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From rem-conf Sun Oct 01 15:25:07 2000 
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From rem-conf Sun Oct 01 20:24:02 2000 
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From rem-conf Sun Oct 01 23:16:59 2000 
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From: "Toshiyuki Nomura" <t-nomura@ccm.cl.nec.co.jp>
To: "John Lazzaro" <lazzaro@cs.berkeley.edu>
Cc: <mpeg4_vm@tnt.uni-hannover.de>, <rem-conf@es.net>
References: <200009251800.LAA05183@snap.CS.Berkeley.EDU>
Subject: Re: LATM specified in a public MPEG document?
Date: Mon, 2 Oct 2000 15:14:20 +0900
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Dear John and members,

As you pointed out, it seems difficult to transmit SA data
by LATM. I have forwarded this issue to MPEG Audio Experts
and have not been received any objections.
Therefore, I'll modify the I/D text so that SA data
MUST NOT be handled by the LATM-based RTP specification,
following the suggestion below:

John Lazzaro wrote:
> 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.  

The modified text will be published as  
draft-ietf-avt-rtp-mpeg4-es-05.txt.

Best Regards,

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






From rem-conf Mon Oct 02 03:26:48 2000 
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Date: Mon, 02 Oct 2000 03:21:53 -0700
To: rem-conf@es.net
From: Ross Finlayson <finlayson@live.com>
Subject: Announcing: An open-source library for RTP/RTCP streaming
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[This is something that I've been promising people for a long time.  I'm 
pleased to report that I finally have it in a releasable state.  (Not 
perfect, but releasable..)]

The C++ source code that forms the basis of my "liveCaster" (& 
"playRTPMPEG") media streaming tools is now available as "open source" 
(released under the LGPL).  This code forms a set of libraries that can be 
used within RTP/RTCP streaming applications, and/or can be extended to 
support additional media types and codecs (in addition to MP3 audio).  For 
a description of these libraries, and instructions for how to install and 
build them, see <http://live.sourceforge.net/>.

You can use this code right now to build real applications - in fact, the 
distribution includes code for building two demo applications: (i) a server 
that reads a MP3 file and streams it (using RTP/RTCP), and (ii) a client 
that reads from the same RTP/RTCP stream, and outputs the resulting MP3 
stream to stdout.

These libraries can be built and run on Unix (various flavors) and 
Windows.  (However, the demo applications mentioned above don't yet compile 
on Windows.)

These libraries are similar to existing toolkits such as "Open Mash", 
although - for now, at least - not nearly as rich in functionality.  Also, 
they do not require the use of a specific scripting language - or any 
scripting language at all.  (However, the libraries can, and have, been 
used with scripting languages - e.g., "liveCaster" and "playRTPMPEG" were 
built using these libraries in combination with Tcl.)

Once again, the URL for this code is <http://live.sourceforge.net/>.

         Ross.





From rem-conf Mon Oct 02 06:19:01 2000 
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From: "Nikos Laoutaris" <laoutaris@noc.uoa.gr>
To: <rem-conf@es.net>
Subject: video frame arrival traces
Date: Mon, 2 Oct 2000 16:06:40 +0300
Organization: University of Athens, Networks Operation Center
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This is a multi-part message in MIME format.

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Dear list members

    I am searching for traces of interarrival distributions or (better) =
one-way delay distributions of streaming video, at the frame level. That =
is, i would like to have a real measurement of the distortion of =
interstream synchronization of a series of equally spaced frames (at the =
source) as they travel over a (multi hop) network path. This information =
is to be used for the evaluation of an adaptive playout controller for =
packet video receivers.

Thanking you in advance

Nikos Laoutaris
Communication Networks Laboratory
National and Kapodestrian University of Athens, Greece
e-mail : laoutaris@noc.uoa.gr
tel. : +301 7275603
fax : +301 7275601

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Diso-8859-7" =
http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.2919.6307" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT size=3D2>Dear list members</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT size=3D2>&nbsp;&nbsp;&nbsp; I am searching for traces&nbsp;of =

interarrival distributions or (better) one-way delay distributions of =
streaming=20
video, at the frame level. That is, i would like to have a real =
measurement of=20
the distortion of interstream synchronization of a&nbsp;series of =
equally=20
spaced&nbsp;frames (at the source)&nbsp;as&nbsp;they travel over a =
(multi hop)=20
network path. This information is to be used for the evaluation of an =
adaptive=20
playout controller for packet video receivers.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT size=3D2>Thanking you in advance</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT size=3D2>Nikos Laoutaris<BR>Communication Networks=20
Laboratory<BR>National and Kapodestrian University of Athens, =
Greece<BR>e-mail :=20
<A href=3D"mailto:laoutaris@noc.uoa.gr">laoutaris@noc.uoa.gr</A><BR>tel. =
: +301=20
7275603<BR>fax : +301 7275601</FONT></DIV></BODY></HTML>

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From rem-conf Mon Oct 02 06:35:56 2000 
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Date: Mon, 02 Oct 2000 15:32:45 +0200
From: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
Organization: Ericsson Research
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Hello,

The payload format is designed with IP networks in mind. The problem we see is
that IP is transported over a wide range of different links with different
behavior, e.g. different radio and wireline links. Radio and wire do not have
the same error characteristics. AMR will be widly used over wireless links and
on these links bandwidth efficiency is extremely important. This lead to the
bitwise sorting. Octet sorting needs octet alignement for each block of the
payload, i.e. padding at the end of each block. Which we considered to be to
expensive. The optional sensitivity sorting facilitates the use of different
error robustness schemes. The need for different robustness enhancing schemes
may depend on the application and the lower transport layers. The application
can be for example conversational speech or streaming. Lower layers are here
layer 1 and 2.

Regards

Magnus W.


Qiaobing Xie wrote:

> 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

--

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





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Subject: DDMA workshop CFP
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--=====================_970498602==_
Content-Type: text/plain; charset="us-ascii"

Dear Colleague,

You are invited to submit a paper to the International "Distributed Dynamic
Multiservice Architectures" (DDMA) Workshop. This workshop is in conjunction
with the 21st International Conference on Distributed Computing Systems
(ICDCS2001), one of the major conferences in the Distributed Computing
Systems Area.

DDMA workshop covers aspects concerning adaptive architectures and service
composition and management.

The submission deadline is November 1st. Further details are at the web
site http://ddma.enst-bretagne.fr/. A text version is also in attachment.

The DDMA workshop will be held in Phoenix, Arizona; April 16-19, 2001.

[We apologize if you receive multiple copies of this announcement]

--=====================_970498602==_
Content-Type: text/plain; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
Content-Disposition: attachment; filename="ddma-cfp.txt"

					Call for Papers

				International Workshop on
			Distributed Dynamic Multiservice Architectures
				http://ddma.enst-bretagne.fr/

				in conjunction with ICDCS2001
				sponsored by IEEE Computer Society

				April 16 =96 19, 2001
				Phoenix, Arizona, USA

  The growing need to integrate new services while complying with the=
 requirements of end=20
users puts strong emphasis on the management and combination of future=
 services and the QoS=20
they should be able to provide and guarantee. Dynamic Multiservice=
 Architectures constitute a=20
challenging paradigm for distribution theory and practice due to their=
 unique requirements in terms=20
of adaptability, behavioural integration, separation of concerns, service=
 composition and service=20
management. In particular, the deployment of multimedia telecommunication=
 services is a highly=20
challenging area where those paradigms should apply because advanced=
 services require that the=20
user's profiles, preferences and context be taken into account. "Distributed=
 Dynamic=20
Multiservice Architectures" hence add a new dimension to distributed=
 computing to help the=20
composition and deployment of new services guaranteeing final user's=
 requirements.
  The goals of this workshop are to bring together users and researchers to=
 present their=20
recent work related to diverse aspects of dynamic multiservice=
 architectures, their fundamental=20
issues, their paradigms and some appropriate approaches and experiments. The=
 workshop will=20
thus present opportunities for discussing further evolutions and their=
 expected benefits.

Scope of Workshop
-----------------
The scope of this workshop encompasses but is not limited to :

Evolution of distributed multiservice architectures
	Separation of aspects and composition of aspects.
	Methodological approaches for aspect oriented programming.
	Components as cornerstones for dynamic composition.
	Composition requirements for reflection.
	Auto-adaptive ad hoc architectures and protocols.
Impact on the management and control
	Management of adaptive architectures
	Management and monitoring of composed services.
Platforms, tools, languages, ...=20
	Languages, tools and platforms for auto-adaptive systems.
	Environments for specification and design of dynamic architectures.
	Experience with design, implementation and use of auto-adaptive systems.
Application to telecommunication services
	Open critical issues in terms of composition of services.
	Panorama of QoS integration approaches for multimedia telecom services.
	Paradigms and requirements for telecom services architectures.
	Architectures for 3rd generation multimedia services.
	Distribution protocols for telecom (WAP, CORBA, CAMEL, mobile agents).

Submission Instructions and Workshop Format
-------------------------------------------
	Interested participants should submit a 4-8 pages position paper that=
 focuses on one of the=20
relevant topics or related issues. Papers will be reviewed by the program=
 committee. Electronic=20
submissions should be sent to zied.choukair@enst-bretagne.fr. Mail body=
 should include name,=20
address, affiliation and a brief biography along with paper's title and=
 abstract.
	The workshop will follow the following format: The morning and part of the=
 afternoon will=20
be dedicated to presentations and discussions of accepted proposals,=
 including demonstrations of=20
existing software. The afternoon session will end with an identification of=
 issues, followed by=20
discussion. The organisers will then elaborate a summary of the workshop=
 based upon a summary=20
of the papers and the major outlined points of the discussion.

Program Committee Members :
	Mehmet Aksit, Twente University, NL
	Zi=E8d Choukair, ENST Bretagne, FR
	Jaime Delgado, Universitat Pompeu Fabra, ES
	Byung-Guk Kim, University of Massachusetts Lowell, USA
	Marwan Krunz, University of Arizona, USA
	Abdul H. Sadka, University of Surrey, GB
	Antonio Rito Silva, University of Lisbon, PT
	Timothy K. Shih, Tamkang University, TW
	Katsuya Tanaka, Tokyo Denki University, JP

Important Dates :
	Paper submission due : 	Nov 1, 2000
	Notification : 		Dec 30, 2000
	Camera ready : 	 	Jan 20, 2001
	Conference : 		April 16-19, 2001

Web Pages :
	Conference URL : http://cactus.eas.asu.edu/ICDCS2001/

--=====================_970498602==_--




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                            Call for Papers

                         2nd IP Telephony Workshop
	 =20
            April 2-3, 2001 - Columbia University, New York City
                 http://www.fokus.gmd.de/events/iptel2001/


Objectives
----------=20
Internet telephony is rapidly evolving from research to design
and deployment. The objectives of the IP Telephony Workshop are
to bring together researchers, developers, vendors and service=20
providers active in this area and stimulate discussion on=20
innovation, research, implementation, deployment experiences and
future directions.

Scope & Topics=20
--------------
Original technical articles related to IP telephony are solicited.=20
Only papers with significant technical content, not "white papers"
or tutorials, will be considered for publication:

     - Research papers (unique ideas, novel algorithms,=20
       architectures, measurements, theoretical and/or analytical=20
       contributions)=20
     - Surveys, state-of-the-art studies, technology comparisons=20
     - Implementation and deployment reports=20
     - Standardization reports=20

Particular areas of interest include, but are not limited to, the
following:=20

     - Integration with Internet services (e.g., web, instant=20
       messaging, games)=20
     - Added-value services (e.g., call centers, conferencing)=20
     - Mobility and 3rd generation wireless=20
     - Authentication, authorization, accounting, charging,=20
       settlement
     - QoS support=20
     - Security (e.g., privacy, authentication, certification=20
       authorities, firewall traversal)=20
     - Call signaling & processing=20
     - Feature creation=20
     - Supporting services (e.g., call routing, lookup services)=20
     - Audio & video encoding and transmission=20
     - Management and provisioning=20
     - Interworking with the PSTN=20
     - Design and deployment considerations (e.g., performance,=20
       scalability, reliability)=20

Important Dates
---------------=20
Full paper due                November 27th, 2000
Notification of acceptance    January 12th, 2001
Final version due             January 31st, 2001
Program published and         February 5th, 2001
registration opens
Workshop                      April 2nd-3rd, 2001

iptel2001 Organizing Committee=20
------------------------------
Program Chair        =20
 H. Schulzrinne       Columbia University=20
Program Committee=20
 M. Arango            Sun Microsystems
 F. Baker             Cisco
 W. Bauerfeld         T-Nova
 G. Bond              AT&T Research
 S. Bradner           Harvard University
 G. Carle             GMD FOKUS
 J. Crowcroft         UCL
 C. Huitema           Microsoft
 G. S. Kuo            National Central University, Taiwan
 J. Kuthan            GMD Fokus
 T. Magedanz          IKV++ GmbH
 W. Marshall          AT&T Research
 K. Mehdi             University of Kansas
 D. Oran              Cisco
 J. Ott               University of Bremen
 T. La Porta          Bell Labs
 B. Rosen             Marconi
 J. Rosenberg         dynamicsoft
 H. Sinnreich         MCI WorldCom
 R. Steinmetz         Technical University of Darmstadt
 H. St=FCttgen          NEC CCRLE
 W. Wimmreuter        Siemens
 L. Wolf              University of Karlsruhe
 A. Wolisz            Technical University of Berlin
 M. Zitterbart        Technical University of Braunschweig=20

Submission Instructions=20
-----------------------
Authors are invited to submit full papers written in English=20
before November 27th, 2000. The submissions will be reviewed, and
accepted papers will be included in the program. Notifications of
acceptance will be sent out on January 12th, 2000. Deadline for=20
submission of camera-ready copies is January 31st, 2001. Authors=20
of accepted papers will need to sign a Copyright Transfer Form and
submit a Netbib entry.=20

Papers must be submitted electronically using the Web site at
          http://www.cs.columbia.edu/iptel
Submissions must be in PDF or Postscript; any other documents=20
cannot be accepted. Postscript papers must use only standard=20
PostScript fonts: Times Roman, Courier, Symbol, and Helvetica.=20
Papers must be formatted according to the IEEE Transactions format
except for the font size, which MUST be 11pt. Templates are=20
available at the Web site
          http://www.fokus.gmd.de/events/iptel2001/cfp/=20
Because of the size limitation on the final manuscript, and to=20
ensure that the reviewed paper and the final version have a=20
similar size, papers with more than 11 pages cannot be reviewed.
Submissions must include: title, authors, affiliation, abstract,=20
list of keywords, and contact information. One of the authors of=20
each accepted paper must present the paper at iptel'2001.=20

Contact Address
---------------
Please, send all your inquiries regarding iptel2001 to=20
               iptel2001@egroups.com.



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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
time to read the attached program, 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 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'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 youdon'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 justthat, 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:
 
 Becki Owens
 30 Sea Star Court
 Dana Point, CA 92629

 REPORT #2 "The Insider's Guide to Sending Bulk E-mail
 on the Internet"
 ORDER REPORT #2 FROM:
 
 Kristin Butler
 32 McCullough Rd.
 Cody, WY 82414



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

  ADRP10
 PO Box 7794
 Athens, GA 30604-7794


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

 P. Drucker
 1517A Rolling Glen Rd.
 Boothwyn, PA 19061
                            
              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.
Orderthe reports and follow the program outlined above--SUCCESSwill
be yourreward.

                 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
localoffice of the Small Business Administration (a Federal Agency)
1-800-827-5722 for free help and answers to questions. Also, the
InternalRevenue Service offers free help via telephone and free
seminars aboutbusiness tax requirements. Your earnings are highly
dependant on youractivities 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 derr9@goplay.com
/////////////////////////////////////////////////////////////////











From rem-conf Mon Oct 02 16:50:39 2000 
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Date: Mon, 02 Oct 2000 18:49:07 -0500
From: Qiaobing Xie <xieqb@cig.mot.com>
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To: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
CC: Qiaobing Xie <Qiaobing_Xie-QXIE1@email.mot.com>, rem-conf@es.net
Subject: Re: Comments on draft-ietf-avt-rtp-amr-00.txt
References: <39D203BA.561299A1@era.ericsson.se> <200009271749.MAA02664@agevole.cig.mot.com> <39D88E7D.DFCD1BC7@era.ericsson.se>
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Precedence: list

Hi, Magnus,

See my comments below....

Magnus Westerlund wrote:
> 
> Hello,
> 
> The payload format is designed with IP networks in mind. The problem we see is
> that IP is transported over a wide range of different links with different
> behavior, e.g. different radio and wireline links. Radio and wire do not have
> the same error characteristics. AMR will be widly used over wireless links and
> on these links bandwidth efficiency is extremely important. This lead to the
> bitwise sorting. Octet sorting needs octet alignement for each block of the
> payload, i.e. padding at the end of each block. Which we considered to be to
> expensive. 

This further convinces me that the current design presented in the AMR
draft is a radio-link-centric design, which is wrong in my opinion,
because:

1) bandwidth efficiency over wireless links is more of a concern for 
ROHC working group. Architecturally, it should be delt with below IP by
header compression/stripping, as being worked upon in rohc. It doesn't
belong to RTP payload layer (unless it is something facilitating the
underlying header compression/stripping).

2) AMR itself is not specfic to radio links, it is a general purpose
codec. It is in the best interest of the AVT working group to define a
*general* RTP payload format for carrying AMR speech over any IP
connection, not something optimized for radio links only. This is why I
am against the bitwise sorting - it may be inevitable for a radio link
(even for radio links, it is expensive and often done at a lower layer
by DSP or with hardware assistance), but it is a magnitude more
expensive than byte-oriented handling to most applications running on a
general purpose machine.

> The optional sensitivity sorting facilitates the use of different
> error robustness schemes. The need for different robustness enhancing schemes
> may depend on the application and the lower transport layers. The application
> can be for example conversational speech or streaming. Lower layers are here
> layer 1 and 2.

Since the bitwise sorting is so expensive, I think it is more than
necessary that you justify it by showing exactly what "robustness
echancing schemes" you have in mind and exactly how they would work
together.

Here you also brought out another concern of mine: the current design
put no consideration on frame quality classification support (you would
need CRC and bad frame indicator to do so, as I pointed out in my
original email). Is this because you assume those functions are provided
by "the lower transport layers", such as by the Iu-UP protocols defined
in 3GPP? Unfortunately, in a general IP network, there is no such
support from the transport layer. In order to make the AMR over IP work
effectively over any IP connections (not only for a 3GPP Iu-UP
interface), you will need to support those functions in your RTP 
payload format.


regards,
-Qiaobing

> 
> Regards
> 
> Magnus W.
> 
> Qiaobing Xie wrote:
> 
> > 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
> 
> --
> 
> Magnus Westerlund
> 
> Audio Technology, Ericsson Research
> ----------------------------------------------------------------------
> Ericsson Radio Systems AB  | Phone +46 8 4048287
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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: draft text for MPEG-4 Audio/Visual RTP I-D revision
Date: Tue, 3 Oct 2000 10:36:25 +0900
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Dear experts,

Attached find please a draft text for the revision of MPEG-4 Audio/Visual
RTP Internet-Draft.

The revisions are mostly to reflect suggestions by the AVT chair. Most of
them are editorial and there are some minor revisions and clarifications.
The following is a summary of the changes:

(1) MIME subtype names were changed to "video/MP4V-ES" for video and
"audio/MP4A-LATM" for audio.
(2) The use of H.263 RTP payload formats (RFC 2190, 2429) in the short
video header mode was changed to "MAY" to "SHOULD".
(3) Clarification of timestamp definition
  There was a suggestion not to describe the detailed definition of the
video timestamp in the RTP spec. and change it as a reference to MPEG-4
visual document, so as to keep consistency with the MPEG-4 visual. A
description about the RTP timestamp in the RTCP SR packet was added.
(4) SA/TTSI over LATM
    A description was added to restrict the LATM usage so as not to transmit
 SA data by LATM.
(5) Other editorial and wording changes.

Please let us know if there is comment on the attached draft. The authors
need to submit it soon as an updated I-D and hopefully final before RFC.

Best regards,
Yoshihiro Kikuchi
and authors of the Internet 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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Internet Engineering Task Force                 Yoshihiro Kikuchi - =
Toshiba
Internet Draft                                       Toshiyuki Nomura - =
NEC
Document: draft-ietf-avt-rtp-mpeg4-es-05.txt         Shigeru Fukunaga - =
Oki
                                              Yoshinori Matsui - =
Matsushita
                                                       Hideaki Kimata - =
NTT
                                                         September 29, =
2000


             RTP payload format for MPEG-4 Audio/Visual streams


Status of this Memo

   This document is an Internet-Draft and is in full conformance with =
all
      provisions of Section 10 of RFC2026 [1].

   Internet-Drafts are working documents of the Internet Engineering =
Task
   Force (IETF), its areas, and its working groups. Note that other =
groups
   may also distribute working documents as Internet-Drafts. =
Internet-Drafts
   are draft documents valid for a maximum of six months and may be =
updated,
   replaced, or obsoleted by other documents at any time. It is
   inappropriate to use Internet- Drafts as reference material or to =
cite
   them other than as "work in progress."
   The list of current Internet-Drafts can be accessed at
   http://www.ietf.org/ietf/1id-abstracts.txt
   The list of Internet-Draft Shadow Directories can be accessed at
   http://www.ietf.org/shadow.html.






                                   Abstract

   This document describes RTP payload formats for carrying each of =
MPEG-4
   Audio and MPEG-4 Visual bitstreams without using MPEG-4 Systems. For =
the
   purpose of directly mapping MPEG-4 Audio/Visual bitstreams onto RTP
   packets, it provides specifications for the use of RTP header fields =
and
   also specifies fragmentation rules. It also provides specifications =
for
   MIME type registrations and the use of SDP.













Kikuchi/Nomura/Fukunaga/Kimata/Matsui                           [Page 1]
=0CRTP payload format for MPEG-4 Audio/Visual streams      September =
2000=0D


1. Introduction

   The RTP payload formats described in this document specify how MPEG-4
   Audio [3][5] and MPEG-4 Visual streams [2][4] are to be fragmented =
and
   mapped directly onto RTP packets.

   These RTP payload formats enable transport of MPEG-4 Audio/Visual =
streams
   without using the synchronization and stream management functionality =
of
   MPEG-4 Systems [6]. Such RTP payload formats will be used in systems =
that
   have intrinsic stream management functionality and thus require no =
such
   functionality from MPEG-4 Systems. H.323 terminals are an example of =
such
   a systems, where MPEG-4 Audio/Visual streams are not managed by =
MPEG-4
   Systems Object Descriptors but by H.245. The streams are directly =
mapped
   onto RTP packets without using MPEG-4 Systems Sync Layer. Other =
examples
   are SIP and RTSP where MIME and SDP are used. MIME types and SDP =
usages
   of the RTP payload formats described in this document are defined to
   directly specify the attribute of Audio/Visual streams (e.g. media =
type,
   packetization format and codec configuration) without using MPEG-4
   Systems. The obvious benefit is that these MPEG-4 Audio/Visual RTP
   payload formats can be handled in an unified way together with those
   formats defined for non-MPEG-4 codecs. The disadvantage is that
   interoperability with environments using MPEG-4 Systems may be =
difficult,
   other payload formats may be better suited to those applications.

   The semantics of RTP headers in such cases need to be clearly =
defined,
   including the association with MPEG-4 Audio/Visual data elements. In
   addition, it is beneficial to define the fragmentation rules of RTP
   packets for MPEG-4 Video streams so as to enhance error resiliency by
   utilizing the error resilience tools provided inside the MPEG-4 Video
   stream.


1.1 MPEG-4 Visual RTP payload format

   MPEG-4 Visual is a visual coding standard with many new features: =
high
   coding efficiency; high error resiliency; multiple, arbitrary shape
   object-based coding; etc. [2]. It covers a wide range of bitrates =
from
   scores of Kbps to several Mbps. It also covers a wide variety of
   networks, ranging from those guaranteed to be almost error-free to =
mobile
   networks with high error rates.

   With respect to the fragmentation rules for an MPEG-4 visual =
bitstream
   defined in this document, since MPEG-4 Visual is used for a wide =
variety
   of networks, it is desirable not to apply too much restriction on
   fragmentation, and a fragmentation rule such as "a single video =
packet
   shall always be mapped on a single RTP packet" may be inappropriate. =
On
   the other hand, careless, media unaware fragmentation may cause
   degradation in error resiliency and bandwidth efficiency. The
   fragmentation rules described in this document are flexible but =
manage to
   define the minimum rules for preventing meaningless fragmentation =
while
   utilizing the error resilience functionalities of MPEG-4 Visual.


Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 2]
=0CRTP payload format for MPEG-4 Audio/Visual streams      September =
2000=0D


   The fragmentation rule recommends not to map more than one VOP in an =
RTP
   packet so that the 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 an empty VOP (vop_coded=3D0) containing only VOP header or =
an
   arbitrary shaped VOP with a small number of coding blocks. To reduce =
the
   overhead for such cases, the fragmentation rule permits concatenating
   multiple VOPs in an RTP packet. (See fragmentation rule (4) in =
section
   3.2 and marker bit and timestamp in section 3.1.)

   While the additional media specific RTP header defined for such video
   coding tools as H.261 or MPEG-1/2 is effective in helping to recover
   picture headers corrupted by packet losses, MPEG-4 Visual has already
   error resilience functionalities for recovering corrupt headers, and
   these can be used on RTP/IP networks as well as on other networks
   (H.223/mobile, MPEG-2/TS, etc.). Therefore, no extra RTP header =
fields
   are defined in this MPEG-4 Visual RTP payload format.


1.2 MPEG-4 Audio RTP payload format

   MPEG-4 Audio is a new kind of audio standard that integrates many
   different types of audio coding tools. Low-overhead MPEG-4 Audio
   Transport Multiplex (LATM) manages the sequences of audio data with
   relatively small overhead. In audio-only applications, then, it is
   desirable for LATM-based MPEG-4 Audio bitstreams to be directly =
mapped
   onto the RTP packets without using MPEG-4 Systems.

   While LATM has several multiplexing features as follows;
   - Carrying configuration information with audio data,
   - Concatenation of multiple audio frames in one audio stream,
   - Multiplexing multiple objects (programs),
   - Multiplexing scalable layers,
   in RTP transmission there is no need for the last two features.
   Therefore, these two features MUST NOT be used in applications based =
on
   RTP packetization specified by this document. Since LATM has been
   developed for only natural audio coding tools, i.e. not for synthesis
   tools, it seems difficult to transmit Structured Audio (SA) data and =
Text
   to Speech Interface (TTSI) data by LATM. Therefore, SA data and TTSI =
data
   MUST NOT be transported by the RTP packetization in this document

   For transmission of scalable streams, audio data of each layer should =
be
   packetized onto different RTP packets allowing for the different =
layers
   to be treated differently at the IP level, for example via some means =
of
   differentiated service. On the other hand, all configuration data of =
the
   scalable streams are contained in one LATM configuration data
   "StreamMuxConfig" and every scalable layer shares the =
StreamMuxConfig.
   The mapping between each layer and its configuration data is achieved =
by
   LATM header information attached to the audio data. In order to =
indicate
   the dependency information of the scalable streams, a restriction is
   applied to the dynamic assignment rule of payload type (PT) values =
(see
   section 4.2).


Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 3]
=0CRTP payload format for MPEG-4 Audio/Visual streams      September =
2000=0D


   For MPEG-4 Audio coding tools, as is true for other audio coders, if =
the
   payload is a single audio frame, packet loss will not impair the
   decodability of adjacent packets. Therefore, the additional media
   specific header for recovering errors will not be required for MPEG-4
   Audio. Existing RTP protection mechanisms, such as Generic Forward =
Error
   Correction (RFC 2733) and Redundant Audio Data (RFC 2198), MAY be =
applied
   to improve error resiliency.




2. Conventions used in this document

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED",  "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC-2119 [7].




3. RTP Packetization of MPEG-4 Visual bitstream

   This section specifies RTP packetization rules for MPEG-4 Visual =
content.
   An MPEG-4 Visual bitstream is mapped directly onto RTP packets =
without
   the addition of extra header fields or any removal of Visual syntax
   elements. The Combined Configuration/Elementary stream mode MUST be =
used
   so that configuration information will be carried to the same RTP =
port as
   the elementary stream. (see 6.2.1 "Start codes" of ISO/IEC 14496-2
   [2][9][4]) The configuration information MAY additionally be =
specified by
   some out-of-band means. If needed for an H.323 terminal, H.245 =
codepoint
   "decoderConfigurationInformation" MUST be used for this purpose. If
   needed by systems using MIME content type and SDP parameters, e.g. =
SIP
   and RTSP, the optional parameter "config" MUST be used to specify the
   configuration information. (see 5.1 and 5.2)

   When the short video header mode is used, RTP payload format for =
H.263
   (e.g., RFC 2190 or RFC 2429) SHOULD be used.
















Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 4]
=0CRTP payload format for MPEG-4 Audio/Visual streams      September =
2000=0D


   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=3D2|P|X|  CC   |M|     PT      |       sequence number         | =
RTP
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                           timestamp                           | =
Header
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |           synchronization source (SSRC) identifier            |
   =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+
   |            contributing source (CSRC) identifiers             |
   |                             ....                              |
   =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+
   |                                                               | RTP
   |       MPEG-4 Visual stream (byte aligned)                     | =
Payload
   |                                                               |
   |                               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                               :...OPTIONAL RTP padding        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

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




3.1 Use of RTP header fields for MPEG-4 Visual

   Payload Type (PT): The assignment of an RTP payload type for this new
   packet format is outside the scope of this document, and will not be
   specified here. It is expected that the RTP profile for a particular
   class of applications will assign a payload type for this encoding, =
or if
   that is not done then a payload type in the dynamic range shall be =
chosen
   by means of an out of band signaling protocol (e.g. H.245, SIP, etc).


   Extension (X) bit: Defined by the RTP profile used.


   Sequence Number: Incremented by one for each RTP data packet sent,
   starting, for security reasons, with a random initial value.


   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: The timestamp indicates the composition time, or the
   presentation time in a no-compositor decoder. It is the same time
   instance as that conveyed in the timestamp field (modulo_time_base =
and
   vop_time_increment) of the video stream. (See ISO/IEC 14496-2[2], =
section
   6 for the detailed definition.) A constant offset, which is random, =
is


Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 5]
=0CRTP payload format for MPEG-4 Audio/Visual streams      September =
2000=0D


   added for security reasons. The detailed definition of the timestamp =
is
   as follows:
   - When multiple VOPs are carried in the same RTP packet, the =
timestamp
     indicates the earliest of the presentation times 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).
   - If the RTP packet contains only configuration information and/or
     Group_of_VideoObjectPlane() fields, the presentation time of the =
next
     VOP in the coding order is used.
   - If the RTP packet contains only visual_object_sequence_end_code
     information, the presentation time of the immediately preceding VOP =
in
     the coding order is used.

   The resolution of the timestamp is set to its default value of 90KHz,
   unless specified by an out-of-band means (e.g. SDP parameter or MIME
   parameter as defined in section 5).


   Other header fields are used as described in RFC 1889 [8].


   RTP timestamps in RTCP SR packets: According to the RTP timing model, =
the
   RTP timestamp that is carried into an RTCP SR packet is the same as =
the
   presentation time that would be applied to an RTP packet for data =
that
   was sampled at the instant the SR packet is being generated and sent. =
The
   RTP timestamp value is calculated from the NTP timestamp for the =
current
   time which also goes in the RTCP SR packet. To perform that =
calculation,
   an implementation needs to periodically establish a correspondence
   between the presentation time of a data packet and the NTP time at =
which
   that data was sampled.


3.2 Fragmentation of MPEG-4 Visual bitstream

   A fragmented MPEG-4 Visual bitstream is mapped directly onto the RTP
   payload without any addition of extra header fields or any removal of
   Visual syntax elements. The Combined Configuration/Elementary streams
   mode is used. The following rules apply for the fragmentation.

   In the following header means one of the following:
   - Configuration information (Visual Object Sequence Header, Visual =
Object
     Header and Video Object Layer Header)
   - visual_object_sequence_end_code
   - The header of the entry point function for an elementary stream
     (Group_of_VideoObjectPlane() or the header of VideoObjectPlane(),
     video_plane_with_short_header(), MeshObject() or FaceObject())
   - The video packet header (video_packet_header() excluding
     next_resync_marker())
   - The header of gob_layer()
   See 6.2.1 "Start codes" of ISO/IEC 14496-2[2][9][4] for the =
definition of
   the configuration information and the entry point functions.

Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 6]
=0CRTP payload format for MPEG-4 Audio/Visual streams      September =
2000=0D



   (1) Configuration information and Group_of_VideoObjectPlane() fields
   SHALL be placed at the beginning of the RTP payload (just after the =
RTP
   header) or just after the header of the syntactically upper layer
   function.

   (2) If one or more headers exist in the RTP payload, the RTP payload
   SHALL begin with the header of the syntactically highest function.
   Note: The visual_object_sequence_end_code is regarded as the lowest
   function.

   (3) A header SHALL NOT be split into a plurality of RTP packets.

   (4) Different 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 consecutive VOPs MAY =
be
   carried within one RTP packet in the decoding order if the size of =
the
   VOPs is small.
   Note: 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 in which the marker =
bit
   equals to one and the beginning of RTP payload corresponds to a start
   code. (See timestamp and marker bit in section 3.1)

   (5) It is RECOMMENDED that a single video packet is sent as a single =
RTP
   packet. The size of a video packet SHOULD be adjusted in such a way =
that
   the resulting RTP packet is not larger than the path-MTU.
   Note: Rule (5) does not apply when the video packet is disabled by =
the
   coder configuration (by setting resync_marker_disable in the VOL =
header
   to 1), or in coding tools where the video packet is not supported. In
   this case, a VOP MAY be split at arbitrary byte-positions.

   The video packet starts with the VOP header or the video packet =
header,
   followed by motion_shape_texture(), and ends with =
next_resync_marker() or
   next_start_code().


3.3 Examples of packetized MPEG-4 Visual bitstream

   Figure 2 shows examples of RTP packets generated based on the =
criteria
   described in 3.2

   (a) is an example of the first RTP packet or the random access point =
of
   an MPEG-4 visual bitstream containing the configuration information.
   According to criterion (1), the Visual Object Sequence Header(VS =
header)
   is placed at the beginning of the RTP payload, preceding the Visual
   Object Header and the Video Object Layer Header(VO header, VOL =
header).
   Since the fragmentation rule defined in 3.2 guarantees that the
   configuration information, starting with
   visual_object_sequence_start_code, is always placed at the beginning =
of
   the RTP payload, RTP receivers can detect the random access point by

Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 7]
=0CRTP payload format for MPEG-4 Audio/Visual streams      September =
2000=0D


   checking if the first 32-bit field of the RTP payload is
   visual_object_sequence_start_code.

   (b) is another example of the RTP packet containing the configuration
   information. It differs from example (a) in that the RTP packet also
   contains a video packet in the VOP following the configuration
   information. Since the length of the configuration information is
   relatively short (typically scores of bytes) and an RTP packet =
containing
   only the configuration information may thus increase the overhead, =
the
   configuration information and the immediately following GOV and/or (a
   part of) VOP can be packetized into a single RTP packet as in this
   example.

   (c) is an example of an RTP packet that contains
   Group_of_VideoObjectPlane(GOV). Following criterion (1), the GOV is
   placed at the beginning of the RTP payload. It would be a waste of =
RTP/IP
   header overhead to generate an RTP packet containing only a GOV whose
   length is 7 bytes. Therefore, (a part of) the following VOP can be =
placed
   in the same RTP packet as shown in (c).

   (d) is an example of the case where one video packet is packetized =
into
   one RTP packet. When the packet-loss rate of the underlying network =
is
   high, this kind of packetization is recommended. Even when the RTP =
packet
   containing the VOP header is discarded by a packet loss, the other =
RTP
   packets can be decoded by using the HEC(Header Extension Code)
   information in the video packet header. No extra RTP header field is
   necessary.

   (e) is an example of the case where more than one video packet is
   packetized into one RTP packet. This kind of packetization is =
effective
   to save the overhead of RTP/IP headers when the bit-rate of the
   underlying network is low. However, it will decrease the packet-loss
   resiliency because multiple video packets are discarded by a single =
RTP
   packet loss. The optimal number of video packets in an RTP packet and =
the
   length of the RTP packet can be determined considering the =
packet-loss
   rate and the bit-rate of the underlying network.

   (f) is an example of the case when the video packet is disabled by
   setting resync_marker_disable in the VOL header to 1. In this case, a =
VOP
   may be split into a plurality of RTP packets at arbitrary =
byte-positions.
   For example, it is possible to split a VOP into fixed-length packets.
   This kind of coder configuration and RTP packet fragmentation may be =
used
   when the underlying network is guaranteed to be error-free. On the =
other
   hand, it is not recommended to use it in error-prone environment =
since it
   provides only poor packet loss resiliency.


   Figure 3 shows examples of RTP packets prohibited by the criteria of =
3.2.

   Fragmentation of a header into multiple RTP packets, as in (a), will =
not
   only increase the overhead of RTP/IP headers but also decrease the =
error
   resiliency. Therefore, it is prohibited by the criterion (3).

Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 8]
=0CRTP payload format for MPEG-4 Audio/Visual streams      September =
2000=0D



   When concatenating more than one video packets into an RTP packet, =
VOP
   header or video_packet_header() shall not be placed in the middle of =
the
   RTP payload. The packetization as in (b) is not allowed by criterion =
(2)
   due to the aspect of the error resiliency. Comparing this example =
with
   Figure 2(d), although two video packets are mapped onto two RTP =
packets
   in both cases, the packet-loss resiliency is not identical. Namely, =
if
   the second RTP packet is lost, both video packets 1 and 2 are lost in =
the
   case of Figure 3(b) whereas only video packet 2 is lost in the case =
of
   Figure 2(d).











































Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 9]
=0CRTP payload format for MPEG-4 Audio/Visual streams      September =
2000=0D


       +------+------+------+------+
   (a) | RTP  |  VS  |  VO  | VOL  |
       |header|header|header|header|
       +------+------+------+------+

       +------+------+------+------+------------+
   (b) | RTP  |  VS  |  VO  | VOL  |Video Packet|
       |header|header|header|header|            |
       +------+------+------+------+------------+

       +------+-----+------------------+
   (c) | RTP  | GOV |Video Object Plane|
       |header|     |                  |
       +------+-----+------------------+

       +------+------+------------+  +------+------+------------+
   (d) | RTP  | VOP  |Video Packet|  | RTP  |  VP  |Video Packet|
       |header|header|    (1)     |  |header|header|    (2)     |
       +------+------+------------+  +------+------+------------+

       =
+------+------+------------+------+------------+------+------------+
   (e) | RTP  |  VP  |Video Packet|  VP  |Video Packet|  VP  |Video =
Packet|
       |header|header|     (1)    |header|    (2)     |header|    (3)    =
 |
       =
+------+------+------------+------+------------+------+------------+

       +------+------+------------+  +------+------------+
   (f) | RTP  | VOP  |VOP fragment|  | RTP  |VOP fragment|
       |header|header|    (1)     |  |header|    (2)     | ___
       +------+------+------------+  +------+------------+

        Figure 2 - Examples of RTP packetized MPEG-4 Visual bitstream


       +------+-------------+  +------+------------+------------+
   (a) | RTP  |First half of|  | RTP  |Last half of|Video Packet|
       |header|  VP header  |  |header|  VP header |            |
       +------+-------------+  +------+------------+------------+

       +------+------+----------+  =
+------+---------+------+------------+
   (b) | RTP  | VOP  |First half|  | RTP  |Last half|  VP  |Video =
Packet|
       |header|header| of VP(1) |  |header| of VP(1)|header|    (2)     =
|
       +------+------+----------+  =
+------+---------+------+------------+

   Figure 3 - Examples of prohibited RTP packetization for MPEG-4 Visual
   bitstream








Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 10]
=0CRTP payload format for MPEG-4 Audio/Visual streams      September =
2000=0D


4. RTP Packetization of MPEG-4 Audio bitstream

   This section specifies RTP packetization rules for MPEG-4 Audio
   bitstreams. MPEG-4 Audio streams MUST be formatted by LATM =
(Low-overhead
   MPEG-4 Audio Transport Multiplex) tool[5], and the LATM-based streams =
are
   then mapped onto RTP packets as described the three sections below.

4.1 RTP Packet Format

   LATM-based streams consist of a sequence of audioMuxElements that =
include
   one or more audio frames. A complete audioMuxElement or a part of one
   SHALL be mapped directly onto an RTP payload without any removal of
   audioMuxElement syntax elements (see Figure 4). The first byte of =
each
   audioMuxElement SHALL be located at the first payload location in an =
RTP
   packet.


   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=3D2|P|X|  CC   |M|     PT      |       sequence number         =
|RTP
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                           timestamp                           =
|Header
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |           synchronization source (SSRC) identifier            |
   =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+
   |            contributing source (CSRC) identifiers             |
   |                             ....                              |
   =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+
   |                                                               |RTP
   :                 audioMuxElement (byte aligned)                =
:Payload
   |                                                               |
   |                               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                               :...OPTIONAL RTP padding        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
                Figure 4 - An RTP packet for MPEG-4 Audio

   In order to decode the audioMuxElement, the following =
muxConfigPresent
   information is required to be indicated by an out-of-band means. When =
SDP
   is utilized for this indication, MIME parameter "cpresent" =
corresponds to
   the muxConfigPresent information (see section 5.3).

   muxConfigPresent: If this value is set to 1 (in-band mode), the
   audioMuxElement SHALL include an indication bit "useSameStreamMux" =
and
   MAY include the configuration information for audio compression
   "StreamMuxConfig". The useSameStreamMux bit indicates whether the
   StreamMuxConfig element in the previous frame is applied in the =
current
   frame. If the useSameStreamMux bit indicates to use the =
StreamMuxConfig
   from the previous frame, but if the previous frame has been lost, the
   current frame may not be decodable. Therefore, in case of in-band =
mode,
   the StreamMuxConfig element SHOULD be transmitted repeatedly =
depending on
   the network condition. On the other hand, if muxConfigPresent is set =
to 0

Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 11]
=0CRTP payload format for MPEG-4 Audio/Visual streams      September =
2000=0D


   (out-band mode), the StreamMuxConfig element is required to be
   transmitted by an out-of-band means. In case of SDP, MIME parameter
   "config" is utilized (see section 5.3).

4.2 Use of RTP Header Fields for MPEG-4 Audio

   Payload Type (PT): The assignment of an RTP payload type for this new
   packet format is outside the scope of this document, and will not be
   specified here. It is expected that the RTP profile for a particular
   class of applications will assign a payload type for this encoding, =
or if
   that is not done then a payload type in the dynamic range shall be =
chosen
   by means of an out of band signaling protocol (e.g. H.245, SIP, etc). =
In
   the dynamic assignment of RTP payload types for scalable streams, a
   different value should be assigned to each layer. The assigned values
   should be in order of enhance layer dependency, where the base layer =
has
   the smallest value.

   Marker (M) bit: The marker bit indicates audioMuxElement boundaries. =
It
   is set to one to indicate that the RTP packet contains a complete
   audioMuxElement or the last fragment of an audioMuxElement.

   Timestamp: The timestamp indicates the sampling instance of the first
   audio frame contained in the RTP packet. Timestamps are recommended =
to
   start at a random value for security reasons.

   Unless specified by an out-of-band means, the resolution of the =
timestamp
   is set to its default value of 90 kHz.

   Sequence Number: Incremented by one for each RTP packet sent, =
starting,
   for security reasons, with a random value.

   Other header fields are used as described in RFC 1889 [8].


4.3 Fragmentation of MPEG-4 Audio bitstream

   It is RECOMMENDED to put one audioMuxElement in each RTP packet. If =
the
   size of an audioMuxElement can be kept small enough that the size of =
the
   RTP packet containing it does not exceed the size of the path-MTU, =
this
   will be no problem. If it cannot, the audioMuxElement MAY be =
fragmented
   and spread across multiple packets.




5. MIME type registration for MPEG-4 Audio/Visual streams

   The following sections describe the MIME type registrations for =
MPEG-4
   Audio/Visual streams. MIME type registration and SDP usage for the =
MPEG-4
   Visual stream are described in Sections 5.1 and 5.2, respectively, =
while
   MIME type registration and SDP usage for MPEG-4 Audio stream are
   described in Sections 5.3 and 5.4, respectively.

Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 12]
=0CRTP payload format for MPEG-4 Audio/Visual streams      September =
2000=0D



   (In the following sections, the RFC number "XXXX" represents the RFC
   number, which should be assigned for this document.)


5.1 MIME type registration for MPEG-4 Visual

   MIME media type name: video

   MIME subtype name: MP4V-ES

   Required parameters: none

   Optional parameters:
     rate: This parameter is used only for RTP transport. It indicates =
the
     resolution of the timestamp field in the RTP header. If this =
parameter
     is not specified, its default value of 90000 (90KHz) is used.

     profile-level-id: A decimal representation of MPEG-4 Visual Profile
     Level indication value (profile_and_level_indication) defined in =
Table
     G-1 of ISO/IEC 14496-2 [2][4]. This parameter MAY be used in the
     capability exchange or session setup procedure to indicate MPEG-4
     Visual Profile and Level combination of which the MPEG-4 Visual =
codec
     is capable. If this parameter is not specified by the procedure, =
its
     default value of 1 (Simple Profile/Level 1) is used.

     config: This parameter indicates the configuration of the
     corresponding MPEG-4 visual bitstream. It SHALL NOT be used to
     indicate the codec capability in the capability exchange procedure. =
It
     is a hexadecimal representation of an octet string that expresses =
the
     MPEG-4 Visual configuration information, as defined in subclause =
6.2.1
     Start codes of ISO/IEC14496-2[2][4][9]. The configuration =
information
     is mapped onto the octet string in an MSB-first basis. The first =
bit
     of the configuration information SHALL be located at the MSB of the
     first octet. The configuration information indicated by this =
parameter
     SHALL be the same as the configuration information in the
     corresponding MPEG-4 Visual stream, except for
     first_half_vbv_occupancy and latter_half_vbv_occupancy, if exist,
     which may vary in the repeated configuration information inside an
     MPEG-4 Visual stream (See 6.2.1 Start codes of ISO/IEC14496-2).


     Example usages for these parameters are:
       - MPEG-4 Visual Simple Profile/Level 1:
          Content-type: video/mp4v-es; profile-level-id=3D1

       - MPEG-4 Visual Core Profile/Level 2:
          Content-type: video/mp4v-es; profile-level-id=3D34

       - MPEG-4 Visual Advanced Real Time Simple Profile/Level 1:
          Content-type: video/mp4v-es; profile-level-id=3D145


Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 13]
=0CRTP payload format for MPEG-4 Audio/Visual streams      September =
2000=0D



   Published specification:
     The specifications for MPEG-4 Visual streams are presented in =
ISO/IEC
     14469-2[2][4][9]. The RTP payload format is described in RFCXXXX.

   Encoding considerations:
     Video bitstreams must be generated according to MPEG-4 Visual
     specifications (ISO/IEC 14496-2). A video bitstream is binary data =
and
     must be encoded for non-binary transport (for Email, the Base64
     encoding is sufficient).  This type is also defined for transfer =
via
     RTP. The RTP packets MUST be packetized according to the MPEG-4 =
Visual
     RTP payload format defined in RFCXXXX.

   Security considerations:
     See section 6 of RFCXXXX.

   Interoperability considerations:
     MPEG-4 Visual provides a large and rich set of tools for the coding =
of
     visual objects. For effective implementation of the standard, =
subsets
     of the MPEG-4 Visual tool sets have been provided for use in =
specific
     applications. These subsets, called 'Profiles', limit the size of =
the
     tool set a decoder is required to implement. In order to restrict
     computational complexity, one or more Levels are set for each =
Profile.
     A Profile@Level combination allows:

     o a codec builder to implement only the subset of the standard he
     needs, while maintaining interworking with other MPEG-4 devices
     included in the same combination, and

     o checking whether MPEG-4 devices comply with the standard
     ('conformance testing').

     The visual stream SHALL be compliant with the MPEG-4 Visual
     Profile@Level specified by the parameter "profile-level-id".
     Interoperability between a sender and a receiver may be achieved by
     specifying the parameter "profile-level-id" in MIME content, or by
     arranging in the capability exchange/announcement procedure to set =
this
     parameter mutually to the same value.


   Applications which use this media type:
     Audio and visual streaming and conferencing tools, Internet =
messaging
     and Email applications.

   Additional information: none

   Person & email address to contact for further information:
     The authors of RFCXXXX. (See section 8)

   Intended usage: COMMON

   Author/Change controller:

Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 14]
=0CRTP payload format for MPEG-4 Audio/Visual streams      September =
2000=0D


     The authors of RFCXXXX. (See section 8)


5.2 SDP usage of MPEG-4 Visual

   The MIME media type video/MP4V-ES string is mapped to fields in the
   Session Description Protocol (SDP), RFC 2327, as follows:

   o The MIME type (video) goes in SDP "m=3D" as the media name.

   o The MIME subtype (MP4V-ES) goes in SDP "a=3Drtpmap" as the encoding =
name.

   o The optional parameter "rate" goes in "a=3Drtpmap" as the clock =
rate.

   o The optional parameter "profile-level-id" and "config" go in the
   "a=3Dfmtp" line to indicate the coder capability and configuration,
   respectively. These parameters are expressed as a MIME media type =
string,
   in the form of as a semicolon separated list of parameter=3Dvalue =
pairs.

   The following are some examples of media representation in SDP:

   Simple Profile/Level 1, rate=3D90000(90KHz), "profile-level-id" and
   "config" are present in "a=3Dfmtp" line:
     m=3Dvideo 49170/2 RTP/AVP 98
     a=3Drtpmap:98 MP4V-ES/90000
     a=3Dfmtp:98 profile-level-
     =
id=3D1;config=3D000001B001000001B5090000010000000120008440FA282C2090A21F

   Core Profile/Level 2, rate=3D90000(90KHz), "profile-level-id" is =
present in
   "a=3Dfmtp" line:
     m=3Dvideo 49170/2 RTP/AVP 98
     a=3Drtpmap:98 MP4V-ES/90000
     a=3Dfmtp:98 profile-level-id=3D34

   Advance Real Time Simple Profile/Level 1, rate=3D25(25Hz), =
"profile-level-
   id" is present in "a=3Dfmtp" line:
     m=3Dvideo 49170/2 RTP/AVP 98
     a=3Drtpmap:98 MP4V-ES/25
     a=3Dfmtp:98 profile-level-id=3D145




5.3 MIME type registration of MPEG-4 Audio

   MIME media type name: audio

   MIME subtype name: MP4A-LATM

   Required parameters:
     rate: the rate parameter indicates the RTP time stamp clock rate. =
The
     default value is 90000. Other rates MAY be specified only if they =
are

Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 15]
=0CRTP payload format for MPEG-4 Audio/Visual streams      September =
2000=0D


     set to the same value as the audio sampling rate (number of samples
     per second).

   Optional parameters:
     profile-level-id: a decimal representation of MPEG-4 Audio Profile
     Level indication value defined in ISO/IEC 14496-1 [10]. This =
parameter
     indicates which MPEG-4 Audio tool subsets the decoder is capable of
     using. If this parameter is not specified in the capability =
exchange
     or session setup procedure, its default value of 30 (Natural Audio
     Profile/Level 1) is used.

     object: a decimal representation of the MPEG-4 Audio Object Type =
value
     defined in ISO/IEC 14496-3 [5]. This parameter specifies the tool =
to
     be used by the coder. It CAN be used to limit the capability within
     the specified "profile-level-id".

     bitrate: the data rate for the audio bit stream.

     cpresent: this parameter indicates whether audio payload =
configuration
     data has been multiplexed into an RTP payload (see section 4.1). If
     not specified, the default value is 1.

     config: a hexadecimal representation of an octet string that =
expresses
     the audio payload configuration data "StreamMuxConfig", as defined =
in
     ISO/IEC 14496-3 [5] (see section 4.1). Configuration data is mapped
     onto the octet string in an MSB-first basis. The first bit of the
     configuration data SHALL be located at the MSB of the first octet. =
In
     the last octet, zero-padding bits, if necessary, shall follow the
     configuration data.

     ptime: RECOMMENDED duration of each packet in milliseconds.

   Published specification:
     Payload format specifications are described in this document. =
Encoding
     specifications are provided in ISO/IEC 14496-3 [3][5].

   Encoding considerations:
     This type is only defined for transfer via RTP.

   Security considerations:
     See Section 6 of RFCXXXX.

   Interoperability considerations:
     MPEG-4 Audio provides a large and rich set of tools for the coding =
of
     audio objects. For effective implementation of the standard, =
subsets of
     the MPEG-4 Audio tool sets similar to those used in MPEG-4 Visual =
have
     been provided (see section 5.1).

     The audio stream SHALL be compliant with the MPEG-4 Audio
     Profile@Level specified by the parameter "profile-level-id".
     Interoperability between a sender and a receiver may be achieved by
     specifying the parameter "profile-level-id" in MIME content, or by

Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 16]
=0CRTP payload format for MPEG-4 Audio/Visual streams      September =
2000=0D


     arranging in the capability exchange procedure to set this =
parameter
     mutually to the same value. Furthermore, the "object" parameter can =
be
     used to limit the capability within the specified Profile@Level in
     capability exchange.


   Applications which use this media type:
     Audio and video streaming and conferencing tools.

   Additional information: none

   Personal & email address to contact for further information:
     See Section 8 of RFCXXXX.

   Intended usage: COMMON

   Author/Change controller:
     See Section 8 of RFCXXXX.


5.4 SDP usage of MPEG-4 Audio

   The MIME media type audio/MP4A-LATM string is mapped to fields in the
   Session Description Protocol (SDP), RFC 2327, as follows:

   o The MIME type (audio) goes in SDP "m=3D" as the media name.

   o The MIME subtype (MP4A-LATM) goes in SDP "a=3Drtpmap" as the =
encoding
   name.

   o The required parameter "rate" goes in "a=3Drtpmap" as the clock =
rate.

   o The optional parameter "ptime" goes in SDP "a=3Dptime" attribute.

   o The optional parameter "profile-level-id" goes in the "a=3Dfmtp" =
line to
   indicate the coder capability. The "object" parameter goes in the
   "a=3Dfmtp" attribute. The payload-format-specific parameters =
"bitrate",
   "cpresent" and "config" go in the "a=3Dfmtp" line. These parameters =
are
   expressed as a MIME media type string, in the form of as a semicolon
   separated list of parameter=3Dvalue pairs.

   The following are some examples of the media representation in SDP:

   For 6 kb/s CELP bitstreams (with an audio sampling rate of 8 kHz),
     m=3Daudio 49230 RTP/AVP 96
     a=3Drtpmap:96 MP4A-LATM/8000
     a=3Dfmtp:96 =
profile-level-id=3D9;object=3D8;cpresent=3D0;config=3D9128B1071070
     a=3Dptime:20

   For 64 kb/s AAC LC stereo bitstreams (with an audio sampling rate of =
24
   kHz),


Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 17]
=0CRTP payload format for MPEG-4 Audio/Visual streams      September =
2000=0D


     m=3Daudio 49230 RTP/AVP 96
     a=3Drtpmap:96 MP4A-LATM/24000
     a=3Dfmtp:96 profile-level-id=3D1; bitrate=3D64000; cpresent=3D0;
     config=3D9122620000

   In the above two examples, audio configuration data is not =
multiplexed
   into the RTP payload and is described only in SDP. Furthermore, the
   "clock rate" is set to the audio sampling rate.

   If the clock rate has been set to its default value and it is =
necessary
   to obtain the audio sampling rate, this can be done by parsing the
   "config" parameter (see the following example).

     m=3Daudio 49230 RTP/AVP 96
     a=3Drtpmap:96 MP4A-LATM/90000
     a=3Dfmtp:96 object=3D8; cpresent=3D0; config=3D9128B1071070

   The following example shows that the audio configuration data appears =
in
   the RTP payload.

   m=3Daudio 49230 RTP/AVP 96
   a=3Drtpmap:96 MP4A-LATM/90000
   a=3Dfmtp:96 object=3D2; cpresent=3D1



6. Security Considerations

   RTP packets using the payload format defined in this specification =
are
   subject to the security considerations discussed in the RTP =
specification
   [8]. This implies that confidentiality of the media streams is =
achieved
   by encryption. Because the data compression used with this payload =
format
   is applied end-to-end, encryption may be performed on the compressed =
data
   so there is no conflict between the two operations.

   The complete MPEG-4 system allows for transport of a wide range of
   content, including Java applets (MPEG-J) and scripts.  Since this =
payload
   format is restricted to audio and video streams, it is not possible =
to
   transport such active content in this format.


7. References


   1  Bradner, S., "The Internet Standards Process -- Revision 3", BCP =
9,
      RFC 2026, October 1996.

   2 ISO/IEC 14496-2:1999, "Information technology - Coding of =
audio-visual
      objects - Part2: Visual", December 1999.




Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 18]
=0CRTP payload format for MPEG-4 Audio/Visual streams      September =
2000=0D



   3 ISO/IEC 14496-3:1999, "Information technology - Coding of =
audio-visual
      objects - Part3: Audio", December 1999.

   4 ISO/IEC 14496-2:1999/FDAM1:2000, December 1999.

   5 ISO/IEC 14496-3:1999/FDAM1:2000, December 1999.

   6 ISO/IEC 14496-1:1999, "Information technology - Coding of =
audio-visual
      objects - Part1: Systems", December 1999.

   7  Bradner, S., "Key words for use in RFCs to Indicate Requirement
      Levels", BCP 14, RFC 2119, March 1997

   8 H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson "RTP: A =
Transport
      Protocol for Real Time Applications",  RFC 1889, Internet =
Engineering
      Task Force, January 1996.

   9  ISO/IEC 14496-2:1999/COR1:2000, "Information technology - Coding =
of
      audio-visual objects - Part2: Visual, Technical corrigendum 1", =
August
      2000.

   10 ISO/IEC 14496-1:1999/FDAM1:2000, December 1999.

8. Author's Addresses


   Yoshihiro Kikuchi
   Toshiba corporation
   1, Komukai Toshiba-cho, Saiwai-ku, Kawasaki, 212-8582, Japan
   Email: yoshihiro.kikuchi@toshiba.co.jp

   Yoshinori Matsui
   Matsushita Electric Industrial Co., LTD.
   1006, Kadoma, Kadoma-shi, Osaka, Japan
   Email: matsui@drl.mei.co.jp

   Toshiyuki Nomura
   NEC Corporation
   4-1-1,Miyazaki,Miyamae-ku,Kawasaki,JAPAN
   Email: t-nomura@ccm.cl.nec.co.jp

   Shigeru Fukunaga
   Oki Electric Industry Co., Ltd.
   1-2-27 Shiromi, Chuo-ku, Osaka 540-6025 Japan.
   Email: fukunaga444@oki.co.jp

   Hideaki Kimata
   Nippon Telegraph and Telephone Corporation
   1-1, Hikari-no-oka, Yokosuka-shi, Kanagawa, Japan
   Email: kimata@nttvdt.hil.ntt.co.jp


Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 19]
=0CRTP payload format for MPEG-4 Audio/Visual streams      September =
2000=0D



Full Copyright Statement

   "Copyright (C) The Internet Society (date). All Rights Reserved.

   This document and translations of it may be copied and furnished to
   others, and derivative works that comment on or otherwise explain it
   or assist in its implementation may be prepared, copied, published
   and distributed, in whole or in part, without restriction of any
   kind, provided that the above copyright notice and this paragraph
   are included on all such copies and derivative works. However, this
   document itself may not be modified in any way, such as by removing
   the copyright notice or references to the Internet Society or other
   Internet organizations, except as needed for the purpose of
   developing Internet standards in which case the procedures for
   copyrights defined in the Internet Standards process must be
   followed, or as required to translate it into languages other than
   English.

   The limited permissions granted above are perpetual and will not be
   revoked by the Internet Society or its successors or assigns.
=0C
------=_NextPart_000_0243_01C02D25.C7F165E0--




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To: IETF AVT <rem-conf@es.net>,
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 'Yoshihiro Kikuchi' <kiku@eel.rdc.toshiba.co.jp>
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Change (2) means that significantly different packets are generated in case
of Short Video Headers (H263) between the original proposal and this new
version. Am I right, or is this just an editorial matter ?

Best regards
Guido

> ----------
> From: 	Yoshihiro Kikuchi[SMTP:kiku@eel.rdc.toshiba.co.jp]
> Sent: 	Tuesday, October 03, 2000 3:36 AM
> To: 	IETF AVT; 4on2andIP-sys
> Cc: 	Yoshihiro Kikuchi
> Subject: 	draft text for MPEG-4 Audio/Visual RTP I-D revision
> 
> <<File: draft-ietf-avt-rtp-mpeg4-es-05-Sep29.txt>>
> Dear experts,
> 
> Attached find please a draft text for the revision of MPEG-4 Audio/Visual
> RTP Internet-Draft.
> 
> The revisions are mostly to reflect suggestions by the AVT chair. Most of
> them are editorial and there are some minor revisions and clarifications.
> The following is a summary of the changes:
> 
> (1) MIME subtype names were changed to "video/MP4V-ES" for video and
> "audio/MP4A-LATM" for audio.
> (2) The use of H.263 RTP payload formats (RFC 2190, 2429) in the short
> video header mode was changed to "MAY" to "SHOULD".
> (3) Clarification of timestamp definition
>   There was a suggestion not to describe the detailed definition of the
> video timestamp in the RTP spec. and change it as a reference to MPEG-4
> visual document, so as to keep consistency with the MPEG-4 visual. A
> description about the RTP timestamp in the RTCP SR packet was added.
> (4) SA/TTSI over LATM
>     A description was added to restrict the LATM usage so as not to
> transmit
>  SA data by LATM.
> (5) Other editorial and wording changes.
> 
> Please let us know if there is comment on the attached draft. The authors
> need to submit it soon as an updated I-D and hopefully final before RFC.
> 
> Best regards,
> Yoshihiro Kikuchi
> and authors of the Internet 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 Tue Oct 03 02:04:22 2000 
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From: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
To: "Franceschini Guido" <Guido.Franceschini@cselt.it>,
        "IETF AVT" <rem-conf@es.net>,
        "4on2andIP-sys" <4on2andIP-sys@advent.ee.columbia.edu>
Cc: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
References: <85077463E51D6A498C453073E167794039BD69@exc2k02.cselt.it>
Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision
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Dear Guido,

Excerpt from the new text:

>   When the short video header mode is used, RTP payload format for H.263
<   (e.g., RFC 2190 or RFC 2429) SHOULD be used.

This means that it is RECOMMENDED to use the H.263 RTP payload format for
the short video header which is compatible with H.263 baseline. Regardless
of MAY or SHOULD, there is a possibility that the H.263 RTP payload format
may be used, although the possibility may be higher for SHOULD than MAY. If
you really don't want to use the H.263 RTP payload format in some
circumstances, you don't have to use it. The payload format is signaled
through the MIME type.

(IETF experts, please correct me if my understanding on the definition of
SHOULD is wrong.)

Best regards,
Yoshihiro

----- Original Message -----
From: Franceschini Guido <Guido.Franceschini@cselt.it>
To: IETF AVT <rem-conf@es.net>; 4on2andIP-sys
<4on2andIP-sys@advent.ee.columbia.edu>; 'Yoshihiro Kikuchi'
<kiku@eel.rdc.toshiba.co.jp>
Sent: Tuesday, October 03, 2000 3:20 PM
Subject: RE: draft text for MPEG-4 Audio/Visual RTP I-D revision


> Change (2) means that significantly different packets are generated in
case
> of Short Video Headers (H263) between the original proposal and this new
> version. Am I right, or is this just an editorial matter ?
>
> Best regards
> Guido
>
> > ----------
> > From: Yoshihiro Kikuchi[SMTP:kiku@eel.rdc.toshiba.co.jp]
> > Sent: Tuesday, October 03, 2000 3:36 AM
> > To: IETF AVT; 4on2andIP-sys
> > Cc: Yoshihiro Kikuchi
> > Subject: draft text for MPEG-4 Audio/Visual RTP I-D revision
> >
> > <<File: draft-ietf-avt-rtp-mpeg4-es-05-Sep29.txt>>
> > Dear experts,
> >
> > Attached find please a draft text for the revision of MPEG-4
Audio/Visual
> > RTP Internet-Draft.
> >
> > The revisions are mostly to reflect suggestions by the AVT chair. Most
of
> > them are editorial and there are some minor revisions and
clarifications.
> > The following is a summary of the changes:
> >
> > (1) MIME subtype names were changed to "video/MP4V-ES" for video and
> > "audio/MP4A-LATM" for audio.
> > (2) The use of H.263 RTP payload formats (RFC 2190, 2429) in the short
> > video header mode was changed to "MAY" to "SHOULD".
> > (3) Clarification of timestamp definition
> >   There was a suggestion not to describe the detailed definition of the
> > video timestamp in the RTP spec. and change it as a reference to MPEG-4
> > visual document, so as to keep consistency with the MPEG-4 visual. A
> > description about the RTP timestamp in the RTCP SR packet was added.
> > (4) SA/TTSI over LATM
> >     A description was added to restrict the LATM usage so as not to
> > transmit
> >  SA data by LATM.
> > (5) Other editorial and wording changes.
> >
> > Please let us know if there is comment on the attached draft. The
authors
> > need to submit it soon as an updated I-D and hopefully final before RFC.
> >
> > Best regards,
> > Yoshihiro Kikuchi
> > and authors of the Internet 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 Tue Oct 03 09:30:27 2000 
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Date: Tue, 03 Oct 2000 18:21:15 +0200
From: Franceschini Guido <Guido.Franceschini@CSELT.IT>
Subject: RE: draft text for MPEG-4 Audio/Visual RTP I-D revision
To: Franceschini Guido <guido.franceschini@cselt.it>,
 IETF AVT <rem-conf@es.net>,
 4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>,
 'Yoshihiro Kikuchi' <kiku@eel.rdc.toshiba.co.jp>
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Does it mean that a stream with short video header (H263) SHOULD use the
traditional Payload Format for H263, but could as well (not recommended) use
the new Payload Format for MPEG-4 Video, and that the choosen PF is notified
as usual (SDP etc) ?
If so, OK

Guido

> ----------
> From: 	Yoshihiro Kikuchi[SMTP:kiku@eel.rdc.toshiba.co.jp]
> Sent: 	Tuesday, October 03, 2000 11:00 AM
> To: 	Franceschini Guido; IETF AVT; 4on2andIP-sys
> Cc: 	Yoshihiro Kikuchi
> Subject: 	Re: draft text for MPEG-4 Audio/Visual RTP I-D revision
> 
> Dear Guido,
> 
> Excerpt from the new text:
> 
> >   When the short video header mode is used, RTP payload format for H.263
> <   (e.g., RFC 2190 or RFC 2429) SHOULD be used.
> 
> This means that it is RECOMMENDED to use the H.263 RTP payload format for
> the short video header which is compatible with H.263 baseline. Regardless
> of MAY or SHOULD, there is a possibility that the H.263 RTP payload format
> may be used, although the possibility may be higher for SHOULD than MAY.
> If
> you really don't want to use the H.263 RTP payload format in some
> circumstances, you don't have to use it. The payload format is signaled
> through the MIME type.
> 
> (IETF experts, please correct me if my understanding on the definition of
> SHOULD is wrong.)
> 
> Best regards,
> Yoshihiro
> 
> ----- Original Message -----
> From: Franceschini Guido <Guido.Franceschini@cselt.it>
> To: IETF AVT <rem-conf@es.net>; 4on2andIP-sys
> <4on2andIP-sys@advent.ee.columbia.edu>; 'Yoshihiro Kikuchi'
> <kiku@eel.rdc.toshiba.co.jp>
> Sent: Tuesday, October 03, 2000 3:20 PM
> Subject: RE: draft text for MPEG-4 Audio/Visual RTP I-D revision
> 
> 
> > Change (2) means that significantly different packets are generated in
> case
> > of Short Video Headers (H263) between the original proposal and this new
> > version. Am I right, or is this just an editorial matter ?
> >
> > Best regards
> > Guido
> >
> > > ----------
> > > From: Yoshihiro Kikuchi[SMTP:kiku@eel.rdc.toshiba.co.jp]
> > > Sent: Tuesday, October 03, 2000 3:36 AM
> > > To: IETF AVT; 4on2andIP-sys
> > > Cc: Yoshihiro Kikuchi
> > > Subject: draft text for MPEG-4 Audio/Visual RTP I-D revision
> > >
> > > <<File: draft-ietf-avt-rtp-mpeg4-es-05-Sep29.txt>>
> > > Dear experts,
> > >
> > > Attached find please a draft text for the revision of MPEG-4
> Audio/Visual
> > > RTP Internet-Draft.
> > >
> > > The revisions are mostly to reflect suggestions by the AVT chair. Most
> of
> > > them are editorial and there are some minor revisions and
> clarifications.
> > > The following is a summary of the changes:
> > >
> > > (1) MIME subtype names were changed to "video/MP4V-ES" for video and
> > > "audio/MP4A-LATM" for audio.
> > > (2) The use of H.263 RTP payload formats (RFC 2190, 2429) in the short
> > > video header mode was changed to "MAY" to "SHOULD".
> > > (3) Clarification of timestamp definition
> > >   There was a suggestion not to describe the detailed definition of
> the
> > > video timestamp in the RTP spec. and change it as a reference to
> MPEG-4
> > > visual document, so as to keep consistency with the MPEG-4 visual. A
> > > description about the RTP timestamp in the RTCP SR packet was added.
> > > (4) SA/TTSI over LATM
> > >     A description was added to restrict the LATM usage so as not to
> > > transmit
> > >  SA data by LATM.
> > > (5) Other editorial and wording changes.
> > >
> > > Please let us know if there is comment on the attached draft. The
> authors
> > > need to submit it soon as an updated I-D and hopefully final before
> RFC.
> > >
> > > Best regards,
> > > Yoshihiro Kikuchi
> > > and authors of the Internet 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 Tue Oct 03 09:41:31 2000 
From rem-conf-request@es.net Tue Oct 03 09:41:30 2000
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From: Florissa Colina <florissa@bmrc.berkeley.edu>
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:

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 Tue Oct 03 11:39:16 2000 
From rem-conf-request@es.net Tue Oct 03 11:39:16 2000
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Reply-To: "Olivier Avaro" <olivier.avaro@francetelecom.fr>
From: "Olivier Avaro" <olivier.avaro@francetelecom.fr>
To: <jan.vandermeer@philips.com>,
	"rem-conf" <rem-conf@es.net>,
	"4on2andIP-sys" <4on2andIP-sys@advent.ee.columbia.edu>
Cc: "CURET Dominique FTRD/DMI" <dominique.curet@francetelecom.fr>
References: <0056890014249640000002L902*@MHS>
Subject: Re: Types of MPEG-4 streams over IP
Date: Tue, 3 Oct 2000 12:14:13 +0200
Organization: France Telecom R&D
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Dear Jan,

> 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.

Here is the list you can find in V2 :

0x00 Forbidden
0x01 ObjectDescriptorStream (see 8.5)
0x02 ClockReferenceStream (see 10.2.519)
0x03 SceneDescriptionStream (see 9.2.1)
0x04 VisualStream
0x05 AudioStream
0x06 MPEG7Stream
0x07 IPMPStream (see 8.3.2)
0x08 ObjectContentInfoStream (see 8.4.2)
0x09 MPEGJStream

I don't have anything for Flexmux.

cu,

O.




From rem-conf Tue Oct 03 11:51:51 2000 
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From: Kris Huber <khuber@sorensonlabs.com>
To: Franceschini Guido <Guido.Franceschini@cselt.it>
Cc: IETF AVT <rem-conf@es.net>, 4on2andIP-sys
	 <4on2andIP-sys@advent.ee.columbia.edu>, 'Yoshihiro Kikuchi'
	 <kiku@eel.rdc.toshiba.co.jp>
Subject: RE: draft text for MPEG-4 Audio/Visual RTP I-D revision
Date: Tue, 3 Oct 2000 12:50:10 -0600 
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Hello Guido,

I think you are right.  It appears that the other alternative (change MAY to
SHALL or SHOULD) was taken rather than insertion of the MPEG-4 visual
semantics regarding picture rate in short header mode (which lacks
modulo_time_base, vop_time_increment, and vop_time_increment_resolution).
These were spelled out for short header in
draft-ietf-avt-rtp-mpeg4-es-04.txt based on a suggestion I made (e-mail of
9/8/00 "RE: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt -
timestamp of short header VOP").

Since then, I have learned that a more typical way (I think) that
hardware-assisted compression systems are used with PAL devices is with the
picture clock frequency driving the temporal_reference/TR field set to 25 Hz
rather than giving rounded values of a 30 Hz picture clock.  However, my
suggestion does match baseline H.263 and MPEG-4 short-header mode, although
I think H.263 implementers have been a little loose in their interpretation,
perhaps with good reason.

Another opinion I respect is that a better partitioning of functions would
have placed all timing information in the systems part, as (I think) was
done for the audio part.  But on the other hand you can't properly decode
video without some timing information (sequence number at least) due to
temporal prediction and B-VOPs.

I heard no discussion of the text I proposed for clarification of the
timestamp relationship to temporal_reference in short header mode after Gary
Sullivan's clarifications and suggestion that MAY be changed to SHALL.  I'm
not on the IETF AVT mailing list or at other meetings where the discussion
probably occurred.

Regards,
Kris

-----Original Message-----
From: Franceschini Guido [mailto:Guido.Franceschini@cselt.it]
Sent: Tuesday, October 03, 2000 12:20 AM
To: IETF AVT; 4on2andIP-sys; 'Yoshihiro Kikuchi'
Subject: RE: draft text for MPEG-4 Audio/Visual RTP I-D revision


Change (2) means that significantly different packets are generated in case
of Short Video Headers (H263) between the original proposal and this new
version. Am I right, or is this just an editorial matter ?

Best regards
Guido

> ----------
> From: 	Yoshihiro Kikuchi[SMTP:kiku@eel.rdc.toshiba.co.jp]
> Sent: 	Tuesday, October 03, 2000 3:36 AM
> To: 	IETF AVT; 4on2andIP-sys
> Cc: 	Yoshihiro Kikuchi
> Subject: 	draft text for MPEG-4 Audio/Visual RTP I-D revision
> 
> <<File: draft-ietf-avt-rtp-mpeg4-es-05-Sep29.txt>>
> Dear experts,
> 
> Attached find please a draft text for the revision of MPEG-4 Audio/Visual
> RTP Internet-Draft.
> 
> The revisions are mostly to reflect suggestions by the AVT chair. Most of
> them are editorial and there are some minor revisions and clarifications.
> The following is a summary of the changes:
> 
> (1) MIME subtype names were changed to "video/MP4V-ES" for video and
> "audio/MP4A-LATM" for audio.
> (2) The use of H.263 RTP payload formats (RFC 2190, 2429) in the short
> video header mode was changed to "MAY" to "SHOULD".
> (3) Clarification of timestamp definition
>   There was a suggestion not to describe the detailed definition of the
> video timestamp in the RTP spec. and change it as a reference to MPEG-4
> visual document, so as to keep consistency with the MPEG-4 visual. A
> description about the RTP timestamp in the RTCP SR packet was added.
> (4) SA/TTSI over LATM
>     A description was added to restrict the LATM usage so as not to
> transmit
>  SA data by LATM.
> (5) Other editorial and wording changes.
> 
> Please let us know if there is comment on the attached draft. The authors
> need to submit it soon as an updated I-D and hopefully final before RFC.
> 
> Best regards,
> Yoshihiro Kikuchi
> and authors of the Internet 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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Subject: PCS2001 Final Call for Papers
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------=_NextPart_000_003D_01C02DDA.EC7B5500--




From rem-conf Tue Oct 03 22:15:55 2000 
From rem-conf-request@es.net Tue Oct 03 22:15:53 2000
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	Wed, 4 Oct 2000 00:44:46 -0400
Message-Id: <200010040444.AAA09294@purple.east.isi.edu>
To: Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp>
cc: Franceschini Guido <Guido.Franceschini@cselt.it>,
        IETF AVT <rem-conf@es.net>,
        4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>
Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision 
In-Reply-To: Message from Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp> 
   of "Tue, 03 Oct 2000 18:00:30 +0900." <038001c02d18$6335a920$16c77885@ave> 
Date: Wed, 04 Oct 2000 00:44:46 -0400
From: Colin Perkins <csp@isi.edu>
X-Mailing-List: <rem-conf@es.net> 
X-Loop: rem-conf@es.net
Precedence: list

--> Yoshihiro Kikuchi writes:
>>   When the short video header mode is used, RTP payload format for H.263
><   (e.g., RFC 2190 or RFC 2429) SHOULD be used.

You should specify which of RFC 2190 or RFC 2429 is intended.

>This means that it is RECOMMENDED to use the H.263 RTP payload format for
>the short video header which is compatible with H.263 baseline. Regardless
>of MAY or SHOULD, there is a possibility that the H.263 RTP payload format
>may be used, although the possibility may be higher for SHOULD than MAY. If
>you really don't want to use the H.263 RTP payload format in some
>circumstances, you don't have to use it. The payload format is signaled
>through the MIME type.
>
>(IETF experts, please correct me if my understanding on the definition of
>SHOULD is wrong.)

SHOULD = "there may exist valid reasons in particular circumstances to
	 ignore a particular item, but the full implications must be
	 understood and carefully weighed before choosing a different
	 course."

MAY = "an item is truly optional"

[see RFC 2119 for the full wording]

Colin



From rem-conf Tue Oct 03 22:15:55 2000 
From rem-conf-request@es.net Tue Oct 03 22:15:53 2000
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Message-Id: <200010040450.AAA09317@purple.east.isi.edu>
To: Kris Huber <khuber@sorensonlabs.com>
cc: Franceschini Guido <Guido.Franceschini@cselt.it>,
        IETF AVT <rem-conf@es.net>,
        4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>,
        "'Yoshihiro Kikuchi'" <kiku@eel.rdc.toshiba.co.jp>
Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision 
In-Reply-To: Message from Kris Huber <khuber@sorensonlabs.com> 
   of "Tue, 03 Oct 2000 12:50:10 MDT." <6E031E06378BD311AEF20090273CE1BA81E6EF@el-postino.s-vision.com> 
Date: Wed, 04 Oct 2000 00:50:31 -0400
From: Colin Perkins <csp@isi.edu>
X-Mailing-List: <rem-conf@es.net> 
X-Loop: rem-conf@es.net
Precedence: list

A brief note - the issue of the timestamp is one area where I had some
concerns with this draft, but unfortunately I haven't had enough spare
cycles this past week to follow up on them. This is likely just my mis-
understanding, but input from those who know both MPEG4 and RTP, to ensure
that this is both accurate and consistent with the two timing models, would
be greatly appreciated.

Colin



--> Kris Huber writes:
>Hello Guido,
>
>I think you are right.  It appears that the other alternative (change MAY to
>SHALL or SHOULD) was taken rather than insertion of the MPEG-4 visual
>semantics regarding picture rate in short header mode (which lacks
>modulo_time_base, vop_time_increment, and vop_time_increment_resolution).
>These were spelled out for short header in
>draft-ietf-avt-rtp-mpeg4-es-04.txt based on a suggestion I made (e-mail of
>9/8/00 "RE: IETF AVT last call on draft-ietf-avt-rtp-mpeg4-es-03.txt -
>timestamp of short header VOP").
>
>Since then, I have learned that a more typical way (I think) that
>hardware-assisted compression systems are used with PAL devices is with the
>picture clock frequency driving the temporal_reference/TR field set to 25 Hz
>rather than giving rounded values of a 30 Hz picture clock.  However, my
>suggestion does match baseline H.263 and MPEG-4 short-header mode, although
>I think H.263 implementers have been a little loose in their interpretation,
>perhaps with good reason.
>
>Another opinion I respect is that a better partitioning of functions would
>have placed all timing information in the systems part, as (I think) was
>done for the audio part.  But on the other hand you can't properly decode
>video without some timing information (sequence number at least) due to
>temporal prediction and B-VOPs.
>
>I heard no discussion of the text I proposed for clarification of the
>timestamp relationship to temporal_reference in short header mode after Gary
>Sullivan's clarifications and suggestion that MAY be changed to SHALL.  I'm
>not on the IETF AVT mailing list or at other meetings where the discussion
>probably occurred.
>
>Regards,
>Kris
>
>-----Original Message-----
>From: Franceschini Guido [mailto:Guido.Franceschini@cselt.it]
>Sent: Tuesday, October 03, 2000 12:20 AM
>To: IETF AVT; 4on2andIP-sys; 'Yoshihiro Kikuchi'
>Subject: RE: draft text for MPEG-4 Audio/Visual RTP I-D revision
>
>
>Change (2) means that significantly different packets are generated in case
>of Short Video Headers (H263) between the original proposal and this new
>version. Am I right, or is this just an editorial matter ?
>
>Best regards
>Guido
>
>> ----------
>> From: 	Yoshihiro Kikuchi[SMTP:kiku@eel.rdc.toshiba.co.jp]
>> Sent: 	Tuesday, October 03, 2000 3:36 AM
>> To: 	IETF AVT; 4on2andIP-sys
>> Cc: 	Yoshihiro Kikuchi
>> Subject: 	draft text for MPEG-4 Audio/Visual RTP I-D revision
>> 
>> <<File: draft-ietf-avt-rtp-mpeg4-es-05-Sep29.txt>>
>> Dear experts,
>> 
>> Attached find please a draft text for the revision of MPEG-4 Audio/Visual
>> RTP Internet-Draft.
>> 
>> The revisions are mostly to reflect suggestions by the AVT chair. Most of
>> them are editorial and there are some minor revisions and clarifications.
>> The following is a summary of the changes:
>> 
>> (1) MIME subtype names were changed to "video/MP4V-ES" for video and
>> "audio/MP4A-LATM" for audio.
>> (2) The use of H.263 RTP payload formats (RFC 2190, 2429) in the short
>> video header mode was changed to "MAY" to "SHOULD".
>> (3) Clarification of timestamp definition
>>   There was a suggestion not to describe the detailed definition of the
>> video timestamp in the RTP spec. and change it as a reference to MPEG-4
>> visual document, so as to keep consistency with the MPEG-4 visual. A
>> description about the RTP timestamp in the RTCP SR packet was added.
>> (4) SA/TTSI over LATM
>>     A description was added to restrict the LATM usage so as not to
>> transmit
>>  SA data by LATM.
>> (5) Other editorial and wording changes.
>> 
>> Please let us know if there is comment on the attached draft. The authors
>> need to submit it soon as an updated I-D and hopefully final before RFC.
>> 
>> Best regards,
>> Yoshihiro Kikuchi
>> and authors of the Internet 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 Wed Oct 04 01:38:23 2000 
From rem-conf-request@es.net Wed Oct 04 01:38:22 2000
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From: CURET Dominique FTRD/DMI/REN <dominique.curet@rd.francetelecom.fr>
To: rem-conf <rem-conf@es.net>, 4on2andIP-sys
	 <4on2andIP-sys@advent.ee.columbia.edu>
Cc: CURET Dominique FTRD/DMI <dominique.curet@francetelecom.fr>, 
	jan.vandermeer@philips.com, 'Olivier Avaro'
	 <olivier.avaro@francetelecom.fr>
Subject: RE: Types of MPEG-4 streams over IP
Date: Wed, 4 Oct 2000 10:35:35 +0200 
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dear all,

*********Jan proposed:
- 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

*****Olivier found in V2 :
0x00 Forbidden
0x01 ObjectDescriptorStream (see 8.5)
0x02 ClockReferenceStream (see 10.2.519)
0x03 SceneDescriptionStream (see 9.2.1)
0x04 VisualStream
0x05 AudioStream
0x06 MPEG7Stream
0x07 IPMPStream (see 8.3.2)
0x08 ObjectContentInfoStream (see 8.4.2)
0x09 MPEGJStream (or MPEG-J Streams as Viswanathan said)

*******so, to my understanding, at least:
 3 streams are missing: Flexmux steeams, SL steams & MP4 file streams

******about the muxcode table in band/out of band:
Dave said that :"The MuxCodeTable needs to be delivered out-of-band".
**it certainly needs to be delivered.

Pierre showed how it is possible to deliver it, accurately in time, by=20
out of band means

**But is 'out of band' compatible with a multicast or broadcast =
scenario.
**So, I think agree with Jan that an inband means is necessary.

regards

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
_________________________________________




From rem-conf Wed Oct 04 02:40:42 2000 
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To: <dominique.curet@rd.francetelecom.fr>
Cc: <rem-conf@es.net>, <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: Types of MPEG-4 streams over IP
Message-ID: <0056890014577469000002L992*@MHS>
Date: Wed, 4 Oct 2000 11:38:00 +0200
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Dominique, all,

Dominique wrote:
*******so, to my understanding, at least:
 3 streams are missing: Flexmux steeams, SL steams & MP4 file streams

A few comments :
1) I don't think that SL stream is an additional stream type; in my vie=
w SL
streams are included in the stream types that use MPEG-4 systems.
2) The wording MP4 file stream is confusing; I don't think there is a n=
eed
to "stream" an MP4 file; instead only a mime type is needed for file
transfer.

A question : for which streams we need to specify use of RTP and SDP ? =
In my
understanding for :

- VisualStream (with and without use of MPEG-4 Systems)
- AudioStream (with and without use of MPEG-4 Systems)
- FlexMux stream

Can we assume that the following streams are probably most efficiently
contained in a FlexMux stream, and consequently that is no need to spec=
ify
use of RTP and SDP for those streams:

- ObjectDescriptorStream
- ClockReferenceStream
- SceneDescriptionStream
- MPEG-7Stream
- IPMPStream
- ObjectContentInfoStream
- MPEGJStream

Regards,

Jan

-----Original Message-----
From:	dominique.curet@rd.francetelecom.fr
[mailto:dominique.curet@rd.francetelecom.fr]
Sent:	woensdag, 04 oktober, 2000 10:39
To:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Cc:	olivier.avaro@francetelecom.fr; Jan vanderMeer;
dominique.curet@francetelecom.fr
Subject:	RE: Types of MPEG-4 streams over IP


dear all,

*********Jan proposed:
- 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

*****Olivier found in V2 :
0x00 Forbidden
0x01 ObjectDescriptorStream (see 8.5)
0x02 ClockReferenceStream (see 10.2.519)
0x03 SceneDescriptionStream (see 9.2.1)
0x04 VisualStream
0x05 AudioStream
0x06 MPEG7Stream
0x07 IPMPStream (see 8.3.2)
0x08 ObjectContentInfoStream (see 8.4.2)
0x09 MPEGJStream (or MPEG-J Streams as Viswanathan said)

*******so, to my understanding, at least:
 3 streams are missing: Flexmux steeams, SL steams & MP4 file streams

******about the muxcode table in band/out of band:
Dave said that :"The MuxCodeTable needs to be delivered out-of-band".
**it certainly needs to be delivered.

Pierre showed how it is possible to deliver it, accurately in time, by
out of band means

**But is 'out of band' compatible with a multicast or broadcast scenari=
o.
**So, I think agree with Jan that an inband means is necessary.

regards

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
> 35512           Cesson-Sevigne   Cedex   09     FRANCE
> mailto:dominique.curet@rd.francetelecom.fr
_________________________________________

=



From rem-conf Wed Oct 04 04:18:28 2000 
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Message-ID: <067f01c02df4$61442560$cc51c485@eel.rdc.toshiba.co.jp>
From: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
To: "Colin Perkins" <csp@isi.edu>
Cc: "IETF AVT" <rem-conf@es.net>,
        "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
References: <200010040444.AAA09294@purple.east.isi.edu>
Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision 
Date: Wed, 4 Oct 2000 20:15:19 +0900
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Dear Colin, all,

> >>   When the short video header mode is used, RTP payload format for
H.263
> ><   (e.g., RFC 2190 or RFC 2429) SHOULD be used.
>
> You should specify which of RFC 2190 or RFC 2429 is intended.
>
Can I ask suggestions from ITU-T H.263 experts on this selection of RFCs?
Or, if ITU-T does not have any preference, I'm not sure if it is really
necessary to change the text to decide this in "MPEG-4" RTP payload spec.

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 Wed Oct 04 04:19:36 2000 
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From: "Dave" <Dave_Bo827@mail.sol.dk>
Reply-To: Dave_Bo821@mail.sol.dk
Subject: Re: Here Is The Information As Your Requested
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	<dominique.curet@rd.francetelecom.fr>
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Subject: Re: Types of MPEG-4 streams over IP
Date: Wed, 4 Oct 2000 22:33:20 +0900
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From rem-conf Wed Oct 04 07:27:20 2000 
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To: <young@techway.co.kr>
Cc: <rem-conf@es.net>, <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: Types of MPEG-4 streams over IP
Message-ID: <0056890014590674000002L942*@MHS>
Date: Wed, 4 Oct 2000 16:26:02 +0200
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Dear Young,

You wrote :

>Can we assume that the following streams are probably most efficiently=

>contained in a FlexMux stream, and consequently that is no need to spe=
cify
>use of RTP and SDP for those streams:

>- ObjectDescriptorStream
>- ClockReferenceStream
>- SceneDescriptionStream
>- MPEG-7Stream
>- IPMPStream
>- ObjectContentInfoStream
>- MPEGJStream

In general we are not allowed to assume this condition because there is=
 no
normative limitation to this even though practically it is valid. Do yo=
u
want to put a normative usage of FlexMux for those cases?  And I think =
IPMP
streams should not be multiplexed with other streams.

I am asking just for getting opinions; I am not sure how practical it i=
s to
have a separate session for each of the above streams. You may be right=
 that
IPMP streams should not be multiplexed with other streams, but for that=
 case
FlexMux could offer a useful multiplex solution for multiple IPMP messa=
ges
within an RTP payload.

Kind regards,

Jan


=



From rem-conf Wed Oct 04 08:07:32 2000 
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From: Krister Svanbro <krister.svanbro@lu.erisoft.se>
Organization: Ericsson Research
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Qiaobing, Magnus and others, 

I am Krister Svanbro and work mainly in the ROHC WG. 
I noticed that the ROHC WG name was mentioned below and 
thought I perhaps could bring some information from a ROHC point of
view, which might help somewhat in this issue.

The ROHC WG has spent year 2000 on mainly developing a robust header 
compression scheme for RTP/UDP/IP. We have also created a framework 
based on compression profiles to enable development of other 
compression profiles such as compression of TCP/IP the coming year.

The term robust origins from that fact we realised that we need to 
design header compression schemes more robust to make them cope with 
the relative large amount of packet loss that occur in e.g. cellular 
systems. That does not, however, preclude in any way the usage of 
the ROHC scheme in other types of (possible error prone) 
environments.

We have to this date not looked upon compression of RTP 
payloadformats and we do not plan to that either. I also think that 
there should not be any need for compression of payloadformats. They 
should be as efficient as possible from the beginning. AMR is 
certainly a highly interesting codec in usage together with ROHC 
to enable VoIP to a mobile, hence it is important that the 
AMR format is efficient and does not waste any bits.
It becomes kind of strange if we in the ROHC WG maximize compression
efficiency with rather sofisticated methods to enable to go down
to a compressed header size of 1 byte for the IP/UDP/RTP header, and 
then is 1 byte extra used for the RTP payloadformat only.

Hence, I would like to encourage you to keep efficiency in mind 
when developing this payload format. I look forward to see the 
results of this work.

Cheers, 

/ Krister


Qiaobing Xie wrote:
> 
> Hi, Magnus,
> 
> See my comments below....
> 
> Magnus Westerlund wrote:
> >
> > Hello,
> >
> > The payload format is designed with IP networks in mind. The problem we see is
> > that IP is transported over a wide range of different links with different
> > behavior, e.g. different radio and wireline links. Radio and wire do not have
> > the same error characteristics. AMR will be widly used over wireless links and
> > on these links bandwidth efficiency is extremely important. This lead to the
> > bitwise sorting. Octet sorting needs octet alignement for each block of the
> > payload, i.e. padding at the end of each block. Which we considered to be to
> > expensive.
> 
> This further convinces me that the current design presented in the AMR
> draft is a radio-link-centric design, which is wrong in my opinion,
> because:
> 
> 1) bandwidth efficiency over wireless links is more of a concern for
> ROHC working group. Architecturally, it should be delt with below IP by
> header compression/stripping, as being worked upon in rohc. It doesn't
> belong to RTP payload layer (unless it is something facilitating the
> underlying header compression/stripping).



From rem-conf Wed Oct 04 08:17:50 2000 
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From: "Phil Farrar" <phil@talarian.com>
To: <phil@talarian.com>
Subject: Seminar: Catch the Vision of IP Multicast
Date: Wed, 4 Oct 2000 10:07:05 -0500
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Learn first hand about the latest advances in IP multicast technology in =
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From rem-conf Wed Oct 04 08:35:04 2000 
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From: "Bogdan Ghita" <b.ghita@jack.see.plym.ac.uk>
To: <rem-conf@es.net>
Subject: Routing for UDP/real-time vs TCP
Date: Wed, 4 Oct 2000 16:36:05 +0100
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Dear all

I have a question related to general routing issues:
I am looking into measuring RTT for a specific path using 2 different =
methods: TCP acknowledgements and RTCP NTP/LSR/DLSR fields.
Is there any specification regarding different handling by the servers =
of packets that carry TCP segments vs packets carrying UDP datagrams? I =
had several discussions lately about how some/all routers tend to drop =
first ICMP traffic, then TCP traffic and, at the end, UDP traffic, as =
the first category is not critical and the second can recover. If so, it =
would mean that there are several different transport performance rates =
for the same path, depending on the type of traffic being sent. Are the =
assumption/conclusion correct? Could you point me to some materials that =
detail this matter?

Thank you very much

Best regards
Bogdan Ghita




------=_NextPart_000_0049_01C02E21.30CFFF30
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	charset="Windows-1252"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dwindows-1252">
<META content=3D"MSHTML 5.50.4134.600" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Dear all</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>I have a question related to general =
routing=20
issues:</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>I am looking into measuring RTT for a =
specific path=20
using 2 different methods: TCP acknowledgements and RTCP NTP/LSR/DLSR=20
fields.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Is there any =
specification&nbsp;regarding different=20
handling by the servers of packets that carry TCP segments vs packets =
carrying=20
UDP datagrams? I had several discussions lately about how some/all =
routers tend=20
to drop first ICMP traffic, then TCP traffic and, at the end, UDP =
traffic, as=20
the first category is not critical and the second can recover. If so, it =
would=20
mean that there are several different transport performance rates for =
the same=20
path, depending on the type of traffic being sent. Are the =
assumption/conclusion=20
correct? Could you point me to some materials that detail this=20
matter?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Thank you very much</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Best regards</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Bogdan Ghita</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV></BODY></HTML>

------=_NextPart_000_0049_01C02E21.30CFFF30--




From rem-conf Wed Oct 04 08:50:23 2000 
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To: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
From: Stephan Wenger <stewe@cs.tu-berlin.de>
Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision 
Cc: "Colin Perkins" <csp@isi.edu>, "IETF AVT" <rem-conf@es.net>,
        "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
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Dear Yoshihiro, Group,

To my knowledge the only reason why RFC2190 is not yet OBSOLETE is
that some early H.323 products still use it.  Interoperability with such
products is probably not your main concern.  So I would suggest to select
RFC2429 only.

Stephan

At 08:15 PM 10/4/2000 +0900, Yoshihiro Kikuchi wrote:
>Dear Colin, all,
>
> > >>   When the short video header mode is used, RTP payload format for
>H.263
> > ><   (e.g., RFC 2190 or RFC 2429) SHOULD be used.
> >
> > You should specify which of RFC 2190 or RFC 2429 is intended.
> >
>Can I ask suggestions from ITU-T H.263 experts on this selection of RFCs?
>Or, if ITU-T does not have any preference, I'm not sure if it is really
>necessary to change the text to decide this in "MPEG-4" RTP payload spec.
>
>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 Wed Oct 04 09:19:21 2000 
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Date: Wed, 4 Oct 2000 09:17:04 -0700
To: jan.vandermeer@philips.com
From: Dave Singer <singer@apple.com>
Subject: RE: Types of MPEG-4 streams over IP
Cc: dominique.curet@rd.francetelecom.fr, rem-conf@es.net,
        4on2andIP-sys@advent.ee.columbia.edu
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>Can we assume that the following streams are probably most efficiently
>contained in a FlexMux stream, and consequently that is no need to specify
>use of RTP and SDP for those streams:
>
>- ObjectDescriptorStream
>- ClockReferenceStream
>- SceneDescriptionStream
>- MPEG-7Stream
>- IPMPStream
>- ObjectContentInfoStream
>- MPEGJStream


I would be strongly opposed to mandating the use of flexmux for any 
stream.  What is wrong with Carsten et al.'s sync layer mapping into 
RTP?
-- 
David Singer
Apple Computer/QuickTime



From rem-conf Wed Oct 04 09:19:36 2000 
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Date: Wed, 4 Oct 2000 09:15:55 -0700
To: CURET Dominique FTRD/DMI/REN <dominique.curet@rd.francetelecom.fr>
From: Dave Singer <singer@apple.com>
Subject: RE: Types of MPEG-4 streams over IP
Cc: rem-conf <rem-conf@es.net>,
        4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>,
        CURET Dominique FTRD/DMI <dominique.curet@francetelecom.fr>,
        jan.vandermeer@philips.com,
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>*******so, to my understanding, at least:
>  3 streams are missing: Flexmux steeams, SL steams & MP4 file streams

SL is not a stream type.  It's the layer that carries streams.  MP4 
files are not directly streamed;  they may be downloaded, but that 
requires a MIME type, not a stream type.

>******about the muxcode table in band/out of band:
>Dave said that :"The MuxCodeTable needs to be delivered out-of-band".
>**it certainly needs to be delivered.
>
>Pierre showed how it is possible to deliver it, accurately in time, by
>out of band means
>
>**But is 'out of band' compatible with a multicast or broadcast scenario.
>**So, I think agree with Jan that an inband means is necessary.


of course it is.  my framework proposal showed how.  in a multicast 
or broadcast, there is always some information that is given to the 
terminal to enable it to open the streams.  in IETF world, that is an 
SDP file.  I showed how to put the IOD and the flexmux info into the 
SDP file.
-- 
David Singer
Apple Computer/QuickTime



From rem-conf Wed Oct 04 10:19:08 2000 
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From: CURET Dominique FTRD/DMI/REN <dominique.curet@rd.francetelecom.fr>
To: 'Dave Singer' <singer@apple.com>, jan.vandermeer@philips.com
Cc: CURET Dominique FTRD/DMI/REN <dominique.curet@rd.francetelecom.fr>, 
	rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: Types of MPEG-4 streams over IP
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Dear all, dear Dave,

Dave said:
"I would be strongly opposed to mandating the use of flexmux for any=20
stream."=20

answer:
The applications that you have in mind don't seem to need multiplexing=20
at the application level. Fine with me.
When some other applications find this approach interesting, our job=20
is to give solutions to them. There should not be any technical=20
fundamentalism.

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: mercredi 4 octobre 2000 18:17
=C0: jan.vandermeer@philips.com
Cc: dominique.curet@rd.francetelecom.fr; rem-conf@es.net;
4on2andIP-sys@advent.ee.columbia.edu
Objet: RE: Types of MPEG-4 streams over IP


>Can we assume that the following streams are probably most efficiently
>contained in a FlexMux stream, and consequently that is no need to =
specify
>use of RTP and SDP for those streams:
>
>- ObjectDescriptorStream
>- ClockReferenceStream
>- SceneDescriptionStream
>- MPEG-7Stream
>- IPMPStream
>- ObjectContentInfoStream
>- MPEGJStream


I would be strongly opposed to mandating the use of flexmux for any=20
stream.  What is wrong with Carsten et al.'s sync layer mapping into=20
RTP?
--=20
David Singer
Apple Computer/QuickTime



From rem-conf Wed Oct 04 10:29:10 2000 
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From: "Narasimhan, Sam (SD-EX)" <SNarasimhan@gi.com>
Subject: RE: Types of MPEG-4 streams over IP
To: 'Dave Singer' <singer@apple.com>, jan.vandermeer@philips.com
Cc: dominique.curet@rd.francetelecom.fr, rem-conf@es.net,
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Dave,

 The MPEG requirements document asks for a payload
mapping scheme to carry flexmux streams. This is possible in
MPEG-2 transport and must be possible in RTP. We need to
define a mapping scheme which will be normative for those
applications that want to transmit flexmux streams over
RTP. I will strongly object to a mapping scheme that 
excludes or precludes the carriage of flexmux streams
over RTP.

Regards
Sam Narasimhan
Motorola

-----Original Message-----
From: Dave Singer [mailto:singer@apple.com]
Sent: Wednesday, October 04, 2000 9:17 AM
To: jan.vandermeer@philips.com
Cc: dominique.curet@rd.francetelecom.fr; rem-conf@es.net;
4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: Types of MPEG-4 streams over IP


>Can we assume that the following streams are probably most efficiently
>contained in a FlexMux stream, and consequently that is no need to specify
>use of RTP and SDP for those streams:
>
>- ObjectDescriptorStream
>- ClockReferenceStream
>- SceneDescriptionStream
>- MPEG-7Stream
>- IPMPStream
>- ObjectContentInfoStream
>- MPEGJStream


I would be strongly opposed to mandating the use of flexmux for any 
stream.  What is wrong with Carsten et al.'s sync layer mapping into 
RTP?
-- 
David Singer
Apple Computer/QuickTime



From rem-conf Wed Oct 04 12:56:18 2000 
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From rem-conf Wed Oct 04 14:24:09 2000 
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Date: Wed, 04 Oct 2000 23:23:19 +0200
To: Krister Svanbro <krister.svanbro@lu.erisoft.se>
From: Mikael Degermark <micke@CS.Arizona.EDU>
Subject: Re: Comments on draft-ietf-avt-rtp-amr-00.txt
Cc: Qiaobing Xie <xieqb@cig.mot.com>,
        Magnus Westerlund <magnus.westerlund@era.ericsson.se>, rem-conf@es.net
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ROHC WG co-chair speaking.

I have a couple of comments to this letter:

>Qiaobing Xie wrote:
>> 
>> Hi, Magnus,
>> 
>> See my comments below....
>> 
>> Magnus Westerlund wrote:
>> >
>> > Hello,
>> >
>> > The payload format is designed with IP networks in mind. The problem we see is
>> > that IP is transported over a wide range of different links with different
>> > behavior, e.g. different radio and wireline links. Radio and wire do not have
>> > the same error characteristics. AMR will be widly used over wireless links and
>> > on these links bandwidth efficiency is extremely important. This lead to the
>> > bitwise sorting. Octet sorting needs octet alignement for each block of the
>> > payload, i.e. padding at the end of each block. Which we considered to be to
>> > expensive.
>> 
>> This further convinces me that the current design presented in the AMR
>> draft is a radio-link-centric design, which is wrong in my opinion,
>> because:
>> 
>> 1) bandwidth efficiency over wireless links is more of a concern for
>> ROHC working group.

That statement is too general. ROHC deals with the efficient transmission of 
the RTP/UDP/IP header. The payload format is NOT a concern of the ROHC WG.
If a payload format is to be efficient over a wireless link, it must be made so by
others than ROHC. 

>>         Architecturally, it should be delt with below IP by
>> header compression/stripping, as being worked upon in rohc. It doesn't
>> belong to RTP payload layer (unless it is something facilitating the
>> underlying header compression/stripping).

Architecturally speaking (and here I remove my ROHC chair hat), bandwidth
efficiency over wireless links will best be obtained by 

a) having an efficiently compressed payload, which might be somewhat resilient to 
errors introduced by the wireless link.

b) compressing the RTP/UDP/IP header robustly and efficiently.

ROHC deals with b). Others, perhaps you people, need to deal with a). 
The best result will be obtained if a) is done close to the application,
i.e., above the IP layer. a) is certainly not the task of ROHC, and neither should
it be.

The IETF should not worry ONLY about wireless, of course, but it certainly
seems appropriate to ensure that things work over wireless as well as over
the rest of the currently largely wired Internet.

Cheers, 

Mikael Degermark



From rem-conf Wed Oct 04 15:45:00 2000 
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Mikael and Krister (and other ROHCers on the list):

Ok, I know I was getting myself in trouble when I mentioned the
"dreadful" word ROHC :-) :-) 

Mikael Degermark wrote:
...snip...
> >>         Architecturally, it should be delt with below IP by
> >> header compression/stripping, as being worked upon in rohc. It doesn't
> >> belong to RTP payload layer (unless it is something facilitating the
> >> underlying header compression/stripping).
> 
> Architecturally speaking (and here I remove my ROHC chair hat), bandwidth
> efficiency over wireless links will best be obtained by
> 
> a) having an efficiently compressed payload, which might be somewhat resilient to
> errors introduced by the wireless link.
> 
> b) compressing the RTP/UDP/IP header robustly and efficiently.
> 
> ROHC deals with b). Others, perhaps you people, need to deal with a).
> The best result will be obtained if a) is done close to the application,
> i.e., above the IP layer. a) is certainly not the task of ROHC, and neither should
> it be.

I think we are in "violent" agreement here :) 

The point I was trying to make is, when you are looking at optimizing
the bandwidth efficiency over a *link* with a unique error
characteristics, the framework/approach used by ROHC is more
appropriate, which is to look for header compression/stripping solutions
below the IP layer. This matches your b) above. 

But if you are designing an RTP payload format, which is meant to be
sent end-to-end over a routable IP connection, it does not make much
sense to optimize the payload format design according to the error
characteristics of any one type of the link which, after all, may or may
not be present in the connection. 

I have not problem with a) and I think it is in deed the job of AVT WG.
No doubt that it is universally desirable to design any protocol message
as compact and as compressed as possible (as Krister said: not to waste
a bit)  But in reality, you often have to make trade-off between saving
a bit in the message and the associated processing cost, especially when
the processing is done in a higher stack layer. The error resilience
part (or the lack of it) in the current AMR draft is exactly one of my
big concerns as I talked about in my first email. 

Cheers,
-Qiaobing

> 
> The IETF should not worry ONLY about wireless, of course, but it certainly
> seems appropriate to ensure that things work over wireless as well as over
> the rest of the currently largely wired Internet.
> 
> Cheers,
> 
> Mikael Degermark



From rem-conf Wed Oct 04 16:40:46 2000 
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Date: Wed, 4 Oct 2000 16:36:23 -0700
To: CURET Dominique FTRD/DMI/REN <dominique.curet@rd.francetelecom.fr>
From: Dave Singer <singer@apple.com>
Subject: RE: Types of MPEG-4 streams over IP
Cc: jan.vandermeer@philips.com,
        CURET Dominique FTRD/DMI/REN <dominique.curet@rd.francetelecom.fr>,
        rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
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At 7:15 PM +0200 10/4/00, CURET Dominique FTRD/DMI/REN wrote:
>Dear all, dear Dave,
>
>Dave said:
>"I would be strongly opposed to mandating the use of flexmux for any
>stream."
>
>answer:
>The applications that you have in mind don't seem to need multiplexing
>at the application level. Fine with me.
>When some other applications find this approach interesting, our job
>is to give solutions to them. There should not be any technical
>fundamentalism.

and Sam said:

>  The MPEG requirements document asks for a payload
>mapping scheme to carry flexmux streams. This is possible in
>MPEG-2 transport and must be possible in RTP. We need to
>define a mapping scheme which will be normative for those
>applications that want to transmit flexmux streams over
>RTP. I will strongly object to a mapping scheme that
>excludes or precludes the carriage of flexmux streams
>over RTP.

I am not opposed to allowing the use of flexmux.  The proposal was 
that certain stream types (e.g. MPEG-7) could *only* be carried in 
flexmux.  Flexmux is not always suitable;  it should be possible to 
carry a single MPEG-4 stream in a simple RTP stream, without 
requiring flexmux.

Indeed, I am not being a fundamentalist;  though I personally do not 
see much need for flexmux, I am happy that it be allowed.  I would 
like a technical solution for MPEG-4 which does not *mandate* it, 
that's all.
-- 
David Singer
Apple Computer/QuickTime



From rem-conf Wed Oct 04 18:53:06 2000 
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Subject: Re: Types of MPEG-4 streams over IP
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Dear Jan and all,

>Can we assume that the following streams are probably most efficiently
>contained in a FlexMux stream, and consequently that is no need to specify
>use of RTP and SDP for those streams:
>
>- ObjectDescriptorStream
>- ClockReferenceStream
>- SceneDescriptionStream
>- MPEG-7Stream
>- IPMPStream
>- ObjectContentInfoStream
>- MPEGJStream

In my opinion, those streams should be carried on a reliable channel
rather than RTP.

What I am not sure is the scene description stream.
There are two kind of the scene description stream, one is BIFS-update
and another is BIFS-Anim.  In my understanding, BIFS-Anim should
be streamed like audio/visual streams.

Best regards,
Yoshinori Matsui





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From: SIGNES Julien / ENVIVIO <julien.signes@rd.francetelecom.com>
To: "'Y. Matsui'" <matsui@drl.mei.co.jp>, 
	4on2andIP-sys@advent.ee.columbia.edu
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Subject: RE: Types of MPEG-4 streams over IP
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Hello,

> -----Original Message-----
> From: Y. Matsui [mailto:matsui@drl.mei.co.jp]
> Sent: Wednesday, October 04, 2000 6:42 PM
> To: 4on2andIP-sys@advent.ee.columbia.edu
> Cc: rem-conf@es.net
> Subject: Re: Types of MPEG-4 streams over IP
> 
> 
> Dear Jan and all,
> 
> >Can we assume that the following streams are probably most 
> efficiently
> >contained in a FlexMux stream, and consequently that is no 
> need to specify
> >use of RTP and SDP for those streams:
> >
> >- ObjectDescriptorStream
> >- ClockReferenceStream
> >- SceneDescriptionStream
> >- MPEG-7Stream
> >- IPMPStream
> >- ObjectContentInfoStream
> >- MPEGJStream
> 
> In my opinion, those streams should be carried on a reliable channel
> rather than RTP.
> 

I agree in general they should be fully reliable channels carrying those,as
well as BIFS comands. However, note that a lot of streaming those days is
actually done by RTP over TCP.

> What I am not sure is the scene description stream.
> There are two kind of the scene description stream, one is BIFS-update
> and another is BIFS-Anim.  In my understanding, BIFS-Anim should
> be streamed like audio/visual streams.

I agree.


> 
> Best regards,
> Yoshinori Matsui
> 

Julien 

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<TITLE>RE: Types of MPEG-4 streams over IP</TITLE>
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<BODY>

<P><FONT SIZE=3D2>Hello,</FONT>
</P>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Y. Matsui [<A =
HREF=3D"mailto:matsui@drl.mei.co.jp">mailto:matsui@drl.mei.co.jp</A>]</F=
ONT>
<BR><FONT SIZE=3D2>&gt; Sent: Wednesday, October 04, 2000 6:42 =
PM</FONT>
<BR><FONT SIZE=3D2>&gt; To: 4on2andIP-sys@advent.ee.columbia.edu</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: rem-conf@es.net</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: Re: Types of MPEG-4 streams over =
IP</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Dear Jan and all,</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt;Can we assume that the following streams =
are probably most </FONT>
<BR><FONT SIZE=3D2>&gt; efficiently</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;contained in a FlexMux stream, and =
consequently that is no </FONT>
<BR><FONT SIZE=3D2>&gt; need to specify</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;use of RTP and SDP for those =
streams:</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;- ObjectDescriptorStream</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;- ClockReferenceStream</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;- SceneDescriptionStream</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;- MPEG-7Stream</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;- IPMPStream</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;- ObjectContentInfoStream</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;- MPEGJStream</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; In my opinion, those streams should be carried =
on a reliable channel</FONT>
<BR><FONT SIZE=3D2>&gt; rather than RTP.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

<P><FONT SIZE=3D2>I agree in general they should be fully reliable =
channels carrying those,as well as BIFS comands. However, note that a =
lot of streaming those days is actually done by RTP over =
TCP.</FONT></P>

<P><FONT SIZE=3D2>&gt; What I am not sure is the scene description =
stream.</FONT>
<BR><FONT SIZE=3D2>&gt; There are two kind of the scene description =
stream, one is BIFS-update</FONT>
<BR><FONT SIZE=3D2>&gt; and another is BIFS-Anim.&nbsp; In my =
understanding, BIFS-Anim should</FONT>
<BR><FONT SIZE=3D2>&gt; be streamed like audio/visual streams.</FONT>
</P>

<P><FONT SIZE=3D2>I agree.</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Best regards,</FONT>
<BR><FONT SIZE=3D2>&gt; Yoshinori Matsui</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

<P><FONT SIZE=3D2>Julien </FONT>
</P>

</BODY>
</HTML>
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From rem-conf Wed Oct 04 19:29:26 2000 
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Date: Thu, 05 Oct 2000 04:30:33 +0200
To: Qiaobing Xie <xieqb@cig.mot.com>
From: Mikael Degermark <micke@CS.Arizona.EDU>
Subject: Re: Comments on draft-ietf-avt-rtp-amr-00.txt
Cc: Mikael Degermark <micke@optima.CS.Arizona.EDU>,
        Krister Svanbro <krister.svanbro@lu.erisoft.se>,
        Qiaobing Xie <Qiaobing_Xie-QXIE1@email.mot.com>,
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At 05:41 PM 10/4/00 -0500, Qiaobing Xie wrote:
(snip)
>The point I was trying to make is, when you are looking at optimizing
>the bandwidth efficiency over a *link* with a unique error
>characteristics, the framework/approach used by ROHC is more
>appropriate, which is to look for header compression/stripping solutions
>below the IP layer. This matches your b) above. 

>But if you are designing an RTP payload format, which is meant to be
>sent end-to-end over a routable IP connection, it does not make much
>sense to optimize the payload format design according to the error
>characteristics of any one type of the link which, after all, may or may
>not be present in the connection. 

If one expects that a common case will be that the payload is sent across
such links, it may make a lot of sense. But this is of course difficult to 
predict. 

>I have not problem with a) and I think it is in deed the job of AVT WG.
>No doubt that it is universally desirable to design any protocol message
>as compact and as compressed as possible (as Krister said: not to waste
>a bit)  But in reality, you often have to make trade-off between saving
>a bit in the message and the associated processing cost, especially when
>the processing is done in a higher stack layer. 

The tradeoff is between the cost of sending the bits across the link and
the processing cost at the sender/receiver. As the end user will ultimately 
pay for both, one has to consider the relative costs.

When the amr format is used for voice, it seems that either of two
scenarios will occur:

1) users use some sort of mobile terminal.
2) users use stationary terminals.

For 1, the cost of bandwidth is very likely to be higher than the processing cost
if the mobile uses cellular radio.

For 2, the cost of bandwidth is very low, but on the other hand 
the incremental processing cost is zero when something like a PC is used
as the stationary terminal. 

So this very simple analysis does seem to favor a high degree of compression. 

>The error resilience
>part (or the lack of it) in the current AMR draft is exactly one of my
>big concerns as I talked about in my first email.
>
>Cheers,
>-Qiaobing
>
>> 
>> The IETF should not worry ONLY about wireless, of course, but it certainly
>> seems appropriate to ensure that things work over wireless as well as over
>> the rest of the currently largely wired Internet.
>> 
>> Cheers,
>> 
>> Mikael Degermark
> 



From rem-conf Wed Oct 04 21:39:33 2000 
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From: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
To: "Stephan Wenger" <stewe@cs.tu-berlin.de>
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Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision 
Date: Thu, 5 Oct 2000 13:36:52 +0900
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Dear Stephan, Kris and all,

Thank you very much for your comments.

Seeing a comment from Kris and a suggestion form Stephan about H.263 RTP
format, I think it is reasonable to keep the use of H.263 RTP format as
"SHOULD", specifying only RFC2429 as suggested, and not to add the detailed
timestamp definition of the short video header mode.

    When the short video header mode is used, RTP payload format for H.263
    (RFC 2429) SHOULD be used.

Best regards,
Yoshihiro

----- Original Message -----
From: Stephan Wenger <stewe@cs.tu-berlin.de>
To: Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp>
Cc: Colin Perkins <csp@isi.edu>; IETF AVT <rem-conf@es.net>; Yoshihiro
Kikuchi <kiku@eel.rdc.toshiba.co.jp>
Sent: Thursday, October 05, 2000 12:39 AM
Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision


> Dear Yoshihiro, Group,
>
> To my knowledge the only reason why RFC2190 is not yet OBSOLETE is
> that some early H.323 products still use it.  Interoperability with such
> products is probably not your main concern.  So I would suggest to select
> RFC2429 only.
>
> Stephan
>
> At 08:15 PM 10/4/2000 +0900, Yoshihiro Kikuchi wrote:
> >Dear Colin, all,
> >
> > > >>   When the short video header mode is used, RTP payload format for
> >H.263
> > > ><   (e.g., RFC 2190 or RFC 2429) SHOULD be used.
> > >
> > > You should specify which of RFC 2190 or RFC 2429 is intended.
> > >
> >Can I ask suggestions from ITU-T H.263 experts on this selection of RFCs?
> >Or, if ITU-T does not have any preference, I'm not sure if it is really
> >necessary to change the text to decide this in "MPEG-4" RTP payload spec.
> >
> >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: Re: Types of MPEG-4 streams over IP
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matsui@drl.mei.co.jp wrote:

> Dear Jan and all,
>
> >Can we assume that the following streams are probably most efficient=
ly
> >contained in a FlexMux stream, and consequently that is no need to s=
pecify
> >use of RTP and SDP for those streams:
> >
> >- ObjectDescriptorStream
> >- ClockReferenceStream
> >- SceneDescriptionStream
> >- MPEG-7Stream
> >- IPMPStream
> >- ObjectContentInfoStream
> >- MPEGJStream
>
> In my opinion, those streams should be carried on a reliable channel
> rather than RTP.

I wish that was possible too, however ...

In a truly scalable distribution architecture
you must use connectionless protocols.

RTP on UDP multicast is the key candidate.

NB: Retransmission of lost data requires a return channel
that is not always present and if it were present
would probably be not very scalable so forget
about absolute reliability too, repetition (and FEC)
are what we have to make sure can work

Therefore scalable broadcast requires RTP transport
of ALL these streams as well as SDP in repeated SAP packets.


>
>
> What I am not sure is the scene description stream.
> There are two kind of the scene description stream, one is BIFS-updat=
e
> and another is BIFS-Anim.  In my understanding, BIFS-Anim should
> be streamed like audio/visual streams.
>

Same thing for both, must be possible to broadcast them too.


Regards,

--
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


=



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Date: Thu, 05 Oct 2000 10:23:37 +0200
From: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
Organization: Ericsson Research
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Subject: Re: Comments on draft-ietf-avt-rtp-amr-00.txt
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Hello everyone!

We agree with the comments from Mikael Degermark, and would like to address some
issues raised in the last days:


1) AMR is of course suitable for a multitude of applications. We know
that it will be used for wireless and also over internet. The wireless
case is important and in this case saving an octet per packet is very
desirable. Therefore, bandwidth efficiency should be prioritiesed.

2) The IAB supports, as I see it, the design of bandwidth efficient
solutions, see e.g. section 3.8 in
http://www.ietf.org/internet-drafts/draft-iab-wirelessws-01.txt .

3) What is to complex? We have compared the payload packetization with
the speech codec.  We are using the optimized official ETSI/3GPP
floating point AMR codec and an unoptimized payload packetizer. The load
is expressed in percent of an Windows NT machine with a Pentium III 500 MHz
processor.

Speech encoder: 7.1%
Payload packetizer: 0.058 %
Payload depacketizer: 0.058 %
Speech decoder: 1.1%

All figures are for the AMR mode with the most complex packetization and
the least favorable number of frames per packet, i.e. worst case for
the payload packetizer.

The data bits are already reshuffled bitwise in the speech encoder. If the robust
sorting is done in conjunction with the reshuffling in the speech encoder the extra
complexity would decrease.

Note also that the default operation is the simple and not the robust sorting.

4) UEP/UED schemes under development that can utilise the fact that the payload is
sorted in error sensitivity order.

The UDP Lite Protocol,
http://search.ietf.org/internet-drafts/draft-larzon-udplite-03.txt

An RTP Payload Format for Generic FEC with Uneven Level Protection,
http://search.ietf.org/internet-drafts/draft-li-ulp-00.txt

An RTP Payload Format for Erasure-Resilient Transmission of Progressive Multimedia
Streams, http://search.ietf.org/internet-drafts/draft-lnt-avt-uxp-02.txt


Best regards

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





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Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision
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kiku@eel.rdc.toshiba.co.jp wrote:

> Dear experts,
>
> Attached find please a draft text for the revision of MPEG-4 Audio/Vi=
sual
> RTP Internet-Draft.
>
>
>
> Please let us know if there is comment on the attached draft. The aut=
hors
> need to submit it soon as an updated I-D and hopefully final before R=
FC.
>

I would like to know why the presence of configuration information in t=
he SDP is
OPTIONAL.

I would think much preferable if the profile-level-id and config fields=
 would be
REQUIRED.

The reasons being that:

- it is only a few bytes anyway

- a decoder implementation would always
have the possibility to be initialized using the information in the SDP=
,
which is easier for many implementation because parsing the video
syntax requires a video decoder and when you instanciate
a video decoder it is VERY handy to already have the configuration !

- it would help a receiver discover that it cannot decode a given strea=
m
(without having to request and start receiving the actual stream and pa=
rse it)
and that may be the key reason: in all modes of delivery this
would solve a "security" issue:

A malignant (or stupid) user of a non-capable decoder could generate
a lot of traffic by requesting repeatedly streams that cannot be decode=
d,
which the decoder could discover "at once" by parsing the SDP,
if the SDP would always contain the config info...

- it would save (a little) bandwidth since frequent repetition of VOL i=
n the
stream is
required only for clients who do not use SDP info, you could then
have streams that actually do not repeat the VOL, or only rarely.

In point to point delivery, in case the stream cannot be decoded by cli=
ent, it
saves:
- several round trips
- quite some network traffic
- quite some server load

In broadcast (IP multicast) it  saves:

- some start time since the decoder can start parsing immediatly after
receiving the SDP, without having to wait for repeated VOL

- quite some traffic too (in case the stream cannot be decoded by clien=
t)
since issuing a IGMP "join" in order to
receive the stream will "flood" all the network legs between
broadcaster and client.

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

Furtheremore I would like to know the exact status of video B frames re=
lative
to this proposal.

>

Regards,


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


=



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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: RTP payload format for AMR
	Author(s)	: J. Sjoberg, M. Westerlund, A. Lakaniemi,
                          P. Koskelainen, B. Wimmer, T. Fingscheidt
	Filename	: draft-ietf-avt-rtp-amr-01.txt
	Pages		: 17
	Date		: 04-Oct-00
	
This document describes a proposed real-time transport protocol (RTP)
[8] payload format for AMR speech encoded [1] signals. The AMR
payload format is designed to be able to interoperate with existing
AMR transport formats. This document also includes a MIME type
registration for AMR. The MIME type is specified for both real-time
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From rem-conf Thu Oct 05 04:21:55 2000 
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From: "Jiro Katto" <katto@katto.comm.waseda.ac.jp>
To: "SIGNES Julien / ENVIVIO" <julien.signes@rd.francetelecom.com>,
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Subject: Re: Types of MPEG-4 streams over IP
Date: Thu, 5 Oct 2000 20:17:10 +0900
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Dear all,

I completely agree with Yoshinori,

> Hello,
>
> > Dear Jan and all,
> >
> > >Can we assume that the following streams are probably most
> > efficiently
> > >contained in a FlexMux stream, and consequently that is no
> > need to specify
> > >use of RTP and SDP for those streams:
> > >
> > >- ObjectDescriptorStream
> > >- ClockReferenceStream
> > >- SceneDescriptionStream
> > >- MPEG-7Stream
> > >- IPMPStream
> > >- ObjectContentInfoStream
> > >- MPEGJStream
> >
> > In my opinion, those streams should be carried on a reliable channel
> > rather than RTP.
>
> I agree in general they should be fully reliable channels carrying
those,as
> well as BIFS comands. However, note that a lot of streaming those days is
> actually done by RTP over TCP.

That may be. But, I guess RTP is still used for video and audio
streams even in this case, too.

When we say the listed streams above are conveyed as "data"
or "control" (not "video" nor "audio"), this is fully aligned with the
underlying layers (TransMux, in MPEG world), i.e. H.323 and
SIP/SDP.

I think it is free for MPEG to attach a SL header to the listed
streams in order to treat them as synchronized time events.
However, I cannot find any reason why RTP usage is forced.

> > What I am not sure is the scene description stream.
> > There are two kind of the scene description stream, one is BIFS-update
> > and another is BIFS-Anim.  In my understanding, BIFS-Anim should
> > be streamed like audio/visual streams.
>
> I agree.

I agree, too. This is the final thing to complete the "MPEG4 over
IP" framework, in my mind.

Quite sorry for my noises.
--
Jiro Katto
Waseda University




From rem-conf Thu Oct 05 05:47:42 2000 
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To: philippe.gentric@philips.com
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        rem-conf <rem-conf@es.net>
Subject: Re: Types of MPEG-4 streams over IP 
In-Reply-To: Message from philippe.gentric@philips.com 
   of "Thu, 05 Oct 2000 10:08:11 +0200." <0056900012513789000002L092*@MHS> 
Date: Thu, 05 Oct 2000 08:38:26 -0400
From: Colin Perkins <csp@isi.edu>
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--> philippe.gentric@philips.com writes:
>matsui@drl.mei.co.jp wrote:
>> In my opinion, those streams should be carried on a reliable channel
>> rather than RTP.
>
>I wish that was possible too, however ...
>
>In a truly scalable distribution architecture
>you must use connectionless protocols.
>
>RTP on UDP multicast is the key candidate.
>
>NB: Retransmission of lost data requires a return channel
>that is not always present and if it were present
>would probably be not very scalable so forget
>about absolute reliability too, repetition (and FEC)
>are what we have to make sure can work
>
>Therefore scalable broadcast requires RTP transport
>of ALL these streams as well as SDP in repeated SAP packets.

...or some form of reliable multicast. You will never get 100% reliability
>from RTP, whereas there are reliable multicast protocols which achieve this
with no backchannel. See the work ongoing in the IETF RMT working group.

I'm very nervous about the use of RTP here - it's NOT appropriate for
reliable transport.

Colin



From rem-conf Thu Oct 05 05:58:49 2000 
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Dave, all,

Dave wrote :

>I am not opposed to allowing the use of flexmux.  The proposal was
>that certain stream types (e.g. MPEG-7) could *only* be carried in
>flexmux.

I was not that strong, but only asked whether there is a need for direc=
t
carriage over RTP, given that there is a solution to carry such streams=

inside a FlexMux.

But anyhow, I think the answer is clear, though I still don't understan=
d how
to reduce the huge overhead in case of the (usually) small messages if =
not
by using FlexMux.

Regards,

Jan

=



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Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision
To: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
Cc: "Stephan Wenger" <stewe@cs.tu-berlin.de>, "Colin Perkins" <csp@isi.edu>,
        "IETF AVT" <rem-conf@es.net>,
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Hi Yoshihiro-san, Stephan, and Kris,

Since short-video-header mode of MPEG-4  is essentially H.263V1 baseline,
RFC 2190 and RFC 2429 are both
allowed to be the packetization format in the H.263 world.  For maximal
inter-operability with today's implementation,
and for compatibility with H.263 definition, I would suggest the inclusion
of both RFC2190 and RFC2429 in the
statement. I know for a fact that many H.263 over IP implementations are
still only using RFC2190.

Best regards,

Qunshan
-------------




"Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>@advent.ee.columbia.edu on
10/05/2000 12:36:52 AM

Sent by:  owner-4on2andIP-sys@advent.ee.columbia.edu


To:   "Stephan Wenger" <stewe@cs.tu-berlin.de>
cc:   "Colin Perkins" <csp@isi.edu>, "IETF AVT" <rem-conf@es.net>,
      "4on2andIP-sys" <4on2andIP-sys@advent.ee.columbia.edu>, "Yoshihiro
      Kikuchi" <kiku@eel.rdc.toshiba.co.jp>

Subject:  Re: draft text for MPEG-4 Audio/Visual RTP I-D revision


Dear Stephan, Kris and all,

Thank you very much for your comments.

Seeing a comment from Kris and a suggestion form Stephan about H.263 RTP
format, I think it is reasonable to keep the use of H.263 RTP format as
"SHOULD", specifying only RFC2429 as suggested, and not to add the detailed
timestamp definition of the short video header mode.

    When the short video header mode is used, RTP payload format for H.263
    (RFC 2429) SHOULD be used.

Best regards,
Yoshihiro

----- Original Message -----
From: Stephan Wenger <stewe@cs.tu-berlin.de>
To: Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp>
Cc: Colin Perkins <csp@isi.edu>; IETF AVT <rem-conf@es.net>; Yoshihiro
Kikuchi <kiku@eel.rdc.toshiba.co.jp>
Sent: Thursday, October 05, 2000 12:39 AM
Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision


> Dear Yoshihiro, Group,
>
> To my knowledge the only reason why RFC2190 is not yet OBSOLETE is
> that some early H.323 products still use it.  Interoperability with such
> products is probably not your main concern.  So I would suggest to select
> RFC2429 only.
>
> Stephan
>
> At 08:15 PM 10/4/2000 +0900, Yoshihiro Kikuchi wrote:
> >Dear Colin, all,
> >
> > > >>   When the short video header mode is used, RTP payload format for
> >H.263
> > > ><   (e.g., RFC 2190 or RFC 2429) SHOULD be used.
> > >
> > > You should specify which of RFC 2190 or RFC 2429 is intended.
> > >
> >Can I ask suggestions from ITU-T H.263 experts on this selection of
RFCs?
> >Or, if ITU-T does not have any preference, I'm not sure if it is really
> >necessary to change the text to decide this in "MPEG-4" RTP payload
spec.
> >
> >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 Oct 05 08:18:31 2000 
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To: micke@CS.Arizona.EDU, xieqb@cig.mot.com
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Subject: RE: Comments on draft-ietf-avt-rtp-amr-00.txt
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> The tradeoff is between the cost of sending the bits across the link and
> the processing cost at the sender/receiver.
> ...
> So this very simple analysis does seem to favor a high degree 
> of compression. 

I agree with Mikael (not only because I happen to participate
the rohc work as well :-)). Another reason: it is the physical 
law that limits the bandwidth of radio links. You cannot increase
bandwidth infinitely like it is done in wireline links by putting 
more and more optical fibers/cables. On the other hand, the power 
of chips (per volume) has keep increasing and will continue to 
do so in the future. So, the processing cost at the sender/receiver
will become less significant as time passes by.

Br,
Zhigang




From rem-conf Thu Oct 05 08:46:35 2000 
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Subject: RE: Types of MPEG-4 streams over IP
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Dave,

You wrote :

>What is wrong with Carsten et al.'s sync layer mapping into RTP

Good question, as usual. Let me list a few issues :

1) Substantial overhead for small access units; a solution is being
discussed for audio, but not yet for the other streams I listed; some m=
ay
indeed deserve a more reliable transport method, but an RTP solution wi=
ll be
required too, I think. Perhaps use of FlexMux the only solution here.
2) Inclusion of a plain SL header with associated fields; I would prefe=
r to
start with a byte indicating the length of the subsequent field that
contains the SL header. Such field is easily decodable by non-MPEG-4 sy=
stem
receivers, provides a future proof solution (in future more information=
 may
be contained in the SL header with a length that is unknown by now), an=
d
creates room for fully compatible extensions.
3) Little commonality with the Kikuchi-san drafts for audio and video.
Perhaps it is possible to merge both solutions by extending the Kikuchi=
-san
drafts with the option to carry an SL header (preceded by the above len=
gth
field), for example signaled by a new attribute "MPEG-4 SL header carri=
ed"
at SDP level.

Kind regards,

Jan

>Can we assume that the following streams are probably most efficiently=

>contained in a FlexMux stream, and consequently that is no need to spe=
cify
>use of RTP and SDP for those streams:
>
>- ObjectDescriptorStream
>- ClockReferenceStream
>- SceneDescriptionStream
>- MPEG-7Stream
>- IPMPStream
>- ObjectContentInfoStream
>- MPEGJStream


I would be strongly opposed to mandating the use of flexmux for any
stream.  What is wrong with Carsten et al.'s sync layer mapping into
RTP?
--
David Singer
Apple Computer/QuickTime



=



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 <00b301c02e6d$6663bfa0$156bb684@drl.mei.co.jp>
Date: Thu, 5 Oct 2000 09:30:23 -0700
To: "Y. Matsui" <matsui@drl.mei.co.jp>
From: Dave Singer <singer@apple.com>
Subject: Re: Types of MPEG-4 streams over IP
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At 10:41 AM +0900 10/5/00, Y. Matsui wrote:
>Dear Jan and all,
>
>>Can we assume that the following streams are probably most efficiently
>>contained in a FlexMux stream, and consequently that is no need to specify
>>use of RTP and SDP for those streams:
>>
>>- ObjectDescriptorStream
>>- ClockReferenceStream
>>- SceneDescriptionStream
>>- MPEG-7Stream
>>- IPMPStream
>>- ObjectContentInfoStream
>>- MPEGJStream
>
>In my opinion, those streams should be carried on a reliable channel
>rather than RTP.

relibaility and RTP are not necessarily exclusive;  they happen to be 
at the moment, but there are a number of proposals before the IETF 
addressing this problem.  I'd like to attack it as an orthogonal 
issue, if we can.
-- 
David Singer
Apple Computer/QuickTime



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Date: Thu, 5 Oct 2000 09:35:55 -0700
To: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
From: Dave Singer <singer@apple.com>
Subject: RE: Types of MPEG-4 streams over IP
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At 5:45 PM +0200 10/5/00, jan.vandermeer@philips.com wrote:
>Dave,
>
>You wrote :
>
>>What is wrong with Carsten et al.'s sync layer mapping into RTP
>
>Good question, as usual. Let me list a few issues :
>
>1) Substantial overhead for small access units; a solution is being
>discussed for audio, but not yet for the other streams I listed; some may
>indeed deserve a more reliable transport method, but an RTP solution will be
>required too, I think. Perhaps use of FlexMux the only solution here.

I agree.  Allow flexmux, where this is a problem, but don't mandate it.

>2) Inclusion of a plain SL header with associated fields; I would prefer to
>start with a byte indicating the length of the subsequent field that
>contains the SL header. Such field is easily decodable by non-MPEG-4 system
>receivers, provides a future proof solution (in future more information may
>be contained in the SL header with a length that is unknown by now), and
>creates room for fully compatible extensions.

OK.  This seems fine on the surface.

>3) Little commonality with the Kikuchi-san drafts for audio and video.
>Perhaps it is possible to merge both solutions by extending the Kikuchi-san
>drafts with the option to carry an SL header (preceded by the above length
>field), for example signaled by a new attribute "MPEG-4 SL header carried"
>at SDP level.

Wasn't this where the whole flare-up started;  proposing to add an 
optional SL-header to the Kikuchi proposal.

Perhaps an a=x-mpeg4-sl-header-length:<byte count>
attribute on any RTP stream carrying MPEG4 content should be 
permitted, and required if byte count is non-zero (with zero being 
the default)?

This is just a question, as I realize there are still some messages I 
have not managed to read...

>Kind regards,
>
>Jan
>
>>Can we assume that the following streams are probably most efficiently
>>contained in a FlexMux stream, and consequently that is no need to specify
>>use of RTP and SDP for those streams:
>>
>>- ObjectDescriptorStream
>>- ClockReferenceStream
>>- SceneDescriptionStream
>>- MPEG-7Stream
>>- IPMPStream
>>- ObjectContentInfoStream
>>- MPEGJStream
>
>
>I would be strongly opposed to mandating the use of flexmux for any
>stream.  What is wrong with Carsten et al.'s sync layer mapping into
>RTP?
>--
>David Singer
>Apple Computer/QuickTime

-- 
David Singer
Apple Computer/QuickTime



From rem-conf Thu Oct 05 10:38:11 2000 
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Date: Thu, 05 Oct 2000 12:34:50 -0500
From: Qiaobing Xie <xieqb@cig.mot.com>
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        Mikael Degermark <micke@optima.CS.Arizona.EDU>,
        Krister Svanbro <krister.svanbro@lu.erisoft.se>,
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Subject: Re: Comments on draft-ietf-avt-rtp-amr-00.txt
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Mikael,

see my comments below...

cheers,
-Qiaobing

Mikael Degermark wrote:
..snip...
> >But if you are designing an RTP payload format, which is meant to be
> >sent end-to-end over a routable IP connection, it does not make much
> >sense to optimize the payload format design according to the error
> >characteristics of any one type of the link which, after all, may or may
> >not be present in the connection.
> 
> If one expects that a common case will be that the payload is sent across
> such links, it may make a lot of sense. But this is of course difficult to
> predict.

In my view, AMR is a general purpose codec and a good addition to the
long list of codecs (eg, G711, G726, ...) currently used in many VoIP
applications. To have a radio link in the path is just one of the use
cases. Another thing is, even if you have a mobile unit (as your
analysis goes below) at one end of the IP connection, you probably will
still have a lot more ordinary wireline links to go through end-to-end.
It is definitely desirable to make the payload format as radio-friendly
as possible, but this should be done without compromising the "normal"
link performance too much.

> 
> >I have not problem with a) and I think it is in deed the job of AVT WG.
> >No doubt that it is universally desirable to design any protocol message
> >as compact and as compressed as possible (as Krister said: not to waste
> >a bit)  But in reality, you often have to make trade-off between saving
> >a bit in the message and the associated processing cost, especially when
> >the processing is done in a higher stack layer.
> 
> The tradeoff is between the cost of sending the bits across the link and
> the processing cost at the sender/receiver. As the end user will ultimately
> pay for both, one has to consider the relative costs.
> 
> When the amr format is used for voice, it seems that either of two
> scenarios will occur:
> 
> 1) users use some sort of mobile terminal.
> 2) users use stationary terminals.
> 
> For 1, the cost of bandwidth is very likely to be higher than the processing cost
> if the mobile uses cellular radio.
> 
> For 2, the cost of bandwidth is very low, but on the other hand
> the incremental processing cost is zero when something like a PC is used
> as the stationary terminal.
> 
> So this very simple analysis does seem to favor a high degree of compression.

I dis-agree with you here. 

First of all, you missed the third scenario: a transcoder supporting
thousands of calls and multiple codecs in a heavy duty call center. In
such a case, the computational intensity of the codec is going to be a
big factor to your overall capacity.

Secondly, if your analysis holds, why people are still doing bit-padding
(i.e., "wasting" a few bits in order to gain octet-alignments) in almost
all the protocols designed to run above IP (even IP itself is
octet-aligned by bit-staffing)? 

Maybe we misunderstood each other, but I never said bits should be
wasted, what I've been trying to point out is that the processing
advantage gained by octet alignment in upper layer protocols is
important and the AMR payload design is no exception if we are designing
it for general purpose VoIP applications. 

> 
> >The error resilience
> >part (or the lack of it) in the current AMR draft is exactly one of my
> >big concerns as I talked about in my first email.
> >
> >Cheers,
> >-Qiaobing
> >
> >>
> >> The IETF should not worry ONLY about wireless, of course, but it certainly
> >> seems appropriate to ensure that things work over wireless as well as over
> >> the rest of the currently largely wired Internet.
> >>
> >> Cheers,
> >>
> >> Mikael Degermark
> >



From rem-conf Thu Oct 05 11:11:25 2000 
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From: John Lazzaro <lazzaro@CS.Berkeley.EDU>
Message-Id: <200010051807.LAA31407@snap.CS.Berkeley.EDU>
To: micke@CS.Arizona.EDU, xieqb@cig.mot.com
Subject: Re: Comments on draft-ietf-avt-rtp-amr-00.txt
Cc: krister.svanbro@lu.erisoft.se, magnus.westerlund@era.ericsson.se,
        micke@optima.CS.Arizona.EDU, Qiaobing_Xie-QXIE1@email.mot.com,
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> First of all, you missed the third scenario: a transcoder supporting
> thousands of calls and multiple codecs in a heavy duty call center. In
> such a case, the computational intensity of the codec is going to be a
> big factor to your overall capacity.

Why can't:

[1] 1 bit in the AMR payload design signify "radio-friendly" or 
    "landline-friendly" packetizations.
[2] The "radio-friendly" packetization is designed to be ultra-efficient,
    while the "landline-friendly" packetization is designed to be 
    CPU-friendly to general-purpose CPUs, and
[3] Both packetizations are designed so that building special-purpose
    boxes that transcode large numbers of streams between the two 
    packetizations is a straightforward task for an ASIC, DSP, or FPGA?

This would seem to satisfy everyone's requirements in this thread ...

-------------------------------------------------------------------------
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 Oct 05 12:33:43 2000 
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To: Dave Singer <singer@apple.com>
cc: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
Subject: Re: Types of MPEG-4 streams over IP 
In-Reply-To: Message from Dave Singer <singer@apple.com> 
   of "Thu, 05 Oct 2000 09:35:55 PDT." <p0433010db6025d901572@[17.202.35.52]> 
Date: Thu, 05 Oct 2000 15:30:53 -0400
From: Colin Perkins <csp@isi.edu>
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--> Dave Singer writes:
>At 5:45 PM +0200 10/5/00, jan.vandermeer@philips.com wrote:
...
>>2) Inclusion of a plain SL header with associated fields; I would prefer to
>>start with a byte indicating the length of the subsequent field that
>>contains the SL header. Such field is easily decodable by non-MPEG-4 system
>>receivers, provides a future proof solution (in future more information may
>>be contained in the SL header with a length that is unknown by now), and
>>creates room for fully compatible extensions.
>
>OK.  This seems fine on the surface.

If this is an issue, it seems to be a problem with the definition of SL
headers in MPEG. The SL payload format is semantically identical to the
MPEG-4 SL packets (just the bits get re-ordered on the wire).

>>3) Little commonality with the Kikuchi-san drafts for audio and video.
>>Perhaps it is possible to merge both solutions by extending the Kikuchi-san
>>drafts with the option to carry an SL header (preceded by the above length
>>field), for example signaled by a new attribute "MPEG-4 SL header carried"
>>at SDP level.
>
>Wasn't this where the whole flare-up started;  proposing to add an 
>optional SL-header to the Kikuchi proposal.
>
>Perhaps an a=x-mpeg4-sl-header-length:<byte count>
>attribute on any RTP stream carrying MPEG4 content should be 
>permitted, and required if byte count is non-zero (with zero being 
>the default)?

In which case, signal using the payload type field...

Colin



From rem-conf Thu Oct 05 12:36:58 2000 
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From: "Dr. Carsten Bormann" <cabo@tzi.org>
To: "John Lazzaro" <lazzaro@CS.Berkeley.EDU>, <micke@CS.Arizona.EDU>,
   <xieqb@cig.mot.com>
Cc: <krister.svanbro@lu.erisoft.se>, <magnus.westerlund@era.ericsson.se>,
   <micke@optima.CS.Arizona.EDU>, <Qiaobing_Xie-QXIE1@email.mot.com>,
   <rem-conf@es.net>
Subject: RE: Comments on draft-ietf-avt-rtp-amr-00.txt
Date: Thu, 5 Oct 2000 21:19:22 +0200
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> This would seem to satisfy everyone's requirements in this thread ...

No, adding an option that increases implementation complexity further for
everyone certainly does not meet *my* requirements...
(In case you don't shiver already when you hear the word "option":
How do you make sure the other guy sends you the variant you want?)

Gruesse, Carsten




From rem-conf Thu Oct 05 12:39:15 2000 
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To: Qunshan_Gu@pictel.com
cc: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>,
        "Stephan Wenger" <stewe@cs.tu-berlin.de>, "IETF AVT" <rem-conf@es.net>,
        "4on2andIP-sys" <4on2andIP-sys@advent.ee.columbia.edu>
Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision 
In-Reply-To: Message from Qunshan_Gu@pictel.com 
   of "Thu, 05 Oct 2000 10:54:18 EDT." <OFA5AA56DA.713ED2B3-ON8525696F.0051082C@PicTel.com> 
Date: Thu, 05 Oct 2000 15:37:01 -0400
From: Colin Perkins <csp@isi.edu>
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--> Qunshan_Gu@pictel.com writes:
>
>Hi Yoshihiro-san, Stephan, and Kris,
>
>Since short-video-header mode of MPEG-4  is essentially H.263V1 baseline,
>RFC 2190 and RFC 2429 are both
>allowed to be the packetization format in the H.263 world.  For maximal
>inter-operability with today's implementation,
>and for compatibility with H.263 definition, I would suggest the inclusion
>of both RFC2190 and RFC2429 in the
>statement. I know for a fact that many H.263 over IP implementations are
>still only using RFC2190.

What about...
	"When the short video header mode is used, the RTP payload format
	 for H.263 SHOULD be used (the format defined in RFC 2429 is
	 RECOMMENDED, but the RFC 2190 format MAY be used for compatibility
	 with older implementations)."
Or is that still too strong?

Colin



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> (In case you don't shiver already when you hear the word "option":
> How do you make sure the other guy sends you the variant you want?)

Basically, if you're a cell-phone, you're prepared to send/receive
one option, and if you're not, you're prepared to send/receive the other.
Interoperability would happen because the boxes sitting between the
wired world and the wireless would translate packets on the fly. 

And yes, if you're hooking up a laptop running Windows to the cell network,
as opposed to a cell-phone or designed-for-wireless PDA, your Windows
media app will need to grok both options -- but if you've gone that
route as a laptop user you expect some things might not work out
of the box.

-------------------------------------------------------------------------
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 Oct 05 12:51:25 2000 
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Date: Thu, 5 Oct 2000 12:49:22 -0700
To: Colin Perkins <csp@isi.edu>
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 3:30 PM -0400 10/5/00, Colin Perkins wrote:
>--> Dave Singer writes:
>>At 5:45 PM +0200 10/5/00, jan.vandermeer@philips.com wrote:
>...
>>>2) Inclusion of a plain SL header with associated fields; I would prefer to
>>>start with a byte indicating the length of the subsequent field that
>>>contains the SL header. Such field is easily decodable by non-MPEG-4 system
>  >>receivers, provides a future proof solution (in future more information may
>>>be contained in the SL header with a length that is unknown by now), and
>>>creates room for fully compatible extensions.
>>
>>OK.  This seems fine on the surface.
>
>If this is an issue, it seems to be a problem with the definition of SL
>headers in MPEG. The SL payload format is semantically identical to the
>MPEG-4 SL packets (just the bits get re-ordered on the wire).

OK, I elided the point that any overlapping features with RTP 
(sequence and timestamp) should be omitted in the contained SL 
header.  There are other fields in it also...

>  >>3) Little commonality with the Kikuchi-san drafts for audio and video.
>>>Perhaps it is possible to merge both solutions by extending the Kikuchi-san
>>>drafts with the option to carry an SL header (preceded by the above length
>>>field), for example signaled by a new attribute "MPEG-4 SL header carried"
>>>at SDP level.
>>
>>Wasn't this where the whole flare-up started;  proposing to add an
>>optional SL-header to the Kikuchi proposal.
>>
>>Perhaps an a=x-mpeg4-sl-header-length:<byte count>
>>attribute on any RTP stream carrying MPEG4 content should be
>>permitted, and required if byte count is non-zero (with zero being
>>the default)?
>
>In which case, signal using the payload type field...
>
>Colin

which is what the framework used to say.  oh boy.  I personally have 
little trouble with progressing the framework, and Kikuchi, and 
Carsten, and Flexmux-in-RTP, with them all fitting into the same 
framework.  But that was not the US NB position.
-- 
David Singer
Apple Computer/QuickTime



From rem-conf Thu Oct 05 13:28:21 2000 
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Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision
To: Colin Perkins <csp@isi.edu>
Cc: Qunshan_Gu@pictel.com, "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>,
        "Stephan Wenger" <stewe@cs.tu-berlin.de>, "IETF AVT" <rem-conf@es.net>,
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Colin, et al,

I am happy with your revised statement.

Thanx.

Qunshan
-------------




Colin Perkins <csp@isi.edu>@advent.ee.columbia.edu on 10/05/2000 03:37:01
PM

Sent by:  owner-4on2andIP-sys@advent.ee.columbia.edu


To:   Qunshan_Gu@pictel.com
cc:   "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>, "Stephan Wenger"
      <stewe@cs.tu-berlin.de>, "IETF AVT" <rem-conf@es.net>,
      "4on2andIP-sys" <4on2andIP-sys@advent.ee.columbia.edu>

Subject:  Re: draft text for MPEG-4 Audio/Visual RTP I-D revision


--> Qunshan_Gu@pictel.com writes:
>
>Hi Yoshihiro-san, Stephan, and Kris,
>
>Since short-video-header mode of MPEG-4  is essentially H.263V1 baseline,
>RFC 2190 and RFC 2429 are both
>allowed to be the packetization format in the H.263 world.  For maximal
>inter-operability with today's implementation,
>and for compatibility with H.263 definition, I would suggest the inclusion
>of both RFC2190 and RFC2429 in the
>statement. I know for a fact that many H.263 over IP implementations are
>still only using RFC2190.

What about...
     "When the short video header mode is used, the RTP payload format
      for H.263 SHOULD be used (the format defined in RFC 2429 is
      RECOMMENDED, but the RFC 2190 format MAY be used for compatibility
      with older implementations)."
Or is that still too strong?

Colin






From rem-conf Thu Oct 05 13:31:13 2000 
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From: Qiaobing Xie <xieqb@cig.mot.com>
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I agree with you, and I am one of those shivering when hear the word
"option" :-) Someone once said having options in a standard means
partial failure of the standardization effort, but I won't go that far
:) IMHO, options should only be used as the last resort. 

I still believe there should be a better solution which can satisfy both
types of links. Bitwise swapping and shuffling are obviously very
unfriendly to any general purpose CPU. I will soon put forward a
proposal for a octet-oriented design. I believe that the octet-aligned
payload format (I will clearly lay it out in a draft to avt WG) will be
equally friendly to wireless link centric systems as well as other VoIP
applications (did I mention that my employer Motorola is also a
wireless/radio equipment vendor, meaning I care about wireless no less
than other wireless advocates ;-)).

Cheers,
-Qiaobing

, for the sake of the best thing is we can find something satisfy both
types of links

"Dr. Carsten Bormann" wrote:
> 
> > This would seem to satisfy everyone's requirements in this thread ...
> 
> No, adding an option that increases implementation complexity further for
> everyone certainly does not meet *my* requirements...
> (In case you don't shiver already when you hear the word "option":
> How do you make sure the other guy sends you the variant you want?)
> 
> Gruesse, Carsten



From rem-conf Thu Oct 05 13:52:43 2000 
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Subject: Re: Comments on draft-ietf-avt-rtp-amr-00.txt
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John Lazzaro wrote:
...snip...
> [3] Both packetizations are designed so that building special-purpose
>     boxes that transcode large numbers of streams between the two
>     packetizations is a straightforward task for an ASIC, DSP, or FPGA?

Even most DSPs will have a strong preference/performance gain when the
operations are octet-aligned. Asking for ASIC, FPGA support to make a
user level protocol operate with capacity does not sound encouraging to
vendors/operators at all.

Besides cell phones and laptops, transcoding is a common function for a
lot of Media Gateways. Samll or large, I don't think many of them are
built using special hardware. To lower the cost, the vendor would love
to see that they can be built on off the shelf platforms.

With the bit level shuffling loops as defined in the current payload
format draft, I guess that a 700MHz Pentium II probably will only be
able to sustain a few streams for real time transcoding. Of cause I have
no data to back this up, just guessing. Maybe the authors can give us
some performance numbers if they've even implemented it... 

-Qiaobing



From rem-conf Thu Oct 05 16:03:27 2000 
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Date: Thu, 5 Oct 2000 15:56:24 -0700
To: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
From: Dave Singer <singer@apple.com>
Subject: what Mime type for MP4 files?
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a reply from the gurus at the IETF about the (e.g. HTTP) mime type 
for MP4 files...

>The top-level questions for MPEG are:
>a) do we belong under video, or application, for general MPEG-4
>sessions?

I strongly prefer video.

>I prefer video, as it warns the terminal that a
>presentation is coming, it's parallel to MPEG, and I'm concerned that
>"application" may be treated more circumspectly by security software.
>But the Beijing MPEG meeting preferred application.

Without knowing why I cannot comment, but historically such formats have
gone under video. I expect there to be pushback if application is attempted.

>b) A taste question.  Is the /mpeg4 to match the name of the standard
>(my preference, along with Steve Casner), or /mp4 to match the file
>format name?

I personally prefer the former, but as you say, it is solely a matter of
taste.

>c) I'm fairly sure that we should say a lot more about security, as
>we allow embedded URLs, and have a Java layer that can deliver
>"applets".  What sort of analysis is needed here?

Basically, you need to note the issues. If I were you I'd take a look
at existing registrations of similar things and add what you need.
-- 
David Singer
Apple Computer/QuickTime



From rem-conf Thu Oct 05 16:14:47 2000 
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From: Mikael Degermark <micke@CS.Arizona.EDU>
Subject: Re: Comments on draft-ietf-avt-rtp-amr-00.txt
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        Mikael Degermark <micke@optima.CS.Arizona.EDU>,
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Hi Qiaobing, 

At 12:34 PM 10/5/00 -0500, Qiaobing Xie wrote:
>Mikael Degermark wrote:
>..snip...
>> >But if you are designing an RTP payload format, which is meant to be
>> >sent end-to-end over a routable IP connection, it does not make much
>> >sense to optimize the payload format design according to the error
>> >characteristics of any one type of the link which, after all, may or may
>> >not be present in the connection.
>> 
>> If one expects that a common case will be that the payload is sent across
>> such links, it may make a lot of sense. But this is of course difficult to
>> predict.
>
>In my view, AMR is a general purpose codec and a good addition to the
>long list of codecs (eg, G711, G726, ...) currently used in many VoIP
>applications. To have a radio link in the path is just one of the use
>cases. Another thing is, even if you have a mobile unit (as your
>analysis goes below) at one end of the IP connection, you probably will
>still have a lot more ordinary wireline links to go through end-to-end.
>It is definitely desirable to make the payload format as radio-friendly
>as possible, but this should be done without compromising the "normal"
>link performance too much.

The normal link performance will be improved by having smaller payloads, right?
So what you must be arguing for here cannot really have much to do with 
link performance. 

>> The tradeoff is between the cost of sending the bits across the link and
>> the processing cost at the sender/receiver. As the end user will ultimately
>> pay for both, one has to consider the relative costs.
>> 
>> When the amr format is used for voice, it seems that either of two
>> scenarios will occur:
>> 
>> 1) users use some sort of mobile terminal.
>> 2) users use stationary terminals.
>> 
>> For 1, the cost of bandwidth is very likely to be higher than the processing cost
>> if the mobile uses cellular radio.
>> 
>> For 2, the cost of bandwidth is very low, but on the other hand
>> the incremental processing cost is zero when something like a PC is used
>> as the stationary terminal.
>> 
>> So this very simple analysis does seem to favor a high degree of compression.
>
>I dis-agree with you here. 
>
>First of all, you missed the third scenario: a transcoder supporting
>thousands of calls and multiple codecs in a heavy duty call center. In
>such a case, the computational intensity of the codec is going to be a
>big factor to your overall capacity.

Yes, this is certainly a valid scenario and a valid concern. 

>Secondly, if your analysis holds, why people are still doing bit-padding
>(i.e., "wasting" a few bits in order to gain octet-alignments) in almost
>all the protocols designed to run above IP (even IP itself is
>octet-aligned by bit-staffing)? 

Perhaps because the wireless case has not been frequent enough. When the
bandwidth cost is almost zero, it does make a lot of sense for a good engineer
to save on processing power. 

>Maybe we misunderstood each other, but I never said bits should be
>wasted, what I've been trying to point out is that the processing

>advantage gained by octet alignment in upper layer protocols is
>important and the AMR payload design is no exception if we are designing
>it for general purpose VoIP applications. 

I guess the real issue is not what layer the processing is done at, but rather
if it is done by a general-purpose processor or by a more specialized processor. 
Would the call-center scenario you described above use general-purpose processors
to do the transcoding or would more specialized hardware be deployed?

Cheers, 

Micke



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From: "Mikael Bourges-Sevenier" <mikael@ivast.com>
To: "Dave Singer" <singer@apple.com>, <rem-conf@es.net>,
        <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: what Mime type for MP4 files?
Date: Thu, 5 Oct 2000 16:41:38 -0700
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Even if historically MPEG is more concerned with video, MPEG-4 deals with so
many different kinds of streams that I would prefer application/x-mp4 or
x-mpeg4.
In general as the BIFS stream defines a scene I would view it as a special
kind of application.

Mikael Bourges-Sevenier
iVAST Inc.

> -----Original Message-----
> From: owner-4on2andIP-sys@advent.ee.columbia.edu
> [mailto:owner-4on2andIP-sys@advent.ee.columbia.edu]On Behalf Of Dave
> Singer
> Sent: Thursday, October 05, 2000 3:56 PM
> To: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
> Subject: what Mime type for MP4 files?
>
>
> a reply from the gurus at the IETF about the (e.g. HTTP) mime type
> for MP4 files...
>
> >The top-level questions for MPEG are:
> >a) do we belong under video, or application, for general MPEG-4
> >sessions?
>
> I strongly prefer video.
>
> >I prefer video, as it warns the terminal that a
> >presentation is coming, it's parallel to MPEG, and I'm concerned that
> >"application" may be treated more circumspectly by security software.
> >But the Beijing MPEG meeting preferred application.
>
> Without knowing why I cannot comment, but historically such formats have
> gone under video. I expect there to be pushback if application is
> attempted.
>
> >b) A taste question.  Is the /mpeg4 to match the name of the standard
> >(my preference, along with Steve Casner), or /mp4 to match the file
> >format name?
>
> I personally prefer the former, but as you say, it is solely a matter of
> taste.
>
> >c) I'm fairly sure that we should say a lot more about security, as
> >we allow embedded URLs, and have a Java layer that can deliver
> >"applets".  What sort of analysis is needed here?
>
> Basically, you need to note the issues. If I were you I'd take a look
> at existing registrations of similar things and add what you need.
> --
> David Singer
> Apple Computer/QuickTime
>




From rem-conf Thu Oct 05 17:23:03 2000 
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Date: Thu, 5 Oct 2000 17:19:25 -0700
From: John Lazzaro <lazzaro@CS.Berkeley.EDU>
Message-Id: <200010060019.RAA00975@snap.CS.Berkeley.EDU>
To: rem-conf@es.net
Subject: Re: what Mime type for MP4 files?
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>>b) A taste question.  Is the /mpeg4 to match the name of the standard
>>(my preference, along with Steve Casner), or /mp4 to match the file
>>format name?

>
>>I personally prefer the former, but as you say, it is solely a matter of
>>taste.
>

Has the Global Music One USPTO Trademark issue on 'MP4' been brought
to a resolution yet? This might be of some relevence ...

Last I had heard of the matter is summarized in this M4IF web page:

http://www.m4if.org/mp4.html

but this is quite dated ... 

-------------------------------------------------------------------------
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 Oct 05 17:53:51 2000 
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References: <NDBBKANAIKAJACBMCMKBKEFECHAA.mikael@ivast.com>
Date: Thu, 5 Oct 2000 17:49:52 -0700
To: Mikael Bourges-Sevenier <mikael@ivast.com>
From: Dave Singer <singer@apple.com>
Subject: RE: what Mime type for MP4 files?
Cc: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
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At 4:41 PM -0700 10/5/00, Mikael Bourges-Sevenier wrote:
>Even if historically MPEG is more concerned with video, MPEG-4 deals with so
>many different kinds of streams that I would prefer application/x-mp4 or
>x-mpeg4.
>In general as the BIFS stream defines a scene I would view it as a special
>kind of application.

The problems are that many people see "application" as a potential 
threat to their security, and try to block it.  Another problem is 
that the mime type is supposed to indicate the kind of behavior at 
the client:  video indicates a time-based visual presentation, 
whereas application can be (and often is) anything at all.

The behavior of Mpeg4 is basically intended to be visual, unless an 
audio-only session is wanted (whereupon the mime type audio/mpeg4 
should be used).  In the view of this expert on mime naming, visual 
is much to be preferred (and I agree).

By the way, we are asking for an official registration, and therefore 
there will be no "x-" at all.

>Mikael Bourges-Sevenier
>iVAST Inc.
>
>>  -----Original Message-----
>>  From: owner-4on2andIP-sys@advent.ee.columbia.edu
>>  [mailto:owner-4on2andIP-sys@advent.ee.columbia.edu]On Behalf Of Dave
>>  Singer
>>  Sent: Thursday, October 05, 2000 3:56 PM
>>  To: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
>>  Subject: what Mime type for MP4 files?
>>
>>
>>  a reply from the gurus at the IETF about the (e.g. HTTP) mime type
>>  for MP4 files...
>>
>>  >The top-level questions for MPEG are:
>>  >a) do we belong under video, or application, for general MPEG-4
>>  >sessions?
>>
>>  I strongly prefer video.
>>
>>  >I prefer video, as it warns the terminal that a
>>  >presentation is coming, it's parallel to MPEG, and I'm concerned that
>>  >"application" may be treated more circumspectly by security software.
>>  >But the Beijing MPEG meeting preferred application.
>>
>>  Without knowing why I cannot comment, but historically such formats have
>>  gone under video. I expect there to be pushback if application is
>>  attempted.
>>
>>  >b) A taste question.  Is the /mpeg4 to match the name of the standard
>>  >(my preference, along with Steve Casner), or /mp4 to match the file
>>  >format name?
>>
>>  I personally prefer the former, but as you say, it is solely a matter of
>>  taste.
>>
>>  >c) I'm fairly sure that we should say a lot more about security, as
>>  >we allow embedded URLs, and have a Java layer that can deliver
>>  >"applets".  What sort of analysis is needed here?
>>
>>  Basically, you need to note the issues. If I were you I'd take a look
>>  at existing registrations of similar things and add what you need.
>  > --
>  > David Singer
>  > Apple Computer/QuickTime
>  >

-- 
David Singer
Apple Computer/QuickTime



From rem-conf Thu Oct 05 18:53:21 2000 
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To: John Lazzaro <lazzaro@CS.Berkeley.EDU>
cc: micke@CS.Arizona.EDU, xieqb@cig.mot.com, krister.svanbro@lu.erisoft.se,
        magnus.westerlund@era.ericsson.se, micke@optima.CS.Arizona.EDU,
        Qiaobing_Xie-QXIE1@email.mot.com, rem-conf@es.net
Subject: Re: Comments on draft-ietf-avt-rtp-amr-00.txt 
In-Reply-To: Message from John Lazzaro <lazzaro@CS.Berkeley.EDU> 
   of "Thu, 05 Oct 2000 11:07:42 PDT." <200010051807.LAA31407@snap.CS.Berkeley.EDU> 
Date: Thu, 05 Oct 2000 21:35:05 -0400
From: Colin Perkins <csp@isi.edu>
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--> John Lazzaro writes:
>> First of all, you missed the third scenario: a transcoder supporting
>> thousands of calls and multiple codecs in a heavy duty call center. In
>> such a case, the computational intensity of the codec is going to be a
>> big factor to your overall capacity.
>
>Why can't:
>
>[1] 1 bit in the AMR payload design signify "radio-friendly" or 
>    "landline-friendly" packetizations.

Why waste a bit in the payload? Just use a different RTP payload type
number.....

I strongly recommend we don't go down this road. 

Colin



From rem-conf Thu Oct 05 18:53:21 2000 
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To: Dave Singer <singer@apple.com>
cc: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
Subject: Re: what Mime type for MP4 files? 
In-Reply-To: Message from Dave Singer <singer@apple.com> 
   of "Thu, 05 Oct 2000 15:56:24 PDT." <p04330121b602b74313f4@[17.202.35.52]> 
Date: Thu, 05 Oct 2000 21:19:35 -0400
From: Colin Perkins <csp@isi.edu>
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Dave, all,

We have to be careful with the various RTP payload formats, so we have
consistent types, e.g. if we register video/mpeg4 for the file format,
we should register video/mpeg4-es not video/mp4-es (or whatever...).
Just something to think about when writing the payload format drafts.

--> Dave Singer writes:
>a reply from the gurus at the IETF about the (e.g. HTTP) mime type 
>for MP4 files...
>
>>The top-level questions for MPEG are:
>>a) do we belong under video, or application, for general MPEG-4
>>sessions?
>
>I strongly prefer video.
>
>>I prefer video, as it warns the terminal that a
>>presentation is coming, it's parallel to MPEG, and I'm concerned that
>>"application" may be treated more circumspectly by security software.
>>But the Beijing MPEG meeting preferred application.
>
>Without knowing why I cannot comment, but historically such formats have
>gone under video. I expect there to be pushback if application is attempted.

probably. What about audio/mp4 though? Do we need register both?

>>b) A taste question.  Is the /mpeg4 to match the name of the standard
>>(my preference, along with Steve Casner), or /mp4 to match the file
>>format name?
>
>I personally prefer the former, but as you say, it is solely a matter of
>taste.

/mp4 matches the types defined in the profile more closely...

>>c) I'm fairly sure that we should say a lot more about security, as
>>we allow embedded URLs, and have a Java layer that can deliver
>>"applets".  What sort of analysis is needed here?
>
>Basically, you need to note the issues. If I were you I'd take a look
>at existing registrations of similar things and add what you need.

Agree that there's a strong need for this - we had feedback from our area
directors at the last IETF, who were very concerned about the potential
security issues with MPEG-4. 

Colin



From rem-conf Thu Oct 05 19:11:25 2000 
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From rem-conf Fri Oct 06 02:20:05 2000 
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From: "Helmut Wittmann (EED)" <Helmut.Wittmann@eed.ericsson.se>
To: "'Mikael Degermark'" <micke@CS.Arizona.EDU>,
   Qiaobing Xie
	 <xieqb@cig.mot.com>
Cc: Krister Svanbro <krister.svanbro@lu.erisoft.se>,
   Magnus Westerlund
	 <magnus.westerlund@era.ericsson.se>, rem-conf@es.net
Subject: RE: Comments on draft-ietf-avt-rtp-amr-00.txt
Date: Fri, 6 Oct 2000 11:07:14 +0200 
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Dear Colleages,

time for a small own contribution.

I am working with Ericsson in Nuremberg in a department
that develops realtime software for transcoders for
all current mobile communication standards. My personal
work is focussed more on speech transcoding software
on PC platforms.

I do not share the concerns on processing capacity.
As mentioned before, we are talking about two totally
different application scenarios:

1 Highly specialized hardware with highly optimized
  software, such is the case for our transcoders. 
 
  In this case you are putting much effort to run
  as many speech channels as possible on a given
  piece of hardware. The additional complexity
  caused by some non-optimum bit-shuffling
  should always much smaller than the 
  "speech processing load" itself. One speech
  channel is dimensioned that it can run in
  realtime, given the known worst case known
  for one channel, assuming the worst case
  happenening coincidently on all channels
  plus also "some" extra time because it is
  difficult to judge the worst case, you need
  some complexity reserve for bug fixes,
  work-arounds, small functional changes and 
  simply because a clever designer would never 
  fully go to the theoretical limit.

  In such a scenario we would have always enough
  processing power left to cope with even
  non-optimum bit-suffling tasks. Or, spoken in the
  words of a bussiness man: "We are only talking
  about nr of speech channels per given board,
  nothing else. Everything, that does not impact
  this key factor is beyond our considerations."

  And that is the case for packetizing complexity, 
  or better: the complexity difference between 
  optimized and non-optimized packetizing: It is
  NOT SIGNIFICANT compared to the overall board
  load.

2 General purpose hardware and non-optimized software/
  software optimized to a very low degree.

  To illustrate this, I recall the complexity figures
  Magnus had given a couple of contributions before:

"(...)
We are using the optimized official ETSI/3GPP
floating point AMR codec and an unoptimized payload packetizer. The load
is expressed in percent of an Windows NT machine with a Pentium III 500 MHz
processor.

Speech encoder: 7.1%
Payload packetizer: 0.058 %
Payload depacketizer: 0.058 %
Speech decoder: 1.1%
 (...)"

  This figures derive from an AMR implementation already optimized in C and
  compiled with optimum, machine-specific switches. Hence, the load ratio
  could no further be improved in favor of the speech transcoding with simple 
  measures (assembler level optimization is not a a simple measure).

  And what would be a typical application scenario here? Maybe the 
  much-mentioned call-center. The vendor wants to use PC hardware,
  wants to be fast at the market, machine optimization would not make
  sense because we would have much faster hardware available in short
  time (Moore's Law).

  So, what does he do? Compiles his C programs directly and hopes to
  get as many speech channels onto the CPU as possible. But he would
  not try to do any further optimization to gain a fraction of what
  he now has as nr of channels. And the nr of channels? He will
  make some rough estimates on the actual capacity, given simplifying
  assumptions on actual DTX use, actual codec use distribution
  (different codecs, different AMR modi), other fluctuations such
  as varying background load etc. In any case, this will be a
  rough estimate and he is well-advised not to be too optimistic
  in his assumptions. 

  In this scenario, the packetizing load is also NOT SIGNIFICANT.
  It is even much more insignificant than for scenario 1,
  because there we are really optimizing a high-volume product
  with a very specific software architecture design. In turn,
  in scenario 2 we want use the same platform-non-specific C 
  programs for years.
  
  The following fact may illustrate this further: If you
  are measuring the load of e.g. the AMR codec on a PC, running it several
  times on the same speech data, you are experiencing fluctuations
  which exceed the load contribution of packetizing/de-packetizing
  clearly. (For clarification: Each routine which is part of
  the AMR is measured also seperately and the fluctuations reflect
  also there. The above figures are average figures for a long
  data sequence).


Summarizing, I really do consider the bit-shuffling load as insignificant.
It should not be focussed on as the ultimate optimization target.
The time-to-market IP world, and the IP call center with PC hardware belongs here, 
I guess, would reduce all our concerns to absurdum.


Best regards,
Helmut Wittmann




-----Original Message-----
From: Mikael Degermark [mailto:micke@CS.Arizona.EDU]
Sent: Friday, 06 October, 2000 01:16
To: Qiaobing Xie
Cc: Mikael Degermark; Mikael Degermark; Krister Svanbro; Magnus
Westerlund; rem-conf@es.net
Subject: Re: Comments on draft-ietf-avt-rtp-amr-00.txt


Hi Qiaobing, 

At 12:34 PM 10/5/00 -0500, Qiaobing Xie wrote:
>Mikael Degermark wrote:
>..snip...
>> >But if you are designing an RTP payload format, which is meant to be
>> >sent end-to-end over a routable IP connection, it does not make much
>> >sense to optimize the payload format design according to the error
>> >characteristics of any one type of the link which, after all, may or may
>> >not be present in the connection.
>> 
>> If one expects that a common case will be that the payload is sent across
>> such links, it may make a lot of sense. But this is of course difficult to
>> predict.
>
>In my view, AMR is a general purpose codec and a good addition to the
>long list of codecs (eg, G711, G726, ...) currently used in many VoIP
>applications. To have a radio link in the path is just one of the use
>cases. Another thing is, even if you have a mobile unit (as your
>analysis goes below) at one end of the IP connection, you probably will
>still have a lot more ordinary wireline links to go through end-to-end.
>It is definitely desirable to make the payload format as radio-friendly
>as possible, but this should be done without compromising the "normal"
>link performance too much.

The normal link performance will be improved by having smaller payloads, right?
So what you must be arguing for here cannot really have much to do with 
link performance. 

>> The tradeoff is between the cost of sending the bits across the link and
>> the processing cost at the sender/receiver. As the end user will ultimately
>> pay for both, one has to consider the relative costs.
>> 
>> When the amr format is used for voice, it seems that either of two
>> scenarios will occur:
>> 
>> 1) users use some sort of mobile terminal.
>> 2) users use stationary terminals.
>> 
>> For 1, the cost of bandwidth is very likely to be higher than the processing cost
>> if the mobile uses cellular radio.
>> 
>> For 2, the cost of bandwidth is very low, but on the other hand
>> the incremental processing cost is zero when something like a PC is used
>> as the stationary terminal.
>> 
>> So this very simple analysis does seem to favor a high degree of compression.
>
>I dis-agree with you here. 
>
>First of all, you missed the third scenario: a transcoder supporting
>thousands of calls and multiple codecs in a heavy duty call center. In
>such a case, the computational intensity of the codec is going to be a
>big factor to your overall capacity.

Yes, this is certainly a valid scenario and a valid concern. 

>Secondly, if your analysis holds, why people are still doing bit-padding
>(i.e., "wasting" a few bits in order to gain octet-alignments) in almost
>all the protocols designed to run above IP (even IP itself is
>octet-aligned by bit-staffing)? 

Perhaps because the wireless case has not been frequent enough. When the
bandwidth cost is almost zero, it does make a lot of sense for a good engineer
to save on processing power. 

>Maybe we misunderstood each other, but I never said bits should be
>wasted, what I've been trying to point out is that the processing

>advantage gained by octet alignment in upper layer protocols is
>important and the AMR payload design is no exception if we are designing
>it for general purpose VoIP applications. 

I guess the real issue is not what layer the processing is done at, but rather
if it is done by a general-purpose processor or by a more specialized processor. 
Would the call-center scenario you described above use general-purpose processors
to do the transcoding or would more specialized hardware be deployed?

Cheers, 

Micke




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To: Dave Singer <singer@apple.com>
CC: 
	CURET Dominique FTRD/DMI/REN <dominique.curet@rd.francetelecom.fr>
	, rem-conf <rem-conf@es.net>
	, 4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>
	, CURET Dominique FTRD/DMI <dominique.curet@francetelecom.fr>
	, jan.vandermeer@philips.com,
	 'Olivier Avaro' <olivier.avaro@francetelecom.fr>
Subject: Re: Types of MPEG-4 streams over IP
References: <8D0FCE51EA3DD411B8D80004AC4CCB5B132B1D@l-rmhs.lannion.cnet.fr> <p0433010bb60107c32145@[17.202.35.52]>
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Dear all,

To announce the MuxCodeTable in a multicast scenario, it is possible to s=
end
it periodically thanks to SAP/SDP. Among others, SAP is used to transmit
periodically the SDP file as described in Dave's proposal.

Dave explicitely mentionned the use of a URL in his proposal; it may be
usefull to also mention the fact that the MuxCodeTable can be also direct=
ly
conveyed in a SDP file  in its binary format.

Pierre



Dave Singer a =E9crit :

>
> >Dave said that :"The MuxCodeTable needs to be delivered out-of-band".
> >**it certainly needs to be delivered.
> >
> >Pierre showed how it is possible to deliver it, accurately in time, by
> >out of band means
> >
> >**But is 'out of band' compatible with a multicast or broadcast scenar=
io.
> >**So, I think agree with Jan that an inband means is necessary.
>
> of course it is.  my framework proposal showed how.  in a multicast
> or broadcast, there is always some information that is given to the
> terminal to enable it to open the streams.  in IETF world, that is an
> SDP file.  I showed how to put the IOD and the flexmux info into the
> SDP file.
> --
> David Singer
> Apple Computer/QuickTime




From rem-conf Fri Oct 06 04:13:23 2000 
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To: <pierre.clement@thomcast.thomson-csf.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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Pierre,

Possibly a misunderstanding from my side. I thought that the MuxCode ta=
ble
may change in the course of a stream. If yes, then it needs to be signa=
led
where in the FlexMux stream the new table starts to apply. For this pur=
pose
carriage of the table within the FlexMux stream would be useful. Howeve=
r,
probably the MuxCode table is intended to apply to any FlexMux packet i=
n
MuxCode mode throughout the entire FlexMux stream. Is this right ?

Regards,

Jan

-----Original Message-----
From:	pierre.clement@thomcast.thomson-csf.com
[mailto:pierre.clement@thomcast.thomson-csf.com]
Sent:	vrijdag, 06 oktober, 2000 12:05
To:	singer@apple.com
Cc:	olivier.avaro@francetelecom.fr; Jan vanderMeer;
dominique.curet@francetelecom.fr; 4on2andIP-sys@advent.ee.columbia.edu;=

rem-conf@es.net; dominique.curet@rd.francetelecom.fr
Subject:	Re: Types of MPEG-4 streams over IP



Dear all,

To announce the MuxCodeTable in a multicast scenario, it is possible to=
 send
it periodically thanks to SAP/SDP. Among others, SAP is used to transmi=
t
periodically the SDP file as described in Dave's proposal.

Dave explicitely mentionned the use of a URL in his proposal; it may be=

usefull to also mention the fact that the MuxCodeTable can be also dire=
ctly
conveyed in a SDP file  in its binary format.

Pierre



Dave Singer a =E9crit :

>
> >Dave said that :"The MuxCodeTable needs to be delivered out-of-band"=
.
> >**it certainly needs to be delivered.
> >
> >Pierre showed how it is possible to deliver it, accurately in time, =
by
> >out of band means
> >
> >**But is 'out of band' compatible with a multicast or broadcast scen=
ario.
> >**So, I think agree with Jan that an inband means is necessary.
>
> of course it is.  my framework proposal showed how.  in a multicast
> or broadcast, there is always some information that is given to the
> terminal to enable it to open the streams.  in IETF world, that is an=

> SDP file.  I showed how to put the IOD and the flexmux info into the
> SDP file.
> --
> David Singer
> Apple Computer/QuickTime

=



From rem-conf Fri Oct 06 04:13:23 2000 
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From: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
To: <philippe.gentric@philips.com>
Cc: "IETF AVT" <rem-conf@es.net>,
        "4on2andIP-sys" <4on2andIP-sys@advent.ee.columbia.edu>,
        "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
References: <0056900012516881000002L012*@MHS>
Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision
Date: Fri, 6 Oct 2000 20:08:19 +0900
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Dear Philippe and all,

> I would like to know why the presence of configuration information in the
SDP is
> OPTIONAL.
>
> I would think much preferable if the profile-level-id and config fields
would be
> REQUIRED.
>
I'm wondering if it is better to change profile-level-id and config as
mandate, although reasons are much different from Philippe's.

> The reasons being that:
>
> - it is only a few bytes anyway
>
No. It is at least tens of bytes and may go up to a few hundred. It is not
huge (sufficiently small compared to a VOP size) but not so very small.

> - a decoder implementation would always
> have the possibility to be initialized using the information in the SDP,
> which is easier for many implementation because parsing the video
> syntax requires a video decoder and when you instanciate
> a video decoder it is VERY handy to already have the configuration !
>
I don't think out-of-band transport of configuration information will always
save the codec load. It much depends on the implementation. But decoders
optimized for out-of-band may be relaxed.

> - it would help a receiver discover that it cannot decode a given stream
> (without having to request and start receiving the actual stream and parse
it)
> and that may be the key reason: in all modes of delivery this
> would solve a "security" issue:
>
> A malignant (or stupid) user of a non-capable decoder could generate
> a lot of traffic by requesting repeatedly streams that cannot be decoded,
> which the decoder could discover "at once" by parsing the SDP,
> if the SDP would always contain the config info...
>
> - it would save (a little) bandwidth since frequent repetition of VOL in
the
> stream is
> required only for clients who do not use SDP info, you could then
> have streams that actually do not repeat the VOL, or only rarely.
>
I'm not sure how much it is, but it may be save a little.


Maybe, another good reason for changing config to mandate is an error
resiliency. I think in most cases repetition of VOL header inside video
stream work properly, but not 100% guaranty. For error-prone environment
such as mobile, it should at least be recommended to use config in SDP which
is delivered through reliable channel (by TCP/IP?).

For detailed definition, I don't think there is no problem to change
profile-level-id to required parameter. Maybe we can remove the default
value by this. However, since config is defined that this parameter shall
not be used for the capability exchange, and actually config is not to
indicate the capability but coder configuration, there is a problem to
change it as required. How about changing the first sentence of config as
follows?

"config: This parameter SHALL be used to indicate the configuration of the
corresponding MPEG-4 visual bitstream. It SHALL NOT be used to indicate the
codec capability in the capability exchange procedure. ......"


Since the parameter config is added by requests to keep interlocking with
32x-based systems, I would like to hear if this change matches the request.
I would also like to hear from IETF experts if it is usual in IETF world to
add this kind of parameter as required.

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 Oct 06 05:50:30 2000 
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Date: Fri, 06 Oct 2000 08:51:30 -0400
From: Marshall Eubanks <tme@21rst-century.com>
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Organization: Multicast Technologies
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To: Fethi Filali <Fethi.Filali@sophia.inria.fr>
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Subject: Re: Multicast and Unicast Flows Fairness
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Fethi Filali wrote:
> 
> hi all,
> 
> My question is how the internet community define the fairness between
> reliable multicast and TCP connections sharing the same communication
> link.
> 
> Should a reliable multicast flows be treated as a TCP flow ? Or, should
> the reliable multicast flows be given more bandwidth than TCP flow
> because it is intended to serve more receivers? In the second case, how
> much more bandwidth should be given to the multicast flow and how do the
> Internet community define this fairness.
> 
> Thanks,
> 
> Fethi.

Dear Fethi;

This has been discussed at length on the AVT list, because of the
desire to add something about congestion control to the RTP profile. '
The current wording is (from   draft-avt-profile, 2000])

	        If best-effort service is being used, RTP receivers SHOULD
             monitor packet loss to ensure that the packet loss rate is
             within acceptable parameters. Packet loss is considered
             acceptable if a TCP flow across the same network path and
             experiencing the same network conditions would achieve an
             average throughput that is not less the RTP flow is
             achieving. This condition can be satisfied by implementing
             congestion control mechanisms to adapt the transmission
             rate (or the number of layers subscribed for a layered
             multicast session), or by arranging for a receiver to leave
             the session if the loss rate is unacceptably high.


I did not like, and still do not like, this wording. TCP is not like
UDP, especially UDP multicast, and I see nothing “fair” about having a
multicast stream sharing the same bandwidth as a host of unicast TCP
sessions. In case of large scale receiver based RTP/UDP congestion
control, it is also very unclear how the receiver is to determine the
amount of TCP traffic. I argued that a scheme based on monitoring of
packet loss rates only was sufficient, and suggested a simple minded
congestion monitoring wording based purely on packet loss rates.

There followed a fair amount of discussion about this point, which has
resurfaced a number of time.

The following points have been made :

1.) UDP flows, without congestion control, can ruin a bandwidth
limited pipe - see [Jeffay, 1999] for an example of congestive collapse
(38% packet
loss) when you try and stuff TCP  traffic + a 8 or 9 mbps UDP flow into
a 10 mbps channel.

2.) Conversely, most multicast applications are streaming audio or video,
and these do not respond well to TCP like changes in bandwidth - especially
the slow start / congestion control. It is possible that RMT would help here.

To the extent that there is consensus, I would cautiously say that
people have
focused more on avoiding congestive collapse instead of fairness. I
would also
cautiously claim a VERY rough consensus that multicast streams should be allowed
to take the bandwidth they need as long as there is not signs of serious
congestion, but
that they should throttle back or terminate entirely if there is, and
that this
should be determined by monitoring the rate of packet loss.

I am cc:ing the AVT list, for obvious reasons. I apologize for
those who get this twice.

References :

[draft-avt-profile, 2000] http://www.ietf.org/internet-drafts/draft-ietf-avt-profile-new-09.txt

[Jeffay, 1999] Towards a Better-Than-Best-Effort Forwarding Service for
Multimedia Flows,  K. Jeffay, IEEE Multimedia, 6, 84-88, 1999. The
“long” version is available from 
http://www.cs.unc.edu/~jeffay/papers/IEEE-MM-99-long.pdf


                                   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     

	http://www.on-the-i.com



From rem-conf Fri Oct 06 06:15:40 2000 
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From: "Pierre Clement" <pierre.clement@thomcast.thomson-csf.com>
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Subject: Re: Types of MPEG-4 streams over IP
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Dear Jan,

In case the mux code table change in the course of a flexmux stream, it i=
s true
that an in band signaling can be usefull to tell the deflexmuxer when to =
take
the change into account.

I guess it is also possible to do it by out of band mean, and among other=
s
thanks to SAP/SDP. The Message Identifier Hash for SAP to announce a chan=
ge in a
MuxCodeTable and a "t=3D " line in SDP to announce when the change should=
 be put
in place in the deflexmuxer can be linked with the RTP time stamp of the =
RTP
packets containing the flexmux stream. When thinking of the jitter introd=
uced by
the underlying network, one implementer would have to be careful about th=
e
proper time to send the updated MuxCodeTable via SAP/SDP.

In this issue, my feeling is that the configuration stage should be disso=
ciated
=66rom the maintenance stage. SAP/SDP could be used for the configuration=
 stage of
the deflexmuxer whereas in band signalling can be thought of for maintena=
nce
(i.e change in the flight) of the MuxCodeTable.

Regards,

Pierre



jan.vandermeer@philips.com a =E9crit :

> Pierre,
>
> Possibly a misunderstanding from my side. I thought that the MuxCode ta=
ble
> may change in the course of a stream. If yes, then it needs to be signa=
led
> where in the FlexMux stream the new table starts to apply. For this pur=
pose
> carriage of the table within the FlexMux stream would be useful. Howeve=
r,
> probably the MuxCode table is intended to apply to any FlexMux packet i=
n
> MuxCode mode throughout the entire FlexMux stream. Is this right ?
>
> Regards,
>
> Jan
>
> -----Original Message-----
> From:   pierre.clement@thomcast.thomson-csf.com
> [mailto:pierre.clement@thomcast.thomson-csf.com]
> Sent:   vrijdag, 06 oktober, 2000 12:05
> To:     singer@apple.com
> Cc:     olivier.avaro@francetelecom.fr; Jan vanderMeer;
> dominique.curet@francetelecom.fr; 4on2andIP-sys@advent.ee.columbia.edu;
> rem-conf@es.net; dominique.curet@rd.francetelecom.fr
> Subject:        Re: Types of MPEG-4 streams over IP
>
> Dear all,
>
> To announce the MuxCodeTable in a multicast scenario, it is possible to=
 send
> it periodically thanks to SAP/SDP. Among others, SAP is used to transmi=
t
> periodically the SDP file as described in Dave's proposal.
>
> Dave explicitely mentionned the use of a URL in his proposal; it may be
> usefull to also mention the fact that the MuxCodeTable can be also dire=
ctly
> conveyed in a SDP file  in its binary format.
>
> Pierre
>
> Dave Singer a =E9crit :
>
> >
> > >Dave said that :"The MuxCodeTable needs to be delivered out-of-band"=
=2E
> > >**it certainly needs to be delivered.
> > >
> > >Pierre showed how it is possible to deliver it, accurately in time, =
by
> > >out of band means
> > >
> > >**But is 'out of band' compatible with a multicast or broadcast scen=
ario.
> > >**So, I think agree with Jan that an inband means is necessary.
> >
> > of course it is.  my framework proposal showed how.  in a multicast
> > or broadcast, there is always some information that is given to the
> > terminal to enable it to open the streams.  in IETF world, that is an
> > SDP file.  I showed how to put the IOD and the flexmux info into the
> > SDP file.
> > --
> > David Singer
> > Apple Computer/QuickTime




From rem-conf Fri Oct 06 06:29:12 2000 
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From: Zvi Lifshitz <zvil@csi.com>
To: "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>,
        "singer@apple.com" <singer@apple.com>
Cc: "rem-conf@es.net" <rem-conf@es.net>,
        "4on2andIP-sys@advent.ee.columbia.edu" <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: Types of MPEG-4 streams over IP
Date: Fri, 6 Oct 2000 15:27:47 +0200
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[jan.vandermeer@philips.com:]

Perhaps it is possible to merge both solutions by extending the Kikuchi-san
drafts with the option to carry an SL header (preceded by the above length
field), for example signaled by a new attribute "MPEG-4 SL header carried"
at SDP level.

[Reply:]

Or just have a new payload type which is just a different (and more common) way of signaling.

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
The MPEGathon Developers Seminar,  Brussels Airport Novotel, Brussels, November 15th, 2000






From rem-conf Fri Oct 06 06:38:33 2000 
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From rem-conf Fri Oct 06 06:42:01 2000 
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Message-ID: <01C02FAB.DAA91100.zvil@csi.com>
From: Zvi Lifshitz <zvil@csi.com>
To: "'Dave Singer'" <singer@apple.com>, Colin Perkins <csp@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: Types of MPEG-4 streams over IP
Date: Fri, 6 Oct 2000 15:41:10 +0200
Organization: Optibase Ltd.
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[Dave Singer:]

I personally have 
little trouble with progressing the framework, and Kikuchi, and 
Carsten, and Flexmux-in-RTP, with them all fitting into the same 
framework.  But that was not the US NB position.

[Reply:]

I know at least one other NB which supports exactly that ;-)

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
The MPEGathon Developers Seminar,  Brussels Airport Novotel, Brussels, November 15th, 2000






From rem-conf Fri Oct 06 06:47:04 2000 
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To: "'Dave Singer'" <singer@apple.com>, "rem-conf@es.net" <rem-conf@es.net>,
        "4on2andIP-sys@advent.ee.columbia.edu" <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: what Mime type for MP4 files?
Date: Fri, 6 Oct 2000 15:46:03 +0200
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[Dave Singer:]

I strongly prefer video.

[Reply:]

So do I. Unless there is some MPEG-J stream there that operates the toaster 
or something.

[Original message:]

>I prefer video, as it warns the terminal that a
>presentation is coming, it's parallel to MPEG, and I'm concerned that
>"application" may be treated more circumspectly by security software.
>But the Beijing MPEG meeting preferred application.

Without knowing why I cannot comment, but historically such formats have
gone under video. I expect there to be pushback if application is 
attempted.

[Reply:]

The main reason was your were not there to explain. I don't remember this 
as a big issue and therefore assume we have chances to convince them this 
time.

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
The MPEGathon Developers Seminar,  Brussels Airport Novotel, Brussels, 
November 15th, 2000






From rem-conf Fri Oct 06 06:50:16 2000 
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From: Zvi Lifshitz <zvil@csi.com>
To: "'Mikael Bourges-Sevenier'" <mikael@ivast.com>,
        Dave Singer
	 <singer@apple.com>, "rem-conf@es.net" <rem-conf@es.net>,
        "4on2andIP-sys@advent.ee.columbia.edu" <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: what Mime type for MP4 files?
Date: Fri, 6 Oct 2000 15:48:53 +0200
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All these streams still create an audio/visual presentation, which is what meant by 'video' mime type.

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
The MPEGathon Developers Seminar,  Brussels Airport Novotel, Brussels, November 15th, 2000


-----Original Message-----
From:	Mikael Bourges-Sevenier [SMTP:mikael@ivast.com]
Sent:	Friday, October 06, 2000 1:42 AM
To:	Dave Singer; rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: what Mime type for MP4 files?

 
Even if historically MPEG is more concerned with video, MPEG-4 deals with so
many different kinds of streams that I would prefer application/x-mp4 or
x-mpeg4.
In general as the BIFS stream defines a scene I would view it as a special
kind of application.

Mikael Bourges-Sevenier
iVAST Inc.

> -----Original Message-----
> From: owner-4on2andIP-sys@advent.ee.columbia.edu
> [mailto:owner-4on2andIP-sys@advent.ee.columbia.edu]On Behalf Of Dave
> Singer
> Sent: Thursday, October 05, 2000 3:56 PM
> To: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
> Subject: what Mime type for MP4 files?
>
>
> a reply from the gurus at the IETF about the (e.g. HTTP) mime type
> for MP4 files...
>
> >The top-level questions for MPEG are:
> >a) do we belong under video, or application, for general MPEG-4
> >sessions?
>
> I strongly prefer video.
>
> >I prefer video, as it warns the terminal that a
> >presentation is coming, it's parallel to MPEG, and I'm concerned that
> >"application" may be treated more circumspectly by security software.
> >But the Beijing MPEG meeting preferred application.
>
> Without knowing why I cannot comment, but historically such formats have
> gone under video. I expect there to be pushback if application is
> attempted.
>
> >b) A taste question.  Is the /mpeg4 to match the name of the standard
> >(my preference, along with Steve Casner), or /mp4 to match the file
> >format name?
>
> I personally prefer the former, but as you say, it is solely a matter of
> taste.
>
> >c) I'm fairly sure that we should say a lot more about security, as
> >we allow embedded URLs, and have a Java layer that can deliver
> >"applets".  What sort of analysis is needed here?
>
> Basically, you need to note the issues. If I were you I'd take a look
> at existing registrations of similar things and add what you need.
> --
> David Singer
> Apple Computer/QuickTime
>




From rem-conf Fri Oct 06 07:44:59 2000 
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From: zhigang.liu@nokia.com
To: csp@isi.edu, lazzaro@CS.Berkeley.EDU
Cc: micke@CS.Arizona.EDU, xieqb@cig.mot.com, krister.svanbro@lu.erisoft.se,
   magnus.westerlund@era.ericsson.se, micke@optima.CS.Arizona.EDU,
   Qiaobing_Xie-QXIE1@email.mot.com, rem-conf@es.net
Subject: RE: Comments on draft-ietf-avt-rtp-amr-00.txt 
Date: Fri, 6 Oct 2000 09:39:28 -0500 
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> >[1] 1 bit in the AMR payload design signify "radio-friendly" or 
> >    "landline-friendly" packetizations.
> 
> Why waste a bit in the payload? Just use a different RTP payload type
> number.....
> 
> I strongly recommend we don't go down this road. 
> 
> Colin

That's the way to go. This remind me the situation of G723.1 ...

Zhigang
 



From rem-conf Fri Oct 06 07:45:05 2000 
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To: Pierre Clement <pierre.clement@thomcast.thomson-csf.com>
cc: jan.vandermeer@philips.com, rem-conf@es.net,
        4on2andIP-sys@advent.ee.columbia.edu
Subject: Re: Types of MPEG-4 streams over IP 
In-Reply-To: Message from Pierre Clement <pierre.clement@thomcast.thomson-csf.com> 
   of "Fri, 06 Oct 2000 15:10:53 +0200." <39DDCF5D.30C2FA98@thomcast.thomson-csf.com> 
Date: Fri, 06 Oct 2000 10:17:13 -0400
From: Colin Perkins <csp@isi.edu>
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--> Pierre Clement writes:
>Dear Jan,
>
>In case the mux code table change in the course of a flexmux stream, it is true
>that an in band signaling can be usefull to tell the deflexmuxer when to take
>the change into account.
>
>I guess it is also possible to do it by out of band mean, and among others
>thanks to SAP/SDP. The Message Identifier Hash for SAP to announce a change in a
>MuxCodeTable and a "t= " line in SDP to announce when the change should be put
>in place in the deflexmuxer can be linked with the RTP time stamp of the RTP
>packets containing the flexmux stream. When thinking of the jitter introduced by
>the underlying network, one implementer would have to be careful about the
>proper time to send the updated MuxCodeTable via SAP/SDP.
>
>In this issue, my feeling is that the configuration stage should be dissociated
>from the maintenance stage. SAP/SDP could be used for the configuration stage of
>the deflexmuxer whereas in band signalling can be thought of for maintenance
>(i.e change in the flight) of the MuxCodeTable.

Yes - SAP can't be used for the maintainance stage, because it is too slow
(one announcement every tens of _minutes_).

Colin



From rem-conf Fri Oct 06 07:45:07 2000 
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To: Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp>
cc: "philippe.gentric" <philippe.gentric@philips.com>,
        IETF AVT <rem-conf@es.net>,
        4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>
Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision 
In-Reply-To: Message from Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp> 
   of "Fri, 06 Oct 2000 20:08:19 +0900." <085201c02f85$bc4d1ce0$cc51c485@eel.rdc.toshiba.co.jp> 
Date: Fri, 06 Oct 2000 10:21:24 -0400
From: Colin Perkins <csp@isi.edu>
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--> Yoshihiro Kikuchi writes:
...
>For detailed definition, I don't think there is no problem to change
>profile-level-id to required parameter. Maybe we can remove the default
>value by this. However, since config is defined that this parameter shall
>not be used for the capability exchange, and actually config is not to
>indicate the capability but coder configuration, there is a problem to
>change it as required. How about changing the first sentence of config as
>follows?
>
>"config: This parameter SHALL be used to indicate the configuration of the
>corresponding MPEG-4 visual bitstream. It SHALL NOT be used to indicate the
>codec capability in the capability exchange procedure. ......"
>
>Since the parameter config is added by requests to keep interlocking with
>32x-based systems, I would like to hear if this change matches the request.
>I would also like to hear from IETF experts if it is usual in IETF world to
>add this kind of parameter as required.

RTP payload formats typically don't specify how the negotiation is done 
in so much detail, rather they leave it up to the definition of the setup
protocol (e.g. H.323 may not use this, but what if a SIP or RTSP based
system want's to?)

Colin



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To: <zvil@csi.com>, <csp@isi.edu>
Cc: <rem-conf@es.net>, <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: Types of MPEG-4 streams over IP
Message-ID: <0056890014650733000002L932*@MHS>
Date: Fri, 6 Oct 2000 16:56:08 +0200
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Colin, Zvi, all,

Colin wrote :
>>Perhaps an a=3Dx-mpeg4-sl-header-length:<byte count>
>>attribute on any RTP stream carrying MPEG4 content should be
>>permitted, and required if byte count is non-zero (with zero being
>>the default)?
>In which case, signal using the payload type field...

And Zvi wrote almost the same :
>Or just have a new payload type which is just a different (and more
>common) way of signaling.

There are several arguments for introducing a length byte. The main rea=
son
to introduce such field is that it creates headroom for fully backward
compatible extensions. Specifically in two cases:
a) when applications want to add an SL header to non-MPEG-4 system serv=
ices
(for reasons discussed in earlier threads);
b) when the SL header is extended in future with additional fields of a=

length unknown by now.
In such cases the length byte provides a future proof solution for addi=
ng
additional information using the same payload type. Assuming of course =
that
the receiver is capable to parse the length byte while skipping the par=
t of
the subsequent field that the receiver does not understand.

Using another payload type does also provide a solution, but this one i=
s not
backward compatible. Existing receivers will not be capable to receive =
the
new payload format.

Kind regards,

Jan



=



From rem-conf Fri Oct 06 08:07:14 2000 
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From: Zvi Lifshitz <zvil@csi.com>
To: "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>,
        "csp@isi.edu" <csp@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: Types of MPEG-4 streams over IP
Date: Fri, 6 Oct 2000 17:06:03 +0200
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Jan, I'm in favor of a length byte. In the Generic payload format. All I 
said is that signaling whether the use of extra SL info (the Generic 
format) or not (Kikuchi-san proposal) can be done through payload type.

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
The MPEGathon Developers Seminar,  Brussels Airport Novotel, Brussels, 
November 15th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Friday, October 06, 2000 4:56 PM
To:	zvil@csi.com; csp@isi.edu
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: Types of MPEG-4 streams over IP


Colin, Zvi, all,

Colin wrote :
>>Perhaps an a=x-mpeg4-sl-header-length:<byte count>
>>attribute on any RTP stream carrying MPEG4 content should be
>>permitted, and required if byte count is non-zero (with zero being
>>the default)?
>In which case, signal using the payload type field...

And Zvi wrote almost the same :
>Or just have a new payload type which is just a different (and more
>common) way of signaling.

There are several arguments for introducing a length byte. The main reason
to introduce such field is that it creates headroom for fully backward
compatible extensions. Specifically in two cases:
a) when applications want to add an SL header to non-MPEG-4 system services
(for reasons discussed in earlier threads);
b) when the SL header is extended in future with additional fields of a
length unknown by now.
In such cases the length byte provides a future proof solution for adding
additional information using the same payload type. Assuming of course that
the receiver is capable to parse the length byte while skipping the part of
the subsequent field that the receiver does not understand.

Using another payload type does also provide a solution, but this one is 
not
backward compatible. Existing receivers will not be capable to receive the
new payload format.

Kind regards,

Jan





From rem-conf Fri Oct 06 08:28:09 2000 
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Zvi,

Thanks. So we agree for at least 50 %. Now the remainder. Signaling the=

presence of the length byte by an attribute on SDP level allows the byt=
e to
be present or not. As long as you don't need the byte, don't burn the
overhead.

Regards,

Jan

-----Original Message-----
From:	zvil@csi.com [mailto:zvil@csi.com]
Sent:	vrijdag, 06 oktober, 2000 17:09
To:	csp@isi.edu; Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: Types of MPEG-4 streams over IP


Jan, I'm in favor of a length byte. In the Generic payload format. All =
I
said is that signaling whether the use of extra SL info (the Generic
format) or not (Kikuchi-san proposal) can be done through payload type.=


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.				GSM: +972(54)461-787
http://www.optibase.com		US voice mail/fax  +1(206) 888-4149
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=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
The MPEGathon Developers Seminar,  Brussels Airport Novotel, Brussels,
November 15th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Friday, October 06, 2000 4:56 PM
To:	zvil@csi.com; csp@isi.edu
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: Types of MPEG-4 streams over IP


Colin, Zvi, all,

Colin wrote :
>>Perhaps an a=3Dx-mpeg4-sl-header-length:<byte count>
>>attribute on any RTP stream carrying MPEG4 content should be
>>permitted, and required if byte count is non-zero (with zero being
>>the default)?
>In which case, signal using the payload type field...

And Zvi wrote almost the same :
>Or just have a new payload type which is just a different (and more
>common) way of signaling.

There are several arguments for introducing a length byte. The main rea=
son
to introduce such field is that it creates headroom for fully backward
compatible extensions. Specifically in two cases:
a) when applications want to add an SL header to non-MPEG-4 system serv=
ices
(for reasons discussed in earlier threads);
b) when the SL header is extended in future with additional fields of a=

length unknown by now.
In such cases the length byte provides a future proof solution for addi=
ng
additional information using the same payload type. Assuming of course =
that
the receiver is capable to parse the length byte while skipping the par=
t of
the subsequent field that the receiver does not understand.

Using another payload type does also provide a solution, but this one i=
s
not
backward compatible. Existing receivers will not be capable to receive =
the
new payload format.

Kind regards,

Jan

=



From rem-conf Fri Oct 06 09:33:56 2000 
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Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision
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kiku@eel.rdc.toshiba.co.jp wrote:

> Dear Philippe and all,
>
> > I would like to know why the presence of configuration information =
in the
> SDP is
> > OPTIONAL.
> >
> > I would think much preferable if the profile-level-id and config fi=
elds
> would be
> > REQUIRED.
> >
> I'm wondering if it is better to change profile-level-id and config a=
s
> mandate, although reasons are much different from Philippe's.
>
> > The reasons being that:
> >
> > - it is only a few bytes anyway
> >
> No. It is at least tens of bytes and may go up to a few hundred. It i=
s not
> huge (sufficiently small compared to a VOP size) but not so very smal=
l.
>
> > - a decoder implementation would always
> > have the possibility to be initialized using the information in the=
 SDP,
> > which is easier for many implementation because parsing the video
> > syntax requires a video decoder and when you instanciate
> > a video decoder it is VERY handy to already have the configuration =
!
> >
> I don't think out-of-band transport of configuration information will=
 always
> save the codec load. It much depends on the implementation. But decod=
ers
> optimized for out-of-band may be relaxed.
>
> > - it would help a receiver discover that it cannot decode a given s=
tream
> > (without having to request and start receiving the actual stream an=
d parse
> it)
> > and that may be the key reason: in all modes of delivery this
> > would solve a "security" issue:
> >
> > A malignant (or stupid) user of a non-capable decoder could generat=
e
> > a lot of traffic by requesting repeatedly streams that cannot be de=
coded,
> > which the decoder could discover "at once" by parsing the SDP,
> > if the SDP would always contain the config info...
> >
> > - it would save (a little) bandwidth since frequent repetition of V=
OL in
> the
> > stream is
> > required only for clients who do not use SDP info, you could then
> > have streams that actually do not repeat the VOL, or only rarely.
> >
> I'm not sure how much it is, but it may be save a little.
>
> Maybe, another good reason for changing config to mandate is an error=

> resiliency. I think in most cases repetition of VOL header inside vid=
eo
> stream work properly, but not 100% guaranty. For error-prone environm=
ent
> such as mobile, it should at least be recommended to use config in SD=
P which
> is delivered through reliable channel (by TCP/IP?).
>

yes !

>
> For detailed definition, I don't think there is no problem to change
> profile-level-id to required parameter. Maybe we can remove the defau=
lt
> value by this. However, since config is defined that this parameter s=
hall
> not be used for the capability exchange, and actually config is not t=
o
> indicate the capability but coder configuration, there is a problem t=
o
> change it as required. How about changing the first sentence of confi=
g as
> follows?
>
> "config: This parameter SHALL be used to indicate the configuration o=
f the
> corresponding MPEG-4 visual bitstream. It SHALL NOT be used to indica=
te the
> codec capability in the capability exchange procedure. ......"
>
> Since the parameter config is added by requests to keep interlocking =
with
> 32x-based systems, I would like to hear if this change matches the re=
quest.

>

Well what I would like is to be sure when I get a SDP
file to be able to correctly initialize my decoders

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

Also, I reiter my question about B frames.


Regards,



--
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 Fri Oct 06 09:43:20 2000 
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From: Kris Huber <khuber@sorensonlabs.com>
To: jan.vandermeer@philips.com, zvil@csi.com, csp@isi.edu
Cc: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: Types of MPEG-4 streams over IP
Date: Fri, 6 Oct 2000 10:41:31 -0600 
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Hello Jan,

For devices that (for whatever reason) must use the Kikuchi-san payload
type, and never upgrade to a new payload type, then perhaps a possibility
you haven't considered is placing future configuration information in the
user_data field of the visual streams.  This field is present in the visual
object sequence (VOS) header, the visual object (VO) header, the visual
object layer (VOL) header, and in the group of VOP (GOV) header.  It is not
in the VOP header, so to accommodate a decoding timestamp, for example,
you'd need to send a list of DTS's in the GOV header user data, for the VOPs
that follow.

OK it's a messy and ugly solution (especially if you put audio-related
information in there)!  But it's already carried in MPEG-4 and
(unwittingly?) in the Kikuchi-san draft payload format.  Perhaps MPEG-4
user_data was accepted with "configuration future-proofing" in mind (I
became involved with MPEG after it was already there, so I don't know the
history).

As for using user_data, Annex D of the systems spec. gives a registration
procedure for "the private data format identifiers" and "the IPMP system
type values".  I'm not sure if user_data is included in the meaning of
"private data format", however, or if you just use it and build decoders
that try to figure out what portion of it is yours.  As for contacting the
registration authority, D.3 says "To Be Determined".  Whatever the
organization, it is to report its activities to JTC 1, ITTF (probably
"Internet Theory Task Force") and SC 29 secretariat.

Note that since visual information consumes much more bandwidth than audio,
there is not so much wasted bandwidth associated with duplication of audio
broadcast services due to a need to support an old payload format as there
is when duplication of video broadcast service is required to prevent device
obsolescence.  Of course the whole analysis of bandwidth wastage depends on
numbers of non-upgradeable devices, but I understand there are applications
that would include a large number of MPEG-2 devices in the count, so device
or bandwidth wastage without information backward-compatibility could be
significant.

I'd appreciate clarification of user_data usage conventions by anyone,
please (no, no, don't just say it shouldn't be used!  I know it's messy and
goes against certain design principles!)  But I think it makes some kludges
possible that could prove very valuable in some cases.  Not everything of
value is pretty.  And note that the burden of parsing and using the
user_data fields is primarily on those who wish to use it.  The burden on
everyone else is to carry and skip it.  If it prevents needless replication
of high-bandwidth services for a significant period of time, it is a small
price to pay.  Of course if it gets too large, it is a more significant
price to pay in bandwidth, but that is certainly not a problem at the
present time.

Regards,
Kris

-----Original Message-----
From: jan.vandermeer@philips.com [mailto:jan.vandermeer@philips.com]
Sent: Friday, October 06, 2000 8:56 AM
To: zvil@csi.com; csp@isi.edu
Cc: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: Types of MPEG-4 streams over IP


Colin, Zvi, all,

Colin wrote :
>>Perhaps an a=x-mpeg4-sl-header-length:<byte count>
>>attribute on any RTP stream carrying MPEG4 content should be
>>permitted, and required if byte count is non-zero (with zero being
>>the default)?
>In which case, signal using the payload type field...

And Zvi wrote almost the same :
>Or just have a new payload type which is just a different (and more
>common) way of signaling.

There are several arguments for introducing a length byte. The main reason
to introduce such field is that it creates headroom for fully backward
compatible extensions. Specifically in two cases:
a) when applications want to add an SL header to non-MPEG-4 system services
(for reasons discussed in earlier threads);
b) when the SL header is extended in future with additional fields of a
length unknown by now.
In such cases the length byte provides a future proof solution for adding
additional information using the same payload type. Assuming of course that
the receiver is capable to parse the length byte while skipping the part of
the subsequent field that the receiver does not understand.

Using another payload type does also provide a solution, but this one is not
backward compatible. Existing receivers will not be capable to receive the
new payload format.

Kind regards,

Jan





From rem-conf Fri Oct 06 10:19:49 2000 
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Date: Fri, 6 Oct 2000 10:18:06 -0700
To: jan.vandermeer@philips.com
From: Dave Singer <singer@apple.com>
Subject: RE: Types of MPEG-4 streams over IP
Cc: pierre.clement@thomcast.thomson-csf.com, rem-conf@es.net,
        4on2andIP-sys@advent.ee.columbia.edu
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At 1:08 PM +0200 10/6/00, jan.vandermeer@philips.com wrote:
>Pierre,
>
>Possibly a misunderstanding from my side. I thought that the MuxCode table
>may change in the course of a stream. If yes, then it needs to be signaled
>where in the FlexMux stream the new table starts to apply. For this purpose
>carriage of the table within the FlexMux stream would be useful. However,
>probably the MuxCode table is intended to apply to any FlexMux packet in
>MuxCode mode throughout the entire FlexMux stream. Is this right ?
>
>Regards,
>
>Jan

My understanding is that the multiplex nature of the stream does not 
change, and that therefore reliable up-front signalling is best.  If 
this ins't true, the flexmux packet format may need some way to 
update the muxcodetable, but I hope that isn't true, because of 
reliability concerns.

>
>-----Original Message-----
>From:	pierre.clement@thomcast.thomson-csf.com
>[mailto:pierre.clement@thomcast.thomson-csf.com]
>Sent:	vrijdag, 06 oktober, 2000 12:05
>To:	singer@apple.com
>Cc:	olivier.avaro@francetelecom.fr; Jan vanderMeer;
>dominique.curet@francetelecom.fr; 4on2andIP-sys@advent.ee.columbia.edu;
>rem-conf@es.net; dominique.curet@rd.francetelecom.fr
>Subject:	Re: Types of MPEG-4 streams over IP
>
>
>
>Dear all,
>
>To announce the MuxCodeTable in a multicast scenario, it is possible to sen=
d
>it periodically thanks to SAP/SDP. Among others, SAP is used to transmit
>periodically the SDP file as described in Dave's proposal.
>
>Dave explicitely mentionned the use of a URL in his proposal; it may be
>usefull to also mention the fact that the MuxCodeTable can be also directly
>conveyed in a SDP file  in its binary format.
>
>Pierre
>
>
>
>Dave Singer a =E9crit :
>
>>
>>  >Dave said that :"The MuxCodeTable needs to be delivered out-of-band".
>>  >**it certainly needs to be delivered.
>>  >
>>  >Pierre showed how it is possible to deliver it, accurately in time, by
>>  >out of band means
>>  >
>>  >**But is 'out of band' compatible with a multicast or broadcast scenari=
o.
>>  >**So, I think agree with Jan that an inband means is necessary.
>>
>>  of course it is.  my framework proposal showed how.  in a multicast
>>  or broadcast, there is always some information that is given to the
>>  terminal to enable it to open the streams.  in IETF world, that is an
>>  SDP file.  I showed how to put the IOD and the flexmux info into the
>  > SDP file.
>  > --
>  > David Singer
>  > Apple Computer/QuickTime

-- 
David Singer
Apple Computer/QuickTime



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Date: Fri, 6 Oct 2000 10:16:40 -0700
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From: Dave Singer <singer@apple.com>
Subject: Re: Types of MPEG-4 streams over IP
Cc: CURET Dominique FTRD/DMI/REN <dominique.curet@rd.francetelecom.fr>,
        rem-conf <rem-conf@es.net>,
        4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>,
        CURET Dominique FTRD/DMI <dominique.curet@francetelecom.fr>,
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At 11:50 AM +0200 10/6/00, Pierre Clement wrote:
>Dear all,
>
>To announce the MuxCodeTable in a multicast scenario, it is possible to sen=
d
>it periodically thanks to SAP/SDP. Among others, SAP is used to transmit
>periodically the SDP file as described in Dave's proposal.
>
>Dave explicitely mentionned the use of a URL in his proposal; it may be
>usefull to also mention the fact that the MuxCodeTable can be also directly
>conveyed in a SDP file  in its binary format.
>
>Pierre

Yes, exactly.

I believe that the muxcodetable will often be small enough that a 
data: URL will be usable.

>
>
>Dave Singer a =E9crit :
>
>>
>>  >Dave said that :"The MuxCodeTable needs to be delivered out-of-band".
>>  >**it certainly needs to be delivered.
>>  >
>>  >Pierre showed how it is possible to deliver it, accurately in time, by
>>  >out of band means
>>  >
>>  >**But is 'out of band' compatible with a multicast or broadcast scenari=
o.
>>  >**So, I think agree with Jan that an inband means is necessary.
>>
>>  of course it is.  my framework proposal showed how.  in a multicast
>>  or broadcast, there is always some information that is given to the
>>  terminal to enable it to open the streams.  in IETF world, that is an
>>  SDP file.  I showed how to put the IOD and the flexmux info into the
>  > SDP file.
>  > --
>  > David Singer
>  > Apple Computer/QuickTime

-- 
David Singer
Apple Computer/QuickTime



From rem-conf Fri Oct 06 10:59:43 2000 
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Date: Thu, 05 Oct 2000 23:26:48 +0200
To: Colin Perkins <csp@isi.edu>
From: Stephan Wenger <stewe@cs.tu-berlin.de>
Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision 
Cc: Qunshan_Gu@pictel.com, "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>,
        "Stephan Wenger" <stewe@cs.tu-berlin.de>, "IETF AVT" <rem-conf@es.net>,
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>What about...
>         "When the short video header mode is used, the RTP payload format
>         for H.263 SHOULD be used (the format defined in RFC 2429 is
>         RECOMMENDED, but the RFC 2190 format MAY be used for compatibility
>         with older implementations)."

Fine with me.

Stephan





From rem-conf Fri Oct 06 10:59:45 2000 
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Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision 
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You really don't need to, because RFC2429 is quite verbose in that regard.

At 01:36 PM 10/5/2000 +0900, Yoshihiro Kikuchi wrote:

>[...] and not to add the detailed
>timestamp definition of the short video header mode.

You really don't need to, because RFC2429 is quite verbose in
that regard (as is RFC2190 if we put that in as well) .

Stephan





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Date: Thu, 05 Oct 2000 19:53:07 +0200
To: SIGNES Julien / ENVIVIO <julien.signes@rd.francetelecom.com>
From: Stephan Wenger <stewe@cs.tu-berlin.de>
Subject: RE: Types of MPEG-4 streams over IP
Cc: "'Y. Matsui'" <matsui@drl.mei.co.jp>, 4on2andIP-sys@advent.ee.columbia.edu,
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Julien, Group,


>I agree in general they should be fully reliable channels carrying
>those,as well as BIFS comands. However, note that a lot of
>streaming those days is actually done by RTP over TCP.

This is interesting.  I'm not aware of any commercial system doing
such a thing.  I know that telephony people discuss the use of
RTP over TCP on managed IP networks for conversational
applications, but that is probably not what you are referring to?
I'm also aware of UDP/RTP streaming techniques that use a TCP
channel for repair, (and feedback to report errors).  But again, that
is not what I would understand as RTP over TCP.  So can you
please come up with an example and/or references?  Thanks in
advance.

Also, using TCP as the transport does not automatically render
a reliable RTP channel, at least not under real-time constraints.
There is still a good chance that packets that finally arrive at the
receiver will be thrown out because they do not fall into the jitter
tolerance window of the playout buffer.  (This is true for all
re-transmission based protocols).  You simply can't guaranty that
all packets arrive in time unless you have finally received all packets.

Stephan



--=====================_280664==_.ALT
Content-Type: text/html; charset="us-ascii"

<html>
Julien, Group,<br>
<br>
<br>
<blockquote type=cite class=cite cite><font size=2>I agree in general
they should be fully reliable channels carrying </font><br>
<font size=2>those,as well as BIFS comands. However, note that a lot of
<br>
streaming those days is actually done by RTP over
TCP.</font></blockquote><br>
This is interesting.&nbsp; I'm not aware of any commercial system
doing<br>
such a thing.&nbsp; I know that telephony people discuss the use of<br>
RTP over TCP on managed IP networks for conversational<br>
applications, but that is probably not what you are referring to?<br>
I'm also aware of UDP/RTP streaming techniques that use a TCP<br>
channel for repair, (and feedback to report errors).&nbsp; But again,
that<br>
is not what I would understand as RTP over TCP.&nbsp; So can you<br>
please come up with an example and/or references?&nbsp; Thanks in<br>
advance.<br>
<br>
Also, using TCP as the transport does not automatically render<br>
a reliable RTP channel, at least not under real-time constraints.&nbsp;
<br>
There is still a good chance that packets that finally arrive at the
<br>
receiver will be thrown out because they do not fall into the jitter
<br>
tolerance window of the playout buffer.&nbsp; (This is true for all 
<br>
re-transmission based protocols).&nbsp; You simply can't guaranty that
<br>
all packets arrive in time unless you have finally received all
packets.<br>
<br>
Stephan<br>
<br>
<br>
</html>

--=====================_280664==_.ALT--




From rem-conf Fri Oct 06 13:25:54 2000 
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From: Qiaobing Xie <xieqb@cig.mot.com>
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Hi, Mikael, 

my comments below...

Cheers,
-Qiaobing

Mikael Degermark wrote:
...snip....
> >In my view, AMR is a general purpose codec and a good addition to the
> >long list of codecs (eg, G711, G726, ...) currently used in many VoIP
> >applications. To have a radio link in the path is just one of the use
> >cases. Another thing is, even if you have a mobile unit (as your
> >analysis goes below) at one end of the IP connection, you probably will
> >still have a lot more ordinary wireline links to go through end-to-end.
> >It is definitely desirable to make the payload format as radio-friendly
> >as possible, but this should be done without compromising the "normal"
> >link performance too much.
> 
> The normal link performance will be improved by having smaller payloads, right?
> So what you must be arguing for here cannot really have much to do with
> link performance.

It is not just the size of the payload, it has something to do with the
selection of the error handling/recovering strategy. For example, based
on the link error characteristcs and transmission style, you may choose
to make the payload itself small and less error tolerant but rely on
outside mechanisms such as some FEC mechanisms to control the error, or
you may choose to have a bigger paylaod with more sophisticated built-in
error recover machanism. I don't think there is a universal strategy
which works equally well to all link types :-) 

...snip...
> >advantage gained by octet alignment in upper layer protocols is
> >important and the AMR payload design is no exception if we are designing
> >it for general purpose VoIP applications.
> 
> I guess the real issue is not what layer the processing is done at, but rather
> if it is done by a general-purpose processor or by a more specialized processor.
> Would the call-center scenario you described above use general-purpose processors
> to do the transcoding or would more specialized hardware be deployed?

This has something to do with the way layered protocols are implemented
and optimized. The lower you are in the protocol stack, the more you
will be willing to pay for the extra effort doing optimization, because
the lower layers tend to be more heavily shared in most systems. The
upper layers are usually more application-specific and their
optimization becomes less cost-effective in most cases.

> 
> Cheers,
> 
> Micke



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From: Florissa Colina <florissa@bmrc.berkeley.edu>
Subject: 10/11 The Making of GUI Bloopers -- Jeff Johnson, UI Wizards,
  Inc. 
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Berkeley Multimedia, Graphics and Interfaces Seminar

The Making of GUI Bloopers

Wednesday October 11, 2000, 
1:10-2:30 p.m. PDT
Fujitsu Seminar Room (405 Soda Hall) 

Jeff Johnson 
UI Wizards, Inc. 

Jeff Johnson, user-interface consultant and author of the new book "GUI
Bloopers: DON'Ts and DOs for Software Developers and Web Designers"
(Morgan-Kaufmann), will describe why he began collecting GUI design
bloopers several years ago, why he decided to publish them in a book, how
the bloopers are organized and presented, and how the book was usability
tested. His talk will be illustrated with many examples of bloopers.

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

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:

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 Oct 06 14:55:32 2000 
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Date: Fri, 06 Oct 2000 23:54:58 +0200
To: Qiaobing Xie <xieqb@cig.mot.com>
From: Mikael Degermark <micke@CS.Arizona.EDU>
Subject: Re: Comments on draft-ietf-avt-rtp-amr-00.txt
Cc: Mikael Degermark <micke@optima.CS.Arizona.EDU>,
        Qiaobing Xie <Qiaobing_Xie-QXIE1@email.mot.com>,
        Mikael Degermark <micke@optima.CS.Arizona.EDU>,
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Hi Qiaobing, 

I didn't realize that you were mostly talking about the error resilience properties. 
I thought you were mostly arguing for a more CPU friendly packetization format. 

Seems that the latter issue has been resolved in favor of non-byte aligned if I
correctly understand the performance measurements put forward by Ericsson people
concerning the relative costs of encoding/decoding and packetization. 

Seems like a no-brainer to me. Coding/decoding is so much more expensive
than packetization that optimizing the latter to save on processing is pointless.
This also makes it meaningless to have a bit distinguishing the two formats
as have been proposed.

But your argument against the amr-00 proposal not really about packetization 
efficiency, if I understand your latest letter correctly, but instead about
error resiliency. Isn't that correct? 

If correct, we should stop arguing about packetization efficiency and instead
talk about error resiliency.

(see also a comment below)

Micke

At 03:21 PM 10/6/00 -0500, Qiaobing Xie wrote:
>Hi, Mikael, 
>
>my comments below...
>
>Cheers,
>-Qiaobing
>
>Mikael Degermark wrote:
>...snip....
>> >In my view, AMR is a general purpose codec and a good addition to the
>> >long list of codecs (eg, G711, G726, ...) currently used in many VoIP
>> >applications. To have a radio link in the path is just one of the use
>> >cases. Another thing is, even if you have a mobile unit (as your
>> >analysis goes below) at one end of the IP connection, you probably will
>> >still have a lot more ordinary wireline links to go through end-to-end.
>> >It is definitely desirable to make the payload format as radio-friendly
>> >as possible, but this should be done without compromising the "normal"
>> >link performance too much.
>> 
>> The normal link performance will be improved by having smaller payloads, right?
>> So what you must be arguing for here cannot really have much to do with
>> link performance.
>
>It is not just the size of the payload, it has something to do with the
>selection of the error handling/recovering strategy. For example, based
>on the link error characteristcs and transmission style, you may choose
>to make the payload itself small and less error tolerant but rely on
>outside mechanisms such as some FEC mechanisms to control the error, or
>you may choose to have a bigger paylaod with more sophisticated built-in
>error recover machanism. I don't think there is a universal strategy
>which works equally well to all link types :-) 
>
>...snip...
>> >advantage gained by octet alignment in upper layer protocols is
>> >important and the AMR payload design is no exception if we are designing
>> >it for general purpose VoIP applications.
>> 
>> I guess the real issue is not what layer the processing is done at, but rather
>> if it is done by a general-purpose processor or by a more specialized processor.
>> Would the call-center scenario you described above use general-purpose processors
>> to do the transcoding or would more specialized hardware be deployed?
>
>This has something to do with the way layered protocols are implemented
>and optimized. The lower you are in the protocol stack, the more you
>will be willing to pay for the extra effort doing optimization, because
>the lower layers tend to be more heavily shared in most systems. The
>upper layers are usually more application-specific and their
>optimization becomes less cost-effective in most cases.

It is important to make higher layers efficient when they consume
a large fraction of the total processing time. In the amr case, it is important
to optimize the amr codec. It is less important to optimize the packetization
process, since it consumes so little of the processing time. So although 
the above paragraph generally is true, amr over RTP is a counterexample
to the principle stated there. 

Cheers, 

Micke




From rem-conf Fri Oct 06 23:28:40 2000 
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To: Colin Perkins <csp@isi.edu>,
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From: Toshiro Kawahara <kawahara@spg.yrp.nttdocomo.co.jp>
Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision 
Cc: "philippe.gentric" <philippe.gentric@philips.com>,
        IETF AVT <rem-conf@es.net>,
        4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>
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References: <Message from Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp>
 <085201c02f85$bc4d1ce0$cc51c485@eel.rdc.toshiba.co.jp>
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Dear all,



At 10:21 00/10/06 -0400, Colin Perkins wrote:
>--> Yoshihiro Kikuchi writes:
>...
>>For detailed definition, I don't think there is no problem to change
>>profile-level-id to required parameter. Maybe we can remove the default
>>value by this. However, since config is defined that this parameter shall
>>not be used for the capability exchange, and actually config is not to
>>indicate the capability but coder configuration, there is a problem to
>>change it as required. How about changing the first sentence of config as
>>follows?
>>
>>"config: This parameter SHALL be used to indicate the configuration of the
>>corresponding MPEG-4 visual bitstream. It SHALL NOT be used to indicate the
>>codec capability in the capability exchange procedure. ......"
>>
>>Since the parameter config is added by requests to keep interlocking with
>>32x-based systems, I would like to hear if this change matches the request.
>>I would also like to hear from IETF experts if it is usual in IETF world to
>>add this kind of parameter as required.

I'm one of the proponenst of this parameter, and this 
change matches our intention. 3G-324M, which is a 3GPP
version of H.324/M terminal specification, mandates the
equivalent parameter in OpenLogicalChannel signaling, and
to make the interworking with this terminal, this change
is really beneficial. I strongly support this idea. 

>RTP payload formats typically don't specify how the negotiation is done 
>in so much detail, rather they leave it up to the definition of the setup
>protocol (e.g. H.323 may not use this, but what if a SIP or RTSP based
>system want's to?)

I understand your concern. The mandatory/optional definition
should be in a proper place to avoid inconsistency between
specifications or not to limit the function of other protocol
unnecessarily. However, I believe the proposed revised 
definition is not such kind of definition. It don't define 
the way of negotiation, but provides a clarification of 
the parameter (for what purpose it SHALL or SHALL NOT be 
used). "SHALL/SHALL NOT" type of definition might look too
strong for such generic definition, but I believe this 
is a straightforward translation of the situation.

Regards,
Toshiro






---
   Toshiro Kawahara  <kawahara@spg.yrp.nttdocomo.co.jp>
   NTT DoCoMo, Inc.
   Multimedia Laboratories, Multimedia Signal Processing Laboratory
   TEL: +81 468 40 3518  FAX: +81 468 40 3788
   $B2O86(B $BIRO/(B (in Japanese)



From rem-conf Sat Oct 07 04:26:05 2000 
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From: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
To: "Colin Perkins" <csp@isi.edu>,
        "Toshiro Kawahara" <kawahara@spg.yrp.nttdocomo.co.jp>
Cc: "philippe.gentric" <philippe.gentric@philips.com>,
        "IETF AVT" <rem-conf@es.net>,
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Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision 
Date: Sat, 7 Oct 2000 20:23:08 +0900
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Dear Colin, Kawahara-san, Philippe, and all,

> >>Since the parameter config is added by requests to keep interlocking
with
> >>32x-based systems, I would like to hear if this change matches the
request.
> >>I would also like to hear from IETF experts if it is usual in IETF world
to
> >>add this kind of parameter as required.
>
> I'm one of the proponenst of this parameter, and this
> change matches our intention. 3G-324M, which is a 3GPP
> version of H.324/M terminal specification, mandates the
> equivalent parameter in OpenLogicalChannel signaling, and
> to make the interworking with this terminal, this change
> is really beneficial. I strongly support this idea.
>
I'm happy to hear that it meets your requirement.

And I hope it also matches the desire of Philippe bellow because the
definition explicitly says that config SHALL be used to indicate the codec
configuration.

> Well what I would like is to be sure when I get a SDP
> file to be able to correctly initialize my decoders

> >RTP payload formats typically don't specify how the negotiation is done
> >in so much detail, rather they leave it up to the definition of the setup
> >protocol (e.g. H.323 may not use this, but what if a SIP or RTSP based
> >system want's to?)
>
> I understand your concern. The mandatory/optional definition
> should be in a proper place to avoid inconsistency between
> specifications or not to limit the function of other protocol
> unnecessarily. However, I believe the proposed revised
> definition is not such kind of definition. It don't define
> the way of negotiation, but provides a clarification of
> the parameter (for what purpose it SHALL or SHALL NOT be
> used). "SHALL/SHALL NOT" type of definition might look too
> strong for such generic definition, but I believe this
> is a straightforward translation of the situation.
>
I completely agree with this.

Maybe it is not appropriate to have a description in the payload format
document about one specific framework, such as how to use OpenLogicalChannel
of H.245. However, I think indicating capable codec and payload format at an
negotiation stage is more general not only for H.323. For example, OPTIONS
or INVITE is used to exchange the capability in SIP.

I tried to describe the definition as general as possible, but I appreciate
suggestions for better wording. Anyway, I think it is better that the
definition will be not too much concise so that we don't leave ambiguity.

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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From: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
To: "Colin Perkins" <csp@isi.edu>
Cc: "philippe.gentric" <philippe.gentric@philips.com>,
        "IETF AVT" <rem-conf@es.net>,
        "4on2andIP-sys" <4on2andIP-sys@advent.ee.columbia.edu>,
        "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
References: <200010061421.KAA01338@purple.east.isi.edu>
Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision 
Date: Sat, 7 Oct 2000 20:22:05 +0900
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Dear Colin,

> >For detailed definition, I don't think there is no problem to change
> >profile-level-id to required parameter. Maybe we can remove the default
> >value by this. However, since config is defined that this parameter shall
> >not be used for the capability exchange, and actually config is not to
> >indicate the capability but coder configuration, there is a problem to
> >change it as required. How about changing the first sentence of config as
> >follows?
> >
> >"config: This parameter SHALL be used to indicate the configuration of
the
> >corresponding MPEG-4 visual bitstream. It SHALL NOT be used to indicate
the
> >codec capability in the capability exchange procedure. ......"
> >
> >Since the parameter config is added by requests to keep interlocking with
> >32x-based systems, I would like to hear if this change matches the
request.
> >I would also like to hear from IETF experts if it is usual in IETF world
to
> >add this kind of parameter as required.
>
> RTP payload formats typically don't specify how the negotiation is done
> in so much detail, rather they leave it up to the definition of the setup
> protocol (e.g. H.323 may not use this, but what if a SIP or RTSP based
> system want's to?)
>
Thank you for your suggestion.

Am I correct that REQUIRED parameter of MIME/SDP means that it is present
every time when MIME type is used? If it is correct, we should be much care
for the selection of REQUIRED so that the unnecessary parameters are
associated to MIME
type evey time.

By this meaning, "config" is not appropriate to be REQUIRED. For example,
when indicating codec capability from client to sever at the negotiation
stage, it is enough to use MIME (sub)type (and profile@level if necessary)
indicating MPEG-4 Visual capability plus this payload format.

On the other hands, I'm afraid that putting the parameter to OPTIONAL
without any detailed definition will make misunderstanding and confusion on
the usage of the parameter.

To avoid both these two's above,  I think it is appropriate to keep "config"
as OPTIONAL, but describe the meaning and usage of the parameter as much
while keeping the generality as much as possible, although it may be a
little detailed.

Is this a proper way of the definition?

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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From: Zvi Lifshitz <zvil@csi.com>
To: "'Dave Singer'" <singer@apple.com>,
        Pierre Clement
	 <pierre.clement@thomcast.thomson-csf.com>
Cc: CURET Dominique FTRD/DMI/REN <dominique.curet@rd.francetelecom.fr>,
        rem-conf <rem-conf@es.net>,
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Subject: RE: Types of MPEG-4 streams over IP
Date: Sat, 7 Oct 2000 22:17:22 +0200
Organization: Optibase Ltd.
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[Dave Singer:]

I believe that the muxcodetable will often be small enough that a
data: URL will be usable.

[Reply:]

Recalling Jan remarks about muxcodetable, I think he was right. Since the 
table can change on the fly, affecting subsequent FlexMux PDUs, it may be 
necessary to send it in band. So the question is, do we need to have also 
an out-of-band MuxCodeTable signaling? IMO not.

Just to make sure, we are not talking about SMT, which should always be 
carried out of band, right?

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
The MPEGathon Developers Seminar,  Brussels Airport Novotel, Brussels, 
November 15th, 2000






From rem-conf Sat Oct 07 13:26:32 2000 
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From: Zvi Lifshitz <zvil@csi.com>
To: "'Dave Singer'" <singer@apple.com>,
        "jan.vandermeer@philips.com"
	 <jan.vandermeer@philips.com>
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Subject: RE: Types of MPEG-4 streams over IP
Date: Sat, 7 Oct 2000 22:25:47 +0200
Organization: Optibase Ltd.
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[Dave Singer:]

My understanding is that the multiplex nature of the stream does not
change, and that therefore reliable up-front signalling is best.  If
this ins't true, the flexmux packet format may need some way to
update the muxcodetable, but I hope that isn't true, because of
reliability concerns.

[Reply:]

The problem is, it may very well change (as it happens in H.223). 
Reliability is indeed a concern. So whether MCT is signaled in-band or 
out-of-band it seems we must use MCT version number byte in each RTP 
packet. If we have this, we may achieve in-band reliability by repeating 
the MCT every x packets (also necessary for multicast).

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
The MPEGathon Developers Seminar,  Brussels Airport Novotel, Brussels, 
November 15th, 2000






From rem-conf Sat Oct 07 13:38:29 2000 
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From: Zvi Lifshitz <zvil@csi.com>
To: "'jan.vandermeer@philips.com'" <jan.vandermeer@philips.com>
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Subject: RE: Types of MPEG-4 streams over IP
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OK, Jan. I see. So we are actually talking about either signaling the 
presence of the byte somewhere inside the SDP description, or having two 
payload types (extra to the Kikuchi-san proposal), one with one without. I 
would prefer the second, but certainly would not be orthodox about it. 
Would follow IETF gurus if they have any recommendation.

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
The MPEGathon Developers Seminar,  Brussels Airport Novotel, Brussels, 
November 15th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Friday, October 06, 2000 5:28 PM
To:	zvil@csi.com
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: Types of MPEG-4 streams over IP


Zvi,

Thanks. So we agree for at least 50 %. Now the remainder. Signaling the
presence of the length byte by an attribute on SDP level allows the byte to
be present or not. As long as you don't need the byte, don't burn the
overhead.

Regards,

Jan

-----Original Message-----
From:	zvil@csi.com [mailto:zvil@csi.com]
Sent:	vrijdag, 06 oktober, 2000 17:09
To:	csp@isi.edu; Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: Types of MPEG-4 streams over IP


Jan, I'm in favor of a length byte. In the Generic payload format. All I
said is that signaling whether the use of extra SL info (the Generic
format) or not (Kikuchi-san proposal) can be done through payload type.

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
The MPEGathon Developers Seminar,  Brussels Airport Novotel, Brussels,
November 15th, 2000


-----Original Message-----
From:	jan.vandermeer@philips.com [SMTP:jan.vandermeer@philips.com]
Sent:	Friday, October 06, 2000 4:56 PM
To:	zvil@csi.com; csp@isi.edu
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: Types of MPEG-4 streams over IP


Colin, Zvi, all,

Colin wrote :
>>Perhaps an a=x-mpeg4-sl-header-length:<byte count>
>>attribute on any RTP stream carrying MPEG4 content should be
>>permitted, and required if byte count is non-zero (with zero being
>>the default)?
>In which case, signal using the payload type field...

And Zvi wrote almost the same :
>Or just have a new payload type which is just a different (and more
>common) way of signaling.

There are several arguments for introducing a length byte. The main reason
to introduce such field is that it creates headroom for fully backward
compatible extensions. Specifically in two cases:
a) when applications want to add an SL header to non-MPEG-4 system services
(for reasons discussed in earlier threads);
b) when the SL header is extended in future with additional fields of a
length unknown by now.
In such cases the length byte provides a future proof solution for adding
additional information using the same payload type. Assuming of course that
the receiver is capable to parse the length byte while skipping the part of
the subsequent field that the receiver does not understand.

Using another payload type does also provide a solution, but this one is
not
backward compatible. Existing receivers will not be capable to receive the
new payload format.

Kind regards,

Jan




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From rem-conf Sun Oct 08 22:37:08 2000 
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Hi all/list administrator
Can any one take steps to stop spam in this mailing lists.
Thanks
Ranga Rao Ravuri
rsb@docsj.de wrote:
>=20
> Herbal V: An Incredible All-Natural Healthy Alternative
>=20
>   Herbal V is the All Natural Approach to Male Virility,
>   Vitality and Pleasure.
>=20
> Available N o w !
>=20
> Welcome to the New Sexual Revolution.
>=20
> It's the all natural male potency and pleasure pill that men
> everywhere are buzzing about. Herbal V is safe, natural and
> specifically formulated to help support male sexual function
> and pleasure. You just take two easy-to-swallow tablets
> one hour before sex. And there's more great news - you can
> get Herbal V for less than $1 a pill.
>=20
> Amazing word of mouth praise on Herbal V has been spreading
> like wildfire-already over 1,500,000 men  have chosen
> Herbal V. Since it is 100% natural you will never have
> to worry about safety. Try doctor-recommended Herbal V
> today and have the greatest night of your life!
>=20
> Herbal V... Bringing Back the Magic!
>=20
> 1,585,000 men can't be wrong. To date over 1 million men
> have tried the super supplement Herbal V.
> Here is why:
>=20
> No Doctor Visit Required
> Available Over the Counter
> Not a Drug
> 100% Natural
> Safe, No Worries
> Highest Quality Pharmaceutical-Grade Pure Nutriceuticals
> Guaranteed Potency & Purity
>=20
> Be a Real Man Again!
>=20
> Questions and Answers
>=20
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> Thank You.



From rem-conf Mon Oct 09 01:40:42 2000 
From rem-conf-request@es.net Mon Oct 09 01:40:41 2000
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Message-ID: <39E18160.CFEEC12F@lmf.ericsson.se>
Date: Mon, 09 Oct 2000 11:27:12 +0300
From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
Organization: Oy L M Ericsson Ab
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Hi,

The draft
http://search.ietf.org/internet-drafts/draft-camarillo-sip-sdp-00.txt
was presented in the last IETF in the MMUSIC working group meeting.

It describe certain situations when it might be desirable to send a
single RTP flow to different port numbers depending on the codec used.

For instance, I begin sending PCM u-law to the remote port number 20000.
Then, I change codec and I begin sending GSM. I send the RTP now to port
20002 instead.

http://www2.ietf.org/proceedings/00jul/slides/mmusic-sdp-media-alignment.ppt
contains some slides showing scenarios where this is useful.

For instance, if I have two RTP libraries developed by third parties I
will install the one supporting PCM in port number 20000 and the one
supporting GSM in port number 20002. When the remote party sends me RTP,
I want them to send the RTP to different ports depending on the codec
they use at every moment.

When celullar access is used, it is also necessary to configure radio
bearers depending on the codec used.

Besides these two, there are many other scenarios where this is needed.

People in MMUSIC and SIP WGs think that there might be problems with
some RTP implementations if RTP packets are sent to different ports in
the middle of a session. RTCP is also a concern.

Could you please help indentify all the possible issues?

Thanks a lot,

Gonzalo
-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo



From rem-conf Mon Oct 09 03:50:35 2000 
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To: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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Subject: Re: RTP sent to different ports
In-reply-to: Your message of "Mon, 09 Oct 2000 11:27:12 +0300." <39E18160.CFEEC12F@lmf.ericsson.se>
Date: Mon, 09 Oct 2000 11:46:31 +0100
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<39E18160.CFEEC12F@lmf.ericsson.se>Gonzalo Camarillo writes:
> Hi,
> 
> The draft
> http://search.ietf.org/internet-drafts/draft-camarillo-sip-sdp-00.txt
> was presented in the last IETF in the MMUSIC working group meeting.
> 
> It describe certain situations when it might be desirable to send a
> single RTP flow to different port numbers depending on the codec used.
> 
> For instance, I begin sending PCM u-law to the remote port number 20000.
> Then, I change codec and I begin sending GSM. I send the RTP now to port
> 20002 instead.
> 
> http://www2.ietf.org/proceedings/00jul/slides/mmusic-sdp-media-alignment.ppt
> contains some slides showing scenarios where this is useful.
> 
> For instance, if I have two RTP libraries developed by third parties I
> will install the one supporting PCM in port number 20000 and the one
> supporting GSM in port number 20002. When the remote party sends me RTP,
> I want them to send the RTP to different ports depending on the codec
> they use at every moment.

This doesn't read like a very plausible example.  A good library will
separate packet processing from media packetization and both from
media decoding, ie apple quicktime, the lucent stack, the ucl stack.
Payload identifiers suffices.  It'd be difficult to engineer a
properly functioning RTCP component in such a system.

> When celullar access is used, it is also necessary to configure radio
> bearers depending on the codec used.
 
Is the intent to hash the TFT fields into some type of pre-configured
mobile channels because there is latency associated with the
configuration?  Is the TFT spec set in stone so there is no way
payload inspectino could be used?

If this is not the case then there are several options compatible with
the existing RTP base:

1) At the present time it's possible to have an educated guess at the
content of an RTP audio packet using knowledge of codec frame sizes
and interpacket gaps.  The worst case scenario is a few packets that
do not enjoy the benefits of codec-specific FEC / concealment at the
link layer at the start of a transmission.  In the long run this may
not be a reliable strategy as more codecs and data-rates come into
use.

2) Depending on the relative costs another approach would be to
inspect the payload of the first packet, and cache (src,dst,packet
size).  When the packet size changes or the cache state times out
another packet inspection is required.

3) It seems obvious, but if there is any form of RTP header compression,
then it should be straightforward to the payload information.

> Besides these two, there are many other scenarios where this is needed.
>
> People in MMUSIC and SIP WGs think that there might be problems with
> some RTP implementations if RTP packets are sent to different ports in
> the middle of a session. RTCP is also a concern.

It would be interesting to know what the cost of inspecting the RTP
payload on per packet basis would be and what the latencies are for
configuring the channels.

Kind Regards
- Orion
 



From rem-conf Mon Oct 09 04:44:26 2000 
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From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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CC: rem-conf@es.net
Subject: Spam filter [Was: Re: At last, HERBAL V the all natural alternative!]
References: <200010090432.MAA11941@www.chinaren-inc.com> <39E156EF.5E25766F@netlab.hcltech.com>
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Ranga Rao Ravuri wrote:
> 
> Hi all/list administrator
> Can any one take steps to stop spam in this mailing lists.
> Thanks

I have not seen evidence that there is a list administrator for this
list, so I doubt that your request will have much effect. Your request
has been made numerous times, to no avail. It might be worthwhile
exploring moving this list to a host that has the ability and interest
to provide list maintenance and spam filtering.


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



From rem-conf Mon Oct 09 04:50:46 2000 
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If data is going to two ports, we have two different sessions. That's
what we have SIP (and SAP, RTSP, etc.) for, to allow mid-session changes
of transport destinations. For example, SIP re-INVITE will handle this
just fine. Among other things, adding this complexity will make
firewalls much more difficult.
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



From rem-conf Mon Oct 09 06:18:11 2000 
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To: <khuber@sorensonlabs.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 Kris,

You wrote:
>I'd appreciate clarification of user_data usage conventions by anyone,=

>please (no, no, don't just say it shouldn't be used!  I know it's mess=
y and
>goes against certain design principles!)

Well, my MPEG history started with a very memorable meeting in Kurihama=
,
back in November '89. By that time, user data was not invented yet. I a=
m not
sure who introduced user data, but it already exists in MPEG-1 video. A=
s far
as I know, it is used (if at all) for associating certain data to a vid=
eo
frame, e.g. closed captioning or subtitles, in particular in ATSC. Thou=
gh it
can be used for multiplexing with any other data (such as audio), I am =
not
aware of any such usage. User data could be the solution for people who=

don't like MPEG systems, but I spent some very enjoyable years there, s=
o I
really love MPEG systems !

Coming back to your suggestion, it may in principle work in cases where=

video is used. In other cases you need to initiate a video session with=
 only
user data. May work too. However, to accommodate different RTP packet
"frequencies" several inventions have to be made to associate the user =
data
to the correct SL header. As you say,  very messy indeed; just by think=
ing
of such solution I already get a headache. And all of that to avoid the=

simple solution of flagging a length byte by an SDP attribute. No, I th=
ink
dirty solutions should only be considered if there are no clean ones.

Kind regards,

Jan




-----Original Message-----
From:	khuber@sorensonlabs.com [mailto:khuber@sorensonlabs.com]
Sent:	vrijdag, 06 oktober, 2000 18:44
To:	csp@isi.edu; zvil@csi.com; Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: Types of MPEG-4 streams over IP


Hello Jan,

For devices that (for whatever reason) must use the Kikuchi-san payload=

type, and never upgrade to a new payload type, then perhaps a possibili=
ty
you haven't considered is placing future configuration information in t=
he
user_data field of the visual streams.  This field is present in the vi=
sual
object sequence (VOS) header, the visual object (VO) header, the visual=

object layer (VOL) header, and in the group of VOP (GOV) header.  It is=
 not
in the VOP header, so to accommodate a decoding timestamp, for example,=

you'd need to send a list of DTS's in the GOV header user data, for the=
 VOPs
that follow.

OK it's a messy and ugly solution (especially if you put audio-related
information in there)!  But it's already carried in MPEG-4 and
(unwittingly?) in the Kikuchi-san draft payload format.  Perhaps MPEG-4=

user_data was accepted with "configuration future-proofing" in mind (I
became involved with MPEG after it was already there, so I don't know t=
he
history).

As for using user_data, Annex D of the systems spec. gives a registrati=
on
procedure for "the private data format identifiers" and "the IPMP syste=
m
type values".  I'm not sure if user_data is included in the meaning of
"private data format", however, or if you just use it and build decoder=
s
that try to figure out what portion of it is yours.  As for contacting =
the
registration authority, D.3 says "To Be Determined".  Whatever the
organization, it is to report its activities to JTC 1, ITTF (probably
"Internet Theory Task Force") and SC 29 secretariat.

Note that since visual information consumes much more bandwidth than au=
dio,
there is not so much wasted bandwidth associated with duplication of au=
dio
broadcast services due to a need to support an old payload format as th=
ere
is when duplication of video broadcast service is required to prevent d=
evice
obsolescence.  Of course the whole analysis of bandwidth wastage depend=
s on
numbers of non-upgradeable devices, but I understand there are applicat=
ions
that would include a large number of MPEG-2 devices in the count, so de=
vice
or bandwidth wastage without information backward-compatibility could b=
e
significant.

I'd appreciate clarification of user_data usage conventions by anyone,
please (no, no, don't just say it shouldn't be used!  I know it's messy=
 and
goes against certain design principles!)  But I think it makes some klu=
dges
possible that could prove very valuable in some cases.  Not everything =
of
value is pretty.  And note that the burden of parsing and using the
user_data fields is primarily on those who wish to use it.  The burden =
on
everyone else is to carry and skip it.  If it prevents needless replica=
tion
of high-bandwidth services for a significant period of time, it is a sm=
all
price to pay.  Of course if it gets too large, it is a more significant=

price to pay in bandwidth, but that is certainly not a problem at the
present time.

Regards,
Kris

-----Original Message-----
From: jan.vandermeer@philips.com [mailto:jan.vandermeer@philips.com]
Sent: Friday, October 06, 2000 8:56 AM
To: zvil@csi.com; csp@isi.edu
Cc: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: Types of MPEG-4 streams over IP


Colin, Zvi, all,

Colin wrote :
>>Perhaps an a=3Dx-mpeg4-sl-header-length:<byte count>
>>attribute on any RTP stream carrying MPEG4 content should be
>>permitted, and required if byte count is non-zero (with zero being
>>the default)?
>In which case, signal using the payload type field...

And Zvi wrote almost the same :
>Or just have a new payload type which is just a different (and more
>common) way of signaling.

There are several arguments for introducing a length byte. The main rea=
son
to introduce such field is that it creates headroom for fully backward
compatible extensions. Specifically in two cases:
a) when applications want to add an SL header to non-MPEG-4 system serv=
ices
(for reasons discussed in earlier threads);
b) when the SL header is extended in future with additional fields of a=

length unknown by now.
In such cases the length byte provides a future proof solution for addi=
ng
additional information using the same payload type. Assuming of course =
that
the receiver is capable to parse the length byte while skipping the par=
t of
the subsequent field that the receiver does not understand.

Using another payload type does also provide a solution, but this one i=
s not
backward compatible. Existing receivers will not be capable to receive =
the
new payload format.

Kind regards,

Jan

=



From rem-conf Mon Oct 09 08:00:33 2000 
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From: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
To: "Colin Perkins" <csp@isi.edu>, "Stephan Wenger" <stewe@cs.tu-berlin.de>
Cc: <Qunshan_Gu@pictel.com>, "Stephan Wenger" <stewe@cs.tu-berlin.de>,
        "IETF AVT" <rem-conf@es.net>,
        "4on2andIP-sys" <4on2andIP-sys@advent.ee.columbia.edu>,
        "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
References: <Message from Qunshan_Gu@pictel.com  of "Thu, 05 Oct 2000 10:54:18 EDT." <OFA5AA56DA.713ED2B3-ON8525696F.0051082C@PicTel.com> <5.0.0.25.2.20001005232618.02ebb6d0@mail.cs.tu-berlin.de>
Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision 
Date: Mon, 9 Oct 2000 23:58:50 +0900
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Dear all,

>
> >What about...
> >         "When the short video header mode is used, the RTP payload
format
> >         for H.263 SHOULD be used (the format defined in RFC 2429 is
> >         RECOMMENDED, but the RFC 2190 format MAY be used for
compatibility
> >         with older implementations)."
>
> Fine with me.
>
> Stephan
>
>
I will incorporate the sentence above to the last text of the
Internet-Draft.

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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From rem-conf Mon Oct 09 08:16:31 2000 
From rem-conf-request@es.net Mon Oct 09 08:16:31 2000
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To: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
cc: rem-conf@es.net
Subject: Re: RTP sent to different ports 
In-Reply-To: Message from Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se> 
   of "Mon, 09 Oct 2000 11:27:12 +0300." <39E18160.CFEEC12F@lmf.ericsson.se> 
Date: Mon, 09 Oct 2000 11:16:41 -0400
From: Colin Perkins <csp@isi.edu>
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--> Gonzalo Camarillo writes:
>Hi,
>
>The draft
>http://search.ietf.org/internet-drafts/draft-camarillo-sip-sdp-00.txt
>was presented in the last IETF in the MMUSIC working group meeting.
>
>It describe certain situations when it might be desirable to send a
>single RTP flow to different port numbers depending on the codec used.
>
>For instance, I begin sending PCM u-law to the remote port number 20000.
>Then, I change codec and I begin sending GSM. I send the RTP now to port
>20002 instead.
>
>http://www2.ietf.org/proceedings/00jul/slides/mmusic-sdp-media-alignment.ppt
>contains some slides showing scenarios where this is useful.
>
>For instance, if I have two RTP libraries developed by third parties I
>will install the one supporting PCM in port number 20000 and the one
>supporting GSM in port number 20002. When the remote party sends me RTP,
>I want them to send the RTP to different ports depending on the codec
>they use at every moment.

This seems like a mistake, and poor application design. It also makes the
job of firewalls even harder.

>When celullar access is used, it is also necessary to configure radio
>bearers depending on the codec used.

Which would seem to be a job for the call setup protocol, hence signal two 
RTP sessions (rather than trying to split a single session in an odd manner).

>Besides these two, there are many other scenarios where this is needed.
>
>People in MMUSIC and SIP WGs think that there might be problems with
>some RTP implementations if RTP packets are sent to different ports in
>the middle of a session. RTCP is also a concern.
>
>Could you please help indentify all the possible issues?

Where does the RTCP traffic go? If you send it to two separate ports also,
then you have two RTP sessions: no protocol change needed, just signal this
in an appropriate manner. If sent to a single port, you'll likely confuse 
existing implementations which monitor traffic sent to a single port.

Colin



From rem-conf Mon Oct 09 08:18:45 2000 
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From: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
To: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>,
        "Colin Perkins" <csp@isi.edu>,
        "Toshiro Kawahara" <kawahara@spg.yrp.nttdocomo.co.jp>
Cc: "philippe.gentric" <philippe.gentric@philips.com>,
        "IETF AVT" <rem-conf@es.net>,
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References: <Message from Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp><085201c02f85$bc4d1ce0$cc51c485@eel.rdc.toshiba.co.jp> <4.2.0.58.J.20001007134735.00b23af0@spg.yrp.nttdocomo.co.jp> <01b801c03050$f97fe7c0$1fc77885@ave>
Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision 
Date: Tue, 10 Oct 2000 00:18:18 +0900
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Dear all,

If there is no problem, I will reflect the following change on the MIME/SDP
parameter "config" to the last text of Internet-Draft to meet the requests.

> "config: This parameter SHALL be used to indicate the configuration of the
> corresponding MPEG-4 visual bitstream. It SHALL NOT be used to indicate
the
> codec capability in the capability exchange procedure. ......"

Best regards,
Yoshihiro

----- Original Message -----
From: Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp>
To: Colin Perkins <csp@isi.edu>; Toshiro Kawahara
<kawahara@spg.yrp.nttdocomo.co.jp>
Cc: philippe.gentric <philippe.gentric@philips.com>; IETF AVT
<rem-conf@es.net>; 4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>;
Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp>
Sent: Saturday, October 07, 2000 8:23 PM
Subject: Re: draft text for MPEG-4 Audio/Visual RTP I-D revision


> Dear Colin, Kawahara-san, Philippe, and all,
>
> > >>Since the parameter config is added by requests to keep interlocking
> with
> > >>32x-based systems, I would like to hear if this change matches the
> request.
> > >>I would also like to hear from IETF experts if it is usual in IETF
world
> to
> > >>add this kind of parameter as required.
> >
> > I'm one of the proponenst of this parameter, and this
> > change matches our intention. 3G-324M, which is a 3GPP
> > version of H.324/M terminal specification, mandates the
> > equivalent parameter in OpenLogicalChannel signaling, and
> > to make the interworking with this terminal, this change
> > is really beneficial. I strongly support this idea.
> >
> I'm happy to hear that it meets your requirement.
>
> And I hope it also matches the desire of Philippe bellow because the
> definition explicitly says that config SHALL be used to indicate the codec
> configuration.
>
> > Well what I would like is to be sure when I get a SDP
> > file to be able to correctly initialize my decoders
>
> > >RTP payload formats typically don't specify how the negotiation is done
> > >in so much detail, rather they leave it up to the definition of the
setup
> > >protocol (e.g. H.323 may not use this, but what if a SIP or RTSP based
> > >system want's to?)
> >
> > I understand your concern. The mandatory/optional definition
> > should be in a proper place to avoid inconsistency between
> > specifications or not to limit the function of other protocol
> > unnecessarily. However, I believe the proposed revised
> > definition is not such kind of definition. It don't define
> > the way of negotiation, but provides a clarification of
> > the parameter (for what purpose it SHALL or SHALL NOT be
> > used). "SHALL/SHALL NOT" type of definition might look too
> > strong for such generic definition, but I believe this
> > is a straightforward translation of the situation.
> >
> I completely agree with this.
>
> Maybe it is not appropriate to have a description in the payload format
> document about one specific framework, such as how to use
OpenLogicalChannel
> of H.245. However, I think indicating capable codec and payload format at
an
> negotiation stage is more general not only for H.323. For example, OPTIONS
> or INVITE is used to exchange the capability in SIP.
>
> I tried to describe the definition as general as possible, but I
appreciate
> suggestions for better wording. Anyway, I think it is better that the
> definition will be not too much concise so that we don't leave ambiguity.
>
> 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 Mon Oct 09 08:37:12 2000 
From rem-conf-request@es.net Mon Oct 09 08:37:12 2000
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Date: Mon, 09 Oct 2000 18:36:05 +0300
From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
Organization: Ericsson
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Hi,

> > For instance, if I have two RTP libraries developed by third parties I
> > will install the one supporting PCM in port number 20000 and the one
> > supporting GSM in port number 20002. When the remote party sends me RTP,
> > I want them to send the RTP to different ports depending on the codec
> > they use at every moment.
> 
> This doesn't read like a very plausible example.  A good library will
> separate packet processing from media packetization and both from
> media decoding, ie apple quicktime, the lucent stack, the ucl stack.
> Payload identifiers suffices.  It'd be difficult to engineer a
> properly functioning RTCP component in such a system.

Yes, that is one of the concerns people have - RTCP. Anyway, this 3rd
party developed RTP library is just one among many examples. There are
other scenarios where this feature might be useful. For instance, when
the DTMF tones have to be received in a different device than the voice.

I'd like to know if RCTP is really an unsolvable issue. I am not fully
aware of all the problems related to RCTP we might have when using two
different port numbers.
 
> > When celullar access is used, it is also necessary to configure radio
> > bearers depending on the codec used.
> 
> Is the intent to hash the TFT fields into some type of pre-configured
> mobile channels because there is latency associated with the
> configuration?  Is the TFT spec set in stone so there is no way
> payload inspectino could be used?

Well, this is another example where this mechanism might be needed. The
TFT (Traffic Flow Template) sends the incoming IP packets (IP packets
>from the network to the UMTS terminal) to the proper radio bearer (PDP
context). Since the TFT examines all the incoming IP packets - not just
RTP packets - it seems complicated to inspect the RTP payload type. All
the incoming UDP packets might have to be examined further to look for
an RTP header.

If we can send different RTP payloads to different ports the packets can
be routed to the proper radio bearer that implements the proper UEP
(Unequal Error Protection), for instance.

Anyway, I would not like to go deep into details that are 3GPP specific.
This can be seen as a general QoS problem, where the packets will get a
different treatment depending on the destination UDP port number.

> It would be interesting to know what the cost of inspecting the RTP
> payload on per packet basis would be and what the latencies are for
> configuring the channels.

I do not have exact figures, but the cost of inspecting the payload of
every IP packet entering the access network seems high.

About configuring channels, it is a too slow process to be performed in
"real-time".

Thanks for the feedback,

Best regards,

Gonzalo
-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 31 18
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland



From rem-conf Mon Oct 09 08:44:22 2000 
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Organization: Ericsson
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Subject: Re: RTP sent to different ports
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Hi,

Yes, SIP re-INVITE solves this issue when we can wait 1.5 RTT. I am more
looking into the scenario where we change dynamically the codec without
application layer signalling - SIP - involved.

In a cellular environment, for instance, handling a SIP re-INVITE might
take time, since new radio bearers have to be configured.

About the firewalls, I agree that this adds complexity.

Thanks a lot for your feedback,

Best regards,

Gonzalo

Henning Schulzrinne wrote:
> 
> If data is going to two ports, we have two different sessions. That's
> what we have SIP (and SAP, RTSP, etc.) for, to allow mid-session changes
> of transport destinations. For example, SIP re-INVITE will handle this
> just fine. Among other things, adding this complexity will make
> firewalls much more difficult.
> --
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 31 18
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland



From rem-conf Mon Oct 09 08:48:17 2000 
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To: Stephan Wenger <stewe@cs.tu-berlin.de>
cc: SIGNES Julien / ENVIVIO <julien.signes@rd.francetelecom.com>,
        "'Y. Matsui'" <matsui@drl.mei.co.jp>,
        4on2andIP-sys@advent.ee.columbia.edu, rem-conf@es.net
Subject: Re: Types of MPEG-4 streams over IP 
In-Reply-To: Message from Stephan Wenger <stewe@cs.tu-berlin.de> 
   of "Thu, 05 Oct 2000 19:53:07 +0200." <5.0.0.25.2.20001005193923.02f48eb0@mail.cs.tu-berlin.de> 
Date: Mon, 09 Oct 2000 11:48:51 -0400
From: Colin Perkins <csp@isi.edu>
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--> Stephan Wenger writes:
>Julien, Group,
>
>
>>I agree in general they should be fully reliable channels carrying
>>those,as well as BIFS comands. However, note that a lot of
>>streaming those days is actually done by RTP over TCP.
>
>This is interesting.  I'm not aware of any commercial system doing
>such a thing.  I know that telephony people discuss the use of
>RTP over TCP on managed IP networks for conversational
>applications, but that is probably not what you are referring to?
>I'm also aware of UDP/RTP streaming techniques that use a TCP
>channel for repair, (and feedback to report errors).  But again, that
>is not what I would understand as RTP over TCP.  So can you
>please come up with an example and/or references?  Thanks in
>advance.

And if you do, please let me know also - this is one of the missing entries
in the RTP interoperability matrix which is holding up the move of RTP to 
draft standard.

>Also, using TCP as the transport does not automatically render
>a reliable RTP channel, at least not under real-time constraints.
>There is still a good chance that packets that finally arrive at the
>receiver will be thrown out because they do not fall into the jitter
>tolerance window of the playout buffer.  (This is true for all
>re-transmission based protocols).  You simply can't guaranty that
>all packets arrive in time unless you have finally received all packets.

Right - which is why reliable multicast is in a separate working group,
which has very different concerns to AVT. Once again, I'd urge those
looking for a reliable transfer protocol to use either TCP or one of the
reliable multicast protocols under development in RMT. RTP is not the
right protocol for reliable transport.

Colin



From rem-conf Mon Oct 09 08:51:40 2000 
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To: jan.vandermeer@philips.com
cc: zvil <zvil@csi.com>, rem-conf <rem-conf@es.net>,
        4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>
Subject: Re: Types of MPEG-4 streams over IP 
In-Reply-To: Message from jan.vandermeer@philips.com 
   of "Fri, 06 Oct 2000 17:27:50 +0200." <0056890014651648000002L982*@MHS> 
Date: Mon, 09 Oct 2000 11:52:06 -0400
From: Colin Perkins <csp@isi.edu>
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--> jan.vandermeer@philips.com writes:
>Thanks. So we agree for at least 50 %. Now the remainder. Signaling the
>presence of the length byte by an attribute on SDP level allows the byte to
>be present or not. As long as you don't need the byte, don't burn the
>overhead.

How do you propose to do this in a manner which is not equivalent to
defining two payload formats? I think we are in need of a concrete 
proposal here, since I don't understand what you're driving towards...

Colin



From rem-conf Mon Oct 09 09:05:14 2000 
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Gonzalo Camarillo wrote:
> 

> 
> Yes, that is one of the concerns people have - RTCP. Anyway, this 3rd
> party developed RTP library is just one among many examples. There are
> other scenarios where this feature might be useful. For instance, when
> the DTMF tones have to be received in a different device than the voice.

That is solvable in a far cleaner manner by having two voice sessions,
one for regular audio and one for DTMF. The same applies to other
scenarios. To the end system, this is just like mixing several sources.
Implementing this gets you mixing and conferencing for free, so this is
a Good Thing.


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



From rem-conf Mon Oct 09 09:59:27 2000 
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From: Kris Huber <khuber@sorensonlabs.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
Date: Mon, 9 Oct 2000 10:57:09 -0600 
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Hello Jan,

> Coming back to your suggestion, it may in principle work in cases where
> video is used.

That's what I thought.

Thanks for the additional clarifications and user_data examples.  Like you,
I don't like headaches.  However, I'm afraid that sometimes the headache is
inevitable and a kludgey solution designed for a relatively short-term and
specific situation can sometimes be the less painful one.  I don't know that
this is the case in this situation, as I am not very expert in the systems
and applications of concern.  But I feel good about having the user_data
option available for this and other purposes.  Even if seldom used it is a
mechanism that preserves the MPEG-1, -2 & -4 implementer's freedom to
compensate, at least to a limited extent, for perceived deficiencies in the
specifications agreed upon by the groups (even the groups outside of MPEG).

I do wish you well in your effort to define a better solution than the
insertion of an adaptation header into user_data, which brings with it the
associated complexities and uncertainties of identifying that adaptation
header and isolating it from other user_data.

>No, I think
>dirty solutions should only be considered if there are no clean ones.

I agree.

Best regards,

Kris

-----Original Message-----
From: jan.vandermeer@philips.com [mailto:jan.vandermeer@philips.com]
Sent: Monday, October 09, 2000 7:17 AM
To: khuber@sorensonlabs.com
Cc: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: Types of MPEG-4 streams over IP


Hello Kris,

You wrote:
>I'd appreciate clarification of user_data usage conventions by anyone,
>please (no, no, don't just say it shouldn't be used!  I know it's messy and
>goes against certain design principles!)

Well, my MPEG history started with a very memorable meeting in Kurihama,
back in November '89. By that time, user data was not invented yet. I am not
sure who introduced user data, but it already exists in MPEG-1 video. As far
as I know, it is used (if at all) for associating certain data to a video
frame, e.g. closed captioning or subtitles, in particular in ATSC. Though it
can be used for multiplexing with any other data (such as audio), I am not
aware of any such usage. User data could be the solution for people who
don't like MPEG systems, but I spent some very enjoyable years there, so I
really love MPEG systems !

Coming back to your suggestion, it may in principle work in cases where
video is used. In other cases you need to initiate a video session with only
user data. May work too. However, to accommodate different RTP packet
"frequencies" several inventions have to be made to associate the user data
to the correct SL header. As you say,  very messy indeed; just by thinking
of such solution I already get a headache. And all of that to avoid the
simple solution of flagging a length byte by an SDP attribute. No, I think
dirty solutions should only be considered if there are no clean ones.

Kind regards,

Jan




-----Original Message-----
From:	khuber@sorensonlabs.com [mailto:khuber@sorensonlabs.com]
Sent:	vrijdag, 06 oktober, 2000 18:44
To:	csp@isi.edu; zvil@csi.com; Jan vanderMeer
Cc:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: Types of MPEG-4 streams over IP


Hello Jan,

For devices that (for whatever reason) must use the Kikuchi-san payload
type, and never upgrade to a new payload type, then perhaps a possibility
you haven't considered is placing future configuration information in the
user_data field of the visual streams.  This field is present in the visual
object sequence (VOS) header, the visual object (VO) header, the visual
object layer (VOL) header, and in the group of VOP (GOV) header.  It is not
in the VOP header, so to accommodate a decoding timestamp, for example,
you'd need to send a list of DTS's in the GOV header user data, for the VOPs
that follow.

OK it's a messy and ugly solution (especially if you put audio-related
information in there)!  But it's already carried in MPEG-4 and
(unwittingly?) in the Kikuchi-san draft payload format.  Perhaps MPEG-4
user_data was accepted with "configuration future-proofing" in mind (I
became involved with MPEG after it was already there, so I don't know the
history).

As for using user_data, Annex D of the systems spec. gives a registration
procedure for "the private data format identifiers" and "the IPMP system
type values".  I'm not sure if user_data is included in the meaning of
"private data format", however, or if you just use it and build decoders
that try to figure out what portion of it is yours.  As for contacting the
registration authority, D.3 says "To Be Determined".  Whatever the
organization, it is to report its activities to JTC 1, ITTF (probably
"Internet Theory Task Force") and SC 29 secretariat.

Note that since visual information consumes much more bandwidth than audio,
there is not so much wasted bandwidth associated with duplication of audio
broadcast services due to a need to support an old payload format as there
is when duplication of video broadcast service is required to prevent device
obsolescence.  Of course the whole analysis of bandwidth wastage depends on
numbers of non-upgradeable devices, but I understand there are applications
that would include a large number of MPEG-2 devices in the count, so device
or bandwidth wastage without information backward-compatibility could be
significant.

I'd appreciate clarification of user_data usage conventions by anyone,
please (no, no, don't just say it shouldn't be used!  I know it's messy and
goes against certain design principles!)  But I think it makes some kludges
possible that could prove very valuable in some cases.  Not everything of
value is pretty.  And note that the burden of parsing and using the
user_data fields is primarily on those who wish to use it.  The burden on
everyone else is to carry and skip it.  If it prevents needless replication
of high-bandwidth services for a significant period of time, it is a small
price to pay.  Of course if it gets too large, it is a more significant
price to pay in bandwidth, but that is certainly not a problem at the
present time.

Regards,
Kris

-----Original Message-----
From: jan.vandermeer@philips.com [mailto:jan.vandermeer@philips.com]
Sent: Friday, October 06, 2000 8:56 AM
To: zvil@csi.com; csp@isi.edu
Cc: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: Types of MPEG-4 streams over IP


Colin, Zvi, all,

Colin wrote :
>>Perhaps an a=x-mpeg4-sl-header-length:<byte count>
>>attribute on any RTP stream carrying MPEG4 content should be
>>permitted, and required if byte count is non-zero (with zero being
>>the default)?
>In which case, signal using the payload type field...

And Zvi wrote almost the same :
>Or just have a new payload type which is just a different (and more
>common) way of signaling.

There are several arguments for introducing a length byte. The main reason
to introduce such field is that it creates headroom for fully backward
compatible extensions. Specifically in two cases:
a) when applications want to add an SL header to non-MPEG-4 system services
(for reasons discussed in earlier threads);
b) when the SL header is extended in future with additional fields of a
length unknown by now.
In such cases the length byte provides a future proof solution for adding
additional information using the same payload type. Assuming of course that
the receiver is capable to parse the length byte while skipping the part of
the subsequent field that the receiver does not understand.

Using another payload type does also provide a solution, but this one is not
backward compatible. Existing receivers will not be capable to receive the
new payload format.

Kind regards,

Jan




From rem-conf Mon Oct 09 10:10:19 2000 
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From: Orion Hodson <O.Hodson@cs.ucl.ac.uk>
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To: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
cc: rem-conf@es.net
Subject: Re: RTP sent to different ports
In-reply-to: Your message of "Mon, 09 Oct 2000 18:36:05 +0300." <39E1E5E5.6C938FAB@lmf.ericsson.se>
Date: Mon, 09 Oct 2000 18:08:42 +0100
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<39E1E5E5.6C938FAB@lmf.ericsson.se>Gonzalo Camarillo writes:
> I'd like to know if RCTP is really an unsolvable issue. I am not fully
> aware of all the problems related to RCTP we might have when using two
> different port numbers.

Each RTP session has an associated participant database with SDES and
reporting information.  If there are two RTP ports and these are
associated with two differnt sessions, then a back door mechanism will
be necessary to share information to have RTCP perform properly,
i.e. correct reception report intervals and information conveyed.  If
there are two ports and one session, then most out-of-the-box RTP
libraries/applications don't support this currently.

> If we can send different RTP payloads to different ports the packets can
> be routed to the proper radio bearer that implements the proper UEP
> (Unequal Error Protection), for instance.

The issue of RTP header extensions should at least be considered when
using UEP because the extensions occur before the RTP data.  A
wired-wireless UEP enabled gateway will have to inspect every RTP
header to make sure it selects the right point to start the UEP
operation (or to discard those packets).  From thereon, the
incremental cost of looking at the payload at the start of UEP
processing is going to be small.

> Anyway, I would not like to go deep into details that are 3GPP specific.
> This can be seen as a general QoS problem, where the packets will get a
> different treatment depending on the destination UDP port number.

Out of general interest, how might your multi-port proposal relate to
the current AMR RTP payload proposal (also for 3GPP)?  This uses a
fixed RTP payload id and a field inside the codec payload to indicate
the frame type (and size) present, plus signalling info.  This would
require codec payload inspection to have UEP applied, unless the
gateway has an AMR specific path.

Kind Regards
- Orion.

=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
Networked Multimedia Research Group      Tel: ++(0)171 419 3704
Department of Computer Science           Fax: ++(0)171 419 1397
University College London, Gower Street, London,  WC1E 6BT,  UK     
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-




From rem-conf Mon Oct 09 10:51:33 2000 
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From: "Phil Farrar" <phil@talarian.com>
To: "Henning Schulzrinne" <schulzrinne@cs.columbia.edu>
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        "Gonzalo Camarillo" <Gonzalo.Camarillo@lmf.ericsson.se>,
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Subject: RE: RTP sent to different ports
Date: Mon, 9 Oct 2000 12:37:52 -0500
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Where is a good discussion of SIP?

Thank You,

Phil.

> -----Original Message-----
> From: hgs@cs.columbia.edu [mailto:hgs@cs.columbia.edu]On Behalf Of
> Henning Schulzrinne
> Sent: Monday, October 09, 2000 6:49 AM
> To: Gonzalo Camarillo
> Cc: rem-conf@es.net
> Subject: Re: RTP sent to different ports
> 
> 
> If data is going to two ports, we have two different sessions. That's
> what we have SIP (and SAP, RTSP, etc.) for, to allow mid-session changes
> of transport destinations. For example, SIP re-INVITE will handle this
> just fine. Among other things, adding this complexity will make
> firewalls much more difficult.
> -- 
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
> 
> 



From rem-conf Mon Oct 09 11:29:25 2000 
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Message-ID: <39E20DE1.37AC7C61@lmf.ericsson.se>
Date: Mon, 09 Oct 2000 21:26:41 +0300
From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
Organization: Ericsson
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Hi,

Check the SIP WG web page:
http://www.ietf.org/html.charters/sip-charter.html

To Subscribe: sip-request@lists.bell-labs.com 
In Body: subscribe 

Gonzalo

Phil Farrar wrote:
> 
> Where is a good discussion of SIP?
> 
> Thank You,
> 
> Phil.
> 
> > -----Original Message-----
> > From: hgs@cs.columbia.edu [mailto:hgs@cs.columbia.edu]On Behalf Of
> > Henning Schulzrinne
> > Sent: Monday, October 09, 2000 6:49 AM
> > To: Gonzalo Camarillo
> > Cc: rem-conf@es.net
> > Subject: Re: RTP sent to different ports
> >
> >
> > If data is going to two ports, we have two different sessions. That's
> > what we have SIP (and SAP, RTSP, etc.) for, to allow mid-session changes
> > of transport destinations. For example, SIP re-INVITE will handle this
> > just fine. Among other things, adding this complexity will make
> > firewalls much more difficult.
> > --
> > Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
> >
> >

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 31 18
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland



From rem-conf Mon Oct 09 11:43:01 2000 
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Phil Farrar wrote:
> 
> Where is a good discussion of SIP?

http://www.cs.columbia.edu/sip



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



From rem-conf Tue Oct 10 04:39:34 2000 
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From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
Organization: Oy L M Ericsson Ab
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Hi,

Orion Hodson wrote:

> > Anyway, I would not like to go deep into details that are 3GPP specific.
> > This can be seen as a general QoS problem, where the packets will get a
> > different treatment depending on the destination UDP port number.
> 
> Out of general interest, how might your multi-port proposal relate to
> the current AMR RTP payload proposal (also for 3GPP)?  This uses a
> fixed RTP payload id and a field inside the codec payload to indicate
> the frame type (and size) present, plus signalling info.  This would
> require codec payload inspection to have UEP applied, unless the
> gateway has an AMR specific path.

I think this is too specific to 3GPP, but here is the reason:

The TFT is implemented in the GGSN. The TFT includes UDP port number,
but does not include RTP payload type. Therefore, the GGSN will choose a
GTP tunnel based on the destination UDP port number of the packet, not
on the RTP payload type. The GTP tunnel is related to a specific radio
bearer. Therefore, this packet has been asigned already a readio bearer.

The GTP tunnel will go to the SGSN. The SGSN need to add the Iu header
to the packet. This header contains UEP info. Then, the RNS will perform
the actual UEP based on this info and any kind of header compression.

Thus, it does not really matter to us whether the SGSN has to examine
the payload or not. The GGSN has assigned previously the radio bearer.
Actually, the GTP tunnel has been configured from the terminal with the
codec information.

Anyway, establishing several media streams solves the problem.

Best regards,

Thanks,

Gonzalo


> 
> Kind Regards
> - Orion.
> 
> =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
> Networked Multimedia Research Group      Tel: ++(0)171 419 3704
> Department of Computer Science           Fax: ++(0)171 419 1397
> University College London, Gower Street, London,  WC1E 6BT,  UK
> -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo



From rem-conf Tue Oct 10 04:50:35 2000 
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Subject: Re: RTP sent to different ports
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Hi,

Henning Schulzrinne wrote:
> 
> Gonzalo Camarillo wrote:
> >
> 
> >
> > Yes, that is one of the concerns people have - RTCP. Anyway, this 3rd
> > party developed RTP library is just one among many examples. There are
> > other scenarios where this feature might be useful. For instance, when
> > the DTMF tones have to be received in a different device than the voice.
> 
> That is solvable in a far cleaner manner by having two voice sessions,
> one for regular audio and one for DTMF. The same applies to other
> scenarios. To the end system, this is just like mixing several sources.
> Implementing this gets you mixing and conferencing for free, so this is
> a Good Thing.

I am happy with this solution. However, I would like to signal it
properly to the other end.

That is, I receive the following description (in an INVITE for
instance).

         v=0
         o=John 289085535 289085535 IN IP4 first.example.com
         t=0 0
         c=IN IP4 111.111.111.111
         m=audio 20000 RTP/AVP 0
         m=audio 20002 RTP/AVP 8

This means that I will receive two media streams. The first one on port
number 20000 will use PCM u-law. The second one will arrive on port
number 20002 and will be using PCM A-law.
Actually I might receive media from both media streams simultaneously.
One of the media streams might contain the voice of the singer and the
other the background (public, guitars, ...) of the concert.


I would like to differentiate this case from the following one.
Now I want to have a conversation with somebody. I want just to transmit
my voice. However, sometimes I will use PCM u-law and others PCM A-law.
The receiver wants to receive (due to any reason) different codecs on
different port numbers.

As we have discussed previously in this thread, we will establish two
media streams. But I want the receiver to know that I will never
transmit simultaneouly in both media streams.

I will transmit using one OR the other.

I would like my session description to look like:

         v=0
         o=John 289085535 289085535 IN IP4 first.example.com
         t=0 0
         c=IN IP4 111.111.111.111
         m=audio 20000 RTP/AVP 0
         a=fid:1
         m=audio 20002 RTP/AVP 8
         a=fid:1

If an implementation does not understand the fid attribute it will try
to mix the audio streams. Since one is "empty" all the time, doing this
is not really a big deal. Thus, backwards compatibility is not a
concern.

Do you have any concerns with this approach?

RTP and RTCP remain untouched.

Actually, this discussion began being very much RTP related, but now we
are moving to MMUSIC or SIP probably...

Best regards,

Gonzalo
-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo



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Date: Tue, 10 Oct 2000 14:22:31 +0200
From: Franceschini Guido <Guido.Franceschini@CSELT.IT>
Subject: RE: what Mime type for MP4 files?
To: Mikael Bourges-Sevenier <mikael@ivast.com>, 'Dave Singer' <singer@apple.com>
Cc: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
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I understand the concerns against 'application', but I mantain that 'video'
is not appropriate either.
IMHO we should select a name which is neutral to both audio and video, since
MPEG-4 Systems is neutral.
If 'application' is not good, what about a new one ('multimedia') ?

Guido

> ----------
> From: 	Dave Singer[SMTP:singer@apple.com]
> Sent: 	Friday, October 06, 2000 2:49 AM
> To: 	Mikael Bourges-Sevenier
> Cc: 	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
> Subject: 	RE: what Mime type for MP4 files?
> 
> At 4:41 PM -0700 10/5/00, Mikael Bourges-Sevenier wrote:
> >Even if historically MPEG is more concerned with video, MPEG-4 deals with
> so
> >many different kinds of streams that I would prefer application/x-mp4 or
> >x-mpeg4.
> >In general as the BIFS stream defines a scene I would view it as a
> special
> >kind of application.
> 
> The problems are that many people see "application" as a potential 
> threat to their security, and try to block it.  Another problem is 
> that the mime type is supposed to indicate the kind of behavior at 
> the client:  video indicates a time-based visual presentation, 
> whereas application can be (and often is) anything at all.
> 
> The behavior of Mpeg4 is basically intended to be visual, unless an 
> audio-only session is wanted (whereupon the mime type audio/mpeg4 
> should be used).  In the view of this expert on mime naming, visual 
> is much to be preferred (and I agree).
> 
> By the way, we are asking for an official registration, and therefore 
> there will be no "x-" at all.
> 
> >Mikael Bourges-Sevenier
> >iVAST Inc.
> >
> >>  -----Original Message-----
> >>  From: owner-4on2andIP-sys@advent.ee.columbia.edu
> >>  [mailto:owner-4on2andIP-sys@advent.ee.columbia.edu]On Behalf Of Dave
> >>  Singer
> >>  Sent: Thursday, October 05, 2000 3:56 PM
> >>  To: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
> >>  Subject: what Mime type for MP4 files?
> >>
> >>
> >>  a reply from the gurus at the IETF about the (e.g. HTTP) mime type
> >>  for MP4 files...
> >>
> >>  >The top-level questions for MPEG are:
> >>  >a) do we belong under video, or application, for general MPEG-4
> >>  >sessions?
> >>
> >>  I strongly prefer video.
> >>
> >>  >I prefer video, as it warns the terminal that a
> >>  >presentation is coming, it's parallel to MPEG, and I'm concerned that
> >>  >"application" may be treated more circumspectly by security software.
> >>  >But the Beijing MPEG meeting preferred application.
> >>
> >>  Without knowing why I cannot comment, but historically such formats
> have
> >>  gone under video. I expect there to be pushback if application is
> >>  attempted.
> >>
> >>  >b) A taste question.  Is the /mpeg4 to match the name of the standard
> >>  >(my preference, along with Steve Casner), or /mp4 to match the file
> >>  >format name?
> >>
> >>  I personally prefer the former, but as you say, it is solely a matter
> of
> >>  taste.
> >>
> >>  >c) I'm fairly sure that we should say a lot more about security, as
> >>  >we allow embedded URLs, and have a Java layer that can deliver
> >>  >"applets".  What sort of analysis is needed here?
> >>
> >>  Basically, you need to note the issues. If I were you I'd take a look
> >>  at existing registrations of similar things and add what you need.
> >  > --
> >  > David Singer
> >  > Apple Computer/QuickTime
> >  >
> 
> -- 
> David Singer
> Apple Computer/QuickTime
> 



From rem-conf Tue Oct 10 06:10:11 2000 
From rem-conf-request@es.net Tue Oct 10 06:10:10 2000
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From: Zvi Lifshitz <ZviL@optibase.com>
To: Franceschini Guido <Guido.Franceschini@cselt.it>, 
	Mikael Bourges-Sevenier <mikael@ivast.com>, 'Dave Singer'
	 <singer@apple.com>
Cc: rem-conf@es.net, 4on2andIP-sys@advent.ee.columbia.edu
Subject: RE: what Mime type for MP4 files?
Date: Tue, 10 Oct 2000 15:08:43 +0200
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Guido, when talking about mime types you need to turn the switch in your
MPEG head. "video" in mime types means an audio/visual presentation, and
that's what you do 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.				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
The MPEGathon Developers Seminar,  Brussels Airport Novotel, Brussels,
November 15th, 2000


-----Original Message-----
From:	Franceschini Guido [SMTP:Guido.Franceschini@cselt.it]
Sent:	Tuesday, October 10, 2000 2:23 PM
To:	Mikael Bourges-Sevenier; 'Dave Singer'
Cc:	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
Subject:	RE: what Mime type for MP4 files?

 
I understand the concerns against 'application', but I mantain that 'video'
is not appropriate either.
IMHO we should select a name which is neutral to both audio and video, since
MPEG-4 Systems is neutral.
If 'application' is not good, what about a new one ('multimedia') ?

Guido

> ----------
> From: 	Dave Singer[SMTP:singer@apple.com]
> Sent: 	Friday, October 06, 2000 2:49 AM
> To: 	Mikael Bourges-Sevenier
> Cc: 	rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
> Subject: 	RE: what Mime type for MP4 files?
> 
> At 4:41 PM -0700 10/5/00, Mikael Bourges-Sevenier wrote:
> >Even if historically MPEG is more concerned with video, MPEG-4 deals with
> so
> >many different kinds of streams that I would prefer application/x-mp4 or
> >x-mpeg4.
> >In general as the BIFS stream defines a scene I would view it as a
> special
> >kind of application.
> 
> The problems are that many people see "application" as a potential 
> threat to their security, and try to block it.  Another problem is 
> that the mime type is supposed to indicate the kind of behavior at 
> the client:  video indicates a time-based visual presentation, 
> whereas application can be (and often is) anything at all.
> 
> The behavior of Mpeg4 is basically intended to be visual, unless an 
> audio-only session is wanted (whereupon the mime type audio/mpeg4 
> should be used).  In the view of this expert on mime naming, visual 
> is much to be preferred (and I agree).
> 
> By the way, we are asking for an official registration, and therefore 
> there will be no "x-" at all.
> 
> >Mikael Bourges-Sevenier
> >iVAST Inc.
> >
> >>  -----Original Message-----
> >>  From: owner-4on2andIP-sys@advent.ee.columbia.edu
> >>  [mailto:owner-4on2andIP-sys@advent.ee.columbia.edu]On Behalf Of Dave
> >>  Singer
> >>  Sent: Thursday, October 05, 2000 3:56 PM
> >>  To: rem-conf@es.net; 4on2andIP-sys@advent.ee.columbia.edu
> >>  Subject: what Mime type for MP4 files?
> >>
> >>
> >>  a reply from the gurus at the IETF about the (e.g. HTTP) mime type
> >>  for MP4 files...
> >>
> >>  >The top-level questions for MPEG are:
> >>  >a) do we belong under video, or application, for general MPEG-4
> >>  >sessions?
> >>
> >>  I strongly prefer video.
> >>
> >>  >I prefer video, as it warns the terminal that a
> >>  >presentation is coming, it's parallel to MPEG, and I'm concerned that
> >>  >"application" may be treated more circumspectly by security software.
> >>  >But the Beijing MPEG meeting preferred application.
> >>
> >>  Without knowing why I cannot comment, but historically such formats
> have
> >>  gone under video. I expect there to be pushback if application is
> >>  attempted.
> >>
> >>  >b) A taste question.  Is the /mpeg4 to match the name of the standard
> >>  >(my preference, along with Steve Casner), or /mp4 to match the file
> >>  >format name?
> >>
> >>  I personally prefer the former, but as you say, it is solely a matter
> of
> >>  taste.
> >>
> >>  >c) I'm fairly sure that we should say a lot more about security, as
> >>  >we allow embedded URLs, and have a Java layer that can deliver
> >>  >"applets".  What sort of analysis is needed here?
> >>
> >>  Basically, you need to note the issues. If I were you I'd take a look
> >>  at existing registrations of similar things and add what you need.
> >  > --
> >  > David Singer
> >  > Apple Computer/QuickTime
> >  >
> 
> -- 
> David Singer
> Apple Computer/QuickTime
> 



From rem-conf Tue Oct 10 06:39:15 2000 
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To: Franceschini Guido <Guido.Franceschini@cselt.it>
cc: Mikael Bourges-Sevenier <mikael@ivast.com>,
        "'Dave Singer'" <singer@apple.com>, rem-conf@es.net,
        4on2andIP-sys@advent.ee.columbia.edu, mankin@isi.edu
Subject: Re: what Mime type for MP4 files? 
In-Reply-To: Message from Franceschini Guido <Guido.Franceschini@cselt.it> 
   of "Tue, 10 Oct 2000 14:22:31 +0200." <85077463E51D6A498C453073E167794039BD90@exc2k02.cselt.it> 
Date: Tue, 10 Oct 2000 09:38:49 -0400
From: Colin Perkins <csp@isi.edu>
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--> Franceschini Guido writes:
>I understand the concerns against 'application', but I mantain that
>'video' is not appropriate either.  IMHO we should select a name which is
>neutral to both audio and video, since MPEG-4 Systems is neutral.  If
>'application' is not good, what about a new one ('multimedia') ?

A new top level "multimedia" will not be assigned. The existing approach of
registering both video/mp4 and audio/mp4 would seem to be appropriate.

Colin



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	, CURET Dominique FTRD/DMI/REN <dominique.curet@rd.francetelecom.fr>
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Dear Zvi, Dave and Jan,


As far as the dynamic nature of the flexmux PDU and consequently, of the
MuxCodeTable,  is concerned,  I guess there is nothing that precludes a
change on the fly.

Nevertheless, and from an implementation point of view,  it is more
manageable to stream flexmux whose multiplex nature does not change on the
fly and to announce the flexmux configuration by up-front signalling. For me,
this up-front signalling should me made of 1/ the Stream Map Table (i.e,
which ES_ID(s) in which flexmux channel(s)) and 2/ the MuxCodeTable, to tell
the deflexmuxer where to look at in the flexmux PDU to pickup in the right
place the right FMC.

Let's take the case of a unicast scenario where a player requires an
additionnal ES which is not yet present in any of the incoming flexmux
streams. That case implies to send 1/ an updaded version of  the SMT so that
the deflexmuxer knows where to look at to retrieve the new ES (i.e, in which
flexmux channel), and 2/ an updated version of the MuxCodeTable (i.e, where
to look at the FMC  in the flexmux PDU).

For this unicast scenario, both the SMT and the MuxCodeTable update may be
sent thanks to an out-of band mean, like a SDP announcement in a RTSP
message.

For a multicast scenario, SAP/SDP may be used to send the upgrades of the SMT
and/or MuxCodeTable, which is slow (see Colin's remark, one announcement
every tens of minutes). But then, as Zvi said, SMT needs still to be  sent by
out-of band mean.
In case in band signalling  be used for the MuxCodeTable, that implies to
strengthen reliability by cyclic retransmission, which would be detrimental
for bandwidth. Furthermore and for an application where both the SMT and the
MuxCodeTable would change, one should be carefull so that the upgraded SMT
arrives at the deflexmuxer before the MuxCodeTable Update takes place in the
flexmux stream.


Pierre


> [Dave Singer:]

My understanding is that the multiplex nature of the stream does not
change, and that therefore reliable up-front signalling is best.  If
this ins't true, the flexmux packet format may need some way to
update the muxcodetable, but I hope that isn't true, because of
reliability concerns.

>
>
> [Zvi:]
>
> Recalling Jan remarks about muxcodetable, I think he was right. Since the
> table can change on the fly, affecting subsequent FlexMux PDUs, it may be
> necessary to send it in band. So the question is, do we need to have also
> an out-of-band MuxCodeTable signaling? IMO not.
>
> Just to make sure, we are not talking about SMT, which should always be
> carried out of band, right?
>




From rem-conf Tue Oct 10 07:55:35 2000 
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From: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
To: "IETF AVT" <rem-conf@es.net>
Cc: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
Subject: items to be reflected on the last I-D
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Dear all,

Thank you very much for the comments on the MPEG-4 Audio/Visual RTP
Internet-Draft.

The following is the items to be reflected on the last text of the
Internet-Draft that the authors are considering to reflect comments raised.

1) RTP payload format for the short video header. (section 3)
    "When the short video header mode is used, the RTP payload format
    for H.263 SHOULD be used (the format defined in RFC 2429 is
    RECOMMENDED, but the RFC 2190 format MAY be used for compatibility
    with older implementations)."

2) video MIME/SDP parameter "config" (section 5.1)
     "config: This parameter SHALL be used to indicate the configuration
    of the corresponding MPEG-4 visual bitstream. It SHALL NOT be
    used to indicate the codec capability in the capability exchange
    procedure.
    ....................."

3) video timestamp (section 3.1)
The authors got a suggestion to change video timestamp definition from
"composition time" to an ordinary RTP timestamp definition (capture time).
This change is necessary to align with other RTP payload formats than MPEG-4
video. Note that audio timestamp has already be changed to "capture
time". The following is a definition the authors are considering:
    "Timestamp: The timestamp indicates the sampling instance
    of the VOP contained in the RTP packet. A constant offset,
    which is random, is added for security reasons.
   - When multiple VOPs are carried in the same RTP packet, the timestamp
     indicates the earliest of the VOP times 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).
   - If the RTP packet contains only configuration information and/or
     Group_of_VideoObjectPlane() fields, the timestamp of the next
     VOP in the coding order is used.
   - If the RTP packet contains only visual_object_sequence_end_code
     information, the timestamp of the immediately preceding VOP in
     the coding order is used."
And delete the definition of "RTP timestamps in RTCP SR packets".


If there is no comment, the authors will provide the last I-D to complete
the specification with reflecting these items.

Best regards,
Yoshihhiro Kikuchi
and authos of the Internet-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 Tue Oct 10 09:18:58 2000 
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To: 'Yoshihiro Kikuchi' <kiku@eel.rdc.toshiba.co.jp>, IETF AVT <rem-conf@es.net>
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Kikuchi-san,
  This is to get a clarification. You state, " ... change video timestamp
definition from
"composition time" to an ordinary RTP timestamp definition (capture time) ".
Does it mean removing the Composition TS and replacing it with Capture Time
Stamp or re-defining Composition TS as Capture TS? As I understood, the
suggestion is to replace Composition TS with Capture TS. Will this create
more incompatibility with the current WG-11 effort of developing standards
on how to carry different MPEG4 streams over RTP? Thanks.

With regards,
Ajay

-----Original Message-----
From: Yoshihiro Kikuchi [mailto:kiku@eel.rdc.toshiba.co.jp]
Sent: Tuesday, October 10, 2000 7:53 AM
To: IETF AVT
Cc: Yoshihiro Kikuchi
Subject: items to be reflected on the last I-D


Dear all,

Thank you very much for the comments on the MPEG-4 Audio/Visual RTP
Internet-Draft.

The following is the items to be reflected on the last text of the
Internet-Draft that the authors are considering to reflect comments raised.

1) RTP payload format for the short video header. (section 3)
    "When the short video header mode is used, the RTP payload format
    for H.263 SHOULD be used (the format defined in RFC 2429 is
    RECOMMENDED, but the RFC 2190 format MAY be used for compatibility
    with older implementations)."

2) video MIME/SDP parameter "config" (section 5.1)
     "config: This parameter SHALL be used to indicate the configuration
    of the corresponding MPEG-4 visual bitstream. It SHALL NOT be
    used to indicate the codec capability in the capability exchange
    procedure.
    ....................."

3) video timestamp (section 3.1)
The authors got a suggestion to change video timestamp definition from
"composition time" to an ordinary RTP timestamp definition (capture time).
This change is necessary to align with other RTP payload formats than MPEG-4
video. Note that audio timestamp has already be changed to "capture
time". The following is a definition the authors are considering:
    "Timestamp: The timestamp indicates the sampling instance
    of the VOP contained in the RTP packet. A constant offset,
    which is random, is added for security reasons.
   - When multiple VOPs are carried in the same RTP packet, the timestamp
     indicates the earliest of the VOP times 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).
   - If the RTP packet contains only configuration information and/or
     Group_of_VideoObjectPlane() fields, the timestamp of the next
     VOP in the coding order is used.
   - If the RTP packet contains only visual_object_sequence_end_code
     information, the timestamp of the immediately preceding VOP in
     the coding order is used."
And delete the definition of "RTP timestamps in RTCP SR packets".


If there is no comment, the authors will provide the last I-D to complete
the specification with reflecting these items.

Best regards,
Yoshihhiro Kikuchi
and authos of the Internet-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 Tue Oct 10 10:03:28 2000 
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From: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
To: "Luthra, Ajay (SD-EX)" <ALuthra@gi.com>, "IETF AVT" <rem-conf@es.net>
Cc: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
References: <973597126BDDD11197AA00805FA7EBC903421762@ntas0026.gi.com>
Subject: Re: items to be reflected on the last I-D
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Dear Ajay,

I expect better explanations from IETF experts, but my understanding on this
change is that it is necessary for the synchronization with other
audio/video codecs than MPEG-4 using "capture time" in their RTP payload
formats. Since it is an important role of this RTP payload format to align
with non-MPEG-4 codecs, this change seems reasonable to me. Please note
again that the audio timestamp has been changed to a capture timestamp.

Best regards,
Yoshihiro

----- Original Message -----
From: Luthra, Ajay (SD-EX) <ALuthra@gi.com>
To: 'Yoshihiro Kikuchi' <kiku@eel.rdc.toshiba.co.jp>; IETF AVT
<rem-conf@es.net>
Sent: Wednesday, October 11, 2000 1:18 AM
Subject: RE: items to be reflected on the last I-D


> Kikuchi-san,
>   This is to get a clarification. You state, " ... change video timestamp
> definition from
> "composition time" to an ordinary RTP timestamp definition (capture time)
".
> Does it mean removing the Composition TS and replacing it with Capture
Time
> Stamp or re-defining Composition TS as Capture TS? As I understood, the
> suggestion is to replace Composition TS with Capture TS. Will this create
> more incompatibility with the current WG-11 effort of developing standards
> on how to carry different MPEG4 streams over RTP? Thanks.
>
> With regards,
> Ajay
>
> -----Original Message-----
> From: Yoshihiro Kikuchi [mailto:kiku@eel.rdc.toshiba.co.jp]
> Sent: Tuesday, October 10, 2000 7:53 AM
> To: IETF AVT
> Cc: Yoshihiro Kikuchi
> Subject: items to be reflected on the last I-D
>
>
> Dear all,
>
> Thank you very much for the comments on the MPEG-4 Audio/Visual RTP
> Internet-Draft.
>
> The following is the items to be reflected on the last text of the
> Internet-Draft that the authors are considering to reflect comments
raised.
>
> 1) RTP payload format for the short video header. (section 3)
>     "When the short video header mode is used, the RTP payload format
>     for H.263 SHOULD be used (the format defined in RFC 2429 is
>     RECOMMENDED, but the RFC 2190 format MAY be used for compatibility
>     with older implementations)."
>
> 2) video MIME/SDP parameter "config" (section 5.1)
>      "config: This parameter SHALL be used to indicate the configuration
>     of the corresponding MPEG-4 visual bitstream. It SHALL NOT be
>     used to indicate the codec capability in the capability exchange
>     procedure.
>     ....................."
>
> 3) video timestamp (section 3.1)
> The authors got a suggestion to change video timestamp definition from
> "composition time" to an ordinary RTP timestamp definition (capture time).
> This change is necessary to align with other RTP payload formats than
MPEG-4
> video. Note that audio timestamp has already be changed to "capture
> time". The following is a definition the authors are considering:
>     "Timestamp: The timestamp indicates the sampling instance
>     of the VOP contained in the RTP packet. A constant offset,
>     which is random, is added for security reasons.
>    - When multiple VOPs are carried in the same RTP packet, the timestamp
>      indicates the earliest of the VOP times 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).
>    - If the RTP packet contains only configuration information and/or
>      Group_of_VideoObjectPlane() fields, the timestamp of the next
>      VOP in the coding order is used.
>    - If the RTP packet contains only visual_object_sequence_end_code
>      information, the timestamp of the immediately preceding VOP in
>      the coding order is used."
> And delete the definition of "RTP timestamps in RTCP SR packets".
>
>
> If there is no comment, the authors will provide the last I-D to complete
> the specification with reflecting these items.
>
> Best regards,
> Yoshihhiro Kikuchi
> and authos of the Internet-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 Tue Oct 10 10:38:49 2000 
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Message-ID: <001101c032e0$aa021460$6c81cf87@attnjs.research.att.com>
Reply-To: "M. Reha Civanlar" <civanlar@research.att.com>
From: "M. Reha Civanlar" <civanlar@research.att.com>
To: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>,
	"Luthra, Ajay (SD-EX)" <ALuthra@gi.com>, "IETF AVT" <rem-conf@es.net>
Cc: <4on2andIP-sys@advent.ee.columbia.edu>
References: <973597126BDDD11197AA00805FA7EBC903421762@ntas0026.gi.com> <031801c032db$da397880$1bc77885@ave>
Subject: Re: items to be reflected on the last I-D
Date: Tue, 10 Oct 2000 13:36:15 -0400
Organization: AT&T Labs - Research
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I don't think that using the "capture time" or the "composition time" will
make any difference on synchronization with other payloads as long as the
timing relation reported in the RTCP sender reports gets established
correctly. All RTP payload formats defined so far, except our MPEG-4 SL
payload format, draft-ietf-avt-rtp-mpeg4-03, use the "capture time." So,
choosing the "capture time" for MPEG4 ES payload formats will be consistent
with this convention. The reason for using the "composition time" in the SL
payload format is to eliminate the redundancy between the RTP and the SL
headers. The draft explains how to establish the timing relations in the
RTCP sender reports for synchronization with other payload formats under
this choice. Clearly, the redundancy argument does not apply to the ES
payload formats.

___________________________________________
M. Reha Civanlar
AT&T Labs - Research
A54D04, 200 Laurel Ave., Middletown, NJ 07748, USA
Ph: +1 732 420 9170
civanlar@research.att.com
http://www.research.att.com/info/mrc



----- Original Message -----
From: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
To: "Luthra, Ajay (SD-EX)" <ALuthra@gi.com>; "IETF AVT" <rem-conf@es.net>
Cc: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
Sent: Tuesday, October 10, 2000 1:02 PM
Subject: Re: items to be reflected on the last I-D


> Dear Ajay,
>
> I expect better explanations from IETF experts, but my understanding on
this
> change is that it is necessary for the synchronization with other
> audio/video codecs than MPEG-4 using "capture time" in their RTP payload
> formats. Since it is an important role of this RTP payload format to align
> with non-MPEG-4 codecs, this change seems reasonable to me. Please note
> again that the audio timestamp has been changed to a capture timestamp.
>
> Best regards,
> Yoshihiro
>
> ----- Original Message -----
> From: Luthra, Ajay (SD-EX) <ALuthra@gi.com>
> To: 'Yoshihiro Kikuchi' <kiku@eel.rdc.toshiba.co.jp>; IETF AVT
> <rem-conf@es.net>
> Sent: Wednesday, October 11, 2000 1:18 AM
> Subject: RE: items to be reflected on the last I-D
>
>
> > Kikuchi-san,
> >   This is to get a clarification. You state, " ... change video
timestamp
> > definition from
> > "composition time" to an ordinary RTP timestamp definition (capture
time)
> ".
> > Does it mean removing the Composition TS and replacing it with Capture
> Time
> > Stamp or re-defining Composition TS as Capture TS? As I understood, the
> > suggestion is to replace Composition TS with Capture TS. Will this
create
> > more incompatibility with the current WG-11 effort of developing
standards
> > on how to carry different MPEG4 streams over RTP? Thanks.
> >
> > With regards,
> > Ajay
> >
> > -----Original Message-----
> > From: Yoshihiro Kikuchi [mailto:kiku@eel.rdc.toshiba.co.jp]
> > Sent: Tuesday, October 10, 2000 7:53 AM
> > To: IETF AVT
> > Cc: Yoshihiro Kikuchi
> > Subject: items to be reflected on the last I-D
> >
> >
> > Dear all,
> >
> > Thank you very much for the comments on the MPEG-4 Audio/Visual RTP
> > Internet-Draft.
> >
> > The following is the items to be reflected on the last text of the
> > Internet-Draft that the authors are considering to reflect comments
> raised.
> >
> > 1) RTP payload format for the short video header. (section 3)
> >     "When the short video header mode is used, the RTP payload format
> >     for H.263 SHOULD be used (the format defined in RFC 2429 is
> >     RECOMMENDED, but the RFC 2190 format MAY be used for compatibility
> >     with older implementations)."
> >
> > 2) video MIME/SDP parameter "config" (section 5.1)
> >      "config: This parameter SHALL be used to indicate the configuration
> >     of the corresponding MPEG-4 visual bitstream. It SHALL NOT be
> >     used to indicate the codec capability in the capability exchange
> >     procedure.
> >     ....................."
> >
> > 3) video timestamp (section 3.1)
> > The authors got a suggestion to change video timestamp definition from
> > "composition time" to an ordinary RTP timestamp definition (capture
time).
> > This change is necessary to align with other RTP payload formats than
> MPEG-4
> > video. Note that audio timestamp has already be changed to "capture
> > time". The following is a definition the authors are considering:
> >     "Timestamp: The timestamp indicates the sampling instance
> >     of the VOP contained in the RTP packet. A constant offset,
> >     which is random, is added for security reasons.
> >    - When multiple VOPs are carried in the same RTP packet, the
timestamp
> >      indicates the earliest of the VOP times 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).
> >    - If the RTP packet contains only configuration information and/or
> >      Group_of_VideoObjectPlane() fields, the timestamp of the next
> >      VOP in the coding order is used.
> >    - If the RTP packet contains only visual_object_sequence_end_code
> >      information, the timestamp of the immediately preceding VOP in
> >      the coding order is used."
> > And delete the definition of "RTP timestamps in RTCP SR packets".
> >
> >
> > If there is no comment, the authors will provide the last I-D to
complete
> > the specification with reflecting these items.
> >
> > Best regards,
> > Yoshihhiro Kikuchi
> > and authos of the Internet-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 Tue Oct 10 13:33:57 2000 
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To: <rem-conf@es.net>, <4on2andIP-sys@advent.ee.columbia.edu>
Subject: Mime types for "generic MPEG" RTP payloads
Message-ID: <0056890014727710000002L902*@MHS>
Date: Tue, 10 Oct 2000 22:31:11 +0200
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Dear all,

The Civanlar et al draft defines an RTP payload type for carriage of
"generic MPEG-4 data" over RTP. When using this method, if I understand=
 it
correctly, the RTP payload data is signaled as "generic MPEG-4 data" an=
d a
receiver has to go into the object descriptors to find out whether the =
data
represents for example MPEG-4 audio, video or graphics.

Such method is clean from an MPEG-4 system point of view, but is not op=
timal
when compatibility with non-MPEG-4 system receivers is required. For
example, consider a full fledge MPEG-4 system application that composes=

several MPEG-4 audio, video and graphics streams within one beautiful
presentation. Of course non-MPEG-4 system receivers are not capable to =
show
such presentation. Nevertheless it may be of interest to present a sing=
le
audio and a single video from such presentation on a non-MPEG-4 receive=
r.
However, the non-MPEG-4 receiver will not be capable to understand such=

payload format.

Therefore I have the following proposal. Rather than defining one "gene=
ric
MPEG-4" mime type, we specify elementary stream specific mime types wit=
hin
the RTP payload specification for "generic MEG-4 data" (note that the M=
PEG-4
system signaling remains unchanged). For example :
- MP4VIDEO for MPEG-4 video streams
- MP4AUDIO for MPEG-4 audio streams
- MP4BIFS for MPEG-4 graphics streams
- MP4IPMP for MPEG-4 IPMP streams
- MP4JAVA for MPEG-4 Java streams
- MP4OD for MPEG-4 Object descriptor streams
- MP4CR for MPEG-4 clock reference streams
- MP4SCENE for MPEG-4 scene description streams
- MP4OCI for MPEG-4 object content information streams
- MP4FLEXMUX for MPEG-4 FlexMux streams

The MPEG-4 system data should be in separate field preceded by the leng=
th
byte as discussed in an earlier thread. In this way non-MPEG-4 system
receivers can skip the MPEG-4 system data without any MPEG-4 system
knowledge. Signaling this byte by an SDP attribute will allow full
compatible services, when appropriate.

Any comment ?

Kind 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 Tue Oct 10 18:01:06 2000 
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From: Florissa Colina <florissa@bmrc.berkeley.edu>
Subject: 10/11 The Making of GUI Bloopers -- Jeff Johnson, UI Wizards,
  Inc. 
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Berkeley Multimedia, Graphics and Interfaces Seminar

The Making of GUI Bloopers

Wednesday October 11, 2000, 
1:10-2:30 p.m. PDT
Fujitsu Seminar Room (405 Soda Hall) 

Jeff Johnson 
UI Wizards, Inc. 

Jeff Johnson, user-interface consultant and author of the new book "GUI
Bloopers: DON'Ts and DOs for Software Developers and Web Designers"
(Morgan-Kaufmann), will describe why he began collecting GUI design
bloopers several years ago, why he decided to publish them in a book, how
the bloopers are organized and presented, and how the book was usability
tested. His talk will be illustrated with many examples of bloopers.

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

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:

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 Tue Oct 10 21:59:15 2000 
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Subject: Investor and Stock Holder . . . .PLEASE READ TODAY, always picking winners
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STOCK PROFILER
FOR  THURSDAY, OCTOBER 12,  2000
 
Early Release to Group Leaders, Newsletter Owners & Members Only!
 
STRONG BUY RECOMMENDATION:
CMEC
COMMERCIAL CONCEPTS INC.
 
CMEC Closed at .23 on Tuesday, October 10 
Rumors of Huge News of Additional School Board and Hospital  Contracts !
 
VOLUME ALERT - CMEC ACCUMULATION IS UNDERWAY 

CMEC INVESTMENT RELATIONS IS MAILING OUT SPECIAL I.R. PACKAGE TO THOUSANDS OF BROKERS PLUS
STOCKHOLDERS, NEWSLETTERS, STOCK MAGAZINES AND INVESTMENT INSTITUTIONS.

  NEWS Released September 27 by CMEC CONFIRMS CONTRACTS  with  
  SCHOOL BOARDS, HOSPITALS, and MICRON COMPUTER
 
  Please post this message to all your  Clubs and other sites...
  as well as send it to your email and icq lists...thanks...suzi....
 
  3 Stock Sites Pick This One For Big Rise  Next  Week, Plus 
 
  4 Other Big Groups 
 
  Start Heavy Buying of This Stock as they feel it is VERY UNDERVALUED! 
 
  Picked by: 
3bagger
Yankees in Four 
Hot Stocks
Stock Profile
 
 
  THEIR TARGET FOR CMEC is $1.25 by October 31, 2000
 
  STRONG BUY RECOMMENDATION
  CMEC
  Commercial Concepts Inc. 
  
 CMEC Year High was $ 5.62
 

  Website at: http://www.cmecut.com
  NewsReleases at: 
  http://biz.yahoo.com/prnews/000907/ut_commerc.html 
----------------------------------------
 
Wednesday September 27, 2:20 pm Eastern Time 

Press Release
SOURCE: Commercial Concepts, Inc. 

Commercial Concepts Announces Successful Completion of Second Tranche of Funding
SALT LAKE CITY, Sept. 27 /PRNewswire/ -- Commercial Concepts, Inc. (OTC Bulletin Board: CMEC - news; ``CCI'') announced today the successful funding of a $250,000, 6% convertible note due September 8, 2003 to a private investment group. This note is the second tranche of the previously announced $6.5 million dollar equity line of credit that has been secured by the Company. 

George Richards, President and CEO of Commercial Concepts, commented: ``This ongoing support allows us to concentrate on finalizing our product development and commercial rollout, thereby expediting revenue generation. This funding is a vote of confidence in the Company and its products from a sophisticated international investment organization.'' 

The Company has moved its Web site to LMKI, Inc., a DSL-based ISP. The Company's Web site (www.cmecut.com) and e-mail had been inoperative for approximately two weeks. The initial cause was a sustained service breakdown by the Company's previous internet service provider (ISP). The Company's e-mail and Web site are now fully functioning. 

Additionally, the Company has been made aware that some of its shareholders did not receive their proxy ballots with their copies of the annual report and annual shareholders meeting proxy statement. Because some mailings to shareholders were sent too late for full participation in the shareholders' meeting held September 5, 2000, the Company kept open until October 16, 2000 the tally for proxy votes. Unfortunately, the distribution agent for Commercial Concepts elected to withhold the proxy ballot and insert an ``Information Only'' notice in its place. Commercial Concepts, Inc., has notified the distribution agent of its error and a proxy ballot is being mailed to shareholders for their vote. 

Commercial Concepts, Inc. 

The Company's products and services are based on Internet-related database driven software. The three principal products include CCI Picture Base©, The Wave Network© and X-Card©. Picture Base is a patent pending medical imaging software that captures and stores a variety of images. The Wave Network is a customized screen saver that disseminates host information while generating revenues through local and national advertising. These revenues are shared between the host and the Company. The Company is targeting three distinct host markets: non-profit organizations (schools, charities, and community groups), corporations (that have large intra-company communication needs), and commercial (sports franchises, theme parks, fan clubs). X-Card is an affordable ``multi media brochure'' delivered on a shaped compact disc (CD). Additional information is available at www.cmecut.com. Those interested in receiving e-mails of future press releases can register on the Company's Web site.
-------------------------------------

CATCH CMEC AT .23   AS CMEC "WAVE SCREENS" SOFTWARE MAY BE 
ON EVERY COMPUTER SCREEN IN EVERY SCHOOL AROUND THE WORLD 
AFTER THE NEWS  THAT THE DAVIS COUNTY SCHOOL BOARD HAS 
APPROVED CMEC "WAVE SCREENS" FOR ALL THEIR SCHOOLS' COMPUTERS

PLUS

CMEC'S MEDICAL IMAGING SOFTWARE MAY BE IN 
EVERY HOSPITAL AND MEDICAL CENTRE AROUND THE WORLD AS 
NEWS OF THE HOSPITAL CONTRACTS   REACH 
ALL THE ADMINISTRATORS OF ALL THE HOSPITALS AND CLINICS AROUND THE WORLD.
THIS .25 STOCK WILL START TO SKYROCKET AS THESE 
NEWSRELEASES ARE STARTING TO BE COPIED AND NOTICED BY THE HEADS OF
MOST OF THE MEDICAL AND EDUCATION FIELDS AROUND THE WORLD.
 
CMEC'S most recent newsrelease  is causing quite a flood of calls to 
CMEC offices for contract proposals from all over the world by Hospitals and School Boards.
 
 
Strong Buy Recommendation 
CMEC otc-bb
 
  CMEC was at Year High of $ 5.62  before Product Line was changed from just Y2K Software!
  CMEC has Contracts with MICRON, HOSPITALS and School Boards.
 
  There are Hundreds of Thousands of School Boards Around the World
  With Tens of Millions of Computers that are 
  Immediate Potential Clients FOR CMEC "Wave Screens"
  This Product Alone Could Mean Hundreds of Millions in Revenue For CMEC
  Now that Davis County Utah School Board has approved 
  CMEC "WAVE SCREENS"   FIRST MAJOR CONTRACT!!
  ------------------------
  DAVIS COUNTY SCHOOL BOARD IN UTAH APPROVES 
  CMEC "WAVE SCREENS" CONTRACT FOR APPROX. 30,000 SCREENS!
  ---------------------------------------------

 Thursday September 7, 8:31 am Eastern Time 
  Press Release
  SOURCE: Commercial Concepts, Inc. 
  Commercial Concepts Receives Davis County School District's Approval on Contract for Wave Screens
  Board of Education Approves Screen Saver for Use in Schools

SALT LAKE CITY, Sept. 7 /PRNewswire/ -- Commercial Concepts, Inc. (OTC BB: CMEC) today announced approval by the Board of Education of the Davis County School District of a standardized contract for the use of Wave Screens in each of the 74 schools in its district. Davis County Schools is the third largest school district in the State of Utah, with more than 58,000 students. 
 
  The approval calls for a five-year contract, with a non-compete agreement and it includes a 50-50 revenue sharing agreement for local and national advertising. 
 
  George Richards, President and CEO of Commercial Concepts, commented, ``This approval allows us to aggressively roll out the Wave Screen Program in Northern Utah. Davis County will be the role model for our National Campaign.'' 
 
  As was previously announced, Micron Commercial Computer Systems, a subsidiary of Micron Electronics can provide hardware to the school district in exchange for a percentage of their advertising revenues. 
 
  Commercial Concepts 
 
  The Company's products and services are based on Internet-related database driven software. The three principal products include CCI Picture Base©, The Wave Network© and X-Card©. Picture Base is a patent pending medical imaging software that captures and stores a variety of images. The Wave Network is a customized screen saver that disseminates host information while generating revenues through local and national advertising. These revenues are shared between the host and the Company. The Company is targeting three distinct host markets: non-profit organizations (schools, charities, and community groups), corporations (that have large intra-company communication needs), and commercial (sports franchises, theme parks, fan clubs). X-Card is an affordable ``multimedia brochure'' delivered on a shaped compact disc (CD). Additional information is available at www.cmecut.com . Those interested in receiving e-mails of future press releases can register on the Company's Web site. 
 
  This press release contains ``forward-looking statements'' within the meaning of Private Securities Litigation Reform Act of 1995, Section 27A of the Securities Act of 1933 and Section 21E of the Securities Exchange Act of the 1934. These forward-looking statements can be identified by the use of forward-looking terminology such as, ``This approval allows us to aggressively roll out the Wave Screen Program.'' Actual events or performance involve risks and uncertainties that could differ materially from those anticipated in such forward-looking statements. Factors that could cause actual results to differ materially from those projected in forward-looking statements include competition, the management of the Company's growth and the ability to deliver new products to market on time. Such forward-looking statements are subject to other risks and uncertainties, which are detailed in the Company's filings with the Securities and Exchange Commission. 
 
  SOURCE: Commercial Concepts, Inc. 
 
  --------------------------------------------------
  CMEC "WAVE SCREEN"  REVENUES ESTIMATE:
   
   3200 in just 1 month from 3 schools. using those numbers for 74 schools I come up with  $720,000 in revenue in just 1 yr for this account only on a 50/50 split. not bad numbers. 
 
  Companies would pay BIG bucks to get their banner adds onto the screens in side of schools and also in large and small companies. 
  90 local advertisers and 90 national advertisers per screen (maximum capacity) 
          Future Plans include 
          *Shopping Program 
          *Coupon Program 
 
  --------------------------
   
  DISCLAIMER: 
   
  This is not a solicitation to buy or sell any stocks mentioned herein, and all information 
  provided is believed to be accurate to the best of our knowledge.   Stock Profiler staff & management
  may hold, buy or sell any of the securities mentioned herein.  Always do your own 
  research and consult an investment professional whenever buying stocks, securities
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 ----------------------------------------------------------------------------------------------------------------------------------------------------------------

UPDATE:  
 
PABN   PAN AMERICAN BANK   Best Penny Stock Buy at  .009  STRONG BUY
 
DVDT  DIGITAL VIDEO DISPLAY TECH....STRONG BUY Closed at .25 Monday, October 9
Newsrelease of their $36 Million dollar contract.  
Rumors are out today of More Major News to be Released During the Week!! 
 -----------

 
 
 
 
 
 
 
 
 



From rem-conf Wed Oct 11 03:03:12 2000 
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From: myzhai@990.net
To: rm@mash.CS.Berkeley.EDU
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Subject: about layered video
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Sir,
  I remmeber that someone ever wrote a paper about the perceptual quality causing by losses in the layered video through simulation or theory analysis.
In detail, for example, in MPEG-2 the following frame IPBBPBBPBB conpose a 
GOP,if frame I losses,then the sequential PBBPBBPBB will be useless. and If
frame P losses,the the following BB will be useless. so If loss rate is p,it will cause great quality degradation if frame I and P is lost. Who can provide me some infomation about where I can find the similar papers(better including concrete simulation results in these papers)?

thx

Zhao


__________________________________________________
»¶Ó­Ê¹ÓÃ½ðÁêÈÈÏßÃâ·Ñµç×ÓÓÊ¼þÏµÍ³http://www.990.net



From rem-conf Wed Oct 11 07:02:43 2000 
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To: <rem-conf@es.net>, <4on2andIP-sys@advent.ee.columbia.edu>,
        <singer@apple.com>
Subject: RE: Types of MPEG-4 streams over IP
Message-ID: <0056890014757543000002L932*@MHS>
Date: Wed, 11 Oct 2000 15:55:18 +0200
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Dave,

A late reply. You wrote :

>Perhaps an a=3Dx-mpeg4-sl-header-length:<byte count>
>attribute on any RTP stream carrying MPEG4 content should be
>permitted, and required if byte count is non-zero (with zero being
>the default)?

You are right. But this RTP attribute requires a field of constant leng=
th
for the SL header. As SL headers may vary in length, use of such attrib=
ute
would result in loss of efficiency.

Therefore I proposed to define an a=3D<mpeg4-sl-header-flag> that signa=
ls the
presence of a length byte preceding the SL header and indicating the le=
ngth
of the immediately following SL header.

Kind regards,

Jan




-----Original Message-----
From:	singer@apple.com [mailto:singer@apple.com]
Sent:	donderdag, 05 oktober, 2000 18:46
To:	4on2andIP-sys@advent.ee.columbia.edu; rem-conf@es.net
Subject:	RE: Types of MPEG-4 streams over IP


At 5:45 PM +0200 10/5/00, jan.vandermeer@philips.com wrote:
>Dave,
>
>You wrote :
>
>>What is wrong with Carsten et al.'s sync layer mapping into RTP
>
>Good question, as usual. Let me list a few issues :
>
>1) Substantial overhead for small access units; a solution is being
>discussed for audio, but not yet for the other streams I listed; some =
may
>indeed deserve a more reliable transport method, but an RTP solution w=
ill
be
>required too, I think. Perhaps use of FlexMux the only solution here.

I agree.  Allow flexmux, where this is a problem, but don't mandate it.=


>2) Inclusion of a plain SL header with associated fields; I would pref=
er to
>start with a byte indicating the length of the subsequent field that
>contains the SL header. Such field is easily decodable by non-MPEG-4 s=
ystem
>receivers, provides a future proof solution (in future more informatio=
n may
>be contained in the SL header with a length that is unknown by now), a=
nd
>creates room for fully compatible extensions.

OK.  This seems fine on the surface.

>3) Little commonality with the Kikuchi-san drafts for audio and video.=

>Perhaps it is possible to merge both solutions by extending the Kikuch=
i-san
>drafts with the option to carry an SL header (preceded by the above le=
ngth
>field), for example signaled by a new attribute "MPEG-4 SL header carr=
ied"
>at SDP level.

Wasn't this where the whole flare-up started;  proposing to add an
optional SL-header to the Kikuchi proposal.

Perhaps an a=3Dx-mpeg4-sl-header-length:<byte count>
attribute on any RTP stream carrying MPEG4 content should be
permitted, and required if byte count is non-zero (with zero being
the default)?

This is just a question, as I realize there are still some messages I
have not managed to read...

>Kind regards,
>
>Jan
>
>>Can we assume that the following streams are probably most efficientl=
y
>>contained in a FlexMux stream, and consequently that is no need to sp=
ecify
>>use of RTP and SDP for those streams:
>>
>>- ObjectDescriptorStream
>>- ClockReferenceStream
>>- SceneDescriptionStream
>>- MPEG-7Stream
>>- IPMPStream
>>- ObjectContentInfoStream
>>- MPEGJStream
>
>
>I would be strongly opposed to mandating the use of flexmux for any
>stream.  What is wrong with Carsten et al.'s sync layer mapping into
>RTP?
>--
>David Singer
>Apple Computer/QuickTime

--
David Singer
Apple Computer/QuickTime

=



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<BR><B><FONT COLOR="#000080">Award-Winning Internet Privacy and Protection in an Easy to Use, Cost Effective Environment for Business and Consumers.</FONT></B><BR>

<BR>This company is involved in 3 complementary businesses, encompassing: <B>Internet security software, ISP services, and instant messaging, electronic greeting services. </B><BR>

<BR><B>Communications security software </B>utilizes powerful encryption technologies to secure the transmission of sensitive data over the Internet and across corporate networks. Advantages include voice email, a digital shredder, built-in file compressi

<BR><B><U>Recognition Gained.</U></B> The software has received several positive reviews including a 5-Star rating on ZDNet and selected Editors Picks on popular industry publications. Comparisons with the largest and most noteworthy competitors were extr

<BR><B><U>Acquisition Strategy</U></B>. This strategy involves 2 complementary areas. <BR>

<BR>1-To date, <B><I>5 ISP's in 4 major cities</I></B>, which not only adds direct revenues, it provides direct marketing and distribution channel for software products.<BR>
2-An Instant Messaging/Greeting Card service, which adds complementary services, customers,  visitors, and gives a foothold in international markets. <BR>

<BR>If you would like more information regarding this company and others like it please: <A HREF="mailto:street__news@email.com?Subject=Info001" TARGET="">[CLICK HERE]</A> <BR>

<BR>The company is <B><I>expanding this network</I></B> to offer connectivity, privacy, and protection solutions <B><I>coast to coast</I></B>, as well as to the rest of the world. <BR>

<BR><B><U>The Market.</U></B> The digital information protection business will be a <B><I>US $50 billion</I></B> industry within five years, and the Company&s software can be marketed to all segments of this market. The customer&s knowledge level is not a

<BR><B><U>Distribution.</U></B> is currently available at more than 1000 secure websites. Marketing is also done on its own site, to its ISP client base, and through partnerships with content management and anti-virus solution providers. <BR>

<BR><B><U>Revenue Base.</U></B> Annualized 1999 revenues totaled over $1 million. <BR>

<BR><B><U>Management  Team.</U></B> Collectively the top four executives represent over 50 years of related experience, key management is in place in technical and strategic growth, and a superior Advisory Board is in place. <BR>

<BR><B><U>Why?</U></B> This company has: a superior, recognized product line, well established distribution channels, a revenue base, an acquisition-oriented growth model in complementary areas, and a quality team. <BR>

<BR>If you would like more information regarding this company and others like it please: <A HREF="mailto:street__news@email.com?Subject=Info001" TARGET="">[CLICK HERE]</A> <BR>
<BR>
<BR>REMOVE INSTRUCTIONS: <BR>We apologize if you have received this in error. Simply <A HREF="mailto:optout@ourpostbox.com?Subject=optout" TARGET="">[Click Here]</A> and then click send <BR>

<BR>Disclaimer: This release may contain certain forward-looking statements reflecting the companies current expectations in its respective markets. Investors are cautioned that all forward-looking statements involve risks of course and uncertainties, inc

<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
z1010
#####################################################################
Notice, incidentally, that the theory of syntactic features
developed earlier cannot be arbitrary in the traditional practice of
grammarians.  With respect to specific goals, the characterization of
critically co-optive criteria raises serious doubts about a corpus of
utterance tokens upon which conformity has been defined by the paired
utterance test.  Based on integral subsystem considerations, my
proposed independent structuralistic concept requires considerable
systems analysis and trade-off studies to arrive at the anticipated
epistemological repercussions.  Interestingly enough, the systematic
use of complex symbols presents extremely interesting challenges to
all deeper structuralistic conceptualization.  Specifically, my
proposed independent structuralistic concept is necessary to impose
an interpretation on irrelevant intervening contexts in selectional
rules.
#####################################################################



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********* PLEASE DISTRIBUTE IN YOUR DEPT. OR ORGANIZATION! *********

Apologies if this reaches you in error or multiple times...

ASSETS'2000, the 4th International ACM/SIGCAPH Conference on Assistive
Technologies, will be held November 12-13, 2000, in Arlington, VA (a
suburb of Washington DC, just minutes from Reagan National Airport).

Don't miss this unique and exciting opportunity to participate in ACM's
PREMIER CONFERENCE ON COMPUTER SCIENCE RESEARCH RELATED TO ASSISTIVE
TECHNOLOGIES AND UNIVERSAL ACCESS, which is nevertheless a small and
intimate event where meals are taken together and you have a chance to
really talk to people.

THIS YEAR'S CONFERENCE HOTEL is connected by sky-bridge to the National
Science Foundation, so attendees can plan to take advantage of any free
time to meet with NSF staff and learn about funding opportunities in
their research areas (it is best to make an appointment before-hand).
It is also located directly atop a Metro station, so sight-seeing in
DC is easy.

And if you ACT NOW there's still time to take advantage of DISCOUNTED
EARLY REGISTRATION RATES for the conference!

FOR MORE INFORMATION, please see:

               http://www.acm.org/sigs/conferences/assets00

We look forward to seeing you next month at ASSETS'2000!



From rem-conf Wed Oct 11 13:50:21 2000 
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From rem-conf Wed Oct 11 15:42:16 2000 
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From: Farez <f.abdulrahman@cs.ucl.ac.uk>
Subject: Invitation to AMOC 2000 - 3 weeks to go!
Cc: mobile-systems@cs.ucl.ac.uk, agents@mailserver-ng.cs.umbc.edu, 
    rem-conf@es.net, havinga@cs.utwente.nl, oulusoy@cs.bilkent.edu.tr, 
    helal@cise.ufl.edu, anan@master.cpe.ku.ac.th, b.du@qut.edu.au, 
    habaebi@hotmail.com, borhan@eng.upm.edu.my, lala@asti.dost.gov.ph, 
    hara@ise.eng.osaka-u.ac.jp, jiy@disys.korea.ac.kr, hwang@disys.korea.ac.kr, 
    hara@ise.eng.osaka-u.ac.jp, sri@it.iitb.ernet.in, abhi@it.iitb.ernet.in, 
    rvijay@it.iitb.ernet.in, xwang4@cs.gmu.edu, skjoo@network2.cs.usm.my, 
    tcwan@cs.usm.my, rks@cs.usm.my, Rseptiaw@mail.bond.edu.au, 
    pils@i4.informatik.rwth-aachen.de, jens.hartmann@ericsson.com, 
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Hello,

Apologies for multiple receipt of this. Please pass on this invitation to
those who may be interested.

Regards,
Farez

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


                   Programme & Call for Participation

                               AMOC 2000
         FIRST ASIAN INTERNATIONAL MOBILE COMPUTING CONFERENCE
                    31 October - 3 November, 2000.
              Penang Parkroyal Resort, Penang, Malaysia

                Organised by University Malaya, Malaysia
                     Sponsored by MRCB Multimedia
           with Multimedia Development Corporation, Malaysia
                  In cooperation with ACM SIGMOBILE

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


                           3 weeks to go!

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

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

* 31 October, 2000 - Tutorial only

0900-1215: Tutorial: Mobile Ad-Hoc Wireless Networks

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

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

* 1 November, 2000

0900-0930: Registration

0930-0945: Opening speech 
           
           Prof. Dr. Ir. Mashkuri Yaacob
             Deputy Vice Chancellor (Academic), 
             University of Malaya

0945-1030: Guest Speaker

           Dr. Andreas Fasbender,
             Vice Director,
             Ericsson Application Research Asia Pacific,
             Ericsson Cyberlab, Singapore

           Topic: "Mobile Internet: What's in store for the end user?"

1030-1050: Break

1050-1135: Guest Speaker 2 (TBA)

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
           Media Arbitration 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, University 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 University, 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 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

1030-1045: Break

1045-1215: Session 5: Data Acccess, Display and Management

           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

1215: Conference Close


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

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


============================
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 (USD 100)

Others:      _ RM 600 (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 will be provided during the conference.
Are you a vegetarian?                             _ YES _ NO

____________________________________________________________

Tutorial Registration
---------------------

Will you be attending the tutorial "Mobile Ad Hoc
Wireless Networks" on 31 October 2000.            _ YES _ NO

Registration for the tutorial costs RM 200 (USD 50). This
price includes lunch and tutorial notes.


Your signature: _____________________________ Date: ________

____________________________________________________________

Payment
-------

Please write amount:

  Conference registration: _________

  Tutorial registration:   _________

  Total:                   _________

Please pay the total amount above in a 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

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

For hotel booking info, please see:

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

Important: AMOC will coincide with the local holiday season, so
please book your rooms ASAP.

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



From rem-conf Thu Oct 12 03:47:15 2000 
From rem-conf-request@es.net Thu Oct 12 03:47:14 2000
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          id <g.16335-0@bells.cs.ucl.ac.uk>; Thu, 12 Oct 2000 11:41:19 +0100
From: Orion Hodson <O.Hodson@cs.ucl.ac.uk>
X-Organisation: University College London, CS Dept.
X-Phone: +44 (0)20 7679 3704
To: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
cc: rem-conf@es.net
Subject: Re: RTP sent to different ports
In-reply-to: Your message of "Tue, 10 Oct 2000 14:49:12 +0300." <39E30238.768B1EF9@lmf.ericsson.se>
Date: Thu, 12 Oct 2000 11:41:18 +0100
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Gonzalo Camarillo writes:
> 
> Now I want to have a conversation with somebody. I want just to transmit
> my voice. However, sometimes I will use PCM u-law and others PCM A-law.
> The receiver wants to receive (due to any reason) different codecs on
> different port numbers.
> 
> As we have discussed previously in this thread, we will establish two
> media streams. But I want the receiver to know that I will never
> transmit simultaneouly in both media streams.
> 
> I will transmit using one OR the other.
> 
> I would like my session description to look like:
> 
> v=0
> o=John 289085535 289085535 IN IP4 first.example.com
> t=0 0
> c=IN IP4 111.111.111.111
> m=audio 20000 RTP/AVP 0
> a=fid:1
> m=audio 20002 RTP/AVP 8
> a=fid:1
> 
> If an implementation does not understand the fid attribute it will try
> to mix the audio streams. Since one is "empty" all the time, doing this
> is not really a big deal. Thus, backwards compatibility is not a
> concern.

Backwards compatibility is hard to predict.  How can we be sure the
following won't happen?

i) A separate audio application is launched for both audio streams.
   The first instance to receive audio locks the audio device so that
   any audio received by the second is discarded.

ii) A single instance is launched because the parser chooses to
    observe a single audio stream instance.

These both result in a receiver could be cut out of a conversation
unexpectedly because of a codec change.

> Do you have any concerns with this approach?

My concerns are not with with SDP description per se, but with the
motivation for using two ports for the same media session.

It really would be better to modify TFT to be able to select the radio
channel configs based on RTP payload identifier.  There is no question
of compatibility TFT can be modified.  Is there a categorical
statement on whether this is possible?
 
Supposing TFT is immutable, there is also the issue of whether AMR
will, for all practical purposes, be the only voice codec that is
going to be used to communicate to 3GPP terminals.  The AMR payload
does not benefit from the multi-port approach.  It's also not clear
what the situation will be with music and video, applying UEP for some
popular payload formats will be expensive for a gateway if the stream
requires parsing and analyzing for each packet.  That would DTMF as
the compelling reason for this proposal, but that can use redundancy
rather than UEP anyway (for the wired network this pretty much
essential for DTMF applications).

> RTP and RTCP remain untouched.

It's just all the existing RTP applications and RTP supporting
firewalls that need changing to provide a predictable behaviour with
this proposal.  The scale of 3GPP market might be sufficient market
pressure to see the existing applications get modified quickly.  

- Orion




From rem-conf Thu Oct 12 11:21:56 2000 
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Error Tolerant RTP Payload Format for AMR
	Author(s)	: Q. Xie, S. Gupta
	Filename	: draft-xie-avt-et-rtp-amr-00.txt
	Pages		: 9
	Date		: 09-Oct-00
	
This document defines the RTP payload format for *error tolerant*
delivery of Adaptive Multi-Rate (AMR) speech frames over an RTP
session.

The flexibility on bandwidth requirements and the tolerance to bit
errors of AMR codes are not only beneficial for 'over-the-air'
wireless links, but are also very desirable for any Voice-over-IP
applications. The design is focused on how to best facilitate these
features of AMR codec in an IP environment.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-xie-avt-et-rtp-amr-00.txt

Internet-Drafts are also available by anonymous FTP. Login with the username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-xie-avt-et-rtp-amr-00.txt".

A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html 
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
	mailserv@ietf.org.
In the body type:
	"FILE /internet-drafts/draft-xie-avt-et-rtp-amr-00.txt".
	
NOTE:	The mail server at ietf.org can return the document in
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	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
	a MIME-compliant mail reader.  Different MIME-compliant mail readers
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implementation to automatically retrieve the ASCII version of the
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--OtherAccess
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ENCODING mime
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From rem-conf Thu Oct 12 11:21:56 2000 
From rem-conf-request@es.net Thu Oct 12 11:21:55 2000
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From: The IESG <iesg-secretary@ietf.org>
Subject: Protocol Action: Registration of parityfec MIME types to
         Proposed Standard
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The IESG has approved Registration of parityfec MIME types
<draft-ietf-avt-fecmime-01.txt> for publication as a Proposed Standard.


This document is the product of the Audio/Video Transport Working
Group.  The IESG contact persons are Allison Mankin and Scott Bradner.

 
Technical Summary
 
 The RTP payload format for generic forward error correction (RFC 2733)
 allows RTP participants to improve loss resiliency through the use of
 traditional parity based channel codes. This payload format requires four
 new MIME types, audio/parityfec, video/parityfec, text/parityfec and
 application/parityfec. "The Registration of parityfec MIME types" document
 serves as the MIME type registration for these formats.

Working Group Summary

 The working group supported advancing this document.

Protocol Quality

 This document was reviewed for the IESG by Allison Mankin.

Note to RFC Editor:

Please modify draft-ietf-avt-fecmime-01.txt by replacing all four
occurances of:

          Security considerations: none

with:

          Security considerations:  the same security considerations
          apply to these mime registrations as to the payloads for
          them, as detailed in RFC 2733.




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To: <4on2andIP-sys@advent.ee.columbia.edu>
Cc: <rem-conf@es.net>
Subject: Three documents
Message-ID: <0056890014795482000002L922*@MHS>
Date: Thu, 12 Oct 2000 23:41:21 +0200
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Dear all,

Please find attached three input documents for the La Baule meeting. M6=
447
and M6448 are short documents; the first one is about MIME types for
carriage of MPEG-4 streams in RTP, and the second one about carriage of=
 SL
headers within RTP. The third one, M6449, contains a rough draft of a s=
ingle
specification for carriage of MPEG-4 streams over RTP. It includes majo=
r
parts of the drafts of Civanlar et al, Kikuchi-san et al and Roux et al=
,
while using the approach discussed in M6447 and M6448.

Partly M6449 is in response to the question of Colin for a specific
proposal. Hopefully it clarifies where "I am driving to", to use the wo=
rds
of Colin. More work is needed, but I hope that the basic concepts of M6=
449
can serve as a basis for further convergence. Please feel free to use i=
t for
more elaboration.

After today I will not be able to further participate in the discussion=

until La Baule, as I leave for a small but special vacation. For those =
who
guess right : indeed, indeed.

See you in La Baule, where I will arrive on October 25 (Wednesday). Hav=
e a
good discussion and a very good start of the meeting.

Kind 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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From rem-conf Fri Oct 13 00:40:17 2000 
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Subject: DNAP - THE NEXT $50 BIOTECH STOCK?      
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WE THINK SO !!

Dnaprint (DNAP) is an emerging biotech company and is trading under $1 !!!

1) CEO (Dr. Tony Frudakis) Lead scientist for Corixa starts his own company a 
   few months ago
    
2) Company is formed by a reverse merger of a pinksheet company

3) DNAP has already completed the accounting to get off the pinksheets and 
   should be listed on the OTCBB soon.

4) Dr. Frudakis was  responsible for a patent that sold for $54,000,000 to 
   Smith Kline Beecham while he was at Corixa.
 
5) Corixa's stock has gone from 14 Cents a few years ago to 50 Dollars today.
 
6) According to the Barron's article (9/25/00 pg 52), Dnaprint is in the late stages of forming 
   strategic alliances with major companies.


MAJOR WEB PAGES RELATED TO DNAP 
dnap 
http://www.dnaprint.com/ 
orchid biosciences inc 
http://www.orchidbio.com/ 

INVEST IN DNAP 
email for free investment package= hresources@dnaprint.com 
or call 1-877-362-3453(toll free) 
investment club web page=http://clubs.yahoo.com/clubs/dnaprintgenomicsinvestors 
investors report= 
http://www.companyreporter.com/index.php?u=companies.php 
great dnap investor links: 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=22034 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=23118 
http://cbs.marketwatch.com/news/current/molinski.htx?source=htx/http2_mw =invest in biotech 
http://www.loonycat.com/Genomics.htm =the human genome investors portal 
clearstation rates danp 
http://www.clearstation.com/cgi-bin/details?Event=peek&Symbol=DNAP&Refer=http://www.purepennies.com/ 
purepenny&3bagger rate dnap 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=PCBM&read=132938 
Pennystockprofits assess dnap 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=15730 
tipreporter rates dnap http://www.tipreporter.com/search.php?search=dnap 
dnap on stocktribes hotlist for 21 weeks 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=24335 
analysts tips and reports on dnap 
http://www.tipreporter.com/search.php?search=dnap 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=15730 

why is dnap unique in the bio tech sector?? 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=22654 

NEWSLINKS GENERAL FOR DNAP 
dnap company news 
http://www.bell2bell.newsalert.com/bin/digest?Symbol=DNAP& =bell2bell news 
http://quicken.excite.com/investments/quotes/?s=DNAP&t=0 =excite news 
http://biz.yahoo.com/n/D/DNAP.html =yahoo news 
dnap in barrons http://www.sockthestocks.com/profiles/dnapbarrons.htm 
bioinform news service 
http://www.bioinform.com/ 
online journal of bioinformatics 
http://www.cpb.uokhsc.edu/ojvr/xmlpaper.html#Sites 

UPCOMING EVENTS RELATED TO DNAP 
bio partnering Europe- London England-16-17 oct 2000 
http://www.biopartnering.com/bpe/index.html 
structural genomics conference-yokohama,japan,2-5nov2000 
http://icsg2000.riken.go.jp/ 

TECHNICAL SITES RELATED TO DNAP 
dnap patents 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=21559 
dnap is about pharmacogentics 
http://www.nigms.nih.gov/funding/medforyou.html 
stem cell research 
http://www.stemcellresearchnews.com/index.html 
personalized medicine no longer science fiction 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=25033 
safe drugs that fit like a glove 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=11203 

NEW STAFF ARRIVING AT DNAP SARASOTA 
dr. ramin mirashemi, a new dnap addition—check this track record!!!!!!!!!!!! 
http://www.google.com/search?q=Ramin+Mirhashemi&hl=en&lr=&safe=off&start=0&sa=N 
dr.jose arena joins dnap scientific advisory board 03 oct 2000 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=23532 
scientific advisory board makeup 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=24575 

DNAP COMPARED TO ITS TECHNICAL EQUALS, AND RIVALS: 
genomica(gnom)nasdq-ipo29sept00($19)now$22 
http://www.bell2bell.newsalert.com/bin/story?StoryId=CoDqtWbWbtKvgmda1&FQ=genomica&HdlFmt=simple&Title=Headlines%20for%3A%20genomica 
informax(inunx)nasdq-ipo-03oct00($16)now$23 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=23532 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=23552 
incite genomics(incy)nasdq $37 
http://quicken.excite.com/investments/news/?symbol=INCY 
genentech(dna)($172)-2for1split 5 oct00-263million share float---dnap float = 269 million 
http://www.bell2bell.newsalert.com/bin/digest?Symbol=DNA& 

DNAP IN EUROPE 
http://www.8ung.at/zockstocks 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=21485 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=24168 

WHERE IS DNAP LOCATED—HOW DO YOU GET THERE? 
great air fare rates round trip to sarasota 
http://leisure.travelocity.com/RealDeals/Details/0,2941,TRAVELOCITY_AIR_449,00.html 
dnaps new upscale digs in swinging sarasota, fla 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=21059 
motel shopper link for sunny sarasota area 
http://www.roomsaver.com/roomsaver/city/index.html?state=FL&city=Sarasota 
location map of coconut ave digs 
http://maps.yahoo.com/py/maps.py?BFCat=&Pyt=Tmap&newFL=Use+Address+Below&addr=900+COCONUT+AVE&csz=SARASOTA%2CFLORIDA%2C+34234&country=us&Get%A0Map=Get+Map 
avis rent a car sarasota 
http://www.avis.com/cgi-bin/maps_and_directions/driving_maps/us_locations/mqinterconnect 

INTELLIGENT ADVICE OR BIASED ROT?KNOW THE DIFFERENCE OR HOW TO USE THESE MESSAGE BOARDS 
email suspected lying-sham efforts,etc to your federal govt(it,s called the sec) at enforcement@sec.gov 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=PCBM&read=132869 
get the mob out of microcaps 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=PCBM&read=132789 
crush naked shorting –stop being screwed without even knowing it—read me 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=PCBM&read=132581 
suspicious sounding codswallop?? ask them to take the universal basher quiz!!!! 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=PCBM&read=132816 
dnap has unusually good promise—accounting for the sad fact that were in “basher heaven”-read me 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=PCBM&read=132818 
sec offers tips on how to avoid scams on the internet—good article 
http://www.sec.gov/consumer/cyberfr.htm 


CONTENTS 
1.BACKGROUND 
2.FOUNDERS 
3.PRODUCTS&SERVICES 

1.BACKGROUND 

DNAPrint genomics is a young e-biotech company based in southwest Florida. We are developing an informatics platform system that will provide dynamic solutions for disease gene discovery, genetic predisposition and genetic analysis testing. Our work has real-life application to the germinating field of Personalized Medicine and will help lay the foundation for a brand-new area of medical research called Phenomics. 
The PhenomeSM platform system that we are developing will help identify an individual predisposed to develop cancer before the onset of illness so that lifestyles can be altered and/or preventative measures taken. It will be used to identify individuals who are incompatible with certain drug treatments before the drugs are prescribed and damage is done. It will be used to tease out important genetic determinants associated with complex genetic diseases, so that drugs can be developed to target these genes. 
Our aim is ambitious. By partnering our platform with biotech/pharma, we will participate in the downstream profits generated from pharmaceutical products our platform has enabled. By marketing our platform directly to the public, we intend to make comprehensive genotype screening solutions accessible to people all over the world via the internet. Our system will enable a more holistic approach to using genomics information to improve peoples lives. 
The patient of the future will have more information and power than ever imagined. The human genome project, the internet and recent technological advancements all contribute to make this possible, and we intend to take advantage of these synergies to become the primary conduit for the generation and distribution of personalized genetic information. 

The Market 
Pharmacogenomics is currently a 1 billion dollar industry and is projected to be a 10 billion dollar industry by the end of 2003. Several billions of dollars have been spent on the human genome project so far, and an equal amount will likely be spent trying to understand what all of the genomics information means (i.e. bioinformatics, proteomics and phenomics). Genomics is a multi billion dollar industry world-wide and the market for personalized medicine is expected to exceed $25 billion dollars by 2010. Recently congress has approved a $2.3 billion increase in NIH funding to a total of $18 billion dollars for the year 2000. 

2.FOUNDERS 
Tony Frudakis, Ph.D.(tfrudakis@dnaprint.com) 
Chief Executive Officer, President and Laboratory Director. Dr. Tony Frudakis is a Phi Beta Kappa, Magna Cumme Laude graduate of the University of California system. He received his doctorate degree in molecular and cell biology from the University of California Berkeley and has 14 total years of experience as a molecular biologist. Dr. Frudakis' technical experience encompass a range of areas within biology, including the molecular biology of eukaryotic transcriptional regulation, the creation and study of transgenic animals, the development of novel nucleic acid isolation techniques and gene discovery protocols, bioinformatics processing and anatomical gene expression profiling. At Corixa Corporation in Seattle, WA he developed several new techniques for RNA fingerprinting, managed and executed high-throughput gene discovery programs for various cancers and was instrumental in the companies early success in attracting R and D partners. Founding GAFF biologic in 1998, Dr. Frudakis established a good reputation for high quality DNA sequencing and library construction/screening services, until its evolution into the DNAPrint Genomics that exists today. In all, his work has resulted in a patent portfolio for over 350 unique genes and 2 products 

George Frudakis(gfrudakis@dnaprint.com) 
Vice President of Business Affairs. George Frudakis has vast experience with start-up company development. He started and developed a successful multi-component company called GAFF group, which today enjoys over 15 million dollars annually in revenue. During his life’s work as an entrepreneur and businessman, George has learned how to run and successfully manage the financial and operational aspects of both small and large companies alike. He is a graduate from the California State University at Long Beach in California. 

Myung Ho Kim, Ph.D. Mathematics(mkim@dnaprint.com) 
Director of Informatics and Lead Mathemetician. Myung recieved his Ph.D. in Mathematics from the University of Michigan in Ann Arbor in 1988 and M.S. in Mathematics from The Ohio State University in Columbus Ohio. Most recently, Myung worked on the Ant World Server project supported by DARPA (Defense Advanced Research Progect Agency) and implemented Kolmogorov-Smirnov tests, linear/nonlinear regression, nonparametric analysis and modeling to a TREC 6 (Text Retrieval Conference) project using SAS, C/C++ and Perl languages. Prior to this, Myung simulated mathematical modeling for breast tumors and implemented/developed various image processing algorithms. He has served as an Honorary research Fellow in the Department of Mathematics and Information Science at the University of Auckland, New Zealand and a Visiting Research Scholor in the Department of Mathematics at the University of California at Santa Cruz in California. Prior to coming to the United States, Myung held the position of Assistant Professor in the Department of Mathematics at Sungkyunkwan University in Soel Korea. 

Venkateswarlu Kondragunta, Ph.D. (vkondragunta@dnaprint.com) 
Director of Biostatistics. Dr. Kondragunta obtained his Ph.D. in Statistics from the University of Madras, Madras, India. He most recently served as a Postdoctoral Fellow at the Schoolof Public Health, University of Texas, Houston, TX under the tutelage of Dr. Ranajit Chakraborty, one of the most famous and accomplished genetic mathematicians in the world. His research interests include variance components, linear regression models, general linear models, design and analysis, discriminant analysis, principal component analysis and statistical genetics. He has taught courses in Variance Components, Design and Analysis of Experiments and Linear Regression Models and Probability Theory at Concordia University in Montreal, Canada. Dr. Kondragunta is an accomplished programmer and has amassed considerable experience with implementing statistical genetic ideas into C++, S-plus, FORTRAN (7 years experience) and JAVA code. His most recent work used the REG, ANOVA, GLM, VARCOMP, CANDISC, CLUSTER, DISCRIM, FACTOR, PRINCOMP and LOGISTIC tools in the SAS computing environment on real and simulated data sets. 
A partial list of publications for Dr. Kondragunta include: 
Chaubey, Y.P. and K.Venkateswarlu. A numerical study of some robust estimators of variance components in one-way model. International Indian Statistical Association Conference proceedings, October 10-11, 1998, McMaster University, Hamilton, Canada. 
Venkateswarlu, K. K.N.Ponnuswamy, and M.R.Srinivasan. Estimation of variance components based on diallel model. Mathematical Biosciences, V-150, 105-112. 
Venkateswarlu, K, and K.N.Ponnuswamy. Estimation of variance components based on bio-model of diallel crosses (balanced). Communications in Statistics (Theory & Methods), Vol-27, 139-152. 
Venkateswarlu, K. Estimation of variance components based on diallel model involving maternal and maternal interaction effects. Biometrical Journal, Vol-39, 287-295. 

3.PRODUCTS&SERVICES 
see dnap web page for a complete description of products & services http://www.dnaprint.com/ 
SNiPdocTM 
High-throughput SNP profiling-To help us with our bottom line, we provide contract SNP profiling services to the drug development community. We call this service our SNiPdocTM service, and it is targeted towards pharmaceutical companies. Many pharmaceutical companies are seeking to enter into the field of Pharmacogenomics. This new field of study could have a dramatic impact on overall health care costs in the future. Furthermore, by having access to pharmacogenomics data, pharmaceutical companies can focus their clinical trials on segments of the population that are genetically receptive to the treatment. Experimental drugs can be targeted to individuals of a particular genotype, and genetic variations shown to underlie a poor-response to a trial drug could be used as a basis for eliminating the non-responding segment of the trial population from the study. In this way, pharmaceutical companies can reduce drug trial failures and the costs associated with them. Contracting this work out to service providers is a viable option for many smaller pharmaceutical companies who do not have the volume of work necessary to justify the price of the equipment and expertise. At DNAPrint Genomics, we offer multiple patient – multiple SNP sampling matricies designed on a custom, semi-custom or predefined basis. We can use our proprietary PhenomixM databases of medically relevant SNPs (numbering in the thousands of SNPs at present), or design chips for your application de-novo. We will work with your scientists to develop the experiments from beginning to end, that suit your needs, and we can handle your project from blood/tissue or DNA starting points. Data handling will be customized for each client; data can be deposited in our secure SNP relational databases system offering you the option of accessing your data over the internet. Alternatively, we can burn and hand-delivered CDs to 



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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: RTP payload format for MPEG-4 Audio/Visual streams
	Author(s)	: Y. Kikuchi, T. Nomura, S. Fukunaga, 
                          Y. Matsui, H. Kimata
	Filename	: draft-ietf-avt-rtp-mpeg4-es-05.txt
	Pages		: 18
	Date		: 12-Oct-00
	
This document describes RTP payload formats for carrying each of MPEG-4
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also specifies fragmentation rules. It also provides specifications for
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From rem-conf Fri Oct 13 08:51:34 2000 
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In-Reply-To: Message from jan.vandermeer@philips.com 
   of "Thu, 12 Oct 2000 23:41:21 +0200." <0056890014795482000002L922*@MHS> 
Date: Fri, 13 Oct 2000 09:42:05 -0400
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Jan, all,

--> jan.vandermeer@philips.com writes:
>Please find attached three input documents for the La Baule meeting. M6447
>and M6448 are short documents; the first one is about MIME types for
>carriage of MPEG-4 streams in RTP, and the second one about carriage of SL
>headers within RTP. The third one, M6449, contains a rough draft of a single
>specification for carriage of MPEG-4 streams over RTP. It includes major
>parts of the drafts of Civanlar et al, Kikuchi-san et al and Roux et al,
>while using the approach discussed in M6447 and M6448.
>
>Partly M6449 is in response to the question of Colin for a specific
>proposal. Hopefully it clarifies where "I am driving to", to use the words
>of Colin. More work is needed, but I hope that the basic concepts of M6449
>can serve as a basis for further convergence. Please feel free to use it for
>more elaboration.

A few comments....

The proposal in M6447 is identical in effect to defining two RTP payload
formats, one with the length byte and one without. Saying

	m=audio 49230 RTP/AVP 96
	a=rtpmap:96 MP4SL/90000
	a=fmtp:96 MPEG4-SL-length-present=1

is the same as

	m=audio 49230 RTP/AVP 96
	a=rtpmap:96 MP4SL-length/90000

and has exactly the same interoperability issues. If a length field is
needed it will have to be mandatory, else you have potential problems.

As an aside - are SL headers not self-describing? I was assuming that a 
terminal could parse them and decide when they were complete - is there 
no header extension bit?

Regarding M6449: please submit this as an internet-draft! It gets very
confusing otherwise...

Colin



From rem-conf Fri Oct 13 10:55:34 2000 
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From: "M. Reha Civanlar" <civanlar@research.att.com>
To: <jan.vandermeer@philips.com>, "Colin Perkins" <csp@isi.edu>
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	"rem-conf" <rem-conf@es.net>
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Subject: Re: Three documents 
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----- Original Message -----
From: "Colin Perkins" <csp@isi.edu>

> The proposal in M6447 is identical in effect to defining two RTP payload
> formats, one with the length byte and one without. Saying
>
> m=audio 49230 RTP/AVP 96
> a=rtpmap:96 MP4SL/90000
> a=fmtp:96 MPEG4-SL-length-present=1
>
> is the same as
>
> m=audio 49230 RTP/AVP 96
> a=rtpmap:96 MP4SL-length/90000
>
> and has exactly the same interoperability issues. If a length field is
> needed it will have to be mandatory, else you have potential problems.

I completely agree with Colin. If the consensus is that such a length field
is necessary, it should be mandatory and we can simply add it to the SL
payload format. However, so far, I am not sure if such a consensus exists. I
hope, this issue can be resolved in the MPEG meeting.

M. Reha Civanlar




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Subject: AVT minutes from Pittsburgh
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Folks,

Attached are the minutes from the Pittsburgh AVT meeting.
Colin


------------------------------------------------------------------------------
Minutes of the Audio/Video Transport working group

Reported by Colin Perkins and Stephen Casner.

The audio/video transport working group met twice at the 48th IETF in
Pittsburgh.  In the first session we discussed IP encapsulation,
header compression, RTCP extensions, forward error correction, and MP3
audio.  The second session was for discussion of the payload formats
for AMR, SMPTE 292M, DSR, SONET over IP, and MPEG-4.

The meeting opened with an introduction and status update by Steve
Casner.  A number of drafts have been published as RFCs since the last
meeting:

 - RFC 2793: RTP payload for text conversation
 - RFC 2833: RTP payload for DTMF and other tones
 - RFC 2862: RTP payload for real-time pointers

The RTP MIB, MIME type registration for the parity FEC payload format,
and the payload formats for DV audio and video are awaiting
publication.  The payload format for G.722.1 is in working group last
call, comments are solicited.

The working group last call on the revised RTP specification and
audio/video profile completed in November 1999, with an agreed set of
edits.  These edits have now been completed.  However, the issue of
congestion control was raised: other protocols are required to have
congestion control, why not RTP?  Following on from discussion in
Adelaide, a number of changes have been made to the RTP specification
and profile to address this concern: the RTP specification has text to
note that congestion control is required, but that the requirements
are different to TCP and are somewhat context dependent, the details
of the required congestion control SHOULD be specified in profiles.
The audio/video profile has text from Mark Handley which notes that:

 - If on a network with QoS support, monitor to ensure service was
   received 
 - If on a best-effort network, monitor loss and reduce the
   transmission rate or stop if average throughput is greater than
   what TCP would get 

Comments on whether the text in the drafts is sufficient, acceptable
and appropriate are solicited.

There were several comments from meeting participants.  Some felt this
text was not specific enough, especially for the multicast case, while
others felt it was too early to specify the details of what could be
done without further experimentation.  Perhaps that means this text is
vague to the appropriate degree?  In discussion on the mailing list
shortly after the meeting, the primary concern was that the user may
want an RTP stream to get more than its share of the bandwidth of a
narrow link near the user.  It is expected that QoS support should be
used in that case so that appropriate service can be requested, as in
the first case listed above.

The revised profile also includes changes received from the ITU for G.729 
(new annexes), G.723.1 (diagrams).  The latest revisions are available as 
draft-ietf-avt-rtp-new-08.txt and draft-ietf-avt-profile-new-09.txt.

The chairs also noted that we have still to complete the RTP interoperability 
statement.  A number of contact people were noted, and a meeting was arranged 
between the sessions to discuss this subject, and to assign work items (see 
below). Help from all implementors would be appreciated. The current drafts are draft-ietf-avt-rtp-interop-03.txt and draft-ietf-avt-profile-interop-01.txt.

There are also a number of drafts which accompany the RTP specification:

 - The draft-ietf-avt-rtptest-03.txt on RTP/RTCP testing strategies is
   unchanged from the previous meeting. It is intended for informational. 
 - The draft-ietf-avt-rtcp-bw-01.txt on RTCP bandwidth modifiers is
   also unchanged.  It is intended to be a proposed standard.  
 - The specification of the comfort noise payload is believed ready
   for working group last call.
 - The MIME type registration for the codecs referenced by the A/V
   profile (draft-ietf- avt-rtp-mime-03.txt) has been modified to add
   optional parameters for G.723.1 and G.729.  H263++ has been added
   with `profile' and `level'.  This draft is also intended for
   proposed standard. 

These drafts are believed to be complete.  Anyone who has concerns
about these drafts is urged to send comments to the mailing list.

Stephan Wenger made a brief presentation on the applicability of RFC
2429 to the next revision of H.263 -- known as H.263++ -- which is in
progress.  This revision adds a number of annexes to H.263:

 - Annex U: enhanced reference picture selection
 - Annex V: data partitioning
 - Annex W: enhanced supplementary information
 - Appendix II on profiles and levels

Of these, annex U does not affect the RTP packetization.  Annex V does
not affect the packetization so long as each slice is split into a
minimum of two RTP packets (header plus motion vectors, and up to 8
bits of the DCT info in one packet; rest of the DCT in another).
Annex W has many new features, the only one relevant to the RTP
packetization being header repetition.  This can be supported by
including the repeated header in the current picture header, so long
as the header is smaller than the maximum size imposed by RFC 2429.
The appendix on profiles and levels is intended to ensure
interoperability without very extensive capability negotiation, these
profile and levels are referenced by MIME parameters.  In summary, no
technical changes are required to RFC 2429 to support H.263++ although
some additional explanatory text on how to use the new features may be
useful.  It was suggested that this text could be added in an appendix
to the payload format spec; Stephan said this would be done for the
next meeting.

Steve Casner noted that MIME subtype video/h263-2000 was added to the
draft-ietf-avt-rtp-mime-03.txt spec for H.263++, so there are two MIME
subtypes referencing RFC 2429, but this does not introduce an
incompatibility since old implementations would not understand the new
annexes.

Tmima Koren presented enhancements to IP/UDP/RTP Header Compression
(draft-ietf-avt-crtp-enhance-00.txt). There are three drafts which depend
on these enhancements, two presented in AVT and a third on voice over MPLS.
The change since the previous meeting is an additional section on bandwidth
efficiency.  Feedback is solicited on whether all, some, or none of these
enhancements should be included in a revision of RFC 2508 for Draft
Standard, especially from implementors of 2508 as to whether they would
plan to implement these enhancements.

Tmima also presented IP/UDP/RTP Header Compression over AAL2
(draft-buffam-avt-crtp-over-aal2-00.txt).  This is an alternative to
the CRTP/PPP/PPPMUX/AAL5 multiplex (draft-ietf-avt-tcrtp-01.txt) which
AVT working group has agreed upon as the recommendation for
multiplexing of RTP streams.  The AAL2 method has comparable overheads
for small voice packets but is more robust to cell loss.  This is
outside the scope of the AVT working group since what is required is a
new ATM AAL2 SCCS identifier, but was presented here so the WG would
be aware of this work.  There was a comment that this falls into ITU-T
Study Group 13 Question 5.  A second comment was that this scheme
would work for other RTP header compression schemes, such as ROCCO.

Mooi Chuah presented a Light Weight IP Encapsulation (LIPE) Scheme
(draft-chuah-avt-lipe-01.txt).  This is another multiplexing proposal
aimed at reducing the bandwidth used and processing load (compared
with the header compression/tunneling in the TCRTP framework) at the
expense of reduced functionality.  Although presented as a means of
transporting voice frames, this work is actually a general purpose
layer 2 multiplex based on a new transport protocol as an alternative
to RTP.  This is outside the scope of the AVT working group charter,
so it would require a new charter or new working group.  There was a
comment that introducing a new transport protocol, with a new IP
protocol number, is a step that must considered carefully.

Timur Friedman presented some RTCP reporting extensions
(draft-friedman-avt-rtcp-report-extns-01.txt).  These allow reporting
of considerably more detailed statistics than the standard SR/RR
packets: loss bitmaps (RLE encoded), receiver timestamp, and delay
since last receiver report.  These last two allow receivers to
calculate the round trip time to senders -- it was suggested that
allowing receivers to send SR packets with the packet and octet count
fields set to zero would have similar effect.  The major open issue is
the use extended SR/RR blocks (the current approach) compared to the
definition of new packet types.  The authors are of the opinion that
defining new RTCP packet types would be cleaner, and the consensus of
the room agreed with them.

The next section of the meeting was devoted to discussion of RTCP
backchannel proposals.  Colin Perkins introduced this by noting the
issues to be considered during the presentations:

 - Restrict to unicast?  Or allow limited multicast?
 - How is the timing of RTCP handled?  How does this affect existing
   applications?
 - Are backchannel messages sent in compound RTCP packets or not?  How
   does this affect the system?
 - Should backchannel packets be sent to the same port as normal RTCP?
 - Use multiple RTCP packet types or one plus subtypes?
 - Congestion control?
 - Can repair data packets be distinguished from normal packets?  Does
   it matter?
 - Should we define one or many new profiles? Can we merge these drafts?

The first RTCP backchannel proposal (draft-ietf-avt-rtprx-00.txt) was
an RTP profile for RTCP-based retransmission request for unicast
sessions presented by Koichi Yano.  The changes made since the
previous meeting include additions to the NACK packet format:

 - some payload specific fields which give flexibility for different
   deployment, but a default format is specified.
 - R-bit (Request bit) is for congestion control.  It allows the
   receiver to give feedback on packet loss via a NACK, but by setting
   this bit to zero the sender does not retransmit the lost packet
   (and hence does not use bottleneck bandwidth).

There are also a number of changes to the sender and receiver
behavior: 

 - A receiver should send NACKs for all lost packets (using the R bit
   to prevent retransmission, if necessary)
 - The loss fraction in receiver reports should take into account only
   original data loss
 - Senders should send retransmissions as soon as possible after a
   NACK is received
 - Calculation of metrics for congestion control is to be based on
   fine grained RTCP feedback, but no specific scheme is proposed.

Finally, a brief explanation of the changes needed to SDP and
discussion of deployment with existing payload types was made.  There
have been no changes to the RTCP interval recommendation.

A number of issues were raised for discussion: On congestion control
it was noted that although the draft describes how to calculate
metrics like loss or RTT, it does not propose and particular scheme,
should it?  The author observed that it may not be possible to specify
a single scheme because congestion control is dependent upon the
payload and the environment.  Also, are NACKs sufficient or might
extended RR packets be needed also?  RR is needed for sender RTT
calculation, so rather than duplicating this functionality in the NACK
packet, the author prefers to use both NACK and RR.  What about use of
this format for other purposes (FIR, NEWPRED)?  The requirements for
the RTCP interval should be similar, but should a new RTCP (sub-)type
be defined?  Should this profile be more general?  This is to be
resolved in discussions among the authors of these drafts.

It was asked if the authors had considered including information in
the NACK packets to indicate the length of time for which a
retransmission is useful?  Yes, but the desire to keep the protocol
simple was thought to outweigh the advantages of this (and Dave Singer
noted that so long as the server is consistent, the receiver can
figure out if the retransmission will be received in time and use the
R bit appropriately).  How big a fraction of the RTP bandwidth is used
with small packets?  The authors discuss use with 1 kByte packets, but
audio packets are a lot smaller than this.  The fraction would be
higher with small packets, but Steve Casner noted that it may be
acceptable (in a unicast scenario) to use more control bandwidth if
that is the appropriate balance for the application.  He also noted
that we would like to have a single profile for retransmission, with
multiple payload formats as needed.  That profile should at least be
able to specify the RTCP timing in common for all the payload formats,
and if possible, common congestion control mechanisms as well, though
perhaps with different RTT calculation methods.  The profile may
support multicast as well as unicast.  Koichi countered that we know
how to do unicast now, but multicast is not so clear, so it may have
to be addressed later.

Roger Kermode noted that this work, and that in the other RTCP
backchannel proposals, overlaps with that being done in the reliable
multicast transport (RMT) working group and that coordination between
the groups would be beneficial.  The AVT chairs agree, and urge the
authors of the backchannel drafts to study the work ongoing in RMT.

The next presentation (draft-miyazaki-avt-rtp-selret-01.txt) was by
Carsten Burmeister: an RTP Payload Format for Multiple Selective
Retransmissions.  It was noted that there patent applications exist
which may pertain to this draft - an IPR statement has been received,
and will be lodged on the IETF web site in the usual manner.  This
proposal is a payload format which would fit well with the profile
specified in the previous proposal.  This work allows for multiple and
selective retransmissions in an attempt to solve the following
problems: how does the receiver know if a lost packet should be
retransmitted (since not all packets are of equal importance)?  How
does the receiver detect a lost retransmitted packet?

To solve these problems, the concept of a second sequence number (SSN)
is introduced.  At the sender each packet is marked with an
SSN-indicator and SSN.  If the packet is `important' and should be
retransmitted if lost, the SSN-indicator is set to one and the SSN is
incremented; otherwise the SSN-indicator is set to zero and the
previous value of the SSN is included in the packet.  On reception,
the receiver compares the SSN in the packet with the stored value from
the previous packet: if it is the same, nothing important has been
lost.  If it equals the previous value plus one, and the SSN-indicator
is one, an important packet has been received correctly.  If it equals
the previous value plus one, and the SSN-indicator is zero, an
important packet - with SSN equal to that in the just received packet
- was lost and a retransmission request should be made.  The claimed
advantages of this approach are selective retransmission (only
important packets are retransmitted) and support for multiple
retransmissions (retransmitted packets get a new SSN value, the
receiver is able to detect lost retransmissions).

It was asked what happens if the NACK packet is lost, is there a
timeout to retransmit it?  No, if the receiver gets other packets with
a SSN, it knows the packet is still lost and can retransmit the NACK.
How about using ACK which is more reliable if the error rate is high.
Yes, but ACK takes more bandwidth.

Stephan Wenger noted that there are only two priority groups here, and
it is not easily enhanced to more than two groups.  Maybe more than
two levels are needed?  The authors believe there are only two
meaningful priorities: retransmit or don't.  Steve Casner said that if
there are several packets lost and a limited retransmission budget it
may make sense to choose to send the most important vs.  medium
importance packets.  He also noted that it may be possible to achieve
the same effect using layering by sending "important" packets on a
different port, with a separate sequence number space, and that this
might avoid the need for the second sequence number.

Next, Stephan Wenger presented RTCP-based feedback for predictive video
coding (draft-wenger-avt-rtcp-feedback-00.txt).  This draft discusses
why feedback is needed for predictive video coding, and how to provide
feedback in a timely manner for both multicast and unicast sessions.
The motivation for support of multicast and unicast together is to
avoid specialized solutions wherever possible, and to provide a
solution which degrades smoothly as the number of members increases.
This is "the RTP way".

The draft defines a number of feedback message types: picture loss
indication, slice loss indication and reference picture selection
indication.  The choice of when to send feedback is based on:

 - avoiding a single receiver which can monopolize the mechanism
 - avoid increasing the overall RTCP bandwidth
 - avoid feedback implosion
 - allow timely feedback if possible
 - support multicast (with graceful degradation as group size
   increases)

The result is a formula which allows for sending after a random
interval which is a function of the group size, retaining timely
feedback for unicast sessions.  Feedback requests inhibit other
receivers, preventing implosion.

There are several open issues on which comments from the WG are
solicited: Do we want to use feedback suppression, as in SRM, or do we
want feedback from multiple receivers to know how bad the loss problem
is?  The spec currently defines feedback only for video, but it should
work for other media as well, should we do that?  The proposal
currently specifies that the feedback message is always combined with
RTCP RR; should we do that?  Or should the RTCP bandwidth budget be
split between the two, maybe 50/50?

Steve Casner asked if the needed RTT calculation is possible with the
delay mechanisms available?  Yes, pretty sure this is true.  It was
asked if the draft assumes real time encoding, rather than streaming
>from a file where the buffer can be bigger?  Yes, this is mostly
intended for real-time.  Henning Schulzrinne noted that may be quite
useful for one of the recurring discussions in the SIP working group
regarding DTMF tones, where there are different perceptions on whether
FEC or retransmission is appropriate.  analysis to see which is
better.  Does this assume only one packet loss per interval since the
second packet cannot be NACKed?  Yes.  Jörg Ott replied that in
unicast multiple NACKs are allowed still keeping within the bandwidth
limit, but in multicast they are not.  That is the tradeoff for
scaling, and another receiver will likely send the needed NACK anyway.
Simulations showed this works well unless the loss is as high as 10%,
and congestion control rules may say you should cease transmission
above that point.

Finally in this section of the meeting, was a presentation by Shigeru
Fukunaga on low delay RTCP for backward messages
(draft-fukunaga-low-delay-rtcp-00.txt).  This is the RTCP format for
NEWPRED backchannel messages which was extracted from
draft-ietf-avt-rtp-mpeg4-es-02.txt.  The main features are as follows:

 - supports unicast sessions only
 - LD_RTCP is sent as soon as possible (without delay, even if this
   means violating the standard RTCP timing rules)
 - LD_RTCP packets are not included in the calculation of the send
   time for other RTCP packets
 - only LD_RTCP packets are included in one compound LD_RTCP packet
   (how does this affect compatibility?  This rule is different from
   that for other packet types)
 - congestion control (actually rate control) since total amount of
   traffic is constrained to under 5% of RTP rate

A number of issues were noted for discussion: the features of this
proposal are similar to Koichi Yano's proposal - e.g.  low delay,
unicast - and it is possible to use that proposal as a profile of low
delay RTCP after some modifications (define NEWPRED RTCP under RTP-RX
profile).  What about Stephan Wenger's proposal?  Not considered low
enough delay which is very important for the performance of NEWPRED;
does not use ACK messages; and multicast may not be useful for
NEWPRED.  Stephan disputes the claim that very low delay is required,
and reminded the list that he had recently sent a spreadsheet to the
mailing list with his analysis.  For a codec running at 10 frames per
second, an extra 100ms delay only increases the size of the refresh
frame from 2x to 2.5x the normal frame size.

Steve Casner pointed out this disagreement will not be settled in the AVT
meeting; the answer must come from those doing the video compression work,
which may mean from another body.  If there is no single answer, it may
mean we have to specify multiple mechanisms.  The chairs asked the authors
of the RTCP backchannel drafts to consider if they could be merged into a
single specification or at least a common framework/profile that can work
for all these schemes, since there is believed to be much overlap between
the proposals.  This needs to be done soon because of timing constraints
for working with other bodies, so the authors were requested to meet later
in the day and report back in the second AVT session.  The authors agreed
to discuss this in a separate meeting, and report back to the second
session of the AVT meeting (see below).  Ron Frederick commented that it
should at least be possible to agree on a common RTCP NACK/ACK packet
format, possibly for either unicast or multicast.

Ross Finlayson presented a more loss-tolerant RTP payload format for
MP3 audio (draft-ietf-avt-rtp-mp3-02.txt).  This format is a data
preserving rearrangement of MP3 frames which removes backpointers
making the result more loss tolerant, and optionally provides for
interleaving.  Changes since the previous draft are that each ADU is
not preceeded by a descriptor giving its length, and the packing is no
longer compatible with RFC 2250, since the 11 bit syncword is used for
interleaving.  It was already the case that a 2250 receiver could not
process the MP3 payload format without the ADUs being rearranged, so
all that is given up by this change is some potential for code re-use.
The interleaving scheme provides for an explicit description of the
ADU order, with an interleave index and cycle count.  This differs
>from the general interleaving proposal submitted on the mailing list
by Orion Hodson, which is not specific to a particular codec and gives
an algorithmic description of the interleaving.  There was some
discussion over the merits of a general purpose interleaving format vs
inclusion into particular formats.  Ross claims that explicit
interleaving is a benefit for MP3 because of the variable frame rate
and variable number of frames per packet.  He would also like to avoid
a dependency on progress of Orion's proposal.  Feedback on this would
be appreciated.

Future steps include an update of the implementation to use the new
ADU descriptor (and an open source code release), plus further tests
of the interleaving.  Ross has submitted a minor revision to the draft
to change the number of bits in the interleave field.  This payload
format is thought to be ready for working group last call once the
draft gets posted.

The second session started with a discussion of unequal error
protection and FEC which was held over from the previous day due to
lack of time.

The first presentation was by Adam Li on an RTP Payload Format for
Generic FEC with Uneven Level Protection (draft-li-ulp-00.txt).  It
was noted that data in multimedia packets is not all of the same
importance, and with some common encodings, the most important data is
at the beginning of a packet.  Hence this draft proposes a scheme for
protecting just that data, and for layered protection (where the data
earlier in the packet has more protection than that later), which is
derived from the RFC 2733 parity FEC format.

It was asked how format independent this payload format is?  The encoder
needs to know the format of the media stream, the decoder does not, and it
can be used for any format with the required properties. Jonathan Rosenberg
asked a similar question: how useful is this?  He is not so sure that many
codecs have the most important information at the start of the packet (e.g.
H.263 has the data spread throughout, although the H.263++ extensions
change this).  Defining a new generic payload format may not be worthwhile
if it is just applicable to H.263++.  Reha Civanlar suggested layered
coding; both techniques require partitioning of the data, but layered
coding allows differential network QoS as well.  Concern was raised that
there may be IPR issue with this work - clarification is sought.

Bernhard Wimmer presented draft-lnt-avt-uxp-00.txt, an RTP Payload
Format for Erasure-Resilient Transmission of Progressive Multimedia
Streams.  The aims of this work are similar to those of the previous,
except that an interleaved Reed-Solomon encoding is adopted (rather
than parity FEC).  Ron Frederick asked how much knowledge of the
encoding scheme needs to be transmitted?  How does one signal the
amount of redundancy used per row?  They have defined a scheme to
signal how much redundancy is transmitted in each row.  It is dynamic
>from frame to frame.  Dave Singer asked about the effects of this
encoding on congestion control: when multimedia data is a significant
fraction of the traffic, increasing the bandwidth for redundancy
coding just causes more loss.  Has there been an analysis of what
would happen if there were many streams using this scheme and filling
a link?  No.  Colin Perkins pointed out that we in AVT need to address
congestion control.  Over and above the basic requirements on RTP,
this payload format has a feedback channel which adds data in the
presence of congestion.  Allison Mankin said an option might be that
this is dependent upon using Endpoint Congestion Management (ECM WG).

Since this draft was not offered in accordance with section 10 of RFC
2026, the chairs declined to progress it in the working group at this
time.  The authors were requested to clarify the IPR issues.  Wimmer
said a clarification would be sent to the mailing list.

The next area of discussion was the RTP payload format for the AMR
speech codec.  This was opened by discussion of the liaison statement
received from ETSI SMG2 (now part of 3GPP) delivered by Peter Barany.
The liaison statement raises a number of procedural questions for the
group:

 - Is the schedule for completion of the AMR payload format sufficient
   to meet the ETSI standardization deadline (December 2000)?  Yes, we
   believe so.
 - How does the AVT working group plan to interface to ETSI?  By
   members of AVT attending ETSI meetings, and vice-versa.

In addition, concern was expressed that the proposed payload format
may not be able to deal gracefully with a feature of the AMR codec,
where frames which have bit errors (for example, due to poor wireless
reception) are marked as damaged, but still transported.

There are two aspects to this latter question: can damaged frames be
delivered to an application, and can those frames be marked during
later transit?  The first issue is not one the AVT working group can
resolve, and is rather for the lower layer protocol stack designers
(e.g.  the recent UDP-Lite proposal); the second can be solved by use
of a `damaged frame' bit in the RTP payload header which is set when
the payload contains errors (there are cross layer issues to resolve,
since this bit must be set in transit, but again these are outside the
scope of the AVT working group).

Allison Mankin said that the UDP-Lite proposal did not get much
support from link layer folks, but maybe now this is evidence of a
link that would support it.  Allison suggested approaching the
transport area directors about following up on this, probably not in
AVT but possibly.

Ron Frederick noted that RTP translators and mixers would have to be
aware of this `damaged frame' bit also, and that the interactions here
are unclear (e.g.  when mixing erroneous and non-erroneous data).  It
was also noted that AMR includes the existing GSM FR/EFR codecs as a
subset, and the `damaged frame' indicator is needed there also, which
may be an issue.  Would these need to have revised payload formats
which are indicated by dynamic payload types.

Following this introduction we had presentations on the two proposed
payload formats for AMR.  First was draft-sjoberg-avt-rtp-amr-01.txt
presented by Johan Sjöberg.  This draft is a merger of the Ericsson
and Nokia drafts from the last meeting.  The main open issue among the
authors is that this format currently supports 4 different Comfort
Noise types.  We probably want to reduce this to just the one
AMR-specific CN format, and switch payload types if other CN formats
are needed.  Steve Casner endorsed this approach.

The second payload format is draft-fingscheidt-avt-rtp-amr-00.txt
together with a companion document that defines the MIME type for
storage and RTP transport, which were presented by Bernhard Wimmer.
This format provides for redundancy and parity FEC via a novel scheme.
It was asked why existing standards (e.g.  RFC 2198) could not be
used.  The claim is that the additional overhead of these more general
purpose schemes is not acceptable.  The issue is one of efficiency
versus generality, and it is unclear if the trade- off selected for
this draft is appropriate.  In addition, the format defines its own
frame type field: is was suggested that using RTP payload types here
would be more appropriate.

Henning Schulzrinne asked about IPR for this codec and availability of
reference code.  The code is downloadable from 3GPP, both fixed point
and soon floating point.  Bernhard did not know the IPR status.

The chairs noted that these two payload formats need to be merged into
a single proposal.  This is the primary gating item for meeting the
short timeline that was requested.  Bernhard said the authors had a
discussion before the meeting and should have a solution soon.

Ladan Gharai presented an RTP Payload Format for SMPTE 292M video
(draft-ietf-avt-smpte292-video-00.txt).  This is the serial digital
interface used for uncompressed HDTV at a data rate of 1.485 Gb/s with
source formats of SMPTE 260M, 295M, 274M and 296M.  The payload format
is relatively simple, although the high data rate makes for a couple
of unusual features: the RTP sequence number is extended to 26 bits in
the payload header, to give a wrap- around time of ~170 seconds (a 16
bit sequence number wraps in less than 1 second).  Also, unlike other
video formats, this one uses a 10MHz timestamp to allow for precise
timing reconstruction of the serial bitstream.

Steve Casner asked if the format really needs the extended sequence
number, since this may cause confusion with respect to the extended
sequence number in RTCP.  It may be that a one second wraparound time
is not an issue, since the network can't buffer that many packets, or
it may be feasible to infer the sequence number epoch from the
timestamp.  David Richardson noted that, whilst that may be so, the
larger space may be needed to reference frames in other packets for
error correction or retransmission, or for references between layers
in layered coding.  This is an area they are still working on.

Ladan also presented an AC3 audio payload format, draft-gharai-ac3-01.txt.  
AC3 is the audio format used with HDTV.  There was not much change to this
draft since it was last presented.  The authors are seeking comments, but
there are no specific open design issues.

Jeff Meunier presented draft-meunier-avt-rtp-dsr-00.txt, an RTP
Payload Format for transport of ETSI ES 201 108 Distributed Speech
Recognition streams.  This is similar to sending compressed audio
except that not all of the information required for playback is
included, just what is necessary for speech recognition processing.
Being new to RTP, the author solicited guidance on use of header
compression, SDP/SIP usage and gotchas, robustness to packet loss and
error concealment (in particular considerations for RTP over IP vs
mobile IP), and packetization.  In particular, the author is seeking
public domain RTP implementations to help them get started.

It was noted that maybe we don't need to send the multiframe header
which is defined in this draft, since it is currently constant.
Rather, it may be possible to omit it from the stream and send the
data out of band.  There was no out-of-band channel when the ETSI spec
was being written (for non-RTP channels).  If you gateway to a non-RTP
channel, the gateway would need to re-insert this info in-band.  A
second version of the ETSI standard may have time-changing parameters
in this header.

Steve Casner noted that the current draft seems reasonable, but asked
if the bandwidth savings (compared to using a normal codec) is
worthwhile.  Yes, in particular within a wireless environment, because
this encoding is 4.8kbps vs 32kbps needed to do this with a normal
audio codec of sufficient quality.  The saved bandwidth is needed for
other data exchange in parallel with the speech recognition.

It was asked how much support there is from speech recognition vendors
to use this common format.  They tend to be very guarded about their
implementations, and they don't care much about interoperation.  This
proposal has gotten some support, but it is a chicken-and-egg problem
between handset providers and application providers.  There is
agreement on this algorithm as the front-end.

The next presentation was preliminary work on an RTP Payload Format
(http://www.tcb.net/plip/draft-white-sonet-format-rtp-00.txt) to carry
SONET traffic.  The motivation is to provide transport of legacy SONET
data on IP networks where the lower layers do not support SONET, such
as IP over photon or 10 gigabit ethernet.  Although somewhat outside
the traditional focus of the AVT working group, this appears to be a
sensible application of RTP, in some ways related to the multiplexing
proposals we have received in the past which seek to emulate a T1
circuit.  The authors want to take advantage of the timing and
sequencing functions of RTP in addition to more advanced functions
like generic FEC.  There may have been some confusion about the
sequence number cycling back to 1 for the first packet of each frame,
but it was explained that the sequence number must increment
continuously across frame boundaries.

Ron Frederick explained why the M bit marks the last rather than the
first packet of a frame for video, and that this choice might be
helpful for similar reasons with this format.  It was also asked to
what extent does this format will preserve SONET SLA characteristics?
It is not intended that SONET error recovery mechanisms will be
supported.  Only the SONET payload will be carried in RTP, and the
overhead would be recreated at the far end.  IP error recovery
mechanisms (e.g., re-routing) would be used instead.  It was also
noted that a similar technique could be used to support transport of
ATM over RTP, should the need arise.

The final area for discussion in the meeting was a framework document
for the transport of MPEG-4 on IP (draft-singer-mpeg4-ip-00.txt),
presented by Dave Singer.  This is intended to provide a common
framework for transport of any part - or all - of the MPEG-4 system on
IP networks, to agree on that which can be agreed on, collect ideas
and current practise, and is intended to be non-controversial.  It
refers to existing drafts and specifications only, explaining how they
can be combined and what options exist.

In terms of RTP payload formats for MPEG-4, the framework document
suggests that one, simple, base-level scheme ought to be available
which can transport any MPEG-4 stream (possibly non-optimally).  Any
receiver ought to be able to receive this format, and it's good if
senders can be persuaded to send it.  However, the draft also endorses
the development of other formats which can be optimized for particular
media types.  MPEG are working on the development of a generic format,
to be based on draft-ietf-avt-rtp-mpeg4-03.txt, and we have a media
specific format in draft-ietf-avt-rtp-mpeg4-es-02.txt being developed
in the IETF.

Steve Casner noted that the chairs endorse this basic approach, and
the consensus of the meeting was also that this was valid (although it
was noted that some in the MPEG community would prefer a single format
only).  Several MPEG committee members were present.  Zvi Lifshitz
said he/they would take this approach back to MPEG to seek agreement.
The timetable is an MPEG-over-IP Ad Hoc group meeting in September and
a full MPEG committee meeting in October.

Guido Franceschini asked if we could resolve the choice between
"application" and "video" MIME type for MPEG-4.  Some people felt that
video is appropriate because MPEG-4 is used primarily for visual
presentations, but others counter that MPEG-4 can carry active content
such as Java.  Discussion continues on the mailing list.

Allison Mankin noted that this generality has caused considerable
concern in the IESG regarding security due to the ability of MPEG-4 to
transport active content (Java, ECMAscript, etc) and other potentially
dangerous media.  This aspect of any MPEG-4 payload format(s) will be
reviewed thoroughly.  In particular, in the view of the Area Directors
the current Security Concerns section of the generic format
(draft-ietf-avt-rtp-mpeg4-03.txt) is not sufficient.  Allison asked
that the next versions of the drafts posted for the WG have enhanced
security sections so they may be reviewed.  Colin asked for comments
on all the drafts, both complaints and statements of support so we
know which direction to take.

The discussion on RTCP backchannel messages in the first day's meeting
led to a separate meeting of the authors of those drafts, to consider
how they could be merged.  Jörg Ott presented a summary of these
discussions.  It was agreed that there is a good set of aggregate
functionality in these documents, but the authors couldn't agree on
what should be in the baseline document.  The major issue is whether
to include support for multicast.  Some desire two drafts: one purely
for unicast, one which also supports multicast.  It was noted that
timing is important due to market pressure, and multicast is perceived
by some to take longer consideration.  Steve Casner noted that the
chairs do not want to make an assumption that this will be two drafts
>from the start, we want to try to make a single document which
supports both multicast and unicast.  The authors agreed to try to
merge the drafts, with a target completion date of December 2000 (last
call after next IETF meeting).

Colin Perkins presented a summary of the discussions on completing the
RTP interoperability matrix, which occurred between the two sessions.
It is believed that we now have a good handle on the RTP protocol
interoperability statement, with a number of implementors volunteering
to provide input.  We are less advanced with the profile
interoperability statement, although some progress was also made
there.  Of particular importance is that implementors of the more
advanced audio and video payload formats provide test results, since
this is an area where we are short of data.

The meeting concluded with a recap of the action items and a call for
acceptance of new work items.  Those present agreed to that all of the
following list of payload format proposals should be accepted as AVT
work items and be resubmitted as AVT drafts: multiple selective
retransmission, generic FEC with ULP, AC3 audio, DSR and SONET.
Confirmation of this agreement must come from the mailing list, so
anyone who believes any of these proposals are not appropriate for AVT
work items is requested to comment on the mailing list.




From rem-conf Sun Oct 15 22:37:51 2000 
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From rem-conf Mon Oct 16 04:51:49 2000 
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RGVhciBhbGwsDQoNCkFzIGl0IGlzIHBsYW5uZWQsIHdlIHdpbGwgaGF2ZSBhIG1lZXRpbmcgb24g
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From rem-conf Mon Oct 16 05:10:43 2000 
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Thanks, will replay tomorrow; will be there.

Regards


Paul Christ

> -----Original Message-----
> From: Young-Kwon LIM [mailto:young@techway.co.kr]
> Sent: Montag, 16. Oktober 2000 13:43
> To: 4on2andIP-sys@advent.ee.columbia.edu
> Cc: rem-conf@es.net
> Subject: AHG meeting announcement
> 
> 
> Dear all,
> 
> As it is planned, we will have a meeting on Saturday before La 
> Baule meeting. Meeting will start at 1:00 pm. There is no  clear 
> meeting plan for Sunday yet. But it will be decided on site to 
> extend the meeting to Suday. Detailed agenda will be posted soon 
> with the AHG report. Please let me know who is preparing what 
> contribution for the next meeting.
> 
> Sincerely,
> Young.
> 



From rem-conf Mon Oct 16 06:25:37 2000 
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Since I'll not be on Saturday I add my voice to extending the meeting into 
Sunday. I may supply a contribution.

Thanks,

z


-----Original Message-----
From:	Young-Kwon LIM [SMTP:young@techway.co.kr]
Sent:	Monday, October 16, 2000 1:43 PM
To:	4on2andIP-sys@advent.ee.columbia.edu
Cc:	rem-conf@es.net
Subject:	AHG meeting announcement


Dear all,

As it is planned, we will have a meeting on Saturday before La Baule 
meeting. Meeting will start at 1:00 pm. There is no  clear meeting plan for 
Sunday yet. But it will be decided on site to extend the meeting to Suday. 
Detailed agenda will be posted soon with the AHG report. Please let me know 
who is preparing what contribution for the next meeting.

Sincerely,
Young.




From rem-conf Mon Oct 16 09:14:08 2000 
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Date: Mon, 16 Oct 2000 11:57:43 -0400
From: "Narasimhan, Sam (SD-EX)" <SNarasimhan@gi.com>
Subject: RE: AHG meeting announcement
To: 'Young-Kwon LIM' <young@techway.co.kr>, 4on2andIP-sys@advent.ee.columbia.edu
Cc: rem-conf@es.net
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 I will attend the meeting. Can you let us know
where will the meeting be held - is it at a hotel or
in the convention center.

Best Regards
Sam Narasimhan
Motorola

-----Original Message-----
From: Young-Kwon LIM [mailto:young@techway.co.kr]
Sent: Monday, October 16, 2000 4:43 AM
To: 4on2andIP-sys@advent.ee.columbia.edu
Cc: rem-conf@es.net
Subject: AHG meeting announcement


Dear all,

As it is planned, we will have a meeting on Saturday before La Baule
meeting. Meeting will start at 1:00 pm. There is no  clear meeting plan for
Sunday yet. But it will be decided on site to extend the meeting to Suday.
Detailed agenda will be posted soon with the AHG report. Please let me know
who is preparing what contribution for the next meeting.

Sincerely,
Young.



From rem-conf Mon Oct 16 09:57:11 2000 
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Reply-To: "Young-Kwon LIM" <young@techway.co.kr>
From: "Young-Kwon LIM" <young@techway.co.kr>
To: "Narasimhan, Sam \(SD-EX\)" <SNarasimhan@gi.com>,
	<4on2andIP-sys@advent.ee.columbia.edu>
Cc: <rem-conf@es.net>
References: <973597126BDDD11197AA00805FA7EBC9034239FE@ntas0026.gi.com>
Subject: Re: AHG meeting announcement
Date: Tue, 17 Oct 2000 01:00:07 +0900
Organization: MP4CAST
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From rem-conf Mon Oct 16 09:59:24 2000 
From rem-conf-request@es.net Mon Oct 16 09:59:23 2000
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From: CURET Dominique FTRD/DMI/REN <dominique.curet@rd.francetelecom.fr>
To: "'Narasimhan, Sam (SD-EX)'" <SNarasimhan@gi.com>, 'Young-Kwon LIM'
	 <young@techway.co.kr>, 4on2andIP-sys@advent.ee.columbia.edu
Cc: rem-conf@es.net
Subject: RE: AHG meeting announcement
Date: Mon, 16 Oct 2000 18:56:24 +0200
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at Atlantia,
CORMORANS room


_________________________________________
> 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: Narasimhan, Sam (SD-EX) [mailto:SNarasimhan@gi.com]
Date: lundi 16 octobre 2000 17:58
=C0: 'Young-Kwon LIM'; 4on2andIP-sys@advent.ee.columbia.edu
Cc: rem-conf@es.net
Objet: RE: AHG meeting announcement



 I will attend the meeting. Can you let us know
where will the meeting be held - is it at a hotel or
in the convention center.

Best Regards
Sam Narasimhan
Motorola

-----Original Message-----
From: Young-Kwon LIM [mailto:young@techway.co.kr]
Sent: Monday, October 16, 2000 4:43 AM
To: 4on2andIP-sys@advent.ee.columbia.edu
Cc: rem-conf@es.net
Subject: AHG meeting announcement


Dear all,

As it is planned, we will have a meeting on Saturday before La Baule
meeting. Meeting will start at 1:00 pm. There is no  clear meeting plan =
for
Sunday yet. But it will be decided on site to extend the meeting to =
Suday.
Detailed agenda will be posted soon with the AHG report. Please let me =
know
who is preparing what contribution for the next meeting.

Sincerely,
Young.



From rem-conf Mon Oct 16 10:24:22 2000 
From rem-conf-request@es.net Mon Oct 16 10:24:21 2000
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Date: Mon, 16 Oct 2000 09:45:19 -0700
To: "4on2andIP-sys@advent.ee.columbia.edu" <4on2andIP-sys@advent.ee.columbia.edu>
From: Dave Singer <singer@apple.com>
Subject: RE: AHG meeting announcement
Cc: "'Young-Kwon LIM'" <young@techway.co.kr>,
        "rem-conf@es.net" <rem-conf@es.net>
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At 3:23 PM +0200 10/16/00, Zvi Lifshitz wrote:
>Since I'll not be on Saturday I add my voice to extending the meeting into
>Sunday. I may supply a contribution.
>
>Thanks,
>
>z

I also cannot arrive until late Saturday, expecting a one-day ad-hoc 
on Sunday, so some meeting that day would be good.

>
>
>-----Original Message-----
>From:	Young-Kwon LIM [SMTP:young@techway.co.kr]
>Sent:	Monday, October 16, 2000 1:43 PM
>To:	4on2andIP-sys@advent.ee.columbia.edu
>Cc:	rem-conf@es.net
>Subject:	AHG meeting announcement
>
>
>Dear all,
>
>As it is planned, we will have a meeting on Saturday before La Baule
>meeting. Meeting will start at 1:00 pm. There is no  clear meeting plan for
>Sunday yet. But it will be decided on site to extend the meeting to Suday.
>Detailed agenda will be posted soon with the AHG report. Please let me know
>who is preparing what contribution for the next meeting.
>
>Sincerely,
>Young.

-- 
David Singer
Apple Computer/QuickTime



From rem-conf Mon Oct 16 14:32:24 2000 
From rem-conf-request@es.net Mon Oct 16 14:32:23 2000
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Date: Mon, 16 Oct 2000 14:15:47 -0700
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From: Florissa Colina <florissa@bmrc.berkeley.edu>
Subject: 10/18  DTV Application Software Environment and Interactive TV
  Platforms - Aninda DasGupta, Philips Consumer Electronics
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Berkeley Multimedia, Graphics and Interfaces Seminar

DTV Application Software Environment and Interactive TV Platforms

October 18, 2000,
1:10-2:30 p.m.
Fujitsu Seminar Room (405 Soda Hall) 

Aninda DasGupta 
Philips Consumer Electronics

Digital television systems are capable of delivering rich media services to
viewers.  With the use of Web and Java content, program producers are
creating exciting interactive television content, and service providers are
launching data broadcast services, over terrestrial, cable and satellite
networks.  In this talk we will briefly take a look at the evolution of
interactive TV solutions.  You will then be introduced to the DTV
Application Software Environment (DASE), which is a proposed middleware
standard that provides platform independence for all applications to run
uniformly over all brands and models of DTV receivers.  If time permits,
some demonstrations will be shown.

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:

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 Mon Oct 16 22:07:49 2000 
From rem-conf-request@es.net Mon Oct 16 22:07:47 2000
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Message-ID: <33D69E5EAD06364FB98F3562E62ACE890D96C7@red-msg-02.redmond.corp.microsoft.com>
From: Gary Sullivan <garysull@microsoft.com>
To: "'rem-conf@es.net'" <rem-conf@es.net>, 
	"'itu-adv-video@standard.pictel.com'" <itu-adv-video@standard.pictel.com>
Cc: "Joerg Ott (E-mail)" <jo@TZI.UNI-BREMEN.DE>, 
	"Stephan Wenger (TUB) (E-mail)" <stewe@cs.tu-berlin.de>, 
	"Stephen Casner (IETF) (E-mail)" <casner@acm.org>, 
	"Simao Campos (Comsat) (E-mail)" <simao@ctd.comsat.com>, 
	"'fabio.bigi@itu.int'" <fabio.bigi@itu.int>
Subject: Collaborative Letter from ITU-T Advanced Video Coding Experts Gro
	up
Date: Mon, 16 Oct 2000 21:56:51 -0700
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The following collaborative letter was drafted by the ITU-T Advanced
Video Coding Experts Group at its last meeting (Late August).
I am sending it to the rem-conf group reflector as a means of
informally conveying the contents to the relevant IETF working group.
Attached is the draft document that is referred to in the letter.
I can be contacted for any further information regarding the
contents of this letter or regarding other issues relating to
the work of the ITU-T Advanced Video Coding Experts Group:

Document: Q15-K-56  (24 Aug. 2000)
Question: Q.15/SG16
Source:   ITU-T Advanced Video Coding Experts Group
          Gary Sullivan, Q.15/SG16 Rapporteur
          Microsoft Corp., One Microsoft Way, Redmond, WA 98053
Tel:      +1 (425) 703-5308
Fax:      +1 (425) 936-7329
Email:    Gary.Sullivan@itu.int

Title:    Collaborative Letter to IETF AVT and MMUSIC Working
          Groups on Profile and Level definitions for ITU-T
          H.263 Standard Video Codec

We are writing this collaborative letter to inform you of the current
status of work in the ITU-T Advanced Video Coding Experts Group (ITU-T
Q.15/SG16) on video codec design for IP-network applications.

In an effort to provide configurations of video coding features that are
customized to specific key applications and to simplify capability
negotiation, we have drafted a new annex for Recommendation H.263
which defines specific normative "profiles" and "levels" of decoder
capabilities.  These profiles and levels will be defined in a normative
manner particularly suitable for use in simplified capability exchange
scenarios, specifically including SIP.  Our plan for the development
of this new Annex (draft attached) is to achieve preliminary
("determination") approval at the November, 2000 meeting of ITU-T
Study Group 16, and then to achieve final ("decision") approval a few
months later.  This new annex is intended to replace the current
informative definitions in H.263 Appendix II.

The draft annex includes definition of a "Conversational Internet"
profile designed specifically to address real-time conversational
services on IP networks.  Also included is a "High Latency" profile
designed for non-conversational services which should also be
suitable for many IP network applications.

-- END --


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`
end



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From rem-conf Tue Oct 17 07:12:09 2000 
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Subject: DDMA workshop CFP
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--=====================_971783547==_
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Dear Colleague,

You are invited to submit a paper to the International "Distributed Dynamic
Multiservice Architectures" (DDMA) Workshop. This workshop is in conjunction
with the 21st International Conference on Distributed Computing Systems
(ICDCS2001), one of the major conferences in the Distributed Computing
Systems Area.

DDMA workshop covers aspects concerning adaptive architectures and service
composition and management.

The submission deadline is November 1st. Further details are at the web
site http://ddma.enst-bretagne.fr/. A text version is also in attachment.

The DDMA workshop will be held in Phoenix, Arizona; April 16-19, 2001.
The proceedings of the workshop will be published by IEEE CS press.

[We apologize if you receive multiple copies of this announcement]
--=====================_971783547==_
Content-Type: text/plain; charset="iso-8859-1"
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					Call for Papers

				International Workshop on
			Distributed Dynamic Multiservice Architectures
				http://ddma.enst-bretagne.fr/

				in conjunction with ICDCS2001
				sponsored by IEEE Computer Society

				April 16 =96 19, 2001
				Phoenix, Arizona, USA

  The growing need to integrate new services while complying with the=
 requirements of end=20
users puts strong emphasis on the management and combination of future=
 services and the QoS=20
they should be able to provide and guarantee. Dynamic Multiservice=
 Architectures constitute a=20
challenging paradigm for distribution theory and practice due to their=
 unique requirements in terms=20
of adaptability, behavioural integration, separation of concerns, service=
 composition and service=20
management. In particular, the deployment of multimedia telecommunication=
 services is a highly=20
challenging area where those paradigms should apply because advanced=
 services require that the=20
user's profiles, preferences and context be taken into account. "Distributed=
 Dynamic=20
Multiservice Architectures" hence add a new dimension to distributed=
 computing to help the=20
composition and deployment of new services guaranteeing final user's=
 requirements.
  The goals of this workshop are to bring together users and researchers to=
 present their=20
recent work related to diverse aspects of dynamic multiservice=
 architectures, their fundamental=20
issues, their paradigms and some appropriate approaches and experiments. The=
 workshop will=20
thus present opportunities for discussing further evolutions and their=
 expected benefits.

Scope of Workshop
-----------------
The scope of this workshop encompasses but is not limited to :

Evolution of distributed multiservice architectures
	Separation of aspects and composition of aspects.
	Methodological approaches for aspect oriented programming.
	Components as cornerstones for dynamic composition.
	Composition requirements for reflection.
	Auto-adaptive ad hoc architectures and protocols.
Impact on the management and control
	Management of adaptive architectures
	Management and monitoring of composed services.
Platforms, tools, languages, ...=20
	Languages, tools and platforms for auto-adaptive systems.
	Environments for specification and design of dynamic architectures.
	Experience with design, implementation and use of auto-adaptive systems.
Application to telecommunication services
	Open critical issues in terms of composition of services.
	Panorama of QoS integration approaches for multimedia telecom services.
	Paradigms and requirements for telecom services architectures.
	Architectures for 3rd generation multimedia services.
	Distribution protocols for telecom (WAP, CORBA, CAMEL, mobile agents).

Submission Instructions and Workshop Format
-------------------------------------------
	Interested participants should submit a 4-8 pages position paper that=
 focuses on one of the=20
relevant topics or related issues. Papers will be reviewed by the program=
 committee. Electronic=20
submissions should be sent to zied.choukair@enst-bretagne.fr. Mail body=
 should include name,=20
address, affiliation and a brief biography along with paper's title and=
 abstract.
	The workshop will follow the following format: The morning and part of the=
 afternoon will=20
be dedicated to presentations and discussions of accepted proposals,=
 including demonstrations of=20
existing software. The afternoon session will end with an identification of=
 issues, followed by=20
discussion. The organisers will then elaborate a summary of the workshop=
 based upon a summary=20
of the papers and the major outlined points of the discussion.

Program Committee Members :
	Mehmet Aksit, Twente University, NL
	Zi=E8d Choukair, ENST Bretagne, FR
	Jaime Delgado, Universitat Pompeu Fabra, ES
	Byung-Guk Kim, University of Massachusetts Lowell, USA
	Marwan Krunz, University of Arizona, USA
	Abdul H. Sadka, University of Surrey, GB
	Antonio Rito Silva, University of Lisbon, PT
	Timothy K. Shih, Tamkang University, TW
	Katsuya Tanaka, Tokyo Denki University, JP

Important Dates :
	Paper submission due : 	Nov 1, 2000
	Notification : 		Dec 30, 2000
	Camera ready : 	 	Jan 20, 2001
	Conference : 		April 16-19, 2001

Web Pages :
	Conference URL : http://cactus.eas.asu.edu/ICDCS2001/

--=====================_971783547==_--




From rem-conf Tue Oct 17 08:38:53 2000 
From rem-conf-request@es.net Tue Oct 17 08:38:52 2000
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From: The IESG <iesg-secretary@ietf.org>
SUBJECT: Last Call: RTP payload format for MPEG-4 Audio/Visual streams
	 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 MPEG-4 Audio/Visual streams
<draft-ietf-avt-rtp-mpeg4-es-05.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 October 31, 2000.

Files can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-mpeg4-es-05.txt




From rem-conf Tue Oct 17 09:42:56 2000 
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From: "Bratakos, Maroula" <mbratakos@nuera.com>
To: "'rem-conf@es.net'" <rem-conf@es.net>
Subject: RTCP BYE
Date: Tue, 17 Oct 2000 09:38:22 -0700
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What is the role of the RTCP BYE when a control protocol, such as SIP, is
being used to create and terminate sessions?

Thanks in advance.
Maroula Bratakos



From rem-conf Tue Oct 17 10:57:19 2000 
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From: "Mozumder, Monir" <m_mozumder@trillium.com>
To: "'Bratakos, Maroula'" <mbratakos@nuera.com>,
        "'rem-conf@es.net'"
	 <rem-conf@es.net>
Subject: RE: RTCP BYE
Date: Tue, 17 Oct 2000 10:11:11 -0700
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Hi Maroula,

I think (correct me if wrong) the answer lies in layering principles. SIP
sits in upper layer which uses the service of RTP. When SIP sends
terminating commands, it will direct RTP to send RTCP BYE, which will
ultimately indicate the other SIP side RTP(s) to notify their respective
user-SIPs as termination indications.

Thanks,
-Monir
Trillium Digital Systems -an Intel Company.

-----Original Message-----
From: Bratakos, Maroula [mailto:mbratakos@nuera.com]
Sent: Tuesday, October 17, 2000 9:38 AM
To: 'rem-conf@es.net'
Subject: RTCP BYE


What is the role of the RTCP BYE when a control protocol, such as SIP, is
being used to create and terminate sessions?

Thanks in advance.
Maroula Bratakos





From rem-conf Tue Oct 17 14:22:47 2000 
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Date: Tue, 17 Oct 2000 11:36:15 -0700
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From: Florissa Colina <florissa@bmrc.berkeley.edu>
Subject: 10/18  DTV Application Software Environment and Interactive TV
  Platforms - Aninda DasGupta, Philips Consumer Electronics
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Berkeley Multimedia, Graphics and Interfaces Seminar

DTV Application Software Environment and Interactive TV Platforms

October 18, 2000,
1:10-2:30 p.m.
Fujitsu Seminar Room (405 Soda Hall) 

Aninda DasGupta 
Philips Consumer Electronics

Digital television systems are capable of delivering rich media services to
viewers.  With the use of Web and Java content, program producers are
creating exciting interactive television content, and service providers are
launching data broadcast services, over terrestrial, cable and satellite
networks.  In this talk we will briefly take a look at the evolution of
interactive TV solutions.  You will then be introduced to the DTV
Application Software Environment (DASE), which is a proposed middleware
standard that provides platform independence for all applications to run
uniformly over all brands and models of DTV receivers.  If time permits,
some demonstrations will be shown.

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:

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 Tue Oct 17 14:28:51 2000 
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Date: Tue, 17 Oct 2000 15:27:19 -0500
From: Qiaobing Xie <xieqb@cig.mot.com>
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To: Mikael Degermark <micke@CS.Arizona.EDU>
CC: Qiaobing Xie <Qiaobing_Xie-QXIE1@email.mot.com>,
        Mikael Degermark <micke@optima.CS.Arizona.EDU>,
        Krister Svanbro <krister.svanbro@lu.erisoft.se>,
        Magnus Westerlund <magnus.westerlund@era.ericsson.se>, rem-conf@es.net
Subject: Re: Comments on draft-ietf-avt-rtp-amr-00.txt
References: <39D203BA.561299A1@era.ericsson.se>
					 <200009271749.MAA02664@agevole.cig.mot.com>
					 <39D88E7D.DFCD1BC7@era.ericsson.se>
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					 <200010042121.OAA19379@baskerville.CS.Arizona.EDU>
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Mikael,

Sorry for not responding to you sooner. Just got back from my mini
vacation...

See my comments below..

regards,
-Qiaobing

Mikael Degermark wrote:
> 
> Hi Qiaobing,
> 
> I didn't realize that you were mostly talking about the error resilience properties.
> I thought you were mostly arguing for a more CPU friendly packetization format.

Yes, error resilience is one of the key reasons people like AMR. I'd
like to see the RTP payload design really taking advantage of this. The
current avt-amr draft is not very effective on this aspect. For example,
with the compound payload format defined in the current avt-amr draft,
if one would use some partial transport checksum to cover the "robust
sorted" most sensitive bits in the payload, then any bit error in that
part of the payload will cause the whole packet being tossed by the
transport layer. This doesn't sound right to me.

CPU friendly of cause is always desirable to any protocol design. The
bitwise sorting in the current avt-amr draft will be a waste of CPU
unless the authors can show us clearly how that would benefit the
general AMR over IP application.
 
> Seems that the latter issue has been resolved in favor of non-byte aligned if I
> correctly understand the performance measurements put forward by Ericsson people
> concerning the relative costs of encoding/decoding and packetization.
> 
> Seems like a no-brainer to me. Coding/decoding is so much more expensive
> than packetization that optimizing the latter to save on processing is pointless.
> This also makes it meaningless to have a bit distinguishing the two formats
> as have been proposed.

Comparing to an LPC-based voice coder such as AMR, nothing is more
expensive CPU-wise. That's why DSPs were invented :-) One should keep in
mind that the RTP payload frame formatting is merely an adaptation layer
between the application and transport. 
 
> But your argument against the amr-00 proposal not really about packetization
> efficiency, if I understand your latest letter correctly, but instead about
> error resiliency. Isn't that correct?

Yes, as I said, my main concern is error resilience. If certain
complexity is what it takes to make the design error resilient, I will
happily accept that. But I fail to see how the complex bit sorting can
contribute to the error resilience of the design. In the contrary, I
think it is harmful to the error resilience. 
 
> If correct, we should stop arguing about packetization efficiency and instead
> talk about error resiliency.
> 
> (see also a comment below)
> 
> Micke
>



From rem-conf Wed Oct 18 00:08:32 2000 
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From: "Gert Olsson (ECS)" <Gert.Olsson@ex1.ecs.ericsson.se>
To: "'rem-conf@es.net'" <rem-conf@es.net>
Subject: RE: RTCP BYE
Date: Wed, 18 Oct 2000 09:02:22 +0200
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.. and an RTCP BYE is also required when an SSRC identifier collision has
been detected.


__________________________________________________________________|
Gert Olsson
Ericsson Mobile Communications AB, SE-221 83  Lund, Sweden


-----Original Message-----
From: Mozumder, Monir [mailto:m_mozumder@trillium.com]
Sent: den 17 oktober 2000 19:11
To: 'Bratakos, Maroula'; 'rem-conf@es.net'
Subject: RE: RTCP BYE


Hi Maroula,

I think (correct me if wrong) the answer lies in layering principles. SIP
sits in upper layer which uses the service of RTP. When SIP sends
terminating commands, it will direct RTP to send RTCP BYE, which will
ultimately indicate the other SIP side RTP(s) to notify their respective
user-SIPs as termination indications.

Thanks,
-Monir
Trillium Digital Systems -an Intel Company.

-----Original Message-----
From: Bratakos, Maroula [mailto:mbratakos@nuera.com]
Sent: Tuesday, October 17, 2000 9:38 AM
To: 'rem-conf@es.net'
Subject: RTCP BYE


What is the role of the RTCP BYE when a control protocol, such as SIP, is
being used to create and terminate sessions?

Thanks in advance.
Maroula Bratakos





From rem-conf Wed Oct 18 00:23:19 2000 
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
To: "Bratakos, Maroula" <mbratakos@nuera.com>,
        "'rem-conf@es.net'"
	 <rem-conf@es.net>
Subject: RE: RTCP BYE
Date: Wed, 18 Oct 2000 03:17:05 -0400
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RTCP BYE is better in loosely coupled sessions, typically large multicast
ones, where there isn't even a SIP relationship between all the
participants. RTCP BYE scales up very well to groups with thousands or
millions of members, whereas SIP BYE would not. 

SIP BYE is, not surprisingly, better for tightly coupled sessions. It is
reliable (RTCP BYE is not), faster (RTCP intervals are several seconds), can
be authenticated, and can be seen by proxies and other sip devices on the
session establishment path that need to see them.

-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: (973) 952-5000
http://www.dynamicsoft.com
 

> -----Original Message-----
> From: Bratakos, Maroula [mailto:mbratakos@nuera.com]
> Sent: Tuesday, October 17, 2000 12:38 PM
> To: 'rem-conf@es.net'
> Subject: RTCP BYE
> 
> 
> What is the role of the RTCP BYE when a control protocol, 
> such as SIP, is
> being used to create and terminate sessions?
> 
> Thanks in advance.
> Maroula Bratakos
> 



From rem-conf Wed Oct 18 07:23:36 2000 
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Date: Wed, 18 Oct 2000 16:06:07 +0200
From: Franceschini Guido <Guido.Franceschini@CSELT.IT>
Subject: RE: 4onIP AHG meeting on Saturday
To: 'Young-Kwon LIM' <young@techway.co.kr>, 4on2andIP-sys@advent.ee.columbia.edu
Cc: rem-conf@es.net
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I will be there on Sunday, not on Saturday. Thus, I support changing the
meeting date.
Guido

-----Original Message-----
From: Young-Kwon LIM [mailto:young@techway.co.kr]
Sent: Tuesday, October 17, 2000 12:24 PM
To: 4on2andIP-sys@advent.ee.columbia.edu
Cc: rem-conf@es.net
Subject: 4onIP AHG meeting on Saturday


Dear all,

I found out that there are many AHG members relatively active so far have
problem to join Saturday AHG meeting. So, I'm considering to postpone it to
Sunday. Please let me know if there are any people having problem to join
the meeting if we have a meeting only on Sunday instead of Saturday. If I
didn't get enough support from members I can not change anything. Please
show your hands.

Sincerely,
Young.



From rem-conf Wed Oct 18 11:00:01 2000 
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From: mike2112@arabia.com
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Removal instructions below

I saw your listing on the internet.

I work for a company that specializes
in getting clients web sites listed
as close to the top of the major 
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submit your site at least twice a
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To get started and put your web site
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888-532-8842

To be removed call: 888-800-6339 X1377






From rem-conf Wed Oct 18 15:38:32 2000 
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Subject: Hi, this is for you.....
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             **************** CALL FOR PAPERS ***************

            2001 IEEE Pacific Rim Conference on Communications,
                Computers and Signal Processing (PACRIM'01)

                 August 26-28, 2001, Victoria, B.C., Canada
        Sponsored by IEEE Canada (Region 7) & IEEE Victoria Section
                 Co-sponsored by the University of Victoria

Established in 1987, Pacific Rim Conference is the premier IEEE biennial
event in the Pacific Northwest. It is our pleasure to invite you to
participate in this event.

The scope of PACRIM'01 encompasses but is not limited to:
 Mobile and Personal Communications    Computer Architecture

 Coding, Modulation and Spread
 Spectrum                              Parallel and Distributed Computing

 ATM Networks                          Internet Applications

 Multimedia Communications             Neural Networks and Genetic
                                       Algorithms

 Optoelectronic Systems and Devices    Hardware/Software Codesign

 Broadband Satellite Communications    Database and Knowledge-Based Systems

 DSP Applications in
 Telecommunication                     Adaptive Filters and Algorithms

 Image and Video Processing            Wavelets and Filter Banks

 Digital Filters                       VLSI

 Speech and Audio Processing           Nonlinear Signals and Systems

                                  DEADLINES

                    1000 word summary:     March 1, 2001

                    Acceptance/Rejection:  April 1, 2001

                    Camera Ready Copy:     May 1, 2001

 Please submit your summary           General enquires on PACRIM'01
 (by March 1, 2001) to the            can be addressed to either of the
 Technical Program Chair:             General Co-Chairs:
 Dr. Panajotis Agathoklis             Dr. Aaron Gulliver or Dr. Kin Li
 Dept. of Electrical & Computer Eng.  Dept. of Electrical & Computer Eng.
 University of Victoria               University of Victoria
 P.O. Box 3055, STN CSC               P.O. Box 3055, STN CSC
 Victoria, BC, Canada V8W 3P6         Victoria, BC, Canada V8W 3P6
 Fax: +1-250-721-6052                 Fax: +1-250-721-6052
 Email: pan@ece.uvic.ca               Email: agullive@ece.uvic.ca or kinli@ece.uvic.ca

Accepted papers will be published by the IEEE in the Conference Proceedings.
They will be available for worldwide post conference sales from the IEEE.

Further information on the conference may be found on the Website:
                   http://www.citr.ece.uvic.ca/pacrim01/

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

  We apologize if you have received this notice more than once.

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



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--=====================_972372683==_
Content-Type: text/plain; charset="us-ascii"

Dear Colleague,

You are invited to submit a paper to the International "Distributed Dynamic
Multiservice Architectures" (DDMA) Workshop. This workshop is in conjunction
with the 21st International Conference on Distributed Computing Systems
(ICDCS2001), one of the major conferences in the Distributed Computing
Systems Area.

DDMA workshop covers aspects concerning adaptive architectures and service
composition and management.

The submission deadline is November 1st. Further details are at the web
site http://ddma.enst-bretagne.fr/. A text version is also in attachment.

The DDMA workshop will be held in Phoenix, Arizona; April 16-19, 2001.
The proceedings of the workshop will be published by IEEE CS press.

[We apologize if you receive multiple copies of this announcement]
--=====================_972372683==_
Content-Type: text/plain; charset="iso-8859-1"
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					Call for Papers

				International Workshop on
			Distributed Dynamic Multiservice Architectures
				http://ddma.enst-bretagne.fr/

				in conjunction with ICDCS2001
				sponsored by IEEE Computer Society

				April 16 =96 19, 2001
				Phoenix, Arizona, USA

  The growing need to integrate new services while complying with the=
 requirements of end=20
users puts strong emphasis on the management and combination of future=
 services and the QoS=20
they should be able to provide and guarantee. Dynamic Multiservice=
 Architectures constitute a=20
challenging paradigm for distribution theory and practice due to their=
 unique requirements in terms=20
of adaptability, behavioural integration, separation of concerns, service=
 composition and service=20
management. In particular, the deployment of multimedia telecommunication=
 services is a highly=20
challenging area where those paradigms should apply because advanced=
 services require that the=20
user's profiles, preferences and context be taken into account. "Distributed=
 Dynamic=20
Multiservice Architectures" hence add a new dimension to distributed=
 computing to help the=20
composition and deployment of new services guaranteeing final user's=
 requirements.
  The goals of this workshop are to bring together users and researchers to=
 present their=20
recent work related to diverse aspects of dynamic multiservice=
 architectures, their fundamental=20
issues, their paradigms and some appropriate approaches and experiments. The=
 workshop will=20
thus present opportunities for discussing further evolutions and their=
 expected benefits.

Scope of Workshop
-----------------
The scope of this workshop encompasses but is not limited to :

Evolution of distributed multiservice architectures
	Separation of aspects and composition of aspects.
	Methodological approaches for aspect oriented programming.
	Components as cornerstones for dynamic composition.
	Composition requirements for reflection.
	Auto-adaptive ad hoc architectures and protocols.
Impact on the management and control
	Management of adaptive architectures
	Management and monitoring of composed services.
Platforms, tools, languages, ...=20
	Languages, tools and platforms for auto-adaptive systems.
	Environments for specification and design of dynamic architectures.
	Experience with design, implementation and use of auto-adaptive systems.
Application to telecommunication services
	Open critical issues in terms of composition of services.
	Panorama of QoS integration approaches for multimedia telecom services.
	Paradigms and requirements for telecom services architectures.
	Architectures for 3rd generation multimedia services.
	Distribution protocols for telecom (WAP, CORBA, CAMEL, mobile agents).

Submission Instructions and Workshop Format
-------------------------------------------
	Interested participants should submit a 4-8 pages position paper that=
 focuses on one of the=20
relevant topics or related issues. Papers will be reviewed by the program=
 committee. Electronic=20
submissions should be sent to zied.choukair@enst-bretagne.fr. Mail body=
 should include name,=20
address, affiliation and a brief biography along with paper's title and=
 abstract.
	The workshop will follow the following format: The morning and part of the=
 afternoon will=20
be dedicated to presentations and discussions of accepted proposals,=
 including demonstrations of=20
existing software. The afternoon session will end with an identification of=
 issues, followed by=20
discussion. The organisers will then elaborate a summary of the workshop=
 based upon a summary=20
of the papers and the major outlined points of the discussion.

Program Committee Members :
	Mehmet Aksit, Twente University, NL
	Zi=E8d Choukair, ENST Bretagne, FR
	Jaime Delgado, Universitat Pompeu Fabra, ES
	Byung-Guk Kim, University of Massachusetts Lowell, USA
	Marwan Krunz, University of Arizona, USA
	Abdul H. Sadka, University of Surrey, GB
	Antonio Rito Silva, University of Lisbon, PT
	Timothy K. Shih, Tamkang University, TW
	Katsuya Tanaka, Tokyo Denki University, JP

Important Dates :
	Paper submission due : 	Nov 1, 2000
	Notification : 		Dec 30, 2000
	Camera ready : 	 	Jan 20, 2001
	Conference : 		April 16-19, 2001

Web Pages :
	Conference URL : http://cactus.eas.asu.edu/ICDCS2001/

--=====================_972372683==_--




From rem-conf Tue Oct 24 11:49:59 2000 
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Message-ID: <004701c03deb$289cafa0$6dfda8c0@wpi.edu>
From: "Abhijit Pandey" <abhijit@ece.WPI.EDU>
To: "Gert Olsson \(ECS\)" <Gert.Olsson@ex1.ecs.ericsson.se>, <rem-conf@es.net>
References: <A9D7A677B724D411A8B600204840355E187B1D@eseldnt102.ld.sw.ericsson.se>
Subject: Adaptive Buffer info needed
Date: Tue, 24 Oct 2000 14:49:37 -0400
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Could anybody let me know where I could information about adaptive jitter
buffer?
How to implement it and its effects on voice quality.
Thanks,
Abhijit Pandey




From rem-conf Tue Oct 24 12:13:11 2000 
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
To: "'Abhijit Pandey'" <abhijit@ece.WPI.EDU>,
        "Gert Olsson (ECS)"
	 <Gert.Olsson@ex1.ecs.ericsson.se>,
        rem-conf@es.net
Subject: RE: Adaptive Buffer info needed
Date: Tue, 24 Oct 2000 15:07:39 -0400
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Here are pointers to a few papers:


Jonathan Rosenberg, Lili Qiu and Henning Schulzrinne, "Integrating Packet
FEC into Adaptive Voice Playout Buffer Algorithms on the
Internet," in Proceedings of the Conference on Computer Communications (IEEE
Infocom), (Tel Aviv, Israel), Mar. 2000.

     Abstract: Transport of real time voice traffic on the Internet is
difficult for two reasons - network loss and network jitter. There
     has been substantial research in developing protocols and algorithms to
combat these problems. Network loss is handled primarily
     through a variety of different forward error correction (FEC)
algorithms and local repair operations at the receiver. Jitter has been
     compensated for by means of adaptive playout buffer algorithms at the
receiver. Traditionally, these two mechanisms have been
     investigated in isolation. In this paper, we show the interactions
between adaptive playout buffer algorithms and FEC, and
     demonstrate the need for coupling. We propose a number of novel playout
buffer algorithms which provide this coupling, and
     demonstrate their effectiveness through simulations based on both
network models and real network traces.



Sue B. Moon, Jim Kurose and Don Towsley, "Packet audio playout delay
adjustment algorithms: performance bounds and algorithms,"
Department of Computer Science, University of Massachusetts at Amherst,
Amherst, Massachusetts, Aug. 1995.

     Abstract: In packet audio applications, packets are buffered at a
receiving site and their playout delayed in order to compensate
     for variable network delays. In this paper, we consider the problem of
adaptively adjusting the playout delay in order to keep this
     delay as small as possible, while at the same time not incurring too
many ``losses'' due to the arrival of packets at the receiver after
     their playout time has already passed. The contributions of this paper
are twofold. First, given a trace of packet audio receptions
     at a receiver, we present efficient algorithms for computing upper and
lower bounds on the optimum (minimum) average playout
     delay for a given number of packet losses (due to late arrivals) at the
receiver for that trace. These bounds, which we show to be
     tight for the range of loss and delay values of interest, are of
particular importance as they provide a bound on the achievable
     performance of any playout delay adjustment algorithm. Second, we
present a new adaptive delay adjustment algorithm that
     tracks the network delay of recently received packets and efficiently
maintains delay percentile information. This information,
     together with a ``delay spike'' detection algorithm based on (but
extending) our earlier work is used to dynamically adjust talkspurt
     playout delay. We show that this algorithm outperforms existing delay
adjustment algorithms over a number of measured audio
     delay traces and performs close to the theoretical optimum over a range
of parameter values of interest.

     Keywords: packet audio; playout delay; multimedia; packet loss; dynamic
programming; computer networks



Ramachandran Ramjee, Jim Kurose, Don Towsley and Henning Schulzrinne,
"Adaptive playout mechanisms for packetized audio
applications in wide-area networks," in Proceedings of the Conference on
Computer Communications (IEEE Infocom), (Toronto,
Canada), pp. 680-688, Jun. 1994.

     Abstract: Recent interest in supporting packet-audio applications over
wide area networks has been fueled by the availability of
     low-cost, toll-quality workstation audio and the demonstration that
limited amounts of interactive audio can be supported by
     today's Internet. In such applications, received audio packets are
buffered, and their playout delayed, at the destination host in
     order to compensate for the variable network delays. In this paper we
investigate the performance of four different algorithms for
     adaptively adjusting the playout delay of audio packets in an
interactive packet-audio terminal application, in the face of such
     varying network delays. We evaluate the playout algorithms using
experimentally-obtained delay measurements of audio traffic
     between several different Internet sites. Our results indicate that an
adaptive algorithm which explicitly adjusts to the sharp,
     spike-like increases in packet delay which we observed in our traces
can achieve a lower rate of lost packets for both a given
     average playout delay and a given maximum buffer size.

     Keywords: packet voice; packet audio; delay adaptation; internet; IP;
UDP; estimation



---
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: (973) 952-5000
http://www.dynamicsoft.com
 

> -----Original Message-----
> From: Abhijit Pandey [mailto:abhijit@ece.WPI.EDU]
> Sent: Tuesday, October 24, 2000 2:50 PM
> To: Gert Olsson (ECS); rem-conf@es.net
> Subject: Adaptive Buffer info needed
> 
> 
> Could anybody let me know where I could information about 
> adaptive jitter
> buffer?
> How to implement it and its effects on voice quality.
> Thanks,
> Abhijit Pandey
> 
> 



From rem-conf Tue Oct 24 12:13:11 2000 
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Date: Tue, 24 Oct 2000 12:07:21 -0700
To: rem-conf@es.net
From: "Michael A. Dolan" <miked@tbt.com>
Subject: SMPTE
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FYI, I called a couple of ID's to the attention of a relevant working group 
in SMPTE. I think it would be valuable to establish some form of 
communication between this WG and SMPTE as things continue to 
merge.  Specifically, I called their attention to the following drafts:


>         http://search.ietf.org/internet-drafts/draft-gharai-hdtv-video-00.txt
>
> 
>http://search.ietf.org/internet-drafts/draft-ietf-avt-smpte292-video-00.txt
>
>         http://search.ietf.org/internet-drafts/draft-ietf-avt-dv-video-03.txt

FYI,
         Mike


------------------------------------------------------
Michael A. Dolan, Representing DIRECTV,  (619)445-9070
PO Box 1673 Alpine, CA 91903        FAX: (208)545-6564




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Date: Tue, 24 Oct 2000 15:43:43 -0400
From: blue@vt.edu (leslie michelle lipscomb)
Subject: Re: Adaptive Buffer info needed
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I was just testing h.323 with two vendor's systems and experienced video=
 jitter from one, and was informed by the technical rep that some systems=
 include packet buffering as a feature and some do not handle it at all --=
 appearantly it may be a proprietary addition and not standards-based.

------

>Could anybody let me know where I could information about adaptive jitter
>buffer?
>How to implement it and its effects on voice quality.
>Thanks,
>Abhijit Pandey

-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-
Interoperability Lab - ISB208
Communications Network Services
Andrews Information Systems Building
1700 Pratt Drive, CRC, Blacksburg, VA 24061-0506
phone : 540/231/2361  email : interop-lab@vt.edu
-=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 Oct 24 13:06:28 2000 
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From: "Dr. Carsten Bormann" <cabo@tzi.org>
To: "leslie michelle lipscomb" <blue@vt.edu>,
   "Abhijit Pandey" <abhijit@ece.WPI.EDU>
Cc: <rem-conf@es.net>
Subject: RE: Adaptive Buffer info needed
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> I was just testing h.323 with two vendor's systems and
> experienced video jitter from one, and was informed by the
> technical rep that some systems include packet buffering as a
> feature and some do not handle it at all -- appearantly it may be
> a proprietary addition and not standards-based.

Of course, whether you play the video stream correctly, distort or discard
it, or replace it by images of hopping monkeys is a "quality of
implementation issue".
On the other hand, RFC1889 would certainly not go to the length of creating
all that mechanism if there wasn't an expectation that a receiver is going
to use the information...
You need to tell that technical rep that playing the video correctly is
certainly more standards-based and less proprietary than showing hopping
monkeys.

Oh, and name that vendor, please.

I would like to take this opportunity to open a round of vendor bashing on
negligent RTP implementations.

Apart from simply not implementing RTP playout (see above), many
implementations seem to skip sending RTCP or implement it in non-conforming
ways.
This hurts interoperability surprisingly little because very few
implementers care about what they receive in RTCP.
I know that some streaming applications at least use the RTCP receiver
reports to find out whether the recipient is still there and listening.
Any other positive news on this front?

While vendor bashing may not be on the charter of this list, it does help
creating interoperability if you know how a standard is being used.

Gruesse, Carsten




From rem-conf Tue Oct 24 13:24:50 2000 
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To: "Dr. Carsten Bormann" <cabo@tzi.org>
cc: leslie michelle lipscomb <blue@vt.edu>,
   Abhijit Pandey <abhijit@ece.WPI.EDU>, rem-conf <rem-conf@es.net>
Subject: Re: Adaptive Buffer info needed 
In-Reply-To: Your message of "Tue, 24 Oct 2000 22:04:30 +0200."
             <NDBBLDHFKCPEPDKNKJBKOECKEPAA.cabo@tzi.org> 
Date: Tue, 24 Oct 2000 16:23:38 -0400
From: Colin Perkins <csp@east.isi.edu>
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--> "Dr. Carsten Bormann" writes:
>> I was just testing h.323 with two vendor's systems and
>> experienced video jitter from one, and was informed by the
>> technical rep that some systems include packet buffering as a
>> feature and some do not handle it at all -- appearantly it may be
>> a proprietary addition and not standards-based.
>
>Of course, whether you play the video stream correctly, distort or discard
>it, or replace it by images of hopping monkeys is a "quality of
>implementation issue".
>On the other hand, RFC1889 would certainly not go to the length of creating
>all that mechanism if there wasn't an expectation that a receiver is going
>to use the information...
>You need to tell that technical rep that playing the video correctly is
>certainly more standards-based and less proprietary than showing hopping
>monkeys.
>
>Oh, and name that vendor, please.
>
>I would like to take this opportunity to open a round of vendor bashing on
>negligent RTP implementations.
>
>Apart from simply not implementing RTP playout (see above), many
>implementations seem to skip sending RTCP or implement it in non-conforming
>ways. This hurts interoperability surprisingly little because very few
>implementers care about what they receive in RTCP.  I know that some
>streaming applications at least use the RTCP receiver reports to find out
>whether the recipient is still there and listening.  Any other positive
>news on this front?

They certainly need to implement RTCP if they want lip-sync, and a number
of products do that. 

There are also implementations which refuse to play-out the media until
they receive a correct RTCP packet from a source. As the author of one of
those, I find it sad that we get complaints after every SIP bakeoff, due to
lack of interoperability with the implementations which just don't implement 
RTCP.

And, of course, implementations which don't care about sending RTP on even
numbered UDP ports are kind-of annoying...

>While vendor bashing may not be on the charter of this list, it does help
>creating interoperability if you know how a standard is being used.

Cheers,
Colin



From rem-conf Tue Oct 24 13:31:44 2000 
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Message-ID: <4FBEA8857476D311A03300204840E1CF01A6E8B2@whq-msgusr-02.pit.comms.marconi.com>
From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Colin Perkins'" <csp@east.isi.edu>,
        "Dr. Carsten Bormann"
	 <cabo@tzi.org>
Cc: leslie michelle lipscomb <blue@vt.edu>,
        Abhijit Pandey
	 <abhijit@ece.wpi.edu>, rem-conf <rem-conf@es.net>
Subject: RE: Adaptive Buffer info needed 
Date: Tue, 24 Oct 2000 16:30:00 -0400
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Care to name one or more implementations you find exemplary in this
regard to use as an interwork reference?

Brian

> -----Original Message-----
> From: Colin Perkins [mailto:csp@east.isi.edu]
> Sent: Tuesday, October 24, 2000 4:24 PM
> To: Dr. Carsten Bormann
> Cc: leslie michelle lipscomb; Abhijit Pandey; rem-conf
> Subject: Re: Adaptive Buffer info needed 
> 
> 
> --> "Dr. Carsten Bormann" writes:
> >> I was just testing h.323 with two vendor's systems and
> >> experienced video jitter from one, and was informed by the
> >> technical rep that some systems include packet buffering as a
> >> feature and some do not handle it at all -- appearantly it may be
> >> a proprietary addition and not standards-based.
> >
> >Of course, whether you play the video stream correctly, 
> distort or discard
> >it, or replace it by images of hopping monkeys is a "quality of
> >implementation issue".
> >On the other hand, RFC1889 would certainly not go to the 
> length of creating
> >all that mechanism if there wasn't an expectation that a 
> receiver is going
> >to use the information...
> >You need to tell that technical rep that playing the video 
> correctly is
> >certainly more standards-based and less proprietary than 
> showing hopping
> >monkeys.
> >
> >Oh, and name that vendor, please.
> >
> >I would like to take this opportunity to open a round of 
> vendor bashing on
> >negligent RTP implementations.
> >
> >Apart from simply not implementing RTP playout (see above), many
> >implementations seem to skip sending RTCP or implement it in 
> non-conforming
> >ways. This hurts interoperability surprisingly little 
> because very few
> >implementers care about what they receive in RTCP.  I know that some
> >streaming applications at least use the RTCP receiver 
> reports to find out
> >whether the recipient is still there and listening.  Any 
> other positive
> >news on this front?
> 
> They certainly need to implement RTCP if they want lip-sync, 
> and a number
> of products do that. 
> 
> There are also implementations which refuse to play-out the 
> media until
> they receive a correct RTCP packet from a source. As the 
> author of one of
> those, I find it sad that we get complaints after every SIP 
> bakeoff, due to
> lack of interoperability with the implementations which just 
> don't implement 
> RTCP.
> 
> And, of course, implementations which don't care about 
> sending RTP on even
> numbered UDP ports are kind-of annoying...
> 
> >While vendor bashing may not be on the charter of this list, 
> it does help
> >creating interoperability if you know how a standard is being used.
> 
> Cheers,
> Colin
> 



From rem-conf Tue Oct 24 14:01:53 2000 
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To: "Rosen, Brian" <Brian.Rosen@marconi.com>
cc: "Dr. Carsten Bormann" <cabo@tzi.org>,
   leslie michelle lipscomb <blue@vt.edu>,
   Abhijit Pandey <abhijit@ece.wpi.edu>, rem-conf <rem-conf@es.net>
Subject: Re: Adaptive Buffer info needed 
In-Reply-To: Your message of "Tue, 24 Oct 2000 16:30:00 EDT."
             <4FBEA8857476D311A03300204840E1CF01A6E8B2@whq-msgusr-02.pit.comms.marconi.com> 
Date: Tue, 24 Oct 2000 17:00:37 -0400
From: Colin Perkins <csp@east.isi.edu>
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--> "Rosen, Brian" writes:
>Care to name one or more implementations you find exemplary in this
>regard to use as an interwork reference?

The mbone tools (rat, vat, vic) include pretty reasonable open-source
implementations of RTP.  I'm not sure it's appropriate for me to name
commercial implementations I like/dislike, but if someone who isn't a
working group chair wishes to do so....... 

I'd also like to draw attention to draft-ietf-avt-rtptest-03.txt and
solicit comments on that draft (which we hope to make an informational
RFC eventually).


Colin
[with the disclaimer that I wrote the RTP stack in rat]



From rem-conf Tue Oct 24 16:00:54 2000 
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Date: Tue, 24 Oct 2000 16:01:27 -0700
To: "Rosen, Brian" <Brian.Rosen@marconi.com>
From: Kevin Marks <kmarks@apple.com>
Subject: RE: Adaptive Buffer info needed
Cc: rem-conf <rem-conf@es.net>
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At 4:30 PM -0400 10/24/00, Rosen, Brian wrote:
>Care to name one or more implementations you find exemplary in this
>regard to use as an interwork reference?

If you are doing interoperability tests on RTP implementations, do 
take a look at the current Public Previews of QuickTime 5 Client and 
QuickTime Streaming Server 3.0

http://www.apple.com/quicktime/preview/



From rem-conf Tue Oct 24 16:32:36 2000 
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From: =?iso-8859-1?Q?=22Colleen_Moffitt_=28ne=E9_Gehrt=29=22?=
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Subject: RE: Adaptive Buffer info needed
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References: <  <4FBEA8857476D311A03300204840E1CF01A6E8B2@whq-msgusr-02.pit.comms.marconi .com>
 <4FBEA8857476D311A03300204840E1CF01A6E8B2@whq-msgusr-02.pit.comms.marconi .com>
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For your RTP interop tests the latest preview version of the RealServer can
be found at:

http://www.realnetworks.com/products/servers/server8.html?src=noref,rnhmpg_1
02300,rnhmsi,srvrmn_081800


At 04:01 PM 10/24/00 -0700, Kevin Marks wrote:
>At 4:30 PM -0400 10/24/00, Rosen, Brian wrote:
>>Care to name one or more implementations you find exemplary in this
>>regard to use as an interwork reference?
>
>If you are doing interoperability tests on RTP implementations, do 
>take a look at the current Public Previews of QuickTime 5 Client and 
>QuickTime Streaming Server 3.0
>
>http://www.apple.com/quicktime/preview/
>
>



From rem-conf Tue Oct 24 22:29:42 2000 
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From rem-conf Wed Oct 25 06:12:26 2000 
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Date: Wed, 25 Oct 2000 15:00:23 +0100
From: "Zacharias Bilalis" <Zacharias.Bilalis@curia.eu.int>
To: <rem-conf@es.net>
Subject: Could you please inicate me how I can become a member of your
	list?
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Could you please inicate me how I can become a member of your list?

Thank you in advance

Cordialement / Best Regards
Zacharias Bilalis
Division Informatique
Cour de Justice des CE
Tel: +352 4303 3106
Fax: +352 4303 3200




From rem-conf Wed Oct 25 12:18:25 2000 
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Message-ID: <39F73055.31737D6F@21rst-century.com>
Date: Wed, 25 Oct 2000 15:11:16 -0400
From: Marshall Eubanks <tme@21rst-century.com>
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Subject: Multicast RTP in IPv6
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Hello;

I hesitate to send to two lists at once, but both seem relevant. My apologies for
duplicates.

Does anyone know what the status of RTP/RTCP is in IPv6? I know that
v6 is "born multicast enabled", but what is going to be required to
actually use it with streaming audio / video ? This is going to be a major
application with 3G wireless devices using IPv6, and we would like to start work on v6
compliant software.


                                 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 Oct 25 13:03:03 2000 
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To: tme@21rst-century.com
cc: rem-conf@es.net, ipng@sunroof.eng.sun.com
Subject: Re: Multicast RTP in IPv6 
In-Reply-To: Your message of "Wed, 25 Oct 2000 15:11:16 EDT."
             <39F73055.31737D6F@21rst-century.com> 
Date: Wed, 25 Oct 2000 15:58:41 -0400
From: Colin Perkins <csp@east.isi.edu>
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--> Marshall Eubanks writes:
>Does anyone know what the status of RTP/RTCP is in IPv6? I know that
>v6 is "born multicast enabled", but what is going to be required to
>actually use it with streaming audio / video ? This is going to be a major
>application with 3G wireless devices using IPv6, and we would like to start work on v6
>compliant software.

No changes are required to RTP/RTCP to support IPv6, and there are a number
of applications which support it already.

See http://www-mice.cs.ucl.ac.uk/multimedia/software/rat/ for one open
source implementation, there are others.

Colin



From rem-conf Thu Oct 26 02:00:42 2000 
From rem-conf-request@es.net Thu Oct 26 02:00:41 2000
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	id 13oiq5-0002Ej-00; Thu, 26 Oct 2000 01:56:45 -0700
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Date: Thu, 26 Oct 2000 03:56:58 -0500
Message-Id: <200010260856.DAA16775@server1.faithfunds.com>
To: rem-conf@es.net
From: jordan@controlmultimedia.com
Subject: FAITHFUNDS....A Place to Give, A Place to Share, A Place to Call Home 
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FAITHFUNDS....A Place to Give, A Place to Share, A Place to Call Home
 
Would you like to know how to reach over 765 millions viewersFaithfunds is designed to assist Christians in locating products and services in their community and nationwide. We also provide a Shop-to-Donate section so that local Chapters can benefit from having their Congregation shop online. 

How does this help you?
You will find that Faithfunds is the premier place to get up-to-date news, weather and sports, shop for Christian products, browse through Church directories nationwide and send Children's E-Cards during those special occasions. 

Faithfunds is the premier place to sell your Christian products on the web. Your site can be listed in this directory for premium placement so that you can sell your books, jewelry and accessories. Our demographic is geared toward people just like you -- interested in Christian products! And now we've merged with InfoSpace so that our Reach is even greater! 

Faithfunds REACH

Faithfunds Directory powered by InfoSpace and its network of over 3,100 affiliates guarantees the widest reach.  Ranked 20th overall among sites on the Web and #1 among directory services, receiving more than 765 million impressions per month with more than 29 million unique visits. 

ADVERTISING OPTIONS 

Faithfunds offers advertisers and vendors an ideal medium to cost-effectively target and reach their audience. Whether your primary goal is to drive traffic to your Web site, heighten brand awareness, or generate e-commerce transactions, we will help you reach your goals. 
  

YOUR OPTIONS INCLUDE

You can start your national advertising campaign and reach over 765 million viewers for as little as $1,200 per year!!! 


Christian Directory (MONTHLY RATES)

I. SILVER PACKAGE:
Features: Bold, Enlarged Font, URL link & Email Link 

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a. City: $5 
b. Multi-City: $15 
c. State: $25 
d. Multi-State: $60 
e. National: $100 

II. GOLD PACKAGE 
Features: Bold, Enlarged Font, URL link & Email Link PLUS Ad Link (any graphic) or Page Express 

Listings: 
a. City: $15 
b. Multi-City: $25 
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d. Multi-State: $250 
e. National: $500 

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c. State: $250 
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Please note that these are introductory prices for the Faithfunds launch. Please call us now before the Holiday Season begins! 

May God Bless You, 

The Faithfunds Family 
http://www.faithfunds.com





From rem-conf Thu Oct 26 09:37:26 2000 
From rem-conf-request@es.net Thu Oct 26 09:37:26 2000
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Message-ID: <39F8678C.E4F749F0@thomcast.thomson-csf.com>
Date: Thu, 26 Oct 2000 18:19:08 +0100
From: "Marc Le Naour" <marc.lenaour@thomcast.thomson-csf.com>
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To: rem-conf <rem-conf@es.net>
CC: Nathalie Cabel <Nathalie.Cabel@Thomcast.Thomson-csf.com>
Subject: UNICODE RTSP syntax and binary data
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Dear all,

In RFC 2326 it's defined the RTSP messages are a text-based and RTSP
protocol uses the ISO 10646 character set in UTF-8 encoding.

Parsing a RTSP message including multiple packets of binary data (binary
content, data url, ...)  is quite difficult : the parser has to track
characters delimitaions or string delimitation for each packets before
the next character packets ....

This implies an indirect access to the informations in the RTSP messages
...

To avoid this problem it would be interesting to enforced a pair length
for each binary packet included in a RTSP message with a parity byte
....

This solution would allow quick and easy jumping over non-interesting
header field ...

Marc




From rem-conf Thu Oct 26 12:07:43 2000 
From rem-conf-request@es.net Thu Oct 26 12:07:43 2000
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To: Targeted-Search-Engine@25centspervisitor.com
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Date: Thu, 26 Oct 2000 11:15:21 -0700
From: searchenginemechanic@excite.com
Message-Id: <psbgoeinglcvbciyn.fofafemik@Pbedboy.com>
Subject: FWD: info on Targeted Search Engine Traffic for $0.25 per visitor!!!
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Guaranteed Traffic from major Search Engines 

When using a search engine, if you don ’t find what you ’re looking for 
in the first 10 to 30 matches what do you do? Well, most people go to 
the next search engine and try again. If your site is not coming up in 
the first 10 to 30 matches of a search, then you ’re losing business! 

IT ’S THAT SIMPLE! 

Improving search engine traffic means: 

MORE HITS 
MORE BUSINESS 
MORE SUCCESS 

The most important advertising dollar you spend should be for 
search engine traffic. 

WHY?

The top three reasons are: 

1. Approximately 95% of all Internet users start with a search 
engine query. 

2. Anyone who comes to your site from a major search engine is 
100 times more likely to become a customer because they were 
specifically looking for your product, goods, or services. 

3. Search engine traffic gives you substantially more for your 
advertising dollar than banners. 


We guarantee traffic generated from the.following.search	engines:
Yahoo, HotBot, Excite, Alta Vista, Lycos, Infoseek, MSN, Netscape, AOL Search,
InfoSeek/Go Network,Direct Hit,Fast/Alltheweb,Goto,Iwon and NBCi.

Your initial payment covers setup expenses including: 
domain name registration, web page construction, 
Dir. reviews, DNS setup, etc.  Plus, 
you'll get 2,000 unique visits to your web site.

You'll have 24 hour access to a our web-based interface allowing 
you to see all your traffic, what search engine the visitor used to 
reach you, and even the terms they used to search for you. 

The cost to start getting targeted traffic to YOUR web site is:
$500.00 this equals out to be 2,000 target visitors to 
YOUR web site, A mere twenty five cents ($0.25) 
Per unique targeted visitor.  The cost per visitor
will never go up. After the 2,000 visitors you simply 
pre-pay for more target visitors.

We Look forward to working with you!

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

To get the ball rolling fax the following request to 1.208-445-2319
This Form MUST be completely filled out!.
Note... Do Not provide any payment information at this time!

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

Company:

Address: 

State:                         City:                            Zip:

Contact Name: 

Contact Telephone: 

Contact Fax: 

Contact E-mail Address: 

Website Address: 

A few of your top Keywords:



__ Yes, I want to start this service asap, please contact me today!

__ Yes, I want to start this service please contact me __Mon__Tue__Wed__Thur__Fri 

between these hours__________________________________________________!


Questions:




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

Remove:	searchenginemechanic@excite.com

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




From rem-conf Fri Oct 27 07:58:56 2000 
From rem-conf-request@es.net Fri Oct 27 07:58:55 2000
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Message-ID: <00001587132f$0000428f$00001ff7@localhost>
To: <undiclsed5@erols.com>
From: lexx@cairomail.com
Subject: tbeltz@erols.com
Date: Wed, 25 Oct 2000 21:11:44 -0400
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" style=3D"BACKGROUND-COLOR: #ffffff" SIZE=3D2 PTSIZE=3D9 FAMILY=3D"SANSSE=
RIF" FACE=3D"Arial" LANG=3D"0">out performs, and the price beats TV advert=
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  </FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
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=3D"0">Call 954-974-9903</FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" s=
tyle=3D"BACKGROUND-COLOR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERI=
F" FACE=3D"Arial" LANG=3D"0"><BR>
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OUND-COLOR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Ari=
al" LANG=3D"0">LibidoMagic</FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff"=
 style=3D"BACKGROUND-COLOR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSE=
RIF" FACE=3D"Arial" LANG=3D"0">!</FONT><FONT  COLOR=3D"#000000" BACK=3D"#f=
fffff" style=3D"BACKGROUND-COLOR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"=
SANSSERIF" FACE=3D"Arial" LANG=3D"0">   <BR>
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  VNR<BR>
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  <BR>
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OR: #ffffff" SIZE=3D1 PTSIZE=3D8 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=3D=
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  For Fax Orders, please print and fax the following to:<BR>
  </FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
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From rem-conf Sun Oct 29 08:55:30 2000 
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WE THINK SO !!

Dnaprint (DNAP) is an emerging biotech company and is trading under $1 !!!

1) CEO (Dr. Tony Frudakis) Lead scientist for Corixa starts his own company a 
   few months ago
    
2) Company is formed by a reverse merger of a pinksheet company

3) DNAP has already completed the accounting to get off the pinksheets and 
   should be listed on the OTCBB soon.

4) Dr. Frudakis was  responsible for a patent that sold for $54,000,000 to 
   Smith Kline Beecham while he was at Corixa.
 
5) Corixa's stock has gone from 14 Cents a few years ago to 50 Dollars today.
 
6) According to the Barron's article (9/25/00 pg 52), Dnaprint is in the late stages of forming 
   strategic alliances with major companies.


MAJOR WEB PAGES RELATED TO DNAP 
dnap 
http://www.dnaprint.com/ 
orchid biosciences inc 
http://www.orchidbio.com/ 

INVEST IN DNAP 
email for free investment package= hresources@dnaprint.com 
or call 1-877-362-3453(toll free) 
investment club web page=http://clubs.yahoo.com/clubs/dnaprintgenomicsinvestors 
investors report= 
http://www.companyreporter.com/index.php?u=companies.php 
great dnap investor links: 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=22034 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=23118 
http://cbs.marketwatch.com/news/current/molinski.htx?source=htx/http2_mw =invest in biotech 
http://www.loonycat.com/Genomics.htm =the human genome investors portal 
clearstation rates danp 
http://www.clearstation.com/cgi-bin/details?Event=peek&Symbol=DNAP&Refer=http://www.purepennies.com/ 
purepenny&3bagger rate dnap 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=PCBM&read=132938 
Pennystockprofits assess dnap 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=15730 
tipreporter rates dnap http://www.tipreporter.com/search.php?search=dnap 
dnap on stocktribes hotlist for 21 weeks 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=24335 
analysts tips and reports on dnap 
http://www.tipreporter.com/search.php?search=dnap 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=15730 

why is dnap unique in the bio tech sector?? 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=22654 

NEWSLINKS GENERAL FOR DNAP 
dnap company news 
http://www.bell2bell.newsalert.com/bin/digest?Symbol=DNAP& =bell2bell news 
http://quicken.excite.com/investments/quotes/?s=DNAP&t=0 =excite news 
http://biz.yahoo.com/n/D/DNAP.html =yahoo news 
dnap in barrons http://www.sockthestocks.com/profiles/dnapbarrons.htm 
bioinform news service 
http://www.bioinform.com/ 
online journal of bioinformatics 
http://www.cpb.uokhsc.edu/ojvr/xmlpaper.html#Sites 

UPCOMING EVENTS RELATED TO DNAP 
bio partnering Europe- London England-16-17 oct 2000 
http://www.biopartnering.com/bpe/index.html 
structural genomics conference-yokohama,japan,2-5nov2000 
http://icsg2000.riken.go.jp/ 

TECHNICAL SITES RELATED TO DNAP 
dnap patents 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=21559 
dnap is about pharmacogentics 
http://www.nigms.nih.gov/funding/medforyou.html 
stem cell research 
http://www.stemcellresearchnews.com/index.html 
personalized medicine no longer science fiction 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=25033 
safe drugs that fit like a glove 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=11203 

NEW STAFF ARRIVING AT DNAP SARASOTA 
dr. ramin mirashemi, a new dnap addition—check this track record!!!!!!!!!!!! 
http://www.google.com/search?q=Ramin+Mirhashemi&hl=en&lr=&safe=off&start=0&sa=N 
dr.jose arena joins dnap scientific advisory board 03 oct 2000 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=23532 
scientific advisory board makeup 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=24575 

DNAP COMPARED TO ITS TECHNICAL EQUALS, AND RIVALS: 
genomica(gnom)nasdq-ipo29sept00($19)now$22 
http://www.bell2bell.newsalert.com/bin/story?StoryId=CoDqtWbWbtKvgmda1&FQ=genomica&HdlFmt=simple&Title=Headlines%20for%3A%20genomica 
informax(inunx)nasdq-ipo-03oct00($16)now$23 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=23532 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=23552 
incite genomics(incy)nasdq $37 
http://quicken.excite.com/investments/news/?symbol=INCY 
genentech(dna)($172)-2for1split 5 oct00-263million share float---dnap float = 269 million 
http://www.bell2bell.newsalert.com/bin/digest?Symbol=DNA& 

DNAP IN EUROPE 
http://www.8ung.at/zockstocks 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=21485 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=24168 

WHERE IS DNAP LOCATED—HOW DO YOU GET THERE? 
great air fare rates round trip to sarasota 
http://leisure.travelocity.com/RealDeals/Details/0,2941,TRAVELOCITY_AIR_449,00.html 
dnaps new upscale digs in swinging sarasota, fla 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=DNAP&read=21059 
motel shopper link for sunny sarasota area 
http://www.roomsaver.com/roomsaver/city/index.html?state=FL&city=Sarasota 
location map of coconut ave digs 
http://maps.yahoo.com/py/maps.py?BFCat=&Pyt=Tmap&newFL=Use+Address+Below&addr=900+COCONUT+AVE&csz=SARASOTA%2CFLORIDA%2C+34234&country=us&Get%A0Map=Get+Map 
avis rent a car sarasota 
http://www.avis.com/cgi-bin/maps_and_directions/driving_maps/us_locations/mqinterconnect 

INTELLIGENT ADVICE OR BIASED ROT?KNOW THE DIFFERENCE OR HOW TO USE THESE MESSAGE BOARDS 
email suspected lying-sham efforts,etc to your federal govt(it,s called the sec) at enforcement@sec.gov 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=PCBM&read=132869 
get the mob out of microcaps 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=PCBM&read=132789 
crush naked shorting –stop being screwed without even knowing it—read me 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=PCBM&read=132581 
suspicious sounding codswallop?? ask them to take the universal basher quiz!!!! 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=PCBM&read=132816 
dnap has unusually good promise—accounting for the sad fact that were in “basher heaven”-read me 
http://www.ragingbull.altavista.com/mboard/boards.cgi?board=PCBM&read=132818 
sec offers tips on how to avoid scams on the internet—good article 
http://www.sec.gov/consumer/cyberfr.htm 


CONTENTS 
1.BACKGROUND 
2.FOUNDERS 
3.PRODUCTS&SERVICES 

1.BACKGROUND 

DNAPrint genomics is a young e-biotech company based in southwest Florida. We are developing an informatics platform system that will provide dynamic solutions for disease gene discovery, genetic predisposition and genetic analysis testing. Our work has real-life application to the germinating field of Personalized Medicine and will help lay the foundation for a brand-new area of medical research called Phenomics. 
The PhenomeSM platform system that we are developing will help identify an individual predisposed to develop cancer before the onset of illness so that lifestyles can be altered and/or preventative measures taken. It will be used to identify individuals who are incompatible with certain drug treatments before the drugs are prescribed and damage is done. It will be used to tease out important genetic determinants associated with complex genetic diseases, so that drugs can be developed to target these genes. 
Our aim is ambitious. By partnering our platform with biotech/pharma, we will participate in the downstream profits generated from pharmaceutical products our platform has enabled. By marketing our platform directly to the public, we intend to make comprehensive genotype screening solutions accessible to people all over the world via the internet. Our system will enable a more holistic approach to using genomics information to improve peoples lives. 
The patient of the future will have more information and power than ever imagined. The human genome project, the internet and recent technological advancements all contribute to make this possible, and we intend to take advantage of these synergies to become the primary conduit for the generation and distribution of personalized genetic information. 

The Market 
Pharmacogenomics is currently a 1 billion dollar industry and is projected to be a 10 billion dollar industry by the end of 2003. Several billions of dollars have been spent on the human genome project so far, and an equal amount will likely be spent trying to understand what all of the genomics information means (i.e. bioinformatics, proteomics and phenomics). Genomics is a multi billion dollar industry world-wide and the market for personalized medicine is expected to exceed $25 billion dollars by 2010. Recently congress has approved a $2.3 billion increase in NIH funding to a total of $18 billion dollars for the year 2000. 

2.FOUNDERS 
Tony Frudakis, Ph.D.(tfrudakis@dnaprint.com) 
Chief Executive Officer, President and Laboratory Director. Dr. Tony Frudakis is a Phi Beta Kappa, Magna Cumme Laude graduate of the University of California system. He received his doctorate degree in molecular and cell biology from the University of California Berkeley and has 14 total years of experience as a molecular biologist. Dr. Frudakis' technical experience encompass a range of areas within biology, including the molecular biology of eukaryotic transcriptional regulation, the creation and study of transgenic animals, the development of novel nucleic acid isolation techniques and gene discovery protocols, bioinformatics processing and anatomical gene expression profiling. At Corixa Corporation in Seattle, WA he developed several new techniques for RNA fingerprinting, managed and executed high-throughput gene discovery programs for various cancers and was instrumental in the companies early success in attracting R and D partners. Founding GAFF biologic in 1998, Dr. Frudakis established a good reputation for high quality DNA sequencing and library construction/screening services, until its evolution into the DNAPrint Genomics that exists today. In all, his work has resulted in a patent portfolio for over 350 unique genes and 2 products 

George Frudakis(gfrudakis@dnaprint.com) 
Vice President of Business Affairs. George Frudakis has vast experience with start-up company development. He started and developed a successful multi-component company called GAFF group, which today enjoys over 15 million dollars annually in revenue. During his life’s work as an entrepreneur and businessman, George has learned how to run and successfully manage the financial and operational aspects of both small and large companies alike. He is a graduate from the California State University at Long Beach in California. 

Myung Ho Kim, Ph.D. Mathematics(mkim@dnaprint.com) 
Director of Informatics and Lead Mathemetician. Myung recieved his Ph.D. in Mathematics from the University of Michigan in Ann Arbor in 1988 and M.S. in Mathematics from The Ohio State University in Columbus Ohio. Most recently, Myung worked on the Ant World Server project supported by DARPA (Defense Advanced Research Progect Agency) and implemented Kolmogorov-Smirnov tests, linear/nonlinear regression, nonparametric analysis and modeling to a TREC 6 (Text Retrieval Conference) project using SAS, C/C++ and Perl languages. Prior to this, Myung simulated mathematical modeling for breast tumors and implemented/developed various image processing algorithms. He has served as an Honorary research Fellow in the Department of Mathematics and Information Science at the University of Auckland, New Zealand and a Visiting Research Scholor in the Department of Mathematics at the University of California at Santa Cruz in California. Prior to coming to the United States, Myung held the position of Assistant Professor in the Department of Mathematics at Sungkyunkwan University in Soel Korea. 

Venkateswarlu Kondragunta, Ph.D. (vkondragunta@dnaprint.com) 
Director of Biostatistics. Dr. Kondragunta obtained his Ph.D. in Statistics from the University of Madras, Madras, India. He most recently served as a Postdoctoral Fellow at the Schoolof Public Health, University of Texas, Houston, TX under the tutelage of Dr. Ranajit Chakraborty, one of the most famous and accomplished genetic mathematicians in the world. His research interests include variance components, linear regression models, general linear models, design and analysis, discriminant analysis, principal component analysis and statistical genetics. He has taught courses in Variance Components, Design and Analysis of Experiments and Linear Regression Models and Probability Theory at Concordia University in Montreal, Canada. Dr. Kondragunta is an accomplished programmer and has amassed considerable experience with implementing statistical genetic ideas into C++, S-plus, FORTRAN (7 years experience) and JAVA code. His most recent work used the REG, ANOVA, GLM, VARCOMP, CANDISC, CLUSTER, DISCRIM, FACTOR, PRINCOMP and LOGISTIC tools in the SAS computing environment on real and simulated data sets. 
A partial list of publications for Dr. Kondragunta include: 
Chaubey, Y.P. and K.Venkateswarlu. A numerical study of some robust estimators of variance components in one-way model. International Indian Statistical Association Conference proceedings, October 10-11, 1998, McMaster University, Hamilton, Canada. 
Venkateswarlu, K. K.N.Ponnuswamy, and M.R.Srinivasan. Estimation of variance components based on diallel model. Mathematical Biosciences, V-150, 105-112. 
Venkateswarlu, K, and K.N.Ponnuswamy. Estimation of variance components based on bio-model of diallel crosses (balanced). Communications in Statistics (Theory & Methods), Vol-27, 139-152. 
Venkateswarlu, K. Estimation of variance components based on diallel model involving maternal and maternal interaction effects. Biometrical Journal, Vol-39, 287-295. 

3.PRODUCTS&SERVICES 
see dnap web page for a complete description of products & services http://www.dnaprint.com/ 
SNiPdocTM 
High-throughput SNP profiling-To help us with our bottom line, we provide contract SNP profiling services to the drug development community. We call this service our SNiPdocTM service, and it is targeted towards pharmaceutical companies. Many pharmaceutical companies are seeking to enter into the field of Pharmacogenomics. This new field of study could have a dramatic impact on overall health care costs in the future. Furthermore, by having access to pharmacogenomics data, pharmaceutical companies can focus their clinical trials on segments of the population that are genetically receptive to the treatment. Experimental drugs can be targeted to individuals of a particular genotype, and genetic variations shown to underlie a poor-response to a trial drug could be used as a basis for eliminating the non-responding segment of the trial population from the study. In this way, pharmaceutical companies can reduce drug trial failures and the costs associated with them. Contracting this work out to service providers is a viable option for many smaller pharmaceutical companies who do not have the volume of work necessary to justify the price of the equipment and expertise. At DNAPrint Genomics, we offer multiple patient – multiple SNP sampling matricies designed on a custom, semi-custom or predefined basis. We can use our proprietary PhenomixM databases of medically relevant SNPs (numbering in the thousands of SNPs at present), or design chips for your application de-novo. We will work with your scientists to develop the experiments from beginning to end, that suit your needs, and we can handle your project from blood/tissue or DNA starting points. Data handling will be customized for each client; data can be deposited in our secure SNP relational databases system offering you the option of accessing your data over the internet. Alternatively, we can burn and hand-delivered CDs to 



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From rem-conf Sun Oct 29 12:40:20 2000 
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From rem-conf Sun Oct 29 17:47:28 2000 
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From: Colin Perkins <csp@isi.edu>
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The preliminary agenda for the 49th IETF has AVT scheduled for two slots:
Wednesday 15:30-17:30 and Thursday 13:00-15:00 (these are subject to change,
of course).

Anyone wanting agenda time in the AVT meeting should contact me and Steve,
and we'll put a draft together.

Cheers,
Colin



From rem-conf Mon Oct 30 12:22:20 2000 
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Date: Mon, 30 Oct 2000 11:41:05 -0800
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From: "Michael A. Dolan" <miked@tbt.com>
Subject: RTP content type questions/comments
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I am new to this WG, but understand MPEG and am active in SMPTE and ATSC 
standards efforts. So in that context, I have some questions on the IETF 
AVT WG ID (draft-ietf-avt-profile-09) related to the supported RTP content 
types and the meaning of the names:

1. Regarding "video/mp2t": The document says "either audio or video". Since 
a transport stream is usually both, what does this mean exactly?  Do you 
mean that these are really elementary streams just encapsulated in 188-byte 
transport packets, or do you mean a real transport stream constrained only 
by 13818-1?

2. If the former in (2), then how does one carry a "real" transport stream?

3. If the latter in (2), then does this stream have to strictly adhere to 
13818-1? No mention of this is made, and if it is truly a 13818-1 transport 
stream, then without some PSI tables, one cannot unwrap the elementary 
streams from the transport, or even know what they are.

4. Either way, what elementary stream types are supported?  13818-1 is 
basically open ended on stream types.  So this applies to both mp2t and 
mp2p.  As written, the document is unbounded.  One might infer that the 
stream types are bounded by the elementary streams enumerated either in 
this document or in RFC 2250, but this is not stated.

5. Any reason not to include AC-3 (analogous to 13818-3)? This is the 
broadcast audio standard in the US, and is starting to creep into some 
European digital systems.  FYI, an ID was recently submitted by Dolby to 
register the content type, "audio/ac3".

The reason I care is that these content types need to coexist in the SMPTE 
and ATSC system environments so they can be re-used there with the same 
meaning.

Thanks,
         Mike


------------------------------------------------------
Michael A. Dolan, Representing DIRECTV,  (619)445-9070
PO Box 1673 Alpine, CA 91903        FAX: (208)545-6564




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Subject: optional CRC field in AMR payload
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Hi, all,

I'd like to get your opinion on whether an 8 bit optional CRC field is
useful in the AMR payload design. The coauthors of the two RTP AMR
payload drafts (draft-xie-avt-et-rtp-amr-00.txt and
draft-ietf-avt-rtp-amr-01.txt) have been having an off-line discussion
on this topic.

Opinion #1: since the transport layer and link layer of Internet are
currently bit-error intolerant (they have their own CRC or checksum to
filter out any bad data packet before it can reach the RTP receiver), a
CRC field in AMR payload won't add any value to the design and therefore
should not be included.

Opinion #2: bit error-tolerant to speech data is one big advantage of
AMR codec, we should not *design out* this feature when we define the
AMR payload simply because the current transport and link layer are not
ready for error-tolerant data transfer. There have already been talks
about the idea of error-tolerant data transfer support at the transport
and link layers and some initial work has already started (see
Pittsburgh AVT meeting minutes and UDP-Lite draft). We should be
forward-looking and include an optional CRC field in the AMR payload,
especially when the size cost of adding this optional CRC field is only
1 bit.

And my opinion is of cause Opinion #2 :-)

regards,
-Qiaobing



From rem-conf Mon Oct 30 23:48:27 2000 
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From: "Dr. Carsten Bormann" <cabo@tzi.org>
To: "Qiaobing Xie" <xieqb@cig.mot.com>, "AVT List" <rem-conf@es.net>
Cc: <lln@cdt.luth.se>, "Allison Mankin" <mankin@east.isi.edu>
Subject: RE: optional CRC field in AMR payload
Date: Tue, 31 Oct 2000 08:42:34 +0100
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> There have already been talks
> about the idea of error-tolerant data transfer support at the transport
> and link layers and some initial work has already started (see
> Pittsburgh AVT meeting minutes and UDP-Lite draft).

Another data point: The ROHC working group is trying hard to make header
compression tolerant not only of packet losses but also of residual bit
errors handed up from the link layer.
Together with IPv4's ability to switch off the UDP checksum, we already have
some support from the transport layer (and by the nature of ROHC, we are
adding protection to the RTP header, in some modes only by reducing its
cross-section, in others also by adding a 3-bit or larger CRC).
Work is needed for IPv6, though.

Gruesse, Carsten




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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Error Tolerant RTP Payload Format for AMR
	Author(s)	: Q. Xie, S. Gupta
	Filename	: draft-xie-avt-et-rtp-amr-01.txt
	Pages		: 
	Date		: 30-Oct-00
	
This document defines the RTP payload format for *error tolerant*
delivery of Adaptive Multi-Rate (AMR) speech frames over an RTP
session.

The flexibility on bandwidth requirements and the tolerance to bit
errors of AMR codes are not only beneficial for 'over-the-air'
wireless links, but are also very desirable for any Voice-over-IP
applications. The design is focused on how to best facilitate these
features of AMR codec in an IP environment.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-xie-avt-et-rtp-amr-01.txt

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From: "Henning G. Schulzrinne" <hgs@cs.columbia.edu>
Organization: Dept. of Computer Science, Columbia University
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 and Video (NOSSDAV) 2001
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 The 11th International Workshop on Network and Operating Systems
          Support for Digital Audio and Video (NOSSDAV 2001)

      Sponsored by the Association for Computing Machinery
      In cooperation with USENIX, ACM SIGCOMM, ACM SIGMM, ACM SIGOPS
      Technically cosponsored by IEEE Communications Society

                          June 25-27, 2001
         Danfords on the Sound, Port Jefferson, New York
             http://www.nossdav.org/2001

                          CALL FOR PAPERS

The 11th International Workshop on Network and Operating Systems
Support for Digital Audio and Video (NOSSDAV 2001) will bring together
participants from academia and industry to present and discuss new
ideas and future directions in networking and operating system support
for multimedia.  Exponential improvements in networking
bandwidth, cost, and ubiquity coupled with the growing availability
and use of streaming media content pose new networking and operating
system research opportunities and challenges at the global systems
level.  We solicit papers in current or emerging areas including but
not limited to ubiquitous multimedia services, broadband streaming
media content distribution, webcasting, end-to-end resource
management, real-time interactive streaming audio services,
internet telephony, interactive network games, wireless multimedia
systems, and multimedia information appliances and consumer devices.
The workshop particularly encourages submission of papers describing
implementations, measurements and field experience.

To ensure a productive workshop environment, attendance will be limited
to about 70 participants who are active in the field.  Each potential
participant should submit a position paper or extended abstract that
identifies new problems and explains why they are important,
challenges conventional wisdom, advocates a specific solution, and/or
reports on actual experience with real systems.  Papers will be
subject to a normal review process and are restricted to five pages.
Participants will be invited based on the originality, technical merit
and topical relevance of their submissions, as well as the likelihood
that the ideas expressed in their submissions will lead to insightful
technical discussions at the workshop.  Papers submitted to NOSSDAV
must not have been published or submitted elsewhere.  Please do not
submit abbreviated versions of journal or conference papers.  Authors
of accepted papers will be invited to present at the workshop and
publish full-length versions of their papers in the workshop
proceedings.

All submissions must be made through the web. For detailed guidelines
for submission, please refer to the workshop web site at
http://www.nossdav.org/2001.


Important Dates:
================

Submission Deadline for Papers:         February 15, 2001
Acceptance Notification:                April 2, 2001 
Camera Ready Deadline:                  May 15, 2001
Workshop:                               June 25-27, 2001

NOSSDAV 2001 Organizing Committee:
=================================

Chairs:
Jason Nieh, Columbia University
Henning Schulzrinne, Columbia University

Peter Danzig, Akamai
Christophe Diot, Sprint Advanced Technology Labs
Wu-chi Feng, Ohio State University
Anoop Gupta, Microsoft Research
Chuck Kalmanek, AT&T Research
Duane Northcutt, Sun Microsystems
Larry Peterson, Princeton University
Larry Rowe, University of California, Berkeley
Doug Shepherd, Lancaster University

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



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