
moreover, the framework contexts, to make it a superset of the Kikuchi-san 
proposal (which doesn't need a change ;-).

Two suggestions I have in mind right now, on top of the SL configuration 
issues we discussed in Paris:

1. Precede the extra (in-band) header in the Generic format by a byte 
indicating its length. This would enable non-MPEG systems to read it.

2. Allowing LATM as valid stream content in MPEG-4.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm 
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October 
10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Thursday, September 14, 2000 4:02 PM
To:	csp@east.isi.edu; zvil@csi.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to 
the last call


Hello Zvi,

I am glad we came to technical understanding. In your email you ask many
questions, some of which I think are answered in the email I sent meanwhile
on Kikuchi-san's proposal as a subset of the generic format. Is that right 
?
Perhaps we could continue our discussion within the "generic context".

Kind regards,

Jan

-----Original Message-----
From:	zvil@csi.com [mailto:zvil@csi.com]
Sent:	donderdag, 14 september, 2000 09:52
To:	csp@east.isi.edu; Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to
the last call


[jan.vandermeer@philips.com:]

2) Define the meaning of the adaptation field in the "generic MPEG-4
specification". For carriage of MPEG-4 ES and MPEG-4 SL streams the
"generic
specification" defines the use of exactly_the_same payload type as defined
in 1). The only difference is the defined use of the adaptation field.
Whether or not an SL header is carried is signaled in the adaptation field.

[Reply:]

If I understand you right, you seem to suggest what I asked you about few
emails ago. To define one payload format, with an empty extension, and
later to define the content of the extension. Same payload format, same
name. Well, if this is your meaning at least I can understand what you want
to do technically. Theoretically this can work, as it solves the problem of
old devices having to recognize new name.

[Over night I thought about another scenario which solves the
old-device-new-name issue, and planned to ask you (and Sam) if this was
your scenario. This is when streams are broadcast but session description
is interactively negotiated point-to-point. Then, assuming a stream is
broadcast in the "new" format and a client starts negotiation to locate
this stream, the server asks the client what formats it recognizes. If the
client says the new format is OK, the server tells it the truth that this
is in the new format. But if the client says it knows only the "old"
format, the server bluffs and tells this is the old format].

Still, I don't believe this can work. I think also Colin said that. First,
I believe this is against the IETF practice. Then, the Generic format
differs from the ES format in much more than just the extension content.
The ES format uses LATM. The Generic format is not only for audio/video
elementary streams. The ES format defines fragmentation rules, which in the
Generic format are configured by the SLConfig. The evolution that one needs
to reach the other is Darwinian and might take 2 billion years... The main
question is, assuming old devices can peacefully skip the extension field,
how do we guarantee that they correctly handle the content beyond?

And, again, I don't see why we need it. If a stream that is delivered with
the ES format can be played and synchronized, why would you ever want to
put something in its adaptation field? What kind of extra SL info do you
think can enhance it? The only thing I can think of is combining it with
other streams that require the Generic format, but this is possible without
touching the stream.

----
at this point I'm getting a phone call from Jan :-)............
---

(20 minutes later).. OK, we clarified between us few more things. At least
we better understand now the disagreement... I'll send my message is as and
let Jan continue...

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000




From rem-conf Thu Sep 14 08:51:32 2000 
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To: Tmima Koren <tmima@cisco.com>
cc: "W. Mark Townsley" <townsley@cisco.com>, brucet@cisco.com, dwing@cisco.com,
   Stephen Casner <casner@packetdesign.com>, gurdeep@microsoft.com,
   palter@zev.net, l2tp@diameter.org, rem-conf@es.net
Subject: Re: 0 byte L2TPHC header and implied PPPMUX 
In-reply-to: Your message of "Thu, 14 Sep 2000 01:40:08 PDT."
             <4.3.1.2.20000914010737.01f5b750@omega.cisco.com> 
Date: Thu, 14 Sep 2000 08:44:29 -0700
From: Andy Valencia <vandys@zendo.com>
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[Tmima Koren <tmima@cisco.com> writes:]

>>flag 2:  value 1 means all packets have implied P=1
>>             value 0 means all packets have implied P=0

I'm concerned with this one.  Since L2TPHC is basically useless except for
payload, what you've achieved with this flag is to ensure that your control
traffic will be interrupted if you *ever* permit all payload traffic to be
given preference by choosing P=1.  My experience is that this will make your
link unworkable.  And no, using the UDP side doesn't help, since priority is
(and should be) global to the PPP connection.

I would suggest that when omitting the L2TPHC header entirely, that P=0
always be the mode of operation.  This shouldn't affect your TCRTP mode of
operation, because the only thing aside from those payload packets would be
control packets which should be given priority anyway (and they can go via
UDP).

>>When flags 1 and 3 are on (=1), both L2TPHC header and PPPMUX protocol id can
> be omitted 

How about A/C fields?  Isn't it a little weird that a field in the middle of
the PPP packet is coming and going based on this, but the bytes in front
depend on PPP negotiation?

Also, why hard-wire it to PPPMUX?  Why not include the implicit protocol
value, so that these savings could apply to IP-only connections which
weren't using PPPMUX and/or TCRTP?

>>PPP packets that have a different priority or that are not PPPMUX can still b
>e sent in the tunnel using the UDP/L2TP format, same as the L2TP control packe
>ts
>>TCRTP (Tunneled Compressed multiplexed RTP) can benefit from these changes: t
>he tunneling overhead will be reduced by 2 bytes.

I guess I'm OK with a 0-byte L2TPHC header option (subject to a nod from the
chair).  I'd sure like to hear from some people (not just infer something
>from silence) on the PPP header aspect.

Andy Valencia



From rem-conf Thu Sep 14 08:54:41 2000 
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Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the
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Zvi,

We agree to a large extend, with the one exception that you know very w=
ell.

Kind regards, anyhow.

Jan

-----Original Message-----
From:	zvil@csi.com [mailto:zvil@csi.com]
Sent:	donderdag, 14 september, 2000 17:34
To:	csp@east.isi.edu; Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response=
 to
the last call


Jan,

I would welcome continuing the discussion within the "generic" and,
moreover, the framework contexts, to make it a superset of the Kikuchi-=
san
proposal (which doesn't need a change ;-).

Two suggestions I have in mind right now, on top of the SL configuratio=
n
issues we discussed in Paris:

1. Precede the extra (in-band) header in the Generic format by a byte
indicating its length. This would enable non-MPEG systems to read it.

2. Allowing LATM as valid stream content in MPEG-4.

Regards,

z

=



From rem-conf Thu Sep 14 10:01:52 2000 
From rem-conf-request@es.net Thu Sep 14 10:01:51 2000
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Message-ID: <39C1038A.ED4ED543@rp.lip6.fr>
Date: Thu, 14 Sep 2000 18:57:46 +0200
From: Vida Rolland <Rolland.Vida@lip6.fr>
Organization: Laboratoire Lip6
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Hello,

I'm new on this mailing list, maybe you've discussed already this
subject. If it is the case, i'm sorry.
My question is related to the new PIM-SSM protocol. What do you think
about a modification of RTP/RTCP in order to work in a PIM-SSM
environment?

Regards,
Rolland Vida






From rem-conf Thu Sep 14 10:15:44 2000 
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Date: Thu, 14 Sep 2000 19:12:01 +0200
From: Franceschini Guido <Guido.Franceschini@CSELT.IT>
Subject: RE: Kikuchi-san proposal as subset of generic MPEG-4 formats
To: 4on2andIP-sys@advent.ee.columbia.edu,
 "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>
Cc: rem-conf@es.net
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Dear Jan,

> With respect to a solution in the control plane, that is fine too, as long
> as it is required that clients MUST support both cases (with and without
> the
> additional bytes). I agree with your analysis that such solution may be
> more
> elegant; however, taking more time for study is only OK as long as we take
> into account that the ITU needs their "compatible solution" really fast.
> If
> the "control plane" solution can not be achieved quickly, I propose to
> focus
> on the solution I suggested.
The point is I believe that your extra-byte proposal does not work as is,
since the 'old' device, in presence of 'new' content, would know from the
control plane only that the RTP Payload Format is the 'generic' one, but it
would not know what media is being carried.

Bye
Guido




From rem-conf Thu Sep 14 10:21:15 2000 
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Date: Thu, 14 Sep 2000 10:15:15 -0700
To: Andy Valencia <vandys@zendo.com>
From: Tmima Koren <tmima@cisco.com>
Subject: Re: 0 byte L2TPHC header and implied PPPMUX 
Cc: "W. Mark Townsley" <townsley@cisco.com>, brucet@cisco.com, dwing@cisco.com,
        Stephen Casner <casner@packetdesign.com>, gurdeep@microsoft.com,
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At 08:44 AM 9/14/00 -0700, Andy Valencia wrote:
>[Tmima Koren <tmima@cisco.com> writes:]
>
> >>flag 2:  value 1 means all packets have implied P=1
> >>             value 0 means all packets have implied P=0
>
>I'm concerned with this one.  Since L2TPHC is basically useless except for
>payload, what you've achieved with this flag is to ensure that your control
>traffic will be interrupted if you *ever* permit all payload traffic to be
>given preference by choosing P=1.  My experience is that this will make your
>link unworkable.  And no, using the UDP side doesn't help, since priority is
>(and should be) global to the PPP connection.
>
>I would suggest that when omitting the L2TPHC header entirely, that P=0
>always be the mode of operation.  This shouldn't affect your TCRTP mode of
>operation, because the only thing aside from those payload packets would be
>control packets which should be given priority anyway (and they can go via
>UDP).

In the typical case of a TCRTP tunnel both ends of the tunnel are tied to virtual PPP interfaces.
RTP packets arrive to one side, compressed and tunneled to the other side where they are decompressed and routed to their destination.
Those RTP packets carry voice and may have certain priority. Those packets should regain their priority when they leave the tunnel.
Can this be done with P=0?


> >>When flags 1 and 3 are on (=1), both L2TPHC header and PPPMUX protocol id can
> > be omitted 
>
>How about A/C fields?  Isn't it a little weird that a field in the middle of
>the PPP packet is coming and going based on this, but the bytes in front
>depend on PPP negotiation?
>
>Also, why hard-wire it to PPPMUX?  Why not include the implicit protocol
>value, so that these savings could apply to IP-only connections which
>weren't using PPPMUX and/or TCRTP?

I thought about this option. The difference is that the PPPMux protocol ID at the head of the packet really acts as a flag to indicate that there is more then one packet in this packet..
but each multiplex carries its own PPP protocol ID.
If you think it's ok to generalize for any protocol, I support that.

Tmima


> >>PPP packets that have a different priority or that are not PPPMUX can still b
> >e sent in the tunnel using the UDP/L2TP format, same as the L2TP control packe
> >ts
> >>TCRTP (Tunneled Compressed multiplexed RTP) can benefit from these changes: t
> >he tunneling overhead will be reduced by 2 bytes.
>
>I guess I'm OK with a 0-byte L2TPHC header option (subject to a nod from the
>chair).  I'd sure like to hear from some people (not just infer something
>from silence) on the PPP header aspect.
>
>Andy Valencia




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Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
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Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
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From rem-conf Thu Sep 14 10:57:35 2000 
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To: Young-Kwon LIM <young@techway.co.kr>
cc: Herpel Carsten <HerpelC@thmulti.com>,
        Stephan Wenger <stewe@cs.tu-berlin.de>, rem-conf <rem-conf@es.net>,
        4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
In-Reply-To: Your message of "Fri, 15 Sep 2000 01:40:43 +0900."
             <000401c01e73$034b9050$a4d2c59d@Young> 
Date: Thu, 14 Sep 2000 13:56:31 -0400
From: Colin Perkins <csp@east.isi.edu>
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--> Young-Kwon LIM writes:
>Dear Stephe and all,
>
>> 
>> Very simple to achieve.  Have two payload types, MPEG-4-SL and MPEG-4-ES.
>> Try to negotiate MPEG-4-SL.  If the other system rejects, use MPEG-4-ES.
>> This is how things are done, at least in H.323, all the time.
>> 
>
>With additional one byte indicating the length of the payload header at
>the beginning of the payload, bitstream for MPEG-4 SL can be comsumed by
>the device only supporting MPEG-4 ES without any modification. This is one
>of the advantages using this byte.  

This does not work, since a device supporting the ES format will only
understand LATM audio or video in Combined Configuration/Elementary 
stream mode. If the content sent after the payload header is in a format
other than these, the old receiver will break (even if it can skip the
payload header).

Surely you are not proposing a generic format which can only transport
this particular subset of MPEG4?

Colin



From rem-conf Thu Sep 14 11:09:43 2000 
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To: Young-Kwon LIM <young@techway.co.kr>
cc: "Narasimhan, Sam (SD-EX)" <SNarasimhan@gi.com>,
        "'Zvi Lifshitz'" <ZviL@optibase.com>,
        "jan.vandermeer" <jan.vandermeer@philips.com>,
        rem-conf <rem-conf@es.net>,
        4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
In-Reply-To: Your message of "Fri, 15 Sep 2000 01:51:21 +0900."
             <000501c01e73$0444bd60$a4d2c59d@Young> 
Date: Thu, 14 Sep 2000 14:07:39 -0400
From: Colin Perkins <csp@east.isi.edu>
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--> Young-Kwon LIM writes:
>> The payload type will have to be negotiated out of band anyway, there will
>> be no more static payload type assignments. Since there must be negotiation
>> having this extra byte in the payload format makes no sense.
>
>I'd like to be more clear on this point. The payload type of MPEG-4 ES
>over RTP bitstream and MPEG-4 SL over RTP bitstream can have the same
>payload type semantically. Right?

I don't understand your question. If the formats are different, they will
have different MIME types. What payload type those MIME types are mapped
to is not relevent.

Colin



From rem-conf Thu Sep 14 11:13:24 2000 
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To: Young-Kwon LIM <young@techway.co.kr>
cc: Herpel Carsten <HerpelC@thmulti.com>, rem-conf <rem-conf@es.net>,
        4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
In-Reply-To: Your message of "Fri, 15 Sep 2000 01:29:27 +0900."
             <000301c01e73$023ba6f0$a4d2c59d@Young> 
Date: Thu, 14 Sep 2000 14:11:54 -0400
From: Colin Perkins <csp@east.isi.edu>
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--> Young-Kwon LIM writes:
>Dear Carsten, and all
>
>> 
>> One assumption is that huge numbers of devices will be out there that use
>> the ES payload format. Assuming that these devices cannot be upgraded to a
>> newer payload format!
>> Then applications come up with more functionality, using MPEG-4 Systems. New
>> devices will exist that know how to use this functionality. Old devices
>> should at least be able to understand the (unmodified!) MPEG-4 Video over
>> RTP stream. Since the old devices are not reprogrammable, they cannot be
>> upgraded to another payload format. So the old format needs to be by design
>> compatible to any new format.
>
>I do belive that this scenario is more realistic and more useful than the
>second case.

The existing ES format is intended for devices which do not implement
MPEG-4 systems. I fail to see why anyone would use it for an application
which depended on the systems framework, since a number of features of 
the draft are in direct opposition to the systems framework.

Square peg. Round hole.

Colin



From rem-conf Thu Sep 14 11:14:46 2000 
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To: Young-Kwon LIM <young@techway.co.kr>
cc: "jan.vandermeer" <jan.vandermeer@philips.com>,
        stewe <stewe@cs.tu-berlin.de>, rem-conf <rem-conf@es.net>,
        4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
In-Reply-To: Your message of "Fri, 15 Sep 2000 01:27:21 +0900."
             <000001c01e73$01567710$a4d2c59d@Young> 
Date: Thu, 14 Sep 2000 14:13:12 -0400
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--> Young-Kwon LIM writes:
>> In RTP, that parameter is the payload type field. 
>> 
>> I really don't understand why this additional field will add value. It 
>> allows for systems to skip data which they don't understand, how is this
>> different from using the payload type for the same effect?
>
>The length of additional header will be variable. So, the device should
>know the length to skip the additional headers it doesn't understand.

But this doesn't help, since the ES draft imposes strict limitations on
the format of the elementary stream following this additional header. If
you try to use it for generic elementary stream transport, you will break
existing receivers.

Colin



From rem-conf Thu Sep 14 11:29:48 2000 
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From: John Lazzaro <lazzaro@CS.Berkeley.EDU>
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To: 4on2andIP-sys@advent.ee.columbia.edu, jan.vandermeer@philips.com
Subject: Re: Kikuchi-san proposal as subset of generic MPEG-4 formats
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[jan.vandermeer@philips.com writes]

> If we agree on his, then
> the important question is whether we need also define carriage of MPEG-4
> audio ES over IP without LATM. Perhaps not. LATM adds only little overhead
> in case of a single large AU and for a small AU it is likely that LATM is
> applied anyhow. 

Here's an app where it may be worthwhile to carry MPEG-4 audio ES over IP 
without LATM: using SA_access_units to transmit MIDI Events between a set
of Structured Audio decoders over a low-latency network for real-time 
musical performance. Since SA_access_units's are defined to hold a single
MIDI command, most often 3 bytes, and in many cases there won't be another
audio codec being used along with the Structured Audio decoder, the LATM
overhead will be significant and won't be adding any value.

-------------------------------------------------------------------------
John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
-------------------------------------------------------------------------




From rem-conf Thu Sep 14 11:47:21 2000 
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From: Kris Huber <khuber@sorensonlabs.com>
To: Colin Perkins <csp@east.isi.edu>, Young-Kwon LIM <young@techway.co.kr>
Cc: Herpel Carsten <HerpelC@thmulti.com>, rem-conf <rem-conf@es.net>, 
	4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to
	 the last call 
Date: Thu, 14 Sep 2000 12:43:37 -0600
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Colin,

>The existing ES format is intended for devices which do not implement
>MPEG-4 systems. I fail to see why anyone would use it for an application
>which depended on the systems framework, since a number of features of 
>the draft are in direct opposition to the systems framework.
>
>Square peg. Round hole.

Existing ES may become used by systems that implement the system framework
simply to avoid sending duplicate broadcasts to non-updateable devices and
newer devices utilizing systems tools.  I think this is the reason why.
Remember video ES data probably dominates bandwidth consumption which is not
unlimited (yet!?) and therefore we should consider a precious resource.

I'm assuming services worried about not obsoleting nonupgradable devices can
take the  approach of using Kikuchi-san format in their systems-based
services for video ES if they want.  Of course, it is not guaranteed that
nonupgradeable devices won't become obsolete quite soon, especially if the
generic format is rapidly deployed, but since it seems to be an option of
the services provider to avoid it, I am satisfied and have no further
comment other than to wish the generic format definition to continue along
well.

I may attend the pre-meeting Saturday (if I'm not bought off by the airlines
to delay!
$:>).  Actually, I have relatives living at a few hours distance that I hope
to visit, too.  I have full confidence in the superiority of the rest of the
group in this area!

regards,
Kris

-----Original Message-----
From: Colin Perkins [mailto:csp@east.isi.edu]
Sent: Thursday, September 14, 2000 12:12 PM
To: Young-Kwon LIM
Cc: Herpel Carsten; rem-conf; 4on2andIP-sys
Subject: Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response
to the last call 


--> Young-Kwon LIM writes:
>Dear Carsten, and all
>
>> 
>> One assumption is that huge numbers of devices will be out there that use
>> the ES payload format. Assuming that these devices cannot be upgraded to
a
>> newer payload format!
>> Then applications come up with more functionality, using MPEG-4 Systems.
New
>> devices will exist that know how to use this functionality. Old devices
>> should at least be able to understand the (unmodified!) MPEG-4 Video over
>> RTP stream. Since the old devices are not reprogrammable, they cannot be
>> upgraded to another payload format. So the old format needs to be by
design
>> compatible to any new format.
>
>I do belive that this scenario is more realistic and more useful than the
>second case.

The existing ES format is intended for devices which do not implement
MPEG-4 systems. I fail to see why anyone would use it for an application
which depended on the systems framework, since a number of features of 
the draft are in direct opposition to the systems framework.

Square peg. Round hole.

Colin



From rem-conf Thu Sep 14 12:03:39 2000 
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From: Zvi Lifshitz <zvil@csi.com>
To: "'Young-Kwon LIM'" <young@techway.co.kr>,
        "Narasimhan, Sam (SD-EX)"
	 <SNarasimhan@gi.com>,
        Colin Perkins <csp@east.isi.edu>
Cc: "jan.vandermeer@philips.com" <jan.vandermeer@philips.com>,
        "rem-conf@es.net" <rem-conf@es.net>,
        "4on2andIP-sys@advent.ee.columbia.edu" <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
Date: Thu, 14 Sep 2000 22:01:23 +0200
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[Young-Kwon LIM:]

I'd like to be more clear on this point. The payload type of MPEG-4 ES over 
RTP bitstream and MPEG-4 SL over RTP bitstream can have the same tayload 
type semantically. Right?

[Reply:]

Wrong.
The option that was raised when I talked to Jan was that a server would 
advertise same content twice, once as having the ES format and then as 
having the Generic format. This would probably work but IMO is useless...

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm 
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October 
10th-12th, 2000






From rem-conf Thu Sep 14 12:08:40 2000 
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From: Zvi Lifshitz <zvil@csi.com>
To: "'Young-Kwon LIM'" <young@techway.co.kr>,
        Herpel Carsten
	 <HerpelC@thmulti.com>,
        Stephan Wenger <stewe@cs.tu-berlin.de>
Cc: "rem-conf@es.net" <rem-conf@es.net>,
        "4on2andIP-sys@advent.ee.columbia.edu" <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to the last call 
Date: Thu, 14 Sep 2000 22:07:17 +0200
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If you want to make the stream consumable to devices knowing only MPEG4-ES, 
use MPEG-4 ES. What value will the MPEG-4 SL give you?

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm 
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October 
10th-12th, 2000


-----Original Message-----
From:	Young-Kwon LIM [SMTP:young@techway.co.kr]
Sent:	Thursday, September 14, 2000 6:41 PM
To:	Herpel Carsten; Stephan Wenger
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	Re: AHG comments on draft-ietf-avt-rtp-mpeg4-es-03 in response to 
the last call


Dear Stephe and all,

>
> Very simple to achieve.  Have two payload types, MPEG-4-SL and MPEG-4-ES.
> Try to negotiate MPEG-4-SL.  If the other system rejects, use MPEG-4-ES.
> This is how things are done, at least in H.323, all the time.
>

With additional one byte indicating the length of the payload header at the 
beginning of the payload, bitstream for MPEG-4 SL can be comsumed by the 
device only supporting MPEG-4 ES without any modification. This is one of 
the advantages using this byte.

Sincerely,
Young.




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From: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
To: "IETF AVT" <rem-conf@es.net>
Cc: "Zvi Lifshitz" <ZviL@optibase.com>, "Colin Perkins" <csp@east.isi.edu>,
        "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
Subject: comments on in-band-signaling proposal
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Dear all experts,


The authors of the Internet-Draft have discussed the issue of
in-band-signaling proposal. We understand the importance of
interoperability, however, we do not believe the specific proposal of an
extra byte in the RTP header is a good solution. It is our common and firm
desire that any proposal will not degrade the original intention, purpose
and functionality of the RTP payload format.

Especially, the following points are important.

o Overhead caused by an extra RTP header
The in-band-signaling proposal will cause at least one byte overhead per
each RTP packet. Since this RTP payload format may be used in very low
bitrate applications such as mobile and analog modem, the overhead is a
serious problem. On the other hand, the ways suggested by Colin, Stephan,
etc. of using different payload types will not cause any overhead problem.

o Stability and reliability
As was pointed out by many, there is no guaranty that receivers of ES-based
RTP payload format can receive SL-based RTP packets correctly, even if the
extra byte in the RTP header is added. It is dangerous to adopt such
proposal at this moment without any guaranty. It is our strong desire that
only verified solution will be adopted in the RTP payload format
specification.

o Schedule
There is no specific description of changes about the proposal, so far. It
is hard to accept such proposal. Even if the proposal has some value
(although we doubt it), it should not be too late to adopt it in
SL-based proposal, by keeping backword compatibility with ES-based RTP
payload format.


These points are very important to keep the original functionality of the
proposal. Constructive comments are welcome, however, we cannot accept
any proposals to harm the original intention and functionality. By this
point, it is hard to accept the in-band-signaling proposal which causes
overhead problem and which is not well verified. It should be remarked that
the alternatives proposed by IETF experts will not cause any such problems.


Kind regards,
Yoshihiro Kikuchi
and authors of the Intenet Draft

----
        Yoshihiro Kikuchi

E-mail: kiku@eel.rdc.toshiba.co.jp
Corporate Research and Development Center
TOSHIBA Corporation
TEL: +81 +44 549 2288    FAX: +81 +44 520 1267





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From rem-conf Thu Sep 14 17:15:46 2000 
From rem-conf-request@es.net Thu Sep 14 17:15:45 2000
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Date: Thu, 14 Sep 2000 19:45:43 -0400
From: "Narasimhan, Sam (SD-EX)" <SNarasimhan@gi.com>
Subject: RE: comments on in-band-signaling proposal
To: 'Yoshihiro Kikuchi' <kiku@eel.rdc.toshiba.co.jp>,
 IETF AVT <rem-conf@es.net>,
 "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>
Cc: Zvi Lifshitz <ZviL@optibase.com>, Colin Perkins <csp@east.isi.edu>,
 "'4on2andIP-sys@advent.ee.columbia.edu'" <4on2andIP-sys@advent.ee.columbia.edu>
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Dear Kikuchi-San and experts,

 I am coming from a world where content creators and
servers are very 'receiver-aware' and will not do anything
in the content creation that will crash receivers. In this
regard, the service providers who use the content want to
play this on as many receivers as possible and they are also
aware of the fact that receivers vary in their capabilities.
Service providers, on the other hand, will not send 2 separate
content streams to play on 2 receivers on the same wire as this 
consumes precious bandwidth. 

 2 examples of streams that one may want to play on receivers
that implement your draft are 

1. A video ES that required CTS (in the RTP header) and DTS (which
   could be in the adaptation field). This may be needed for receivers
   that implement MPEG buffer models. This stream can play on
   receivers that implement your draft as it has a bigger buffer
   and does not need DTS, as long as your draft enabled these
   receivers to use the adaptation length field to offset where
   the video ES payload was picked up.
2. One may author SL packetized video where there was information
   in the SL header and this will be in the adaptation field. In addition,
   the authors will make sure that the video ES in the SL packet 
   would be independently decodable by receivers that implemented your 
   draft. This will require the ability to parse the first byte in
   the adaptation header and use it to offset where the video ES
   payload would be picked up.

 If your draft can accommodate this simple extension, then it will enable
the receivers to play more content from the future and it will enable
service providers to keep this in mind when they author MPEG-4 based
content in the future.

 Regards to your concerns,

1.  The overhead is 1 byte/RTP packet and this is a 6.25% addition
    to the 16 byte RTP header. If we sent 1 VOP per packet, then it
    is 30 bytes/sec (assuming 30 Hz video) or 240 bits/sec overhead.
    This is very minimal. The FEC proposal proposes the same type
    of overhead.
2.  The content authors will always make sure they follow all the
    other recommendations from your draft to make sure that the
    future content will play on receivers that implement the draft
    (such as sending data in decoding order, not splitting the headers
    etc). The only change from an implementation of today is the 
    extra parsing of one byte before the ES payload to offset where
    the ES payload is picked up.
3.  The change to your draft would to be add 2 fields after the CSRC 
    and before the video ES. These would be
    'adaptation length' and
    'length (if adaptation length is greater than 0)'.
    The semantics will say, the first byte after CSRC will contain the
    length of adaptation field. If this length value is zero, then the
    video payload will start in the next byte. Otherwise, the video
    payload will start after the number of bytes indicated in the
    adaptation length field. The content of the adaptation field after
    the length is private and left for future extensions.

 Jan Vandermeer may also have some suggested changes to the draft, if this
can be accommodated. Thank you for your kind patience.

Best Regards
Sam Narasimhan
Motorola
 
-----Original Message-----
From: Yoshihiro Kikuchi [mailto:kiku@eel.rdc.toshiba.co.jp]
Sent: Thursday, September 14, 2000 12:38 PM
To: IETF AVT
Cc: Zvi Lifshitz; Colin Perkins; Yoshihiro Kikuchi
Subject: comments on in-band-signaling proposal


Dear all experts,


The authors of the Internet-Draft have discussed the issue of
in-band-signaling proposal. We understand the importance of
interoperability, however, we do not believe the specific proposal of an
extra byte in the RTP header is a good solution. It is our common and firm
desire that any proposal will not degrade the original intention, purpose
and functionality of the RTP payload format.

Especially, the following points are important.

o Overhead caused by an extra RTP header
The in-band-signaling proposal will cause at least one byte overhead per
each RTP packet. Since this RTP payload format may be used in very low
bitrate applications such as mobile and analog modem, the overhead is a
serious problem. On the other hand, the ways suggested by Colin, Stephan,
etc. of using different payload types will not cause any overhead problem.

o Stability and reliability
As was pointed out by many, there is no guaranty that receivers of ES-based
RTP payload format can receive SL-based RTP packets correctly, even if the
extra byte in the RTP header is added. It is dangerous to adopt such
proposal at this moment without any guaranty. It is our strong desire that
only verified solution will be adopted in the RTP payload format
specification.

o Schedule
There is no specific description of changes about the proposal, so far. It
is hard to accept such proposal. Even if the proposal has some value
(although we doubt it), it should not be too late to adopt it in
SL-based proposal, by keeping backword compatibility with ES-based RTP
payload format.


These points are very important to keep the original functionality of the
proposal. Constructive comments are welcome, however, we cannot accept
any proposals to harm the original intention and functionality. By this
point, it is hard to accept the in-band-signaling proposal which causes
overhead problem and which is not well verified. It should be remarked that
the alternatives proposed by IETF experts will not cause any such problems.


Kind regards,
Yoshihiro Kikuchi
and authors of the Intenet Draft

----
        Yoshihiro Kikuchi

E-mail: kiku@eel.rdc.toshiba.co.jp
Corporate Research and Development Center
TOSHIBA Corporation
TEL: +81 +44 549 2288    FAX: +81 +44 520 1267





From rem-conf Thu Sep 14 22:35:34 2000 
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From: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
To: "Narasimhan, Sam (SD-EX)" <SNarasimhan@gi.com>,
        "IETF AVT" <rem-conf@es.net>, <jan.vandermeer@philips.com>
Cc: "Zvi Lifshitz" <ZviL@optibase.com>, "Colin Perkins" <csp@east.isi.edu>,
        "4on2andIP-sys" <4on2andIP-sys@advent.ee.columbia.edu>,
        "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
References: <973597126BDDD11197AA00805FA7EBC9034239BA@ntas0026.gi.com>
Subject: Re: comments on in-band-signaling proposal
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Dear Sam and all,

Additional comment:

When one content is delivered to different applications and environments,
such as set-top-box and mobile terminals, there are many differences on the
conditions. Network conditions, such as bitrate, error rate, jitter rate,
are
different in most cases. Profile@Level for set-top-box and tiny mobile
terminals are probably different. For some applications, non-MPEG-4 codec
may be used in combined. To adjust to these differences, we need to use
different set of parameters, such as bitrate, image size, refresh rate,
etc. at codec level anyway and different types of codec may be necessary.

>  I am coming from a world where content creators and
> servers are very 'receiver-aware' and will not do anything
> in the content creation that will crash receivers. In this
> regard, the service providers who use the content want to
> play this on as many receivers as possible and they are also
> aware of the fact that receivers vary in their capabilities.
> Service providers, on the other hand, will not send 2 separate
> content streams to play on 2 receivers on the same wire as this
> consumes precious bandwidth.
>
Considering above, having the same format ONLY at the RTP level may not help
the situation so much. It may be useful for some kind of mobile terminals
with Mbps bandwidth, error-free, large display size, but maybe it is not so
usual case. Or, such powerful terminal can easily implement SL-based
format.

>  2 examples of streams that one may want to play on receivers
> that implement your draft are
>
> 1. A video ES that required CTS (in the RTP header) and DTS (which
>    could be in the adaptation field). This may be needed for receivers
>    that implement MPEG buffer models. This stream can play on
>    receivers that implement your draft as it has a bigger buffer
>    and does not need DTS, as long as your draft enabled these
>    receivers to use the adaptation length field to offset where
>    the video ES payload was picked up.
> 2. One may author SL packetized video where there was information
>    in the SL header and this will be in the adaptation field. In addition,
>    the authors will make sure that the video ES in the SL packet
>    would be independently decodable by receivers that implemented your
>    draft. This will require the ability to parse the first byte in
>    the adaptation header and use it to offset where the video ES
>    payload would be picked up.
>
I think these streams are covered by SL-based payload. Why don't you use
this?

>  If your draft can accommodate this simple extension, then it will enable
> the receivers to play more content from the future and it will enable
> service providers to keep this in mind when they author MPEG-4 based
> content in the future.
>
>  Regards to your concerns,
>
> 1.  The overhead is 1 byte/RTP packet and this is a 6.25% addition
>     to the 16 byte RTP header. If we sent 1 VOP per packet, then it
>     is 30 bytes/sec (assuming 30 Hz video) or 240 bits/sec overhead.
>     This is very minimal. The FEC proposal proposes the same type
>     of overhead.

I'm not convinced by this explanation. A VOP may be fragmented into many RTP
packets for error resiliency and/or low-delay requirements, and the size of
the RTP packet is relatively small. The fixed RTP header is compressed by
the header compression technique, but the extra field is not. The overhead
is still a serious problem since this RTP payload format may be used in very
low bitrate environment.

> 2.  The content authors will always make sure they follow all the
>     other recommendations from your draft to make sure that the
>     future content will play on receivers that implement the draft
>     (such as sending data in decoding order, not splitting the headers
>     etc). The only change from an implementation of today is the
>     extra parsing of one byte before the ES payload to offset where
>     the ES payload is picked up.

Is there any guaranty on this? There were many concerns raised by both IETF
and MPEG experts on whether the ES payload receiver can decode SL payload
packet correctly only by adding the extra byte fields. We must be care not
to bring any danger to the implementers of ES payload format.

> 3.  The change to your draft would to be add 2 fields after the CSRC
>     and before the video ES. These would be
>     'adaptation length' and
>     'length (if adaptation length is greater than 0)'.
>     The semantics will say, the first byte after CSRC will contain the
>     length of adaptation field. If this length value is zero, then the
>     video payload will start in the next byte. Otherwise, the video
>     payload will start after the number of bytes indicated in the
>     adaptation length field. The content of the adaptation field after
>     the length is private and left for future extensions.
>
Maybe I'm not a correct person to decide whether this change is small or
not. But again, we must be care not to bring any danger to the implementers
of ES payload format.

>  Jan Vandermeer may also have some suggested changes to the draft, if this
> can be accommodated. Thank you for your kind patience.
>
We are not so stubborn to reject all changes if it is really useful and
supported. Actually, many people helped us for the editorial clarification
of the Internet-Draft. However, the proposal of adding extra byte has many
serious problems:

- Clear disadvantage of the overhead.
- No guaranty to work..
- There are many concerns raised by both MPEG and IETF members.

How can we adopt such proposal?

Best regards,
Yoshihiro Kikuchi

> -----Original Message-----
> From: Yoshihiro Kikuchi [mailto:kiku@eel.rdc.toshiba.co.jp]
> Sent: Thursday, September 14, 2000 12:38 PM
> To: IETF AVT
> Cc: Zvi Lifshitz; Colin Perkins; Yoshihiro Kikuchi
> Subject: comments on in-band-signaling proposal
>
>
> Dear all experts,
>
>
> The authors of the Internet-Draft have discussed the issue of
> in-band-signaling proposal. We understand the importance of
> interoperability, however, we do not believe the specific proposal of an
> extra byte in the RTP header is a good solution. It is our common and firm
> desire that any proposal will not degrade the original intention, purpose
> and functionality of the RTP payload format.
>
> Especially, the following points are important.
>
> o Overhead caused by an extra RTP header
> The in-band-signaling proposal will cause at least one byte overhead per
> each RTP packet. Since this RTP payload format may be used in very low
> bitrate applications such as mobile and analog modem, the overhead is a
> serious problem. On the other hand, the ways suggested by Colin, Stephan,
> etc. of using different payload types will not cause any overhead problem.
>
> o Stability and reliability
> As was pointed out by many, there is no guaranty that receivers of
ES-based
> RTP payload format can receive SL-based RTP packets correctly, even if the
> extra byte in the RTP header is added. It is dangerous to adopt such
> proposal at this moment without any guaranty. It is our strong desire that
> only verified solution will be adopted in the RTP payload format
> specification.
>
> o Schedule
> There is no specific description of changes about the proposal, so far. It
> is hard to accept such proposal. Even if the proposal has some value
> (although we doubt it), it should not be too late to adopt it in
> SL-based proposal, by keeping backword compatibility with ES-based RTP
> payload format.
>
>
> These points are very important to keep the original functionality of the
> proposal. Constructive comments are welcome, however, we cannot accept
> any proposals to harm the original intention and functionality. By this
> point, it is hard to accept the in-band-signaling proposal which causes
> overhead problem and which is not well verified. It should be remarked
that
> the alternatives proposed by IETF experts will not cause any such
problems.
>
>
> Kind regards,
> Yoshihiro Kikuchi
> and authors of the Intenet Draft
>
> ----
>         Yoshihiro Kikuchi
>
> E-mail: kiku@eel.rdc.toshiba.co.jp
> Corporate Research and Development Center
> TOSHIBA Corporation
> TEL: +81 +44 549 2288    FAX: +81 +44 520 1267
>
>
>
>






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From: Zvi Lifshitz <zvil@csi.com>
To: "'Narasimhan, Sam (SD-EX)'" <SNarasimhan@gi.com>,
        "'Yoshihiro Kikuchi'"
	 <kiku@eel.rdc.toshiba.co.jp>,
        IETF AVT <rem-conf@es.net>,
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Cc: Colin Perkins <csp@east.isi.edu>,
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Subject: RE: comments on in-band-signaling proposal
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[Narasimhan, Sam (SD-EX):]

1. A video ES that required CTS (in the RTP header) and DTS (which
   could be in the adaptation field). This may be needed for receivers
   that implement MPEG buffer models. This stream can play on
   receivers that implement your draft as it has a bigger buffer
   and does not need DTS, as long as your draft enabled these
   receivers to use the adaptation length field to offset where
   the video ES payload was picked up.

[Reply:]

The jitter on IP network is such that practically receivers may always assume CTS=DTS for implementation of the buffer model.

[Original message:]

1.  The overhead is 1 byte/RTP packet and this is a 6.25% addition
    to the 16 byte RTP header. If we sent 1 VOP per packet, then it
    is 30 bytes/sec (assuming 30 Hz video) or 240 bits/sec overhead.
    This is very minimal. The FEC proposal proposes the same type
    of overhead.

[Reply:]

There is header compression. I think Kikuchi-san mentioned that, along with few other good arguments.

Regards,

z
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From rem-conf Thu Sep 14 23:55:15 2000 
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Date: Fri, 15 Sep 2000 08:51:41 +0200
From: Franceschini Guido <Guido.Franceschini@CSELT.IT>
Subject: RE: comments on in-band-signaling proposal
To: "Narasimhan, Sam (SD-EX)" <SNarasimhan@gi.com>, IETF AVT <rem-conf@es.net>,
 jan.vandermeer@philips.com, 'Yoshihiro Kikuchi' <kiku@eel.rdc.toshiba.co.jp>
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Sorry MPEG friends,

but I fully support Kikuchi-sun.

At least until I see a proposal that does not address just the extra-byte
detail, but solves the real problem (negotiation). So that everybody can
convince himself that there are different ways to fit the requirement at the
level of RTP packets ...
Remember, when using the SL mapping an 'old' receiver has currently no way
to know the media being carried. It could even be BIFS !
[ps: in addition there is a need to clean-up the LATM issue, but this is
orthogonal IMHO.]

Best regards
Guido



> ----------
> From: 	Yoshihiro Kikuchi[SMTP:kiku@eel.rdc.toshiba.co.jp]
> Sent: 	Friday, September 15, 2000 7:29 AM
> To: 	Narasimhan, Sam (SD-EX); IETF AVT; jan.vandermeer@philips.com
> Cc: 	Zvi Lifshitz; Colin Perkins; 4on2andIP-sys; Yoshihiro Kikuchi
> Subject: 	Re: comments on in-band-signaling proposal
> 
> Dear Sam and all,
> 
> Additional comment:
> 
> When one content is delivered to different applications and environments,
> such as set-top-box and mobile terminals, there are many differences on
> the
> conditions. Network conditions, such as bitrate, error rate, jitter rate,
> are
> different in most cases. Profile@Level for set-top-box and tiny mobile
> terminals are probably different. For some applications, non-MPEG-4 codec
> may be used in combined. To adjust to these differences, we need to use
> different set of parameters, such as bitrate, image size, refresh rate,
> etc. at codec level anyway and different types of codec may be necessary.
> 
> >  I am coming from a world where content creators and
> > servers are very 'receiver-aware' and will not do anything
> > in the content creation that will crash receivers. In this
> > regard, the service providers who use the content want to
> > play this on as many receivers as possible and they are also
> > aware of the fact that receivers vary in their capabilities.
> > Service providers, on the other hand, will not send 2 separate
> > content streams to play on 2 receivers on the same wire as this
> > consumes precious bandwidth.
> >
> Considering above, having the same format ONLY at the RTP level may not
> help
> the situation so much. It may be useful for some kind of mobile terminals
> with Mbps bandwidth, error-free, large display size, but maybe it is not
> so
> usual case. Or, such powerful terminal can easily implement SL-based
> format.
> 
> >  2 examples of streams that one may want to play on receivers
> > that implement your draft are
> >
> > 1. A video ES that required CTS (in the RTP header) and DTS (which
> >    could be in the adaptation field). This may be needed for receivers
> >    that implement MPEG buffer models. This stream can play on
> >    receivers that implement your draft as it has a bigger buffer
> >    and does not need DTS, as long as your draft enabled these
> >    receivers to use the adaptation length field to offset where
> >    the video ES payload was picked up.
> > 2. One may author SL packetized video where there was information
> >    in the SL header and this will be in the adaptation field. In
> addition,
> >    the authors will make sure that the video ES in the SL packet
> >    would be independently decodable by receivers that implemented your
> >    draft. This will require the ability to parse the first byte in
> >    the adaptation header and use it to offset where the video ES
> >    payload would be picked up.
> >
> I think these streams are covered by SL-based payload. Why don't you use
> this?
> 
> >  If your draft can accommodate this simple extension, then it will
> enable
> > the receivers to play more content from the future and it will enable
> > service providers to keep this in mind when they author MPEG-4 based
> > content in the future.
> >
> >  Regards to your concerns,
> >
> > 1.  The overhead is 1 byte/RTP packet and this is a 6.25% addition
> >     to the 16 byte RTP header. If we sent 1 VOP per packet, then it
> >     is 30 bytes/sec (assuming 30 Hz video) or 240 bits/sec overhead.
> >     This is very minimal. The FEC proposal proposes the same type
> >     of overhead.
> 
> I'm not convinced by this explanation. A VOP may be fragmented into many
> RTP
> packets for error resiliency and/or low-delay requirements, and the size
> of
> the RTP packet is relatively small. The fixed RTP header is compressed by
> the header compression technique, but the extra field is not. The overhead
> is still a serious problem since this RTP payload format may be used in
> very
> low bitrate environment.
> 
> > 2.  The content authors will always make sure they follow all the
> >     other recommendations from your draft to make sure that the
> >     future content will play on receivers that implement the draft
> >     (such as sending data in decoding order, not splitting the headers
> >     etc). The only change from an implementation of today is the
> >     extra parsing of one byte before the ES payload to offset where
> >     the ES payload is picked up.
> 
> Is there any guaranty on this? There were many concerns raised by both
> IETF
> and MPEG experts on whether the ES payload receiver can decode SL payload
> packet correctly only by adding the extra byte fields. We must be care not
> to bring any danger to the implementers of ES payload format.
> 
> > 3.  The change to your draft would to be add 2 fields after the CSRC
> >     and before the video ES. These would be
> >     'adaptation length' and
> >     'length (if adaptation length is greater than 0)'.
> >     The semantics will say, the first byte after CSRC will contain the
> >     length of adaptation field. If this length value is zero, then the
> >     video payload will start in the next byte. Otherwise, the video
> >     payload will start after the number of bytes indicated in the
> >     adaptation length field. The content of the adaptation field after
> >     the length is private and left for future extensions.
> >
> Maybe I'm not a correct person to decide whether this change is small or
> not. But again, we must be care not to bring any danger to the
> implementers
> of ES payload format.
> 
> >  Jan Vandermeer may also have some suggested changes to the draft, if
> this
> > can be accommodated. Thank you for your kind patience.
> >
> We are not so stubborn to reject all changes if it is really useful and
> supported. Actually, many people helped us for the editorial clarification
> of the Internet-Draft. However, the proposal of adding extra byte has many
> serious problems:
> 
> - Clear disadvantage of the overhead.
> - No guaranty to work..
> - There are many concerns raised by both MPEG and IETF members.
> 
> How can we adopt such proposal?
> 
> Best regards,
> Yoshihiro Kikuchi
> 
> > -----Original Message-----
> > From: Yoshihiro Kikuchi [mailto:kiku@eel.rdc.toshiba.co.jp]
> > Sent: Thursday, September 14, 2000 12:38 PM
> > To: IETF AVT
> > Cc: Zvi Lifshitz; Colin Perkins; Yoshihiro Kikuchi
> > Subject: comments on in-band-signaling proposal
> >
> >
> > Dear all experts,
> >
> >
> > The authors of the Internet-Draft have discussed the issue of
> > in-band-signaling proposal. We understand the importance of
> > interoperability, however, we do not believe the specific proposal of an
> > extra byte in the RTP header is a good solution. It is our common and
> firm
> > desire that any proposal will not degrade the original intention,
> purpose
> > and functionality of the RTP payload format.
> >
> > Especially, the following points are important.
> >
> > o Overhead caused by an extra RTP header
> > The in-band-signaling proposal will cause at least one byte overhead per
> > each RTP packet. Since this RTP payload format may be used in very low
> > bitrate applications such as mobile and analog modem, the overhead is a
> > serious problem. On the other hand, the ways suggested by Colin,
> Stephan,
> > etc. of using different payload types will not cause any overhead
> problem.
> >
> > o Stability and reliability
> > As was pointed out by many, there is no guaranty that receivers of
> ES-based
> > RTP payload format can receive SL-based RTP packets correctly, even if
> the
> > extra byte in the RTP header is added. It is dangerous to adopt such
> > proposal at this moment without any guaranty. It is our strong desire
> that
> > only verified solution will be adopted in the RTP payload format
> > specification.
> >
> > o Schedule
> > There is no specific description of changes about the proposal, so far.
> It
> > is hard to accept such proposal. Even if the proposal has some value
> > (although we doubt it), it should not be too late to adopt it in
> > SL-based proposal, by keeping backword compatibility with ES-based RTP
> > payload format.
> >
> >
> > These points are very important to keep the original functionality of
> the
> > proposal. Constructive comments are welcome, however, we cannot accept
> > any proposals to harm the original intention and functionality. By this
> > point, it is hard to accept the in-band-signaling proposal which causes
> > overhead problem and which is not well verified. It should be remarked
> that
> > the alternatives proposed by IETF experts will not cause any such
> problems.
> >
> >
> > Kind regards,
> > Yoshihiro Kikuchi
> > and authors of the Intenet Draft
> >
> > ----
> >         Yoshihiro Kikuchi
> >
> > E-mail: kiku@eel.rdc.toshiba.co.jp
> > Corporate Research and Development Center
> > TOSHIBA Corporation
> > TEL: +81 +44 549 2288    FAX: +81 +44 520 1267
> >
> >
> >
> >
> 
> 
> 



From rem-conf Fri Sep 15 09:39:29 2000 
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Tmima,

I am just getting around to reading this thread myself, so a week is
really not a lot of time for people to consider this. I would like to
hear from others. Paolo? 

To summarize, the current l2tphc provides a 1, 2 or 3 byte header and
now we need to consider a 0 byte header. Bandwidth is precious, the more
flexible the header, the more difficult it is to switch. Our variable
length l2tp header is no joy to switch as is. That said, you know your
market better than I and if you have to get rid of this byte, even with
the processing cost, so be it. I have some new concerns about this first
byte in general that have come up during my most recent hard look at
this.

The first byte of the l2tphc header carries more than the P bit of
course, it carries at least one bit for future use and a few more bits
which are essentially static for all packets. These bits may be more
precious that we think. Given the number of suggestions for changing
this header and the ways it which it is being used that were never
dreamed of in the beginning (l2tphc was really designed for voluntary
tunnels across slow dialup links which have yet to be implemented, where
it *is* being deployed today is on ATM networks and voice applications!)
I would not dare to guess what will be next. Further, we may wish to
carry a uncompressed version of L2TP directly over IP at some point and
Ip protocol IDs are hard to come by. So, I think the static bits should
be put to better use, and a version field seems to be in order so that I
can sleep well at night.

This is what I would like to see in the first byte:

       0                   1                   2                   3
       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |T|P|I|J|x|x| ver |  Session ID...             |  PPP packet... |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

T,P,I,J are as defined in the l2tphc spec. ver is 1 (or 2 to match l2tp
if you like). 

As for the compression of the first byte via l2tphc, as long as one side
can always force a peer to include this byte if need be then I am OK
with its removal. This has to be a very strong MUST though as it is the
local box that knows if it is going to need this first byte to multiplex
l2tphc with other l2tp or l2tphc versions, not its peer. It should be
very explicit that this is the case (e.g. An l2tphc implementation MUST
be able to send the header format requested by its peer... and then
perhaps even an explanation of why).

More comments below on the mechanisms. Some of these concerns certainly
overlap with Andy's.

Tmima Koren wrote:
> 
> I suggest a few minor changes to the L2TPHC draft draft-ietf-l2tpext-l2tphc-02.txt
> to improve bandwidth utilization in special cases.
> 
> When the L2TPHC header is 1 byte long, actually only 1 bit is significant: the P (Priority) bit.
> 
> If all the packets in the tunnel will have the same priority, the L2TPHC byte is redundant.
> 
> We could add to the L2TPHC AVP one more byte (byte 9)
> We really need only 2 flag bits:
> 
> flag 1:  value 1 means: L2TPHC byte will be omitted
>             value 0 means: L2TPHC byte is there, get P (Priority) from the L2TPHC byte
> 
> flag 2:  value 1 means all packets have implied P=1
>             value 0 means all packets have implied P=0
> 

Since we now have potentially more information in the first byte,
perhaps you should compress it away in a general manner. A new AVP which
defines what the static byte should look like would accomplish this.
e.g. something like this in the AVP:

       0 1 2 3 4 5 6 7 8 
      +-+-+-+-+-+-+-+-+-+
      |T|P|I|J|x|x| ver |
      +-+-+-+-+-+-+-+-+-+

...which sets the proper bits that are "always on" for this session
(and, really, the entire box!)

I would include cautionary text about setting the P bit for all packets
as Andy pointed out (perhaps even stating that it MUST be 0), as well as
the loss of functionality for prioritizing LCP echos in general. 

> consult flag 2 only when flag 1 is 1
> 
> If the packet in the L2TPHC tunnel is always PPPMUX, there is another useful flag:
> 
> flag 3: value 1 means 'PPPMUX only'
>            value 0 means: any PPP packet
> 
> When flags 1 and 3 are on (=1), both L2TPHC header and PPPMUX protocol id can be omitted
> PPP packets that have a different priority or that are not PPPMUX can still be sent in the tunnel using the UDP/L2TP format, same as the L2TP control packets

This actually falls nicely in the new Service Type model. As soon as
Suhail and Danny finish this up (its my fault, I keep sending it back to
them with corrections) then you will have a framework to identify a new
Service Type for TCRTP/PPPMUX. PPPMUX at least is really a different
framing method and protocol for l2tp so it should be treated as such.
You will need to write a draft identifying this new service type. Nice
thing is that you get l2tp or l2tphc functionality to boot. It also
provides a good place to talk about any pppmux/rtp/l2tphc concerns, as
well as a vehicle for idenifying any specific new avps you might need
for the session that you have not thought of yet.

> 
> TCRTP (Tunneled Compressed multiplexed RTP) can benefit from these changes: the tunneling overhead will be reduced by 2 bytes.

Some of the lengths (no pun intended) we go to for a byte or two still
amaze me. 

> 
> Tmima



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ANY country, including of course, to anywhere in the USA!    They offer a
FREE 1 yr. subscription to your choice of over 200 of the titles in their
catalogue to any new customer using them for the first time.       The 
dollar value of the freebies, based on the subscription prices directly
>from the publishers, ranges from $6.97 all the way up to $50.00!

For new customers in the USA, there is no charge for FPH (foreign postage &
handling), so the freebie is 100% free!   For new customers living
overseas, the only charge on the freebie would be for the FPH (foreign
postage & handling).

Their president has been in the magazine subscription business since 1973
and they are very customer-service oriented.   They will even help you with
address changes on your magazines, even if you move from one country to
another country.   They have thousands of happy customers in over 59
countries.

Their price guarantee is very simple:       they guarantee that their
subscription prices are the lowest available and they will BEAT any
legitimate, verifiable offer before you pay them or match it afterwards, by
refunding you the difference in price PLUS the cost of the postage stamp
you would use sending in the special offer to them, even 6 months after you
pay them, as long as it was current at the time of your offer.    Does that
sound fair?       Wouldn't it be great if everything you bought came with
that price guarantee?  

Sometimes they are less than half of the next best deal out there,
sometimes just a little cheaper, but always you get the lowest rates
without having to shop around.     With 1,100+ titles on their list, they
would like to think that they have also the best selection around!

Within the USA, for their USA customers, they are cheaper than all their
competitors and even the publishers themselves.  This is their price
guarantee.         The 1 yr. freebie that you get with your first order is
completely free!   

Overseas, (even after you factor in the cost of the FPH (foreign postage &
handling) and the conversion from USA Dollars to your currency), on the
average, they are generally around one-fourth to one-half of what the
newsstands overseas charge locally for USA magazines.  On some titles they
are as little as one-tenth of what the newsstands charge.  They are also
the cheapest subscription source for delivery overseas, including directly
>from the publishers themselves!   Some publishers don't even offer
subscriptions overseas.........but overseas subscriptions are this
company's specialty!  They feel that magazines should not be a luxury
overseas.   In the USA, people buy magazines and then toss them after
reading them for just a few minutes or hours.  They are so cheap in the
USA!   Well, this company would like to make it the same way for their
overseas customers.  They are also cheaper than all their competitors in
the USA and overseas, including the publishers themselves!   It is also
*highly unlikely* you will find any of their USA competitors calling you
overseas, in order to offer that personal touch, just to sell you a couple
of magazines!  But that is what this company specializes in and loves
doing!     Around one-half their business comes from overseas, so they are
very patient with new customers who only speak limited English as a 2nd
language.    Subscription prices quoted for overseas consist of the
subscription price, plus the FPH.   You add the two together and that is
your total cost.   The exception is the 1 yr. freebie you get with your
first order.   On that title, you pay *only* the FPH for the 1 yr. term.

Their prices are so cheap because when you deal with them, you cut-out all
the middlemen.


HERE IS HOW YOU CAN GET MORE INFO AND GET STARTED WITH THEM:

Simply email or fax or smail back to us the reply form listed at the top of
this message.   We will then forward your form on to the subscription
agency.  They will then email their "big and juicy" catalogue to you, in
whichever of the two formats you chose.   The catalogue is FREE and makes
for hours of fascinating reading, on its own. It includes the complete list
of freebies, a complete list of all the titles they sell, as well as
detailed descriptions on most of the titles, along with lists of titles by
category of interest and their terms of sale.    

They will then give you a friendly, no-pressure, no obligation, 5-minute
call to go over how they work and to answer any questions that you might
have, as well as give you up-to-the minute price quotes on any titles you
might be considering.     They will call you in whatever country you live
in, taking the time difference into account.        As they like to
emphasize the personal touch they give to each new customer, all first-time
orders can only be done via phone, so they can answer all your questions
completely and personally.   Once you have placed your first order via
phone, you will be able to place future orders and make inquiries on your
account, get price quotes, etc., all via email, if that is most convenient
for you.

Within the USA, they accept payment via check over the phone, Mastercard,
Visa, American Express, Diner's Club and Carte Blanche.    Overseas, they
accept Mastercard, Visa, American Express, Diner's Club and Carte Blanche,
even if your credit card is a local one in local currency (that most
merchants in the USA would not normally be willing to accept).

That's our introduction of our client that we represent.   We hope that we
have piqued your interest and that you will take the next step to get their
free catalogue!   Thank you for your time and interest.

--
Tempting Tear-Outs.
For more info on marketing & consulting rates, please write us on your
company letterhead, w/business card, via smail to:   Tempting Tear-Outs,
PMB 200, 3835 Richmond Ave., Staten Island NY  10312-3828, USA.     

This email message has been sent to you by:  Tempting Tear-Outs, PMB 200,
3835 Richmond Ave., Staten Island NY  10312-3828, USA.









































































































From rem-conf Sat Sep 16 03:50:56 2000 
From rem-conf-request@es.net Sat Sep 16 03:50:53 2000
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Cc: <ZviL@optibase.com>, <csp@east.isi.edu>,
        <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: comments on in-band-signaling proposal
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This one did not came through yesterday, due to network problems
Sam, Kikuchi-san, others
While I am personally confident that it is possible to define a feasibl=
e and
good solution, I share some of the concern of Kikuchi-san. Within MPEG =
there
are far too many misunderstandings and disagreements to approve anythin=
g at
this time. MPEG clearly needs more time to identify the most desirable
approach.

Therefore I would like to suggest the following way forward :

1) The IETF proceeds with the Kikuchi-san draft without implementing th=
e
in-band signaling proposal so as to meet the ITU deadline with minimum =
risk.
2) MPEG continues to consider the most desirable format for MPEG-4 ES a=
nd
other MPEG-4 streams.
3) In the La Baule meeting MPEG concludes its discussion by producing i=
ts
"generic" MPEG-4 over IP draft.
4) If MPEG at the La Baule meeting is confident about the feasibility a=
nd
usefulness of a forward and backward compatible solution using the
Kikuchi-san draft with in-band signaling, then MPEG will request IETF t=
o
start the publication process of an "update" of the Kikuchi-san draft, =
that
will obsolete the current one.
5) In case of such "update", MPEG commits to neither change the "update=
" of
the Kikuchi-san draft, nor elements in the "generic MPEG-4 A/V ES" draf=
t
that impact the forward and backward compatibility with the updated
Kikuchi-san draft.

Kind regards,

Jan

-----Original Message-----
From:	SNarasimhan@gi.com [mailto:SNarasimhan@gi.com]
Sent:	vrijdag, 15 september, 2000 02:15
To:	Jan vanderMeer; rem-conf@es.net; kiku@eel.rdc.toshiba.co.jp
Cc:	4on2andIP-sys@advent.ee.columbia.edu; csp@east.isi.edu;
ZviL@optibase.com
Subject:	RE: comments on in-band-signaling proposal


Dear Kikuchi-San and experts,

 I am coming from a world where content creators and
servers are very 'receiver-aware' and will not do anything
in the content creation that will crash receivers. In this
regard, the service providers who use the content want to
play this on as many receivers as possible and they are also
aware of the fact that receivers vary in their capabilities.
Service providers, on the other hand, will not send 2 separate
content streams to play on 2 receivers on the same wire as this
consumes precious bandwidth.

 2 examples of streams that one may want to play on receivers
that implement your draft are

1. A video ES that required CTS (in the RTP header) and DTS (which
   could be in the adaptation field). This may be needed for receivers
   that implement MPEG buffer models. This stream can play on
   receivers that implement your draft as it has a bigger buffer
   and does not need DTS, as long as your draft enabled these
   receivers to use the adaptation length field to offset where
   the video ES payload was picked up.
2. One may author SL packetized video where there was information
   in the SL header and this will be in the adaptation field. In additi=
on,
   the authors will make sure that the video ES in the SL packet
   would be independently decodable by receivers that implemented your
   draft. This will require the ability to parse the first byte in
   the adaptation header and use it to offset where the video ES
   payload would be picked up.

 If your draft can accommodate this simple extension, then it will enab=
le
the receivers to play more content from the future and it will enable
service providers to keep this in mind when they author MPEG-4 based
content in the future.

 Regards to your concerns,

1.  The overhead is 1 byte/RTP packet and this is a 6.25% addition
    to the 16 byte RTP header. If we sent 1 VOP per packet, then it
    is 30 bytes/sec (assuming 30 Hz video) or 240 bits/sec overhead.
    This is very minimal. The FEC proposal proposes the same type
    of overhead.
2.  The content authors will always make sure they follow all the
    other recommendations from your draft to make sure that the
    future content will play on receivers that implement the draft
    (such as sending data in decoding order, not splitting the headers
    etc). The only change from an implementation of today is the
    extra parsing of one byte before the ES payload to offset where
    the ES payload is picked up.
3.  The change to your draft would to be add 2 fields after the CSRC
    and before the video ES. These would be
    'adaptation length' and
    'length (if adaptation length is greater than 0)'.
    The semantics will say, the first byte after CSRC will contain the
    length of adaptation field. If this length value is zero, then the
    video payload will start in the next byte. Otherwise, the video
    payload will start after the number of bytes indicated in the
    adaptation length field. The content of the adaptation field after
    the length is private and left for future extensions.

 Jan Vandermeer may also have some suggested changes to the draft, if t=
his
can be accommodated. Thank you for your kind patience.

Best Regards
Sam Narasimhan
Motorola

-----Original Message-----
From: Yoshihiro Kikuchi [mailto:kiku@eel.rdc.toshiba.co.jp]
Sent: Thursday, September 14, 2000 12:38 PM
To: IETF AVT
Cc: Zvi Lifshitz; Colin Perkins; Yoshihiro Kikuchi
Subject: comments on in-band-signaling proposal


Dear all experts,


The authors of the Internet-Draft have discussed the issue of
in-band-signaling proposal. We understand the importance of
interoperability, however, we do not believe the specific proposal of a=
n
extra byte in the RTP header is a good solution. It is our common and f=
irm
desire that any proposal will not degrade the original intention, purpo=
se
and functionality of the RTP payload format.

Especially, the following points are important.

o Overhead caused by an extra RTP header
The in-band-signaling proposal will cause at least one byte overhead pe=
r
each RTP packet. Since this RTP payload format may be used in very low
bitrate applications such as mobile and analog modem, the overhead is a=

serious problem. On the other hand, the ways suggested by Colin, Stepha=
n,
etc. of using different payload types will not cause any overhead probl=
em.

o Stability and reliability
As was pointed out by many, there is no guaranty that receivers of ES-b=
ased
RTP payload format can receive SL-based RTP packets correctly, even if =
the
extra byte in the RTP header is added. It is dangerous to adopt such
proposal at this moment without any guaranty. It is our strong desire t=
hat
only verified solution will be adopted in the RTP payload format
specification.

o Schedule
There is no specific description of changes about the proposal, so far.=
 It
is hard to accept such proposal. Even if the proposal has some value
(although we doubt it), it should not be too late to adopt it in
SL-based proposal, by keeping backword compatibility with ES-based RTP
payload format.


These points are very important to keep the original functionality of t=
he
proposal. Constructive comments are welcome, however, we cannot accept
any proposals to harm the original intention and functionality. By this=

point, it is hard to accept the in-band-signaling proposal which causes=

overhead problem and which is not well verified. It should be remarked =
that
the alternatives proposed by IETF experts will not cause any such probl=
ems.


Kind regards,
Yoshihiro Kikuchi
and authors of the Intenet Draft

----
        Yoshihiro Kikuchi

E-mail: kiku@eel.rdc.toshiba.co.jp
Corporate Research and Development Center
TOSHIBA Corporation
TEL: +81 +44 549 2288    FAX: +81 +44 520 1267


=



From rem-conf Sat Sep 16 08:57:14 2000 
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To: "W. Mark Townsley" <townsley@cisco.com>
cc: Tmima Koren <tmima@cisco.com>, brucet@cisco.com, dwing@cisco.com,
   Stephen Casner <casner@packetdesign.com>, l2tp@diameter.org,
   rem-conf@es.net
Subject: Re: 0 byte L2TPHC header and implied PPPMUX 
In-reply-to: Your message of "Fri, 15 Sep 2000 12:13:28 EDT."
             <39C24AA8.12666180@cisco.com> 
Date: Sat, 16 Sep 2000 08:46:01 -0700
From: Andy Valencia <vandys@zendo.com>
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["W. Mark Townsley" <townsley@cisco.com> writes:]

>I would not dare to guess what will be next. Further, we may wish to
>carry a uncompressed version of L2TP directly over IP at some point and
>Ip protocol IDs are hard to come by. So, I think the static bits should
>be put to better use, and a version field seems to be in order so that I
>can sleep well at night.

But but but... the approach I was trying to take was the version fields
(being invariant) would belong in the L2TPHC *AVP*'s.  So version--like any
other negotiation-phase parameter--do not take bits from the payload.

>This is what I would like to see in the first byte:
>       0                   1                   2                   3
>       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2
>      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>      |T|P|I|J|x|x| ver |  Session ID...             |  PPP packet... |
>      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Is it just me, or is that a 9-bit byte? :->

>T,P,I,J are as defined in the l2tphc spec. ver is 1 (or 2 to match l2tp
>if you like). 

If you want a version field for L2TPHC, let me define Yet Another AVP!

>As for the compression of the first byte via l2tphc, as long as one side
>can always force a peer to include this byte if need be then I am OK
>with its removal. This has to be a very strong MUST though as it is the
>local box that knows if it is going to need this first byte to multiplex
>l2tphc with other l2tp or l2tphc versions, not its peer. It should be
>very explicit that this is the case (e.g. An l2tphc implementation MUST
>be able to send the header format requested by its peer... and then
>perhaps even an explanation of why).

Understood.  I can be sure to strengthen that verbage.

>Since we now have potentially more information in the first byte,
>perhaps you should compress it away in a general manner. A new AVP which
>defines what the static byte should look like would accomplish this.
>e.g. something like this in the AVP:
>       0 1 2 3 4 5 6 7 8 
>      +-+-+-+-+-+-+-+-+-+
>      |T|P|I|J|x|x| ver |
>      +-+-+-+-+-+-+-+-+-+

Do you want the AVP to say "This is a version 0 format L2TPHC header"?  Or
do you actually want a general bit assignment description (bleh)?

Andy



From rem-conf Sat Sep 16 11:38:16 2000 
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From: "W. Mark Townsley" <townsley@cisco.com>
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        Stephen Casner <casner@packetdesign.com>, l2tp@diameter.org,
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Subject: Re: 0 byte L2TPHC header and implied PPPMUX
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Andy Valencia wrote:
> 
> ["W. Mark Townsley" <townsley@cisco.com> writes:]
> 
> >I would not dare to guess what will be next. Further, we may wish to
> >carry a uncompressed version of L2TP directly over IP at some point and
> >Ip protocol IDs are hard to come by. So, I think the static bits should
> >be put to better use, and a version field seems to be in order so that I
> >can sleep well at night.
> 
> But but but... the approach I was trying to take was the version fields
> (being invariant) would belong in the L2TPHC *AVP*'s.  So version--like any
> other negotiation-phase parameter--do not take bits from the payload.

Unless we have another l2tp header variant come along that we would like
to carry on protocol ID 115. l2tphc only leaves one bit to make the
distinction. Utilizing AVPs to determine the l2tp version assumes that
you can read the session ID, lookup state, and identify the l2tp version
for that packet. This is fine when the session id is in a known location
in the header, but if I have a completely different header from that
defined in l2tphc I am out of luck. 

Since you are defining a completely new header anyway, I don't see much
benefit it not allowing the Offset, Sequence and Length bits to be put
to good use (or at least reserved for good use). Really, the ver field
is no different than using the I and J bits to identify presence of the
one or two byte session ID that l2tphc does now. We could simply leave
the bits reserved or call them something else if you like.

> 
> >This is what I would like to see in the first byte:
> >       0                   1                   2                   3
> >       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2
> >      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
> >      |T|P|I|J|x|x| ver |  Session ID...             |  PPP packet... |
> >      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
> 
> Is it just me, or is that a 9-bit byte? :->

Whoops. Just remove one of the x's here.

> 
> >T,P,I,J are as defined in the l2tphc spec. ver is 1 (or 2 to match l2tp
> >if you like).
> 
> If you want a version field for L2TPHC, let me define Yet Another AVP!
> 
> >As for the compression of the first byte via l2tphc, as long as one side
> >can always force a peer to include this byte if need be then I am OK
> >with its removal. This has to be a very strong MUST though as it is the
> >local box that knows if it is going to need this first byte to multiplex
> >l2tphc with other l2tp or l2tphc versions, not its peer. It should be
> >very explicit that this is the case (e.g. An l2tphc implementation MUST
> >be able to send the header format requested by its peer... and then
> >perhaps even an explanation of why).
> 
> Understood.  I can be sure to strengthen that verbage.
> 
> >Since we now have potentially more information in the first byte,
> >perhaps you should compress it away in a general manner. A new AVP which
> >defines what the static byte should look like would accomplish this.
> >e.g. something like this in the AVP:
> >       0 1 2 3 4 5 6 7 8
> >      +-+-+-+-+-+-+-+-+-+
> >      |T|P|I|J|x|x| ver |
> >      +-+-+-+-+-+-+-+-+-+
> 
> Do you want the AVP to say "This is a version 0 format L2TPHC header"?  Or
> do you actually want a general bit assignment description (bleh)?
> 
> Andy



From rem-conf Sun Sep 17 02:18:18 2000 
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From: Zvi Lifshitz <zvil@csi.com>
To: "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>,
        "kiku@eel.rdc.toshiba.co.jp" <kiku@eel.rdc.toshiba.co.jp>,
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	 <csp@east.isi.edu>,
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Subject: RE: comments on in-band-signaling proposal
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With the exception that I didn't understand 5, this is ok with me.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				GSM: +972(54)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October 10th-12th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Saturday, September 16, 2000 12:47 PM
To:	kiku@eel.rdc.toshiba.co.jp; rem-conf@es.net; SNarasimhan@gi.com
Cc:	ZviL@optibase.com; csp@east.isi.edu; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: comments on in-band-signaling proposal

 
This one did not came through yesterday, due to network problems
Sam, Kikuchi-san, others
While I am personally confident that it is possible to define a feasible and
good solution, I share some of the concern of Kikuchi-san. Within MPEG there
are far too many misunderstandings and disagreements to approve anything at
this time. MPEG clearly needs more time to identify the most desirable
approach.

Therefore I would like to suggest the following way forward :

1) The IETF proceeds with the Kikuchi-san draft without implementing the
in-band signaling proposal so as to meet the ITU deadline with minimum risk.
2) MPEG continues to consider the most desirable format for MPEG-4 ES and
other MPEG-4 streams.
3) In the La Baule meeting MPEG concludes its discussion by producing its
"generic" MPEG-4 over IP draft.
4) If MPEG at the La Baule meeting is confident about the feasibility and
usefulness of a forward and backward compatible solution using the
Kikuchi-san draft with in-band signaling, then MPEG will request IETF to
start the publication process of an "update" of the Kikuchi-san draft, that
will obsolete the current one.
5) In case of such "update", MPEG commits to neither change the "update" of
the Kikuchi-san draft, nor elements in the "generic MPEG-4 A/V ES" draft
that impact the forward and backward compatibility with the updated
Kikuchi-san draft.

Kind regards,

Jan




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International Federation for Information Processing 
CMS 2001 
Communications and Multimedia Security 
Joint working conference IFIP TC6 and TC11 
May 21-22, 2001 
GMD Darmstadt, Germany 

GMD-German National Research Center for Information Technology, 
Institute IPSI - Dept. Mobile Interactive Media Darmstadt, Germany 
and 
Darmstadt Univ. of Technology, Industrial Process and System Communications, 
Dept.of Electrical Engineering & Information Technology 

--------------------------------------------------------------------------------

Call For Papers
GOALS and TOPICS of INTEREST 

CMS 2001 is the sixth working conference on Communications and Multimedia Security since 1995. State-of-the-art issues as well as practical experiences and new trends in the areas will be the topics of interest again, as proven by preceding conferences. 

This year the organisers want to focus the attention on watermarking and copyright protection for e-commerce applications and multimedia data. We want to evaluate the progress of digital watermarking, the robustness and the practical usage for authentication and also for integrity checks. 

Topics of interest include, but are not limited to 

Applied cryptography 
Biometry 
Combined multimedia security 
Communications systems security 
Cryptography  steganography  watermarking aspects 
Digital watermarking applications 
Digital watermarking technologies 
Electronic commerce and digital signatures 
Internet, intranet and extranet security 
Legal, social and ethical aspects of communication systems security 
Mobile communications security 
Multimedia systems security 
Possible attacks on multimedia systems 
SUBMISSION DETAILS 
Authors are strongly encouraged to submit their papers electronically. 

Please email your submission in postscript or pdf format to: cms2001@darmstadt.gmd.de 

Electronic submissions must be received by December 4, 2000. 

Authors unable to submit electronically are invited to contact us (cms2001@darmstadt.gmd.de) or send the papers to the  postal address below. 

Authors are asked to submit original papers only. Papers that have previously been published and papers that are currently being considered for publication by another journal or conference are not eligible. All submitted papers will be refereed for correctness, originality, relevance to the conference, and quality of presentation. 

All submissions must be in English. The paper must be anonymous, with no author names, affiliations, acknowledgements, or obvious references. It should begin with a title, a short abstract, and a list of key words, and its introduction should summarise the contributions of the paper at a level appropriate for a non-specialist reader. The paper should be at most 5000 words long. A full page figure is 500 words. 

All submitted papers will be refereed by at least three members of the International Program Committee according to the standard blind refereeing procedures. The Conference Proceedings will be published by an international publisher; copies of the proceedings will be available at the conference. 

Notification of acceptance or rejection will be sent to authors by 15th of January. Authors of accepted papers must guarantee that their paper will be presented at the conference. 



--------------------------------------------------------------------------------

Important dates : 
Submission Deadline: December 4, 2000 

Notification: January 15, 2001 

Final camera-ready version: February 28, 2001 

Conference: May 20-21, 2001 

To submit a paper, or for further details, please contact: 

Yvonne Sobon 

German National Research Center for Information Technology 

Institute IPSI - Dept. Mobile Interactive Media 

Dolivostr.15 

D-64293 Darmstadt 

Germany 

tel: +49-6151-869-847 | fax:-6847 

cms2001@darmstadt.gmd.de 
  

For further details see http://www.darmstadt.gmd.de/mobile/cms/start.html 



--------------------------------------------------------------------------------

Program Committee Chair: 

Chair: R. Steinmetz 

Co-Chair: J. Dittmann 

Program Committee: 

I. Cox, NEC Research Institut, USA 
E. Delp, Purdue University, USA 
J. Fridrich, Center for Intelligent Systems SUNY Binghamton, USA 
D. Gollmann, Microsoft Research, UK 
R. Grimm, GMD SIT, Germany 
P. Horster, Universitaet Klagenfurt, Austria 
T. Kalker, Philips Research Eindhoven, The Netherlands 
K. Keus, BSI, Germany 
P. Kraaibeek, ConSecur, Germany 
D. Kundur, University of Toronto, Canada 
N. Memon, Polytechnic University Brooklyn, USA 
K. Nahrstedt, University of Illinois at Urbana-Champaign, USA 
G. Pernul, University of Essen, Germany 
F. Petitcolas, Microsoft, UK 
B. Preneel, Katholieke Universiteit Leuven, Belgium 
C. Schmidt, Software Professional GmbH & Co. KG, Germany 
J. Schwenk, Telekom, Germany 
A. Tirkel, Scientific Technology, Australia 
P. Wohlmacher, Universitaet Klagenfurt, Austria 
R. Zuccherato, Entrust Technologies, Canada 

Organising Committee Chair: 

Chair: J. Dittmann 

Co-Chair: M. Steinebach 

Organising Committee: 

Yvonne Sobon 

Main Organiser: 
IFIP TC 11 and TC 6 



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Message-ID: <008f01c0215c$b1ebc3a0$cc51c485@eel.rdc.toshiba.co.jp>
From: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
To: <jan.vandermeer@philips.com>, <rem-conf@es.net>, <SNarasimhan@gi.com>
Cc: <ZviL@optibase.com>, <csp@east.isi.edu>,
        <4on2andIP-sys@advent.ee.columbia.edu>,
        "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
References: <0056890014123034000002L942*@MHS>
Subject: Re: comments on in-band-signaling proposal
Date: Mon, 18 Sep 2000 19:39:16 +0900
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Dear Jan and all,

Following suggestion 1, the authors will produce an updated Internet-Draft
of MPEG-4 Audio/Visual RTP soon with reflecting comments raised and agreed
during the WG last call period. It does not included in-band signaling
proposal.

It is too early for us to comment on suggestions 2-5, because we have not
seen detailed proposal. In any case, we cannot accept any disadvantage
(overhead, efficiency, risk, documentation, ...) compared to the current
spec. We should also be care that we will not fall into "an MPEG only
solution".

But if we can keep on these, maybe we will have a good solution in the end.

Kind regards,
Yoshihiro Kikuchi

----- Original Message -----
From: <jan.vandermeer@philips.com>
To: <kiku@eel.rdc.toshiba.co.jp>; <rem-conf@es.net>; <SNarasimhan@gi.com>
Cc: <ZviL@optibase.com>; <csp@east.isi.edu>;
<4on2andIP-sys@advent.ee.columbia.edu>
Sent: Saturday, September 16, 2000 7:47 PM
Subject: RE: comments on in-band-signaling proposal


> This one did not came through yesterday, due to network problems
> Sam, Kikuchi-san, others
> While I am personally confident that it is possible to define a feasible
and
> good solution, I share some of the concern of Kikuchi-san. Within MPEG
there
> are far too many misunderstandings and disagreements to approve anything
at
> this time. MPEG clearly needs more time to identify the most desirable
> approach.
>
> Therefore I would like to suggest the following way forward :
>
> 1) The IETF proceeds with the Kikuchi-san draft without implementing the
> in-band signaling proposal so as to meet the ITU deadline with minimum
risk.
> 2) MPEG continues to consider the most desirable format for MPEG-4 ES and
> other MPEG-4 streams.
> 3) In the La Baule meeting MPEG concludes its discussion by producing its
> "generic" MPEG-4 over IP draft.
> 4) If MPEG at the La Baule meeting is confident about the feasibility and
> usefulness of a forward and backward compatible solution using the
> Kikuchi-san draft with in-band signaling, then MPEG will request IETF to
> start the publication process of an "update" of the Kikuchi-san draft,
that
> will obsolete the current one.
> 5) In case of such "update", MPEG commits to neither change the "update"
of
> the Kikuchi-san draft, nor elements in the "generic MPEG-4 A/V ES" draft
> that impact the forward and backward compatibility with the updated
> Kikuchi-san draft.
>
> Kind regards,
>
> Jan
>
> -----Original Message-----
> From: SNarasimhan@gi.com [mailto:SNarasimhan@gi.com]
> Sent: vrijdag, 15 september, 2000 02:15
> To: Jan vanderMeer; rem-conf@es.net; kiku@eel.rdc.toshiba.co.jp
> Cc: 4on2andIP-sys@advent.ee.columbia.edu; csp@east.isi.edu;
> ZviL@optibase.com
> Subject: RE: comments on in-band-signaling proposal
>
>
> Dear Kikuchi-San and experts,
>
>  I am coming from a world where content creators and
> servers are very 'receiver-aware' and will not do anything
> in the content creation that will crash receivers. In this
> regard, the service providers who use the content want to
> play this on as many receivers as possible and they are also
> aware of the fact that receivers vary in their capabilities.
> Service providers, on the other hand, will not send 2 separate
> content streams to play on 2 receivers on the same wire as this
> consumes precious bandwidth.
>
>  2 examples of streams that one may want to play on receivers
> that implement your draft are
>
> 1. A video ES that required CTS (in the RTP header) and DTS (which
>    could be in the adaptation field). This may be needed for receivers
>    that implement MPEG buffer models. This stream can play on
>    receivers that implement your draft as it has a bigger buffer
>    and does not need DTS, as long as your draft enabled these
>    receivers to use the adaptation length field to offset where
>    the video ES payload was picked up.
> 2. One may author SL packetized video where there was information
>    in the SL header and this will be in the adaptation field. In addition,
>    the authors will make sure that the video ES in the SL packet
>    would be independently decodable by receivers that implemented your
>    draft. This will require the ability to parse the first byte in
>    the adaptation header and use it to offset where the video ES
>    payload would be picked up.
>
>  If your draft can accommodate this simple extension, then it will enable
> the receivers to play more content from the future and it will enable
> service providers to keep this in mind when they author MPEG-4 based
> content in the future.
>
>  Regards to your concerns,
>
> 1.  The overhead is 1 byte/RTP packet and this is a 6.25% addition
>     to the 16 byte RTP header. If we sent 1 VOP per packet, then it
>     is 30 bytes/sec (assuming 30 Hz video) or 240 bits/sec overhead.
>     This is very minimal. The FEC proposal proposes the same type
>     of overhead.
> 2.  The content authors will always make sure they follow all the
>     other recommendations from your draft to make sure that the
>     future content will play on receivers that implement the draft
>     (such as sending data in decoding order, not splitting the headers
>     etc). The only change from an implementation of today is the
>     extra parsing of one byte before the ES payload to offset where
>     the ES payload is picked up.
> 3.  The change to your draft would to be add 2 fields after the CSRC
>     and before the video ES. These would be
>     'adaptation length' and
>     'length (if adaptation length is greater than 0)'.
>     The semantics will say, the first byte after CSRC will contain the
>     length of adaptation field. If this length value is zero, then the
>     video payload will start in the next byte. Otherwise, the video
>     payload will start after the number of bytes indicated in the
>     adaptation length field. The content of the adaptation field after
>     the length is private and left for future extensions.
>
>  Jan Vandermeer may also have some suggested changes to the draft, if this
> can be accommodated. Thank you for your kind patience.
>
> Best Regards
> Sam Narasimhan
> Motorola
>
> -----Original Message-----
> From: Yoshihiro Kikuchi [mailto:kiku@eel.rdc.toshiba.co.jp]
> Sent: Thursday, September 14, 2000 12:38 PM
> To: IETF AVT
> Cc: Zvi Lifshitz; Colin Perkins; Yoshihiro Kikuchi
> Subject: comments on in-band-signaling proposal
>
>
> Dear all experts,
>
>
> The authors of the Internet-Draft have discussed the issue of
> in-band-signaling proposal. We understand the importance of
> interoperability, however, we do not believe the specific proposal of an
> extra byte in the RTP header is a good solution. It is our common and firm
> desire that any proposal will not degrade the original intention, purpose
> and functionality of the RTP payload format.
>
> Especially, the following points are important.
>
> o Overhead caused by an extra RTP header
> The in-band-signaling proposal will cause at least one byte overhead per
> each RTP packet. Since this RTP payload format may be used in very low
> bitrate applications such as mobile and analog modem, the overhead is a
> serious problem. On the other hand, the ways suggested by Colin, Stephan,
> etc. of using different payload types will not cause any overhead problem.
>
> o Stability and reliability
> As was pointed out by many, there is no guaranty that receivers of
ES-based
> RTP payload format can receive SL-based RTP packets correctly, even if the
> extra byte in the RTP header is added. It is dangerous to adopt such
> proposal at this moment without any guaranty. It is our strong desire that
> only verified solution will be adopted in the RTP payload format
> specification.
>
> o Schedule
> There is no specific description of changes about the proposal, so far. It
> is hard to accept such proposal. Even if the proposal has some value
> (although we doubt it), it should not be too late to adopt it in
> SL-based proposal, by keeping backword compatibility with ES-based RTP
> payload format.
>
>
> These points are very important to keep the original functionality of the
> proposal. Constructive comments are welcome, however, we cannot accept
> any proposals to harm the original intention and functionality. By this
> point, it is hard to accept the in-band-signaling proposal which causes
> overhead problem and which is not well verified. It should be remarked
that
> the alternatives proposed by IETF experts will not cause any such
problems.
>
>
> Kind regards,
> Yoshihiro Kikuchi
> and authors of the Intenet Draft
>
> ----
>         Yoshihiro Kikuchi
>
> E-mail: kiku@eel.rdc.toshiba.co.jp
> Corporate Research and Development Center
> TOSHIBA Corporation
> TEL: +81 +44 549 2288    FAX: +81 +44 520 1267
>
>
>




From rem-conf Mon Sep 18 05:31:08 2000 
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Message-ID: <01c101c0216b$ac58ec60$cc51c485@eel.rdc.toshiba.co.jp>
From: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
To: "IETF AVT" <rem-conf@es.net>,
        "4on2andIP-sys" <4on2andIP-sys@advent.ee.columbia.edu>
Cc: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
Subject: New I/D submitted for MPEG-4 A/V RTP
Date: Mon, 18 Sep 2000 21:26:30 +0900
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Dear all,

I have submitted a new version (-04) of the internet draft of MPEG-4
Audio/Visual RTP payload format to reflect comments raised and agreed during
the WG last call period. The document should be available soon at the
internet draft archive.

The following is the summary of changes from the previous version (-03).

1) Default value of MIME type
2) Editorial update of abstract and introduction
3) Clarification related to multiple VOPs concatenation (timestamp,
fragmentation rule 4)
4) LATM related modification (section 1.2, 4.2)
5) Some other editorial clarifications


The authors would like to thank for those who raised comments.

Best regards,
Yoshihiro Kikuchi

----
        Yoshihiro Kikuchi

E-mail: kiku@eel.rdc.toshiba.co.jp
Corporate Research and Development Center
TOSHIBA Corporation
TEL: +81 +44 549 2288    FAX: +81 +44 520 1267







From rem-conf Mon Sep 18 07:27:54 2000 
From rem-conf-request@es.net Mon Sep 18 07:27:52 2000
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	id 13b1od-0001jW-00; Mon, 18 Sep 2000 07:22:40 -0700
From:     998766financebuyout@888.nu
To:       
Subject:  ___We have foreigners who want to buy or finance your business/speak to them right now
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As per your inquiry.................phone calls only.....

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 We have foreigners who want to buy or finance your business/speak to them right now... 





From rem-conf Mon Sep 18 14:36:29 2000 
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 <8D0FCE51EA3DD411B8D80004AC4CCB5B132A10@l-rmhs.lannion.cnet.fr>
References: <8D0FCE51EA3DD411B8D80004AC4CCB5B132A10@l-rmhs.lannion.cnet.fr>
Date: Mon, 18 Sep 2000 14:30:38 -0700
To: CURET Dominique FTRD/DMI/REN <dominique.curet@rd.francetelecom.fr>
From: Dave Singer <singer@apple.com>
Subject: Re: Pre-MPEG meeting for the '4onIP' AHG
Cc: 4on2andIP-sys@advent.ee.columbia.edu, rem-conf@es.net
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At 6:01 PM +0200 9/13/00, CURET Dominique FTRD/DMI/REN wrote:
>Dear all, dear Young-Kwon,
>
>we are supposed to meet on Saturday(21/10) and Sunday (22/10)
>before the MPEG meeting in La Baule.
>
>Can we say something like:
>      Saturday from 14 to 2000
>      Sunday from 0900 to 1800
>Or only the Sunday from 0900 to 1800
>
>comments


whoa!  I'm completely snowed by the mail on this...and read as fast 
as I can, I am not catching up.  hence no comment from me yet!

I'll be in La Baule via SFO, JFK, LHR, CDG, CDG TGV, Nantes TGV, La 
Baule SNCF on Saturday night, if all those connections work...
-- 
David Singer
Apple Computer/QuickTime



From rem-conf Mon Sep 18 19:22:59 2000 
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From rem-conf Mon Sep 18 20:29:34 2000 
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Date: Mon, 18 Sep 2000 23:27:58 -0400
From: "Narasimhan, Sam (SD-EX)" <SNarasimhan@gi.com>
Subject: RE: comments on in-band-signaling proposal
To: 'Yoshihiro Kikuchi' <kiku@eel.rdc.toshiba.co.jp>,
 jan.vandermeer@philips.com, rem-conf@es.net, SNarasimhan@GI.COM
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Dear Jan and Kikuchi-San,

 I support Jan's proposal and I am sorry to see no
support for this from Kikuchi-San. MPEG will define a
format for carriage of MPEG-4 ES only which will require
CTS and DTS (like we did for MPEG-2 carriage) and this
will not be backward compatible with Kikuchi-San's draft.
If so, then one has to choose a more generic proposal.

Best Regards
Sam Narasimhan
Motorola

-----Original Message-----
From: Yoshihiro Kikuchi [mailto:kiku@eel.rdc.toshiba.co.jp]
Sent: Monday, September 18, 2000 3:39 AM
To: jan.vandermeer@philips.com; rem-conf@es.net; SNarasimhan@GI.COM
Cc: ZviL@optibase.com; csp@east.isi.edu;
4on2andIP-sys@advent.ee.columbia.edu; Yoshihiro Kikuchi
Subject: Re: comments on in-band-signaling proposal


Dear Jan and all,

Following suggestion 1, the authors will produce an updated Internet-Draft
of MPEG-4 Audio/Visual RTP soon with reflecting comments raised and agreed
during the WG last call period. It does not included in-band signaling
proposal.

It is too early for us to comment on suggestions 2-5, because we have not
seen detailed proposal. In any case, we cannot accept any disadvantage
(overhead, efficiency, risk, documentation, ...) compared to the current
spec. We should also be care that we will not fall into "an MPEG only
solution".

But if we can keep on these, maybe we will have a good solution in the end.

Kind regards,
Yoshihiro Kikuchi

----- Original Message -----
From: <jan.vandermeer@philips.com>
To: <kiku@eel.rdc.toshiba.co.jp>; <rem-conf@es.net>; <SNarasimhan@gi.com>
Cc: <ZviL@optibase.com>; <csp@east.isi.edu>;
<4on2andIP-sys@advent.ee.columbia.edu>
Sent: Saturday, September 16, 2000 7:47 PM
Subject: RE: comments on in-band-signaling proposal


> This one did not came through yesterday, due to network problems
> Sam, Kikuchi-san, others
> While I am personally confident that it is possible to define a feasible
and
> good solution, I share some of the concern of Kikuchi-san. Within MPEG
there
> are far too many misunderstandings and disagreements to approve anything
at
> this time. MPEG clearly needs more time to identify the most desirable
> approach.
>
> Therefore I would like to suggest the following way forward :
>
> 1) The IETF proceeds with the Kikuchi-san draft without implementing the
> in-band signaling proposal so as to meet the ITU deadline with minimum
risk.
> 2) MPEG continues to consider the most desirable format for MPEG-4 ES and
> other MPEG-4 streams.
> 3) In the La Baule meeting MPEG concludes its discussion by producing its
> "generic" MPEG-4 over IP draft.
> 4) If MPEG at the La Baule meeting is confident about the feasibility and
> usefulness of a forward and backward compatible solution using the
> Kikuchi-san draft with in-band signaling, then MPEG will request IETF to
> start the publication process of an "update" of the Kikuchi-san draft,
that
> will obsolete the current one.
> 5) In case of such "update", MPEG commits to neither change the "update"
of
> the Kikuchi-san draft, nor elements in the "generic MPEG-4 A/V ES" draft
> that impact the forward and backward compatibility with the updated
> Kikuchi-san draft.
>
> Kind regards,
>
> Jan
>
> -----Original Message-----
> From: SNarasimhan@gi.com [mailto:SNarasimhan@gi.com]
> Sent: vrijdag, 15 september, 2000 02:15
> To: Jan vanderMeer; rem-conf@es.net; kiku@eel.rdc.toshiba.co.jp
> Cc: 4on2andIP-sys@advent.ee.columbia.edu; csp@east.isi.edu;
> ZviL@optibase.com
> Subject: RE: comments on in-band-signaling proposal
>
>
> Dear Kikuchi-San and experts,
>
>  I am coming from a world where content creators and
> servers are very 'receiver-aware' and will not do anything
> in the content creation that will crash receivers. In this
> regard, the service providers who use the content want to
> play this on as many receivers as possible and they are also
> aware of the fact that receivers vary in their capabilities.
> Service providers, on the other hand, will not send 2 separate
> content streams to play on 2 receivers on the same wire as this
> consumes precious bandwidth.
>
>  2 examples of streams that one may want to play on receivers
> that implement your draft are
>
> 1. A video ES that required CTS (in the RTP header) and DTS (which
>    could be in the adaptation field). This may be needed for receivers
>    that implement MPEG buffer models. This stream can play on
>    receivers that implement your draft as it has a bigger buffer
>    and does not need DTS, as long as your draft enabled these
>    receivers to use the adaptation length field to offset where
>    the video ES payload was picked up.
> 2. One may author SL packetized video where there was information
>    in the SL header and this will be in the adaptation field. In addition,
>    the authors will make sure that the video ES in the SL packet
>    would be independently decodable by receivers that implemented your
>    draft. This will require the ability to parse the first byte in
>    the adaptation header and use it to offset where the video ES
>    payload would be picked up.
>
>  If your draft can accommodate this simple extension, then it will enable
> the receivers to play more content from the future and it will enable
> service providers to keep this in mind when they author MPEG-4 based
> content in the future.
>
>  Regards to your concerns,
>
> 1.  The overhead is 1 byte/RTP packet and this is a 6.25% addition
>     to the 16 byte RTP header. If we sent 1 VOP per packet, then it
>     is 30 bytes/sec (assuming 30 Hz video) or 240 bits/sec overhead.
>     This is very minimal. The FEC proposal proposes the same type
>     of overhead.
> 2.  The content authors will always make sure they follow all the
>     other recommendations from your draft to make sure that the
>     future content will play on receivers that implement the draft
>     (such as sending data in decoding order, not splitting the headers
>     etc). The only change from an implementation of today is the
>     extra parsing of one byte before the ES payload to offset where
>     the ES payload is picked up.
> 3.  The change to your draft would to be add 2 fields after the CSRC
>     and before the video ES. These would be
>     'adaptation length' and
>     'length (if adaptation length is greater than 0)'.
>     The semantics will say, the first byte after CSRC will contain the
>     length of adaptation field. If this length value is zero, then the
>     video payload will start in the next byte. Otherwise, the video
>     payload will start after the number of bytes indicated in the
>     adaptation length field. The content of the adaptation field after
>     the length is private and left for future extensions.
>
>  Jan Vandermeer may also have some suggested changes to the draft, if this
> can be accommodated. Thank you for your kind patience.
>
> Best Regards
> Sam Narasimhan
> Motorola
>
> -----Original Message-----
> From: Yoshihiro Kikuchi [mailto:kiku@eel.rdc.toshiba.co.jp]
> Sent: Thursday, September 14, 2000 12:38 PM
> To: IETF AVT
> Cc: Zvi Lifshitz; Colin Perkins; Yoshihiro Kikuchi
> Subject: comments on in-band-signaling proposal
>
>
> Dear all experts,
>
>
> The authors of the Internet-Draft have discussed the issue of
> in-band-signaling proposal. We understand the importance of
> interoperability, however, we do not believe the specific proposal of an
> extra byte in the RTP header is a good solution. It is our common and firm
> desire that any proposal will not degrade the original intention, purpose
> and functionality of the RTP payload format.
>
> Especially, the following points are important.
>
> o Overhead caused by an extra RTP header
> The in-band-signaling proposal will cause at least one byte overhead per
> each RTP packet. Since this RTP payload format may be used in very low
> bitrate applications such as mobile and analog modem, the overhead is a
> serious problem. On the other hand, the ways suggested by Colin, Stephan,
> etc. of using different payload types will not cause any overhead problem.
>
> o Stability and reliability
> As was pointed out by many, there is no guaranty that receivers of
ES-based
> RTP payload format can receive SL-based RTP packets correctly, even if the
> extra byte in the RTP header is added. It is dangerous to adopt such
> proposal at this moment without any guaranty. It is our strong desire that
> only verified solution will be adopted in the RTP payload format
> specification.
>
> o Schedule
> There is no specific description of changes about the proposal, so far. It
> is hard to accept such proposal. Even if the proposal has some value
> (although we doubt it), it should not be too late to adopt it in
> SL-based proposal, by keeping backword compatibility with ES-based RTP
> payload format.
>
>
> These points are very important to keep the original functionality of the
> proposal. Constructive comments are welcome, however, we cannot accept
> any proposals to harm the original intention and functionality. By this
> point, it is hard to accept the in-band-signaling proposal which causes
> overhead problem and which is not well verified. It should be remarked
that
> the alternatives proposed by IETF experts will not cause any such
problems.
>
>
> Kind regards,
> Yoshihiro Kikuchi
> and authors of the Intenet Draft
>
> ----
>         Yoshihiro Kikuchi
>
> E-mail: kiku@eel.rdc.toshiba.co.jp
> Corporate Research and Development Center
> TOSHIBA Corporation
> TEL: +81 +44 549 2288    FAX: +81 +44 520 1267
>
>
>



From rem-conf Mon Sep 18 23:35:44 2000 
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To: "'Narasimhan, Sam (SD-EX)'" <SNarasimhan@gi.com>,
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	 <kiku@eel.rdc.toshiba.co.jp>,
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[Narasimhan, Sam (SD-EX):]

I support Jan's proposal and I am sorry to see no
support for this from Kikuchi-San. MPEG will define a
format for carriage of MPEG-4 ES only which will require
CTS and DTS (like we did for MPEG-2 carriage) and this
will not be backward compatible with Kikuchi-San's draft.
If so, then one has to choose a more generic proposal.

[Reply:]

In Paris we said that in cases when DTS is not required (we never said it 
would always be required as you seem to propose) then it can be backward 
compatible.

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				GSM: +972-54-461787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
==================================================================
> Come see us at:
The ISP Forum, Cavalieri Hilton, Rome, October 2nd-5th, 2000 
www.isp-conferences.com/ispf2000
Streaming Media Europe Earls Court Centre, Booth # 628, London, October 
10th-12th, 2000






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A New Internet-Draft is available from the on-line Internet-Drafts directories.
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	Title		: RTP payload format for MPEG-4 Audio/Visual streams
	Author(s)	: Y. Kikuchi, T. Nomura, S. Fukunaga, 
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	Filename	: draft-ietf-avt-rtp-mpeg4-es-04.txt
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	Date		: 18-Sep-00
	
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From rem-conf Tue Sep 19 09:14:44 2000 
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From: Bill Nowicki <BNowicki@Omneon.com>
To: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: Pre-MPEG meeting for the '4onIP' AHG
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Ah, is there a rule that people going to standards meetings must travel on
every mode of transportation that has an acronym? Probably would not be
allowed to go to a "group" meeting in a country that just has "trains" and
"airports"?

Sorry I could not resist. Seriously, we appreciate standards efforts, but
would appreciate the acronyms spelled out once in a while for the benefit of
those of us unable to attend the meetings in person. Thanks.



From rem-conf Tue Sep 19 09:39:36 2000 
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You mean these:

I'll be in La Baule via SFO, JFK, LHR, CDG, CDG TGV, Nantes TGV, La Baule SNCF on Saturday night, if all those connections work... ???

Well, those who are unable to attend the meeting in person don't need these acronyms spelled out...

z
soon the whole world will STREAM (but hopefully we'll still get through these ;-)

==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				GSM: +972-54-461787
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==================================================================
> Come see us at:
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Streaming Media Europe Earls Court Centre, Booth # 628, London, October 10th-12th, 2000


-----Original Message-----
From:	Bill Nowicki [SMTP:BNowicki@Omneon.com]
Sent:	Tuesday, September 19, 2000 6:06 PM
To:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: Pre-MPEG meeting for the '4onIP' AHG

 
Ah, is there a rule that people going to standards meetings must travel on
every mode of transportation that has an acronym? Probably would not be
allowed to go to a "group" meeting in a country that just has "trains" and
"airports"?

Sorry I could not resist. Seriously, we appreciate standards efforts, but
would appreciate the acronyms spelled out once in a while for the benefit of
those of us unable to attend the meetings in person. Thanks.




From rem-conf Tue Sep 19 12:37:55 2000 
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From: "Accountant's Marketing Enterprises" <randy.hirsch@belizemail.net>
To: <rem-conf@es.net>
Subject: CPAs, QuickBooks Advisors and Accountants -- New Year-Round Business Clients
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Hello:  We found your firm's name recently on one of the search directories...  We offer a unique and affordable marketing and appointment 
setting service for Accountants, QuickBooks Advisors, CPAs and Tax Advisors...  We'd like to introduce you to several business owners (with 
1-20 employees, annual revenues of $250,000 to $750,000) who have an immediate need for accounting, tax, payroll and consulting 
services... Most of these business owners have questions regarding incorporation, what software to buy for their businesses, retirement, tax and 
business planning, etc... Many have had problems with QuickBooks...  What we're finding this time of year, are business owners who have 
immediate tax needs, and also year-round accounting and "tax planning" needs as well... With our service, you pay as you go -- for results!  We 
charge $350 for every two year-round monthly or quarterly business clients that you actually sign up from our leads, business prospects and 
appointments...  Let us know that you're interested, and would like more information by calling us, or you may fax your request for more info... 
We'll send you more information about our program, including: where we're located, how we find qualified business clients (85-90% of our 
business owner leads are coming "inbound" directly from our web site),  references, the four ways that we generate appointments with 
interested business owners, the size and types of businesses that we're finding, the average Accounting fees that are being generated by our 
500+ member Accountants and CPAs nationwide, and our exclusive area guarantee (we give every customer of ours an exclusive 10 to 50 mile 
area)...   Looking forward to hearing from you soon...  Best wishes to you and your family...  From my wife Jennifer, and daughters Michelle and 
Molly, and myself, "Have a GREAT year!"  Randy Hirsch

OUR CONTACT INFORMATION:
Accountant's Marketing Enterprises 
"New Year-Round Business Clients for CPAs and Accountants"
Call us Toll-Free: (888) 305-0651   
For the fastest response, FAX your request for information to: (509) 472-7479  
Be sure to include your email address, firm name, phone and fax numbers
or you may E-mail your request for more information...

For remove instructions, please see below...

P.S.  Respond right away and we'll email or fax a map of the exclusive area that we have in mind for your firm, and give you a $25 discount on 
your first two business clients... AND just to say "Thanks for responding so promptly" we're sending you a complimentary copy of our best 
selling: BLUEPRINTS Marketing Report for Growing and Expanding Accounting & Tax Firms."  The report is packed with good ideas on 
growing your business and adding new clients, and sample introduction letters, two postcards, flyers, brochures and three sample Accounting 
Service agreements that we designed...  We hope that you find the report helpful and informative...

*************************************************************************************************************************************

REMOVAL INSTRUCTIONS:  

This message is being sent to you in compliance with the proposed Federal Legislation commercial e-mail S.1618 -- Section 301... "Pursuant to 
Section 301, Paragraph (a) (2)  (C) of S. 1618, further transmissions to you by the sender of this email may be stopped at no cost to you by 
submitting a request to "REMOVE" on the reply line of this message.... Further, this message cannot be considered  SPAM as long as we 
include sender contact information...  This is a one time mailing, done by an independent marketing company...  We apologize for any 
inconvenience this may have caused you...  We respect your online privacy and apologize if you have received this message in error...  We do 
respect our remove lists!  Your name was provided to us after extensive research and screening, as either an Accountant, CPA, QuickBooks 
Advisor,Tax Professional, Business Owner or Self-Employed Person...  If you would like to be removed from any future mailings, please reply to 
this email and send us a remove request email...  Please be aware that any disruption to our remove request email link above, also prevents 
those that want to be removed from our database from being removed...  Thank You...  

 



From rem-conf Tue Sep 19 13:04:22 2000 
From rem-conf-request@es.net Tue Sep 19 13:04:22 2000
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Date: Tue, 19 Sep 2000 15:27:51 -0400
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From: Dan <livevideo@speedycam.com>
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Dear  ISP / Webmaster:

Get new members and keep them with Speedycam!  You can now offer your 
members a way to interact with each other via live video and chat !  Your 
members can have their own live video web page with chat and real time 
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  You may also email info@speedycam.com  for a demo and for custom 
video/audio solutions for your company.



You were sent this email because you are part of a  webmaster list, if you 
wish to be removed from our list send a email to : remove@speedycam.com
Thank you. 



From rem-conf Tue Sep 19 18:10:12 2000 
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From: infomarketing4@china.com
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Date: Tue, 19 Sep 00 19:41:22 EST
To: infomarketing4@china.com
Subject: INTERNET MARKETING PROGRAM
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THE INTERNET MARKETING PROGRAM
                $$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$

                   INCREDIBLE $0 to $50,000 in 90 days!!!

Dear Friend,

You can earn $50,000 or more in the next 90 days sending e-mail. Seem
impossible? Read on for details.

                          "AS SEEN ON NATIONAL TV"

Thank you for your time and interest. This is the letter you've been reading
about in the news lately.  Due to the popularity of this letter on the
Internet, a major nightly news program recently devoted an entire show to
the investigation of the program described below to see if it really can
make people money.

The show also investigated whether or not the program was legal.  Their
findings proved once and for all that there are absolutely no laws
prohibiting the participation in the program. This has helped to show people
that this is a simple, harmless and fun way to make some extra money at
home.

The results of this show have been truly remarkable. So many people are
participating that those involved are doing much better than ever before.
Since everyone makes more as more people try it out, it's been very
exciting. You will understand once you experience it.

                              HERE IT IS BELOW:

                 *** Print This Now For Future Reference ***

The following income opportunity is one you may be interested in taking a
look at. It can be started with VERY LITTLE investment and the income return
is TREMENDOUS!!!

               $$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$
            If you would like to make at least $50,000 in less than 90 days

            Please read the enclosed program...THEN READ IT AGAIN!!!
               $$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$

THIS IS A LEGITIMATE, LEGAL, MONEY MAKING OPPORTUNITY. 
It does not require you to come into contact with people, do any hard work,
and best of all, you never have to leave the house except to get the mail. 
If you believe that someday you'll get that big break that you've been waiting
for, THIS IS IT!  Simply follow the instructions, and your dreams will come true.
This multi-level email order marketing program works perfectly...100% EVERY TIME.

E-mail is the sales tool of the future. Take advantage of this
non-commercialized method of advertising NOW!!! The longer you wait, the
more people will be doing business using email. Get your piece of this
action!!!

MULTI-LEVEL MARKETING (MLM) has finally gained respectability.  It is being
taught in the Harvard Business School, and both Stanford Research and the
Wall Street Journal have stated that between 50%
and 65% of all goods and services will be sold through multi-level methods
by the late 1990's.  This is a Multi-Billion Dollar industry and of the
500,000 millionaires in the U.S., 20% (100,000) made their fortune in the
last several years in MLM.  Moreover, statistics show that 45 people become
millionaires everyday through Multi-Level Marketing.

You may have heard this story before, but Donald Trump recently made an
appearance on the David Letterman show. Dave asked him what he would do if
he lost everything and had to start over from
scratch. Without hesitating, Trump said he would find a good network
marketing company and get to work. The audience started to hoot and boo him.
He looked out at the audience and deadpanned his response:
"That's why I'm sitting up here and you are all sitting out there!"

A PERSONAL NOTE FROM THE ORIGINATOR OF THIS PROGRAM:

By the time you have read the enclosed program and reports, you should have
concluded that such a program, and one that is legal, could not have been
created by an amateur.

Let me tell you a little about myself. I had a profitable business for 10
years. Then in 1979 my business began falling off.  I was doing the same
things that were previously successful for me, but it wasn't
working. Finally, I figured it out. It wasn't me, it was the economy.
Inflation and recession had replaced the stable economy that had been with
us since 1945.  I don't have to tell you what happened to the
unemployment rate... because many of you know from first hand experience.
There were more failures and bankruptcies than ever before.

The middle class was vanishing.  Those who knew what they were doing
invested wisely and moved up. Those who did not, including those who never
had anything to save or invest, were moving down into the ranks of the poor.
As the saying goes, "THE RICH GET RICHER AND THE POOR GET POORER." The
traditional methods of making money will never allow you to "move up" or
"get rich", inflation will see to that.

You have just received information that can give you financial freedom for
the rest of your life, with "NO RISK" and "JUST A LITTLE BIT OF EFFORT." You
can make more money in the next few months than you have ever imagined.  I
should also point out that I will not see a penny of this money, nor anyone
else who has provided a testimonial for this program. I have already made
over 4 MILLION DOLLARS! I have retired from the program after sending
thousands and thousands of programs.

Follow the program EXACTLY AS INSTRUCTED.  Do not change it in any way.  It
works exceedingly well as it is now. Remember to e-mail a copy of this
exciting report to everyone you can think of. One of the people you send
this to may send out 50,000...and your name will be on everyone of them!

Remember though, the more you send out the more potential customers you will
reach.

So my friend, I have given you the ideas, information, materials and
opportunity to become financially independent. IT IS UP TO YOU NOW!

                              "THINK ABOUT IT"

Before you delete this program from your mailbox, as I almost did, take a
little time to read it and REALLY THINK ABOUT IT.  Get a pencil and figure
out what could happen when YOU participate. Figure out the worst possible
response and no matter how you calculate it, you will still make a lot of
money!  You will definitely get back what you invested.  Any doubts you have
will vanish when your first orders come in.  IT WORKS!

                         Jody Jacobs, Richmond, VA

     HERE'S HOW THIS AMAZING PROGRAM WILL MAKE YOU THOUSANDS OF
DOLLAR$

                               INSTRUCTIONS:

This method of raising capital REALLY WORKS 100% EVERY TIME.  I am sure that
you could use up to $50,000 or more in the next 90 days. Before you say
"BULL... ", please read this program carefully.

This is not a chain letter, but a perfectly legal money making opportunity.
Basically, this is what you do: As with all multi-level businesses, we build
our business by recruiting new partners and selling our products. Every
state in the USA allows you to recruit new multi-level business partners,
and we offer a product for EVERY dollar sent.  YOUR ORDERS COME BY MAIL
 AND ARE FILLED BY E-MAIL, so you are not involved in personal selling. You do it
privately in your own home, store or office.  This is the GREATEST
Multi-Level Mail Order Marketing anywhere.

                          This is what you MUST do:

1. Order all 4 reports shown on the list below (you can't sell them
if you don't order them).
-- For each report, send $5.00 CASH, the NAME & NUMBER OF THE REPORT
YOU ARE ORDERING, YOUR E-MAIL ADDRESS, and YOUR NAME & RETURN
ADDRESS (in case of a problem) to the person whose name appears on
the list next to the report.  MAKE SURE YOUR RETURN ADDRESS IS ON
YOUR ENVELOPE IN CASE OF ANY MAIL PROBLEMS!
-- When you place your order, make sure you order each of the four
reports. You will need all four reports so that you can save them on
your computer and resell them.
-- Within a few days you will receive, via e-mail, each of the four
reports. Save them on your computer so they will be accessible for you to
send to the 1,000's of people who will order them from you.

2. IMPORTANT DO NOT alter the names of the people who are listed next
to each report, or their sequence on the list, in any way other than
is instructed below in steps "a" through "f" or you will lose out on
the majority of your profits. Once you understand the way this works,
you'll also see how it doesn't work if you change it. Remember, this
method has been tested,and if you alter it, it will not work.
a. Look below for the listing of available reports.
b. After you've ordered the four reports, take this advertisement and
remove the name and address under REPORT #4. This person has
made it through the cycle and is no doubt counting their $50,000!
c. Move the name and address under REPORT #3 down to REPORT #4.
d. Move the name and address under REPORT #2 down to REPORT #3.
e. Move the name and address under REPORT #1 down to REPORT #2.
f.  Insert your name/address in the REPORT #1 position.

     Please make sure you COPY ALL INFORMATION, every name and
address,
     ACCURATELY!

3. Take this entire letter, including the modified list of names, and
save it to your computer. Make NO changes to the instruction portion
of this letter.

   Your cost to participate in this is practically nothing (surely
you can afford $20). You obviously already have an Internet
connection and e-mail is FREE!



          There are two primary methods of building your downline:

                       METHOD #1: SENDING BULK E-MAIL

Let's say that you decide to start small, just to see how it goes,
and we'll assume you and all those involved send out only 2,000
programs each. Let's also assume that the mailing receives a 0.5%
response. Using a good list the response could be much better. Also,
many people will send out hundreds of
thousands of programs instead of 2,000. But continuing with this
example, you send out only 2,000 programs. With a 0.5% response, that
is only 10 orders for REPORT #1. Those 10 people respond by sending
out 2,000 programs each for a total of 20,000. Out of those 0.5%, 100
people respond and order REPORT #2. Those 100 mail out 2,000 programs
each for a total of 200,000.
The 0.5% response to that is 1,000 orders for REPORT #3. Those 1,000
send out 2,000 programs each for a 2,000,000 total. The 0.5% response
to that is 10,000 orders for REPORT #4. That's 10,000 $5 bills for
you. CASH!!! Your total income in this example is $50 + $500 + $5,000
+ $50,000 for a total of $55,550!!! REMEMBER FRIEND, 
THIS IS ASSUMING 1,990 OUT OF THE 2,000 PEOPLE YOU MAIL
 TO WILL DO ABSOLUTELY NOTHING AND TRASH THIS
PROGRAM! DARE TO THINK FOR A MOMENT WHAT WOULD HAPPEN IF
EVERYONE, OR HALF SENT OUT 100,000 PROGRAMS INSTEAD OF 2,000.
Believe me, many people will do just that, and more! By the way, your cost
to participate in this is practically nothing.  You obviously already have an
Internet connection and e-mail is FREE!!! REPORT #2 will show you the best
methods for bulk e-mailing, tell you where to obtain free bulk e-mail
software and where to obtain e-mail lists.


                METHOD #2 - PLACING FREE ADS ON THE INTERNET

Advertising on the internet is very, very inexpensive, and there are
HUNDREDS of FREE places to advertise. Let's say you decide to start
small just to see how well it works. Assume your goal is to get ONLY
10 people to participate on your first level. (Placing a lot of FREE
ads on the Internet will EASILY get a larger response.) Also assume that
everyone else in YOUR ORGANIZATION gets ONLY 10 downline members.
Follow this example to achieve the STAGGERING results below:

1st level--your 10 members with $5.......................................$50
2nd level--10 members from those 10 ($5 x 100)..................$500
3rd level--10 members from those 100 ($5 x 1,000)...........$5,000
4th level--10 members from those 1,000 ($5 x 10,000).....$50,000
                   THIS TOTALS ---------->$55,550

Remember friends, this assumes that the people who participate only
recruit 10 people each. Think for a moment what would happen if they
got 20 people to participate! Most people get 100's of participants!
THINK ABOUT IT! For every $5.00 you receive, all you must do is e-mail them
the report they ordered. THAT'S IT! ALWAYS PROVIDE SAME-DAY SERVICE
ON ALL ORDERS! This will guarantee that the e-mail THEY send out with YOUR
name and address on it will be prompt because they can't advertise
until they receive the report!

                              AVAILABLE REPORTS

                *** Order Each REPORT by NUMBER and NAME ***
                                   Notes:
-- ALWAYS SEND $5 CASH (U.S. CURRENCY) FOR EACH REPORT. CHECKS NOT
   ACCEPTED.
-- ALWAYS SEND YOUR ORDER VIA FIRST CLASS MAIL.
-- Make sure the cash is concealed by wrapping it in at least two
sheets of paper. On one of those sheets of paper, include:
     (a) the number & name of the report you are ordering, (b) your
e-mail address, and (c) your name & postal address.

                  PLACE YOUR ORDER FOR THESE REPORTS NOW:


REPORT #1 "The Insider's Guide to Advertising for Free on the
Internet"

ORDER REPORT #1 FROM:
                                 M & M
                                 PO BOX 1862
                                 Santa Rosa Beach, FL 32459

REPORT #2 "The Insiders Guide to Sending Bulk E-mail on the Internet"

ORDER REPORT #2 FROM:
                                Steve Goodwin
                                 PO BOX 283
                                 Johnston,  IA 50131


REPORT #3 "The Secrets to Multilevel Marketing on the Internet"

ORDER REPORT #3 FROM:
                                 M & H Consulting
                                 2714 W 5th
                                  North Platte, NE 69101

REPORT #4 "How to become a Millionaire utilizing the Power of
Multilevel Marketing and the Internet"

ORDER REPORT #4 FROM:

                                  Nancy Brooks
                                 3182 Lakeway ST
                                 Akron, Ohio 44319


               About 50,000 new people get online every month!

                         ******* TIPS FOR SUCCESS *******
-- TREAT THIS AS YOUR BUSINESS! Be prompt, professional, and follow
the directions accurately.
-- Send for the four reports IMMEDIATELY so you will have them when
the orders start coming in because: When you receive a $5 order, you
MUST send out the requested product/report.
-- ALWAYS PROVIDE SAME-DAY SERVICE ON THE ORDERS YOU RECEIVE.
-- Be patient and persistent with this program. If you follow the
     instructions exactly, your results WILL BE SUCCESSFUL!
-- ABOVE ALL, HAVE FAITH IN YOURSELF AND KNOW YOU WILL SUCCEED!

                   ******* YOUR SUCCESS GUIDELINES *******
             Follow these guidelines to guarantee your success:

If you don't receive 20 orders for REPORT #1 within two weeks,
continue advertising or sending e-mails until you do. Then, a couple of
weeks
later you should receive at least 100 orders for REPORT#2. If you don't,
continue advertising or sending e-mails until you do. Once you have received
100 or more orders for REPORT #2, YOU CAN RELAX,
because the system is already working for you, and the cash will continue to
roll in!

                       THIS IS IMPORTANT TO REMEMBER:
Every time your name is moved down on the list, you are placed in
front of a DIFFERENT report. You can KEEP TRACK of your PROGRESS by
watching which report people are ordering from you. If you want to
generate more income, send another batch of e-mails or continue
placing ads and start the whole process again! There is no limit to
the income you will generate from this business!

Before you make your decision as to whether or not you participate in
this program. Please answer one question. DO YOU WANT TO CHANGE YOUR
LIFE? If the answer is yes, please look at the following facts about
this program:

1. You are selling a product which does not Cost anything to PRODUCE,
SHIP OR ADVERTISE.
2. All of your customers pay you in CASH!
3. E-mail is without question the most powerful method of
distributing information on earth. This program combines the
distribution power of e-mail together with the revenue generating
power of multi-level marketing.
4. Your only expense--other than your initial $20 investment--is your
time!
5. Virtually all of the income you generate from this program is PURE
PROFIT!
6. This program will change your LIFE FOREVER.

ACT NOW!  Take your first step toward achieving financial independence.
Order the reports and follow the program outlined above-SUCCESS will
be your reward.

                 Thank you for your time and consideration.

PLEASE NOTE: If you need help with starting a business, registering a
business name, learning how income tax is handled, etc., contact your
local office of the Small Business Administration (a Federal Agency)
1-800-827-5722 for free help and answers to questions. Also, the
Internal Revenue Service offers free help via telephone and free
seminars about business tax requirements. Your earnings are highly
dependant on your activities and advertising. The information
contained on this site and in the report constitutes no guarantees
stated nor implied. In the event that it is determined that this site
or report constitutes a guarantee of any kind, that guarantee is now
void. The earnings amounts listed on this site and in the report are
estimates only. If you have any questions of the legality of this
program, contact the Office of Associate Director for Marketing
Practices, Federal Trade Commission, Bureau of Consumer Protection in
Washington, DC.


/////////////////////////////////////////////////////////////////
Remove at rivers@dcemail.com
/////////////////////////////////////////////////////////////////

This message is sent in compliance of the new e-mail bill: SECTION 301. Per
Section 301, Paragraph (a)(2)(C) of S. 1618,
http://www.senate.gov/~murkowski/commercialemail/S771index.html










From rem-conf Tue Sep 19 18:15:21 2000 
From rem-conf-request@es.net Tue Sep 19 18:15:20 2000
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Message-ID: <010d01c0229f$129b83a0$03000004@hariv.domain>
Reply-To: "Hari Krishna Vuppaladhadiam" <hariv@chiplogic.com>
From: "Hari Krishna Vuppaladhadiam" <hariv@chiplogic.com>
To: <rem-conf@es.net>
Subject: Voice Playout for VoATM 
Date: Tue, 19 Sep 2000 18:06:56 -0700
MIME-Version: 1.0
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	charset="iso-8859-1"
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Hi,
  I am looking for Voice playout schemes for VoATM. Are there any
references?
hari





From rem-conf Tue Sep 19 21:25:08 2000 
From rem-conf-request@es.net Tue Sep 19 21:25:07 2000
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To: Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp>
cc: IETF AVT <rem-conf@es.net>,
        4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>
Subject: Re: New I/D submitted for MPEG-4 A/V RTP 
In-Reply-To: Message from Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp> 
   of "Mon, 18 Sep 2000 21:26:30 +0900." <01c101c0216b$ac58ec60$cc51c485@eel.rdc.toshiba.co.jp> 
Date: Tue, 19 Sep 2000 15:16:10 -0400
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Folks,

Now that this update has made it to the i-d archives, please can you send
any final comments this week, such that we can progress this document.

Thanks,
Colin



--> Yoshihiro Kikuchi writes:
>Dear all,
>
>I have submitted a new version (-04) of the internet draft of MPEG-4
>Audio/Visual RTP payload format to reflect comments raised and agreed during
>the WG last call period. The document should be available soon at the
>internet draft archive.
>
>The following is the summary of changes from the previous version (-03).
>
>1) Default value of MIME type
>2) Editorial update of abstract and introduction
>3) Clarification related to multiple VOPs concatenation (timestamp,
>fragmentation rule 4)
>4) LATM related modification (section 1.2, 4.2)
>5) Some other editorial clarifications
>
>
>The authors would like to thank for those who raised comments.
>
>Best regards,
>Yoshihiro Kikuchi
>
>----
>        Yoshihiro Kikuchi
>
>E-mail: kiku@eel.rdc.toshiba.co.jp
>Corporate Research and Development Center
>TOSHIBA Corporation
>TEL: +81 +44 549 2288    FAX: +81 +44 520 1267
>
>
>
>



From rem-conf Wed Sep 20 03:13:39 2000 
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From: CURET Dominique FTRD/DMI/REN <dominique.curet@rd.francetelecom.fr>
To: 'Dave Singer' <singer@apple.com>
Cc: 4on2andIP-sys@advent.ee.columbia.edu, rem-conf@es.net, 
	CURET Dominique FTRD/DMI/REN <dominique.curet@rd.francetelecom.fr>
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Dear Dave, dear all,

It seems that going from SFO to La Baule is quite tricky, as you said:
" I'll be in La Baule via SFO, JFK, LHR, CDG, CDG TGV, Nantes TGV, La=20
Baule SNCF"=20

I don't have any advice to give, but if it can help you and some =
others:
*  there are some direct flights from SFO to CDG (or London?)
*  If you were coming via London you may catch a flight to Nantes,=20
   but it is from Gatwick (3 flights per day)
*  If you were coming via Bruxelles you may catch a flight to Nantes,=20
   (3 flights per day)
* you can check that there are a large number of flights from other=20
Euoropean airports=20

Dominique
 ________________________________________
> Dominique Curet
> France T=E9l=E9com R&D - DMI/PIA
> % + 33 (0)2 99 12 40 05   Fax : + 33 (0)2 99 12 40 98
> )  CCETT       B.P. 59            4, rue du Clos Courtel=20
> 35512           Cesson-Sevigne   Cedex   09     FRANCE
> mailto:dominique.curet@rd.francetelecom.fr
_________________________________________




-----Message d'origine-----
De: Dave Singer [mailto:singer@apple.com]
Date: lundi 18 septembre 2000 23:31
=C0: CURET Dominique FTRD/DMI/REN
Cc: 4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Objet: Re: Pre-MPEG meeting for the '4onIP' AHG


At 6:01 PM +0200 9/13/00, CURET Dominique FTRD/DMI/REN wrote:
>Dear all, dear Young-Kwon,
>
>we are supposed to meet on Saturday(21/10) and Sunday (22/10)
>before the MPEG meeting in La Baule.
>
>Can we say something like:
>      Saturday from 14 to 2000
>      Sunday from 0900 to 1800
>Or only the Sunday from 0900 to 1800
>
>comments


whoa!  I'm completely snowed by the mail on this...and read as fast=20
as I can, I am not catching up.  hence no comment from me yet!

I'll be in La Baule via SFO, JFK, LHR, CDG, CDG TGV, Nantes TGV, La=20
Baule SNCF on Saturday night, if all those connections work...
--=20
David Singer
Apple Computer/QuickTime



From rem-conf Wed Sep 20 06:03:21 2000 
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Hari Krishna Vuppaladhadiam'" <hariv@chiplogic.com>, rem-conf@es.net
Subject: RE: Voice Playout for VoATM 
Date: Wed, 20 Sep 2000 08:58:06 -0400
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You wouldn't use RTP for VoATM except for a case of VoIP over ATM.
So, I suspect this group can't help you much.

The ATM Forum has some specs on carrying voice.  The most recent
is the BLES spec for VoDSL, which is ATM AAL2 based.  
The Q.BICC work in ITU covers VoATM using an extension to ISUP
to carry an NSAP address.

H.323 in ITU offers Annex C, which is an AAL5 VoATM bearer.

Finally, there is work in MMUSIC to extend SDP to include
ATM bearers, which would allow SIP, MEGACO and MGCP to
support VoATM. MEGACO (H.248) has VoATM support in the
"native" tags media format using binary encoding.  The SDP
work extends similar support for text encoding.

Brian

> -----Original Message-----
> From: Hari Krishna Vuppaladhadiam [mailto:hariv@chiplogic.com]
> Sent: Tuesday, September 19, 2000 9:07 PM
> To: rem-conf@es.net
> Subject: Voice Playout for VoATM 
> 
> 
> Hi,
>   I am looking for Voice playout schemes for VoATM. Are there any
> references?
> hari
> 
> 
> 



From rem-conf Wed Sep 20 06:18:29 2000 
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Date: Wed, 20 Sep 2000 15:16:04 +0200
From: ICSM2001 (Canfora) <icsm2001@unisannio.it>
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To: rem-conf@es.net
Subject: CFPs: IEEE Int. Conf. on Software Maint., Florence, Italy
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Dear Colleague

I hope that this CFPs could be useful for your work.
Please forward the following to anybody who you think may be interested.
Apologies if you have already seen this.

If you would like to be removed from our list please send an email to
icsm2001@unisannio.it  with REMOVE in the subject.

ICSM2001
Technical Committee

_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=

		               CALL---FOR---PAPERS

	IEEE International Conference on Software Maintenance 2001

		FLORENCE, ITALY, 5-9 November 2001, 

		        http://www.dsi.unifi.it/icsm2001

	Theme: Systems and Software Evolution in the era of the Internet

_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=_=

ICSM is the major international conference in the field of software and 
systems maintenance, evolution, and management.

In the era of the Internet, businesses and end-users have invested in new 
technologies and small and large software organizations around the world 
are looking for Internet related solutions to evolve and maintain their 
new Internet software products. 

Internet technologies are strongly impacting system architectures and 
business processes and rules. In some cases businesses and end-users 
have been overwhelmed trying to keep up with software development and 
evolution processes and practices. In addition to novel solutions to 
enable the life-cycle of new web-based software systems, huge investments 
are necessary to migrate aginglegacy applications to web-enabled 
contemporary systems.

ICSM 2001 will address these major changes in the software landscape and 
their impact on maintenance and evolution. The focus of the conference 
will be to explore the new challenges that the Internet, as a driver for 
business changes, poses for software maintenance, and the new opportunities
it opens as infrastructure and enabling technology.

The purpose of the conference is to promote discussion and interaction 
between researchers and practitioners. We are particularly interested in 
exchanging concepts, prototypes, research ideas, and other results which 
could contribute to the academic arena and also benefit business and the 
industrial community. ICSM 2001 will be participatory, with working 
collaborative sessions and presentations of industry projects. ICSM 2001 
will bring together researchers, practitioners, developers and users
of tools, technology transfer experts, and project managers.

The Conference will be held in conjunction with WESS, the Workshop on 
Empirical Studies of Software.

Topics of interest include but are not restricted to the following 
aspects of maintenance and evolution:
- Methods and theories			-Processes and strategies
- Organizational frameworks		-Life cycle and process control 
- Design for maintenance 		-Tools and environments 
- Internet and distributed systems 	-Multimedia systems 
- User interface evolution 		-Commercial off-the-shelf (COTS) 
- Third party maintenance 		-Freeware and open source applications
- Program comprehension 		-Software and system visualization 
- Knowledge based systems 		-Formal methods 
- Impact of new software practices	-Empirical studies 
- Software reusability 			-Programming languages 
- Source code analysis and manipulation 	-Testing and regression testing 
- Models and methods for error prediction 	-Measurement of software 
- Maintenance and/or productivity metrics 	-Preventive maintenance 
- Personnel aspects of maintenance 	-Reengineering and reverse engineering
- Version and configuration management 	-Legal aspects and standards 
- Management and organization 		-Remote, tele-work, and co-operative applications

RESEARCH PAPERS
Research papers should describe original and significant work in the 
research and practice of software maintenance. Research case studies, 
empirical research, and experiments are particularly welcome. Papers should be 2000 - 5000 words in length, in English. Submit them 
in PDF or PostScript via email to icsm2001@unisannio.it by 15 January 2001.
A prize of the Journal of Software Maintenance will be assigned at the 
Best submitted Paper.

INDUSTRIAL APPLICATIONS
We welcome proposals for presentations of Industrial Applications. 
These can be experience reports from real projects, industrial practices 
and models, or tool demonstrations. Submit proposals for Industrial Application
 presentations via email to icsm2001.industry@unisannio.it by 12 March 2001.
 Industrial Applications proposals will be reviewed by a dedicated 
sub-committee of the program committee and a 1 page summary of accepted 
proposals will be included in the conference proceedings.

TUTORIALS
Tutorials should present software maintenance and evolution topics of 
interest to practitioners. Tutorials may be full-day or half-day in length. 
Submit tutorial proposals via email to
icsm2001.tutorial@unisannio.it by 12 February 2001.

IMPORTANT DATES
Research Paper submission 15 January 2001, notification of acceptance 1 June 2001
Industrial Application submission 12 March 2001
Tutorial submission 12 February 2001

------------------------------

General chair:
	Paolo Nesi, University of Florence, Italy, nesi@dsi.unifi.it

Financial chair:
	Vaclav Rajlich, Wayne State University, USA, vtr@cs.wayne.edu

Program co-chairs:
	Gerardo Canfora, University of Sannio, Italy, gerardo.canfora@unisannio.it

	Anneliese von Mayrhauser, Colorado State University, USA, avm@CS.ColoState.EDU

Tutorials co-chairs:
	Lionel C. Briand, Carleton University, briand@sce.carleton.ca

	Alessandro Fantechi, University of Florence, Italy, Fantechi@dsi.unifi.it

Industrial Applications co-chairs:
	Panagiotis K. Linos, Tennessee Technological University, USA, linos@tntech.edu

	Harry Sneed, Software Engineering Service GmbH, Germany, Harry.Sneed@t-online.de

	Chris Verhoef, University of Amsterdam, NL, x@wins.uva.nl

Publicity co-chairs:
	Nicholas Zvegintzov (General Co-chair), Software Management Network, USA, 
	zvegint@attglobal.net

	Malcolm Munro (Co-chair for Europe), University of Durham, UK, 
	malcolm.munro@durham.ac.uk

	William Cheng-Chung Chu (Co-chair for East), TungHai University, Taiwan, 
	chu@cis.thu.edu.tw

Local Arrangements co-chairs:
	Fabrizio Fioravanti, University of Florence, Italy, fioravan@dsi.unifi.it

	Pierfrancesco Bellini (Industrial Applications, and Demos), 
        University of Florence, Italy, 	bellini@hpcn.dsi.unifi.it

WEB Master:
	Marius Bogdan Spinu, University of Florence, Italy, spinu@hpcn.dsi.unifi.it

--------------------------------------------------------------------------------------------------------------



From rem-conf Wed Sep 20 08:12:08 2000 
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From: Baese Gero <Gero.X.Baese@mchp.siemens.de>
To: rem-conf@es.net
Subject: new version of the ID "An RTP Payload Format for Erasure-Resilien
	t Transmission of Progressive Multimedia Streams"
Date: Wed, 20 Sep 2000 17:09:05 +0200
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Dear Experts,

we are glad to announce the completion of the new version of the I-D "An RTP
Payload Format for Erasure-Resilient Transmission of Progressive Multimedia
Streams" (draft-lnt-avt-uxp-02). 
This document specifies an efficient way to ensure erasure-resilient
transmission of progressively encoded multimedia sources via RTP using
Reed-Solomon codes. The level of erasure protection can be explicitly
adapted to the importance of the respective parts in the source stream, thus
allowing a graceful degradation of application quality with increasing
packet loss rate on the network. Nevertheless, backward compatibility to
currently standardized non-progressive multimedia codecs is ensured, since
equal erasure protection (EXP) represents a subset of generic unequal
erasure protection (UXP). The new version includes effective in-band
signaling for erasure protection profiles.
We would highly appreciate any comments. If questions remaining don't
hesitate to contact anyone of us.

Best Wishes
Gero Baese






From rem-conf Wed Sep 20 09:39:15 2000 
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Subject: Re: new version of the ID "An RTP Payload Format for Erasure-Resilient 
 Transmission of Progressive Multimedia Streams"
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Baese Gero wrote:

> Dear Experts,
>
> we are glad to announce the completion of the new version of the I-D "An RTP
> Payload Format for Erasure-Resilient Transmission of Progressive Multimedia
> Streams" (draft-lnt-avt-uxp-02).
> This document specifies an efficient way to ensure erasure-resilient
> transmission of progressively encoded multimedia sources via RTP using
> Reed-Solomon codes. The level of erasure protection can be explicitly
> adapted to the importance of the respective parts in the source stream, thus
> allowing a graceful degradation of application quality with increasing
> packet loss rate on the network. Nevertheless, backward compatibility to
> currently standardized non-progressive multimedia codecs is ensured, since
> equal erasure protection (EXP) represents a subset of generic unequal
> erasure protection (UXP). The new version includes effective in-band
> signaling for erasure protection profiles.
> We would highly appreciate any comments. If questions remaining don't
> hesitate to contact anyone of us.
>
> Best Wishes
> Gero Baese

Hello;

 This (draft-lnt-avt-uxp-02) does not seem to be present in

http://www.ietf.org/ids.by.wg/none.html

Could you tell me where I can find it ?


--
                                 Regards
                                 Marshall Eubanks

T.M. Eubanks
Multicast Technologies, Inc
10301 Democracy Lane, Suite 410
Fairfax, Virginia 22030
Phone : 703-293-9624
Fax     : 703-293-9609
e-mail : tme@on-the-i.com     tme@multicasttech.com

http://www.on-the-i.com http://www.buzzwaves.com





From rem-conf Wed Sep 20 10:32:36 2000 
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Message-ID: <00a601c02327$441caac0$03000004@hariv.domain>
Reply-To: "Hari Krishna Vuppaladhadiam" <hariv@chiplogic.com>
From: "Hari Krishna Vuppaladhadiam" <hariv@chiplogic.com>
To: "Rosen, Brian" <Brian.Rosen@marconi.com>, <rem-conf@es.net>
Subject: Re: Voice Playout for VoATM 
Date: Wed, 20 Sep 2000 10:21:51 -0700
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Thanks Brian for the information.

I am trying to capture the differences as per the jitter buffer requirements
for VoIP and VoATM case. How does it differ for these two?

Regards
Hari V

-----Original Message-----
From: Rosen, Brian <Brian.Rosen@marconi.com>
To: 'Hari Krishna Vuppaladhadiam' <hariv@chiplogic.com>; rem-conf@es.net
<rem-conf@es.net>
Date: Wednesday, September 20, 2000 6:11 AM
Subject: RE: Voice Playout for VoATM


>You wouldn't use RTP for VoATM except for a case of VoIP over ATM.
>So, I suspect this group can't help you much.
>
>The ATM Forum has some specs on carrying voice.  The most recent
>is the BLES spec for VoDSL, which is ATM AAL2 based.
>The Q.BICC work in ITU covers VoATM using an extension to ISUP
>to carry an NSAP address.
>
>H.323 in ITU offers Annex C, which is an AAL5 VoATM bearer.
>
>Finally, there is work in MMUSIC to extend SDP to include
>ATM bearers, which would allow SIP, MEGACO and MGCP to
>support VoATM. MEGACO (H.248) has VoATM support in the
>"native" tags media format using binary encoding.  The SDP
>work extends similar support for text encoding.
>
>Brian
>
>> -----Original Message-----
>> From: Hari Krishna Vuppaladhadiam [mailto:hariv@chiplogic.com]
>> Sent: Tuesday, September 19, 2000 9:07 PM
>> To: rem-conf@es.net
>> Subject: Voice Playout for VoATM
>>
>>
>> Hi,
>>   I am looking for Voice playout schemes for VoATM. Are there any
>> references?
>> hari
>>
>>
>>
>




From rem-conf Wed Sep 20 11:30:16 2000 
From rem-conf-request@es.net Wed Sep 20 11:30:16 2000
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Date: Wed, 20 Sep 2000 11:26:17 -0700
From: John Lazzaro <lazzaro@CS.Berkeley.EDU>
Message-Id: <200009201826.LAA05456@snap.CS.Berkeley.EDU>
To: purnhage@tnt.uni-hannover.de, tmn@iis.fhg.de
Subject: Re: LATM specified in a public MPEG document?
Cc: grl@iis.fhg.de, lazzaro@CS.Berkeley.EDU, mpeg4_vm@tnt.uni-hannover.de,
        purnhagen@tnt.uni-hannover.de, rem-conf@es.net,
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[ Bodo Teichmann <tmn@iis.fhg.de> writes] 

> LATM was actually developed for only "natural"  audio coders, 
> i.e. not fo Structured Audio. it is not possible to transmitt
> multiple objects, no scene desctipion  and no time stamps
> can be transmitted . in your case you  better should  either use
> SyncLayer packets (thats an Acess unit with a Sync Layer header)
> with rtp or use flexmux with rtp. but both options are just
> drafts at the moment.

Thanks for the info, it sounds like the earlier suggestions I
made on the draft-ietf-avt-rtp-mpeg4-es-04.txt document about
Structured Audio issues, which made it into the current draft,
are inappropriate, since they assume the header described in
"4.1 RTP Packet Format" is meant for all audio, not only 
"natural" audio. Perhaps the I-D should simply state that 
Structured Audio isn't currently transportable over any of 
the packet formats described in the document.  

I'm currently prototyping using SA_access_units for MIDI 
messages for low-latency "jam session" sorts of applications,
and its becoming apparent that a SA-specific profile for RTP
(particularly RTCP) may have enough value to simply pursue
it as an another valid packetization scheme, described as an
I-D -- the draft-singer-mpeg4-ip-00.txt states that these
application-specific packetization schemes are appropriate
complements to the major ones like draft-ietf-avt-rtp-mpeg4-es-04.txt.

-------------------------------------------------------------------------
John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
-------------------------------------------------------------------------




From rem-conf Wed Sep 20 11:33:57 2000 
From rem-conf-request@es.net Wed Sep 20 11:33:57 2000
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Message-ID: <4FBEA8857476D311A03300204840E1CF01A6E6D6@whq-msgusr-02.pit.comms.marconi.com>
From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Hari Krishna Vuppaladhadiam'" <hariv@chiplogic.com>, rem-conf@es.net
Subject: RE: Voice Playout for VoATM 
Date: Wed, 20 Sep 2000 14:29:32 -0400
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There is no simple answer for VoATM, as the underlying
ATM network has mechanisms to control transmission characteristics the
affect jitter.  With ATM, one can maintain Stratum 2 timing
end to end and not need much jitter buffer.  Of course, one
can also use UBR, and be under the same basic position that
VoIP is.

I usually tell folks "less, how much less depends on what
you are doing".

Brian

> -----Original Message-----
> From: Hari Krishna Vuppaladhadiam [mailto:hariv@chiplogic.com]
> Sent: Wednesday, September 20, 2000 1:22 PM
> To: Rosen, Brian; rem-conf@es.net
> Subject: Re: Voice Playout for VoATM 
> 
> 
> Thanks Brian for the information.
> 
> I am trying to capture the differences as per the jitter 
> buffer requirements
> for VoIP and VoATM case. How does it differ for these two?
> 
> Regards
> Hari V
> 
> -----Original Message-----
> From: Rosen, Brian <Brian.Rosen@marconi.com>
> To: 'Hari Krishna Vuppaladhadiam' <hariv@chiplogic.com>; 
> rem-conf@es.net
> <rem-conf@es.net>
> Date: Wednesday, September 20, 2000 6:11 AM
> Subject: RE: Voice Playout for VoATM
> 
> 
> >You wouldn't use RTP for VoATM except for a case of VoIP over ATM.
> >So, I suspect this group can't help you much.
> >
> >The ATM Forum has some specs on carrying voice.  The most recent
> >is the BLES spec for VoDSL, which is ATM AAL2 based.
> >The Q.BICC work in ITU covers VoATM using an extension to ISUP
> >to carry an NSAP address.
> >
> >H.323 in ITU offers Annex C, which is an AAL5 VoATM bearer.
> >
> >Finally, there is work in MMUSIC to extend SDP to include
> >ATM bearers, which would allow SIP, MEGACO and MGCP to
> >support VoATM. MEGACO (H.248) has VoATM support in the
> >"native" tags media format using binary encoding.  The SDP
> >work extends similar support for text encoding.
> >
> >Brian
> >
> >> -----Original Message-----
> >> From: Hari Krishna Vuppaladhadiam [mailto:hariv@chiplogic.com]
> >> Sent: Tuesday, September 19, 2000 9:07 PM
> >> To: rem-conf@es.net
> >> Subject: Voice Playout for VoATM
> >>
> >>
> >> Hi,
> >>   I am looking for Voice playout schemes for VoATM. Are there any
> >> references?
> >> hari
> >>
> >>
> >>
> >
> 
> 



From rem-conf Wed Sep 20 19:58:06 2000 
From rem-conf-request@es.net Wed Sep 20 19:58:04 2000
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Message-ID: <007e01c02376$f2313c80$152c380a@dsp.cl.nec.co.jp>
From: "Toshiyuki Nomura" <t-nomura@ccm.cl.nec.co.jp>
To: "John Lazzaro" <lazzaro@cs.berkeley.edu>, <tmn@iis.fhg.de>
Cc: <grl@iis.fhg.de>, <mpeg4_vm@tnt.uni-hannover.de>,
        <purnhagen@tnt.uni-hannover.de>, <rem-conf@es.net>
References: <200009201826.LAA05456@snap.CS.Berkeley.EDU>
Subject: Re: LATM specified in a public MPEG document?
Date: Thu, 21 Sep 2000 11:52:12 +0900
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Dear John and Bodo,


> [ Bodo Teichmann <tmn@iis.fhg.de> writes] 
> 
> > LATM was actually developed for only "natural"  audio coders, 
> > i.e. not fo Structured Audio. it is not possible to transmitt
> > multiple objects, no scene desctipion  and no time stamps
> > can be transmitted . in your case you  better should  either use
> > SyncLayer packets (thats an Acess unit with a Sync Layer header)
> > with rtp or use flexmux with rtp. but both options are just
> > drafts at the moment.
> 
> Thanks for the info, it sounds like the earlier suggestions I
> made on the draft-ietf-avt-rtp-mpeg4-es-04.txt document about
> Structured Audio issues, which made it into the current draft,
> are inappropriate, since they assume the header described in
> "4.1 RTP Packet Format" is meant for all audio, not only 
> "natural" audio. Perhaps the I-D should simply state that 
> Structured Audio isn't currently transportable over any of 
> the packet formats described in the document.  

Simple SA session without scene description can be realized
by the LATM-based RTP specification.

Therefore, I'll modify the current draft to claim
"Since LATM has no funciton to handle scene description,
Structured Audio with scene description MUST NOT be 
transported using RTP packetization specified in 
this document".

Best Regards,

Toshiyuki Nomura, NEC
--------------------------




From rem-conf Thu Sep 21 00:29:50 2000 
From rem-conf-request@es.net Thu Sep 21 00:29:48 2000
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To: "Rosen, Brian" <Brian.Rosen@marconi.com>
cc: 'Hari Krishna Vuppaladhadiam' <hariv@chiplogic.com>, rem-conf@es.net
Subject: Re: Voice Playout for VoATM
In-reply-to: Your message of "Wed, 20 Sep 2000 14:29:32 EDT." <4FBEA8857476D311A03300204840E1CF01A6E6D6@whq-msgusr-02.pit.comms.marconi.com>
Date: Thu, 21 Sep 2000 08:19:35 +0100
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From: Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>
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In message <4FBEA8857476D311A03300204840E1CF01A6E6D6@whq-msgusr-02.pit.comms.ma
rconi.com>, "Rosen, Brian" typed:

 >>There is no simple answer for VoATM, as the underlying
 >>ATM network has mechanisms to control transmission characteristics the
 >>affect jitter.  With ATM, one can maintain Stratum 2 timing
 >>end to end and not need much jitter buffer.  Of course, one
 >>can also use UBR, and be under the same basic position that
 >>VoIP is.

right - you can run VOIP with 40 byte rtp+udp+ip headers over a TDM
network, and get whatever bit level jitte the TDM net guarantees, or
at the other weirder extreme, you can run PCM voice in ATM cells in
cells-in-frames on an ethernet - then you get absolutely nada in the
way of jitter control.....
 
 >>I usually tell folks "less, how much less depends on what
 >>you are doing".

usually less, but see above...

 
 >>
 >>> -----Original Message-----
 >>> From: Hari Krishna Vuppaladhadiam [mailto:hariv@chiplogic.com]
 >>> Sent: Wednesday, September 20, 2000 1:22 PM
 >>> To: Rosen, Brian; rem-conf@es.net
 >>> Subject: Re: Voice Playout for VoATM 
 >>> 
 >>> 
 >>> Thanks Brian for the information.
 >>> 
 >>> I am trying to capture the differences as per the jitter 
 >>> buffer requirements
 >>> for VoIP and VoATM case. How does it differ for these two?
 >>> 
 >>> Regards
 >>> Hari V
 >>> 
 >>> -----Original Message-----
 >>> From: Rosen, Brian <Brian.Rosen@marconi.com>
 >>> To: 'Hari Krishna Vuppaladhadiam' <hariv@chiplogic.com>; 
 >>> rem-conf@es.net
 >>> <rem-conf@es.net>
 >>> Date: Wednesday, September 20, 2000 6:11 AM
 >>> Subject: RE: Voice Playout for VoATM
 >>> 
 >>> 
 >>> >You wouldn't use RTP for VoATM except for a case of VoIP over ATM.
 >>> >So, I suspect this group can't help you much.
 >>> >
 >>> >The ATM Forum has some specs on carrying voice.  The most recent
 >>> >is the BLES spec for VoDSL, which is ATM AAL2 based.
 >>> >The Q.BICC work in ITU covers VoATM using an extension to ISUP
 >>> >to carry an NSAP address.
 >>> >
 >>> >H.323 in ITU offers Annex C, which is an AAL5 VoATM bearer.
 >>> >
 >>> >Finally, there is work in MMUSIC to extend SDP to include
 >>> >ATM bearers, which would allow SIP, MEGACO and MGCP to
 >>> >support VoATM. MEGACO (H.248) has VoATM support in the
 >>> >"native" tags media format using binary encoding.  The SDP
 >>> >work extends similar support for text encoding.
 >>> >
 >>> >Brian
 >>> >
 >>> >> -----Original Message-----
 >>> >> From: Hari Krishna Vuppaladhadiam [mailto:hariv@chiplogic.com]
 >>> >> Sent: Tuesday, September 19, 2000 9:07 PM
 >>> >> To: rem-conf@es.net
 >>> >> Subject: Voice Playout for VoATM
 >>> >>
 >>> >>
 >>> >> Hi,
 >>> >>   I am looking for Voice playout schemes for VoATM. Are there any
 >>> >> references?
 >>> >> hari
 >>> >>
 >>> >>
 >>> >>
 >>> >
 >>> 
 >>> 
 >>

 cheers

   jon




From rem-conf Thu Sep 21 01:13:16 2000 
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From: nile333@kadet.co.uk
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To: nile333@kadet.co.uk
Subject: So, How in the heck have you been?
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So, How in the heck have you been?

Do you remember holding previous conversations regarding business and
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little time to read it and REALLY THINK ABOUT IT. Get a pencil and
figure out what could happen when YOU participate. Figure out the worst
possible response and no matter how you calculate it, you will still
make a
lot of money! You will definitely get back what you invested. Any doubts
you
have will vanish when your first orders come in. $$$ IT WORKS!!! $$$

Jody Jacobs Richmond, VA.

HERES HOW THIS AMAZING PROGRAM WILL MAKE YOU THOUSANDS OF
DOLLARS$$$$!!!!

This method of raising capital REALLY WORKS 100% EVERY TIME. I am sure
that you could use up to $50,000 or more in the next 90 days. Before you
say
BULL, please read this program carefully. This is not a chain letter,but
a
perfectly legal money making business. As with all multi-level
businesses,
we build our business by recruiting new partners and selling our
products.
Every state in the USA allows you to recruit new multi-level business
partners, and we sell and deliver a product for EVERY dollar received.

YOUR ORDERS COME BY MAIL AND ARE FILLED BY E-MAIL, so you are not
involved in personal selling. You do it privately in your own home,
store or
office. This is the EASIEST marketing plan anywhere! It is simply order
filling by e-mail! The product is informational and instructional
material,
keys to the secrets for everyone on how to open the doors to the magic
world
of E-COMMERCE, the information highway, the wave of the future !

PLAN SUMMARY:

(1) You order the 4 reports listed below ($5 each) They come to you by
e-mail.

(2)  Save a copy of this entire letter and put your name after Report #1
and
move the other names down.

(3)  Via the internet, access Yahoo.com or any of the other major search

engines to locate hundreds of bulk e-mail service companies (search for
bulk
email) and have them send 25,000  50,000 emails for you about $49+.

(4)  Orders will come to you by postal mail simply e-mail them the
Report they ordered. Let me ask you  isnt this about as easy as it
gets?

By the way there are over 50 MILLION e-mail address with millions more
joining the internet each year so dont worry about running out or
saturation. People are used to seeing and hearing the same
advertisements every day on radio/TV. How many times have you received
the same pizza flyers on your door? Then one day you are hungry for
pizza
and order one. Same thing with this letter. I received this letter many
times  then one day I decided it was time to try it.

YOU CAN START TODAY UST DO THESE EASY STEPS: STEP #1 ORDER THE FOUR
REPORTS

Order the four reports shown on the list below (you cant sell them if
you dont order them).  For each report, send $5.00 CASH, the NAME &
NUMBER
OF THE REPORT YOU ARE ORDERING, YOUR E-MAIL ADDRESS, and YOUR NAME &
RETURN
ADDRESS (in case of a problem) to the person whose name appears on the
list
next to the report.MAKE SURE YOUR RETURN ADDRESS IS ON YOUR ENVELOPE IN
CASE OF ANY MAIL PROBLEMS! Within a few days you will receive, by e-mail

each of the four reports.Save them on your computer so you can send them
to
the 1,000s of people who will  order them from you.

STEP #2. ADD YOUR MAILING ADDRESS TO THIS LETTER

a. Look below for the listing of the four reports.
b. After youve ordered the four reports, delete the name and address
under REPORT #4. This person has made it through the cycle.
c. Move the name and address under REPORT #3 down to REPORT #4.
d. Move the name and address under REPORT #2 down to REPORT #3.
e. Move the name and address under REPORT #1 down to REPORT #2.
f. Insert your name/address in the REPORT #1 position. Please make sure
you

COPY ALL INFORMATION, every name and address, ACCURATELY!

STEP #3. Take this entire letter, including the modified list of names,
and save it to your computer. Make NO changes to these instructions. Now
you
are ready to use this entire e-mail to send by e-mail to prospects.

Report #1 will tell you how to download bulk email software and email
address so you can send it out to thousands of people while you sleep!
Remember that 50,000+ new people are joining the internet every month!
Your cost to participate in this is practically nothing ( surely you can

afford $20 and initial bulk mailing cost). You obviously already have a
computer and an Internet connection and e-mail is FREE! There are two
primary methods of building your downline: METHOD #1: SENDING BULK
E-MAIL
lets say that you decide to start small, just to see how it goes, and
well
assume you and all those involved email out only 2,000 programs each.
Lets
also assume that the mailing receives a 0.5% response. The response
could be
much better. Also, many people will email out thousands of thousands of
programs instead of 2,000 (Why stop at 2000?) But continuing with this
example, you send out only 2,000 programs. With a 0.5% response, that is

only 10 orders for REPORT #1. Those 10 people respond by sending out
2,000
programs each for a total of 20,000. Out of those 0.5%, 100 people
respond
and order REPORT #2.Those 100 mail out 2,000 programs each for a total
of
200,000. The 0.5% response to that is 1,000 orders for REPORT #3. Those
1,000 send out 2,000  programs each for a 2,000,000 total. The 0.5%
response
to that is 10,000 orders for REPORT #4. Thats 10,000 $5 bills for you.
CASH!!! Your total income in this example is $50 + $500 + $5000 +
$50,000
for a total of $55,550!!!

REMEMBER FRIEND, THIS IS ASSUMING 1,990 OUT OF THE 2,000 PEOPLE YOU MAIL
TO
WILL DO ABSOLUTELY NOTHING AND TRASH THIS PROGRAM! DARE TO THINK FOR A
MOMENT WHAT WOULD HAPPEN IF EVERYONE, OR HALF SENT OUT 100,000 PROGRAMS
INSTEAD OF 2,000. Believe me, many people will do just that, and more!

METHOD #2 PLACING FREE ADS ON THE INTERNET Advertising on the internet
is very, very inexpensive, and there are HUNDREDS of FREE places to
advertise. Lets say you decide to start small to see how well it works.

Assume your goal is to get ONLY 10 people to participate on your first
level. (Placing a lot of FREE ads on the Internet will EASILY get a
larger
response). Also assume that everyone else in YOUR ORGANIZATION gets only
10
downline members. Look how this small number accumulates to achieve the
STAGGERING results below:

1St level  your first 10 send you $5........................$50
2nd level  10 members from those 10 ($5 x 100)............$500
3rd level  10 members from those 100 ($5 x 1,000)......$5,000
4th level 10 members from those 1,000 ($5 x 10,000)..$50,000
$$$$$$ THIS TOTALS
------------------------------------------------55,5550
$$$$$

AMAZING ISNT IT Remember friends, this assumes that the people who
participate only recruit 10 people each. Think for a moment what would
happen if they got 20 people to participate! Most people get 100s of
participants and many will continue to work this program, sending out
programs WITH YOUR NAME ON THEM for years! THINK ABOUT IT!
People are going to get emails about this plan from you or somebody else
and
many will work this plan  the question is Dont you want your name to be
on
the emails they will send out?

*** DONT MISS OUT !!!***
***JUST TRY IT ONCE !!!***
***SEE WHAT HAPPENS !!!***
***YOU'LL BE AMAZED !!!***

ALWAYS PROVIDE SAME DAY SERVICE ON ALL ORDERS! This will guarantee that
the e-mail THEY send out with YOUR name and address on it will be prompt

because they cant advertise until they receive the report!

GET STARTED TODAY: PLACE YOUR ORDER FOR THE FOUR REPORTS NOW. Note:--
ALWAYS SEND $5 CASH (U.S. CURRENCY) FOR EACH REPORT. CHECKS NOT
ACCEPTED.
Make sure the cash is concealed by wrapping it in two sheets of paper.
On
one of those sheets write:

(a) the number & name of the report you are ordering
(b) your e-mail address, and
(c) your name & postal address.

REPORT #1b The Insiders Guide to Advertising for Free on the Internet
ORDER REPORT #1 FROM:

NICK NICHOLAS
473 MICHIGAN ST
ST.PAUL, MN 55102

NOTE: I and every member below are dedicated at helping you with this
program so it will work for you also. TRY US!

REPORT #2 The Insiders Guide to Sending Bulk E-Mail on the Internet
ORDER REPORT #2 FROM:

DIANE COLON
1811 TAMARIND AVE # 206
LOS ANGELES, CA. 90028

REPORT #3 The Secrets to Multilevel Marketing on the Internet
ORDER REPORT #3 FROM:

MELISSA HOGENMILLER
3709 MONHEIM ROAD
CONOVER, WI 54519

REPORT #4 How to become a Millionaire utilizing the Power of Multilevel
Marketing and the Internet
ORDER REPORT #4 FROM:

CATHY BARROW
10 SYCAMORE STREET
CONWAY, SC 29527

*************TIPS FOR SUCCESS***************
TREAT THIS AS YOUR BUSINESS! Be prompt, professional, and follow the
directions accurately.  Send for the four reports IMMEDIATELY so you
will have them when the orders start coming in because: When you
receive a $5 order you MUST send out the requested product/report.
It is required for this to be a legal business and they need the
reports to send out their letter (with your name on them).

--ALWAYS PROVIDE SAME-DAY SERVICE ON THE ORDERS YOU RECEIVE. Be
patient and persistent with this program- If you follow the
instructions exactly results WILL FOLLOW. $$$$

************ YOUR SUCCESS GUIDELINES ***************

Follow these guidelines to guarantee your success: If you dont receive
20 orders for REPORT #1 within two weeks, continue advertising or
sending
e-mail until you do. Then a couple of weeks later you should receive at
least 100 orders for REPORT #2. If you dont continue advertising or
sending
e-mail until you do. Once you have received 100 or more orders for
REPORT
#2, YOU CAN RELAX, because the system is already working for you, and
the
cash will continue to roll in! THIS IS IMPORTANT TO REMEMBER:  Every
time
your name is moved down on the list, you are placed in front of a
DIFFERENT
report. You can KEEP TRACK of your  PROGRESS by watching which report
people
are ordering from you. To generate more income, simply send another
batch of
e-mails or continue placing ads and start the whole process again! There
is
no limit to the income you will generate from this business! Before you
make
your decision as to whether or not you participate in this program.
Please
answer one question:

ARE YOU HAPPY WITH YOUR PRESENT INCOME OR JOB?

1. If the answer is no, then please look at the following facts about
this super simple MLM program: NO face to face selling, NO meetings, NO
inventory! NO Telephone calls, NO big cost to start! Nothing to learn,
No skills needed! (Surely you know how to send email?)

2. No equipment to buy you already have a computer and internet
connection so you have everything you need to fill orders!

3. You are selling a product which does NOT COST ANYTHING TO PRODUCE OR
SHIP! (Email copies of the reports are FREE!)

4. All of your customers pay you in CASH! This program will change your
LIFE  FOREEVER!! Look at the potential for you to be able to quit your
job and live a life of luxury you could only dream about! Imagine
getting out of debt and buying the car and home of your dreams and
being able to work a super-high paying leisurely easy business from
home!

$$$ FINALLY MAKE SOME DREAMS COME TRUE! $$$ ACT NOW!
Take your first step toward achieving financial independence.  Order
the reports and follow the program outlined above __ SUCCESS will be
your reward.

Thank you for your time and consideration. PLEASE NOT: If you need
help with starting a business, registering a business name, learning
now income tax is handled, etc., contact your local office of the
Small Business Administration  (A Federal Agency) 1-800-827-5722
for free help and answers to questions. Also the Internal Revenue
Service offers free help via telephone and free seminars about
business tax requirements. Your earnings are highly dependent on
your activities and advertising. The information contained on this
site and in the report constitutes no guarantees stated nor implied.
In the event that it is determined that this site or report
constitutes a guarantee of any kind, that guarantee is now void. The
earnings amounts listed on this site and in the report are estimates
only. If you have any questions of the legality of this program,
contact the Office of Associate Director for Marketing Practices,
Federal Trade Commission, Bureau of Consumer Protection in
Washington DC.

Under Bill s.1618 TITLE III passed by the 105th US Congress this
letter cannot be considered spam as long as the sender includes
contact information and a method of removal. This is a one time
e-mail transmission. No request for removal is necessary.






From rem-conf Thu Sep 21 01:30:11 2000 
From rem-conf-request@es.net Thu Sep 21 01:30:11 2000
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Date: Thu, 21 Sep 2000 03:28:08 -0500
From: "Brian F. G. Bidulock" <bidulock@dallas.net>
To: rem-conf@es.net
Subject: Re: So, How in the heck have you been?
Message-ID: <20000921032808.A7095@dallas.net>
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Organization: Brian F. G. Bidulock, P. Eng.
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Could someone PLEEEEEASE put a spam filter on this list!

--brian

On Thu, 21 Sep 2000, nile333@kadet.co.uk wrote:

>=20
> So, How in the heck have you been?
>=20
> Do you remember holding previous conversations regarding business and
> money making opportunities? I did not send this to you in error!
>=20
> You Said:
>=20
> If only I could find an easier way to make a higher income!
>=20
> and
>=20
> If I had more money, I could spend more time with my Family, and less
> time at work and I sure could use more money so I could pay off my
> bills once and for all!
>=20
> And
>=20
> I would love to get involved in a business in which will generate money
> while I am not at work (like a Gas Pump)!
>=20
> Dear Friend,
>=20
> There is a possibility that we haven=92t met, but you were chosen by
> someone to receive this E-Mail. Please, please, print this off and
> read thoroughly. Be sure that you don=92t miss any of the points
> outlined.  Then put it down, and then read it again. I am sending
> you a whole lot of information in which you might not understand
> the first time you read it. If you don=92t believe this  program
> will work for you, send it to 10-20 of your closest friends
> (in which you trust deeply),  and ask them what they think?
> This really works! Have faith, don=92t miss this opportunity,
> get involved also, and it will work for you as it does for us!!!!
>=20
> Due to the popularity of this letter on the Internet, A Major Nightly
> News Program recently dedicated an entire show to the investigation of
> the
> program described below to see if it really can make people money.
> The show also investigated whether or not the program was legal. Their
> findings proved that there are absolutely no laws prohibiting the
> participation in the program. This has helped to show people that this
> is a simple, harmless and fun way to make extra money at home. The
> results have been truly remarkable. So many people are participating
> that those involved are doing much better than ever before. Since
> everyone makes more as more people try it out, its been very exciting.
>=20
> You will understand only if you get involved!
> ********** THE ENTIRE PLAN IS HERE BELOW **********
> **** Print This Now For Future Reference ****
>=20
> $$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$
> If you would like to make AT LEAST $50,000 in less than 90 days! If not=
,
>=20
> forward this to someone who would like to make this kind of money.
> It works (like designed) but only for those who follow it to the letter=
!
>=20
> Please read this program THEN READ IT AGAIN!!
> $$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$
>=20
> THIS IS A LEGITIMATE. LEGAL, MONEY MAKING OPPORTUNITY!! It does NOT
> require you to come into contact with people or make or take any
> telephone
> calls. Just follow the instructions, and you will make money. This
> simplified e-mail marketing program works perfectly 100% EVERY TIME!
>=20
> E-mail is the sales tool of the future. Take advantage of this virtuall=
y
>=20
> free method of advertising NOW!!! The longer you wait, the more people
> will
> be doing business using e-mail. Get your piece of this action!!!
>=20
> Hello, My name is Johnathon Rourke, I=92m from Rhode Island.  The enclo=
sed
>=20
> information is something I almost let slip through my fingers.
> Fortunately, sometime later I re-read everything and gave some thought
> and study to it. Two years ago, the corporation I worked for the past
> twelve yearsdown-sized and my position was eliminated. After
> unproductive
> job interviews, I decided to open my own business. Over the past year,I
> incurred many unforeseen financial problems. I owed my family, friends
> and
> creditors over$35,000. The economy was taking a toll on my business and
> I
> just could not seem to make ends meet. I had to refinance and borrow
> against
> my home to support my family and struggling business.
>=20
> AT THAT MOMENT something significant happened in my life. I am writing
> to share the experience I hopes that this could change your life
> FOREVER.
> FINANCIALLY$$$!!!
>=20
> In mid December, I received this program in my e-mail. Six months prior
> to
> receiving this program I had been sending away for information on
> various
> business opportunities. All of the programs I received, in my
> opinion,were
> not cost effective. They were either toodifficult for me to comprehend
> or
> the initial investment was too muchfor me to risk to see if they would
> work.
> But as I was saying, in December of 1997 I received this program.I
> didn=92t
> send for it, or ask for it, they just got my name off a mailing list.
>=20
> THANK GOODNESS FOR THAT!!!
>=20
> After reading it several times, to make sure I was reading it correctly.
> I
> couldn=92t believe my eyes! Here was a MONEY MAKING MACHINE I could sta=
rt
> immediately without any debt. Like most of you I was still a little
> skeptical and a little worried about the legalaspects of it all. So I
> checked it out with the U.S. Post Office (1-800-725-2161 24-hrs)  and
> they
> confirmed that it is indeed legal ! After determining the program was
> LEGAL
> I decided WHY NOT!?!??
>=20
> Initially I sent out 10,000 e-mails. It cost me about $15 for my time
> on-line. The great thing about e-mail is that I don=92t need any paper =
for
>=20
> printing to send out the program, and because I also send the product
> (reports) by e-mail, my only expense is my time. In less than one week,=
I
> was
> starting to receive orders for REPORT #1.
>=20
> By January 13, I had received 26 orders for REPORT #1. Your goal is to
> RECEIVE at least 20 ORDERS FOR REPORT #1 WITHIN 2 WEEKS. IF YOU DON=92T
> SEND OUT MORE PROGRAMS UNTIL YOU DO. My first step in making $50,000 in
> 90
> days was done. By January 30, I had received 196 orders for REPORT #2.
> Your
> goal is to RECEIVE AT LEAST 100+ ORDERS FOR REPORT #2 WITHIN
> 2 WEEKS. IF NOT SEND OUT MORE PROGRAMS UNTIL YOU! DO. ONCE YOU HAVE
> 100 ORDERS, THE REST IS EASY, RELAX, YOU WILL MAKE YOUR $50,000 GOAL.
>=20
> Well, I had 196 orders for REPORT #2. 96 more than I needed. So I
> sat back and relaxed.
>=20
> By March 1, of my e-mailing of 10,000, received $58,000 with more comin=
g
> in
> every day. I paid off ALL my debts and bought a much need new car!
> Please
> take your time to read this plan, IT WILL CHANGE YOUR LIFE FOREVER$!!!
> Remember, it won=92t work if you don=92t try it. This program does work=
, But
> you
> must follow it EXACTLY! Especially the rules of not trying to place you=
r
>=20
> name in a different place. It won=92t work and you=92ll lose out on a l=
ot of
>=20
> money! In order for this program to work, you must meet your goal of 20=
+
>=20
> orders for REPORT #1, and 100+ orders for REPORT #2 and you will make
> $50,000 or more in 90 days.
>=20
> I AM LIVING PROOF THAT IT WORKS!!!
>=20
> If you choose not to participate in this program, I am sorry. It really
> is a great opportunity with little cost or risk to you.  If you choose
> toparticipate, follow the program and you will be on your way to
> financial security. If you are a fellow business owner and
> are financial trouble like I was, or you want to start your own
> business, consider this a sign. I DID! $$
>=20
> Sincerely,
> Johnathon Rourke
>=20
> A PERSONAL NOTE FROM THE ORIGINATOR OF THIS PROGRAM: By the time you
> have read the enclosed program and reports, you should have concluded
> that
> such a program, and one that is legal, cpuld not have been created by a=
n
>=20
> amateur. Let me tell you a little about myself. I had a profitable
> business
> for 10 years. Then in 1979 my business began falling off. I was doing
> the
> same things that were previously successful for me, but it wasn=92t
> working.
> Finally, I figured it out. It wasn=92t me, it was the economy. Inflatio=
n
> and
> recession had replaced the stable economy that had been with us since
> 1945.
> I don=92t have to tell you what happened to the unemployment rate becau=
se
> many
> of you know from first hand experience. There were more failures and
> bankruptcies than ever before. The middle class was vanishing. Those wh=
o
>=20
> knew what they were doing invested wisely and moved up. Those who did
> not,
> including those who never had anything to save or invest, were moving
> down into the ranks of the poor. As the saying goes, THE RICH GET RICHE=
R
>=20
> ANDTHE POOR GET POORER.  The traditional methods of making money will
> never
> allow you to move up or get rich, inflation will see to that You have
> just
> received the rest of  your life, with NO RISK and JUST A LITTLE BIT OF
> EFFORT. You can make more money in the next few months than you have
> everimagined.I should also point out that I will not see a penny of thi=
s
>=20
> money, nor anyone else who has provided a testimonial for this program.
> I
> retired from the program after sending thousands and thousands of
> programs.
> Follow the program EXACTLY AS INSTRUCTED. Do not change it in any way.
> It
> works exceedingly well as it is now. Remember to e-mail a copyof this
> exciting report to everyone you can think of. One of the people you sen=
d
>=20
> this to may send out 50,000 and your name will be on everyone of them!
> REMEMBER though, ------ the MORE YOU SEND OUT, the more potential
> customers
> you will reach. So my friend, I have given you the ideas,  information,
> materials and opportunity to become financially independent.
>=20
> IT IS UP TO YOU!! NOW DO IT!!
>=20
> BEFORE YOU delete this program from your in box, as I almost did, take =
a
>=20
> little time to read it and REALLY THINK ABOUT IT. Get a pencil and
> figure out what could happen when YOU participate. Figure out the worst
> possible response and no matter how you calculate it, you will still
> make a
> lot of money! You will definitely get back what you invested. Any doubt=
s
> you
> have will vanish when your first orders come in. $$$ IT WORKS!!! $$$
>=20
> Jody Jacobs Richmond, VA.
>=20
> HERE=92S HOW THIS AMAZING PROGRAM WILL MAKE YOU THOUSANDS OF
> DOLLARS$$$$!!!!
>=20
> This method of raising capital REALLY WORKS 100% EVERY TIME. I am sure
> that you could use up to $50,000 or more in the next 90 days. Before yo=
u
> say
> BULL, please read this program carefully. This is not a chain letter,bu=
t
> a
> perfectly legal money making business. As with all multi-level
> businesses,
> we build our business by recruiting new partners and selling our
> products.
> Every state in the USA allows you to recruit new multi-level business
> partners, and we sell and deliver a product for EVERY dollar received.
>=20
> YOUR ORDERS COME BY MAIL AND ARE FILLED BY E-MAIL, so you are not
> involved in personal selling. You do it privately in your own home,
> store or
> office. This is the EASIEST marketing plan anywhere! It is simply order
> filling by e-mail! The product is informational and instructional
> material,
> keys to the secrets for everyone on how to open the doors to the magic
> world
> of E-COMMERCE, the information highway, the wave of the future !
>=20
> PLAN SUMMARY:
>=20
> (1) You order the 4 reports listed below ($5 each) They come to you by
> e-mail.
>=20
> (2)  Save a copy of this entire letter and put your name after Report #=
1
> and
> move the other names down.
>=20
> (3)  Via the internet, access Yahoo.com or any of the other major searc=
h
>=20
> engines to locate hundreds of bulk e-mail service companies (search for
> bulk
> email) and have them send 25,000  50,000 emails for you about $49+.
>=20
> (4)  Orders will come to you by postal mail simply e-mail them the
> Report they ordered. Let me ask you  isn=92t this about as easy as it
> gets?
>=20
> By the way there are over 50 MILLION e-mail address with millions more
> joining the internet each year so don=92t worry about running out or
> saturation. People are used to seeing and hearing the same
> advertisements every day on radio/TV. How many times have you received
> the same pizza flyers on your door? Then one day you are hungry for
> pizza
> and order one. Same thing with this letter. I received this letter many
> times  then one day I decided it was time to try it.
>=20
> YOU CAN START TODAY UST DO THESE EASY STEPS: STEP #1 ORDER THE FOUR
> REPORTS
>=20
> Order the four reports shown on the list below (you can=92t sell them i=
f
> you don=92t order them).  For each report, send $5.00 CASH, the NAME &
> NUMBER
> OF THE REPORT YOU ARE ORDERING, YOUR E-MAIL ADDRESS, and YOUR NAME &
> RETURN
> ADDRESS (in case of a problem) to the person whose name appears on the
> list
> next to the report.MAKE SURE YOUR RETURN ADDRESS IS ON YOUR ENVELOPE IN
> CASE OF ANY MAIL PROBLEMS! Within a few days you will receive, by e-mai=
l
>=20
> each of the four reports.Save them on your computer so you can send the=
m
> to
> the 1,000=92s of people who will  order them from you.
>=20
> STEP #2. ADD YOUR MAILING ADDRESS TO THIS LETTER
>=20
> a. Look below for the listing of the four reports.
> b. After you=92ve ordered the four reports, delete the name and address
> under REPORT #4. This person has made it through the cycle.
> c. Move the name and address under REPORT #3 down to REPORT #4.
> d. Move the name and address under REPORT #2 down to REPORT #3.
> e. Move the name and address under REPORT #1 down to REPORT #2.
> f. Insert your name/address in the REPORT #1 position. Please make sure
> you
>=20
> COPY ALL INFORMATION, every name and address, ACCURATELY!
>=20
> STEP #3. Take this entire letter, including the modified list of names,
> and save it to your computer. Make NO changes to these instructions. No=
w
> you
> are ready to use this entire e-mail to send by e-mail to prospects.
>=20
> Report #1 will tell you how to download bulk email software and email
> address so you can send it out to thousands of people while you sleep!
> Remember that 50,000+ new people are joining the internet every month!
> Your cost to participate in this is practically nothing ( surely you ca=
n
>=20
> afford $20 and initial bulk mailing cost). You obviously already have a
> computer and an Internet connection and e-mail is FREE! There are two
> primary methods of building your downline: METHOD #1: SENDING BULK
> E-MAIL
> let=92s say that you decide to start small, just to see how it goes, an=
d
> we=92ll
> assume you and all those involved email out only 2,000 programs each.
> Let=92s
> also assume that the mailing receives a 0.5% response. The response
> could be
> much better. Also, many people will email out thousands of thousands of
> programs instead of 2,000 (Why stop at 2000?) But continuing with this
> example, you send out only 2,000 programs. With a 0.5% response, that i=
s
>=20
> only 10 orders for REPORT #1. Those 10 people respond by sending out
> 2,000
> programs each for a total of 20,000. Out of those 0.5%, 100 people
> respond
> and order REPORT #2.Those 100 mail out 2,000 programs each for a total
> of
> 200,000. The 0.5% response to that is 1,000 orders for REPORT #3. Those
> 1,000 send out 2,000  programs each for a 2,000,000 total. The 0.5%
> response
> to that is 10,000 orders for REPORT #4. That=92s 10,000 $5 bills for yo=
u.
> CASH!!! Your total income in this example is $50 + $500 + $5000 +
> $50,000
> for a total of $55,550!!!
>=20
> REMEMBER FRIEND, THIS IS ASSUMING 1,990 OUT OF THE 2,000 PEOPLE YOU MAI=
L
> TO
> WILL DO ABSOLUTELY NOTHING AND TRASH THIS PROGRAM! DARE TO THINK FOR A
> MOMENT WHAT WOULD HAPPEN IF EVERYONE, OR HALF SENT OUT 100,000 PROGRAMS
> INSTEAD OF 2,000. Believe me, many people will do just that, and more!
>=20
> METHOD #2 PLACING FREE ADS ON THE INTERNET Advertising on the internet
> is very, very inexpensive, and there are HUNDREDS of FREE places to
> advertise. Let=92s say you decide to start small to see how well it wor=
ks.
>=20
> Assume your goal is to get ONLY 10 people to participate on your first
> level. (Placing a lot of FREE ads on the Internet will EASILY get a
> larger
> response). Also assume that everyone else in YOUR ORGANIZATION gets onl=
y
> 10
> downline members. Look how this small number accumulates to achieve the
> STAGGERING results below:
>=20
> 1St level  your first 10 send you $5........................$50
> 2nd level  10 members from those 10 ($5 x 100)............$500
> 3rd level  10 members from those 100 ($5 x 1,000)......$5,000
> 4th level 10 members from those 1,000 ($5 x 10,000)..$50,000
> $$$$$$ THIS TOTALS
> ------------------------------------------------55,5550
> $$$$$
>=20
> AMAZING ISN=92T IT Remember friends, this assumes that the people who
> participate only recruit 10 people each. Think for a moment what would
> happen if they got 20 people to participate! Most people get 100=92s of
> participants and many will continue to work this program, sending out
> programs WITH YOUR NAME ON THEM for years! THINK ABOUT IT!
> People are going to get emails about this plan from you or somebody els=
e
> and
> many will work this plan  the question is Don=92t you want your name to=
 be
> on
> the emails they will send out?
>=20
> *** DON=92T MISS OUT !!!***
> ***JUST TRY IT ONCE !!!***
> ***SEE WHAT HAPPENS !!!***
> ***YOU'LL BE AMAZED !!!***
>=20
> ALWAYS PROVIDE SAME DAY SERVICE ON ALL ORDERS! This will guarantee that
> the e-mail THEY send out with YOUR name and address on it will be promp=
t
>=20
> because they can=92t advertise until they receive the report!
>=20
> GET STARTED TODAY: PLACE YOUR ORDER FOR THE FOUR REPORTS NOW. Note:--
> ALWAYS SEND $5 CASH (U.S. CURRENCY) FOR EACH REPORT. CHECKS NOT
> ACCEPTED.
> Make sure the cash is concealed by wrapping it in two sheets of paper.
> On
> one of those sheets write:
>=20
> (a) the number & name of the report you are ordering
> (b) your e-mail address, and
> (c) your name & postal address.
>=20
> REPORT #1b The Insider=92s Guide to Advertising for Free on the Interne=
t
> ORDER REPORT #1 FROM:
>=20
> NICK NICHOLAS
> 473 MICHIGAN ST
> ST.PAUL, MN 55102
>=20
> NOTE: I and every member below are dedicated at helping you with this
> program so it will work for you also. TRY US!
>=20
> REPORT #2 The Insider=92s Guide to Sending Bulk E-Mail on the Internet
> ORDER REPORT #2 FROM:
>=20
> DIANE COLON
> 1811 TAMARIND AVE # 206
> LOS ANGELES, CA. 90028
>=20
> REPORT #3 The Secrets to Multilevel Marketing on the Internet
> ORDER REPORT #3 FROM:
>=20
> MELISSA HOGENMILLER
> 3709 MONHEIM ROAD
> CONOVER, WI 54519
>=20
> REPORT #4 How to become a Millionaire utilizing the Power of Multilevel
> Marketing and the Internet
> ORDER REPORT #4 FROM:
>=20
> CATHY BARROW
> 10 SYCAMORE STREET
> CONWAY, SC 29527
>=20
> *************TIPS FOR SUCCESS***************
> TREAT THIS AS YOUR BUSINESS! Be prompt, professional, and follow the
> directions accurately.  Send for the four reports IMMEDIATELY so you
> will have them when the orders start coming in because: When you
> receive a $5 order you MUST send out the requested product/report.
> It is required for this to be a legal business and they need the
> reports to send out their letter (with your name on them).
>=20
> --ALWAYS PROVIDE SAME-DAY SERVICE ON THE ORDERS YOU RECEIVE. Be
> patient and persistent with this program- If you follow the
> instructions exactly results WILL FOLLOW. $$$$
>=20
> ************ YOUR SUCCESS GUIDELINES ***************
>=20
> Follow these guidelines to guarantee your success: If you don=92t recei=
ve
> 20 orders for REPORT #1 within two weeks, continue advertising or
> sending
> e-mail until you do. Then a couple of weeks later you should receive at
> least 100 orders for REPORT #2. If you don=92t continue advertising or
> sending
> e-mail until you do. Once you have received 100 or more orders for
> REPORT
> #2, YOU CAN RELAX, because the system is already working for you, and
> the
> cash will continue to roll in! THIS IS IMPORTANT TO REMEMBER:  Every
> time
> your name is moved down on the list, you are placed in front of a
> DIFFERENT
> report. You can KEEP TRACK of your  PROGRESS by watching which report
> people
> are ordering from you. To generate more income, simply send another
> batch of
> e-mails or continue placing ads and start the whole process again! Ther=
e
> is
> no limit to the income you will generate from this business! Before you
> make
> your decision as to whether or not you participate in this program.
> Please
> answer one question:
>=20
> ARE YOU HAPPY WITH YOUR PRESENT INCOME OR JOB?
>=20
> 1. If the answer is no, then please look at the following facts about
> this super simple MLM program: NO face to face selling, NO meetings, NO
> inventory! NO Telephone calls, NO big cost to start! Nothing to learn,
> No skills needed! (Surely you know how to send email?)
>=20
> 2. No equipment to buy you already have a computer and internet
> connection so you have everything you need to fill orders!
>=20
> 3. You are selling a product which does NOT COST ANYTHING TO PRODUCE OR
> SHIP! (Email copies of the reports are FREE!)
>=20
> 4. All of your customers pay you in CASH! This program will change your
> LIFE  FOREEVER!! Look at the potential for you to be able to quit your
> job and live a life of luxury you could only dream about! Imagine
> getting out of debt and buying the car and home of your dreams and
> being able to work a super-high paying leisurely easy business from
> home!
>=20
> $$$ FINALLY MAKE SOME DREAMS COME TRUE! $$$ ACT NOW!
> Take your first step toward achieving financial independence.  Order
> the reports and follow the program outlined above __ SUCCESS will be
> your reward.
>=20
> Thank you for your time and consideration. PLEASE NOT: If you need
> help with starting a business, registering a business name, learning
> now income tax is handled, etc., contact your local office of the
> Small Business Administration  (A Federal Agency) 1-800-827-5722
> for free help and answers to questions. Also the Internal Revenue
> Service offers free help via telephone and free seminars about
> business tax requirements. Your earnings are highly dependent on
> your activities and advertising. The information contained on this
> site and in the report constitutes no guarantees stated nor implied.
> In the event that it is determined that this site or report
> constitutes a guarantee of any kind, that guarantee is now void. The
> earnings amounts listed on this site and in the report are estimates
> only. If you have any questions of the legality of this program,
> contact the Office of Associate Director for Marketing Practices,
> Federal Trade Commission, Bureau of Consumer Protection in
> Washington DC.
>=20
> Under Bill s.1618 TITLE III passed by the 105th US Congress this
> letter cannot be considered spam as long as the sender includes
> contact information and a method of removal. This is a one time
> e-mail transmission. No request for removal is necessary.
>=20
>=20
>=20

--=20
Brian F. G. Bidulock, P. Eng.
bidulock@dallas.net



From rem-conf Thu Sep 21 05:30:29 2000 
From rem-conf-request@es.net Thu Sep 21 05:30:28 2000
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To: <rem-conf@es.net>, <4on2andIP-sys@advent.ee.columbia.edu>
Subject: Types of MPEG-4 streams over IP
Message-ID: <0056890014249640000002L902*@MHS>
Date: Thu, 21 Sep 2000 14:25:05 +0200
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Dear all,

I would like to verify my understanding on the types of MPEG-4 streams =
that
need tools for transport over IP. I summarized my understanding by the
following list :

- ObjectDescriptorStream
- ClockReferenceStream (is this needed over IP ?)
- SceneDescriptionStream
- VisualStream (with and without use of MPEG-4 Systems)
- AudioStream (with and without use of MPEG-4 Systems)
- MPEG-7Stream
- IPMPStream
- ObjectContentInfoStream
- FlexMux stream
- MP4 file

For MP4 files only a standard MIME type is required. For all other stre=
ams
also use of RTP and SDP need to be defined. Please correct me if I am w=
rong.

The framework document (Singer et al) indicates that also a MIME type i=
s
needed for an MPEG-4 MuxCodeTable. Is this required indeed, or can we a=
ssume
(/require) that the MuxCodeTable is transported inside the same FlexMux=

stream (in a packet with simple mode). This could provide an easy solut=
ion
for the timing of applicability of the table that would need to be reso=
lved
otherwise when using e.g. HTTP; however it would require MuxCodeTable t=
o
become a new entry in the table for streamType values, right ?

Regards,

Jan

************************
Jan van der Meer
Philips - Digital Networks
Building OAN-4
P.O. Box 80002
5600 JB Eindhoven
The Netherlands
Phone +31 402 735 774
Fax +31 402 735 545
Email jan.vandermeer@philips.com
************************


=



From rem-conf Thu Sep 21 10:50:16 2000 
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Date: Thu, 21 Sep 2000 13:16:31 -0400
From: "Luthra, Ajay (SD-EX)" <ALuthra@gi.com>
Subject: RE: comments on in-band-signaling proposal
To: 'Zvi Lifshitz' <zvil@csi.com>,
 "'Narasimhan, Sam (SD-EX)'" <SNarasimhan@GI.COM>,
 'Yoshihiro Kikuchi' <kiku@eel.rdc.toshiba.co.jp>, IETF AVT <rem-conf@es.net>,
 "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>
Cc: Colin Perkins <csp@east.isi.edu>, "'4on2andIP-sys@advent.ee.columbia.edu'"
 <4on2andIP-sys@advent.ee.columbia.edu>
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Zvi,
  Assuming CTS=DTS is confusing. If you have B-frames then there can be a
large (arbitrary with no legal upper limit put by MPEG) difference between
CTS and DTS. All the profiles except Simple use B-frames. So, unless we are
defining this only for the carriage of MPEG4 Simple Profile ES over IP, this
assumption does not seem to make sense. Due to instantaneous decoding model
at the System Level, one can assume that the picture is instantaneously
decoded and available for the prediction of next frame at DTS time (if DTS
is present). However, this is not same as assuming CTS=DTS. Actually, if you
are sending only one Time Stamp then you send CTS and your assumption will
be DTS=CTS which will be a disaster. So, if there is no DTS and you have
B-frames then only reasonable choice is to not to assume it to be anything
(like DTS=CTS) and  hope / pray receiver decodes the picture as fast it can
and does not overflow. In MPEG-2 kind of environment with
fixed/limited/pre-defined frame rates, it was easier to live without paying
too much attention to DTS. I am not so sure about MPEG-4 where there is no
fixed frame rate. That is one of the reasons the buffer model is so messy in
the Video part of MPEG-4. Praying may not help here.

  Why is jitter forcing you to assume CTS=DTS? Are you saying that due to
jitter in IP networks B-frames are not allowed and we need only to use
Simple Profile? 

  Please clarify. Thanks.

With regards,
Ajay 

-----Original Message-----
From: Zvi Lifshitz [mailto:zvil@csi.com]
Sent: Friday, September 15, 2000 12:34 AM
To: 'Narasimhan, Sam (SD-EX)'; 'Yoshihiro Kikuchi'; IETF AVT;
'jan.vandermeer@philips.com'
Cc: Colin Perkins; '4on2andIP-sys@advent.ee.columbia.edu'
Subject: RE: comments on in-band-signaling proposal


[Narasimhan, Sam (SD-EX):]

1. A video ES that required CTS (in the RTP header) and DTS (which
   could be in the adaptation field). This may be needed for receivers
   that implement MPEG buffer models. This stream can play on
   receivers that implement your draft as it has a bigger buffer
   and does not need DTS, as long as your draft enabled these
   receivers to use the adaptation length field to offset where
   the video ES payload was picked up.

[Reply:]

The jitter on IP network is such that practically receivers may always
assume CTS=DTS for implementation of the buffer model.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000






From rem-conf Thu Sep 21 11:09:48 2000 
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From: Zvi Lifshitz <zvil@csi.com>
To: "'Luthra, Ajay (SD-EX)'" <ALuthra@gi.com>,
        "'Narasimhan, Sam (SD-EX)'"
	 <SNarasimhan@gi.com>,
        "'Yoshihiro Kikuchi'"
	 <kiku@eel.rdc.toshiba.co.jp>,
        IETF AVT <rem-conf@es.net>,
        "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>
Cc: Colin Perkins <csp@east.isi.edu>,
        "'4on2andIP-sys@advent.ee.columbia.edu'" <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: comments on in-band-signaling proposal
Date: Thu, 21 Sep 2000 20:59:48 +0200
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Ajay,

You're right assuming DTS=CTS is not sufficient, but I believe we need to 
assume CTS-DTS < delta, when delta has an upper limit. If it doesn't we are 
in troubles because we don't know how much memory to allocate to the 
composition buffer. If indeed delta is known, it can be used to calculate 
the extra memory the decoding buffer would need to compensate for its 
ignorance of DTSs.

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				GSM: +972(54)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
==================================================================
> Come see us at:
The ISP Forum, Cavalieri Hilton, Rome, October 2nd-5th, 2000 
www.isp-conferences.com/ispf2000
Streaming Media Europe Earls Court Centre, Booth # 628, London, October 
10th-12th, 2000


-----Original Message-----
From:	Luthra, Ajay (SD-EX) [SMTP:ALuthra@gi.com]
Sent:	Thursday, September 21, 2000 7:17 PM
To:	'Zvi Lifshitz'; 'Narasimhan, Sam (SD-EX)'; 'Yoshihiro Kikuchi'; IETF 
AVT; 'jan.vandermeer@philips.com'
Cc:	Colin Perkins; '4on2andIP-sys@advent.ee.columbia.edu'
Subject:	RE: comments on in-band-signaling proposal


Zvi,
  Assuming CTS=DTS is confusing. If you have B-frames then there can be a
large (arbitrary with no legal upper limit put by MPEG) difference between
CTS and DTS. All the profiles except Simple use B-frames. So, unless we are
defining this only for the carriage of MPEG4 Simple Profile ES over IP, 
this
assumption does not seem to make sense. Due to instantaneous decoding model
at the System Level, one can assume that the picture is instantaneously
decoded and available for the prediction of next frame at DTS time (if DTS
is present). However, this is not same as assuming CTS=DTS. Actually, if 
you
are sending only one Time Stamp then you send CTS and your assumption will
be DTS=CTS which will be a disaster. So, if there is no DTS and you have
B-frames then only reasonable choice is to not to assume it to be anything
(like DTS=CTS) and  hope / pray receiver decodes the picture as fast it can
and does not overflow. In MPEG-2 kind of environment with
fixed/limited/pre-defined frame rates, it was easier to live without paying
too much attention to DTS. I am not so sure about MPEG-4 where there is no
fixed frame rate. That is one of the reasons the buffer model is so messy 
in
the Video part of MPEG-4. Praying may not help here.

  Why is jitter forcing you to assume CTS=DTS? Are you saying that due to
jitter in IP networks B-frames are not allowed and we need only to use
Simple Profile?

  Please clarify. Thanks.

With regards,
Ajay

-----Original Message-----
From: Zvi Lifshitz [mailto:zvil@csi.com]
Sent: Friday, September 15, 2000 12:34 AM
To: 'Narasimhan, Sam (SD-EX)'; 'Yoshihiro Kikuchi'; IETF AVT;
'jan.vandermeer@philips.com'
Cc: Colin Perkins; '4on2andIP-sys@advent.ee.columbia.edu'
Subject: RE: comments on in-band-signaling proposal


[Narasimhan, Sam (SD-EX):]

1. A video ES that required CTS (in the RTP header) and DTS (which
   could be in the adaptation field). This may be needed for receivers
   that implement MPEG buffer models. This stream can play on
   receivers that implement your draft as it has a bigger buffer
   and does not need DTS, as long as your draft enabled these
   receivers to use the adaptation length field to offset where
   the video ES payload was picked up.

[Reply:]

The jitter on IP network is such that practically receivers may always
assume CTS=DTS for implementation of the buffer model.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000





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From: "Luthra, Ajay (SD-EX)" <ALuthra@gi.com>
Subject: RE: comments on in-band-signaling proposal
To: 'Zvi Lifshitz' <zvil@csi.com>, 'IETF AVT' <rem-conf@es.net>
Cc: "'4on2andIP-sys@advent.ee.columbia.edu'"
 <4on2andIP-sys@advent.ee.columbia.edu>
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But MPEG does not put any limit on that Delta (CTS-DTS). In MPEG-2,
different user groups (like ATSC, SMPTE, DVD, DVB etc) put their own  limits
for their own applications. It is not clear at this stage if those limits
will be carried in the Internet world or not. They may not, as many Internet
applications and users are lot more delay tolerant. So, if we have to assume
some Delta at this stage then I would say that the proposal is broken as
with in MPEG, currently, there is no assigned value to it, except in Simple
Profile. Am I missing something?

With regards,
Ajay

-----Original Message-----
From: Zvi Lifshitz [mailto:zvil@csi.com]
Sent: Thursday, September 21, 2000 12:00 PM
To: 'Luthra, Ajay (SD-EX)'; 'Narasimhan, Sam (SD-EX)'; 'Yoshihiro
Kikuchi'; IETF AVT; 'jan.vandermeer@philips.com'
Cc: Colin Perkins; '4on2andIP-sys@advent.ee.columbia.edu'
Subject: RE: comments on in-band-signaling proposal


Ajay,

You're right assuming DTS=CTS is not sufficient, but I believe we need to 
assume CTS-DTS < delta, when delta has an upper limit. If it doesn't we are 
in troubles because we don't know how much memory to allocate to the 
composition buffer. If indeed delta is known, it can be used to calculate 
the extra memory the decoding buffer would need to compensate for its 
ignorance of DTSs.

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				GSM: +972(54)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
==================================================================
> Come see us at:
The ISP Forum, Cavalieri Hilton, Rome, October 2nd-5th, 2000 
www.isp-conferences.com/ispf2000
Streaming Media Europe Earls Court Centre, Booth # 628, London, October 
10th-12th, 2000


-----Original Message-----
From:	Luthra, Ajay (SD-EX) [SMTP:ALuthra@gi.com]
Sent:	Thursday, September 21, 2000 7:17 PM
To:	'Zvi Lifshitz'; 'Narasimhan, Sam (SD-EX)'; 'Yoshihiro Kikuchi'; IETF

AVT; 'jan.vandermeer@philips.com'
Cc:	Colin Perkins; '4on2andIP-sys@advent.ee.columbia.edu'
Subject:	RE: comments on in-band-signaling proposal


Zvi,
  Assuming CTS=DTS is confusing. If you have B-frames then there can be a
large (arbitrary with no legal upper limit put by MPEG) difference between
CTS and DTS. All the profiles except Simple use B-frames. So, unless we are
defining this only for the carriage of MPEG4 Simple Profile ES over IP, 
this
assumption does not seem to make sense. Due to instantaneous decoding model
at the System Level, one can assume that the picture is instantaneously
decoded and available for the prediction of next frame at DTS time (if DTS
is present). However, this is not same as assuming CTS=DTS. Actually, if 
you
are sending only one Time Stamp then you send CTS and your assumption will
be DTS=CTS which will be a disaster. So, if there is no DTS and you have
B-frames then only reasonable choice is to not to assume it to be anything
(like DTS=CTS) and  hope / pray receiver decodes the picture as fast it can
and does not overflow. In MPEG-2 kind of environment with
fixed/limited/pre-defined frame rates, it was easier to live without paying
too much attention to DTS. I am not so sure about MPEG-4 where there is no
fixed frame rate. That is one of the reasons the buffer model is so messy 
in
the Video part of MPEG-4. Praying may not help here.

  Why is jitter forcing you to assume CTS=DTS? Are you saying that due to
jitter in IP networks B-frames are not allowed and we need only to use
Simple Profile?

  Please clarify. Thanks.

With regards,
Ajay

-----Original Message-----
From: Zvi Lifshitz [mailto:zvil@csi.com]
Sent: Friday, September 15, 2000 12:34 AM
To: 'Narasimhan, Sam (SD-EX)'; 'Yoshihiro Kikuchi'; IETF AVT;
'jan.vandermeer@philips.com'
Cc: Colin Perkins; '4on2andIP-sys@advent.ee.columbia.edu'
Subject: RE: comments on in-band-signaling proposal


[Narasimhan, Sam (SD-EX):]

1. A video ES that required CTS (in the RTP header) and DTS (which
   could be in the adaptation field). This may be needed for receivers
   that implement MPEG buffer models. This stream can play on
   receivers that implement your draft as it has a bigger buffer
   and does not need DTS, as long as your draft enabled these
   receivers to use the adaptation length field to offset where
   the video ES payload was picked up.

[Reply:]

The jitter on IP network is such that practically receivers may always
assume CTS=DTS for implementation of the buffer model.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000




From rem-conf Thu Sep 21 11:41:58 2000 
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From: Zvi Lifshitz <zvil@csi.com>
To: "'Luthra, Ajay (SD-EX)'" <ALuthra@gi.com>,
        "'IETF AVT'"
	 <rem-conf@es.net>
Cc: "'4on2andIP-sys@advent.ee.columbia.edu'"
	 <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: comments on in-band-signaling proposal
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I think whatever delta an application assumes would be implemented by 
extending the terminal buffers accordingly. I still don't see how knowing 
the explicit DTS helps the terminal. Maybe I don't understand something, 
but thinking about it as a developer, I don't care whether the difficulties 
are on implementing the decoding buffer model, or accommodating DTS < CTS 
at the composition buffer.

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				GSM: +972(54)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
==================================================================
> Come see us at:
The ISP Forum, Cavalieri Hilton, Rome, October 2nd-5th, 2000 
www.isp-conferences.com/ispf2000
Streaming Media Europe Earls Court Centre, Booth # 628, London, October 
10th-12th, 2000


-----Original Message-----
From:	Luthra, Ajay (SD-EX) [SMTP:ALuthra@gi.com]
Sent:	Thursday, September 21, 2000 8:31 PM
To:	'Zvi Lifshitz'; 'IETF AVT'
Cc:	'4on2andIP-sys@advent.ee.columbia.edu'
Subject:	RE: comments on in-band-signaling proposal


But MPEG does not put any limit on that Delta (CTS-DTS). In MPEG-2,
different user groups (like ATSC, SMPTE, DVD, DVB etc) put their own 
 limits
for their own applications. It is not clear at this stage if those limits
will be carried in the Internet world or not. They may not, as many 
Internet
applications and users are lot more delay tolerant. So, if we have to 
assume
some Delta at this stage then I would say that the proposal is broken as
with in MPEG, currently, there is no assigned value to it, except in Simple
Profile. Am I missing something?

With regards,
Ajay

-----Original Message-----
From: Zvi Lifshitz [mailto:zvil@csi.com]
Sent: Thursday, September 21, 2000 12:00 PM
To: 'Luthra, Ajay (SD-EX)'; 'Narasimhan, Sam (SD-EX)'; 'Yoshihiro
Kikuchi'; IETF AVT; 'jan.vandermeer@philips.com'
Cc: Colin Perkins; '4on2andIP-sys@advent.ee.columbia.edu'
Subject: RE: comments on in-band-signaling proposal


Ajay,

You're right assuming DTS=CTS is not sufficient, but I believe we need to
assume CTS-DTS < delta, when delta has an upper limit. If it doesn't we are 
in troubles because we don't know how much memory to allocate to the
composition buffer. If indeed delta is known, it can be used to calculate
the extra memory the decoding buffer would need to compensate for its
ignorance of DTSs.

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				GSM: +972(54)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
==================================================================
> Come see us at:
The ISP Forum, Cavalieri Hilton, Rome, October 2nd-5th, 2000
www.isp-conferences.com/ispf2000
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000


-----Original Message-----
From:	Luthra, Ajay (SD-EX) [SMTP:ALuthra@gi.com]
Sent:	Thursday, September 21, 2000 7:17 PM
To:	'Zvi Lifshitz'; 'Narasimhan, Sam (SD-EX)'; 'Yoshihiro Kikuchi'; IETF

AVT; 'jan.vandermeer@philips.com'
Cc:	Colin Perkins; '4on2andIP-sys@advent.ee.columbia.edu'
Subject:	RE: comments on in-band-signaling proposal


Zvi,
  Assuming CTS=DTS is confusing. If you have B-frames then there can be a
large (arbitrary with no legal upper limit put by MPEG) difference between
CTS and DTS. All the profiles except Simple use B-frames. So, unless we are
defining this only for the carriage of MPEG4 Simple Profile ES over IP,
this
assumption does not seem to make sense. Due to instantaneous decoding model
at the System Level, one can assume that the picture is instantaneously
decoded and available for the prediction of next frame at DTS time (if DTS
is present). However, this is not same as assuming CTS=DTS. Actually, if
you
are sending only one Time Stamp then you send CTS and your assumption will
be DTS=CTS which will be a disaster. So, if there is no DTS and you have
B-frames then only reasonable choice is to not to assume it to be anything
(like DTS=CTS) and  hope / pray receiver decodes the picture as fast it can
and does not overflow. In MPEG-2 kind of environment with
fixed/limited/pre-defined frame rates, it was easier to live without paying
too much attention to DTS. I am not so sure about MPEG-4 where there is no
fixed frame rate. That is one of the reasons the buffer model is so messy
in
the Video part of MPEG-4. Praying may not help here.

  Why is jitter forcing you to assume CTS=DTS? Are you saying that due to
jitter in IP networks B-frames are not allowed and we need only to use
Simple Profile?

  Please clarify. Thanks.

With regards,
Ajay

-----Original Message-----
From: Zvi Lifshitz [mailto:zvil@csi.com]
Sent: Friday, September 15, 2000 12:34 AM
To: 'Narasimhan, Sam (SD-EX)'; 'Yoshihiro Kikuchi'; IETF AVT;
'jan.vandermeer@philips.com'
Cc: Colin Perkins; '4on2andIP-sys@advent.ee.columbia.edu'
Subject: RE: comments on in-band-signaling proposal


[Narasimhan, Sam (SD-EX):]

1. A video ES that required CTS (in the RTP header) and DTS (which
   could be in the adaptation field). This may be needed for receivers
   that implement MPEG buffer models. This stream can play on
   receivers that implement your draft as it has a bigger buffer
   and does not need DTS, as long as your draft enabled these
   receivers to use the adaptation length field to offset where
   the video ES payload was picked up.

[Reply:]

The jitter on IP network is such that practically receivers may always
assume CTS=DTS for implementation of the buffer model.

Regards,

z
soon the whole world will STREAM
==================================================================
Zvi Lifshitz				Phone +972(2)679-4788
zvil@optibase.com			Fax +972(2)679-4789
Optibase Ltd.				Mobile: +972(53)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
Send SMS from: 			http://isend.cellcom.co.il/English/Login.htm
==================================================================
> Come see us at:
Streaming Media Europe Earls Court Centre, Booth # 628, London, October
10th-12th, 2000




From rem-conf Thu Sep 21 12:18:14 2000 
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Date: Thu, 21 Sep 2000 12:18:08 -0700 (PDT)
From: Viswanathan Swaminathan <vishy@inbam.eng.sun.com>
Reply-To: Viswanathan Swaminathan <vishy@inbam.eng.sun.com>
Subject: Re: Types of MPEG-4 streams over IP
To: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu,
        jan.vandermeer@philips.com
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Jan,

In your list you missed MPEG-J Streams.

Regards,
Vishy

> X-Authentication-Warning: advent.ctr.columbia.edu: mjrdomo set sender to 
owner-4on2andIP-sys@advent.ee.columbia.edu using -f
> From: jan.vandermeer@philips.com
> To: <rem-conf@es.net>, <4on2andIP-sys@advent.ee.columbia.edu>
> Subject: Types of MPEG-4 streams over IP
> Date: Thu, 21 Sep 2000 14:25:05 +0200
> MIME-Version: 1.0
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> X-MIME-Autoconverted: from quoted-printable to 8bit by advent.ctr.columbia.edu 
id IAA24511
> 
> Dear all,
> 
> I would like to verify my understanding on the types of MPEG-4 streams that
> need tools for transport over IP. I summarized my understanding by the
> following list :
> 
> - ObjectDescriptorStream
> - ClockReferenceStream (is this needed over IP ?)
> - SceneDescriptionStream
> - VisualStream (with and without use of MPEG-4 Systems)
> - AudioStream (with and without use of MPEG-4 Systems)
> - MPEG-7Stream
> - IPMPStream
> - ObjectContentInfoStream
> - FlexMux stream
> - MP4 file
> 
> For MP4 files only a standard MIME type is required. For all other streams
> also use of RTP and SDP need to be defined. Please correct me if I am wrong.
> 
> The framework document (Singer et al) indicates that also a MIME type is
> needed for an MPEG-4 MuxCodeTable. Is this required indeed, or can we assume
> (/require) that the MuxCodeTable is transported inside the same FlexMux
> stream (in a packet with simple mode). This could provide an easy solution
> for the timing of applicability of the table that would need to be resolved
> otherwise when using e.g. HTTP; however it would require MuxCodeTable to
> become a new entry in the table for streamType values, right ?
> 
> Regards,
> 
> Jan
> 
> ************************
> Jan van der Meer
> Philips - Digital Networks
> Building OAN-4
> P.O. Box 80002
> 5600 JB Eindhoven
> The Netherlands
> Phone +31 402 735 774
> Fax +31 402 735 545
> Email jan.vandermeer@philips.com
> ************************
> 
> 

------------------------------------------------------
Viswanathan Swaminathan
Sun Microsystems Inc. 
901 San Antonio Road, M/S UMPK15-214  
Palo Alto, California  94303

Tel:(650) 786 4258     Fax:(650) 786 6445
email: vishy@eng.sun.com,  viswanathan.swaminathan@eng.sun.com





From rem-conf Thu Sep 21 16:12:05 2000 
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Date: Tue, 19 Sep 2000 04:05:09 -0700
To: Andy Valencia <vandys@zendo.com>, Tmima Koren <tmima@cisco.com>
From: Bruce A Thompson <brucet@cisco.com>
Subject: Re: 0 byte L2TPHC header and implied PPPMUX 
Cc: "W. Mark Townsley" <townsley@cisco.com>, dwing@cisco.com,
        Stephen Casner <casner@packetdesign.com>, gurdeep@microsoft.com,
        palter@zev.net, l2tp@diameter.org, rem-conf@es.net
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Sorry for the late response here. I just went through this thread.

At 08:44 AM 09/14/2000 -0700, Andy Valencia wrote:
>[Tmima Koren <tmima@cisco.com> writes:]
>
> >>flag 2:  value 1 means all packets have implied P=1
> >>             value 0 means all packets have implied P=0
>
>I'm concerned with this one.  Since L2TPHC is basically useless except for
>payload, what you've achieved with this flag is to ensure that your control
>traffic will be interrupted if you *ever* permit all payload traffic to be
>given preference by choosing P=1.  My experience is that this will make your
>link unworkable.  And no, using the UDP side doesn't help, since priority is
>(and should be) global to the PPP connection.

I'm trying to understand this issue a little better.

I assume normally, the P bit in the L2TP header would be mapped to IP precedence/DSCP at IP level. It sounnds like you are saying that when we negotiate for the P bit using the L2TPHC AVP, we are negotiating for the PPP session and not for the L2TPHC connection. However, L2TPHC does not preclude sending uncompressed L2TP packets down the UDP side. These frames will have the P bit set in them. If the P bit is negotiated using the L2TPHC AVP, are you saying that it should be ignored in uncompressed L2TP frames?

>I would suggest that when omitting the L2TPHC header entirely, that P=0
>always be the mode of operation.  This shouldn't affect your TCRTP mode of
>operation, because the only thing aside from those payload packets would be
>control packets which should be given priority anyway (and they can go via
>UDP).
>
> >>When flags 1 and 3 are on (=1), both L2TPHC header and PPPMUX protocol id can
> > be omitted 
>
>How about A/C fields?  Isn't it a little weird that a field in the middle of
>the PPP packet is coming and going based on this, but the bytes in front
>depend on PPP negotiation?
>
>Also, why hard-wire it to PPPMUX?  Why not include the implicit protocol
>value, so that these savings could apply to IP-only connections which
>weren't using PPPMUX and/or TCRTP?

After thinking about PPPMUX negotiation using L2TP AVP negotiation, I think there's a layer violation here. PPPMUX is supposed to be negotiated using LCP. If we want to get rid of the PPPMUX protocol ID in the encapsulation, then why not negotiate this using another LCP option for PPPMUX?

My bet is that the PPPEXT working group would accept this option. If PPPMUX was negotiated to not include the PPPMUX protocol ID, then the PPPMUX encapsulation would now not be RFC 1661 compliant. If this is an important option for bandwidth savings with multiplexed encapsulations, then it should be important for both L2TP and PPP.

I think that this option should be proposed in PPPEXT and not as an extension to L2TP.

         Bruce T

> >>PPP packets that have a different priority or that are not PPPMUX can still b
> >e sent in the tunnel using the UDP/L2TP format, same as the L2TP control packe
> >ts
> >>TCRTP (Tunneled Compressed multiplexed RTP) can benefit from these changes: t
> >he tunneling overhead will be reduced by 2 bytes.
>
>I guess I'm OK with a 0-byte L2TPHC header option (subject to a nod from the
>chair).  I'd sure like to hear from some people (not just infer something
>from silence) on the PPP header aspect.
>
>Andy Valencia

Bruce Thompson
Cisco Systems : IOS Products
email: brucet@cisco.com
office: (408)527-0446
home office: (408)868-0112
San Jose, CA




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Dear Webmaster,

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From rem-conf Thu Sep 21 19:49:59 2000 
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Date: Fri, 22 Sep 2000 08:17:51 +0530 (IST)
From: "Kuntal D. Sampat" <sampat@miel.mot.com>
To: jan.vandermeer@philips.com
cc: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
Subject: Re: Types of MPEG-4 streams over IP
In-Reply-To: <0056890014249640000002L902*@MHS>
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Jan,
isn't the "ERROR_PROTECTION_STREAM" (from 14496-3/Amd.1:1999 Part 3, 
Section 9, pp. 124), which is the error protected audio stream also a
unique streamType?

-Kuntal.

DSP Group,
Motorola, India.

On Thu, 21 Sep 2000 jan.vandermeer@philips.com wrote:

> Dear all,
> 
> I would like to verify my understanding on the types of MPEG-4 streams that
> need tools for transport over IP. I summarized my understanding by the
> following list :
> 
> - ObjectDescriptorStream
> - ClockReferenceStream (is this needed over IP ?)
> - SceneDescriptionStream
> - VisualStream (with and without use of MPEG-4 Systems)
> - AudioStream (with and without use of MPEG-4 Systems)
> - MPEG-7Stream
> - IPMPStream
> - ObjectContentInfoStream
> - FlexMux stream
> - MP4 file
> 
> For MP4 files only a standard MIME type is required. For all other streams
> also use of RTP and SDP need to be defined. Please correct me if I am wrong.
> 
> The framework document (Singer et al) indicates that also a MIME type is
> needed for an MPEG-4 MuxCodeTable. Is this required indeed, or can we assume
> (/require) that the MuxCodeTable is transported inside the same FlexMux
> stream (in a packet with simple mode). This could provide an easy solution
> for the timing of applicability of the table that would need to be resolved
> otherwise when using e.g. HTTP; however it would require MuxCodeTable to
> become a new entry in the table for streamType values, right ?
> 
> Regards,
> 
> Jan
> 
> ************************
> Jan van der Meer
> Philips - Digital Networks
> Building OAN-4
> P.O. Box 80002
> 5600 JB Eindhoven
> The Netherlands
> Phone +31 402 735 774
> Fax +31 402 735 545
> Email jan.vandermeer@philips.com
> ************************
> 
> 
> 




From rem-conf Fri Sep 22 00:43:11 2000 
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To: 'Hari Krishna Vuppaladhadiam' <hariv@chiplogic.com>, rem-conf@es.net, 
    J.Crowcroft@cs.ucl.ac.uk
Subject: Re: Voice Playout for VoATM #2
In-reply-to: Your message of "Thu, 21 Sep 2000 08:19:35 BST." <937.969520775@cs.ucl.ac.uk>
Date: Fri, 22 Sep 2000 08:35:25 +0100
Message-ID: <1387.969608125@cs.ucl.ac.uk>
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after a chat with orion hodson and dimitris terzis here who've done
lots of voice of ip and atm:

2 more important consequences of playout for ATM versus IP

1/ native atm never re-orders packets - IP frequently does - this adds
to the playout for an IP receiver potentially in odd ways  - the
re-ordering statistcs are not necessarily distributed the way that jitter is
distributed when causesd by statistical multiplexing so algoruithms
based on assu,mptions about that distribution (e.g. rolling averasge
+ some number times the variance of inter-arrival times) might not
accomoadate this well, whereas ATM wont have this problem

2/ there's a strong incentive not to go for packetisation larger than
a single cell in native atm, whereas in ip there's nothing to stop ou
picking anything up to an mtu really - this leads to sender side
delays in sampling and packetising and serialising the packet for
transmission from an IP box - all of these may be subject to
operating system scheduling effects too (see papre by kouvelas on
cushioning these, or henning schulzrinnes work way back on various
band pass filter ways of remioving these glitches) - again, vice on
atm would be more like to be written device to device, and a cell
payload at a time, so much less susceptible 

so basically, voice on atm playout is easier than voip, once you can
swallow the idea of using ATM :-)

 cheers

   jon




From rem-conf Fri Sep 22 04:42:04 2000 
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From: "Dr. Carsten Bormann" <cabo@tzi.org>
To: "Jon Crowcroft" <J.Crowcroft@cs.ucl.ac.uk>,
   "'Hari Krishna Vuppaladhadiam'" <hariv@chiplogic.com>, <rem-conf@es.net>
Subject: RE: Voice Playout for VoATM #2
Date: Fri, 22 Sep 2000 13:31:17 +0200
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> [...] vice on atm [...]

Is there an RFC for vice on ATM?

Gruesse, Carsten




From rem-conf Fri Sep 22 07:18:22 2000 
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Date: Thu, 21 Sep 2000 21:56:01 +0200
From: Ralph Sperschneider <sps@iis.fhg.de>
Organization: Fraunhofer IIS Audio & Multimedia (http://www.iis.fhg.de/amm)
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To: Toshiyuki Nomura <t-nomura@ccm.cl.nec.co.jp>
CC: John Lazzaro <lazzaro@cs.berkeley.edu>, tmn <tmn@iis.fhg.de>,
        grl <grl@iis.fhg.de>, mpeg4_vm <mpeg4_vm@tnt.uni-hannover.de>,
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Subject: Re: LATM specified in a public MPEG document?
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Dear Toshiyuki,

> Simple SA session without scene description can be realized
> by the LATM-based RTP specification.

Are these facts already stated within the current LATM description? We otherwise would now have the chance to clear these matters by means of the Corrigendum (deadline is the end of September).

Best regards,

Ralph

-- 
Dipl.-Ing. Ralph Sperschneider  | Phone: +49 9131 776 344
FhG IIS-A                       | Fax:   +49 9131 776 398
Am Weichselgarten 3             | mailto:sps@iis.fhg.de 
D 91058 Erlangen                | http://www.iis.fhg.de/amm/

Facts do not cease to exist because they are ignored. (Aldous Huxley)





From rem-conf Fri Sep 22 07:30:07 2000 
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Date: Fri, 22 Sep 2000 09:29:03 -0500
From: "Brian F. G. Bidulock" <bidulock@dallas.net>
To: "Dr. Carsten Bormann" <cabo@tzi.org>
Cc: Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>,
        "'Hari Krishna Vuppaladhadiam'" <hariv@chiplogic.com>, rem-conf@es.net
Subject: Re: Voice Playout for VoATM #2
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> > [...] vice on atm [...]
> 
> Is there an RFC for vice on ATM?

it be rc01.xt at ftp.iet.og

-- 
Brian F. G. Bidulock
bidulock@dallas.net



From rem-conf Fri Sep 22 10:24:23 2000 
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Subject: RDRT CEO - Rapid Growth- Fibre Optic company IPO  
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<font class="\&quot;headline\&quot;"><b><i><span style="text-transform: capitalize"><font face="Times New Roman" size="5" color="#0000FF">Read-Rite
launches Scion, a new </font></span></i></b></font><font size="5" color="#0000FF"><b><i>Fibre</i></b></font><font class="\&quot;headline\&quot;"><b><i><span style="text-transform: capitalize"><font face="Times New Roman" size="5" color="#0000FF">
optics company</font><font size="4" color="#000080"><br>
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<p><strong><font face="Arial" size="5" color="#000080">Read-Rite CEO gets into
growth</font></strong></p>
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  <table border="0" cellSpacing="0" width="100%">
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        <td width="66%"><em><font color="#000080">By Janet Haney,
          CBS.MarketWatch.com<br>
          <font size="2">Last Update: <!--webbot bot="HTMLMarkup" startspan
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<p><font color="#000080">SAN FRANCISCO (CBS.MW) -- Read-Rite Chief Executive
Officer Alan Lowe waxed positive about the future growth prospects for the
magnetic-recording-head supplier.</font></p>
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<TD bgColor=#99cc99><STRONG><FONT color=#ffffff face=Arial size=2>Today on CBS 
MarketWatch</FONT></STRONG></TD></TR>
<TR>
<TD><FONT face=Arial size=2><A 
href="/archive/20000921/news/current/snapshot.htx?dist=hdlnbug&amp;source=htx/http2_mw">Dow 
gets sucked into tech retreat</A></FONT></TD></TR>
<TR>
<TD><FONT face=Arial size=2><A 
href="/archive/20000921/news/current/mwd.htx?dist=hdlnbug&amp;source=htx/http2_mw">Morgan 
Stanley profit falls short of target</A></FONT></TD></TR>
<TR>
<TD><FONT face=Arial size=2><A 
href="/archive/20000921/news/current/f.htx?dist=hdlnbug&amp;source=htx/http2_mw">Ford 
bids to buy back 18.5% stake in Hertz</A></FONT></TD></TR>
<TR>
<TD><FONT face=Arial size=2><A 
href="/archive/20000921/news/current/greenspan.htx?dist=hdlnbug&amp;source=htx/http2_mw">Greenspan 
calls for education reform</A></FONT></TD></TR>
<TR>
<TD><FONT face=Arial size=2><A 
href="/archive/20000921/news/current/molinski.htx?dist=hdlnbug&amp;source=htx/http2_mw">FundWatch: 
Finding funds on the Web</A></FONT></TD></TR>
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face=Arial size=2>More top stories</A>...</FONT></TD></TR>
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face=Arial size=2>CBS MarketWatch Columns</A></FONT></TD></TR>
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<TD><FONT face=Arial size=1>Updated:<BR>9/21/2000 11:47:04 
AMET</FONT></TD></TR></TBODY></TABLE></TD></TR></TBODY></TABLE><!--webbot
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<p><font color="#000080">Lowe told a crowd of investors and analysts during a
presentation at the Banc of America Securities Investment Conference in San
Francisco on Thursday that he expects huge unit growth potential for Read-Rite's
(<a href="http://cbs.marketwatch.com/data/squote.htx?TICKER=RDRT&amp;TABLES=table&amp;SOURCE=htx/http2_mw&amp;dist=newsq">RDRT</a>:
<a href="http://www2.marketwatch.com/news/search.asp?Query=RDRT&amp;SearchOption=ticker">news</a>,
<a href="http://messages.marketwatch.com/mwclub/tickerLink.asp?ticker=RDRT&amp;dist=newsm">msgs</a>)
December quarter, as well as the possibility of profitability.</font></p>
<p><font color="#000080">Lowe added that the company is hiring people as fast as
it can for its wafer fabrication facility.</font></p>
<p><font color="#000080">For the September quarter, the CEO said Read-Rite has a
&quot;lot of product to ship in the last 10 days of the quarter.&quot;</font></p>
<p><font color="#000080">Additionally, Lowe talked about Read-Rite's recent
formation of an independent fiber optic company called Scion Photonics which he
said hopes to go public.</font></p>
<p><font color="#000080">Scion received funding from Tyco, which will make a
presentation at the conference later Thursday.&nbsp; <img src="http://cbs.marketwatch.com/news/images/cbsmw_fin.gif" width="51" height="14"></font></p>
<hr>
<em><font color="#000080">Janet Haney is a reporter for CBS.MarketWatch.com.<br>
</font></em>
<p>&nbsp;</p>
<p><font size="4" color="#000080">Read-Rite Corp &lt;</font><font size="4" color="#CC0000">RDRT</font><font size="4" color="#000080">&gt;
said late on Wednesday it has formed a new fibre optics company, Scion Photonics
Inc, with $25 million in initial funding from Tyco Ventures and Integral Capital
Partners. Read-Rite also said it was in discussions with several other
interested parties for more technology and capital investments into Scion.</font>
</p>
<p class="\&quot;body\&quot;"><font size="4" color="#000080">Scion's business
charter is &quot;to design, manufacture and market high-performance optical
components to the fibre optics communications market,&quot; Read-Rite said in a
statement.</font>
<p class="\&quot;body\&quot;"><font size="4" color="#000080">Scion will act as a
separate company, but will have access to Read-Rite's resources and
manufacturing infrastructure, Read-Rite chairman Cyril Yansouni said in a
statement. Read-Rite is contributing its Milpitas, California wafer fabrication
plant, technical and management resources and its knowledge and experience in
thin-film technology and materials science to Scion, which will be based in
Milpitas.</font>
<p class="\&quot;body\&quot;"><font size="4" color="#000080">Tyco Ventures, the
venture capital unit of Tyco International Ltd. &lt;TYC&gt;, invested $15
million in Scion and also invested $15 million in Read-Rite through a private
placement with the company, Tyco Ventures president Richard Kashnow said in a
statement.</font>
<p class="\&quot;body\&quot;">&nbsp;</p>
<div>
  <font face="Arial" size="2"><font color="#ff0000"><b><i>CBS article:</i></b><br>
  </font><a href="http://cbs.marketwatch.com/archive/20000907/news/current/screamers.htx?source=blq/yhoo&amp;dist=yhoo"><font color="#008000">http://cbs.marketwatch.com/archive/20000907/news/current/screamers.htx?source=blq/yhoo&amp;dist=yhoo</font></a></font>
</div>
<div>
  &nbsp;
</div>
<div style="width: 893; height: 32">
  <font face="Arial" size="2"><font color="#ff0000"><b><i>For the complete
  streaming audio story users should access</i></b></font><br>
  <a href="http://www.on24.com/index.html?id=36870&amp;type=av&amp;ref=bizwire"><font color="#008000">http://www.on24.com/index.html?id=36870&amp;type=av&amp;ref=bizwire</font></a></font>
</div>
<div>
  &nbsp;
</div>
<div>
  <font face="Arial" size="2"><br>
  ALWAYS DO RESEARCH BEFORE INVESTING IN ANY STOCK!!!</font>
</div>
<p>&nbsp;</p>
<p>&nbsp;</p>

</body>

</html>






From rem-conf Fri Sep 22 11:21:12 2000 
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Date: Fri, 22 Sep 2000 11:15:20 -0700
From: John Lazzaro <lazzaro@CS.Berkeley.EDU>
Message-Id: <200009221815.LAA01556@snap.CS.Berkeley.EDU>
To: sps@iis.fhg.de, t-nomura@ccm.cl.nec.co.jp
Subject: Re: LATM specified in a public MPEG document?
Cc: grl@iis.fhg.de, lazzaro@cs.berkeley.edu, mpeg4_vm@tnt.uni-hannover.de,
        purnhagen@tnt.uni-hannover.de, rem-conf@es.net, tmn@iis.fhg.de
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>> Simple SA session without scene description can be realized
>> by the LATM-based RTP specification.

> Are these facts already stated within the current LATM description?
>  We otherwise would now have the chance to clear these matters by means
> of the Corrigendum (deadline is the end of September).

Since there's no publicly-available LATM document, I'm on shakey ground
commenting on this, but nonetheless -- note that the MIDI
SA_access_unit's don't include their own timestamp (unlike SASL SA_access_units
that do have an optional timestamp in "scorebeats", which are tempo
modulatable), and since Bodo Teichmann's comment stated:

> [ Bodo Teichmann <tmn@iis.fhg.de> writes] 
> 
> > LATM was actually developed for only "natural"  audio coders, 
> > i.e. not fo Structured Audio. it is not possible to transmitt
> > multiple objects, no scene desctipion  and no time stamps
> > can be transmitted .

This means that taking LATM by itself, there would be no way to have
timestamps on MIDI SA_access_units.

For RTP over LATM this shouldn't be insurmountable, because of the RTP
timestamps, but if the Corrigendum you're mentioning isn't about LATM
over RTP but just LATM itself, it would seem to break using MIDI in
SA_access_units for all but "use immediately" semantics, which is 
what I assumed the quoted section above meant to convey.

For my own work, its becoming clear that making an RTP packetization
just for Structured Audio is the right thing to do -- one lost MIDI
NoteOff command can really wreck the entire performance (for those   
not familiar with MIDI, NoteOff's end individual notes of a performance
which are begun by a NoteOn -- a lost NoteOff means a note that sounds
"forever"), and a packetization meant for interactive performance over
low-latency networks really has to address this issue to be useful,
either by schemes similiar to draft-ietf-avt-rtprx-00.txt for unicast,
or by various FEC approaches for multicast ...

-------------------------------------------------------------------------
John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
-------------------------------------------------------------------------



From rem-conf Fri Sep 22 11:41:21 2000 
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Date: Fri, 22 Sep 2000 11:34:30 -0700
To: jan.vandermeer@philips.com
From: Dave Singer <singer@apple.com>
Subject: Re: Types of MPEG-4 streams over IP
Cc: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
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At 2:25 PM +0200 9/21/00, jan.vandermeer@philips.com wrote:
>Dear all,
>
>I would like to verify my understanding on the types of MPEG-4 streams that
>need tools for transport over IP. I summarized my understanding by the
>following list :
>
>- ObjectDescriptorStream
>- ClockReferenceStream (is this needed over IP ?)
>- SceneDescriptionStream
>- VisualStream (with and without use of MPEG-4 Systems)
>- AudioStream (with and without use of MPEG-4 Systems)
>- MPEG-7Stream
>- IPMPStream
>- ObjectContentInfoStream
>- FlexMux stream
>- MP4 file
>
>For MP4 files only a standard MIME type is required. For all other streams
>also use of RTP and SDP need to be defined. Please correct me if I am wrong.
>
>The framework document (Singer et al) indicates that also a MIME type is
>needed for an MPEG-4 MuxCodeTable. Is this required indeed, or can we assume
>(/require) that the MuxCodeTable is transported inside the same FlexMux
>stream (in a packet with simple mode). This could provide an easy solution
>for the timing of applicability of the table that would need to be resolved
>otherwise when using e.g. HTTP; however it would require MuxCodeTable to
>become a new entry in the table for streamType values, right ?


The MuxCodeTable needs to be delivered out-of-band.  If you see my 
proposal, it was to deliver this via a URL in SDP, which then 
requires a MIME type if HTTP or DATA are to be used.
-- 
David Singer
Apple Computer/QuickTime



From rem-conf Fri Sep 22 11:41:26 2000 
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Date: Fri, 22 Sep 2000 11:35:40 -0700
To: John Lazzaro <lazzaro@CS.Berkeley.EDU>
From: Dave Singer <singer@apple.com>
Subject: Re: LATM specified in a public MPEG document?
Cc: sps@iis.fhg.de, t-nomura@ccm.cl.nec.co.jp, grl@iis.fhg.de,
        lazzaro@CS.Berkeley.EDU, mpeg4_vm@tnt.uni-hannover.de,
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At 11:15 AM -0700 9/22/00, John Lazzaro wrote:
>  >> Simple SA session without scene description can be realized
>>>  by the LATM-based RTP specification.
>
>>  Are these facts already stated within the current LATM description?
>>   We otherwise would now have the chance to clear these matters by means
>>  of the Corrigendum (deadline is the end of September).
>
>Since there's no publicly-available LATM document,


well, this is a show-stopper.  we cannot base a standard on something 
that is not publicly defined.
-- 
David Singer
Apple Computer/QuickTime



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From: John Lazzaro <lazzaro@CS.Berkeley.EDU>
Message-Id: <200009221901.MAA01790@snap.CS.Berkeley.EDU>
To: lazzaro@cs.berkeley.edu, singer@apple.com
Subject: Re: LATM specified in a public MPEG document?
Cc: grl@iis.fhg.de, mpeg4_vm@tnt.uni-hannover.de,
        purnhagen@tnt.uni-hannover.de, rem-conf@es.net, sps@iis.fhg.de,
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> well, this is a show-stopper.  we cannot base a standard on something 
> that is not publicly defined.

Oh, this is just temporary -- eventually anyone can send a
check to MPEG and buy the FDIS that defines LATM, which I think would
meet the requirement of "publicly defined". Usually the FCD's are posted
on publically-available websites like this one:

http://sound.media.mit.edu/mpeg4/audio/

but you'll notice the last FCD is Dec 1999 (scroll down to MPEG 4) which
is pre-LATM. So in that sense, I'm in the dark as someone outside of
MPEG trying to comment on an I-D that references something not in a 
publically available working document ... 

-------------------------------------------------------------------------
John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
-------------------------------------------------------------------------



From rem-conf Fri Sep 22 18:02:48 2000 
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Date: Fri, 22 Sep 2000 03:00:40 +0200
To: "Luthra, Ajay (SD-EX)" <ALuthra@gi.com>
From: Stephan Wenger <stewe@cs.tu-berlin.de>
Subject: RE: comments on in-band-signaling proposal
Cc: "'Zvi Lifshitz'" <zvil@csi.com>, "'IETF AVT'" <rem-conf@es.net>,
        "'4on2andIP-sys@advent.ee.columbia.edu'" <4on2andIP-sys@advent.ee.columbia.edu>
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At 02:31 PM 9/21/00 -0400, Luthra, Ajay (SD-EX) wrote:
>But MPEG does not put any limit on that Delta (CTS-DTS). In MPEG-2,
>different user groups (like ATSC, SMPTE, DVD, DVB etc) put their own  limits
>for their own applications. It is not clear at this stage if those limits
>will be carried in the Internet world or not. They may not, as many Internet
>applications and users are lot more delay tolerant. So, if we have to assume
>some Delta at this stage then I would say that the proposal is broken as
>with in MPEG, currently, there is no assigned value to it, except in Simple
>Profile. Am I missing something?

My remark to that thread would be that it may be desirable that a payload spec
allows for the full functionality of a media coding, but that there is no 
obligation
to have such an attribute.

Maybe a sentence indicating that, when B-frames are used, it is necessarily
to carefully tune the size of the playout buffer may be enough?

I personally would object to add more timestamp fields in the payload header
just to accommodate some theoretically possible applications that are more
than unlikely to happen in the field.

Stephan




>With regards,
>Ajay
>
>-----Original Message-----
>From: Zvi Lifshitz [mailto:zvil@csi.com]
>Sent: Thursday, September 21, 2000 12:00 PM
>To: 'Luthra, Ajay (SD-EX)'; 'Narasimhan, Sam (SD-EX)'; 'Yoshihiro
>Kikuchi'; IETF AVT; 'jan.vandermeer@philips.com'
>Cc: Colin Perkins; '4on2andIP-sys@advent.ee.columbia.edu'
>Subject: RE: comments on in-band-signaling proposal
>
>
>Ajay,
>
>You're right assuming DTS=CTS is not sufficient, but I believe we need to
>assume CTS-DTS < delta, when delta has an upper limit. If it doesn't we are
>in troubles because we don't know how much memory to allocate to the
>composition buffer. If indeed delta is known, it can be used to calculate
>the extra memory the decoding buffer would need to compensate for its
>ignorance of DTSs.
>
>z
>soon the whole world will STREAM
>==================================================================
>Zvi Lifshitz                            Phone +972(2)679-4788
>zvil@optibase.com                       Fax +972(2)679-4789
>Optibase Ltd.                           GSM: +972(54)461-787
>http://www.optibase.com         US voice mail/fax  +1(206) 888-4149
>==================================================================
> > Come see us at:
>The ISP Forum, Cavalieri Hilton, Rome, October 2nd-5th, 2000
>www.isp-conferences.com/ispf2000
>Streaming Media Europe Earls Court Centre, Booth # 628, London, October
>10th-12th, 2000
>
>
>-----Original Message-----
>From:   Luthra, Ajay (SD-EX) [SMTP:ALuthra@gi.com]
>Sent:   Thursday, September 21, 2000 7:17 PM
>To:     'Zvi Lifshitz'; 'Narasimhan, Sam (SD-EX)'; 'Yoshihiro Kikuchi'; IETF
>
>AVT; 'jan.vandermeer@philips.com'
>Cc:     Colin Perkins; '4on2andIP-sys@advent.ee.columbia.edu'
>Subject:        RE: comments on in-band-signaling proposal
>
>
>Zvi,
>   Assuming CTS=DTS is confusing. If you have B-frames then there can be a
>large (arbitrary with no legal upper limit put by MPEG) difference between
>CTS and DTS. All the profiles except Simple use B-frames. So, unless we are
>defining this only for the carriage of MPEG4 Simple Profile ES over IP,
>this
>assumption does not seem to make sense. Due to instantaneous decoding model
>at the System Level, one can assume that the picture is instantaneously
>decoded and available for the prediction of next frame at DTS time (if DTS
>is present). However, this is not same as assuming CTS=DTS. Actually, if
>you
>are sending only one Time Stamp then you send CTS and your assumption will
>be DTS=CTS which will be a disaster. So, if there is no DTS and you have
>B-frames then only reasonable choice is to not to assume it to be anything
>(like DTS=CTS) and  hope / pray receiver decodes the picture as fast it can
>and does not overflow. In MPEG-2 kind of environment with
>fixed/limited/pre-defined frame rates, it was easier to live without paying
>too much attention to DTS. I am not so sure about MPEG-4 where there is no
>fixed frame rate. That is one of the reasons the buffer model is so messy
>in
>the Video part of MPEG-4. Praying may not help here.
>
>   Why is jitter forcing you to assume CTS=DTS? Are you saying that due to
>jitter in IP networks B-frames are not allowed and we need only to use
>Simple Profile?
>
>   Please clarify. Thanks.
>
>With regards,
>Ajay
>
>-----Original Message-----
>From: Zvi Lifshitz [mailto:zvil@csi.com]
>Sent: Friday, September 15, 2000 12:34 AM
>To: 'Narasimhan, Sam (SD-EX)'; 'Yoshihiro Kikuchi'; IETF AVT;
>'jan.vandermeer@philips.com'
>Cc: Colin Perkins; '4on2andIP-sys@advent.ee.columbia.edu'
>Subject: RE: comments on in-band-signaling proposal
>
>
>[Narasimhan, Sam (SD-EX):]
>
>1. A video ES that required CTS (in the RTP header) and DTS (which
>    could be in the adaptation field). This may be needed for receivers
>    that implement MPEG buffer models. This stream can play on
>    receivers that implement your draft as it has a bigger buffer
>    and does not need DTS, as long as your draft enabled these
>    receivers to use the adaptation length field to offset where
>    the video ES payload was picked up.
>
>[Reply:]
>
>The jitter on IP network is such that practically receivers may always
>assume CTS=DTS for implementation of the buffer model.
>
>Regards,
>
>z
>soon the whole world will STREAM
>==================================================================
>Zvi Lifshitz                            Phone +972(2)679-4788
>zvil@optibase.com                       Fax +972(2)679-4789
>Optibase Ltd.                           Mobile: +972(53)461-787
>http://www.optibase.com         US voice mail/fax  +1(206) 888-4149
>Send SMS from:                  http://isend.cellcom.co.il/English/Login.htm
>==================================================================
> > Come see us at:
>Streaming Media Europe Earls Court Centre, Booth # 628, London, October
>10th-12th, 2000





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Subject: Re: 0 byte L2TPHC header and implied PPPMUX 
In-reply-to: Your message of "Tue, 19 Sep 2000 04:05:09 PDT."
             <4.3.1.0.20000918173607.01236230@sigma.cisco.com> 
Date: Sat, 23 Sep 2000 12:37:26 -0700
From: Andy Valencia <vandys@zendo.com>
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[Bruce A Thompson <brucet@cisco.com> writes:]

>I assume normally, the P bit in the L2TP header would be mapped to IP preceden
>ce/DSCP at IP level.

No.  This is packet queueing on the transmitter of an HDLC device carrying
PPP.

>After thinking about PPPMUX negotiation using L2TP AVP negotiation, I think th
>ere's a layer violation here. PPPMUX is supposed to be negotiated using LCP. I
>f we want to get rid of the PPPMUX protocol ID in the encapsulation, then why 
>not negotiate this using another LCP option for PPPMUX?

Because we're not talking about removing the PPP protocol ID.  We're talking
about L2TP not carrying it across the link, because it's implicitly
understood, and can be inserted after the PPP packet (sans protocol ID) has
traversed the L2TP link.  So this is an L2TP(HC)-only consideration.  PPP
continues to see what it always saw.  And you can't do it at the PPP level,
because then there'd be no way for NCP (not even LCP) to talk across the
link--which PPPEXT would assuredly reject!

Andy Valencia



From rem-conf Sat Sep 23 17:12:06 2000 
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From: "Alan J. Flavell" <flavell@a5.ph.gla.ac.uk>
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To: "Brian F. G. Bidulock" <bidulock@dallas.net>
cc: postmaster@dallas.net, rem-conf@es.net
Subject: Email abuse, was Re: So, How in the heck have you been?
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On Thu, 21 Sep 2000, Brian F. G. Bidulock wrote:

> Could someone PLEEEEEASE put a spam filter on this list!

Please would you (and anyone else tempted to copy you) desist from
relaying additional copies of spam, and complaints about it, to all
the other victims, and refer your proposals to the list owner.


Postmaster, please instruct your user about appropriate use of
email.  Thank you.  Fortunately I won't be seeing anything else from
this particular user, as their address is going straight into my
filter list.


(24K of drivel deleted)






From rem-conf Sun Sep 24 12:01:14 2000 
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From rem-conf Sun Sep 24 21:18:58 2000 
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From: "huang tingxue" <huangtx@xiyou.edu.cn>
To: <rem-conf@es.net>
Subject: RTP payload format for G.723.1?
Date: Mon, 25 Sep 2000 12:13:29 +0800
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I don't know whether RTP payload format for G.723.1 exists. If someone =
know , please tell me. I am very thankful!

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<DIV><FONT color=3D#000000 size=3D2>I don't know whether RTP payload =
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From rem-conf Mon Sep 25 05:27:08 2000 
From rem-conf-request@es.net Mon Sep 25 05:27:07 2000
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To: <singer@apple.com>
Cc: <rem-conf@es.net>, <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: Types of MPEG-4 streams over IP
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Hello David,

Yes, I understand and I agree that a URL is a fine solution for a stati=
c
table. However, I understood that the MuxCode may change in the course =
of a
FlexMux. If yes, then the timing of the change need be specified. In th=
at
case in-band carriage of the MuxCode maybe a more convenient solution. =
If
such in-band carriage was defined, would there still be a need for a UR=
L ?

But I may miss something. Perhaps we constrain the use of MuxCode to st=
atic
cases ?

Kind regards,

Jan van der Meer


The MuxCodeTable needs to be delivered out-of-band.  If you see my
proposal, it was to deliver this via a URL in SDP, which then
requires a MIME type if HTTP or DATA are to be used.
--
David Singer
Apple Computer/QuickTime

=



From rem-conf Mon Sep 25 05:45:53 2000 
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Date: Mon, 25 Sep 2000 14:35:33 +0200
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To: jan.vandermeer@philips.com
CC: singer@apple.com, rem-conf@es.net,
	 4on2andIP-sys@advent.ee.columbia.edu
Subject: Re: Types of MPEG-4 streams over IP
References: <0056890014340965000002L952*@MHS>
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Dear Jan,

I'm not sure, but the change of the MuxCode may also be announced through=
 an
out of band mean.

As an example, in SDP/SAP, a server may automatically update the
mpeg4-flexmuxinfo as described in Dave's framework, and announce the chan=
ge by
updating the Message Identifier Hash of the SAP header to announce that t=
he
session description has been modified. The change timing capacity in SDP =
 may
be used to specify  the time at which the change is valid.The session
description may use a "t=3D " line for that purpose.

in case RTSP is used to convey the flexmuxinfo,  I guess the Cseq paramet=
er may
be used to announce a change has occured in the SDP session.

Regards

Pierre

jan.vandermeer@philips.com a =E9crit :

> Hello David,
>
> Yes, I understand and I agree that a URL is a fine solution for a stati=
c
> table. However, I understood that the MuxCode may change in the course =
of a
> FlexMux. If yes, then the timing of the change need be specified. In th=
at
> case in-band carriage of the MuxCode maybe a more convenient solution. =
If
> such in-band carriage was defined, would there still be a need for a UR=
L ?
>
> But I may miss something. Perhaps we constrain the use of MuxCode to st=
atic
> cases ?
>
> Kind regards,
>
> Jan van der Meer
>
> The MuxCodeTable needs to be delivered out-of-band.  If you see my
> proposal, it was to deliver this via a URL in SDP, which then
> requires a MIME type if HTTP or DATA are to be used.
> --
> David Singer
> Apple Computer/QuickTime




From rem-conf Mon Sep 25 08:31:32 2000 
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To: huang tingxue <huangtx@xiyou.edu.cn>
cc: rem-conf <rem-conf@es.net>
Subject: Re: RTP payload format for G.723.1? 
In-Reply-To: Your message of "Mon, 25 Sep 2000 12:13:29 +0800."
             <001501c026a6$f636ff80$060a0a0a@atm06> 
Date: Mon, 25 Sep 2000 11:21:59 -0400
From: Colin Perkins <csp@east.isi.edu>
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--> huang tingxue writes:
>I don't know whether RTP payload format for G.723.1 exists. If someone 
>know , please tell me. I am very thankful!

See section 4.5.4 of draft-ietf-avt-profile-new-09.txt

Colin



From rem-conf Mon Sep 25 11:03:27 2000 
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Date: Mon, 25 Sep 2000 11:00:30 -0700
From: John Lazzaro <lazzaro@CS.Berkeley.EDU>
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To: lazzaro@cs.berkeley.edu, sps@iis.fhg.de, t-nomura@ccm.cl.nec.co.jp
Subject: Re: LATM specified in a public MPEG document?
Cc: grl@iis.fhg.de, mpeg4_vm@tnt.uni-hannover.de,
        purnhagen@tnt.uni-hannover.de, rem-conf@es.net, tmn@iis.fhg.de
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> By the way, could you let me know what is "use immediately semantics"?

>From 5.5.2 of 14496-3:1999(E), starting at the bottom of page 20:

  The Structured Audio access unit contains real-time
  streaming control information to be provided to a
  running Structured Audio decoding process. It may
  contain as many control instructions as desired and as
  permitted by the available bandwidth. It shall not
  contain new instrument definitions; the orchestra
  configuration is fixed at decoder start-up. It may
  contain score lines, MIDI events, and new sample
  data. When provided as part of an access unit, the
  score line is not required to contain a timestamp.
  When has_time is cleared in the score_line class, the
  event is dispatched immediately according to the rules
  in subclause 5.7.3.3.6. Score lines without timestamps
  are not responsive to orchestra tempo changes. 

  Annex 5.E discusses when the random access point
  flag, conveyed in the Access Unit packaging in the
  Systems specification, may be set. 

The "score line is not required to contain a timestamp"
section is what I meant by "use immediately semantics".

> I think such limitation should not be described in the LATM
> itself definitions. 
> This is because these matters depend on the functions of the
> transport layer.

I guess the issue I'm trying to bring up is this: 

[1] Let's say you're sending an SA_access_unit using MIDI,
whose semantics are NOT "use immediately semantics". If
so, you're dependent on a timestamp to say "when to fire"
supplied from elsewhere. Note this means the timestamp 
has two bounds on it -- too early and the note fires too
early, too late and the note fires too late. The precision
of the timestamp becomes the precision of the performance --
too sloppy and things sound bad.

[2] If I understand the earlier info on LATM correctly (remember,
I don't have access to the spec), LATM doesn't supply timestamps.

[3] So, whatever packetization is used (for example, RTP), needs
to supply this timestamp -- however, the accuracy requirements
for this timestamp (as described in [1] above) is sufficiently
unusual for someone used to "natural audio" codecs, that an 
explicit heads up may be in order, in both the LATM and the
RTP documents.

Hope this helps ...

						--john lazzaro

-------------------------------------------------------------------------
John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
-------------------------------------------------------------------------



From rem-conf Mon Sep 25 11:03:44 2000 
From rem-conf-request@es.net Mon Sep 25 11:03:42 2000
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From: The IESG <iesg-secretary@ietf.org>
SUBJECT: Last Call: RTP Payload Format for 12-bit DAT, 20- and 24-bit
	 Linear Sampled Audio to Proposed Standard
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The IESG has received a request from the Audio/Video Transport Working
Group to consider RTP Payload Format for 12-bit DAT, 20- and 24-bit
Linear Sampled Audio <draft-ietf-avt-dv-audio-02.txt> as a Proposed
Standard.

Note that a companion document, RTP Payload Format for DV Format Video
<draft-ietf-avt-dv-video-03.txt> has already been last called.

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send any comments to the
iesg@ietf.org or ietf@ietf.org mailing lists by October 9, 2000.

Files can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-avt-dv-audio-02.txt
http://www.ietf.org/internet-drafts/draft-ietf-avt-dv-video-03.txt




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------=_NextPart_000_000A_01C02728.F463A500--




From rem-conf Tue Sep 26 06:27:01 2000 
From rem-conf-request@es.net Tue Sep 26 06:27:00 2000
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	id 13duhW-0000NO-00; Tue, 26 Sep 2000 06:23:14 -0700
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To: rem-conf@es.net
Subject: RDRT CEO-  EARLY PROFITS- IPO SPINOFF     
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Read-Rite launches Scion, a new Fibre optics company

Read-rite symbol= RDRT

Read-Rite CEO gets into growth
By Janet Haney, CBS.MarketWatch.com
Last Update: [Timestamp]NewsWatch
Latest headlines
SAN FRANCISCO (CBS.MW) -- Read-Rite Chief Executive Officer Alan Lowe waxed positive about the future growth prospects for the magnetic-recording-head supplier.



Lowe told a crowd of investors and analysts during a presentation at the Banc of America Securities Investment Conference in San Francisco on Thursday that he expects huge unit growth potential for Read-Rite's (RDRT: news, msgs) December quarter, as well as the possibility of profitability.
Lowe added that the company is hiring people as fast as it can for its wafer fabrication facility.
For the September quarter, the CEO said Read-Rite has a "lot of product to ship in the last 10 days of the quarter."
Additionally, Lowe talked about Read-Rite's recent formation of an independent fiber optic company called Scion Photonics which he said hopes to go public.
Scion received funding from Tyco, which will make a presentation at the conference later Thursday.  
Janet Haney is a reporter for CBS.MarketWatch.com.


Read-Rite Corp <RDRT> said late on Wednesday it has formed a new fibre optics company, Scion Photonics Inc, with $25 million in initial funding from Tyco Ventures and Integral Capital Partners. Read-Rite also said it was in discussions with several other interested parties for more technology and capital investments into Scion.
Scion's business charter is "to design, manufacture and market high-performance optical components to the fibre optics communications market," Read-Rite said in a statement.
Scion will act as a separate company, but will have access to Read-Rite's resources and manufacturing infrastructure, Read-Rite chairman Cyril Yansouni said in a statement. Read-Rite is contributing its Milpitas, California wafer fabrication plant, technical and management resources and its knowledge and experience in thin-film technology and materials science to Scion, which will be based in Milpitas.
Tyco Ventures, the venture capital unit of Tyco International Ltd. <TYC>, invested $15 million in Scion and also invested $15 million in Read-Rite through a private placement with the company, Tyco Ventures president Richard Kashnow said in a statement.

CBS article:
http://cbs.marketwatch.com/archive/20000907/news/current/screamers.htx?source=blq/yhoo&dist=yhoo

For the complete streaming audio story users should access
http://www.on24.com/index.html?id=36870&type=av&ref=bizwire


ALWAYS DO RESEARCH BEFORE INVESTING IN ANY STOCK!!!







From rem-conf Tue Sep 26 11:40:20 2000 
From rem-conf-request@es.net Tue Sep 26 11:40:20 2000
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	id 13dzT7-0006cD-00; Tue, 26 Sep 2000 11:28:41 -0700
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	id 13dzT4-0006bl-00; Tue, 26 Sep 2000 11:28:38 -0700
Date: Wed, 27 Sep 2000 03:08:20 +0900
From: donald453@bbc.co.uk
Message-Id: <200009261808.DAA00933@www.daemyung.co.kr>
To: donald453@bbc.co.uk
Subject: Porche Boxter or Luxury Cruise Earn $$$ In Days This Works!!!
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In less than one week, I was starting to receive orders for REPORT #1.
By January 13, I had received 26 orders for REPORT #1. Your goal is
to "RECEIVE at least 20 ORDERS FOR REPORT #1 WITHIN 2 WEEKS. IF YOU
DON4, SEND OUT MORE PROGRAMS UNTIL YOU DO.

My first step in making $50,000 in 90 days was done. By January 30, I
had received 196 orders for REPORT #2. Your goal is to "RECEIVE
AT LEAST 100+ ORDERS FOR REPORT #2 WITHIN 2 WEEKS. IF NOT, SEND OUT MORE

PROGRAMS UNTIL YOU DO. ONCE YOU
HAVE 100 ORDERS, THE REST IS EASY, RELAX, YOU WILL MAKE YOUR $50,000
GOAL."

Well, I had 196 orders for REPORT #2, 96 more than I needed. So I sat
back and relaxed. By March 1, of my e-mailing of 10,000, I received
$58,000 with more coming in every day. I paid off ALL my debts and
bought a much needed new car!

Please take your time to read this plan, IT WILL CHANGE YOUR LIFE
FOREVER$!!! Remember, it won4 work if you don4 try it. This program
does work, but you must follow it EXACTLY!

Especially the rules of not trying to place your name in a different
place.
It won4 work and you4l lose out on a lot of money! In order for this
program to work, you must meet your goal of 20+ orders for REPORT #1,
and 100+ orders for REPORT #2 and you will make $50,000 or more in 90 days.

I AM LIVING PROOF THAT IT WORKS!!! If you choose not to participate in
this program, I am sorry. It really is a great opportunity with little
cost or risk to you. If you choose to participate, follow the program
and you will be on your way to financial security. If you are a fellow
business owner and are in financial trouble like I was, or you want to
start your own business, consider this a sign. I DID! $$

Sincerely,

Johnathon Rourke
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

A PERSONAL NOTE FROM THE ORIGINATOR OF THIS PROGRAM:

By the time you have read the enclosed program and reports, you should
have
concluded that such a program, and one that is legal, could not
have been created by an amateur. Let me tell you a little about myself.
I
had a profitable business for 10 years. Then in 1979 my business
began falling off. I was doing the same things that were previously
successful for me, but it wasn4 working. Finally, I figured it out. It
wasn4
me, it was the economy. Inflation and recession had replaced the stable
economy that had been with us since 1945.

I don4 have to tell you what happened to the unemployment
rate...because
many of you know from first hand experience. There were more
failures and bankruptcies than ever before. The middle class was
vanishing.
Those who knew what they were doing invested wisely and
moved up. Those who did not, including those who never had anything to
save
or invest, were moving down into the ranks of the poor. As
the saying goes, "THE RICH GET RICHER AND THE POOR GET POORER." The
traditional methods of making money will never allow you
to "move up" or "get rich." Inflation will see to that.

You have just received information that can give you financial freedom
for
the rest of your life, with "NO RISK" and "JUST A LITTLE BIT OF
EFFORT." You can make more money in the next few months than you have
ever
imagined. I should also point out that I will not see a penny
of this money, nor anyone else who has provided a testimonial for this
program.

I have retired from the program after sending thousands and thousands of

programs. Follow the program EXACTLY AS INSTRUCTED. Do not
change it in any way. It works exceedingly well as it is now. Remember
to
e-mail a copy of this exciting report to everyone you can think of.
One of the people you send this to may send out 50,000...and your name
will
be on every one of them! Remember though, the more you send
out, the more potential customers you will reach. So my friend, I have
given you the ideas, information, materials and opportunity to become
financially independent.

IT IS UP TO YOU!! NOW DO IT!!

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

Before you delete this program from your in box, as I almost did, take a

little time to read it and REALLY THINK ABOUT IT. Get a pencil and
figure out what could happen when YOU participate. Figure out the worst
possible response and no matter how you calculate it, you will still
make a lot of money! You will definitely get back what you invested. Any

doubts you have will vanish when your first orders come in.

$$$ IT WORKS!!! $$$

Jody Jacobs Richmond, VA

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

HERE4 HOW THIS AMAZING PROGRAM WILL MAKE YOU THOUSANDS OF
DOLLAR$$$$!!!!

This method of raising capital REALLY WORKS 100% EVERY TIME. I am sure
that
you could use up to $50,000 or more in the next 90 days.
Before you say "BULL... ", please read this program carefully. This is
not
a chain letter, but a perfectly legal money making business.

As with all multi-level businesses, we build our business by recruiting
new
partners and selling our products. Every state in the USA allows you
to recruit new multi-level business partners, and we sell and deliver a
product for EVERY dollar received.

YOUR ORDERS COME BY MAIL AND ARE FILLED BY E-MAIL, so you are not
involved
in personal selling. You do it privately in your own
home, store or office. This is the EASIEST marketing plan anywhere! It
is
simply order-filling by email!

*******************************************************************
The product is informational and instructional material, keys to the
secrets
for everyone on how to open the doors to the magic world of E-
COMMERCE, the information highway, the wave of the future!

PLAN SUMMARY:

(1) You order the 4 reports listed below ($5 US each.) They come to you
by
email.

(2) Save a copy of this entire letter and put your name after Report #1
and
move the other names down.

(3) Via the internet, access Yahoo.com or any of the other major search
engines to locate hundreds of bulk email service companies (search for
"bulk email") and have them send 25,000 - 50,000 emails for you about
$49+).

(4) Orders will come to you by postal mail - simply email them the
Report
they ordered. Let me ask you - isn4 this about as easy as it gets?

************************************************************

By the way there are over 50 MILLION email addresses with millions more
joining the internet each year so don4 worry about "running out" or
"saturation". People are used to seeing and hearing the same
advertisements
every day on radio/TV. How many times have you received the
same pizza flyers on your door? Then one day you are hungry for pizza
and
you order one. Same thing with this letter. I received this letter
many times - then one day I decided it was time to try it.

************************************************************

YOU CAN START TODAY - JUST DO THESE EASY STEPS:

STEP #1. ORDER THE FOUR REPORTS

Order the four reports shown on the list below (you can4 sell them if
you
don4 order them). -- For each report, send $5.00 (US) CASH, the NAME
& NUMBER OF THE REPORT YOU ARE ORDERING, YOUR E-MAIL ADDRESS, and YOUR
NAME
& RETURN ADDRESS (in case of a
problem) to the person whose name appears on the list next to the
report.
MAKE SURE YOUR RETURN ADDRESS IS ON YOUR
ENVELOPE IN CASE OF ANY MAIL PROBLEMS! Within a few days you will
receive,
by e-mail, each of the four reports. Save them on your
computer so you can send them to the 1,0004 of people who will order
them
>from you.

STEP #2. ADD YOUR MAILING ADDRESS TO THIS LETTER

a. Look below for the listing of the four reports.
b. After you4e ordered the four reports, delete the name and address
under
REPORT #4. This person has made it through the cycle.
c. Move the name and address under REPORT #3 down to REPORT #4.
d. Move the name and address under REPORT #2 down to REPORT #3.
e. Move the name and address under REPORT #1 down to REPORT #2.
f. Insert your name and address in the REPORT #1 position.
Please make sure you COPY ALL INFORMATION, every name and address,
ACCURATELY!

STEP #3. SAVE THIS LETTER

Take this entire letter, including the modified list of names, and save
it
to your computer. Make NO changes to these instructions. Now you are
ready to use this entire email to send by email to prospects.

Report #1 will tell you how to download bulk email software and email
addresses so you can send it out to thousands of people while you sleep!

Remember that 50,000+ new people are joining the internet every month.

Your cost to participate in this is practically nothing; surely you can
afford $20 (US) and initial bulk mailing cost. You obviously already
have a
computer and an Internet connection and e-mail is FREE!

There are two primary methods of building your downline:

METHOD #1: SENDING BULK E-MAIL Let4 say that you decide to start small,

just to see how it goes, and we4l assume you and all those
involved email out only 2,000 programs each. Let4 also assume that the
mailing receives a 0.5% response. The response could be much
better. Also, many people will email out hundreds of thousands of
programs
instead of 2,000. (Why stop at 2000?). But continuing with this
example, you send out only 2,000 programs. With a 0.5% response, that is

only 10 orders for

REPORT #1. Those 10 people respond by sending out 2,000 programs each
for a
total of 20,000. Out of those 0.5% 100 people respond and
order

REPORT #2. Those 100 mail out 2,000 programs each for a total of
200,000.
The 0.5% response to that is 1,000 orders for

REPORT #3. Those 1,000 send out 2,000 programs each for a 2,000,000
total.
The 0.5% response to that is 10,000 orders for

REPORT #4. That4 10,000 $5 bills for you. CASH!!!

Your total income in this example is $50 + $500 + $5,000 + $50,000 for a

total of $55,550!!!

REMEMBER FRIEND, THIS IS ASSUMING 1,990 OUT OF THE 2,000 PEOPLE YOU MAIL
TO
WILL DO ABSOLUTELY NOTHING AND
TRASH THIS PROGRAM! DARE TO THINK FOR A MOMENT WHAT WOULD HAPPEN IF
EVERYONE, OR HALF SENT OUT 100,000
PROGRAMS INSTEAD OF 2,000. Believe me, many people will do just that,
and
more!

METHOD #2: PLACING FREE ADS ON THE INTERNET
Advertising on the internet is very, very inexpensive, and there are
HUNDREDS of FREE places to advertise. Let4 say you decide to start
small just to see how well it works. Assume your goal is to get ONLY 10
people to participate on your first level. (Placing a lot of FREE ads on

the Internet will EASILY get a larger response.) Also assume that
everyone
else in YOUR ORGANIZATION gets ONLY 10 downline members.
Look how this small number accumulates to achieve the STAGGERING results

below:

1st level--your first 10 send you
$5..................................$50
2nd level--10 members from those 10 ($5 x 100).............$500
3rd level--10 members from those 100 ($5 x 1000)........$5,000
4th level--10 members from those 1000 ($5 x 10,000)..$50,000

$$$$$$ THIS TOTALS ----------$55,550 $$$$$$

AMAZING ISN4 IT? Remember friends, this assumes that the people who
participate only recruit 10 people each. Think for a moment what
would happen if they got 20 people to participate! Most people get 1004
of
participants and many will continue to work this program, sending
out programs WITH YOUR NAME ON THEM for years! THINK ABOUT IT!

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

People are going to get emails about this plan from you or somebody else
and
many will work this plan. The question is, don4 you want your
name to be on the emails they will send out?

* * * DON4 MISS OUT!!! * * * JUST TRY IT ONCE!!! * * *

* * SEE WHAT HAPPENS!!! *** YOU4L BE AMAZED!!!* *

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

ALWAYS PROVIDE SAME-DAY SERVICE ON ALL ORDERS!

This will guarantee that the e-mail THEY send out with YOUR name and
address
on it will be prompt because they can4 advertise until they
receive the report!

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

GET STARTED TODAY: PLACE YOUR ORDER FOR THE FOUR REPORTS NOW.

Notes: ALWAYS SEND $5 CASH (U.S. CURRENCY) FOR EACH REPORT. CHECKS NOT
ACCEPTED. Make sure the cash is concealed
by wrapping it in two sheets of paper. On one of those sheets of paper
write:

(a) the number & name of the report you are ordering

(b) your e-mail address, and

(c) your name & postal address.

REPORT #1 "The Insider4 Guide to Advertising for Free on the Internet"

ORDER REPORT #1 FROM:
Andrew Skidmore
9379 Alexander Rd
Alexander, NY  14005
USA

REPORT #2 "The Insider4 Guide to Sending Bulk E-mail on the Internet"

ORDER REPORT #2 FROM:
Lars Pedersen
Skejbygaardsvej 7, 1, 10
8240 Risskov
Denmark

REPORT #3 "The Secrets to Multilevel Marketing on the Internet"

ORDER REPORT #3 FROM:
John Cole Werner
PO Box 3281
Lihue, HI 96766

REPORT #4 "How to become a Millionaire utilizing the Power of Multilevel

Marketing and the Internet"

ORDER REPORT #4 FROM:
Zac Majors
2242 E Woodchuck Way
Sandy, UT 84093

******* TIPS FOR SUCCESS *******

TREAT THIS AS YOUR BUSINESS! Be prompt, professional, and follow the
directions accurately. Send for the four reports IMMEDIATELY
so you will have them when the orders start coming in because: When you
receive a $5 order, you MUST send out the requested
product/report. It is required for this to be a legal business and they
need the reports to send out their letters (with your name on them!)

-- ALWAYS PROVIDE SAME-DAY SERVICE ON THE ORDERS YOU RECEIVE. -- Be
patient
and persistent with this program - If you follow
the instructions exactly - results WILL FOLLOW. $$$$

******* YOUR SUCCESS GUIDELINES *******

Follow these guidelines to guarantee your success: If you don4 receive
20
orders for REPORT #1 within two weeks, continue advertising or
sending e-mails until you do. Then, a couple of weeks later you should
receive at least 100 orders for REPORT #2. If you don4, continue
advertising or sending e-mails until you do. Once you have received 100
or
more orders for REPORT #2, YOU CAN RELAX, because the
system is already working for you, and the cash will continue to roll
in!

THIS IS IMPORTANT TO REMEMBER: Every time your name is moved down on the

list, you are placed in front of a DIFFERENT report. You
can KEEP TRACK of your PROGRESS by watching which report people are
ordering
>from you.

To generate more income, simply send another batch of e-mails or
continue
placing ads and start the whole process again! There is no limit to
the income you will generate from this business!

Before you make your decision as to whether or not you participate in
this
program, please answer one question. ARE YOU HAPPY WITH
YOUR PRESENT INCOME OR JOB? If the answer is no, then please look at the

following facts about this super simple MLM program:

1. NO face to face selling, NO meetings, NO inventory! NO Telephone
calls,
NO big cost to start! NOthing to learn, NO skills needed! (Surely
you know how to send email?)

2. No equipment to buy - you already have a computer and internet
connection
- so you have everything you need to fill orders!

3. You are selling a product which does NOT COST ANYTHING TO PRODUCE OR
SHIP! (Emailing copies of the reports is FREE!)

4. All of your customers pay you in CA$H! This program will change your
LIFE FOREVER!! Look at the potential for you to be able to quit your
job and live a life of luxury you could only dream about! Imagine
getting
out of debt and buying the car and home of your dreams and being
able to work a super-high paying leisurely easy business from home!

$$$ FINALLY MAKE SOME DREAMS COME TRUE! $$$

ACT NOW! Take your first step toward achieving financial independence.
Order the reports and follow the program outlined above--
SUCCESS will be your reward.

Thank you for your time and consideration.

PLEASE NOTE: If you need help with starting a business, registering a
business name, learning how income tax is handled, etc., contact your
local office of the Small Business Administration (a Federal Agency) at
1-800-827-5722 for free help and answers to questions.

Also, the Internal Revenue Service offers free help via telephone and
free
seminars about business tax requirements. Your earnings are highly
dependent on your activities and advertising. The information contained
on
this site and in the report constitutes no guarantees stated nor
implied. In the event that it is determined that this site or report
constitutes a guarantee of any kind, that guarantee is now void. The
earnings
amounts listed on this site and in the report are estimates only. If you

have any questions of the legality of this program, contact the Office
of
Associate Director for Marketing Practices, Federal Trade Commission,
Bureau
of Consumer Protection in Washington, DC.

================================================

Under Bill s.1618 TITLE III passed by the 105th US Congress this letter
cannot be considered spam as long as the sender includes contact
information and a method of removal.

This is a one time e-mail transmission. No request for removal is
necessary.






From rem-conf Tue Sep 26 16:57:03 2000 
From rem-conf-request@es.net Tue Sep 26 16:57:02 2000
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Message-Id: <200009262351.SAA20998@agevole.cig.mot.com>
Date: Tue, 26 Sep 2000 18:51:29 -0500
From: Qiaobing Xie <xieqb@cig.mot.com>
X-Mailer: Mozilla 4.75 [en] (Win98; U)
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To: AVT List <rem-conf@es.net>
Subject: Comments on draft-ietf-avt-rtp-amr-00.txt
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Hello, everyone:

I have some serious concerns on the design of the AMR payload format as
defined in <draft-ietf-avt-rtp-amr-00.txt>. In brief, the design seems
very limiting in terms of taking advantage of the error tolerant
features built into the AMR codec algorithm. Also, it seems that not
enough consideration has been given to implemenation efficiency and
performance.

Since my concerns are rather fundamental to the design, the email is a
little long, please bear with me..

1) Basic idea

The basic idea of the draft is to transport AMR frames over IP in an RTP
payload, i.e., AMR-frame/RTP/UDP/IP. I have no problem with that. 

Therefore, the packet *passed to the UDP layer* will contain three types
of bits:

   +--------------+
   | header bits  |
   +--------------+
   | Class A bits |
   |    and/or    |
   |  other bits  |
   +--------------+

Here, the "header bits" will include all the AMR speech frame control
bits (ie., frame header), the RTP payload header bits, and the RTP
header bits.

"Class A bits" is the most sensitive coded speech data bits, as defined
by the AMR codec. The "other bits" (also called Class B and C bits in
AMR terminology) are less sensitive coded speech data bits. 

2) Problems with the current AMR payload format design

The design in the current draft apparently was guided under the
misconception that the "Class A bits" of an AMR speech frame need to be
delivered error-free, and if that part of the frame is corrupted the
whole frame should be tossed. This is not correct.

What the AMR codec really wants from the transport is: "if Class A bits
is found corrupted in a frame, raise a Bad Frame flag but still pass the
whole frame to the decoder since the decoder still may want to use Class
B and C bits to help error concealment." The purpose of the flag is to
tell the decoder that Class A bits are bad and do not use them.

However, the draft seems to suggest on the use of a partial checksum
(such as the one provided by the proposed UDP-lite) to protect the
"header bits" AND the "Class A bits" at the transport leverl. This means
if Class A bits have any error, the whole packet will be tossed by the
transport. Even worse, if the packet is a compound payload, you will
have all the AMR frames tossed by the transport if a bit error occurs in
the Class A bits of any one of the enclosed frames.

It becomes more problematic when, in order to support multiple AMR
frames in a compound RTP payload and to use 
the partial checksum, a *very ugly* bit-wise shuffling scheme is
proposed to copy (bit by bit) all the "Class A bits" of each enclosed
AMR speech frame to the front of the packet!! The hugely computational
intensive operation is required no each RTP packet on both ends of the
RTP session.

3) A better way to do it:

To get what AMR codec wants, we should only checksum the "header bits",
and carry an 8-bit CRC in the AMR frame header to cover the "Class A
bits" of the frame. This CRC is for the RTP receiver (that is above the
transport) to verify the integrity of the Class A bits and to raise the
Bad Frame flag before passing the AMR frame to the decoder. 

In other words,

 1. Partial checksum at the transport level to cover all the "header
    bits" -- if checksum fails, toss the whole packet.
 2. CRC in AMR frame header to cover the "Class A bits" -- if CRC fails,
    raise the bad frame flag.
 3. No error checking on the "other bits".

When bundling multiple AMR frames in one RTP packet (the compound
payload), we should use a table of contents (TOC) structure to avoid
bit-wise copying (AMR frames are bit-stuffed to byte boundary). and then
use the transport level checksum to cover the header bits and TOC bits
only (the Class A CRC of each AMR frame is in the TOC). 


   +------------------+
   | RTP header bits  |  -- RTP and payload headers
   +------------------+
   | Table of Contents|  -- list of frame headers of the 
   +==================+ \   included AMR frames
   | Fmr1 Cls A bits  |  \
   +------------------+  |
   | Fmr1 Cls B bits  |  |
   +------------------+  |
   | Fmr1 Cls C bits  |  |
   +==================+  |
   | Fmr2 Cls A bits  |   > not covered by transport level chksum
   +------------------+  |
   | Fmr2 Cls B bits  |  |
   +------------------+  |
   | Fmr2 Cls C bits  |  |
   +==================+  |
   | Fmr3 Cls A bits  |  /
   +------------------+ /
         .....
         .....

This way, you get what the AMR codec asks for and yet avoid bit-wise
copying.


===============================================================
  Qiaobing Xie, Ph.D
  Principal Staff Engineer
  Network Architecture & Technology
  Motorola, Inc.
  TEL: (847) 632-3028 
===============================================================



From rem-conf Tue Sep 26 17:18:02 2000 
From rem-conf-request@es.net Tue Sep 26 17:18:01 2000
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DATE: 26 Sep 00 7:07:44 PM
FROM: 51UL2PxUA@golferslist.com
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SUBJECT: potentially explosive
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 Greetings:

 This Opportunity is Exploding on the Internet!!!!
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 *******************************************************************
 We apologize if this message has reached you in error.
 If you would like to be removed from future mailings,
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 *******************************************************************




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From: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
Organization: Ericsson Research
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Dear Qiaobing, everyone,

The unequal error protection/correction scheme is not designed just for
UDP-Lite. Any mechanism that can exploit the fact that the data is
sorted in sensitivity order is possible to use, e.g. a partial FEC
scheme based on the same techniques as RFC2733.

We have made some design choices that makes the payload format flexible
and bit efficient, i.e. usefull for a wide range of networks and
applications.

Best regards

Magnus Westerlund


Qiaobing Xie wrote:

> Hello, everyone:
>
> I have some serious concerns on the design of the AMR payload format as
> defined in <draft-ietf-avt-rtp-amr-00.txt>. In brief, the design seems
> very limiting in terms of taking advantage of the error tolerant
> features built into the AMR codec algorithm. Also, it seems that not
> enough consideration has been given to implemenation efficiency and
> performance.
>
> Since my concerns are rather fundamental to the design, the email is a
> little long, please bear with me..
>
> 1) Basic idea
>
> The basic idea of the draft is to transport AMR frames over IP in an RTP
> payload, i.e., AMR-frame/RTP/UDP/IP. I have no problem with that.
>
> Therefore, the packet *passed to the UDP layer* will contain three types
> of bits:
>
>    +--------------+
>    | header bits  |
>    +--------------+
>    | Class A bits |
>    |    and/or    |
>    |  other bits  |
>    +--------------+
>
> Here, the "header bits" will include all the AMR speech frame control
> bits (ie., frame header), the RTP payload header bits, and the RTP
> header bits.
>
> "Class A bits" is the most sensitive coded speech data bits, as defined
> by the AMR codec. The "other bits" (also called Class B and C bits in
> AMR terminology) are less sensitive coded speech data bits.
>
> 2) Problems with the current AMR payload format design
>
> The design in the current draft apparently was guided under the
> misconception that the "Class A bits" of an AMR speech frame need to be
> delivered error-free, and if that part of the frame is corrupted the
> whole frame should be tossed. This is not correct.
>
> What the AMR codec really wants from the transport is: "if Class A bits
> is found corrupted in a frame, raise a Bad Frame flag but still pass the
> whole frame to the decoder since the decoder still may want to use Class
> B and C bits to help error concealment." The purpose of the flag is to
> tell the decoder that Class A bits are bad and do not use them.
>
> However, the draft seems to suggest on the use of a partial checksum
> (such as the one provided by the proposed UDP-lite) to protect the
> "header bits" AND the "Class A bits" at the transport leverl. This means
> if Class A bits have any error, the whole packet will be tossed by the
> transport. Even worse, if the packet is a compound payload, you will
> have all the AMR frames tossed by the transport if a bit error occurs in
> the Class A bits of any one of the enclosed frames.
>
> It becomes more problematic when, in order to support multiple AMR
> frames in a compound RTP payload and to use
> the partial checksum, a *very ugly* bit-wise shuffling scheme is
> proposed to copy (bit by bit) all the "Class A bits" of each enclosed
> AMR speech frame to the front of the packet!! The hugely computational
> intensive operation is required no each RTP packet on both ends of the
> RTP session.
>
> 3) A better way to do it:
>
> To get what AMR codec wants, we should only checksum the "header bits",
> and carry an 8-bit CRC in the AMR frame header to cover the "Class A
> bits" of the frame. This CRC is for the RTP receiver (that is above the
> transport) to verify the integrity of the Class A bits and to raise the
> Bad Frame flag before passing the AMR frame to the decoder.
>
> In other words,
>
>  1. Partial checksum at the transport level to cover all the "header
>     bits" -- if checksum fails, toss the whole packet.
>  2. CRC in AMR frame header to cover the "Class A bits" -- if CRC fails,
>     raise the bad frame flag.
>  3. No error checking on the "other bits".
>
> When bundling multiple AMR frames in one RTP packet (the compound
> payload), we should use a table of contents (TOC) structure to avoid
> bit-wise copying (AMR frames are bit-stuffed to byte boundary). and then
> use the transport level checksum to cover the header bits and TOC bits
> only (the Class A CRC of each AMR frame is in the TOC).
>
>    +------------------+
>    | RTP header bits  |  -- RTP and payload headers
>    +------------------+
>    | Table of Contents|  -- list of frame headers of the
>    +==================+ \   included AMR frames
>    | Fmr1 Cls A bits  |  \
>    +------------------+  |
>    | Fmr1 Cls B bits  |  |
>    +------------------+  |
>    | Fmr1 Cls C bits  |  |
>    +==================+  |
>    | Fmr2 Cls A bits  |   > not covered by transport level chksum
>    +------------------+  |
>    | Fmr2 Cls B bits  |  |
>    +------------------+  |
>    | Fmr2 Cls C bits  |  |
>    +==================+  |
>    | Fmr3 Cls A bits  |  /
>    +------------------+ /
>          .....
>          .....
>
> This way, you get what the AMR codec asks for and yet avoid bit-wise
> copying.
>
> ===============================================================
>   Qiaobing Xie, Ph.D
>   Principal Staff Engineer
>   Network Architecture & Technology
>   Motorola, Inc.
>   TEL: (847) 632-3028
> ===============================================================

--

Magnus Westerlund

Audio Technology, Ericsson Research
----------------------------------------------------------------------
Ericsson Radio Systems AB  | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se



From rem-conf Wed Sep 27 10:54:27 2000 
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From: Qiaobing Xie <xieqb@cig.mot.com>
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Magnus,

Correct me if I am wrong, but I find hard to understand your reasoning.

What you are saying is basically that, in order to make the current
payload format design general enough and flexible enough to be used "for
a wide range of networks and applications" and with "any mechanism that
exploit" unequal bit sensitivity, you've chosen to compromise the
implementation efficiency and performance (and even some functionality)
for AMR over RTP/whatever-partial-checksum-UDP/IP.

I think it should be the other way around - the AMR payload design
should primarily focus on finding the optimal solution for
AMR/RTP/UDP*/IP. It doesn't sound right to me for IETF AVT WG to design
something that is meant for a wide range of networks but is
suboptimal/compromised (even functionally handicapped) for IP networks.

Regards,
-Qiaobing Xie

Magnus Westerlund wrote:
> 
> Dear Qiaobing, everyone,
> 
> The unequal error protection/correction scheme is not designed just for
> UDP-Lite. Any mechanism that can exploit the fact that the data is
> sorted in sensitivity order is possible to use, e.g. a partial FEC
> scheme based on the same techniques as RFC2733.
> 
> We have made some design choices that makes the payload format flexible
> and bit efficient, i.e. usefull for a wide range of networks and
> applications.
> 
> Best regards
> 
> Magnus Westerlund
> 
> Qiaobing Xie wrote:
> 
> > Hello, everyone:
> >
> > I have some serious concerns on the design of the AMR payload format as
> > defined in <draft-ietf-avt-rtp-amr-00.txt>. In brief, the design seems
> > very limiting in terms of taking advantage of the error tolerant
> > features built into the AMR codec algorithm. Also, it seems that not
> > enough consideration has been given to implemenation efficiency and
> > performance.
> >
> > Since my concerns are rather fundamental to the design, the email is a
> > little long, please bear with me..
> >
> > 1) Basic idea
> >
> > The basic idea of the draft is to transport AMR frames over IP in an RTP
> > payload, i.e., AMR-frame/RTP/UDP/IP. I have no problem with that.
> >
> > Therefore, the packet *passed to the UDP layer* will contain three types
> > of bits:
> >
> >    +--------------+
> >    | header bits  |
> >    +--------------+
> >    | Class A bits |
> >    |    and/or    |
> >    |  other bits  |
> >    +--------------+
> >
> > Here, the "header bits" will include all the AMR speech frame control
> > bits (ie., frame header), the RTP payload header bits, and the RTP
> > header bits.
> >
> > "Class A bits" is the most sensitive coded speech data bits, as defined
> > by the AMR codec. The "other bits" (also called Class B and C bits in
> > AMR terminology) are less sensitive coded speech data bits.
> >
> > 2) Problems with the current AMR payload format design
> >
> > The design in the current draft apparently was guided under the
> > misconception that the "Class A bits" of an AMR speech frame need to be
> > delivered error-free, and if that part of the frame is corrupted the
> > whole frame should be tossed. This is not correct.
> >
> > What the AMR codec really wants from the transport is: "if Class A bits
> > is found corrupted in a frame, raise a Bad Frame flag but still pass the
> > whole frame to the decoder since the decoder still may want to use Class
> > B and C bits to help error concealment." The purpose of the flag is to
> > tell the decoder that Class A bits are bad and do not use them.
> >
> > However, the draft seems to suggest on the use of a partial checksum
> > (such as the one provided by the proposed UDP-lite) to protect the
> > "header bits" AND the "Class A bits" at the transport leverl. This means
> > if Class A bits have any error, the whole packet will be tossed by the
> > transport. Even worse, if the packet is a compound payload, you will
> > have all the AMR frames tossed by the transport if a bit error occurs in
> > the Class A bits of any one of the enclosed frames.
> >
> > It becomes more problematic when, in order to support multiple AMR
> > frames in a compound RTP payload and to use
> > the partial checksum, a *very ugly* bit-wise shuffling scheme is
> > proposed to copy (bit by bit) all the "Class A bits" of each enclosed
> > AMR speech frame to the front of the packet!! The hugely computational
> > intensive operation is required no each RTP packet on both ends of the
> > RTP session.
> >
> > 3) A better way to do it:
> >
> > To get what AMR codec wants, we should only checksum the "header bits",
> > and carry an 8-bit CRC in the AMR frame header to cover the "Class A
> > bits" of the frame. This CRC is for the RTP receiver (that is above the
> > transport) to verify the integrity of the Class A bits and to raise the
> > Bad Frame flag before passing the AMR frame to the decoder.
> >
> > In other words,
> >
> >  1. Partial checksum at the transport level to cover all the "header
> >     bits" -- if checksum fails, toss the whole packet.
> >  2. CRC in AMR frame header to cover the "Class A bits" -- if CRC fails,
> >     raise the bad frame flag.
> >  3. No error checking on the "other bits".
> >
> > When bundling multiple AMR frames in one RTP packet (the compound
> > payload), we should use a table of contents (TOC) structure to avoid
> > bit-wise copying (AMR frames are bit-stuffed to byte boundary). and then
> > use the transport level checksum to cover the header bits and TOC bits
> > only (the Class A CRC of each AMR frame is in the TOC).
> >
> >    +------------------+
> >    | RTP header bits  |  -- RTP and payload headers
> >    +------------------+
> >    | Table of Contents|  -- list of frame headers of the
> >    +==================+ \   included AMR frames
> >    | Fmr1 Cls A bits  |  \
> >    +------------------+  |
> >    | Fmr1 Cls B bits  |  |
> >    +------------------+  |
> >    | Fmr1 Cls C bits  |  |
> >    +==================+  |
> >    | Fmr2 Cls A bits  |   > not covered by transport level chksum
> >    +------------------+  |
> >    | Fmr2 Cls B bits  |  |
> >    +------------------+  |
> >    | Fmr2 Cls C bits  |  |
> >    +==================+  |
> >    | Fmr3 Cls A bits  |  /
> >    +------------------+ /
> >          .....
> >          .....
> >
> > This way, you get what the AMR codec asks for and yet avoid bit-wise
> > copying.
> >
> > ===============================================================
> >   Qiaobing Xie, Ph.D
> >   Principal Staff Engineer
> >   Network Architecture & Technology
> >   Motorola, Inc.
> >   TEL: (847) 632-3028
> > ===============================================================
> 
> --
> 
> Magnus Westerlund
> 
> Audio Technology, Ericsson Research
> ----------------------------------------------------------------------
> Ericsson Radio Systems AB  | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se



From rem-conf Thu Sep 28 12:33:57 2000 
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Subject: 10/4 Studying and Modeling Real Audio Traffic --  John
  Heidemann, USC Information Sciences Institute  
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Berkeley Multimedia, Graphics and Interfaces Seminar

Studying and Modeling Real Audio Traffic

Wednesday October 4, 2000, 
1:10-2:30 p.m. PDT
Fujitsu Seminar Room (405 Soda Hall) 

John Heidemann 
USC Information Sciences Institute  

The delivery of multimedia content on the Internet is rapidly growing in
importance. While there have been extensive studies on the growth and
effects of Hyper Text Transfer Protocol (HTTP) traffic used on the Web,
little work has been performed in analyzing streaming multimedia traffic.
Based on traces from broadcast.com, we will present a study of Real Audio
traffic. We will describe the characteristics of typical real audio flows
and compare them to web traffic. In addition, we observed consistencies in
audio traffic packet sizes and data rate patterns that may be useful as a
tool for identifying audio data flows. Finally, we will present recent
analysis of real audio traffic and show how it has guided our approaches to
model this traffic in simulation. This is joint work with Art Mena and
Kun-chan Lan at USC.

---------------

The seminar is broadcast live on the Internet Mbone and as a Real Networks
G2 broadcast.

You can connect to the live RealNetworks broadcast at:
http://bmrc.berkeley.edu/bibs/cs298-5

You can also directly put in this url into the Real Player: 
rtsp://media2.bmrc.berkeley.edu/encoder/cs298-5.rm

To do so you will need to have the "Real Player G2" installed, which is
available from:
http://www.real.com/products/player/downloadrealplayer.html

To tune into the Internet MBone broadcast, look for the announcement in
your MBone session directory program ('sdr').  If you are not receiving the
announcement you may be able to receive the session by manually configuring
the client programs ('vic', and 'vat') with the session addresses:

low bit rate
	video: 233.0.25.1/22334
	audio: 233.0.25.2/22446

medium bit rate
	video: 233.0.25.129/22334
	audio: 233.0.25.2/22446

You can get further information about this seminar, and access to replays
of previous seminars at the MIG Seminar web page:

http://media2.bmrc.berkeley.edu/bibs/archive.cfm

Sponsored by the Berkeley Multimedia Research Center 




From rem-conf Fri Sep 29 01:55:31 2000 
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To: <sampat@miel.mot.com>
Cc: <rem-conf@es.net>, <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: Types of MPEG-4 streams over IP
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Dear Kuntal, all,

Kuntal wrote :
isn't the "ERROR_PROTECTION_STREAM" (from 14496-3/Amd.1:1999 Part 3,
Section 9, pp. 124), which is the error protected audio stream also a
unique streamType?

I contacted our audio expert Ralf Funken and he gave the reply below. M=
y
conclusion : an error protection stream is a special audio stream as
signaled in the AudioSpecificInfo. It is not another type of stream tha=
t
contains audio and consequently there is no need for another unique
streamType.

Kind regards,

Jan

---------------------------

The Error Protection (EP) tool in MPEG-4 Audio Version 2 acts as an
intermediate operation between the audio core coder and the MPEG-4 Syst=
ems
multiplexer. Each audio elementary stream coming from the audio encoder=
 is
packed by the EP-tool (like adding CRC checks, applying interleaving,
applying FEC coding etc.) and transferred to the multiplexer. All
information required to decode the EP protected audio stream is contain=
ed in
the packed EP elementary stream itself; there is no specific channel fo=
r EP
side-information or whatsoever.

The usage of the EP-tool is signaled by the AudioSpecificInfo data (thi=
s is
the audio-related header information which is transmitted at the start =
of
each elementary stream). In the AudioSpecificInfo, the epConfig field
signals if and how the EP-tool must be used.

The use of the EP-tool does not affect the audio object type indication=
 in a
stream. The audio object type determines the audio encoding/decoding
algorithm. Whether you use EP or not is signaled separately, see above.=


The indication of profile@level defines limits on whether a MPEG-4 rece=
iver
must implement the EP-decoder or not. Some levels allow the EP-tool to =
be
used, others do not.

---------------------------


************************
Jan van der Meer
Philips - Digital Networks
Building OAN-4
P.O. Box 80002
5600 JB Eindhoven
The Netherlands
Phone +31 402 735 774
Fax +31 402 735 545
Email jan.vandermeer@philips.com
************************


-----Original Message-----
From:	sampat@miel.mot.com [mailto:sampat@miel.mot.com]
Sent:	vrijdag, 22 september, 2000 04:51
To:	Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	Re: Types of MPEG-4 streams over IP


Jan,
isn't the "ERROR_PROTECTION_STREAM" (from 14496-3/Amd.1:1999 Part 3,
Section 9, pp. 124), which is the error protected audio stream also a
unique streamType?

-Kuntal.

DSP Group,
Motorola, India.

On Thu, 21 Sep 2000 jan.vandermeer@philips.com wrote:

> Dear all,
>
> I would like to verify my understanding on the types of MPEG-4 stream=
s
that
> need tools for transport over IP. I summarized my understanding by th=
e
> following list :
>
> - ObjectDescriptorStream
> - ClockReferenceStream (is this needed over IP ?)
> - SceneDescriptionStream
> - VisualStream (with and without use of MPEG-4 Systems)
> - AudioStream (with and without use of MPEG-4 Systems)
> - MPEG-7Stream
> - IPMPStream
> - ObjectContentInfoStream
> - FlexMux stream
> - MP4 file
>
> For MP4 files only a standard MIME type is required. For all other st=
reams
> also use of RTP and SDP need to be defined. Please correct me if I am=

wrong.
>
> The framework document (Singer et al) indicates that also a MIME type=
 is
> needed for an MPEG-4 MuxCodeTable. Is this required indeed, or can we=

assume
> (/require) that the MuxCodeTable is transported inside the same FlexM=
ux
> stream (in a packet with simple mode). This could provide an easy sol=
ution
> for the timing of applicability of the table that would need to be
resolved
> otherwise when using e.g. HTTP; however it would require MuxCodeTable=
 to
> become a new entry in the table for streamType values, right ?
>
> Regards,
>
> Jan
>
> ************************
> Jan van der Meer
> Philips - Digital Networks
> Building OAN-4
> P.O. Box 80002
> 5600 JB Eindhoven
> The Netherlands
> Phone +31 402 735 774
> Fax +31 402 735 545
> Email jan.vandermeer@philips.com
> ************************
>
>
>

=



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Subject: Why So Much Smart Money Is So High on Read-Rite !!         
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Commentary: Herb on TheStreet.com
--------------------------------------------------------------------------------
Why So Much Smart Money Is So High on Read-Rite
By Herb Greenberg 
Senior Columnist
9/26/00 6:30 AM ET 

Read-Rite(symbol= RDRT)

Feeling benevolent, so let's get bullish: 

Reading right: As I've written in RealMoney.com's Columnist Conversation, the 
one stock quite a few of my smartest sources are yapping about is Read-Rite 
(RDRT:Nasdaq - news - boards), the (until very recently) long-forgotten maker of 
heads for disk drives. The only reason I take it seriously is because of the variety of 
investors in the stock (from seasoned and savvy traders to some of the 
most-dogged researchers I know). 

Most of the sizzle surrounding the company in recent weeks has been tied to the 
launch of a new division called Scion Photonics, which is developing optical 
wafers for use in the fiber-optic networks, and which was initially funded with $25 
mil from Tyco Ventures and Roger McNamee's Integral Capital, which got a 
quarter of the company in return. 

But that's only part of the story: 

According to Scott Turkel of TCM Partners, who has had his share of hits and 
misses in this column, and who also happens to be the only on-the-record holder 
among my sources, the company without Scion is worth about where it trades 
today, $10.50, or around 1 times sales. "They're completely sold out in their core 
business in the fourth quarter," he says, "and for the first time, they have pricing 
power." (One reason is that disk drives are no longer sold mostly for PCs; they've 
become a staple in storage networks.) 

Scion, meanwhile, is currently valued at around $100 million (based on 
Tyco/Integral's 25% stake). "Chump change," says Turkel. That's because the 
valuation is without even having a marketable product; the first wafer isn't expected 
to hit the market until next year (which, I should point out, is why some skeptics 
are, uh, skeptical). 

But another very sharp manager I know, who is often short stocks, said he saw 
Read-Rite at several recent conferences, and "I thought the story got better 
between Salomon Smith Barney and Banc of America, both on the optics side 
and on their base business. They actually showed a slide of the optical wafer 
prototype they had made; they said they are sampling product with several 
customers. They said, Our customers have said, 'If you can make them, we'll buy 
them.' In other words, the move to optical is less theoretical than it was a month 
ago." 

What's more, according to this money manger, who is great at spotting nuances, 
"They went from saying, 'We'll be break-even cash flow in Q-4 from core 
businesses,' to saying, 'We are on allocation and we may actually make money in 
Q4' from the core business.' " 

Based on that, Turkel (who first bought the stock when it was $4 not long ago) 
thinks he now owns a $10 stock that is worth $25. 

P.S.: Read-Rite recently paid the first installment of interest on a convertible bond 
with cash, rather than stock, which was an alternative. (The cash came from the 
State of Wisconsin Investment Board, already a large Read-Rite investor.) 
Translation to some investors: The only reason the state paid with cash is because 
it thinks the stock is going higher. Or, put another way, Wisconsin, which already 
owns a 20% stake, wouldn't have sunk in even more cash if it didn't think it would 
make a decent return. (Did I really write something that glowing? Must be some 
kind of a market top!) 

(Voluntary Disclosure: Position- Long)



Read-Rite CEO gets into growth
By Janet Haney, CBS.MarketWatch.com
Last Update: [Timestamp]NewsWatch
Latest headlines
SAN FRANCISCO (CBS.MW) -- Read-Rite Chief Executive Officer Alan Lowe waxed positive about the future growth prospects for the magnetic-recording-head supplier.

Lowe told a crowd of investors and analysts during a presentation at the Banc of America Securities Investment Conference in San Francisco on Thursday that he expects huge unit growth potential for Read-Rite's (RDRT: news, msgs) December quarter, as well as the possibility of profitability.
Lowe added that the company is hiring people as fast as it can for its wafer fabrication facility.
For the September quarter, the CEO said Read-Rite has a "lot of product to ship in the last 10 days of the quarter."
Additionally, Lowe talked about Read-Rite's recent formation of an independent fiber optic company called Scion Photonics which he said hopes to go public.
Scion received funding from Tyco, which will make a presentation at the conference later Thursday.  
Janet Haney is a reporter for CBS.MarketWatch.com.











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From: "Lawrence A. Rowe" <Rowe@bmrc.berkeley.edu>
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Hi everyone -

It is not perfect yet - no Berkeley Lecture Browser (BLB) until we
finish a set of database/UI changes we are making to <a
href="http://bmrc.berkeley.edu/">BIBS</a> to support miscellaneous
lectures - but I have published the video+slides from the Open Mash
Workshop. The workshop material is available at
	http://www.openmash.org/events/00workshop
You'll have to flip slides yourself while watching the video.  Once the 

In addition, I want to announce that we have *finally* hired one of the
full-time staff programmers for the project.  His name is Lloyd Lim. He
will be making a big difference once he gets up to speed.  NO, NO, don't
try to hire him away  :)!

Watch the consortium website over the next couple of months - we should
start to make some visible progress.
	Larry 
-- 
Professor Lawrence A. Rowe          Internet:  Rowe@BMRC.Berkeley.EDU
Computer Science Division - EECS       Phone: 510-642-5117
University of California, Berkeley       Fax: 510-642-5615
Berkeley, CA 94720-1776            URL: http://bmrc.berkeley.edu/~larry



