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Date: 01 May 1998 16:08 EDT
Sender: "Vahe Balabanian" <balabani@nortel.ca>
To: rem-conf@es.net
From: "Vahe Balabanian" <balabani@nortel.ca>
Subject: AVT Consensus on MPEG4/DMIF
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This e-mail is to enable AVT to form a consensus on 
its approach to MPEG-4 and DMIF. It essentially 
addresses the various features of MPEG-4 and DMIF, 
their relevance to AVT in particular and to  IETF 
in general and finally it derives alternative courses 
of action within AVT. MPEG-4 and DMIF (Delivery 
Multimedia Integration Framework) are addressed 
separately since they present complementary and 
independent functions. 

This e-mail aims to stimulate the debate in AVT by  
inviting the audience on the rem-conf list to 
participate in the analysis. Interested members from 
the MPEG community will join AVT on the rem-conf list. 
At the conclusion of this debate, with the kind  help 
of Steve Casner,  I will summarize the result of the 
consensus for the course of action to take in AVT. 

I- What is MPEG-4 and how is it relevant to AVT and
IETF?
---------------------------------------------------- 

1-  MPEG-4 divides the screen into AudioVisualObjects 
(AVO) each having a desired shape (including geometric 
and natural shapes) and multiplicity of coded streams 
to be chosen from by the receiving end-user. 

Analysis: This approach is useful to AVT for enabling 
an end-user to adapt its viewing to changing network 
conditions.
 
2-  MPEG-4 composites the AVOs and renders them on the 
screen with the help of a Scene tree and related Object 
Descriptor information. Both of these are also 
classified as streams in MPEG-4. MPEG-4 uses a variety 
of timestamps and clocks to synchronize the display of 
the streams within and across the AVOs in a scene.

Analysis: AVT has already embarked on defining generic 
format RTP (GF-RTP) streams and is using Session 
Description Protocol to identify the characteristics of 
the GF-RTPs to the receiver. The similarities of MPEG-4 
and AVT in this area are in the equivalence of GF-RTP to
the MPEG-4 stream and the SDP to the Scene and Object 
Descriptor. Except in the latter case SDP is a queried 
entity as opposed to a stream. AVT is not involved in 
compositing or rendering of AVOs which remain purely 
MPEG-4 functions.

II- What is DMIF and how is it relevant to AVT and 
IETF?
---------------------------------------------------

1-  The abstract DMIF Application Interface (DAI) 
isolates MPEG-4  from the underlying delivery  
mechanisms. In this DAI makes use of a media-based QoS 
(expressed in access unit metrics of priority, loss, 
delay and jitter) and an explicit format description 
for each stream.

Analysis: This is useful to AVT in maintaining 
applications intact while transiting from IPV4 to IPV6 
and adopting multiple implementations of the TOS byte 
to carry Diff Serv information.

2-  DMIF allows tagging of resources used in a given 
presentation session. The tags are used to log and apply
policy to multiple streams on different source and 
destination addresses and over different networks.

Analysis: As IETF is moving towards usage accountability
with RSVP, this concept allows the collection of usage 
information for a given presentation. A joint server and
client policy can thus be applied to limit the usage of 
resources for a given presentation.

3-  DMIF allows efficient and agile packing of streams 
into transports and the reuse of existing transports to 
reduce the visual impact of addition of new higher 
priority streams at the expense of existing lower 
priority streams.

Analysis: This allows AVT users to experiment with 
different streams in order to optimize their viewing 
experience perhaps also in the face of changing network 
conditions reflected in RTCP reports.

III- Options for going forward with MPEG-4 and DMIF in 
AVT
-------------------------------------------------------

a)  Adopt MPEG-4 FlexMux for agile packing of multiple 
streams on a UDP flow as opposed to using RTP

Analysis: This will create a parallel track to RTP 
(including mixers, translators and multicast) and in 
time it could supersede RTP. Although this may be a 
realistic course of action it can only be adopted after 
the limit of RTP is duly tested first. AVT and IETF have
demonstrated ingenuity in adapting the existing 
structures to new evolving conditions.

b)  Adopt MPEG-4 FlexMux for agile packing of multiple 
streams on RTP. This is the proposal put on the table by
draft-ietf-avt-mpeg4-00.txt/ps.

Analysis: To some in AVT, FlexMux on RTP appears to be a
reasonable way to pack a large number of thin streams. 
It may require MPEG-4 to increase the size of the 
FlexMux MTU to 64 Kbytes. In contrast, others in AVT 
believe this to be unnecessary and would like to use 
only RTP. This may require restricting MPEG-4 streams 
to an MPEG-4 Profile created expressly for RTP. 

c)  Redesign the whole of MPEG-4 for IETF or instead use
DMIF as in draft-ietf-avt-mpeg4-dmif-00.txt

Analysis: Certain functions such as compositing and 
rendering are clearly not in the scope of AVT, in other 
words the skill set does not exist in AVT. This suggests
that a clear demarcation point must exist where both 
MPEG-4 and AVT can use their best skill sets and achieve
an optimum overall performance.  This demarcation point 
is described in draft-ietf-avt-mpeg4-dmif-00.txt. The 
argument is that this approach may compromise efficiency
in order to satisfy  the requirements of complete 
separation of application from the underlying delivery 
mechanisms. As resources on Internet are better utilized
and as more bandwidth capacity is available to RTP, the 
benefits of this approach will outweigh the extracting 
of the last few bits of efficiency for RTP.

Note: It is conceivable to think of the transfer of the 
DMIF design expertise to AVT since it falls within its 
domain of expertise. DMIF could be used to regulate the 
stream traffic on Internet in a similar fashion to 
regulating its TCP traffic.

d)  Use a subset of MPEG-4 suitable for AVT

Analysis: The stream formats in MPEG-4 cover a wide 
range of timestamps and clocks including implicit 
timestamps. It may be possible to subset these for 
operation over RTP.  This argues for the creation of 
MPEG-4 profile for RTP. But this breaks the clean 
separation adopted in DMIF for isolating the 
applications from the delivery mechanisms, thus MPEG-4 
profiles are normally reserved to applications and not 
to transports. But one can argue that within the 
confines of the RTP subset, the DAI principle can still 
be applied across both the storage and broadcast media. 
As experience is gained with RTP the subset can be 
enlarged to encompass more of MPEG-4.

IV-  Summary
------------

First consider option "d". A clear understanding why 
MPEG-4 uses a wide range of timestamps and clock 
mechanisms is required by AVT in order to arrive at an 
answer whether a useful subset for RTP can be found.  

Failing the above consider option "c". This allows AVT 
to propose a delivery mechanism which is most suitable 
for RTP. 

Failing the above consider option "b". This considers 
the use of FlexMux over RTP and ensures that the 
benefits of RTP  (including mixers, translators and 
multicast) can still be applied to MPEG-4. 

Finally failing the above use option "a". In this a 
parallel track is set for MPEG-4.


Regards,

Vahe Balabanian
Nortel
balabani@nortel.ca



From rem-conf Fri May 01 17:07:49 1998 
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From: Irene Yam <ireney@bmrc.berkeley.edu>
Subject: Re: UCB MIG Seminar, Wed., May 6, "The SoundFisher Sound
  Browser " Erling Wold, Muscle Fish, LLC 
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                                      The SoundFisher Sound Browser 

                (Wednesday May 6, 1998 12:30-2:00 PDT 405 Soda Hall) 

                                                     Erling Wold 
                                                 Muscle Fish, LLC 

Content-based retrieval of audio has found a place in automatic
segmentation and classification of audio and video recordings, multimedia
databases and file systems, surveillance and content-based audio processing
equipment. This talk will describe our approach to audio analysis and
comparison and how it is used in one particular application: the
SoundFisher sound effects browser and editor. Our audio analysis reduces
sounds to low-level perceptual and acoustical attributes as well as
higher-level statistical models of the behaviors of these attributes. Given
such an analysis, sounds can then be searched, classified or retrieved by
any one or a combination of the attributes, by specifying previously learned
categories based on these attributes, or by selecting or entering reference
sounds and asking the engine to retrieve sounds that are similar (or
dissimilar) to them. The talk will also cover the use of this technology in
some related applications. 

_______________________________________
This seminar will be broadcast on the Internet MBONE.  The broadcast will
begin at 12:40PST (GMT - 8 hrs).  See sd or http://bmrc.berkeley.edu/298 for
instructions on setting up, connecting to, and operating the MBONE tools.


                           

____________________________________________

Irene Yam
Berkeley Multimedia Research Center (BMRC)
626 Soda Hall #1776
University of California
Berkeley, CA 94720-1776
Phone: (510) 643-0800   FAX: (510) 642-5615  
Email: ireney@bmrc.berkeley.edu  
Url:  http://bmrc.berkeley.edu/people/irene/



From rem-conf Sat May 02 04:40:39 1998 
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Date: Sat, 02 May 1998 13:05:48 +0200
From: Arantza <etsit@tsc.upna.es>
Organization: Universidad Pública de Navarra
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II Medical Engeneering Course.

>From Public University of Navarra, Spain. Organized  by the department
of Electric and Electronic Ingeneering of this University and in
colaboration with the  Health Service of Navarra (Osasunbidea).
Daily, from 27th  April to 15th of May. 16:00 - 19:30 (GMT+1 hr).

For more information see:
http://www.iee.upna.es/ingemed/introduc.htm
ingemed@upna.es

This course will be broadcast on the Internet MBONE.  The broadcast will

begin at 16:00  (GMT +1 hrs).






From rem-conf Sat May 02 20:20:30 1998 
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        rem-conf@es.net
Subject: Re: Is www.mbone.net down ?
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www.mbone.com is back up now.
www.mbone.net has never been up.

-Pete-

Denis DeLaRoca wrote:
> 
> On Sat, 25 Apr 1998, Yiorgos Adamopoulos wrote:
> 
> >
> > I keep getting ``connection refused'' from any *.gr site that I have tried
> 
> Same with www.mbone.com. The DNS entries are gone so I presume that site
> is history... can anyone clarify that is the case and wether someone else
> will maintain the mbone archives and resource directory that used to
> reside at www.mbone.com?
> 
> -- Denis



From rem-conf Mon May 04 14:50:25 1998 
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From: "Anders Klemets (Exchange)" <anderskl@exchange.microsoft.com>
To: rem-conf@es.net
Subject: RE: AVT Consensus on MPEG4/DMIF
Date: Mon, 4 May 1998 14:27:50 -0700 
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> b)  Adopt MPEG-4 FlexMux for agile packing of multiple 
> streams on RTP. This is the proposal put on the table by
> draft-ietf-avt-mpeg4-00.txt/ps.
> 
> Analysis: To some in AVT, FlexMux on RTP appears to be a
> reasonable way to pack a large number of thin streams. 

It is clear that if you have a large number of very low bit rate streams, it
might be reasonable to bundle data from different streams into a single RTP
packet.  

However, FlexMux seems to be an unnecessarily complex approach for achieving
this goal.  It is not clear how useful a FlexMux PDU in MuxCode mode is,
when used over RTP.  (See pages 161-164 in the MPEG-4 Systems CD, starting
with section 7.5.2.2, for a detailed description of MuxCode mode.)

MuxCode mode is used for byte interleaving of data from different streams.
A MuxCodeTableEntry specifies exactly how the interleaving is done.  Unless
you have the MuxCodeTableEntry, you will not be able to decode the FlexMux
PDU.

Here is an example of how you can use the MuxCodeTableEntry.  Suppose a
FlexMux PDU contains data from two streams, A and B.  A possible
interleaving algorithm that could be expressed by a MuxCodeTableEntry could
look like this:

do
  do 
    1 byte from A and 1 byte from B
  repeat 3 times
  do 
    2 bytes from A and 3 bytes from B
  repeat 2 times
repeat until end of FlexMux PDU

It would seem to me that MuxCode mode could be useful on packet radio
networks, where you have to worry about bit errors.  If you decide to
process a received packet even though the UDP checksum failed, byte
interleaving could spread a burst of bit errors across data from multiple
streams, making the errors easier to correct using FEC.  But other than this
particular scenario, I don't see any particular benefit from having to
implement support for MuxCode mode.

Anders



From rem-conf Mon May 04 20:14:55 1998 
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Date: Mon, 4 May 1998 20:07:38 -0700 (Pacific Daylight Time)
From: Stephen Casner <casner@precept.com>
To: GOULEAU Emmanuel CNET/DSM/REN <emmanuel.gouleau@cnet.francetelecom.fr>
cc: rem-conf@es.net
Subject: Re: Payload format for "MPEG1 System"
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On Fri, 24 Apr 1998, GOULEAU Emmanuel CNET/DSM/REN wrote:

> I would like to know if a Payload Format has been defined for "MPEG1
> System".

I'm not sure whether you are asking about a payload format or a static
payload type number.  The answer to the former is yes, and to the
latter, no.  Assuming you're asking about the latter, here's more
detail which I'm sending because I recently needed to answer the same
question in a different context.

The payload format specification for MPEG1/2 in RFC 2250 defines five
different data types that may be carried, but only three of them have
static RTP payload types assigned.  Since the payload type space is
small, we were reluctant to assign five values all at once.  Instead,
at the time the spec was written the three modes expected most likely
to be used were assigned values, and the other two were left to
dynamic payload type assignment or possible future static assignment
if the other modes were found to be needed.

When experiments with RTP began, it was necessary to use statically
assigned payload types because that was the only mechanism available.
However, in the time since the RTP specification and A/V Profile were
published as RFCs 1889 and 1890, the mechanisms for defining dynamic
payload type mappings have been specified in the Session Description
Protocol (recently published as RFC 2327) and in other protocols such
as H.323/H.245.  The way a dynamic payload type works is that a longer
name for the encoding/payload-format combination is mapped to a
payload type number in the range 96-127.  That mapping applies only to
the RTP session for which it is made, thus the numbers can be re-used
for different encodings in different sessions so the space limitation
is avoided.  There are now many systems implementing these dynamic
payload type mechanisms.

It is recognized that there would eventually be a problem with running
out of space to make further static payload type assignments, and the
mechanisms for using dynamic payload types are now established.
Therefore, at the past couple of IETF meetings the AVT working group
has discussed revising the policy for static payload type assignments
such that no further assignments would be made.  The minutes of the
most recent meeting include the following:

    As had been discussed in Munich, it was agreed that the RTP
    Profile draft should be revised to say no more static payload
    types would be assigned.  Those that are already assigned should
    be considered "default" values in the dynamic payload type table,
    and may be overwritten once all the unassigned values have been
    used for dynamic types within a session.

This new policy has not been published yet, but we plan to get the
profile draft (draft-ietf-avt-profile-new-02.txt/ps) revised and
published soon.

Please consider the use of a dynamic payload type value for your
applications.  See the SDP spec (RFC 2327) for examples of how the
dynamic payload type mapping is expressed using the "rtpmap"
attribute.  If the system you are implementing does not use SDP to
describe sessions, the mapping between the dynamic payload type value
and the longer name may be conveyed along with the address and port
number using whatever communication path you use now for that address
and port number.  If you consider it implicit in your connection
establishment procedure that MPEG1 Systems Streams will always be
used, then you could simply use PT 96 and assume that it is implicitly
mapped.  However, I advise against making that assumption because if
you decide later to use another encoding/payload-format, or if you try
to interoperate with another implementation that has made a different
assumption, then communication will fail.

The revision of the profile draft needs to include some additional
encoding/payload-format names so that those names may be used in
dynamic payload type mappings.  In looking at the profile draft, I
see that names are not defined for MPEG1 Systems and MPEG2 Programs
streams.  Following the model of MP2T, I suggest that we define the
names MP1S and MP2P, however that may be open to discussion.

It was also resolved at the recent IETF meeting that the payload
format names would be registered in the MIME type namespace, so that
is the registration which needs to occur now for all the existing
names in the draft plus others such as the two new ones for MPEG.

							-- Steve




From rem-conf Mon May 04 20:43:23 1998 
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Date: Mon, 04 May 1998 23:45:50 -0400
From: Yasser Rasheed <yrasheed@jupiter.nal.utoronto.ca>
Organization: University Of Toronto
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Dear colleagues,
I am preparing a survey on MPEG2 tools available in both hardware and
software.
I am looking for encoders, decoders and testing tools, for layered and
non-layered MPEG2.
I would appreciate any feedback regarding this issue and please also let
me know if such
information is compiled anywhere.

Thank you for your attention.

--Yasser Rasheed.

_________________________________________________________________________________

YASSER RASHEED

University of Toronto, ECE Dept.,
10 king's College Road
Toronto Ont M5S 3G4 Canada

Tel:   (416)-978-4829
Fax:   (416)-978-4425
Email: yrasheed@nal.utoronto.ca

http://www.comm.utoronto.ca/~yrasheed





From rem-conf Tue May 05 01:24:17 1998 
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[Anders:]

It would seem to me that MuxCode mode could be useful on packet radio
networks, where you have to worry about bit errors.  If you decide to
process a received packet even though the UDP checksum failed, byte
interleaving could spread a burst of bit errors across data from multiple
streams, making the errors easier to correct using FEC.  But other than 
this
particular scenario, I don't see any particular benefit from having to
implement support for MuxCode mode.

[Reply:]

MuxCode can save bandwidth in case of multiple streams with very small 
access units of known size. This may happen in animation streams, for 
instance. Sometimes the units may be 2-3 bytes long, and the use of MuxCode 
can reduce bandwidth by 20-30%.

Zvi (newbi in this forum, not so much in MPEG :-)

=====================
Zvi Lifshitz (zvil@vdo.net)
VDOnet Corp. (www.vdo.net)
Phone +972(2)679-4788
Fax +972(2)679-4789
=====================





From rem-conf Tue May 05 02:05:18 1998 
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> Dear colleagues,
> I am preparing a survey on MPEG2 tools available in both hardware and
> software.
> I am looking for encoders, decoders and testing tools, for layered and
> non-layered MPEG2.
> I would appreciate any feedback regarding this issue and please also let
> me know if such
> information is compiled anywhere.
> 
> Thank you for your attention.
> 
> --Yasser Rasheed.
> 

see http://www.mpeg.org/index.html/



	DAVID
	-----



From rem-conf Tue May 05 06:17:52 1998 
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I would need the exact specification of the tunneling used by moruted. We
have to enable a tunnel through a firewall. The firewall expert asked me
about the type of tunneling: Is it GRE? We use mrouted 3.8 and Cisco
routers.

Thanks in advance.

-----------------------------------------------------------
  Andras Micsik              micsik@sztaki.hu
  MTA SZTAKI Hungary         http://www.sztaki.hu/~micsik




From rem-conf Tue May 05 17:25:23 1998 
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From: "Anders Klemets (Exchange)" <anderskl@exchange.microsoft.com>
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Subject: RE: AVT Consensus on MPEG4/DMIF
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> MuxCode can save bandwidth in case of multiple streams with 
> very small 
> access units of known size. This may happen in animation streams, for 
> instance. Sometimes the units may be 2-3 bytes long, and the 
> use of MuxCode 
> can reduce bandwidth by 20-30%.

Thanks for your reply.  Perhaps I am missing something obvious.  Why would
byte interleaving cause the bandwidth to be reduced?  You can shuffle the
bytes around any way you want, and it's still going to be the same number of
bytes.

One way to save bandwidth would be to eliminate certain common fields from
the AL-PDU headers.  Suppose, for example, you want to multiplex three
AL-PDU's from different streams into one FlexMux-PDU.  If the three AL-PDU
headers are sufficiently similar, (same timestamps, etc.) it may pay off to
compress the headers.  However, FlexMux does not say that any such AL-PDU
header compression should be done.  What am I missing?

Anders



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Subject: EDOC '98: Final Call for Papers & Invited Speakers
Date: Tue, 5 May 1998 21:08:21 -0500
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[Please forward this message to colleagues who might be interested.
We apologize if you receive multiple copies.]

The following is the final Call for Papers for the 2nd International
Enterprise Distributed Object Computing Workshop (EDOC '98).

A HTML version of the CFP can be found at:
    http://uml.systemhouse.mci.com/events/edoc98/home.htm
A PDF version of the CFP can be found at:
    http://uml.systemhouse.mci.com/events/edoc98/docs/edoc98-cfp.pdf

Invited speakers for the workshop include:

    Ivar Jacobson, VP of Business Engineering, Rational Software Corp.
    Phil Bernstein, Repository Architect, Microsoft Corp.

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

***** CALL FOR PAPERS *****

2nd International Enterprise Distributed Object Computing Workshop (EDOC
'98)

November 3-5, 1998

Hyatt Regency La Jolla, San Diego, California, USA

http://uml.systemhouse.mci.com/events/edoc98/home.htm

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

Hosted by:
    MCI Systemhouse Corporation

Technical Co-Sponsors:
    Object Management Group (OMG)
    IEEE Communications Society, Enterprise Networking Technical
Committee

Endorsed by:
    Institute of Electronics, Information and Communication Engineers of
Japan
    Information Processing Society of Japan (IPSJ)

Corporate sponsors:
    Distributed Systems Technology Centre (DSTC)
    Unisys Corp.
    Fujitsu Corp.

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

SCOPE:
~~~~~~
The software industry is in the midst of a major shift in technology
from
static, monolithic systems to the era of distributed objects and
components.
The remarkable level of interest in the first Enterprise Distributed
Object
Computing (EDOC '97) Workshop demonstrated that this transformation is
well
underway. However, there remain some major challenges to be overcome if
distributed objects and components are to fulfill their full potential
for
solving enterprise problems.

The goal of the EDOC workshops is to provide a leading forum for the
exchange of ideas and results in this rapidly developing field of
computing,
and to establish a new community of researchers and industry experts to
address the enterprise issues associated with distributed object
technology.
Building on the success of the first EDOC workshop, EDOC '98 will focus
on
securing high-quality papers and tutorials from around the world, and
will
provide ample opportunities for discussion and debate. In addition,
there
will be relevant panel sessions with substantial time for open
discussions.

EDOC '98 will target two interrelated categories of topics. The first
group
will deal with the specification of enterprise models for distributed
systems. This includes issues related to the specification of roles,
policies, business objects and their relationships in open distributed
systems. The second group will address the problems of implementing
systems
specified by enterprise models using currently available distributed
object
technologies (e.g., CORBA, DCOM, Java-RMI, TINA, RM-ODP). This includes
issues related to architectural models, technology bridges and the
general
problem of middleware selection and application.

TOPICS:
~~~~~~~
The program committee seeks research results, experience reports and
case
studies related to all aspects of enterprise distributed object
computing,
including such topics as:

- Distributed object architectures and frameworks
- Distributed business objects
- Analysis and design methods
- Metamodeling architectures
- Collaborative and workflow systems
- ODP enterprise language and business modeling
- UML for enterprise applications
- Enterprise policy, management and security
- Enterprise Quality of Service
- Traders, brokers, and agents for enterprise solutions
- Application of the above technologies in vertical application
  domains such as telecomunnications, business and financial systems

Papers must be of high quality and will be evaluated for originality,
significance, insight, clarity and their relevance to the enterprise
aspects
of distributed object computing.

IMPORTANT DATES:
~~~~~~~~~~~~~~~~
June 1st, 1998: Papers due
August 10, 1998: Notification of acceptance
September 11, 1998: Final papers due
November 3-5, 1998: Workshop

INFORMATION FOR AUTHORS:
~~~~~~~~~~~~~~~~~~~~~~~~
PAPERS:
Authors are invited to submit full papers according to the submission
guidelines below. Papers will be refereed by an international committee
of experts. Presented papers will be published by the IEEE
Computer Society Press.

SUBMISSIONS:
The papers should not exceed 12 pages, according to IEEE format
(double column, single space, 10 point Times font). The preferred method
for submission is electronic mail (Word, RTF, PostScript or PDF format).
Please send your submissions to the following address:
    edoc98-submissions@omg.org

Enquiries should be directed to:
    edoc98-info@omg.org

WORKSHOP ORGANIZERS:
~~~~~~~~~~~~~~~~~~~~
General Chair:
    Cris Kobryn (MCI Systemhouse, USA)
Program Chairs:
     Colin Atkinson (IESE, Germany)
     Zoran Milosevic (DSTC, Australia)
Program Committee:
     Nigel Baker (University of West England, UK)
     Phil Bernstein (Microsoft, USA)
     Eng Chew (Optus, Australia)
     Jim Coplien (Bell Labs, USA)
     Fred Cummins (EDS, USA)
     Takeo Hamada (Fujitsu Laboratories of America, USA)
     Sridar Iyengar (Unisys, USA)
     Peter Linington (University of Kent at Canterbury, UK)
     Claudia Linnhoff-Popien (RWTH Aachen, Germany)
     George Karetsos (National Technical University of Athens, Greece)
     Subrata Mazumdar (Bell Labs, USA)
     Osamu Miyagishi (NTT, Japan)
     Luke O'Connor (IBM Zurich, Switzerland)
     Maria Orlowska (The University of Queensland, Australia)
     Jong-Tae Park (Kyungpook National University, Korea)
     Thomas Preuss (Brandenburg University of Technology, Germany)
     Kerry Raymond (DSTC, Australia)
     Roberto Saracco (CSELT, Italy)
     Marc-Thomas Schmidt (IBM, USA)
     Morris Sloman (Imperial College, UK)
     Richard Mark Soley (OMG, USA)
     Jean-Bernard Stefani (CNET France Telecom, France)
     Keith Swenson (Netscape, USA)
     Sandy Tyndale-Biscoe (Birchquest Ltd, UK)
     Kevin Tyson (Enterprise Engineering, USA)
     Sanya Uehara  (Fujitsu, Japan)
     Karl-Heinz Weiss (Public Administration Berlin, Germany)
     Edward White (MCI, USA)
     Bryan Wood (Open-IT, UK)
     Douglas Zuckerman (Bellcore, USA)

Local Organizing Committee:
     Cheryl Bissonnette (OMG, USA)

VENUE:
~~~~~~
EDOC '98 will be held in San Diego, California at one of Southern
California's premier venues: the Hyatt Regency La Jolla at Aventine. The
hotel is a San Diego landmark destination and a Mobil 4-Star and AAA
4-Diamond Award winner. It provides the perfect complement of business
and
recreation facilities.

The Aventine complex includes the Hyatt Regency La Jolla, a 32,000 sq.
ft.
world-class Sporting Club and Spa, a restaurant village with
internationally
themed restaurants, lighted tennis courts and a heated pool. The hotel
is
five minutes from golf and the beach and 15 minutes from downtown San
Diego.

ENQUIRIES AND REGISTRATION DETAILS:
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Accommodation at a discounted conference rate is available at the
workshop
venue. Registration and accommodation details will be available in the
near
future.
For enquiries, please email to Cheryl Bissonette at:
    cheryl@omg.org
Postal address:
    Object Management Group
    492 Old Connecticut Path
    Framingham, MA 01701
    USA

RELATED EVENTS:
~~~~~~~~~~~~~~~
November 9-13, 1998     OMG Technical Committee meeting, Burlingame, CA,
USA

WWW PAGE:
~~~~~~~~~
Further details about the EDOC'98 event and this Call For Papers can be
found at:
    http://uml.systemhouse.mci.com/events/edoc98/home.htm



From rem-conf Tue May 05 20:05:34 1998 
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Date: Tue, 5 May 1998 20:00:15 -0700 (Pacific Daylight Time)
From: Stephen Casner <casner@precept.com>
To: rem-conf@es.net
Subject: Re: AVT WG last call on two drafts
In-Reply-To: <Pine.WNT.3.96.980410112154.-3908883F-100000@oak.precept.com>
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To the AVT working group:

The "working group last call for comments" period has concluded for
the two drafts:

    "Options for Repair of Streaming Media"
    draft-ietf-avt-info-repair-03.txt
    Informational RFC

    "RTP Payload Format for the 1998 Version of ITU-T Rec. H.263 Video
    (H.263+)" 
    draft-ietf-avt-rtp-h263-video-01.txt
    Proposed Standard RFC

I believe there were no comments on the first draft.

There was some discussion of the relationship between the second draft
and RFC 2190 on the earlier version of H.263.  My interpretation of
the discussion was that everyone agreed that including a paragraph in
the new spec explaining its relationship to 2190 is a good idea.
Furthermore, although there were several opinions stated pro & con
regarding the designation of 2190 as "updated by" the new one, the
folks who were concerned about that designation were persuaded that it
would be OK.

I will now send a request for IESG Last Call on the info-repair
draft.  I will send a similar request for the h263-video draft as soon
as the authors have posted the updated version including the
aforementioned paragraph and other minor fixes previously discussed.

							-- Steve




From rem-conf Wed May 06 00:09:57 1998 
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        "'Anders Klemets (Exchange)'"
	 <anderskl@exchange.microsoft.com>
Subject: RE: AVT Consensus on MPEG4/DMIF
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[Anders:]

Thanks for your reply.  Perhaps I am missing something obvious.  Why would
byte interleaving cause the bandwidth to be reduced?  You can shuffle the
bytes around any way you want, and it's still going to be the same number 
of
bytes.

[Reply:]

Example:

We have 4 access units, from streams 1, 2, 3 and 4, each 4 bytes long. 
Without MuxCode, we will convey them in 4 FlexMux PDUs, each consists of 1 
byte for PDU length, 1 byte for stream number, and 4 bytes for payload. In 
total 4*(2+4)=24.

With MuxCode this will be 1 byte for PDU length, 1 byte for MuxCode (which 
tells "payload contains AUs from stream 1, 2, 3 and 4, each 4 bytes long), 
and payloads for the 4 PDUs. In total 2+4*4=18. 25% reduction if you pay in 
cash.

[Original Message:]

One way to save bandwidth would be to eliminate certain common fields from
the AL-PDU headers.  Suppose, for example, you want to multiplex three
AL-PDU's from different streams into one FlexMux-PDU.  If the three AL-PDU
headers are sufficiently similar, (same timestamps, etc.) it may pay off to
compress the headers.  However, FlexMux does not say that any such AL-PDU
header compression should be done.  What am I missing?

[Reply:]

Could be done, but was not :-(.

Zvi

=====================
Zvi Lifshitz (zvil@vdo.net)
VDOnet Corp. (www.vdo.net)
Phone +972(2)679-4788
Fax +972(2)679-4789
=====================





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        mm-all@bmrc.Berkeley.EDU
From: Irene Yam <ireney@bmrc.berkeley.edu>
Subject: Re: UCB MIG Seminar, Wed., May 6, "The SoundFisher Sound
  Browser " Erling Wold, Muscle Fish, LLC 
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Please join us for our last MIG Seminar for Spring 1998!  


                                      The SoundFisher Sound Browser 

                (Wednesday May 6, 1998 12:30-2:00 PDT 405 Soda Hall) 

                                                     Erling Wold 
                                                 Muscle Fish, LLC 

Content-based retrieval of audio has found a place in automatic
segmentation and classification of audio and video recordings, multimedia
databases and file systems, surveillance and content-based audio processing
equipment. This talk will describe our approach to audio analysis and
comparison and how it is used in one particular application: the
SoundFisher sound effects browser and editor. Our audio analysis reduces
sounds to low-level perceptual and acoustical attributes as well as
higher-level statistical models of the behaviors of these attributes. Given
such an analysis, sounds can then be searched, classified or retrieved by
any one or a combination of the attributes, by specifying previously learned
categories based on these attributes, or by selecting or entering reference
sounds and asking the engine to retrieve sounds that are similar (or
dissimilar) to them. The talk will also cover the use of this technology in
some related applications. 

_______________________________________
This seminar will be broadcast on the Internet MBONE.  The broadcast will
begin at 12:40PST (GMT - 8 hrs).  See sd or http://bmrc.berkeley.edu/298 for
instructions on setting up, connecting to, and operating the MBONE tools.


                           

____________________________________________

Irene Yam
Berkeley Multimedia Research Center (BMRC)
626 Soda Hall #1776
University of California
Berkeley, CA 94720-1776
Phone: (510) 643-0800   FAX: (510) 642-5615  
Email: ireney@bmrc.berkeley.edu  
Url:  http://bmrc.berkeley.edu/people/irene/



From rem-conf Thu May 07 06:12:07 1998 
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         ESPRIT Information Day with EC and CEEC
on Last Call for Proposals in the 4th Framework Programme

              Friday, 8 May 1998, Budapest, Hungary

We are pleased to announce that the ESPRIT Information Day 
will be broadcasted over the MBONE. (More about the broadcast
at the end of this mail.)

The aim of the event is to disseminate information on
(1) the Last Call for Proposals in ESPRIT,
(2) the Fifth Framework Programme.

The speakers include Mr. Klaus Woelcken, Mr.  Anton Schrag and Mr. Bill
Collin (EU DG III), and Mr. Sandor Bottka (OMFB).

The programme on Friday, 8th May 1998, is as follows:

09:00-09.30  Welcome

09:30-11:00  K. Woelcken: The Fifth Framework programme:
              current status and options for the CEE countries 

             A. Schrag: The current ESPRIT call for        
              proposals in the 4th Framework Programme,        
              participation possibilities for the CEE countries

             Speakers from CEE countries on their project 
              experiences in the national R&D programmes.

11:30-13.00  Discussion, Q&A session on proposal preparation

14:00-15:30  Regional discussion of project participants - practical 
                   advice on proposals (Chair: B. Collin)

---------------------------------------------------------------------
About the broadcast over MBONE

Please note that you can only watch and listen to the event but 
cannot actively participate (i.e. put questions or make comments). 

1. How to connect?
    Join the session called "ESPRIT Info Day Budapest" in the 
    session directory tool. 

2. More info about the MBONE
    See: <http://www.mbone.com> or
           <http://www-mice.cs.ucl.ac.uk/merci/> in English, or
           <http://www.sztaki.hu/services/mbone> in Hungarian





From rem-conf Thu May 07 15:49:48 1998 
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From: "Gim Deisher" <gim.l.deisher@intel.com>
To: "IETF Draft Submission (E-mail)" <internet-drafts@ietf.org>,
        "Rem-Conf Mailing List (E-mail)" <rem-conf@es.net>,
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Cc: "Carsten Bormann (E-mail)" <cabo@tzi.org>,
        "Linda Cline (E-mail)" <lscline@jf.intel.com>,
        "Gim Deisher (E-mail)" <gim.l.deisher@intel.com>,
        "Tom Gardos (E-mail)" <thomas.r.gardos@intel.com>,
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        "Stephan Wenger (E-mail)" <stewe@cs.tu-berlin.de>,
        "Chad Zhu (E-mail)" <chad.zhu@intel.com>
Subject: New H.263+ RTP Payload Draft
Date: Thu, 7 May 1998 15:33:05 -0700
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Enclosed is a new revision of the H.263+ RTP payload format 
specification which is to replace the 1/98 IETF internet draft, 
draft-ietf-avt-rtp-h263-video-01.txt.

Here is a brief summary on the changes:

1. Extended the first paragraph to describe the relationship of this 
   payload draft and RFC2190. (section 1)
2. Mentioned P-picture temporal scalability (sections 1 & 4.2).
3. Made a correction in the RTP timestamp description. (section 2.1)
4. Clarified GOB boundary packetization. (section 2.2)
5. Adjusted diagrams.  No change in payload format. (sections 5.1.1, 
   5.1.2, 5.1.3, 5.2)
6. Added/deleted references. (section 8)
7. A few other minor editorial changes.

Comments welcome as usual.  Thank you.

------------------------------------------------------------------------
Internet Engineering Task Force                 Audio-Video Transport WG
draft-ietf-avt-rtp-h263-video-02.txt           C. Bormann / Univ. Bremen
May 7, 1998                                             L. Cline / Intel
                                                      G. Deisher / Intel
                                                       T. Gardos / Intel
                                                     C. Maciocco / Intel
                                                       D. Newell / Intel
                                                   J. Ott / Univ. Bremen
                                                G. Sullivan / PictureTel
                                                   S. Wenger / TU Berlin
                                                          C. Zhu / Intel


               RTP Payload Format for the 1998 Version of
                    ITU-T Rec. H.263 Video (H.263+)


Status of This Memo

This document is an Internet-Draft.  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 made obsolete 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."

To learn the current status of any Internet-Draft, please check the
"1id-abstracts.txt" listing contained in the Internet-Drafts Shadow
Directories on ftp.is.co.za (Africa), nic.nordu.net (Europe), 
munnari.oz.au (Pacific Rim), ds.internic.net (US East Coast), or 
ftp.isi.edu (US West Coast).

Distribution of this document is unlimited.


1. Introduction

This document specifies an RTP payload header format applicable to the 
transmission of video streams generated based on the 1998 version of 
ITU-T Recommendation H.263 [4].  Because the 1998 version of H.263 is a 
superset of the 1996 syntax, this format can also be used with the 1996 
version of H.263 [3], and is recommended for this use by new 
implementations.  This format does not replace RFC 2190, which continues 
to be used by existing implementations, and may be required for backward 
compatibility in new implementations.  Implementations using the new 
features of the 1998 version of H.263 shall use the format described in 
this document.

The 1998 version of ITU-T Recommendation H.263 added numerous coding 
options to improve codec performance over the 1996 version.  The 1998 
version is referred to as H.263+ in this document.  Among the new 
options, the ones with the biggest impact on the RTP payload 
specification and the error resilience of the video content are the 
slice structured mode, the independent segment decoding mode, the 
reference picture selection mode, and the scalability mode.  This 
section summarizes the impact of these new coding options on 
packetization.  Refer to [4] for more information on coding options.

The slice structured mode was added to H.263+ for three purposes: to 
provide enhanced error resilience capability, to make the bitstream more 
amenable to use with an underlying packet transport such as RTP, and to 
minimize video delay.  The slice structured mode supports fragmentation 
at macroblock boundaries.

With the independent segment decoding option, a video picture frame is 
broken into segments and encoded in such a way that each segment is 
independently decodable.  Utilizing ISD in a lossy network environment 
helps to prevent the propagation of errors from one segment of the 
picture to others.

The reference picture selection mode allows the use of an older 
reference picture rather than the one immediately preceding the current 
picture.  Usually, the last transmitted frame is implicitly used as the 
reference picture for inter-frame prediction.  If the reference picture 
selection mode is used, the data stream carries information on what 
reference frame should be used, indicated by the temporal reference as 
an ID for that reference frame.  The reference picture selection mode 
can be used with or without a back channel, which provides information 
to the encoder about the internal status of the decoder.  However, no 
special provision is made herein for carrying back channel information.

H.263+ also includes bitstream scalability as an optional coding mode.
Three kinds of scalability are defined: temporal, signal-to-noise ratio
(SNR), and spatial scalability.  Temporal scalability is achieved via 
the disposable nature of bi-directionally predicted frames, or B-frames. 
(A low-delay form of temporal scalability known as P-picture temporal 
scalability can also be achieved by using the reference picture 
selection mode described in the previous paragraph.)  SNR scalability 
permits refinement of encoded video frames, thereby improving the 
quality (or SNR).  Spatial scalability is similar to SNR scalability 
except the refinement layer is twice the size of the base layer in the 
horizontal dimension, vertical dimension, or both.


2. Usage of RTP

When transmitting H.263+ video streams over the Internet, the output of 
the encoder can be packetized directly.  All the bits resulting from the 
bitstream including the fixed length codes and variable length codes 
will be included in the packet, with the only exception being that when 
the payload of a packet begins with a Picture, GOB, Slice, EOS, or EOSBS 
start code, the first two (all-zero) bytes of the start code are removed 
and replaced by setting an indicator bit in the payload header.

For H.263+ bitstreams coded with temporal, spatial, or SNR scalability, 
each layer may be transported to a different network address.  More 
specifically, each layer may use a unique IP address and port number 
combination.  The temporal relations between layers shall be expressed 
using the RTP timestamp so that they can be synchronized at the 
receiving ends in multicast or unicast applications.

The H.263+ video stream will be carried as payload data within RTP 
packets.  A new H.263+ payload header is defined in section 4.  This 
section defines the usage of the RTP fixed header and H.263+ video 
packet structure.


2.1 RTP Header Usage

Each RTP packet starts with a fixed RTP header.  The following fields of 
the RTP fixed header are used for H.263+ video streams:

Marker bit (M bit): The Marker bit of the RTP header is set to 1 when 
the current packet carries the end of current frame, and is 0 otherwise.

Payload Type (PT): The Payload Type shall specify the H.263+ video 
payload format.

Timestamp: The RTP Timestamp encodes the sampling instance of the first 
video frame data contained in the RTP data packet.  The RTP timestamp 
shall be the same on successive packets if a video frame occupies more 
than one packet.  In a multilayer scenario, all pictures corresponding 
to the same temporal reference should use the same timestamp.  If 
temporal scalability is used (if B-frames are present), the timestamp 
may not be monotonically increasing in the RTP stream.  If B-frames are 
transmitted on a separate layer and address, they must be synchronized 
properly with the reference frames.  Refer to the 1998 ITU-T 
Recommendation H.263 [4] for information on required transmission order 
to a decoder.  For an H.263+ video stream, the RTP timestamp is based on 
a 90 kHz clock, the same as that of the RTP payload for H.261 stream 
[5].  Since both the H.263+ data and the RTP header contain time 
information, it is required that those timing information run 
synchronously.  That is, both the RTP timestamp and the temporal 
reference (TR in the picture header of H.263) should carry the same 
relative timing information.  Any H.263+ picture clock frequency can be 
expressed as 1800000/(cd*cf) source pictures per second, in which cd is 
an integer from 1 to 127 and cf is either 1000 or 1001.  Using the 
90 kHz clock of the RTP timestamp, the time increment between each coded 
H.263+ picture should therefore be a integer multiple of (cd*cf)/20.  
This will always be an integer for any "reasonable" picture clock 
frequency (for example, it is 3003 for 29.97 Hz NTSC, 3600 for 25 Hz 
PAL, 3750 for 24 Hz film, and 1500, 1250 and 1200 for the computer 
display update rates of 60, 72 and 75 Hz, respectively).  For RTP 
packetization of hypothetical H.263+ bitstreams using "unreasonable" 
custom picture clock frequencies, mathematical rounding could become 
necessary for generating the RTP timestamps.


2.2 Video Packet Structure

A section of an H.263+ compressed bitstream is carried as a payload 
within each RTP packet.  For each RTP packet, the RTP header is followed 
by an H.263+ payload header, which is followed by a number of bytes of a 
standard H.263+ compressed bitstream.  The size of the H.263+ payload 
header is variable depending on the payload involved as detailed in the 
section 4.  The layout of the RTP H.263+ video packet is shown as:

   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
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  |    RTP Header                                               ...
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  |    H.263+ Payload Header                                    ...
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  |    H.263+ Compressed Data Stream                            ...
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Any H.263+ start codes can be byte aligned by an encoder by using the 
stuffing mechanisms of H.263+.  As specified in H.263+, picture, slice, 
and EOSBS starts codes shall always be byte aligned, and GOB and EOS 
start codes may be byte aligned.  For packetization purposes, GOB start 
codes should be byte aligned; however, since this is not required in 
H.263+, there may be some cases where GOB start codes are not aligned, 
such as when transmitting existing content, or when using H.263 encoders 
that do not support GOB start code alignment.  In this case, follow-on 
packets (see section 5.2) should be used for packetization.

All H.263+ start codes (Picture, GOB, Slice, EOS, and EOSBS) begin with 
16 zero-valued bits.  If a start code is byte aligned and it occurs at 
the beginning of a packet, these two bytes shall be removed from the 
H.263+ compressed data stream in the packetization process and shall 
instead be represented by setting a bit (the P bit) in the payload 
header.


3. Design Considerations

The goals of this payload format are to specify an efficient way of 
encapsulating an H.263+ standard compliant bitstream and to enhance the 
resiliency towards packet losses.  Due to the large number of different 
possible coding schemes in H.263+, a copy of the picture header with 
configuration information is inserted into the payload header when 
appropriate.  The use of that copy of the picture header along with the 
payload data can allow decoding of a received packet even in such cases 
in which another packet containing the original picture header becomes 
lost.

There are a few assumptions and constraints associated with this H.263+ 
payload header design.  The purpose of this section is to point out 
various design issues and also to discuss several coding options 
provided by H.263+ that may impact the performance of network-based 
H.263+ video.


o The optional slice structured mode described in Annex K of H.263+ [4]
  enables more flexibility for packetization.  Similar to a picture
  segment that begins with a GOB header, the motion vector predictors in
  a slice are restricted to reside within its boundaries.  However,
  slices provide much greater freedom in the selection of the size and
  shape of the area which is represented as a distinct decodable region. 
  In particular, slices can have a size which is dynamically selected to 
  allow the data for each slice to fit into a chosen packet size.  
  Slices can also be chosen to have a rectangular shape which is
  conducive for minimizing the impact of errors and packet losses on 
  motion compensated prediction.  For these reasons, the use of the 
  slice structured mode is strongly recommended for any applications 
  used in environments where significant packet loss occurs.

o In non-rectangular slice structured mode, only complete slices should
  be included in a packet.  In other words, slices should not be
  fragmented across packet boundaries.  The only reasonable need for a 
  slice to be fragmented across packet boundaries is when the encoder 
  which generated the H.263+ data stream could not be influenced by an 
  awareness of the packetization process (such as when sending H.263+ 
  data through a network other than the one to which the encoder is 
  attached, as in network gateway implementations).  Optimally, each 
  packet will contain only one slice.

o The independent segment decoding (ISD) described in Annex R of [4]
  prevents any data dependency across slice or GOB boundaries in the
  reference picture.  It can be utilized to further improve resiliency
  in high loss conditions.

o If ISD is used in conjunction with the slice structure, the 
  rectangular slice submode shall be enabled and the dimensions and 
  quantity of the slices present in a frame shall remain the same 
  between each two intra-coded frames (I-frames), as required in H.263+. 
  The individual ISD segments may also be entirely intra coded from time 
  to time to realize quick error recovery without adding the latency 
  time associated with sending complete INTRA-pictures.

o When the slice structure is not applied, the insertion of a 
  (preferably byte-aligned) GOB header can be used to provide resync 
  boundaries in the bitstream, as the presence of a GOB header 
  eliminates the dependency of motion vector prediction across GOB 
  boundaries.  These resync boundaries provide natural locations for 
  packet payload boundaries.

o H.263+ allows picture headers to be sent in an abbreviated form in 
  order to prevent repetition of overhead information that does not 
  change from picture to picture.  For resiliency, sending a complete 
  picture header for every frame is often advisable.  This means, that 
  especially in cases with high packet loss probability in which picture 
  header contents are not expected to be highly predictable, the sender 
  may always set the subfield UFEP in PLUSPTYPE to '001' in the H.263+ 
  video bitstream.

o In a multi-layer scenario, each layer may be transmitted to a 
  different network address.  The configuration of each layer such as 
  the enhancement layer number (ELNUM), reference layer number (RLNUM), 
  and scalability type should be determined at the start of the session 
  and should not change during the course of the session.

o All start codes can be byte aligned, and picture, slice, and EOSBS 
  start codes are always byte aligned.  The boundaries of these
  syntactical elements provide ideal locations for placing packet 
  boundaries.

o We assume that a maximum Picture Header size of 504 bits is
  sufficient.  The syntax of H.263+ does not explicitly prohibit larger 
  picture header sizes, but the use of such extremely large picture 
  headers is not expected.


4. H.263+ Payload Header

For H.263+ video streams, each RTP packet carries only one H.263+ video 
packet.  The H.263+ payload header is always present for each H.263+ 
video packet.  The payload header is of variable length.  A 16 bit field 
of the basic payload header may be followed by an 8 bit field for Video 
Redundancy Coding information, and/or by a variable length extra picture 
header as indicated by PLEN. These optional fields appear in the order 
given above when present.

If an extra picture header is included in the payload header, the length 
of the picture header in number of bytes is specified by PLEN.  The 
minimum length of the payload header is 16 bits, corresponding to PLEN 
equal to 0 and no VRC information present.

The remainder of this section defines the various components of the RTP 
payload header.  Section five defines the various packet types that are 
used to carry different types of H.263+ coded data, and section six 
summarizes how to distinguish between the various packet types.


4.1 General H.263+ payload header

The H.263+ payload header is structured as follows:

   0                   1
   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  |   RR    |P|V|   PLEN    |PEBIT|
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

RR: 5 bits
  Reserved bits.  Shall be zero.

P: 1 bit
  Indicates the picture start or a picture segment (GOB/Slice) start or 
  a video sequence end (EOS or EOSBS).  Two bytes of zero bits then have 
  to be prefixed to the payload of such a packet to compose a complete
  picture/GOB/slice/EOS/EOSBS start code.  This bit allows the omission 
  of the two first bytes of the start codes, thus improving the 
  compression ratio.

V: 1 bit
  Indicates the presence of an 8 bit field containing information for 
  Video Redundancy Coding (VRC), which follows immediately after the 
  initial 16 bits of the payload header if present.  For syntax and 
  semantics of that 8 bit VRC field see section 4.2.

PLEN: 6 bits
  Length in bytes of the extra picture header.  If no extra picture 
  header is attached, PLEN is 0.  If PLEN>0, the extra picture header is 
  attached immediately following the rest of the payload header.  Note 
  the length reflects the omission of the first two bytes of the picture 
  start code (PSC).  See section 5.1.

PEBIT: 3 bits
  Indicates the number of bits that shall be ignored in the last byte of 
  the picture header.  If PLEN is zero, then PEBIT shall also be zero.


4.2 Video Redundancy Coding Header Extension

Video Redundancy Coding (VRC) is an optional mechanism intended to 
improve error resilience over packet networks.  Implementing VRC in 
H.263+ will require the Reference Picture Selection option described in 
Annex N.  By having multiple "threads" of independently inter-frame 
predicted pictures, damage of individual frame will cause distortions 
only within its own thread but leave the other threads unaffected.  From 
time to time, all threads converge to a so-called sync frame (an INTRA 
picture or a non-INTRA picture which is redundantly represented within 
multiple threads); from this sync frame, the independent threads are 
started again.  For more information on codec support for VRC see [7].

P-picture temporal scalability is another use of the reference picture 
selection mode and can be considered a special case of VRC in which only 
one copy of each sync frame may be sent.  It offers a thread-based 
method of temporal scalability without the increased delay caused by the 
use of B pictures.  In this use, sync frames sent in the first thread of 
pictures are also used for the prediction of a second thread of pictures 
which fall temporally between the sync frames to increase the resulting 
frame rate.  In this use, the pictures in the second thread can be 
discarded in order to obtain a reduction of bit rate or decoding 
complexity without harming the ability to decode later pictures.  A 
third or more threads can also be added as well, but each thread is 
predicted only from the sync frames (which are sent at least in thread 
0) or from frames within the same thread.

While a VRC data stream is - like all H.263+ data - totally self-
contained, it may be useful for the transport hierarchy implementation 
to have knowledge about the current damage status of each thread.  On 
the Internet, this status can easily be determined by observing the 
marker bit, the sequence number of the RTP header, and the thread-id and 
a circling "packet per thread" number.  The latter two numbers are coded 
in the VRC header extension.

The format of the VRC header extension is as follows:

   0 1 2 3 4 5 6 7
  +-+-+-+-+-+-+-+-+
  | TID | Trun  |S|
  +-+-+-+-+-+-+-+-+

TID: 3 bits
  Thread ID.  Up to 7 threads are allowed. Each frame of H.263+ VRC data 
  will use as reference information only sync frames or frames within 
  the same thread.  By convention, thread 0 is expected to be the 
  "canonical" thread, which is the thread from which the sync frame 
  should ideally be used.  In the case of corruption or loss of the 
  thread 0 representation, a representation of the sync frame with a 
  higher thread number can be used by the decoder.  Lower thread numbers 
  are expected to contain equal or better representations of the sync 
  frames than higher thread numbers in the absence of data corruption or 
  loss.  See [7] for a detailed discussion of VRC.

Trun: 4 bits
  Monotonically increasing (modulo 16) 4 bit number counting the packet
  number within each thread.

S: 1 bit
  A bit that indicates that the packet content is for a sync frame.  An
  encoder using VRC may send several representations of the same "sync"
  picture, in order to ensure that regardless of which thread of 
  pictures is corrupted by errors or packet losses, the reception of at 
  least one representation of a particular picture is ensured (within at 
  least one thread).  The sync picture can then be used for the 
  prediction of any thread.  If packet losses have not occurred, then 
  the sync frame contents of thread 0 can be used and those of other 
  threads can be discarded (and similarly for other threads).  Thread 0 
  is considered the "canonical" thread, the use of which is preferable 
  to all others.  The contents of packets having lower thread numbers 
  shall be considered as generally preferred over those with higher 
  thread numbers.


5. Packetization schemes

5.1 Picture Segment Packets and Sequence Ending Packets (P=1)

A picture segment packet is defined as a packet that starts at the 
location of a Picture, GOB, or slice start code in the H.263+ data 
stream.  This corresponds to the definition of the start of a video 
picture segment as defined in H.263+.  For such packets, P=1 always.

An extra picture header can sometimes be attached in the payload header 
of such packets.  Whenever an extra picture header is attached as 
signified by PLEN>0, only the last six bits of its picture start code, 
'100000', are included in the payload header.  A complete H.263+ picture 
header with byte aligned picture start code can be conveniently 
assembled on the receiving end by prepending the sixteen leading '0' 
bits.

When PLEN>0, the end bit position corresponding to the last byte of the 
picture header data is indicated by PEBIT.  The actual bitstream data 
shall begin on an 8-bit byte boundary following the payload header.

A sequence ending packet is defined as a packet that starts at the 
location of an EOS or EOSBS code in the H.263+ data stream.  This 
delineates the end of a sequence of H.263+ video data (more H.263+ video 
data may still follow later, however, as specified in ITU-T 
Recommendation H.263).  For such packets, P=1 and PLEN=0 always.

The optional header extension for VRC may or may not be present as 
indicated by the V bit flag.


5.1.1 Packets that begin with a Picture Start Code

Any packet that contains the whole or the start of a coded picture shall 
start at the location of the picture start code (PSC), and should 
normally be encapsulated with no extra copy of the picture header. In 
other words, normally PLEN=0 in such a case.   However, if the coded 
picture contains an incomplete picture header (UFEP = "000"), then a 
representation of the complete (UFEP = "001") picture header may be 
attached during packetization in order to provide greater error 
resilience.  Thus, for packets that start at the location of a picture 
start code, PLEN shall be zero unless both of the following conditions 
apply:
1) The picture header in the H.263+ bitstream payload is incomplete
   (PLUSPTYPE present and UFEP="000"), and
2) The additional picture header which is attached is not incomplete
   (UFEP="001").

A packet which begins at the location of a Picture, GOB, slice, EOS, or 
EOSBS start code shall omit the first two (all zero) bytes from the 
H.263+ bitstream, and signify their presence by setting P=1 in the 
payload header.

Here is an example of encapsulating the first packet in a frame (without 
an attached redundant complete picture header):

   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
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  |   RR    |1|V|0|0|0|0|0|0|0|0|0| bitstream data without the    |
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  | first two 0 bytes of the PSC                                ...
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


5.1.2 Packets that begin with GBSC or SSC

For a packet that begins at the location of a GOB or slice start code, 
PLEN may be zero or may be nonzero, depending on whether a redundant 
picture header is attached to the packet.  In environments with very low 
packet loss rates, or when picture header contents are very seldom 
likely to change (except as can be detected from the GFID syntax of 
H.263+), a redundant copy of the picture header is not required.  
However, in less ideal circumstances a redundant picture header should 
be attached for enhanced error resilience, and its presence is indicated 
by PLEN>0.

Assuming a PLEN of 9 and P=1, below is an example of a packet that 
begins with a byte aligned GBSC or a SSC:

   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
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  |   RR    |1|V|0 0 1 0 0 1|PEBIT|1 0 0 0 0 0| picture header    | 
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  | starting with TR, PTYPE ...                                   |
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  | ...                                           | bitstream     |
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  | data starting with GBSC/SSC without its first two 0 bytes   ...
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Notice that only the last six bits of the picture start code, '100000', 
are included in the payload header.  A complete H.263+ picture header 
with byte aligned picture start code can be conveniently assembled if 
needed on the receiving end by prepending the sixteen leading '0' bits.


5.1.3 Packets that Begin with an EOS or EOSBS Code

For a packet that begins with an EOS or EOSBS code, PLEN shall be zero, 
and no Picture, GOB, or Slice start codes shall be included within the 
same packet.  As with other packets beginning with start codes, the two 
all-zero bytes that begin the EOS or EOSBS code at the beginning of the 
packet shall be omitted, and their presence shall be indicated by 
setting the P bit to 1 in the payload header.

System designers should be aware that some decoders may interpret the 
loss of a packet containing only EOS or EOSBS information as the loss of 
essential video data and may thus respond by not displaying some 
subsequent video information.  Since EOS and EOSBS codes do not actually 
affect the decoding of video pictures, they are somewhat unnecessary to 
send at all.  Because of the danger of misinterpretation of the loss of 
such a packet, encoders are generally to be discouraged from sending EOS 
and EOSBS.

Below is an example of a packet containing an EOS code:

   0                   1                   2
   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  |   RR    |1|V|0|0|0|0|0|0|0|0|0|1|1|1|1|1|1|0|0|
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


5.2 Encapsulating Follow-On Packet (P=0)

A Follow-on packet contains a number of bytes of coded H.263+ data which 
does not start at a synchronization point.  That is, a Follow-On packet 
does not start with a Picture, GOB, Slice, EOS, or EOSBS header, and it 
may or may not start at a macroblock boundary.  Since Follow-on packets 
do not start at synchronization points, the data at the beginning of a 
follow-on packet is not independently decodable.  For such packets, P=0 
always.  If the preceding packet of a Follow-on packet got lost, the 
receiver may discard that Follow-on packet as well as all other 
following Follow-on packets.  Better behavior, of course, would be for 
the receiver to scan the interior of the packet payload content to 
determine whether any start codes are found in the interior of the 
packet which can be used as resync points.  The use of an attached copy 
of a picture header for a follow-on packet is useful only if the 
interior of the packet or some subsequent follow-on packet contains a 
resync code such as a GOB or slice start code.  PLEN>0 is allowed, since 
it may allow resync in the interior of the packet.  The decoder may also 
be resynchronized at the next segment or picture packet.

Here is an example of a follow-on packet (with PLEN=0):

   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
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  |   RR    |0|V|0|0|0|0|0|0|0|0|0| bitstream data              ...
  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


6. Use of this payload specification

There is no syntactical difference between a picture segment packet and 
a Follow-on packet, other than the indication P=1 for picture segment or 
sequence ending packets and P=0 for Follow-on packets.  See the 
following for a summary of the entire packet types and ways to 
distinguish between them.

It is possible to distinguish between the different packet types by 
checking the P bit and the first 6 bits of the payload along with the 
header information.  The following table shows the packet type for 
permutations of this information (see also the picture/GOB/Slice header 
descriptions in H.263+ for details):

--------------+--------------+----------------------+-------------------
 First 6 bits | P-Bit | PLEN |  Packet              |  Remarks
 of Payload   |(payload hdr.)|                      |
--------------+--------------+----------------------+-------------------
 100000       |   1   |  0   |  Picture             |  Typical Picture
 100000       |   1   | > 0  |  Picture             |  Note UFEP
 1xxxxx       |   1   |  0   |  GOB/Slice/EOS/EOSBS |  See possible GNs
 1xxxxx       |   1   | > 0  |  GOB/Slice           |  See possible GNs
 Xxxxxx       |   0   |  0   |  Follow-on           |
 Xxxxxx       |   0   | > 0  |  Follow-on           |  Interior Resync
--------------+--------------+----------------------+-------------------

See [4] for details regarding the possible values of the six bits (a "1" 
bit followed by a five bit GN field explicit or emulated) of GOB, Slice, 
EOS, and EOSBS codes.

As defined in this specification, every start of a coded frame (as 
indicated by the presence of a PSC) has to be encapsulated as a picture 
segment packet.  If the whole coded picture fits into one packet of 
reasonable size (which is dependent on the connection characteristics), 
this is the only type of packet that needs to be observed.  Due to the 
high compression ratio achieved by H.263+ it is often possible to use 
this mechanism, especially for small spatial picture formats such as 
QCIF and typical Internet packet sizes around 1500 bytes.

If the complete coded frame does not fit into a single packet, two 
different ways for the packetization may be chosen.  In case of very low 
or zero packet loss probability, one or more Follow-on packets may be 
used for coding the rest of the picture.  Doing so leads to minimal 
coding and packetization overhead as well as to an optimal use of the 
maximal packet size, but does not provide any added error resilience.

The alternative is to break the picture into reasonably small partitions 
- called Segments - (by using the Slice or GOB mechanism), that do offer 
synchronization points.  By doing so and using the Picture Segment 
payload with PLEN>0, decoding of the transmitted packets is possible 
even in such cases in which the Picture packet containing the picture 
header was lost (provided any necessary reference picture is available). 
Picture Segment packets can also be used in conjunction with Follow-on 
packets for large segment sizes.


7. Security Considerations

RTP packets using the payload format defined in this specification are 
subject to the security considerations discussed in the RTP 
specification [1], and any appropriate RTP profile (for example [2]).
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 after compression so 
there is no conflict between the two operations.

A potential denial-of-service threat exists for data encodings using 
compression techniques that have non-uniform receiver-end computational 
load.  The attacker can inject pathological datagrams into the stream 
which are complex to decode and cause the receiver to be overloaded.
However, this encoding does not exhibit any significant non-uniformity.

As with any IP-based protocol, in some circumstances a receiver may be 
overloaded simply by the receipt of too many packets, either desired or 
undesired.  Network-layer authentication may be used to discard packets 
>from undesired sources, but the processing cost of the authentication 
itself may be too high.  In a multicast environment, pruning of specific 
sources may be implemented in future versions of IGMP [5] and in 
multicast routing protocols to allow a receiver to select which sources 
are allowed to reach it.

A security review of this payload format found no additional 
considerations beyond those in the RTP specification.


8. References

[1] H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson, "RTP : A
    Transport Protocol for Real-Time Applications," RFC 1889, January
    1996.

[2] H. Schulzrinne, "RTP Profile for Audio and Video Conference with 
    Minimal Control," RFC 1890, January 1996.

[3] "Video Coding for Low Bit Rate Communication," ITU-T Recommendation 
    H.263, March 1996.

[4] "Video Coding for Low Bit Rate Communication," ITU-T Recommendation 
    H.263, January 1998.

[5] T. Turletti, C. Huitema, "RTP Payload Format for H.261 Video
    Streams," RFC 2032, October 1996.

[6] C. Zhu, "RTP Payload Format for H.263 Video Streams," RFC 2190,
    September 1997.

[7] S. Wenger, "Video Redundancy Coding in H.263+," Proc. Audio-Visual
    Services over Packet Networks, Aberdeen, U.K., September 1997.




From rem-conf Fri May 08 17:57:24 1998 
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To: mbone@isi.edu, rem-conf@es.net
From: Steve Deering <deering@cisco.com>
Subject: MBone apps for Windows?
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I could have sworn that I saw a message within the last week, *probably*
sent to either the mbone or rem-conf mailing list, explaining where the
most functional versions of vat/rat/vic/sdr/(wb?) for Windows can be found,
but I can't seem to find that message in any of my folders.  If someone
knows the message I am talking about, would you please forward it to me
(privately, assuming everyone else who might have been interested already
saw it, and I'm the only idiot to lose it).

(No, I haven't given up my Mac.  I'm seeking the info for a colleague. :-)

Steve





From rem-conf Fri May 08 20:46:45 1998 
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Date: Fri, 08 May 1998 19:28:12
To: Steve Deering <deering@cisco.com>
From: Ross Finlayson <finlayson@lvn.com>
Subject: Re: MBone apps for Windows?
Cc: mbone@isi.edu, rem-conf@es.net
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At 05:44 PM 5/8/98 -0700, Steve Deering wrote:
>I could have sworn that I saw a message within the last week, *probably*
>sent to either the mbone or rem-conf mailing list, explaining where the
>most functional versions of vat/rat/vic/sdr/(wb?) for Windows can be found,
>but I can't seem to find that message in any of my folders.

(I'm not 100% sure that I saw such a message, and this seems to be a
frequently-asked question anyway, so I'm cc'ing to the "mbone" & "rem-conf"
lists.)

The most up-to-date versions of these tools appear to be the recent
releases from University College, London.  They have these versions listed
in a table at
	<http://scrappy.cs.ucl.ac.uk/shrimp/tools.html>

(BTW, I've also bundled their versions of "vic", "rat", "wbd", and "nte"
into a single self-extracting installer package; you can download this from
	<http://www.lvn.com/multikit/wintools.html>.
This may be more convenient for people in the U.S., during times of
trans-Atlantic network congestion.)

	Ross.





From rem-conf Fri May 08 20:56:00 1998 
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To: mbone@isi.edu, rem-conf@es.net
From: Steve Deering <deering@cisco.com>
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Cc: Ross Finlayson <finlayson@lvn.com>
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A couple people sent me the message I was seeking; it was from Christophe
Orlhac to the mbone list, dated 27 Apr, subject "Hardware for multicasting ?".
It contained the following very useful URL:

	http://www.cs.ucl.ac.uk/staff/K.Hasler/shrimp/bin/

which Ross Finlayson also posted here.

Thanks!

Steve





From rem-conf Sat May 09 03:48:46 1998 
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Date: Sat, 9 May 1998 12:42:20 +0200 (MET DST)
From: =?ISO-8859-1?Q?Tobias_=D6brink?= <nv91-tob@nada.kth.se>
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To: rem-conf@es.net
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Hi all

I'm (so far) not familiar with Tcl/Tk, but have been assigned the
task to write a script that starts vic in Transmit-mode with a
certain Quality and opens a receive-window of a certain format at a
certain geometry.=20
In the man-page it says to use the -u argument to source an
additional script, but I've no idea of what this script should look
like.

/Tobias
.........................................................................=
...
Tobias =D6brink                 Grad. Stud. Dept. of Teleinformatics, KTH
Phone numbers:				Paper mail address:
	+46 8 752 14 75 (IT, work)		KTH-ELECTRUM/204
        +46 8 751 17 93 (Fax)			S-164 40 Kista
						Sweden
E-mail: nv91-tob@it.kth.se		URL: http://www.it.kth.se/~nv91-tob/
See if I'm logged in: http://www.it.kth.se/~nv91-tob/notice.html






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Remove



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 IEEE SYMPOSIUM ON PLANNING AND DESIGN
 OF BROADBAND NETWORKS
 
 Montebello, Quebec, Canada
 October 8-11, 1998
 
 
                      CALL FOR PARTICIPATION
 
 The third IEEE Symposium on Planning and Design of Broadband Networks
 will be held at Le Chateau Montebello, Montebello, Quebec, Canada on
 October 8-11, 1998.  The first two Symposia, held in October 1994 and
 1996, were an outstanding success with representation from North America, 
 Europe and Asia.  The purpose of this symposium is to provide an 
 environment for the discussion and exchange of ideas concerning 
 computer-aided planning and design techniques and tools for broadband 
 networks.  The symposium will include both invited and contributed talks, 
 panel discussions, and demonstrations of broadband network planning and 
 design tools.  This year, the symposium will also include a strong focus 
 on the application of design tools and contributors are encouraged to 
 display their applications at the exhibition.   Abstracts of all 
 presentations will also be distributed at the symposium.
 
 The symposium will address topics in the following areas:
 
 *  Challenges in broadband network planning and design
 
 *  Simulation methodologies and tools for planning and design of
    broadband networks
 
 *  Tool applications and deployment case studies
 
 Please submit by July 15th, 1998, 5 copies of the abstract of proposed
 talk or demos to the Technical Program Chairman:
 
 Professor Hussein Mouftah, 
 Department of Electrical and Computer Engineering,
 Queen's University,
 Kingston, Ontario, Canada K7L 3N6,
 Telephone:  (613) 545-2934, Fax:  (613) 545-6615
 EMAIL:  mouftah@eleceng.ee.queensu.ca
 
 For further information please contact:
 
 Ibrahim Gedeon, Symposium Chair,
 Nortel
 1431 Merivale Road, Nepean,
 P.O. Box 5080, Station F,
 Ottawa, Ontario, Canada K2C 3T1
 Tel: 613 723 4928
 Fax: 613 723 4028
 Email: ibgedeon@nortel.com
 
 Organizing Committee:
 Secretary/Treasurer:		John Hopkins (Nortel Technology)
 Publicity:	        	Kathy Mahoney (Ottawa Centre for Research and 
                                               Innovation)
 IEEE Ottawa Chairperson: 	Hala Tabl
 
 Symposium Secretariat and Local Arrangments:
    Ms. Amy Riordon Jarette
    c/o OCRI, 36 Steacie Drive, Kanata, Ontario, Canada K2K 2A9
    Tel: 613 592 8160 Ext.224 Fax: 613 592 5130 Email: ariordon@ocri.ca
        
 Technical Program Committee:
 Chair  : 	Hussein Mouftah (Queen's University)
 	        Richard Vickers (Nortel Technology)
 	        Thomas Chen (Southern Methodist University)
         	Steven Katz (Lucent Technologies Bell Labs)
 	        Kenichi Sato ( NTT)
 
 
 
 



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	          IEEE International Conference on ATM
                                         ICATM'98
                          June 22-24, 1998 - Colmar, France

	          http://iutsun1.colmar.uha.fr/ICATM98.html




08:00 - 19:00     Conference Registration

                      _____________________________________________________


                                         Monday June 22


09:00 - 09:20     Opening Address

09:20 - 10:30     Plenary Session 1
Chair: G. Pujolle

IP or ATM versus IP over ATM : the Role of the ATM Forum and the IETF in
Setting Standards
S. Ritzenthaler, Newbridge, France

                                    10:30 - 11:00     Coffee Break

11:00 - 12:30     Plenary Session 2
Chair: J.P. Coudreuse

IETF Multiprotocol Label Switching (MPLS) Architecture
=46. Le Faucheur, Cisco, France

                            12:30 - 14:00     Lunch - Restaurant Universitai=
re

14:00 - 15:30     Plenary Session 3
Chair: P. Rolin

ATM for Wireless Networks
B. Bing, Ngee Ann Polytechnic, Singapore

14:00 - 15:30     Plenary Session 4
Chair: P. Lorenz

=46ourth generation ATM switches for public carrier and their role for
telephone service at the transit exchange level

M. Dickmann, Siemens, France

                             15:30 - 16:00     Coffee Break and exhibitions

16:00 - 17:30     Wireless ATM 1
Chair: G. Pujolle

Power Saving Mode of Operation in the WATM MAC Protocol
G. Sfikas, C. Apostolas, R. Tafozolli, University of Surrey, UK

Support for Multimedia Applications in a Wireless ATM System
A. Kassler, A. Lupper, University of Ulm, Germany

LMDS Systems: a possible solution for Wireless ATM Access Networks
B. Conaglia, R. Santaniello, CSELT, Italy ; E. Leonardi, R. Lo Cigno, M.
Meo, F. Neri, D. Saracino, Politecnico di Torino,
Italy

16:00 - 17:30     Video 1
Chair: J.J. Pansiot

Scalable Video on Demand on ABR in ATM Networks
K. Ben Younes, University of Erlangen-Nuremberg, Germany ; K. Begain,
Mu'tah, Jordan

MPEG Video Aggregation over ABR-ATM
J.M. Forn=E9s, J.A. Ternero, F.R. Rubio, University of Sevilla, Spain

Efficient Connection Admission Control for Real Time Video Multiplexing
A.M. Ibrahim, ENST, France

                             17:30 - 18:00     Coffee Break and exhibitions

18:00 - 19:30     Wireless ATM 2
Chair: G. Pujolle

A New Approach for Combined Performance Evaluation of Wireless ATM at
Channel and Higher Protocol Levels
K. Ben Younes, University of Erlangen-Nuremberg, Germany ; W. Franz,
Communication technology, Germany ; B. Girod,
University of Erlangen-Nuremberg, Germany

BAAWAM with Wireless Specific Quality of Service
J. Ben-Othman, K. Boussetta, A. Gueroui, University of Versailles, France

Authentication Protocols for Wireless ATM Networks
D. Patiyoot, S.J. Shepherd, University of Bradford, UK

18:00 - 19:30     Video 2
Chair: J.J. Pansiot

Performance Evaluation of Cell Discarding Mechanisms for Hierarchical VBR
MPEG-2 Video Traffic over ATM
Networks
P. Cuenca, R. Casado, A. Garrido, F. Quiles, University of Castilla-La
Mancha, Spain ; L. Orozco-Barbosa, University of
Ottawa, Canada

Multiple Leaky Buckets with Buffer Sharing for MPEG Video in ATM Networks
Y.C. Ouyang, J.L.Sun, University of Chung Hsing, Taiwan

An Experimental Study for Transmitting MPEG-2 Streams over ATM Networks
=46. Meylan, L.G. Kiatake, M.Z. Santos, S.T. Kofuji, University of Sao Paulo=
,
Brazil ; J.P. Courtiat, LAAS, France

                                        20:00     Reception

                      _____________________________________________________


                                         Tuesday June 23


09:00 - 10:30     Traffic Control
Chair: J. Halpern

=46low Control and Bandwidth Management in Next Generation Internets
C.M. Pazos, M. Gerla, UCLA, USA ; G. Rigolio, Italtel, Italy

Analysis and Simulation on a Recently proposed Flow Control Algorithm for
ATM Network: Phantom
E.R. Ruh, M. Juanatey, F. D'Alvano, University of Simon Bolivar, Venezuela

A Measurement Based Connection Admission Control for ATM Networks
M. Zukerman, T.K. Lee, University of Melbourne, Australia

09:00 - 10:30     Multicast
Chair: J.J. Pansiot

Dynamic Multicast on Loss Networks
W.K. Liao, L.M. Ni, Michigan State University, USA

Batcher Banyan Network with Cell Copy Preparation Stages for Multicast
Switching
S. Takagi, Y. Tanaka, H. Tominaga, Waseda University, Japan

CRAM: Cell Re-labeling at Merge-points for ATM Multicast
J. Crowcroft, University College London, UK ; S. Komandur, D. Mosse,
University of Pittsburgh, USA

                             10:30 - 11:00     Coffee Break and exhibitions

11:00 - 12:30     Real-time traffic
Chair: J.P. Coudreuse

Issues on Connection Allocation for Supporting Distributed Real-Time
Systems on ATM Networks
Z. Mammeri, University of Le Havre, France

A Real-Time Communication Service for ATM-based Distributed Systems
C. Lizzi, J. Montiel, CS Technology, France ; E. Gressier-Soudan, CNAM, Fran=
ce

An Architecture For Flexible High Level Communication Services
B. Maaref, S. Nasri, P. Sicard, IMAG, France

11:00 - 12:30     Interconnection
Chair: P. Rolin

Design and Applications of ATM LAN/WAN Adapters
D. Bonjour, G. De Hauteclocque, J. Le Moal, CNET, France

Network Interworking for Narrowband Services over an ATM Network
H. Park, Y.I. Choi, Y.K. Lee, Electronics and Telecommunications Research
Institute, Korea

On the Design of a High-Performance ATM Bridge
W.T. Chen, Y.W. Deng, C.P. Wang, N.F. Huang, H.C. Lin, C.C. Lu, National
Tsing Hua University, Taiwan ; R.C. Chang,
National Taiwan University of Science and Technology, Taiwan

                            12:30 - 14:00     Lunch - Restaurant Universitai=
re

14:00 - 15:30     Traffic 1
Chair: K. Begain

A Detailed Experimental Performance Evaluation on TCP over UBR
L. Jaussi, EPFL, Switzerland ; M. Lorang, University of Stuttgart, Germany
; J. Nelissen, Alcatel Alsthom, Belgium

Scaling of ABR Parameters using a Parallel Control Scheme in ATM Networks
Q.L. Ding, Centre for Wireless Communications, Singapore ; S.C. Liew,
Chinese University of Hong Kong, Hong Kong

Application of AR Based Model in Proactive Management of VBR Traffic
T.S. Randhawa, Applied Digital Access, Canada ; R.H.S. Hardy, Simon Fraser
University, Canada

14:00 - 15:30 Modelization
Chair: Z. Mammeri

A Top-Down Operations Model for ADSL and ATM Based Access Networks
T.K. Lu, DSC Communications, USA

=46PGA Implementation of a Scalable Shared Buffer ATM Switch
J.W. Shim, G.J. Jeong, , M.K. Lee, Yonsei University, Korea ; S.H. Ahn,
Hyundai Electronics Industries Co., Korea

=46air Queuing for Input-Buffered Switches with Back Pressure
S. Li, N. Ansari, New Jersey Institute of Technologies, USA ; J.G. Chen,
Lucent Technologies, USA

                          15:30 - 16:00     Coffee Break and OASICE
exhibitions

16:00 - 17:30     Traffic 2
Chair:   K. Begain

A Dynamic Call Admission Scheme for VBR Traffic in ATM Networks
S. Ramaswamy, Newbridge Networks, Canada ; and P. Gburzynski, University of
Alberta, Canada

Multi-Layer Simulation Approach for Evaluation of Data service Support in
ATM Networks
L. Guijarro, V. Pla, J.R. Vidal, J. Martinez, Polytechnic University of
Valencia, Spain

On the Applicability and Utility of Gaussian models for Broadband Traffic
R.G. Addie, University of Southern Queensland, Australia

16:00 - 17:30     Quality of Service 1
Chair: S. Ritzenthaler

QoS Control: an Application Integrated Framework
J. Bom, P. Marquest, INESC, Portugal ; M. Correia, P. Pinto, University of
Lisboa, Portugal

A Dynamic Bandwidth Allocation for ATM-based Networks Supporting Multimedia
Applications
A.L. Diniz, J.M. Nogueira, Federal University of Minas Gerais, Brazil ;
C.C. Goulart, Federal University of Vi=E7osa, Brazil

Interaction Approaches for Internet and ATM Quality of Service Architectures
J. Schmitt, L. Wolf, R. Steinmetz, University of Darmstadt, Germany, C.
Siebel, Y.O. Lorcy, Deutsche Telekom, Germany

                          17:30 - 18:00     Coffee Break and OASICE
exhibitions

18:00 - 19:30 Quality of Service 2
Chair: S. Ritzenthaler

Priority Encoding of Video Data over ATM
K.K. Lau, M.H. Lee, K.N. Ngan, University of Western Australia, Australia ;
G. Rogers, CSIRO, Australia

Communication Application Programming Interfaces with Quality of Service
Support
R. Eberhardt, C. Rue=DF, Daimler-Benz, Germany ; R. Rusnak, Artem, Germany

Agents Based Approach for QoS Adaptation in Distributed Multimedia
Applications over ATM Networks
=46. Nait-Abdesselam, N. Agoulmine, University of Versailles-Saint Quentin e=
n
Yveline, France ; K. Kasiolas, University of
Western Ontario, Canada

18:00 - 19:30     Flow Control in ATM : ABR
Chair: R. Muraine

ABR Traffic Control in ATM Networks using Optimal Control Theory
B.K. Kim, C. Thompson, University of Massachusetts, USA

A Simulator for and Result of Comparing ABR Flow Controls for ATM
E.C. Foudriat, K. Maly, M. Hou, Old Dominion University, USA

The ABR Congestion Control by the Least Squares Estimation of Spatial
Distribution of Sources
B.G. Kim, G. Pecelli, University of Mass. Lowell, USA

                                       20:00     Gala Dinner

                      _____________________________________________________


                                        Wednesday June 24


09:00 - 10:30     User applications 1
Chair: M. Potts

Intelligent Building Systems: System Integration using ATM
E. Stipidis, S. Li, E.T. Powner, University of Sussex, UK

NETrix: A procedure for the Calculation of the Network Structure and
Traffic Matrix in ATM Networks
A.E. Garcia, K.D. Hackbarth, University of Cantabria, Spain

The LAN-E Network of the Fraunhofer Institute for Computer Graphics
M. G=E4rtner, B. Tritsch, Fraunhofer Institute, Germany

09:00 - 10:30     Protocols 1
Chair: P. Lorenz

Proxy PNNI Augmented Routing (Proxy PAR)
T. Przygienda, Lucent Technologies, USA ; P. Droz, C. West, IBM Research
Division, Switzerland

Routing in a Hierarchical Structure
P. Van Mieghem, Alcatel Corporate Research, Belgium

A Hybrid Spanning Tree Algorithm for Efficient Topology Distribution in PNNI
E. Basturk, Pluris, USA ; P. Stirpe, Reuters, USA

                             10:30 - 11:00     Coffee Break and exhibitions

11:00 - 12:30     User applications 2
Chair: M. Potts

A Design and Implementation of Home PC for Interactive Multimedia Services
J.H. Lee, Electronics and Telecommunication Research Institute, Korea

IP "Telephony" versus ATM: What is There to discuss ?
S. Wright, R. Onvural, Fujitsu Network Communications, USA

Performance measurements on an ATM-based Metropolitan Area Network : OASICE
Case Study
H. Tobiet, Clemessy, France ; P. Lorenz, University of Haute Alsace, France

11:00 - 12:30     Protocols 2
Chair: J. Halpern

Cells in Frames: ATM over Legacy Networks
M. Shore, J. Veronneau, T. Parker, D. Cogger, Cornell University, USA

Resource Location in Mobile ATM Networks
L. Frel=E9choux, D. Dykeman, I. Iliadis, P. Scotton, IBM Research Laboratory=
,
Switzerland

Native ATM support for CORBA Platforms
A. Puder, M. Moscarda, University of Berkeley, USA

                            12:30 - 14:00     Lunch - Restaurant Universitai=
re

14:00 - 15:30     IP over ATM 1
Chair: D. Bonjour

IPv6 over ATM Flow-Handling
M.V. Loukola, Helsinki University of Technology, Finland

TCP/IP over ATM Challenges in Enterprise Network Integration
S. Yousef, C. Strange, Anglia Polytechnic University, UK

IP Switch over ATM and LAN Emulation (IP-Express)
M. Bavant, M. Delattre, O. Gibergues, Thomson-CSF, France

14:00 - 15:30     Performance 1
Chair: H. Tobiet

A Large-Scale ATM Switch: Analysis, Simulation and Implementation
H.S. Yoon, Samsung Electronics Co., Korea

TCP Performance over ATM on Linux and Windows NT
M. Borriss, U. Dannowski, H. H=E4rtig, Dresden University of Technigy, Germa=
ny

Performance Evaluation of Frame Discard Strategies in ATM Networks
C.M. Wu, Napier University, UK

                             15:30 - 16:00     Coffee Break and exhibitions

16:00 - 18:00     IP over ATM 2
Chair: D. Bonjour

A Way to Accommodate IP Services in ATM Access Networks
J.O. Kim, H.S. Choi, I.K. Kim, J.H. Lee, H.J. Kim, Electronics and
Telecommunications Research Institute, Korea

A Study on Accommodation of TCP/IP Traffic using Window Scale Option to
International ATM Network with VBR
Service Category
S. Ano, T. Hasegawa, T. Kato, K. Narita, K. Hokamura, Kokusai Denshin Denwa
Co. ; Japan

Extension of Classical IP over ATM to Support QoS at the Application Level
S. Martignoni, T. K=FChnel, Ascom Autelca AG, Switzerland

Smart IP Switching: a Hybrid System for Fast IP-based Network Backbones
D. Lloyd, Telenor RDI, Ireland ; D. O'Mahony, Trinity College, Ireland

16:00 - 18:00     Performance 2
Chair: H. Tobiet

Comprehensive In-Service Measurement of QoS
P. Kelley, Net2Net Corporation, USA

The Variable Bandwidth Virtual Path ATM Network and its Self-healing
Performance
K. Wipusitwarakun, H. Tode, H. Ikeda, University of Osaka, Japan

Simulation Study of AAL Type 2
M. Han, A.A. Nilsson, North Carolina State University, USA

Performance Implications of QoS Mapping in Heterogeneous Networks Involving
ATM
P. Francis-Cobley, N. Davies, University of Bristol, UK

                                      18:00     Closing Session





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Subject: CFP: Objects, Components and the Virtual Enterprise
References: <c=US%a=MCI%p=SHL%l=DALFW03-980506020821Z-9862@dalmsdoc01.shl.com>
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First Call for Papers

Objects, Components and the Virtual Enterprise
An Interdisciplinary Workshop at OOPSLA'98,
Vancouver (CA)

http://www.cs.tcd.ie/Virtues/ocve98/ 

Monday 19 October 1998


We are currently witnessing a convergence of
several threads of
technology and business imperatives. The idea of a
virtual
enterprise - a business built from both
organisationally and
geographically distributed units - is becoming an
area of
increasing interest to both computer scientists
and business
people. 

Virtual enterprises are becoming feasible on
account of
technology phenomena including CORBA, the World
Wide Web, Java
and component-based software; they are becoming
attractive
because of business trends such as downsizing and
outsourcing.
Set against these positive factors are a number of
disincentives,
including exposing key business processes to the
hostile Internet,
the extra complexities of cross-border contracts,
and the
fluidity of the software marketplace. 

>From a technical perspective, building a virtual
enterprise
involves confronting problems such as
heterogeneity, distribution,
authentication, privacy, and auditing. From a
business perspective
it involves contracts, cross-organisational
management and statutory
obligations. Within the object oriented arena
solutions exist to
each of these problems, but their integration is
still elusive.
So, given that the ingredients exist, what is the
recipe? And what
is the final dish? 


Aims

The workshop aims to bring together a
complementary group of
practitioners and researchers from a range of
fields with a
common interest in constructing virtual
enterprises. The intention
is to facilitate the exchange of ideas between
those working
on the general problems of dynamic component
integration and
those with specific practical problems to solve or
experiences
to offer. The workshop will thus be of interest to
practising
software engineers, to researchers in workflow,
component-based
software and software methodologies, and to
business people interested
in the new oportunities being facilitated by these
new technologies. 

The specific goals of the workshop are: 

   * to identify those factors - technological,
managerial and
     political - which impede the creation of
virtual enterprises
     through the use object technology;
   * to identify any new business models which
emerge from improving
     business communication and interaction; and 
   * to draw a road map of projected future
developments, to aid
     the development of strategies for deploying
virtual enterprises. 


Workshop format

The workshop is a full day event comprising an
invited "scene-setting"
lecture by a speaker from outside computer
science, a small set of
presentations selected from the submitted position
papers, and targetted
discussion sessions addressing the important
emerging themes. 

Topics of interest for position papers include,
but are not restricted
to: 

   * object methodologies for building the virtual
enterprise; 
   * software frameworks and infrastructures;
   * economic models; 
   * managing business processes in the virtual
enterprise; 
   * applications to large-scale scientific
collaborations; and 
   * experiences in building existing virtual
enterprises. 


Proceedings

All position papers will be made available to
workshop attendees on the
day. A post-workshop proceedings will be published
both electronically
and in book form. It will include a selection of
the best papers, a
summary and commentary on the meeting. 


Attendance

Submission of a position paper is a condition of
attendence. A representative
selection will be chosen by the committee for
presentation. The number of
attendees will be limited to around fifty to
facilitate discussion. 

     Submission of position papers : 14 August
1998 
     Notification of selection for presentation :
31 August 1998 
     Workshop date : 19 October 1998 

A position should be 6 pages maximum (including
diagrams and references).
Papers should be submitted to Simon Dobson
(address below). Electronic
submission is strongly encouraged. 


Organising committee

  Paddy Nixon (Chairman)          Trinity College
Dublin, IE 
  Simon Dobson                    Trinity College
Dublin, IE and
                                  CLRC Rutherford
Appleton Laboratory, UK 
  Ian Gorton                      Transarc
Corporation, AU 
  Spyros Lalis                    ICS-FORTH, GR 
  Oliver Sims                     SSA Object
Technology, UK 
  Vincent Wade                    Trinity College
Dublin, IE 
  David Zenie                     NIIIP, USA 


Contacts

 Further information:                 Paper
submissions: 

 Dr Paddy Nixon                       Dr Simon
Dobson
 Department of Computer Science       Department
of Computer Science
 Trinity College                      Trinity
College
 Dublin 2, Ireland                    Dublin 2,
Ireland

 Tel: (+353)-1-608-2666               Tel:
(+353)-1-608-2224
 Fax:(+353)-1-677-2204               
Fax:(+353)-1-677-2204
 paddy.nixon@cs.tcd.ie               
simon.dobson@cs.tcd.ie
 http://www.cs.tcd.ie/Paddy.Nixon/   
http://www.cs.tcd.ie/Simon.Dobson/



-- 
_____________________________________________________________________
Dr. Paddy Nixon

Department of Computer Science  The Warden
Trinity College                 Trinity Hall
Dublin 2, Ireland               Dartry Road
Tel: (+353)-01-608-2666         Dublin 6, Ireland
Fax: (+353)-01-677-2204        
Tel:(+353)-01-497-1772
Email: Paddy.Nixon@cs.tcd.ie              
WWW: http://www.dsg.cs.tcd.ie/~pnixon/



From rem-conf Mon May 11 06:10:49 1998 
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please remove me from your mailing list.

Arshad Hussain
AT&T
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Tel: 732-949-4441
Fax: 732-949-4632  




From rem-conf Mon May 11 08:31:01 1998 
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From rem-conf Mon May 11 10:54:57 1998 
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From rem-conf Mon May 11 11:26:34 1998 
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From: Stephen Casner <casner@precept.com>
To: Vahe Balabanian <balabani@nortel.ca>
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Subject: Re: AVT Consensus on MPEG4/DMIF
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Responding to Vahe Balabanian's summary of approaches for RTP/MPEG4:

Vahe,

I think you have covered the space of options pretty well.  Your use
of the term "AVT" is a bit fuzzy in some instances, as explained in a
couple of the following comments.

> 1-  MPEG-4 divides the screen into AudioVisualObjects 
> (AVO) each having a desired shape (including geometric 
> and natural shapes) and multiplicity of coded streams 
> to be chosen from by the receiving end-user. 
> 
> Analysis: This approach is useful to AVT for enabling 
> an end-user to adapt its viewing to changing network 
> conditions.

Here I think AVT means "RTP applications".  This network adaptivity is
not a function built into RTP (at least not yet), but is a matter of
the application choosing the appropriate streams.  One might say this
selection is enabled by performance measurement done at the receiver
and perhaps sent back to the source using RTCP.  Here is a case where
sending in separate RTP streams is important (at least for multicast)
so selection of RTP streams can be used to control the incoming data
rate.

> the receiver. The similarities of MPEG-4 and AVT in this 
> area are in the equivalence of GF-RTP to the MPEG-4 stream 
> and the SDP to the Scene and Object Descriptor. Except in 
> the latter case SDP is a queried entity as opposed to a 
> stream.

I think the division of information between static out-of-band
delivery and in-band delivery is an important question for RTP
with respect to generic payload formats, and for MPEG-4 (especially
for multicast) even if RTP were not used at all.

> 1-  The abstract DMIF Application Interface (DAI) isolates 
> MPEG-4  from the underlying delivery  mechanisms. In this 
> DAI makes use of a media-based QoS (expressed in access 
> unit metrics of priority, loss, delay and jitter) and an 
> explicit format description for each stream.
> 
> Analysis: This is useful to AVT in maintaining applications 
> intact while transiting from IPV4 to IPV6 and adopting 
> multiple implementations of the TOS byte to carry Diff Serv 
> information.

RTP is already independent of IPv4 vs IPv6.  It does not address QoS
APIs at all.  So this is not exactly an AVT working group concern.
Applications using RTP may want to use QoS and therefore they would
need to use protocols from other working groups (RSVP, Diff-Serv).
One problem here is the DMIF is primarily an API, not a protocol.
IETF work focusses on protocols and only occasionally considers APIs.

> 2-  DMIF allows tagging of resources used in a given 
> presentation session. The tags are used to log and apply 
> policy to multiple streams on different source and 
> destination addresses and over different networks.
> 
> Analysis: As IETF is moving towards usage accountability 
> with RSVP, this concept allows the collection of usage 
> information for a given presentation. A joint server and 
> client policy can thus be applied to limit the usage of 
> resources for a given presentation.

This is again outside the bounds of AVT.

> 3-  DMIF allows efficient and agile packing of streams 
> into transports and the reuse of existing transports to 
> reduce the visual impact of addition of new higher 
> priority streams at the expense of existing lower 
> priority streams.

DMIF allows an expression of priority with the goal of achieving the
desired visual impact, but it depends on the implementation to
and the protocols actually achieve that.  I think this is where some
of the pushback from RTP may come -- that in order to achive the
desired goal, some changes might be needed above the DAI.

> expertise. DMIF could be used to regulate the stream traffic 
> on Internet in a similar fashion to regulating its TCP traffic.

Again, I think the key part of achieving this goal is in the protocols
and mechanisms below the API, not in the API itself.

> Analysis: The stream formats in MPEG-4 cover a wide range of 
> timestamps and clocks including implicit timestamps. It may 
> be possible to subset these for operation over RTP.  This 
> argues for the creation of MPEG-4 profile for RTP. But this 
> breaks the clean separation adopted in DMIF for isolating the 
> applications from the delivery mechanisms, thus MPEG-4 profiles 
> are normally reserved to applications and not to transports. 
> But one can argue that within the confines of the RTP subset, 
> the DAI principle can still be applied across both the storage 
> and broadcast media. As experience is gained with RTP the subset 
> can be enlarged to encompass more of MPEG-4.

This is where we really need to learn more on both sides.  How
flexible are the boundaries?  Is it important to solve all the
problems and scenarios that MPEG-4 has taken on in the RTP context, or
is it reasonable to say that some we just don't expect to solve some
of those problems or support some of those scenarios across the
Internet.  That is, is the problem too hard just because we
artificially made it so?
							-- Steve




From rem-conf Mon May 11 12:49:13 1998 
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please remove me from the list

john meeks
as715@yfn.ysu.edu
thanks




From rem-conf Mon May 11 13:09:09 1998 
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please remove me from the list.




From rem-conf Mon May 11 16:58:14 1998 
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From: almeroth@cs.ucsb.edu (Kevin C. Almeroth)
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Subject: Linux vs FreeBSD
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Which OS has more/better support for MBone tools... Linux or FreeBSD?  

More importantly, what audio and video cards have drivers under both
Linux/FreeBSD and Windows 95/98?

I'd like to run MBone under both OSs with minimal overhead.


It seems like I visit this question at least once every six months,
and this list sees questions about it every couple of weeks.  I've
scanned the archives and none of the answers are very satisfactory.

-Kevin Almeroth



From rem-conf Mon May 11 18:26:19 1998 
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In-reply-to: Your message of "Mon, 11 May 1998 16:55:28 PDT."
             <199805112355.QAA25437@jackson.cs.ucsb.edu> 
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>Which OS has more/better support for MBone tools... Linux or FreeBSD?  

I would say that there isn't much in it, but probably FreeBSD because
it's used by most of the people I know who are developing Mbone tools,
and the audio driver is very solid now.

As of FreeBSD 2.2.6, Luigi Rizzo's excellent audio driver is included,
which removes many of the problems with audio that existed 6 months
ago.  I'm using it at home with an A/Open AW35Pro soundcard ($40) and
a Hauppauge Wincast/TV capture card ($100), and have had very few
problems.

>More importantly, what audio and video cards have drivers under both
>Linux/FreeBSD and Windows 95/98?

I can't tell you if the above hardware is supported under Linux though.

>It seems like I visit this question at least once every six months,
>and this list sees questions about it every couple of weeks.  I've
>scanned the archives and none of the answers are very satisfactory.

If you're interested in this setup under FreeBSD, I put together a
page for the CAIRN community a little while back.  It isn't really
intended for non-CAIRN sites, but the information is application, and
there are a couple of patched binaries that might be useful:
  http://north.east.isi.edu/mbone/

Cheers,
	Mark



From rem-conf Mon May 11 19:34:52 1998 
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From: Rajesh Krishnan <rkrishna@bbn.com>
Message-Id: <199805120226.WAA02913@rkrishnan.bbn.com>
Subject: Re: Linux vs FreeBSD 
To: rem-conf@es.net
Date: Mon, 11 May 1998 22:26:48 -0400 (EDT)
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> 
> 
> Which OS has more/better support for MBone tools... Linux or FreeBSD?  

I do not wish to start a religious flame war here. Let me start by 
stating that I do not see there is a strong advantage to either for 
just running MBone tools. 

Some maintain that the TCP/IP stack implementation in FreeBSD is 
superior, but I am not an expert there. In general I have noticed that 
FreeBSD and NetBSD are favored more by the networking community, possibly 
due to historic reasons.

IMO, Linux has an edge in terms of documentation and growing popularity. 
The general software turnover is also higher, but not specifically so 
for the MBone.

> 
> More importantly, what audio and video cards have drivers under both
> Linux/FreeBSD and Windows 95/98?

Video drivers are not an issue between Linux/FreeBSD since both these 
OS'es use the XFree86 implementation. Well for obscure cards, I believe 
that 3rd party drivers are more readily available for Linux.

As far as audio goes the OSSLite/USSlite/.. drivers are work both
under FreeBSD and Linux. FreeBSD also has an alternate sound driver
(pcm0) which IMO is easier to configure under FreeBSD. Linux sound
configuration is now quite painless, as far as Mbone conferencing
needs go. However getting the funky features of the latest sound 
cards to work correctly under Linux/FreeBSD may require a little 
more work.

Typically most video and audio cards do ship with a Win95 driver and 
Linux hardware compatibility is well documented. This list can get to 
be long. The Linux Hardware Compatibility HOWTO can be found at
www.linuxresources.com,  www.linuxnow.com, www.redhat.com and other 
sites. Its always a good idea to check the latest compatibility
information before ordering new hardware. The same holds for 
Solaris x86, Windows NT as well as other OSes; I'd probably take 
a chance with Win '95 i.e., if I had to.

> 
> I'd like to run MBone under both OSs with minimal overhead.

I am not sure about minimal overhead. MBone tools work well under 
either. I recently tried many of these tools under Linux and did 
not have much trouble in setting up. I could find many tools in 
binary form for Linux, which is a time saver when configuring a bunch
of machines. One useful site for Mbone+Linux is
http://www.cs.virginia.edu/~mke2e/multicast/

> 
> It seems like I visit this question at least once every six months,
> and this list sees questions about it every couple of weeks.  I've
> scanned the archives and none of the answers are very satisfactory.
> 
> -Kevin Almeroth
> 

Please excuse me if I have misunderstood your question and have
answered it a very naive level.

Regards,
Rajesh 





From rem-conf Mon May 11 19:45:11 1998 
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To: almeroth@cs.ucsb.edu (Kevin C. Almeroth)
cc: rem-conf@es.net
Subject: Re: Linux vs FreeBSD 
In-reply-to: Your message of "Mon, 11 May 1998 16:55:28 PDT."
             <199805112355.QAA25437@jackson.cs.ucsb.edu> 
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From: Amancio Hasty <hasty@rah.star-gate.com>
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For now, FreeBSD is better and this is due to its better ip multicast suport.
Further inquiries with respect to using FreeBSD with mbone tools to 
multimedia@freeBSD.org

	Cheers,
	Amancio





From rem-conf Mon May 11 21:33:39 1998 
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From: jhall@UU.NET (Jeremy Hall)
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Subject: Re: Linux vs FreeBSD
To: hasty@rah.star-gate.com (Amancio Hasty)
Date: Tue, 12 May 1998 00:29:42 -0400 (EDT)
Cc: almeroth@cs.ucsb.edu, rem-conf@es.net
In-Reply-To: <199805120242.TAA07922@rah.star-gate.com> from "Amancio Hasty" at May 11, 98 07:42:32 pm
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I've been using Linux apps for over a year now, an while they aren't
without problems, they definitely work.  I was successful in compiling all
the tools, and they all work, including mrouted.

_J

Amancio Hasty said:
> 
> For now, FreeBSD is better and this is due to its better ip multicast suport.
> Further inquiries with respect to using FreeBSD with mbone tools to 
> multimedia@freeBSD.org
> 
> 	Cheers,
> 	Amancio
> 
> 
> 




From rem-conf Mon May 11 21:41:59 1998 
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To: jhall@UU.NET (Jeremy Hall)
cc: almeroth@cs.ucsb.edu, rem-conf@es.net
Subject: Re: Linux vs FreeBSD 
In-reply-to: Your message of "Tue, 12 May 1998 00:29:42 EDT."
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Please qualify "while they aren't without problems" and justify
postings such as:
"> 2. Linux on a similar Pentium

Not recommended.  I spent much time with mrouted.  Couldn't get it to
prune.  I also am aware that mrouted and apps cannot work on the same
linux box.  Possibly ok for just running the apps, though if you don't
need mrouted." 
or more recently:

"
   Are there any patches for mrouted-3.9 to get it to work with
recent versions of Linux 2.1? I tried writing a quick-fix, but
it looked like it'd take more than that to get it to compile. "

	Thank you,
	Amancio



> I've been using Linux apps for over a year now, an while they aren't
> without problems, they definitely work.  I was successful in compiling all
> the tools, and they all work, including mrouted.
> 






From rem-conf Mon May 11 22:53:53 1998 
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Subject: Re: Linux vs FreeBSD
To: hasty@rah.star-gate.com (Amancio Hasty)
Date: Tue, 12 May 1998 01:50:59 -0400 (EDT)
Cc: almeroth@cs.ucsb.edu, rem-conf@es.net
In-Reply-To: <199805120438.VAA08401@rah.star-gate.com> from "Amancio Hasty" at May 11, 98 09:38:09 pm
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Amancio Hasty said:
> "
>    Are there any patches for mrouted-3.9 to get it to work with
> recent versions of Linux 2.1? I tried writing a quick-fix, but
> it looked like it'd take more than that to get it to compile. "
> 
> 	Thank you,
> 	Amancio
> 
I believe this is a challenge. :-)

_J

> 
> 
> > I've been using Linux apps for over a year now, an while they aren't
> > without problems, they definitely work.  I was successful in compiling all
> > the tools, and they all work, including mrouted.
> > 
> 
> 
> 
> 




From rem-conf Mon May 11 23:49:30 1998 
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To: jhall@UU.NET (Jeremy Hall)
cc: almeroth@cs.ucsb.edu, rem-conf@es.net
Subject: Re: Linux vs FreeBSD 
In-reply-to: Your message of "Tue, 12 May 1998 01:50:59 EDT."
             <QQeozv21318.199805120550@neserve0.uu.net> 
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Date: Mon, 11 May 1998 23:46:40 -0700
From: Amancio Hasty <hasty@rah.star-gate.com>
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Sorry to disappoint you, I just don't subscribe to the linux insect-like
propaganda.

	Have Fun,
	Amancio

> Amancio Hasty said:
> > "
> >    Are there any patches for mrouted-3.9 to get it to work with
> > recent versions of Linux 2.1? I tried writing a quick-fix, but
> > it looked like it'd take more than that to get it to compile. "
> > 
> > 	Thank you,
> > 	Amancio
> > 
> I believe this is a challenge. :-)
> 
> _J
> 
> > 
> > 
> > > I've been using Linux apps for over a year now, an while they aren't
> > > without problems, they definitely work.  I was successful in compiling all
> > > the tools, and they all work, including mrouted.
> > > 
> > 
> > 
> > 
> > 
> 





From rem-conf Tue May 12 05:58:12 1998 
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        end2end-interest@isi.edu, f-troup@codex.cis.upenn.edu,
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From: Stan Moyer <stanm@bellcore.com>
Subject: ENCOM '98 Technical Program
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Enterprise Networking and Computing (ENCOM) '98 Technical Program
-----------------------------------------------------------------


ENCOM-98 will be held in conjunction with ICC '98 and SUPERCOMM
on Sunday, June 07, 1998
in the Westin Peachtree Plaza Hotel, Atlanta, GA, USA,
in "Plaza North/Center" and "Plaza South" rooms.
UP-TO-DATE info on ENCOM -98 is at http://www.comsoc.org/confs/encom/98/
REGISTRATION INFO is at http://icc98.bellsouth-hosting.net/regis.html

8:30 am - 9:00 am: Opeing the ENCOM-98
Bhumip Khasnabish, Gen, Chair, ENCOM-98, GTE Labs. Inc., USA
Vijay Bhagavath, TPC Chair, ENCOM-98, AT&T Labs, USA
Roberto Saracco, EnterNet's Chair, CSELT, Italy

9:00 am - 9:45 am: Keynote Speech
"Networking Goes Home"
by Paul W. Shumate, Executive Director,
Broadband Local Access and Premises Networking, Bellcore

9:45 am - 10:30 am: TWO Parallel Sessions
Panel Discussion Session:
Title: Enterprise Network Security Issues
Organiser:Vijay Bhagavath, AT&T Labs, USA
and/or Scott Marcus, GTE Internetworking, USA
Participants: TBA

Paper Session:
Title: Enterprise Network Applications and Services
Chair: Pradeep Ray, Australia

[A] Service supplier chains for the outsourcing of information processing
services, T. Preuss ( Thomas.Preuss@tu-cottbus.de   ) et al, Germany

[B] Supporting enterprise applications with the eternal system,
L. Moser ( moser@ece.ucsb.edu ) et al, USA

[C] An approach to support cooperation in project-enterprises,
G. Canals, C. Godart ( godart@loria.fr ) et al, France

[D] Bilateral protection firewalls based on dynamic rules,
D, Zenchelsky ( dzenc@geoplex.com ) , USA


10:30 am - 10:45 am: Coffee Break

10:45 am - 12:00 noon: TWO Parallel Sessions

Panel Discussion Session:
Title: Internet Technologies and Services in Enterprise
Organiser: Scott Marcus, GTE Internetworking, USA
and/or Vijay Bhagavath, AT&T Labs, USA
Participants: TBA

Paper Session:
Title: Performance Analysis of Enterprise Networks
Chair: Gagan Choudhury, AT&T Labs., USA

[A] Designing an appropriate traffic shaping scheme to reduce
network resource requirements,
Shan and O. Yang ( yang@trix.genie.uottawa.ca , Canada

[B] A Fuzzy Logic Based Call Admission Controller with Feedback for
ATM Networks, N. Shenoy ( nirmala@spri.levels.unisa.edu.au ) and
S. Halgamugue, Australia

[C] Web-Based Enterprise Network Traffic Monitoring and Reporting,
Sung-Uk Park, Young-Min Kang, James W Hong, and Jong-Tae Park
(  park@ee.kyungpook.ac.kr ), Korea

[D] Parallel data mining on a large scale ATM connected PC cluster
and Performance Analysis of TCP retransmission mechanism,
Oguchi, Shintani, Tamura and Kitsuregawa
( oguchi@tkl.iis.u-tokyo.ac.jp ), Japan


12:00 noon - 1:00 pm: Lunch Break

1:00 pm - 3:30 pm: TWO Parallel Sessions:

Panel Discussion Session:
Title: Management of Broadband Enterprise Networks
Organiser: Dr. Shervin Erfani, Lucent Technologies, USA
Possible participants:
Manu Malek, Bell Labs/Lucnet Technologies, USA
Shawn Shokoohi, BellSouth, USA
Robbie Cohen, Lucnet Technologies, USA
Ram Kare, SIAC, USA

Paper Session:
Title: Enterprise Network Design and Implementation
Chair: Julian Newman, Glasgow Caledonian Univ., UK

[A] Video on Demand over ATM: System Design and Networking
Requirements, Zheng ( zhengbin@flyernet.udayton.edu ) and
M. Atiquzamman, USA

[B] Novel Multicasting Scheme over Integrated Services Packet Networks,
Sang-Ha Kim ( shkim@cclab.chungnam.ac.kr ) and Kwang-Il Lee, Korea

[C] Internetworking Architecture Based on a Cells-in-Frames Mechanism,
T Tiradentes Boaventura and P. R. Guardieiro,
( taciana@ufu.br  OR  guardieiro@lrc.deene.ufu.br ), Brazil

[D] Enterprise-wide Networking and Computing in the Health-Care
Industry: A Case Study, Pronab Ganguly and Pradeep Ray
( p.ray@uws.edu.au ), Australia

[E] Unicast Access to Multicast IP Sessions,
David Shur ( d.shur@att.com ),
AT&T Labs, USA


3:30 pm - 3:45 pm: Coffee Break

3:45 pm - 4:45 pm: TWO Parallel Sessions

Panel Discussion Session:
Title: Enterprise-Wide Internet Service Quality

Organise/Moderator: Stan Moyer, Bellcore, USA
Possible participants:
(1) Kai Tesink, Bellcore -- The Automotive Network Exchange, an Extranet
with a Guaranteed QoS
(2) Patricia Wirth, AT&T
(3) Scott Marcus, GTE Internetworking
(4) Chris Baldwin or Steve Willis, Argon Networks


Paper Session:
Title: Enterprise Network Management and Middleware Issues
Chair: Mohammed Atiquzzaman, Univ of Dayton, Ohio, USA

[A] Middleware for Nomadic Access to Enterprise Information
Systems, Ravi Jain ( rjain@bellcore.com ), Bellcore, USA

[B] Design and Implementation of Web-based Internet/Intranet
Application Service Management System, Eun-Ju Ha et al, Kyung pook
( ejha@ain2.kyungpook.ac.kr OR  jwkhong@postech.ac.kr )
National University, Korea

[C] Anonymous Communication Mechanism and its Integration in
CORBA, Maouche Ahmed ( maouche@msisun1.unilim.fr ), France

[D] An implantation of SNMPv3, O. Cherkaoui
( cherkaoui.omar@uqam.ca ) et al, Canada.


4:45 pm - 5:00 pm: Closing Remarks/Discussion
Moderator(s): IEEE ComSoc's EnterNet's Exec. Committee
Particpants: ENCOM-98 Attendees.



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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 the 1998 Version of 
                          ITU-T Rec. H.263 Video (H.263+)
	Author(s)	: C. Maciocco, J. Ott, C. Bormann, C. Zhu, L. Cline, 
                          D. Newell, G. Sullivan, G. Deisher, S. Wenger, T. Gardos
	Filename	: draft-ietf-avt-rtp-h263-video-02.txt
	Pages		: 10
	Date		: 11-May-98
	
This document specifies an RTP payload header format applicable to the
transmission of video streams generated based on the 1998 version of
ITU-T Recommendation H.263 [4].  Because the 1998 version of H.263 is a
superset of the 1996 syntax, this format can also be used with the 1996
version of H.263 [3], and is recommended for this use by new
implementations.  This format does not replace RFC 2190, which continues
to be used by existing implementations, and may be required for backward
compatibility in new implementations.  Implementations using the new
features of the 1998 version of H.263 shall use the format described in
this document.
 
The 1998 version of ITU-T Recommendation H.263 added numerous coding
options to improve codec performance over the 1996 version.  The 1998
version is referred to as H.263+ in this document.  Among the new
options, the ones with the biggest impact on the RTP payload
specification and the error resilience of the video content are the
slice structured mode, the independent segment decoding mode, the
reference picture selection mode, and the scalability mode.  This
section summarizes the impact of these new coding options on
packetization.  Refer to [4] for more information on coding options.

Internet-Drafts are available by anonymous FTP.  Login with the username
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--NextPart--





From rem-conf Tue May 12 14:57:56 1998 
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Below is the final program for IWQoS '98. Registration is now
closed as we're over booked, but proceedings can be ordered
soon from IEEE. Instructions will appear on the workshop's web 
page http://www-ece.rice.edu/conf/iwqos98

Best regards,
-Ed Knightly (IWQoS '98 co-chair)

_________________________________________________________________________


Sixth IEEE/IFIP International Workshop on Quality of Service (IWQoS '98)
                    -Napa, California  USA -
                        May 18-20, 1998
                
         Sponsored by IEEE Communications Society Technical 
         Committees on Computer Communications, Multimedia 
               Communications, Internet, and Software
 
               Technical co-sponsorship by IFIP WG6.1
                  In co-operation with ACM SIGCOMM
                 Patrons: Hewlett Packard and Nokia




===============
Monday May 18th
===============

 8:00 - 8:50 Continental Breakfast

 8:50 - 9:00 Opening Remarks: IWQoS '98 chairs
      Rich Friedrich, HP Labs
      Ed Knightly, Rice University


 9:00 - 10:00 Keynote Address:  Roch Guerin, IBM Research
      "Quality-of-Service: Too little or too much
      (How much optimization do we really need?)" 

 10:00 - 10:30 Coffee Break

 10:30 - 12:00 Technical Session 1: QoS, Scaling, and Large-Scale Networks
       Session Chair: Jorg Liebeherr, University of Virginia

 Full papers:
      Measuring and Analyzing Service Levels: A Scalable Passive
      Approach
      Manjari Asawa (Hewlett-Packard Laboratories)

      On Flow Aggregation for Multicast Video Transport
      Kentarou Fukuda, Naoki Wakamiya, Masayuki Murata, Hideo Miyahara
      (Osaka University)

 Position papers:

      Reservations for Aggregate Traffic: Experiences from an RSVP
      Tunnels Implementation
      Andreas Terzis, Lixia Zhang (UCLA), Ellen Hahne (Bell Labs)

      Aggregation of Internet Service States
      Steven Berson and Subramaniam Vincent (University of Southern
      California)

      Aggregating Resource Reservations over Multiple Routing Domains
      Olov Schelen and Stephen Pink (Lulea University of Technology)

      A Case for Proportional Fair Sharing
      Zheng Wang (Bell Labs)

 Session Panel Discussion

 12:00  Lunch


 1:30 - 3:00 Technical Session 2: Internet Resource Reservation
      Session Chair: Kang Shin, University of Michigan

 Full papers:
      On-line Measurement of QoS for Call Admission Control
      Matthew Siler and Jean Walrand (University of California at
      Berkeley)

      A General Framework for Deterministic Service Guarantees in
      Telecommunication Networks with Variable Length Packets
      C.S. Chang and Y. Lin (National Tsing Hua University)

 Position papers:
      QoS Enhancement and its Implementation in the vBNS
      Laura Cunningham, Chuck Song, and Rick Wilder (MCI)

      QoS Control via Robust Envelope-Based MBAC
      Jingyu Qiu and Edward W. Knightly (Rice University)

      High Quality and High Utilization - Incompatible Objectives for
      the Internet?
      Kalevi Kilkki, Jussi Ruutu, and Ove Strandberg (Nokia Research
      Center)

      Conservative Gaussian Models Applied to Measurement-Based
      Admission Control
      Francois Brichet and Alain Simonian (France Telecom)

 Session Panel Discussion

 3:00 - 3:30 Afternoon Break

 3:30 - 5:00 Technical Session 3: Economics, Pricing, and QoS
 Session Chair: Cormac Sreenan, AT&T Labs Research

 Full papers:
      Paying for QoS: An Optimal Distributed Algorithm for Pricing
      Network Resources
      Errin Fulp (NEC and North Carolina State University), Max Ott and
      Daniel Reininger (NEC), and Douglas Reeves (North Carolina State
      University)

      INDEX: A Platform for Determining how People Value the Quality of
      their Internet Access
      Bjorn Rupp, Richard Edell, and Pravin Varaiya (University of
      California at Berkeley)

      An Embedded Charging Approach for RSVP
      Martin Karsten, Jens Schmitt, Lars Wolf, and Ralf Steinmetz
      (Darmstadt University of Technology)

 Position paper:

      INDEX Project: User Support for Buying QoS with Regard to User's
      Preferences
      Jorn Altmann and Pravin Varaiya (University of California at
      Berkeley)

 Session Panel Discussion

 6:00 Welcome dinner sponsored by Nokia 

================
Tuesday May 19th
================
 
 8:15 - 9:00 Continental Breakfast

 9:00 - 10:00 Keynote Paper: Domenico Ferrari, Universita Cattolica Piacenza
      "Charging for QoS"

 10:00 - 10:30 Coffee Break

 10:30 - 12:00 Technical Session 4: QoS Architecture, Protocols, and Middleware
      Session Chair: Hermann de Meer, University of Hamburg

 Full papers:
      SRP: a Scalable Resource Reservation Protocol for the Internet
      Werner Almesberger (EPFL ICA), Tiziana Ferrari (University of
      Bologna), and Jean-Yves Le Boudec (EPFL ICA)

      SAAM: Integrated Network Architecture for Integrated Services
      Geoffrey Xie, Debra Hensgen, Taylor Kidd, and John Yarger (Naval
      Postgraduate School)

      Dynamic Authentication for High-Performance Networked Applications
      Phyllis Schneck and Karsten Schwan (Georgia Institute of
      Technology)

 Position papers:

      The Quality of Service (QoS) Binding Model
      Jan de Meer (GMD Fokus), Abdelhakim Hafid (University of Western
      Ontario), and Arno Puder (ICSI)

      Qualis: the QoS Component for the Globus Metacomputing System
      Craig Lee (The Aerospace Corporation), Carl Kesselman, Robert
      Lindell, Soonwook Hwang, Joseph Bannister (University of Southern
      California), Ian Foster, Alain Roy (Argonne National Laboratory)

 Session Panel Discussion


 12:15-1:30 Lunch and Panel 

 Panel: Hui Zhang, Carnegie Mellon University
 "The Future of Differential and Integrated Services in the Internet"
     Fred Baker         Cisco
     Yoram Bernet       Microsoft
     Raj Yavatkar       Intel
     Rick Wilder        MCI
     John Wroclawski    MIT



 2:00-3:15 Technical Session 5: Adaptive QoS 1
 Session Chair: Andrew Campbell, Columbia University

 Full papers:
      A Control Theoretical Model for Quality of Service Adaptations
      Baochun Li and Klara Nahrstedt (University of Illinois at
      Urbana-Champaign)

      Soft Real-Time Application Execution with Dynamic QoS Assurance
      Scott Brandt, Gary Nutt, Toby Berk, and Marty Humphrey (University
      of Colorado at Boulder)

 Position papers:

      A Study of Dynamic QoS Negotiation for Multimedia Applications in
      RSVP
      Antonio Loureiro, Vladimir de L. Santos, Carlos de C. Goulart and
      Jose Marcos S. Nogueira (Federal University of Minas Gerais)

      Robust and Secure Light-weight Resource Reservation for Unicast IP
      Traffic
      Anders Eriksson (Ericsson Telecom) and Christian Gehrmann
      (Ericsson Radio Systems)

      Satisfying QoS with a Learning Based Scheduling Algorithm
      Jason Hall and Philip Mars (University of Durham)

 Session Panel Discussion

 3:15 - 3:45 Afternoon Break

 3:45-5:15 Technical Session 6: Application QoS
      Session Chair: Klara Nahrstedt, University of Illinois at Urbana-Champaign

 Full papers:
      User Services Assistant: An End-to-End Reactive QoS Architecture
      Bjorn Landfeldt, Aruna Seneviratne (UTS), and Christophe Diot
      (Inria)

      Network Support for Application-Oriented QoS
      Prashant Chandra, Allan Fisher, Corey Kosak, and Peter Steenkiste
      (Carnegie Mellon University)

      Development of Opinion-Based Audiovisual Quality Models for
      Desktop Video Teleconferencing
      Coleen Jones and D. Atkinson (U.S. Department of Commerce)

 Position paper:

      Content-based VBR Resource Allocation Model and its Application to
      Dynamic Network Resource Allocation
      Paul Bocheck and Shih-Fu Chang (Columbia University)

 Session Panel Discussion

 6:00 Board Bus for V. Sattui Winery, St. Helena for Dinner and Wine Tasting


==================
Wednesday May 20th
==================

 8:15 - 9:00 Continental Breakfast

 9:00 - 10:30 Technical Session 8: Adaptive QoS 2

 Full papers:
      Achieving High Utilization in Guaranteed Services Networks using
      Early-Deadline-First Scheduling
      Fabio Chiussi and Vijay Sivaraman (Bell Laboratories)

      Exact Emulation of an Output Queueing Switch by a Combined Input
      Output Queueing Switch
      Ion Stoica and Hui Zhang (Carnegie Mellon University)

      On the Speedup Required for Work-Conserving Crossbar Switches
      P. Krishna (DEC), N. Patel (Sycamore), A. Charney (Cabletron and
      MIT), and R. Simcoe (Nexabit)

      Algorithms for Providing Bandwidth and Delay Guarantees in
      Input-Buffered Crossbars with Speed Up
      A. Charny (Cabletron and MIT), P. Krishna (DEC), N. Patel
      (Sycamore), and R. Simcoe (Nexabit)

 Session Panel Discussion

 10:30 - 11:00 Coffee Break

 11:00 - 12:00
 Panel: Vaduvur Bharghavan, University of Illinois 
 "QoS in Wireless Networks: the Transition from Myth to Reality"
    Andrew Campbell   Columbia U.
    Randy Katz        UC Berkeley
    Upamanyu Madhow   UIUC


 12:00 Lunch

Technical Session 8: Adaptive QoS 2

 1:30 - 2:45
 Full papers:
      Decision Support in Cooperative QoS Management
      Stefan Fischer (University of Mannheim) and Hermann De Meer
      (University of Hamburg)

      Supporting Utility Fair Adaptive Services in Wireless Networks
      Giuseppe Bianchi, Andrew Campbell, and Raymond Liao (Columbia
      University)

 Position papers:

      UNITE: A Framework for Flexible QoS in Networks
      Gisli Hjalmtysson and K. K. Ramakrishnan (AT&T Labs)

      A Summary of QoS Support in SWAN
      T. Chen (Bell Labs), P. Krzyanowski (VenGen Inc.), C. Sreenan,
      (AT&T Labs Research), and J. Trotter, (Bell Labs)

      A General Model for QoS Adaptation
      D.G. Waddington and D. Hutchison (Lancaster University)

     Session Panel Discussion

  2:45 Closing Remarks and Announcement of IWQoS '99




From rem-conf Wed May 13 01:25:43 1998 
From rem-conf-request@es.net Wed May 13 01:25:42 1998
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Date: Wed, 13 May 1998 10:09:16 +0200 (MET DST)
From: Christoph Fleck <fleck@rcs.urz.tu-dresden.de>
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Reply-To: Multimedia Referenzzentrum <mmt-ref@tu-dresden.de>
To: Amancio Hasty <hasty@rah.star-gate.com>
cc: Jeremy Hall <jhall@UU.NET>, almeroth@cs.ucsb.edu, rem-conf@es.net
Subject: Re: Linux vs FreeBSD
In-Reply-To: <199805120646.XAA08783@rah.star-gate.com>
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Hi,

I would give the "FreeBSD/Linux & MBone" dispute a profitable direction.
How about to list the function with the MBone-Tools tested at your site?
Importrend could be vat, rat, vic, mrouted on different Hardware.
I would use both, FreeBSD & Linux, to work with MBone.
In the case of FreeBSD it was nearly easy to get an working MBone-Box.
With Linux I had no such sucess. I miss some "cooking recipe" that I found
for FreeBSD. Perhaps anybody has such recipe for Linux to. 

Here is my list:

FreeBSD:
* CS4236B, driver Luigi Rizzo:
  - vat fullduplex, volume-slider work
  - rat halfduplex, volume-slider work not 
* SB16, driver Luigi Rizzo:
  - no useful audio
* CS4236B, driver OSS FreeBSD:
  - no matching vat or rat found
* BT848:
  - vic send PAL@(QCIF,CIF,SCIF) up to 25fps
* mrouted pruning o.k.

Linux:
* CS4236B & SB16, OSS Free & OSS Linux:
- no matching vat or rat found, volume-slider work not
* BT848:
  - vic send PAL@(QCIF,CIF,SCIF) up to 15fps
* mrouted no pruning.

Any extensions of the list are welcome.
Who has a solutions for the described restrictions ?

Rgds,
Christoph

,------------------------------------------------------------------.
| Referenzzentrum fuer multimediale Teledienste (MMRZ), TU Dresden |
|         Dr. Klaus Koehler, Christoph Fleck, Rainer Kranz         |
| e-mail: mmt-ref@tu-dresden.de             Tel.: 0351 / 463 5653  |
|    WWW: http://www-mm.urz.tu-dresden.de               (GERMANY)  |
`------------------------------------------------------------------'




From rem-conf Wed May 13 02:06:28 1998 
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To: Multimedia Referenzzentrum <mmt-ref@tu-dresden.de>
cc: almeroth@cs.ucsb.edu, rem-conf@es.net
Subject: Re: Linux vs FreeBSD 
In-reply-to: Your message of "Wed, 13 May 1998 10:09:16 +0200."
             <Pine.GSO.3.95.980513072350.2277B-100000@rncmm1> 
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Date: Wed, 13 May 1998 01:58:39 -0700
From: Amancio Hasty <hasty@rah.star-gate.com>
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Hi,
We , FreeBSD, have been in the MBone for the last 4 years. In fact,
while working at Cisco during 93 to 94, I had a little FreeBSD box
listening and watching IP Multicast broadcasts at Cisco.

This is the list of MBone apps currently ported to FreeBSD and 
mrouted and friends are part of our base distribution:
{hasty} which mrouted
/usr/sbin/mrouted

ncftp ftp.freebsd.org:/pub/FreeBSD/FreeBSD-current/ports/mbone/


nte/            rtptools/       vat/
imm/             sdr/            vic/
mbone_vcr/      rtpmon/         speak_freely/   wb/


I mostly used vat / linux sound driver with gus pnp and
vic with bt848 or meteor and in the past I have used my
box to trouble-shoot the mbone at large . The latter came
about due to  Jim Lowe's Parking Lot Cam broadcast.
He was using a sun box to broadcast on the FreeBSD
MBone Lounge.

On my PPro 200Mhz , vic-2.8/bt848, I get about anywhere between 15 frames
to 30 frames pending on context. Like right now , I am getting 29 frames with 
a
newswoman standing up and talking .

I have used in the past vat / full duplex / guspnp to talk to people
and I wrote the FreeBSD audio component in vat so the volume , mute etc works.


If it matters , the primary development platform for the UCB mash 
project is FreeBSD.

This is a quote and an leaving out the original poster's name out in case
of a flame war:
"
Luigi,

All the mash tools ought to compile on FreeBSD. FreeBSD is the primary 
development platform for mash, so you should not require any porting
to get it to compile on a FreeBSD box.

Thanks.
"

Very recently Luigi modified Vic so we can use the tuner component
in my bt848 driver.

If you guys are really interested in the latest news on 
FreeBSD with Mbone tools just mail to multimedia@freeBSD.org.


	Best Regards,
	Amancio






From rem-conf Wed May 13 02:22:49 1998 
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Date: Wed, 13 May 1998 11:13:59 +0100
To: mbone@isi.edu, rem-conf@es.net
From: Angel Luis Mateo =?iso-8859-1?Q?Mart=EDnez?= <amateo@dif.um.es>
Subject: ISDN and MBone
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	We are installing a videconferencing room that works over ISDN and over
MBone, so we are interested about configure it for internetworking over
both architectures. Does anybody knows a way to installed a "reflector"
that propagates data from ISDN to the MBone the data receive over ISDN? and
>from Mbone to ISDN ?

	Thanks,

Angel L. Mateo


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

Angel L. Mateo Mart=EDnez
Tfo: 968/364640
Laboratorio Redes. Facultad de Inform=E1tica
Universidad de Murcia
Espa=F1a (Spain)




From rem-conf Wed May 13 04:03:34 1998 
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cc: Amancio Hasty <hasty@rah.star-gate.com>, Jeremy Hall <jhall@UU.NET>, 
    almeroth@cs.ucsb.edu, rem-conf@es.net
Subject: Re: Linux vs FreeBSD
In-reply-to: Your message of "Wed, 13 May 1998 10:09:16 +0200." <Pine.GSO.3.95.980513072350.2277B-100000@rncmm1>
Date: Wed, 13 May 1998 11:57:30 +0100
Message-ID: <590.895057050@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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--> Christoph Fleck writes:
>I would give the "FreeBSD/Linux & MBone" dispute a profitable direction.
>How about to list the function with the MBone-Tools tested at your site?
>Importrend could be vat, rat, vic, mrouted on different Hardware.
>I would use both, FreeBSD & Linux, to work with MBone.
>In the case of FreeBSD it was nearly easy to get an working MBone-Box.
>With Linux I had no such sucess. I miss some "cooking recipe" that I found
>for FreeBSD. Perhaps anybody has such recipe for Linux to. 
>
>Here is my list:
> >FreeBSD:
>* CS4236B, driver Luigi Rizzo:
>  - vat fullduplex, volume-slider work
>  - rat halfduplex, volume-slider work not 
>* SB16, driver Luigi Rizzo:
>  - no useful audio
>* CS4236B, driver OSS FreeBSD:
>  - no matching vat or rat found

For rat development on FreeBSD we use Luigi's sound driver. However, older
versions used the OSS driver, and you should be able to compile this into
the FreeBSD version of rat fairly easily.

>* BT848:
>  - vic send PAL@(QCIF,CIF,SCIF) up to 25fps
>* mrouted pruning o.k.
>
>Linux:
>* CS4236B & SB16, OSS Free & OSS Linux:
>- no matching vat or rat found, volume-slider work not

The development machine for rat on linux has a SoundBlaster AWE32 with
OSS/Linux 3.8.1q as the driver (in SoundBlaster16 mode). The volume slider
works, but there's an annoying click every few seconds (this bug is mostly
fixed in the development version of rat). We only have one linux machine to
test on: if anyone has success/failure reports for different cards, or
patches to fix problems, I'd be happy to receive them.

-- 
Colin Perkins                   Email: c.perkins@cs.ucl.ac.uk
Department of Computer Science  Phone: (+44) 171 419 3666
University College London       WWW  : http://www.cs.ucl.ac.uk/staff/c.perkins/



From rem-conf Thu May 14 09:40:42 1998 
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Date: Thu, 14 May 1998 09:21:33 -0700 (PDT)
From: Hugh LaMaster <lamaster@george.arc.nasa.gov>
To: mbone@ISI.EDU, rem-conf@es.net
Subject: Re: MBone apps for Windows?
In-Reply-To: <v03110703b17981bccd88@[171.69.116.90]>
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On Fri, 8 May 1998, Steve Deering wrote:

> A couple people sent me the message I was seeking; it was from Christophe
> Orlhac to the mbone list, dated 27 Apr, subject "Hardware for multicasting ?".
> It contained the following very useful URL:
> 
> 	http://www.cs.ucl.ac.uk/staff/K.Hasler/shrimp/bin/

I concur. I finally got around to downloading the new tools 
(I didn't have very good luck with the previous versions of 
the  W95/Mbone tools) and ran them on W95.  [They should 
also work on NT - reports?]
  
They seemed to work pretty well, and wb is there now, 
also.  Anyone who gave up on using the initial port 
should try these debugged versions.  

[Of course, I use FreeBSD most of the time when running
on x86's, but, these are handy to have and seem usable.]



The bottom line: W95 Mbone users should try these.



-
 Hugh LaMaster, M/S 233-21,    ASCII Email: hlamaster@mail.arc.nasa.gov
 NASA Ames Research Center     Or:          lamaster@george.arc.nasa.gov
 Moffett Field, CA 94035-1000  No Junkmail: USC 18 section 2701
 Phone: 650/604-1056           Disclaimer:  Unofficial, personal *opinion*.




From rem-conf Thu May 14 10:56:16 1998 
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From: "Alan J. Flavell" <flavell@a5.ph.gla.ac.uk>
Reply-To: A.Flavell@physics.gla.ac.uk
To: Hugh LaMaster <lamaster@george.arc.nasa.gov>
cc: mbone@isi.edu, rem-conf@es.net
Subject: Re: MBone apps for Windows?
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On Thu, 14 May 1998, Hugh LaMaster wrote:

> > 	http://www.cs.ucl.ac.uk/staff/K.Hasler/shrimp/bin/
> 
> I concur.

Me too.  Those UCL folks have done good work (Disclaimer: I'm speaking
as a dumb user who has not looked under the covers!). 

The speedup of vic is truly astonishing.

> (I didn't have very good luck with the previous versions of 
> the  W95/Mbone tools) and ran them on W95.  [They should 
> also work on NT - reports?]

They seem to be working fine on two NT4 systems we've tried.

One system has some trouble with audio hangups, but it has trouble
with audio on NetMeeting too, so it seems to be something with its
sound card or with that card's driver software, rather than anything
amiss with rat.

> The bottom line: W95 Mbone users should try these.

NT too.  There's a working sdr for NT.

Accidentally running two instances of rat under W95 can hang the
audio system, I've found (reported to K.H).  That's not a problem 
on NT4.

all the best




From rem-conf Thu May 14 13:02:09 1998 
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        rem-conf@es.net
From: Eric Kozowski <kozowski@structured.net>
Subject: Re: MBone apps for Windows?
In-Reply-To: <Pine.SOL.3.91.980514090611.18043A-100000@george.arc.nasa.g
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At 09:21 AM 5/14/98 , Hugh LaMaster wrote:
>
>On Fri, 8 May 1998, Steve Deering wrote:
>
>> A couple people sent me the message I was seeking; it was from Christophe
>> Orlhac to the mbone list, dated 27 Apr, subject "Hardware for
multicasting ?".
>> It contained the following very useful URL:
>> 
>> 	http://www.cs.ucl.ac.uk/staff/K.Hasler/shrimp/bin/
>
>I concur. I finally got around to downloading the new tools 
>(I didn't have very good luck with the previous versions of 
>the  W95/Mbone tools) and ran them on W95.  [They should 
>also work on NT - reports?]

seems to work ok on nt 4.0 worstation.  minor cosmetic bugs, but it's
fairly stable.




From rem-conf Thu May 14 21:02:56 1998 
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Date: Fri, 15 May 1998 05:57:38 +0200 (MET DST)
From: Hans Mayer <Hans.Mayer@gmd.de>
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To: mbone@ISI.EDU, rem-conf@es.net
Subject: Re: MBone apps for Windows?
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[...deleted...]
>>I concur. I finally got around to downloading the new tools 
>>(I didn't have very good luck with the previous versions of 
>>the  W95/Mbone tools) and ran them on W95.  [They should 
>>also work on NT - reports?]
>
>seems to work ok on nt 4.0 worstation.  minor cosmetic bugs, but it's
>fairly stable.

What version are the current tools based upon? I had a great time trying
getting version 5.0a (the mash version) running (and they still don't :-( )
As I do have problems with multicast with other applications I suspect it's
not the tools but the stack (from M$).
Does anyone know how (and what) to configure multicast on a W'95 / WNT
machine? I looked at the registry but found nothing. I got a few pointers
to things to try but they didn't help either.

Hans



From rem-conf Thu May 14 23:50:13 1998 
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Remove

---------------------------------------------------------------------------
Dr. Issam A. Hamid
Professor,
Faculty of Software and Information Science,
Iwate Prefectural University
152-52 Sugo, Takizawa aza, Iwate Prefecture$B!"(JJapan
Tel:+81-19-694-2578
Fax:+81-19-694-2579
e-mail:issam@soft.iwate-pu.ac.jp
----------------------------------------------------------------------------





From rem-conf Fri May 15 00:10:21 1998 
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Message-ID: <01BD7FE1.13F319B0.michael.stepping@fernuni-hagen.de>
From: "Stepping, Michael" <michael.stepping@FernUni-Hagen.de>
To: 'Hans Mayer' <Hans.Mayer@gmd.de>
Cc: "mbone@ISI.EDU" <mbone@ISI.EDU>, "rem-conf@es.net" <rem-conf@es.net>, 
    "'KS Grosch, Christian'" <Christian.Grosch@FernUni-Hagen.de>, 
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Subject: AW: MBone apps for Windows?
Date: Fri, 15 May 1998 09:08:50 +0200
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Dear Hans,

with our experience according MBone at the University of Hagen, Germany,
you have to do nothing on your local PC (whether Win95 nor WinNT).

If you cannot receive any Multicast sessions, this is normally a problem
of routers and their software. Perhaps a problem from firewalls.

On WinNT we could only establish "Shrimp"-Version-Tools (others do not=20
function.)

Other hints are: http://www.mbone.de
http://www.lvn.com/multikit/wintools.html
http://www.research.microsoft.com/research/barc/mbone/

For all people who want to test VIVA (music sender)(from =
ASTRA-Satellite)
there is a multicast-test-server in Erlangen (Germany).

Hopefully helps,
Michael Stepping
-------------------------------------------------------------------------=
------
Dipl.-Ing. Michael Stepping an der FernUniversitaet-Hagen
Lehrgebiet fuer Kommunikationssysteme, Herr Prof. Dr. Kaderali
Feithstrasse 142 im TGZ in 58084 Hagen
Tel. (0 23 31) 9 87 - 41 04, Fax: - 3 97
EMail: Michael.Stepping@Fernuni-hagen.de
Homepage: http://www-kommsys.fernuni-hagen.de/~stepping
-------------------------------------------------------------------------=
------
FTK Research Institute for Telecommunications
Image Processing & Multimedia Technology
Martin-Schmeisser-Weg 4, D-44227 Dortmund
Phone: (02 31) 97 50 56 - 0, Fax: - 10




> -----Urspr=FCngliche Nachricht-----
> Von:	Hans Mayer [SMTP:Hans.Mayer@gmd.de]
> Gesendet am:	Freitag, 15. Mai 1998 05:58
> An:	mbone@ISI.EDU; rem-conf@es.net
> Betreff:	Re: MBone apps for Windows?
>
> [...deleted...]
> >>I concur. I finally got around to downloading the new tools
> >>(I didn't have very good luck with the previous versions of
> >>the  W95/Mbone tools) and ran them on W95.  [They should
> >>also work on NT - reports?]
> >
> >seems to work ok on nt 4.0 worstation.  minor cosmetic bugs, but it's
> >fairly stable.
>
> What version are the current tools based upon? I had a great time =
trying
> getting version 5.0a (the mash version) running (and they still don't =
:-( )
> As I do have problems with multicast with other applications I suspect =
it's
> not the tools but the stack (from M$).
> Does anyone know how (and what) to configure multicast on a W'95 / WNT
> machine? I looked at the registry but found nothing. I got a few =
pointers
> to things to try but they didn't help either.
>
> Hans
>
>




From rem-conf Fri May 15 00:27:43 1998 
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From: Andreas Rozek <Andreas.Rozek@RUS.Uni-Stuttgart.De>
To: "'Hugh LaMaster'" <lamaster@george.arc.nasa.gov>,
        "mbone@ISI.EDU"
	 <mbone@ISI.EDU>, "rem-conf@es.net" <rem-conf@es.net>
Subject: AW: MBone apps for Windows?
Date: Fri, 15 May 1998 09:28:23 +0200
Organization: Rechenzentrum Uni Stuttgart
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| > A couple people sent me the message I was seeking; it was from
| > Christophe
| > Orlhac to the mbone list, dated 27 Apr, subject "Hardware for
| > multicasting ?".
| > It contained the following very useful URL:
| > 
| > 	http://www.cs.ucl.ac.uk/staff/K.Hasler/shrimp/bin/
| 
| I concur. I finally got around to downloading the new tools 
| (I didn't have very good luck with the previous versions of 
| the  W95/Mbone tools) and ran them on W95.  [They should 
| also work on NT - reports?]
|   
| They seemed to work pretty well, and wb is there now, 
| also.  Anyone who gave up on using the initial port 
| should try these debugged versions.  
| 
| [Of course, I use FreeBSD most of the time when running
| on x86's, but, these are handy to have and seem usable.]
| 
| The bottom line: W95 Mbone users should try these.

Until yesterday,  I had severe problems  with RAT,  the Robust Audio
Tool from UCL, then Kristian Hasler told me to install the Microsoft
WinSock 2.2 upgrade - since then RAT works perfectly!  I include his
mail here for reference:

| Andreas,
| 
| I  recently had a problem with RAT not working on my friends machine.
| The symptoms where a very slow and unusable user interface and broken
| audio.  I'm not sure if it's the same problem you had but an upgrade
| to
| winsock2.2 (from tucows - http://tucows.cableinet.co.uk/tcp95.html)
| solved the problem.
| 
| Hope that helps!  Kris

What shall I say: it DID help!  Thus, the line below the bottom one
could be: W95 MBone users should upgrade to WinSock 2.2 first!

Andreas Rozek
Rechenzentrum Universitaet Stuttgart       ******   **   **   *****
Kommunikationssysteme/BelWue-Entwicklung   **   **  **   **  **
Allmandring 3a                             **   **  **   **  **
D-70550 Stuttgart                          ******   **   **   *****
                                           ***      **   **       **
Telefon: ++49 (711) 685-4514               ** **    **   **       **
Telefax: ++49 (711) 678-8363               **   **   *****   ****** 
E-Mail:  Andreas.Rozek@RUS.Uni-Stuttgart.De




From rem-conf Fri May 15 06:02:01 1998 
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CALL FOR PAPERS
First International Conference on Telecommunications and Electronic
Commerce 
(ICTEC)
Nashville, USA, November 19-22, 1998

http://cottonian.ogsm.vanderbilt.edu/ICTEC/

Sponsored by
Vanderbilt University Institute for Public Policy Studies (VIPPS)
BELLSOUTH
MAGNETEK
National Commercial Bank, Saudi Arabia
INFORMS Colleges on Information Systems and Telecommunications, IFIP WG
7.3, ATSMA

Important Dates:
Paper (or Extended Abstract) Submission Deadline: May 15, 1998
Panel Proposal Submission Deadline: May 15, 1998
Final Paper Submission Deadline: September 15, 1998
Conference Dates: November 19-22, 1998		

Over the past few years, electronic commerce (EC) has emerged as a
dramatic new mode of business. Today, almost every company is either
using or considering EC. Advances in telecommunications and automated
processes are already forcing dramatic changes in a variety of
industries, ranging from banking and finance to music and entertainment.
Yet, the TEC environment is still in a relatively early state of
evolution. The rapid development of the field has also resulted in two
phenomena. First, many of the significant advances in understanding and
implementing EC are occurring concurrently in academia and industry. The
traditional sequential process of technology innovation followed by
technology transfer no longer applies. Thus, it is increasingly
important for dialogue on this field between academia and industry to be
fostered.  Second, study of EC does not conform to traditional academic
disciplines, and spans a wide range of reference disciplines. Thus, new
forums focusing on EC are necessary to stimulate the necessary
interactions and knowledge sharing. Furthermore, the role of
telecommunications networks is central to EC, and advances in
telecommunications technologies are a fundamental force in shaping EC.
Thus, dialogue between telecommunications researchers and EC researchers
is essential. 

ICTEC is intended to address these needs. The goal of the conference is
to bring together both academic and industrial researchers in the fields
of telecommunications and EC on an ongoing, annual basis, to discuss new
technological developments and their implications for EC, as well as
technological issues that need to be addressed to further the
effectiveness and efficiency of EC. The conference will combine research
tracks in which technical papers will be presented, with industrial
tracks consisting of panels, plenary speakers and exhibits. 

Topics of interest for the conference include (but are not limited to)
the following:

   * Impact of EC on supply chain, distribution chain management, EDI
and IOS
   * Impact of EC on business processes
   * Secure transaction-processing methods for electronic commerce 
   * Design and analysis of multimedia applications on the Internet 
   * The impact of mobility on EC mechanisms 
   * Internet traffic management and analysis 
   * Pricing Internet services 
   * EC market mechanisms and business models 
   * EC intermediary roles and their analysis
   * Electronic payment mechanisms
   * Economic analysis of intranets and extranets

   * Design and analysis of intranets 
   * Data management issues in EC
   * Managing response time and content in Web-based information
services 
   * Effective management of electronic commerce transaction servers 
   * EC developments and challenges in specific industries (e.g.,
financial 	services, music, manufacturing, tourism, publishing)
   * Globalization issues in EC (standards, regulation, property rights,
etc.)

Authors should submit an extended abstract of at least 2000 words, or a
complete paper, to the Program Committee Chair by May 15, 1998. Three
hard-copies should be submitted. Electronic submissions are welcome, but
must be augmented with at least one hard-copy.  Authors of accepted
papers will be asked to submit their complete manuscripts to the Program
Chair, by September 15, 1998. Accepted papers that are presented at the
conference will be included in the Conference Proceedings. Formatting
instructions for accepted papers will be sent with the acceptance
notice. Appropriate papers will be considered for publication in the
Telecommunication Systems Journal, and the Journal of Distributed and
Parallel Databases (in a special issue on Internet Electronic Commerce).

Proposals for panels are also invited. Both academic and industry panels
are encouraged. Each proposal should include a list of panelists with
their affiliations. A brief statement explaining the motivation for the
design and composition of the panel should also be included. The
deadline for proposal submission to the program chair is May 15, 1998.  

Address for Submissions: Professor Amit Basu, Owen Graduate School of
Management, Vanderbilt University, Nashville, TN 37203, USA. 
TEL: +(615) 322-7043; FAX: +(615) 343-7177; EMAIL:
Amit.Basu@owen.vanderbilt.edu

Conference General Chair: Bezalel Gavish, Vanderbilt University,
Nashville, USA. 

Program Committee

   Abdullah Al-Dhelaan, King Saud Univ., Saudi Arabia 
   Kemal Altinkemer, Purdue Univ., USA
   Michael Ball, Univ. of Maryland, USA
   Amit Basu, Vanderbilt Univ., USA (Chair)
   Robert Blanning, Vanderbilt Univ., USA
   Imrich Chlamtac, Univ. of Texas, Dallas, USA 
   H. Michael Chung, California State Univ., USA
   Asuman Dogac, Middle East Tech. Univ., Turkey 
   Lawrence Dowdy, Vanderbilt Univ., USA
   Amitava Dutta, George Mason Univ., USA
   Tony Eyers, Univ. of Woolagong, Australia
   Usama Fayyad, Microsoft Corp., USA
   Stuart Feldman, IBM Corp., USA
   Erol Gelenbe, Duke Univ., USA 
   Sigmund Handelman, IBM Corp., USA
   Eric van Heck, Erasmus Univ., Netherlands
   Ravi Kalakota, Georgia State Univ., USA
   Joakim Kalvenes, Univ. of Texas, Dallas, USA
   Ajit Kambil, New York Univ., USA
   Kalevi Kilkki, Nokia Corp., Finland 
   Norihisa Komoda, Osaka Univ., Japan
   Ramayya Krishnan, Carnegie Mellon Univ., USA
   Akhil Kumar, Univ. of Colorado, USA
   Ronald Lee, Erasmus Univ., Netherlands
   John D. C. Little, MIT, USA
   Alberto Mendelzon, Univ. of Toronto, Canada
   Juzar Motiwalla, National Univ. of Singapore
   Prabir Neogi, Industry Canada
   June Park, Univ. of Iowa, USA
   Charles Perkins, Sun Microsystems, USA

   Hasan Pirkul, Univ. of Texas, Dallas, USA
   Sara Porat, IBM Corp., Israel
   Kotagiri Ramamohanarao, Univ. of Melbourne, Australia
   Louiqa Raschid, Univ. of Maryland, USA
   Arie Segev, UC Berkeley, USA
   Olivia Sheng, Univ. of Arizona, USA
   Andrew Whinston, Univ. of Texas, Austin, USA
   Leon Zhao, Univ. of Science & Tech., Hong Kong		

----------------------------------------------------------------------------
--------------
Bezalel Gavish
Owen Graduate School of Management
Vanderbilt University
Nashville, TN 37220

Tel: Office (615) 322-3659            FAX (615) 343-7177
         Home (615) 370-0813

Email: Gavishb@ctral1.vanderbilt.edu



From rem-conf Fri May 15 06:47:47 1998 
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Date: Fri, 15 May 1998 15:45:50 +0200
From: Werner Geyer <geyer@pi4.informatik.uni-mannheim.de>
Organization: University of Mannheim
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To: rem-conf@es.net
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Hi,

I've written a new sdr plugin for whiteboard media and I'm wondering why
this plugin works fine with solaris 2.6 and Linux sdr version but not
with SGI. We have installed sdr v2.3a1 and the plugin is written
according to the description in the sdr help. Has anybody encountered
the same problem or is this a bug on SGI or is something wrong with my
plugin???

Can anybody help me out with this? The plugin looks like this:

media:whiteboard
proto:udp
tool:dlb
fmt:dlb
flags:-t $(TTL)
flags:$(ADDRESS)/$(PORT)

Thank you very much,

Werner
______________________________________________________________
Werner Geyer, University of Mannheim, Praktische Informatik IV
Tel +49-621-292-1526 ,Fax +49-621-292-5745
EMail: geyer@pi4.informatik.uni-mannheim.de
http://www.informatik.uni-mannheim.de/~geyer/



From rem-conf Mon May 18 12:09:06 1998 
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From: Sanjeev Setia <setia@cs.gmu.edu>
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To: sigmetrics98@cs.gmu.edu
Subject: ACM Sigmetrics 98/Performance 98 Early Registration Deadline (May 22)
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(My apologies if you have already received this message.)

Dear Colleagues:

The ACM Sigmetrics '98/Performance '98 Joint Conference will
be held in Madison, Wisconsin from June 22-26.

The EARLY REGISTRATION DEADLINE for the conference is May 22.

Some highlights of the conference are:

  * strong tutorial program, June 22 (pm) and June 23
  * Workshop on Internet Server Performance, June 23 (Tutorial Track 5)
  * Keynote address by Dr. Susan Owicki:
	"The Technological Basis of Electronic Commerce"
  * strong conference program with a nice mixture of theory and applied
	performance evaluation.

The web site contains the full program as well as registration forms
for the tutorial/workshop and conference:

         http://www.cs.gmu.edu/conf/sigmetrics98

Please register early to avail of the discounted registration fees.


	Sanjeev Setia
	ACM Sigmetrics 98 Publicity Co-chair




From rem-conf Mon May 18 13:40:23 1998 
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Subject: CFP: IC3N'98: 7th Int Conf on Comp Communications & Networks
To: rem-conf@es.net
Date: Mon, 18 May 1998 16:28:33 -0400 (EDT)
Reply-To: atiq@engr.udayton.edu (Mohammed Atiquzzaman)
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****************************************************************************
********* Pls note new submission deadline of May 29, 1998 *****************
****************************************************************************

                        IC3N'98 CALL FOR PAPERS                         
                    SEVENTH INTERNATIONAL CONFERENCE ON 
                   COMPUTER COMMUNICATIONS AND NETWORKS                        

             October 12-15, 1998, Lafayette, Louisiana, USA 

     Sponsored (pending approval) by IEEE Communication Society (TCCC Tech.
     Co-Sponsor), BellSouth, Army Research Office, NASA. In cooperation 
     (pending approval) with The Center for Telecommunications Studies
     and the Center for Advanced Computer Studies at USL, NSF, NIST, ACM. 
     Technical co-sponsored by IEEE COMSOC Technical Committee on Personal 
     Communications (TCPC), Technical Committee on Parallel Processing 
     (pending), IEEE ComSoc Technical Committee on Network Operations
     and Managements (CNOM)   

                     http://www.cacs.usl.edu/~ic3n

The objective of this conference is to provide an effective forum for          
original and fundamental advances in Computer Communications and Networks      
and to foster communication among researchers and practitioners working in     
a wide variety of scientific areas with a common interest in improving         
Computer Communications and Networks.                                          

SCOPE                                                                          
=====                                                                          
The primary focus of the conference is on new and original research results    
in the areas of design, implementation and applications of Computer            
Communications and Networks.  We solicit the submission of papers that         
address novel, challenging and innovative results. Therefore the topics        
that will be addressed include, but are not limited to:                        

   o  Optical Communications Networks      
   o  Wireless/Mobile Communication Networks 
   o  ATM Networking                         
   o  Internet Services/Applications                                          
   o  Wireless Multimedia Applications                                      
   o  Real-time Communications                                                 
   o  Quality of Services (QoS) Issues                                         
   o  LAN/WAN Internetworking                                                 
   o  Network Interoperability & Scalability                
   o  Personal Communication Services                                          
   o  Network Control Management 
   o  Broadband Networks 
   o  Intelligent Networks 
   o  Multicast & Routing Protocols  
   o  Network Security                                                         
   o  Media Access Control/Mobility Algorithms 
   o  Network Reliability  
   o  High Speed Network OAM/Protocols                                         
   o  Video-on-Demand                                                          
   o  Data Traffic Engineering 
   o  Network Management 
   o  Global Infrastructure Network Evolution 
   o  Multimedia Human-Machine Interface                                       
   o  Performance Modeling/Analysis                                            
   o  Communication Software                                                   
   o  Protocol Design/Validation/Testing                                 
   o  Networked Databases 
   o  Network Architectures 


SUBMISSION                                                                     
===========                                                                    
Authors are invited to submit complete and original papers.  Papers that       
may be submitted for consideration include those that have not previously      
been published in another forum, or are not currently being published or       
reviewed by another journal or conference.  All submitted papers will be       
refereed for quality, correctness, originality and relevance. The program      
committee reserves the right to accept a submission as long, short or          
poster presentation. Of particular interest are papers which address          
experiences with concrete Computer Communications and applications.  All       
accepted papers will be published in the conference proceedings. Authors       
will be interested to know that special issues of journals containing          
outstanding papers from the conference are being planned.                      

Manuscripts should include an abstract and be limited to 5000 words.           
Submissions should include the title, author(s), author's affiliation,         
e-mail address, fax number and postal address. In case of multiple authors,    
an indication of which author is responsible for correspondence and            
preparing the camera ready paper for the proceedings should also be            
included. Six copies of the manuscript should be submitted by Friday,          
May 29,1998 (extended) to one of the Program Co-Chairs:                                 

Dr. Kia Makki                          
The Center for Telecommunications Studies 
The University of Southwestern Louisiana 
Lafayette, LA 70504-3890 
e-mail: kia@usl.edu 
Tel: (318) 482-5872                                                            
Fax: (318) 482-5872                                                            

Dr. Imrich Chlamtac 
MS EC33, P.O. Box 830688 
The University of Texas at Dallas 
Richardson, TX 75083 
e-mail: chlamtac@utdallas.edu 
Tel: (972) 883-6295 
Fax: (972) 883-2710                                                            

For more information about the conference (as opposed to paper submissions)    
please send e-mail to ic3n@cacs.usl.edu.                                       

IMPORTANT DATES                                                                
================                                                               
Paper submission deadline:   May 29, 1998                                    
Notification of acceptance:  July 17, 1998                                     
Camera ready papers due:     August 10, 1998                                     

WORKSHOPS, TUTORIALS AND INDUSTRIAL EXHIBITS 
============================================
Proposals must clearly identify the intended audience. Tutorial 
proposal should be considerably broader than the communications and 
network research communities. Exhibits should should illustrate 
innovative technology. Please send your proposal by May 29, 1998 to 
Professor O. Frieder, Workshop and Tutorial Chair, College of Engineering, 
Florida Institute of Technology, Melbourne, FL 32901 (ophir@ee.fit.edu). 


SOCIAL ACTIVITIES
==================
Lafayette is situated within 130 miles of New Orleans and 220 miles 
of Houston. It is serviced by the major Airlines.  It is famous for 
its distinct Cajun culture, food and music. Besides the technical 
program, a very entertaining social program and tours are being planned.

CONFERENCE ORGANIZING COMMITTEE 
===============================
General Chairs: 
B. Mukherjee, UC Davis 
N. Pissinou, USL 

Program Chairs: 
I. Chalmtac, UT Dallas 
K. Makki, USL 

Program Vice-Chairs: 
Y. Fang, UT Dallas 
C. Qiao, SUNY Buffalo 

Program Committee: 
I. F. Akyildiz, Georgia Tech. 
M. H. Ammar, Georgia Tech. 
H. Arabnia, U. of Georgia 
B. Chen, UT Dallas 
J. C.I. Chuang, AT&T Labs-Research 
P. Dowd, DoD
D. Everitt, Univ. of Melbourne, Australia 
A. Farago, UT Dallas 
B. Gavish, Vanderbilt U. 
S. Goyal, GTE Labs   
R. Guerin, IBM T.J.Watson Research Center 
Z. Haas, Cornell Univ. 
D. Kazakos, USL 
L. Kleinrock, UCLA 
R. Krishnan, UT Dallas 
Y. B. Lin, NCTU, Taiwan 
T. Liu, Ascend Communications  
M. T. Liu, Ohio State U. 
F. Lombardi, Texas A&M 
S. Makki, RMIT 
P. McKinley, Michigan State.
R. Melhem, Univ of Pittsburgh
W. M. Moh, San Jose St. 
L. E. Moser, UC S. Barbara 
M. Mutka, Michigan State 
L. Ni, Michigan State 
Y. Pan, U. of Dayton 
D. K. Panda, Ohio State 
W. Peng, SW Texas State 
G. Polyzos, UC San Diego 
S. S. Rappaport, SUNY at Stony Brooks 
I. Rubin, UCLA 
S. Sahni, U. of Florida 
G. Sasaki, U. Hawaii 
M. Schwartz, Columbia U. 
M. Singhal,  Ohio State 
K. Sohraby,  U. of Missouri 
A. K. Somani, Iowa State 
T. Todd,  McMaster 
H.C. Torng, Cornell U. 
E. Torng, Michigan State 
S. Tripathi, UC Riverside 
Y. Yang,  U. of Vermont 
T. Znati, Univ. of Pittsburgh
Z. Zhang, Bell Laboratories 
others (See Web Site)  

Industrial Committee: 
K. Bhat, Lucent Tech. 
C. Fayet, INT, France 
A. Gaivoronski, ITALTEL IT 
A. Kogiantis, Lucent Tech. 
P. O'Reilly,  GTE Lab. 
H. Saito,  NTT, Japan 
A. Vernon, BellSouth 
H. Rudin, IBM, Switzerland 

Government Committee: 
S. Balikirsky, US Army 
P. W. Dowd, U.S. Department of Defense  
D. Fisher, NSF 
M. Halem, NASA 
D. Su,  NIST 
D. Grossman, US Gov. 
T. Pressley, ARO 

International Committee: 
I. Cidon, Technion, Israel 
T. S. Dillon, LaTrobe, AU 
J. Misic, Hong U. ST 
Y. Oie,  KIT, Japan 
Z. Y. Zomaya, UWA, AU 

Publicity Chairs: 
M. Atiquzzaman, U. Dayton 
J. Wigglesworth, IBM CA 

Panel Chairs: 
F. Golshani, Arizona State 
M. S. Obaidat, Monmouth 

Local Arrangm. Chair: 
R. Henry,  USL

Social Activities Chair: 
J. V. Lillie, BellSouth 

Treasurer: 
E. K. Park, U. of Missouri 

Registration Chairs: 
S. Busovaca, Cal. State Sac 
X. Jia, City U. Hong Kong 

Information: 
http://www.cacs.usl.edu/~ic3n 




From rem-conf Tue May 19 14:57:51 1998 
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Subject: BayLISA meeting: IP over ATM (Berry Kercheval)
To: baylisa@baylisa.org, rem-conf@es.net, sage-members@usenix.org
Date: Tue, 19 May 1998 14:24:31 -0700 (PDT)
Cc: blw@baylisa.org
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The BayLISA group meets monthly to discuss topics of interest to systems
and network administrators.  The meetings are free and open to the public.

BayLISA holds monthly meetings on the third Thursday of each month at 7:30
PM PST.

***
*** NOTE: As of the upcoming May meeting, we have a new meeting space!
***

We are now at the "Network Meeting Center at Techmart".  Directions can
be found on our web page at http://www.baylisa.org/locations/techmart.html

The meetings are also broadcast via MBONE.

Schedule
--------
Thursday, 21 May, 1998: IP over ATM, Berry Kercheval

Author Berry Kercheval will be talking about his experiences running IP
over ATM.

This talk will be an overview of ATM technology and how IP traffic can be
carried over it.  Following a brief introduction to ATM, I will discuss
the two main approaches in current use: the IETF's "Classical IP" and
the ATM Forum's "LAN Emulation".  Pros and cons of both schemes will
be addressed.


####


For further information on BayLISA, check out our web site:
http://www.baylisa.org/

To get further information on the meeting location, you can also ftp
it from

        ftp.baylisa.org:/BayLISA/location

or you can query the BayLISA mail server by cutting and pasting the
following line to your shell:

        echo "index baylisa" | mail majordomo@baylisa.org

BayLISA makes video tapes of the meetings available to members.  For more
information on available videos, please send email to:

        video@baylisa.org

For any other information, please send email to:

        info@baylisa.org

If you have any questions, please contact me or the info alias listed
above.

-- 
Greg A. Kulosa          | "The avalanche has already started, it is too
Systems Administrator   |  late for the pebbles to vote." - Ambassador Kosh
GNAC, Inc.              |___________________________________________________
greg@gnac.com           999 Main Street - Redwood City, CA  94063



From rem-conf Wed May 20 07:02:54 1998 
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To: IETF-Announce: ;
Cc: rem-conf@es.net
From: Internet-Drafts@ietf.org
Reply-to: Internet-Drafts@ietf.org
Subject: I-D ACTION:draft-ietf-avt-aggregation-00.txt
Date: Wed, 20 May 1998 09:56:18 -0400
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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		: An RTP Payload Format for User Multiplexing
	Author(s)	: J. Rosenberg, H. Schulzrinne
	Filename	: draft-ietf-avt-aggregation-00.txt
	Pages		: 10
	Date		: 19-May-98
	
This memo describes an RTP payload format for multiplexing
data from multiple users into a single RTP packet. Such
multiplexing is especially useful for transporting voice
data between Internet telephony gateways. It causes significant 
reductions in header overheads and improves scalability.

Internet-Drafts are 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-ietf-avt-aggregation-00.txt".
A URL for the Internet-Draft is:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-avt-aggregation-00.txt

Internet-Drafts directories are located at:

	Africa:	ftp.is.co.za
	
	Europe: ftp.nordu.net
		ftp.nis.garr.it
			
	Pacific Rim: munnari.oz.au
	
	US East Coast: ftp.ietf.org
	
	US West Coast: ftp.isi.edu

Internet-Drafts are also available by mail.

Send a message to:	mailserv@ietf.org.  In the body type:
	"FILE /internet-drafts/draft-ietf-avt-aggregation-00.txt".
	
NOTE:	The mail server at ietf.org can return the document in
	MIME-encoded form by using the "mpack" utility.  To use this
	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
	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
	up into multiple messages), so check your local documentation on
	how to manipulate these messages.
		
		
Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.

--NextPart
Content-Type: Multipart/Alternative; Boundary="OtherAccess"

--OtherAccess
Content-Type: Message/External-body;
	access-type="mail-server";
	server="mailserv@ietf.org"

Content-Type: text/plain
Content-ID:	<19980519104700.I-D@ietf.org>

ENCODING mime
FILE /internet-drafts/draft-ietf-avt-aggregation-00.txt

--OtherAccess
Content-Type: Message/External-body;
	name="draft-ietf-avt-aggregation-00.txt";
	site="ftp.ietf.org";
	access-type="anon-ftp";
	directory="internet-drafts"

Content-Type: text/plain
Content-ID:	<19980519104700.I-D@ietf.org>

--OtherAccess--

--NextPart--





From rem-conf Wed May 20 10:58:39 1998 
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From: frank@newquestcity.com
To: rem-conf@es.net
Subject: Directory Listings
Date: Wed, 20 May 1998 12:48:28 -0500
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I just came across your e-mail address on a Web site. Apparently you have an interest in people finding out about you.  Let me suggest a few directories that you may use for FREE advertising and exposure.  (You may list yourself on these directories even if you don't have a Web page.  And, there is absolutely no fee or strings attached.)  

National Business Directory  http://www.newquestcity.com/dir.htm

National Community Directory  http://www.newquestcity.com/comdir.htm

National Realty Directory http://www.newquestcity.com/natdir.htm

These directories allow you to classify your organization or business in one of over 1,000 categories.  Your listing appears simultaneously in one of the over 3,500 New Quest City community or business directories.  You may post your listing in more than one city or nationally.  (http://www.newquestcity.com) 

To make your posting simply find other listings in your category.  On the results page you will find a link for submitting your listing.  Remember that this is a totally FREE service.  

I am contacting you because I think these directories are something that you will benefit from.  If for some reason I have contacted you in error, please accept my sincere apology.  Unless you respond, I will make the utmost effort to ensure that I never contacted you again.  Nevertheless, I do hope that you take the time to visit New Quest City and review our programs and features.



Thanks.

Sincerely,


Frank
Frank@newquestcity.com








From rem-conf Wed May 20 13:47:29 1998 
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From: Paul Reader <paul@wisys.com>
To: "rem-conf@es.net" <rem-conf@es.net>
Subject: remove
Date: Wed, 20 May 1998 16:40:44 -0400
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From rem-conf Wed May 20 13:58:10 1998 
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remove



From rem-conf Wed May 20 14:23:00 1998 
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From: "Andrew G. Malis" <amalis@casc.com>
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To: jdrosen@bell-labs.com, schulzrinne@cs.columbia.edu
cc: "Andrew G. Malis" <malis@casc.com>, rem-conf@es.net
Subject: draft-ietf-avt-aggregation-00.txt
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Jonathan and Henning,

I've just joined the rem-conf list, so please forgive me if I bring up
any issues that have already been discussed.

I just read your draft, and I have a few questions as a result.

It requires that all of the multiplexed streams share a common base
sampling rate, and it can add a variable amount of delay as incoming
samples to the mux are buffered to wait for other samples that will be
combined in a single RTP packet.  I was curious if the practicality of
this requirement and the buffering delay have been discussed in the
group at all. 

I was also curious if possible interactions with
draft-ietf-avt-crtp-04.txt have been considered, since the compression
it provides is a huge win. 

Thanks,
Andy
________________________________________________________________________
Andrew G. Malis  malis@ascend.com  phone:978 952-7414   fax:978 392-1484
Ascend Communications, Inc.        1 Robbins Road     Westford, MA 01886




From rem-conf Wed May 20 17:46:39 1998 
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Andrew G. Malis wrote:
> 
> Jonathan and Henning,
> 
> I've just joined the rem-conf list, so please forgive me if I bring up
> any issues that have already been discussed.

Being that the draft only appeared this morning in the archives, this
discussion is the first :)


> It requires that all of the multiplexed streams share a common base
> sampling rate, and it can add a variable amount of delay as incoming
> samples to the mux are buffered to wait for other samples that will be
> combined in a single RTP packet.  I was curious if the practicality of
> this requirement and the buffering delay have been discussed in the
> group at all.

By sampling rate, I assume you mean frame size (10ms for G.729, for
example), and not A->D sampling rate (8 khz, for example). The draft
does not require all users in a packet to have equal frame sizes. The
draft proposes that users be clustered into groups, and each group has
its media frames encapsulated together in an RTP packet. The only
requirement of users in a group is that the timestamp of their media
frames be the same in the given packet. Lets say, for example, two users
(1 and 2) are multiplexed. User 1 has a frame size of 20ms, and 2 has a
frame size of 10. You can operate with two user groups (A and B), both
with packets sent every 20ms, but out of phase by 10ms. User 1 places a
frame in each packet  of group A, while user 2 places frames in each
packet in both groups A and B. Note that there is no additional delay
introduced here, and that the users are muxed together 50% of the time.

Essentially, the requirement for efficient multiplexing is that the
frame sizes of the codecs have a greatest common divisor that is as
large as possible.  It was my assumption in writing the draft that most
systems would operate with codecs that run at some multiple of a base
size (10ms, say). If this is not the case, I would appreciate input to
that regard. Our earlier draft on multiplexing allowed for mixing of
completely disparate frame sizes in the same packet, but at the cost of
a larger per-user header.

The amount of additional delay introduced by the scheme depends entirely
on how it is used. In the above example, there was no additional delay.
If, instead, the system were run with a single user group, with packets
sent every 20ms, user 2 would suffer an additional 10ms packetization
delay. It is therefore at the discretion of the implementor to determine
the acceptable level of delay/efficiency tradeoff.


> I was also curious if possible interactions with
> draft-ietf-avt-crtp-04.txt have been considered, since the compression
> it provides is a huge win.

Its use of SSRC-based user groups means it should work quite well with
RTP header compression. Each user group will be considered a separate
compression context. As the SN and TS within each group should generally
behave "well", I don't see any problems.


-Jonathan R.
-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
FAX:   (732) 834-5379                       Rm. 4C-526
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen



From rem-conf Thu May 21 00:48:13 1998 
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Date: Wed, 20 May 1998 18:05:00
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From rem-conf Thu May 21 05:30:53 1998 
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From: "Andrew G. Malis" <amalis@casc.com>
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To: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
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Jonathan,

> > It requires that all of the multiplexed streams share a common base
> > sampling rate, and it can add a variable amount of delay as incoming
> > samples to the mux are buffered to wait for other samples that will be
> > combined in a single RTP packet.  I was curious if the practicality of
> > this requirement and the buffering delay have been discussed in the
> > group at all.
> 
> By sampling rate, I assume you mean frame size (10ms for G.729, for
> example), and not A->D sampling rate (8 khz, for example). 

Yes, thanks for clarifying.

> The amount of additional delay introduced by the scheme depends entirely
> on how it is used. In the above example, there was no additional delay.
> If, instead, the system were run with a single user group, with packets
> sent every 20ms, user 2 would suffer an additional 10ms packetization
> delay. It is therefore at the discretion of the implementor to determine
> the acceptable level of delay/efficiency tradeoff.

The case I had in mind was multiple independent IP-based sources having
their streams muxed together (say, by a router at the edge of a WAN). 
If all of the sources were sending their own packets every 10 ms, and
the muxed packets were sent across the WAN every 10 ms, then the
sources would have an average of 5 ms additional delay (and worse case
just under 10 ms).  This could also conceivably affect the playback,
since the original timestamps are lost.  However, if your source is a
PBX or a PSTN gateway, then the syncronization is automatic and there
aren't any playback concerns. 

But of course you're right, it's under the control of the implementer
or network operator (to the extent that the muxing is made
configurable). 

> > I was also curious if possible interactions with
> > draft-ietf-avt-crtp-04.txt have been considered, since the compression
> > it provides is a huge win.
> 
> Its use of SSRC-based user groups means it should work quite well with
> RTP header compression. Each user group will be considered a separate
> compression context. As the SN and TS within each group should generally
> behave "well", I don't see any problems.

Thanks - good to hear.

Cheers,
Andy
________________________________________________________________________
Andrew G. Malis  malis@ascend.com  phone:978 952-7414   fax:978 392-1484
Ascend Communications, Inc.        1 Robbins Road     Westford, MA 01886




From rem-conf Thu May 21 06:40:36 1998 
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From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
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Andrew G. Malis wrote:
> 

> > The amount of additional delay introduced by the scheme depends entirely
> > on how it is used. In the above example, there was no additional delay.
> > If, instead, the system were run with a single user group, with packets
> > sent every 20ms, user 2 would suffer an additional 10ms packetization
> > delay. It is therefore at the discretion of the implementor to determine
> > the acceptable level of delay/efficiency tradeoff.
> 
> The case I had in mind was multiple independent IP-based sources having
> their streams muxed together (say, by a router at the edge of a WAN).
> If all of the sources were sending their own packets every 10 ms, and
> the muxed packets were sent across the WAN every 10 ms, then the
> sources would have an average of 5 ms additional delay (and worse case
> just under 10 ms).  This could also conceivably affect the playback,
> since the original timestamps are lost.  However, if your source is a
> PBX or a PSTN gateway, then the syncronization is automatic and there
> aren't any playback concerns.

Actually, the WAN edge mux example you mention is quite problematic, and
isn't easily supported by the protocol described in the draft. The
assumption in our draft was that the media were generated locally by the
entity performing the muxing, not by outside sources. As a result, I see
the following problems applying the protocol:

1. The SSRC in the RTP packets being muxed will likely never be the
same; RTP timestamps are chosen with a random offset. As the protocol
requires the users in a user group to share a common base timestamp, no
muxing could occur.

2. The SSRC of the RTP packets being muxed will be lost, as these are
not carried by the protocol. This means that (among other things) the
RTCP packets generated by the sources will be ignored if transmitted end
to end. So, they must be dropped, and the mux point has to generate new
RTCP for the muxed source.

3. Any header extension or CSRC information in the muxed packets would
be lost.

4. The UDP header information in the muxed packets also needs to be
conveyed across the multiplex. This is not supported in our draft
either.


Basically, as there is no redundancy of information in the headers of
the packets being muxed, you would need to retain the entire UDP/RTP
header in the multiplex. In that case, the problem is not actually RTP
specific. What you are looking for is a general tunelling mechanism
which allows you to place multiple IP packets within a single tunneled
packet. 
 
-Jonathan R.
-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
FAX:   (732) 834-5379                       Rm. 4C-526
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen



From rem-conf Thu May 21 06:46:23 1998 
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Jonathan,

> Actually, the WAN edge mux example you mention is quite problematic, and
> isn't easily supported by the protocol described in the draft. The
> assumption in our draft was that the media were generated locally by the
> entity performing the muxing, not by outside sources.

OK, thanks again.  I've got a much better feel for the draft and its
application. 

Cheers,
Andy
________________________________________________________________________
Andrew G. Malis  malis@ascend.com  phone:978 952-7414   fax:978 392-1484
Ascend Communications, Inc.        1 Robbins Road     Westford, MA 01886




From rem-conf Thu May 21 07:59:19 1998 
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Date: Thu, 21 May 1998 10:53:30 -0400
From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
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I believe there is a need to be able to associate RTP packets with the
SIP address of the entity which generated them. There are a few examples
of where this is useful:

1. In a multicast conference, you hear audio from someone and want to
invite them to a side conference using SIP. 
2. You sent out a SIP INVITE, and got two 200 responses, both of which
are ACK'ed. The caller now receives audio from both sources to the same
port. How does it know which RTP stream is associated with which SIP
entity? Such correlation is needed for various UI features.

There seem to be a number of possible solutions:

1. Include the CNAME as a header somewhere in the SIP INVITE message or
SDP. This solves problem 2 above, but not problem 1.
2. Include the SSRC as a header somewhere in the SIP INVITE or SDP. This
also solves problem 2, but not 1. It has the additional drawback that
SSRC are not fixed, and may change due to a collision.
3. Use the EMAIL SDES item. For problem (1), you would send a SIP invite
to the address listed in the email identifier. For problem (2), the
email identifier could be matched with the contents of the Location
header to perform the correlation. The problem here is that SIP
addresses need not be the same as the user's email address.
4. Add a new SDES item, called SIP-CADDR, which contains the user's SIP
address. The ITU folks did a similar thing for H.332; they have
registered an H.323-CADDR type used for solving problem (1). 

Comments or other possibilities?

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
FAX:   (732) 834-5379                       Rm. 4C-526
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen



From rem-conf Thu May 21 08:46:06 1998 
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From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
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Jonathan Rosenberg wrote:
> 
> I believe there is a need to be able to associate RTP packets with the
> SIP address of the entity which generated them. There are a few examples
> of where this is useful:
> 
> 1. In a multicast conference, you hear audio from someone and want to
> invite them to a side conference using SIP.
> 2. You sent out a SIP INVITE, and got two 200 responses, both of which
> are ACK'ed. The caller now receives audio from both sources to the same
> port. How does it know which RTP stream is associated with which SIP
> entity? Such correlation is needed for various UI features.
> 

> 4. Add a new SDES item, called SIP-CADDR, which contains the user's SIP
> address. The ITU folks did a similar thing for H.332; they have
> registered an H.323-CADDR type used for solving problem (1).

Only solution (4) seems generic enough, and we definitely want to avoid
making SSRCs visible outside RTP as much as we can. Given the bandwidth
limitation of the RTCP channel, allowing re-use of the same identifier
is helpful, since having

H323-CADDR: foo@bar
SIP-CADDR: foo@bar
WhitePine-CADDR: foo@bar

is very inefficient, in particular since it is likely that many of these
addresses are going to be of the user@host variety. We could use
something like

CADDR: {sip,h323,whitepine}:foo@bar

or a binary equivalent.



From rem-conf Thu May 21 10:37:32 1998 
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From: Jim Gemmell <jgemmell@MICROSOFT.com>
To: 'Julian Highfield' <J.C.Highfield@maybeso.demon.co.uk>, Yves Lepage
	 <yves@cc.mcgill.ca>, rem-conf@es.net
Subject: wbd on NT ( was RE: Declining audiences? (Was: Netshow and the MB
	ONE))
Date: Thu, 21 May 1998 10:17:58 -0700
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I just tried it out on NT 4 and it seemed to work fine. Postscript didn't
work for me, but I'm not sure if I have the right version of ghostsript.

> -----Original Message-----
> From: Julian Highfield [mailto:J.C.Highfield@maybeso.demon.co.uk]
> Sent: Wednesday, April 22, 1998 7:24 PM
> To: Yves Lepage; rem-conf@es.net
> Subject: RE: Declining audiences? (Was: Netshow and the MBONE) JG: WBD
> 
> 
> Yves Lepage <yves@cc.mcgill.ca> wrote:
> >
> >Since then, my desktop is NT. This makes it much more 
> problematic. I am
> >not going to purchase IP/TV or Icast. My only option from 
> that point is
> >to use the freeware apps, which are not really mature on MS OS'es.
> 
> If you noticed the lack of a "wb" whiteboard I can perhaps help:
> 
http://www.stile.lboro.ac.uk/~cojch/winwbd/winwbd.html

Of course, I've only tested it under Windows 95 because I've never met
anyone who will admit to using NT...

Regards,
        Julian.





From rem-conf Thu May 21 12:02:57 1998 
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You are invited to an important Web developer/ISP seminar in beautiful Vancouver British Columbia.
 
As a Web developer, are you
- Interested in increasing your customer base with innovative services?
- Constantly challenged to push the envelope in order to provide top-notch, Web based business 
applications?
- Looking for a more efficient way of building database access over the Internet?
- In need of providing better Web site statistical information?

As an ISP, are you
- Interested in increasing your customer base with innovative services?
- Constantly challenged to push the envelope in order to provide top-notch Web hosting services?
- Looking for a revenue generating way of providing database hosting services?
- In need of providing better Web site statistical information?
 

If so, participate in this action-packed 2-hour FREE seminar sponsored by EveryWare Development 
Inc. 
 
The focus of the presentation will be on 2 key areas of interest for today's Internet professional.
1. Building Database functionality over the Web
2. Tracking and Managing web site usage
 
To find out more go to: http://betawest.everyware.com/sales/index.html 
Or read on...
 
Date: Friday June 5, 1998
Time: From 9:30 AM to 11:30 AM - Web Developers; From 1:30 PM to 3:30 PM - ISP's
Location: 
Simon Fraser University at Harbour Centre http://www.harbour.sfu.ca/
515 West Hastings Street
Vancouver, British Columbia V6B 5K3
Room 1700, Labatt Hall
Contact Linda Hewitt lhewitt@sfu.ca
604/291.5085
 
Over $10,000 in door prizes!
-refreshments
-hands-on training
-meet and learn from other professionals
 
**Space is Limited**
 
Get all the details and Register Now to receive a full Tango Developer license valued at $695 US  
for $95 US!  http://betawest.everyware.com/sales/index.html 
 
For more information contact:
Chris Moffett cmoffett@everyware.com
604/464.8704
www.everyware.com
EveryWare Development Inc.
 
If no longer want to receive messages from this list send "remove from list" in the body of your text 
and e-mail to: northwestsales@everyware.com
 




From rem-conf Thu May 21 12:28:47 1998 
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You are invited to an important Web developer/ISP seminar in beautiful Vancouver British Columbia.
 
As a Web developer, are you
- Interested in increasing your customer base with innovative services?
- Constantly challenged to push the envelope in order to provide top-notch, Web based business 
applications?
- Looking for a more efficient way of building database access over the Internet?
- In need of providing better Web site statistical information?

As an ISP, are you
- Interested in increasing your customer base with innovative services?
- Constantly challenged to push the envelope in order to provide top-notch Web hosting services?
- Looking for a revenue generating way of providing database hosting services?
- In need of providing better Web site statistical information?
 

If so, participate in this action-packed 2-hour FREE seminar sponsored by EveryWare Development 
Inc. 
 
The focus of the presentation will be on 2 key areas of interest for today's Internet professional.
1. Building Database functionality over the Web
2. Tracking and Managing web site usage
 
To find out more go to: http://betawest.everyware.com/sales/index.html 
Or read on...
 
Date: Friday June 5, 1998
Time: From 9:30 AM to 11:30 AM - Web Developers; From 1:30 PM to 3:30 PM - ISP's
Location: 
Simon Fraser University at Harbour Centre http://www.harbour.sfu.ca/
515 West Hastings Street
Vancouver, British Columbia V6B 5K3
Room 1700, Labatt Hall
Contact Linda Hewitt lhewitt@sfu.ca
604/291.5085
 
Over $10,000 in door prizes!
-refreshments
-hands-on training
-meet and learn from other professionals
 
**Space is Limited**
 
Get all the details and Register Now to receive a full Tango Developer license valued at $695 US  
for $95 US!  http://betawest.everyware.com/sales/index.html 
 
For more information contact:
Chris Moffett cmoffett@everyware.com
604/464.8704
www.everyware.com
EveryWare Development Inc.
 
If no longer want to receive messages from this list send "remove from list" in the body of your text 
and e-mail to: northwestsales@everyware.com
 




From rem-conf Thu May 21 12:56:16 1998 
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Subject: IJPDSN: Special Issue
To: ckobryn@shl.com, commsoft@cc.bellcore.com, bomsig@omg.org, adtf@omg.org,
        mof@omg.org, tc@omg.org, ptc@omg.org, dtc@omg.org, ab@omg.org,
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Date: Thu, 21 May 1998 15:53:02 -0400 (EDT)
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Hello,

The following is the Call for Papers for the IJPDSN Special Issue On 
"Network Architectures for End-to-end Quality-of-Service Support". 

Please forward this message to colleagues who might be interested.
My apologies if you receive multiple copies of this message.

Sincerely,

Prof. T. Znati,
Guest Editor


                              CALL FOR PAPERS
----------------------------------------------------------------------------
                     INTERNATIONAL JOURNAL OF PARALLEL AND 
		       DISTRIBUTED SYSTEMS AND NETWORKS 

                              Special Issue On
                     Network Architectures for End-to-end
                          Quality-of-Service Support            
-----------------------------------------------------------------------------




A wide-range of real-time and multimedia applications, such as
interactive multimedia teleconferencing, video-on-demand, and
tele-medicine, rely on the next generation of Integrated Services and
Telecommunication networks for data transport.  Successfully deploying
these networks requires new switching and transmission technologies
and new network and transport protocols capable of supporting a wide
range of Quality-of-Service requirements.  Further, these technologies
and protocols must operate and co-exist within a heterogeneous
infrastructure, encompassing real-time and best-effort networks
and wired and wireless environments.  To address these issues very
intensive research and standardization activities are being carried
out.  The research challenge is to develop these communication systems and
protocols in an efficient and scalable manner. 

The objective of this special issue is the design, implementation and
evaluation of a broad range of network architectures capable of
supporting the full spectrum of distributed real-time and multimedia
applications with both stringent and adaptive QoS guarantees.  Papers
should address current results and future directions of research and
development in new services and frameworks for these next generation
applications. Topics include, but are not limited to:

**  Network protocol architectures and services for end-to-end QoS support
**  New architectures for switching and routing at gigabit speeds
**  Modeling, simulation and implementation of real-time network architectures
**  Routing and congestion control protocols in integrated service networks
**  End-to-end QoS support over heterogeneous networks
**  Network and transport protocols for Integrated Services Networks
**  Scheduling and queue management policies for multiple traffic classes
**  Differentiated Services Architectures
**  End-to-end QoS support over Mobile and Wireless Systems
**  Open Control Architectures and Active Networks
**  Real-time control in mobile and wireless networks



Prospective authors should directly deposit their manuscripts 
(PostScript or PDF format only) by anonymous ftp at 
ftp.cs.pitt.edu:/pub/IJPDSN/incoming, or send five (5) hard copies 
to one of the Guest Editors listed below. 

The following deadlines will apply:

Manuscript Submission Date             September 30, 1998
Acceptance Notification Date           November 15, 1998
Final Manuscript Due Date              December 31, 1998
Anticipated Publication Date           Second quarter 1999

                          Guest Editors


Prof. Taieb Znati                       Prof. Robert Simon                      
Computer Science Department             Computer Science Department
University of Pittsburgh                George Mason University                 
Pittsburgh, PA 15260                    Fairfax, VA  22030                      
U.S.A.                                  U.S.A.

Phone:  +1(412)624-8417                 Phone:  +1(703)993-1556                 
FAX:    +1(412)624-8854                 FAX:    +1(703)993-1710                 
e-mail: znati@cs.pitt.edu               e-mail: simon@cs.gmu.edu                






From rem-conf Fri May 22 06:38:23 1998 
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>From allegre@risq.qc.ca  Thu May 21 19: 24:26 1998
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From: "=?iso-8859-1?Q?Christian_All=E8gre?=" <allegre@risq.qc.ca>
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Resent-From: Yves Boudreau <yboudrea@maverick.risq.qc.ca>


Title  :

RISQ '98 : High Bandwidth InterNetworking in Quebec

Description :

RISQ '98 is the yearly technical Conference of the RISQ, Quebec's Leading
Educational and Research Network. RISQ is the Quebec backbone of CA*Net-2
and of CANARIE's new Canadian Optical Network. Highlights include :
conferences about performance on IP networks, Video server architecture for
CA*Net II, the Canadian Policy regarding Internet Domain Names Registration,
and a high level panel on competing access technologies and IP telephony
(May 26, 1:30-3:00PM). Conferences are mostly in french.

Date : Monday 25 May (PM) and Tuesday 26 May (AM and PM)

URL : http://www.risq.qc.ca/risq98/


--
Christian Allegre, conseiller NTIC
Programme RISQ '98 : <<Batissons l'avenir>>, 25-26 mai a Montreal
http://www.risq.qc.ca/risq98/
RISQ Inc.
(514) 840-1235 ext. 5559
allegre@risq.qc.ca






From rem-conf Fri May 22 06:46:46 1998 
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To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
cc: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>, rem-conf@es.net, 
    confctrl@ISI.EDU
Subject: Re: SIP and RTP correlations
In-reply-to: Your message of "Thu, 21 May 1998 11:43:27 EDT." <35644B9F.646A@cs.columbia.edu>
Date: Fri, 22 May 1998 14:44:32 +0100
Message-ID: <1520.895844672@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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--> Henning Schulzrinne writes:
>Jonathan Rosenberg wrote:
...
>> 4. Add a new SDES item, called SIP-CADDR, which contains the user's SIP
>> address. The ITU folks did a similar thing for H.332; they have
>> registered an H.323-CADDR type used for solving problem (1).
>
>Only solution (4) seems generic enough, and we definitely want to avoid
>making SSRCs visible outside RTP as much as we can. Given the bandwidth
>limitation of the RTCP channel, allowing re-use of the same identifier
>is helpful, since having
>
>H323-CADDR: foo@bar
>SIP-CADDR: foo@bar
>WhitePine-CADDR: foo@bar
>
>is very inefficient, in particular since it is likely that many of these
>addresses are going to be of the user@host variety. We could use
>something like
>
>CADDR: {sip,h323,whitepine}:foo@bar

Could we use the CNAME? Put a sip: url in there? The media tools will have
to be modified anyway, to send out SIP-CADDR. This also avoids having a
proliferation of sdes packet types...

Colin



From rem-conf Fri May 22 08:00:34 1998 
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From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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Colin Perkins wrote:
> 
> --> Henning Schulzrinne writes:
> >Jonathan Rosenberg wrote:
> ...
> >> 4. Add a new SDES item, called SIP-CADDR, which contains the user's SIP
> >> address. The ITU folks did a similar thing for H.332; they have
> >> registered an H.323-CADDR type used for solving problem (1).
> >
> >Only solution (4) seems generic enough, and we definitely want to avoid
> >making SSRCs visible outside RTP as much as we can. Given the bandwidth
> >limitation of the RTCP channel, allowing re-use of the same identifier
> >is helpful, since having
> >
> >H323-CADDR: foo@bar
> >SIP-CADDR: foo@bar
> >WhitePine-CADDR: foo@bar
> >
> >is very inefficient, in particular since it is likely that many of these
> >addresses are going to be of the user@host variety. We could use
> >something like
> >
> >CADDR: {sip,h323,whitepine}:foo@bar
> 
> Could we use the CNAME? Put a sip: url in there? The media tools will have
> to be modified anyway, to send out SIP-CADDR. This also avoids having a
> proliferation of sdes packet types...

I don't think CNAME is appropriate, since it identifies the actual host
the data is coming from. In many cases, I want my SIP requests to go
through various proxies (that do call filtering, firewall drilling,
etc.). Also, we want to maintain backward-compatibility, and CNAME is
the one crucial piece of RTP SDES information that applications rely on
for grouping.

Also, it is sufficient to send the SIP URL in one media stream, while
CNAME is clearly needed everywhere.

> 
> Colin

-- 
Henning Schulzrinne   schulzrinne@cs.columbia.edu
Dept. of Comp. Sci.   ph  +1 212 939-7042
Columbia University   fax +1 212 666-0140
New York, NY 10027    http://www.cs.columbia.edu/~hgs



From rem-conf Sun May 24 01:10:39 1998 
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Date: Sun, 24 May 1998 03:48:46 -0400 (EDT)
From: Ferat Sahin <sferat@ee.vt.edu>
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remove rem-conf@es.net


--------------------------------------------------------------------------
Ferat Sahin
Ph.D. Candidate,
Virginia Polytechnic Institute & SU                 |\/\/\/|
Electrical Engineering Department              	    |	   |            
Control&Systems					    |      | My favorite 
						    | (o)(o)     cartoon
Tel: (H) (540) 953 0743                             C	   _)
     (W) (540) 231 5717  (MIL) 231 5613	            | ,___|    	
email: sferat@birch.ee.vt.edu	     		    |   / 	
homepage: http://www.ee.vt.edu/sferat              /____\
						  /      \
--------------------------------------------------------------------------- 




From rem-conf Sun May 24 07:17:52 1998 
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Date: Sun, 24 May 1998 09:10:16 -0500
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Remove daveh@airmail.net 






From rem-conf Sun May 24 07:35:32 1998 
From rem-conf-request@es.net Sun May 24 07:35:31 1998
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To: rem-conf@es.net
Cc: Frank.Lyonnet@inria.fr, rodeo@inria.fr
Subject: Rendez-Vous 1.0.3 with Layered DCT codec & Appli. level FEC ...
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Date: Sun, 24 May 1998 16:34:08 +0200
From: Frank Lyonnet <Frank.Lyonnet@sophia.inria.fr>
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   A quick word to announce the availability of Rendez-Vous 1.0.3
an Internet audio/video conferencing software developped in INRIA France.

This release of Rendez-Vous should be of interest for people willing
to experiment with multilayer video codec, application level FEC and
bit error resilience.

Extracted from the Rendez-Vous web page (http://www.inria.fr/rodeo/rv) :
------------------------------------------------------------------------

Rendez-Vous is an Internet videoconferencing tool developed at INRIA
by Frank Lyonnet.
It is in a way the successor to the IVS tool developed some time ago by Thierry
Turletti, which was one of the first MBone tools available (however
it is a completely new piece of code compared to IVS).

The audio component of Rendez-Vous is an integration of the application
kernel of the FreePhone 3.5 audio tool developped also at INRIA by Sacha 
Fosse-Parisis
and Andres Vega Garcia (http://www.inria.fr/rodeo/fphone).

Main features of Rendez-Vous :

- RTP protocol support over multicast or unicast IP.
- H261 video standard.
- High quality PCMU, ADPCM, VADPCM (with HiFi support) and low bandwidth GSM 
and LPC audio coding.
- Mpeg 1/2 file reading / transcoding
- An integrated scheduler for multilayer video and audio flow management and 
processing
(layer synchronisation and optimization of machine ressources to maximise the 
suggestive quality
rendered to the user).
- Experimental multilayer DCT based video codec.
- Application level hierarchical FEC.
- Bit error resilience with layered DCT codec.


Who should care for Rendez-Vous ?

Who should NOT care for Rendez-Vous :
Rendez-Vous is not intended to be commercial product. We, the RODEO
team at INRIA are a small research group, part of a French public institute.
In no case we recommend Rendez-Vous for a buisness or even personnal use.
We will not provide any support for such use of Rendez-Vous.

Who should care for Rendez-Vous :
Rendez-Vous is an experimental research tool. It has been developped in
order to be an ideal testbed for some hot topics such as layered video
and audio transport and coding, Forward Error Correction for video and
audio on the Internet, wireless and satellite links access to the Internet.
As a side effect of this primary goal, Rendez-Vous can also be used for
personnal use by people with an adventurous mind.


Supported systems :

- Sun Sparc Solaris machines, internal audio hardware, SunVideo and VigraPix
grabbers
- x86 Linux machines, VoxWare audio by Sacha Fosse-Parisis, no video grabbing
- x86 FreeBSD machines, VoxWare audio by Sacha Fosse-Parisis, no video grabbing
- Windows 95 machines, DirectX 5.0 audio, VideoForWindows (QuickCam only) and 
Matrox Meteor grabbing. Windows NT 4 users
have to wait for offical support for DirectX 5.0 in NT 4.


Frank Lyonnet
Frank.Lyonnet@inria.fr
http://www.inria.fr/rodeo/personnel/Frank.Lyonnet/






From rem-conf Sun May 24 08:14:38 1998 
From rem-conf-request@es.net Sun May 24 08:14:37 1998
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In message <some nerd> writes:
> Remove <some nerd>

Is it possible to filter out this kind of spam from the list ?

/H=E5kan


---------------------------------------
e-mail: Hakan.Lennestal@lu.erisoft.se |
     or Hakan.Lennestal@cdt.luth.se   |
     or hakan@tuttifrutti.cdt.luth.se |
---------------------------------------




From rem-conf Sun May 24 09:38:55 1998 
From rem-conf-request@es.net Sun May 24 09:38:54 1998
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To: Frank Lyonnet <Frank.Lyonnet@sophia.inria.fr>
cc: rem-conf@es.net, Frank.Lyonnet@inria.fr, rodeo@inria.fr
Subject: Re: Rendez-Vous 1.0.3 with Layered DCT codec & Appli. level FEC ... 
In-reply-to: Your message of "Sun, 24 May 1998 16:34:08 +0200."
             <199805241434.QAA26777@tac.inria.fr> 
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Date: Sun, 24 May 1998 09:37:10 -0700
From: Amancio Hasty <hasty@rah.star-gate.com>
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Thank you for supporting FreeBSD!

Any reason why there is no FreeBSD video capture support? 
If you like you can use vic as  model for interfacing to our matrox 
meteor or bt848 driver in fact the vic's freebsd video capture is the same 
for the meteor and bt848  cards.

	Tnks,
	Amancio







From rem-conf Sun May 24 17:36:20 1998 
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From rem-conf Sun May 24 17:58:11 1998 
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Hi, rem-conf.

Your `UNSUBSCRIBE UGST' request has been received by
"Simao F. Campos-Neto" <simao.campos@comsat.com>; however no lists are named `UGST'.

Available lists are:
    UGST             - UGST e-mail discussion group
    UGST-HW          - UGST hardware specific e-mail discussion group
    UGST-MANUAL      - UGST STL Manual editorial group


Please try again.
__________________________________________________________________________




From rem-conf Mon May 25 11:08:41 1998 
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        <mof@omg.org>, <tc@omg.org>, <ptc@omg.org>, <dtc@omg.org>,
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Cc: <ctc-members@redbank.tinac.com>, <dbworld@cs.wisc.edu>,
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Subject: Last CFP for IEEE CNIW'98 
Date: Mon, 25 May 1998 20:34:41 +0530
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Dear recipient:

This is the last CFP for IEEE CNIW'98 to be held at Ahmedabad, India during
9-12 Dec, 1998.


It is my great pleasure to introduce you to the IEEE 1998 Conference on
Networking India & the World (CNIW98) to be held at Ahmedabad
during December 9-12, 1998 as a part of the Annual Convention &
Exhibition (ACE98) programme of the IEEE India Council. It is being
organized and hosted by the IEEE Gujrat Section in cooperation with the
IEEE Computer Society Chapters of the India Council. As you are aware
of, IEEE is the worlds largest technical professional body with the current
motto of Networking the World.
ACE98 is one of the premier events in our profession providing an
excellent forum for learning and sharing ideas and experiences with other
practitioners and organizations. This year the heart of ACE is going to be
the CNIW98 - a leading conference in identifying and focussing modern
trends in the development of new initiatives in networking. Accordingly,
the conference theme has been decided to highlight the importance of
networks in todays ever-changing world. The conceived Technical
Program track also embodies that theme through its multicultural mix.
The evolution of networking techniques resembles a revolution. The
analog to digital transition is virtually complete. The dream of networking
throughout the world is moving quickly towards reality, as advanced
technologies are being developed. Network access is trending from wired
to wireless. Mobility and multimedia are being demanded by end-users. In
near future, networks will be capable of instantly transporting and
processing information between any locations, be they homes, offices or
businesses. Various new technologies, new system solutions and new
business opportunities have created a completely new horizon of
networking that constitutes the nervous system of todays economy and
more generally of tomorrows society. In this context, INTERNET can be
cited as the most dynamic sector of activity and growth worldwide. Being
a part of this global information economy, India is all poised to take the
centre stage in the coming millennium, fuelled by an expanding economy
and increasing affluence of its population.
Service providers are merging and de-merging and deregulating.
Liberalization and globalization of services and networks are now driving
the networking market, with the backdrop of a multi-vendor and multi-
service provider environment. Virtually every network is composed of
equipment from multiple vendors. Vendors are forging alliances amongst
themselves. Still, is it possible to manage the global information
superhighway? What are the real issues? Are standards the answer? Will
your systems and technologies enable it? What are the practical and
theoretical limits? How is the INTERNET being driven by business and
social forces? Millions of such problems, in all facets of networking, are
growing faster than they are being solved. More importantly, these
challenges make it even harder than ever to strive for cooperation among
customers, service providers and vendors. To explore these challenges and
many more, and to debate on their possible solutions, we cordially invite
you to use the platform of CNIW98.
To this end, CNIW98 features an intellectually stimulating technical
program with 8 technical sessions having a broad coverage of topics on
network technologies, applications, operations, management, policies and
future developments. It also offers timely and highly relevant tutorials by
subject matter experts on recently developed techniques. Moreover, four
invited talks by eminent speakers provide a vision into the future and
discuss networking strategies from business, service and technology
perspective. In addition, manufacturers/service providers/consortia will be
presenting technical demonstrations, featuring state-of-art ideas in
networking and illustrating the latest advances of interest to industry
professionals responsible for Networking India and the World.
Given the rapid pace at which developments occur within the networking
community, a major national event like this is vital in keeping
professionals aware of the state-of-the-art technology. It provides a
valuable forum for exchanging knowledge and experience between
academicians, research scientists and technical managers. It also serves as
a showcase of the most advanced products and services, offering industry
leaders a marvelous opportunity to display their revolutionary
technologies, as well as to make contacts which will prove beneficial in
the years to come. Indeed, it is a grand occasion to distinguish yourself in
an elite gathering of leaders and professional from all corners of the
networking world for four days at Ahmedabad.
Lastly, we are sure you will find this Conference to be not only an
excellent forum for innovation and technical discussion, but also a very
natural environment for extending friendship and fellowship here at
Ahmedabad.
So please do join us for this very special event and be a part of
Networking India & the World.
See you in Ahmedabad.
Dr. Debashis Saha.
(Program Committee Vice Chairman)
Dept. of Computer Science & Engg.
Jadavpur University
Calcutta 700 032
Tel. : +91  (33)  413 1766 (O)
         +91  (33)  462 2833 (R)
Fax  : +91  (33)  473 1138
email : d.saha@computer.org, debashis_saha@hotmail.com



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From: d.saha@computer.org
To : tccc@ieee.org

>=20
>                  ANNUAL CONFERENCE & EXHIBITION
> 	            NETWORKING INDIA & THE WORLD
> =20
>=20
> December 9-12, 1998       IRS Campus, ONGC Chandkheda, Ahmedabad
>=20
> =20
>=20
>                           Organised by
>=20
> INDIA  COUNCIL, GUJARAT SECTION, Computer Society Chapter,  India=20
>=20
> Council Computer Society Chapter, Gujarat Section
>=20
> =20
>=20
>                       in co-operation with
>=20
>      INSTITUTION OF ENGINEERS (INDIA), GUJARAT STATE CENTRE
>=20
>     GUJARAT ASSOCIATION OF ELECTRONICS & SOFTWARE INDUSTRIES
>=20
> =20
>=20
> =20
>=20
> For details please see our Web-Site at "ieee-gujarat.org"
>=20
> =20
>=20
> The Conference will provide a vibrant forum to debate the  entire=20
>=20
> gamut   of   continually  exploding   NETWORKING   and   INTERNET=20
>=20
> environment - and their impact, especially in relation to  India.=20
>=20
> The  exchange  of information will be  through  presentation  and=20
>=20
> discussion  of  papers,  tutorials, the  Best  Web  Page  Awards,=20
>=20
> Manufacturers' Presentations and Exhibition.  =20
>=20
> =20
>=20
> =20
>=20
> The Conference  elicits  papers  on  the  following  and  related=20
>=20
> topics :
>=20
> =20
>=20
> 1. Impact of Networking                 4. Applications                =

>=20
>    * Business Impact of the web         * Virtual Reality              =
                             =20
>=20
>        and Intranets                    * Science  & Engineering   =20
>=20
>    * Management Issues                  * Internet Commerce            =
                         =20
>=20
>    * Infrastructure                     * Electronic  Publishing       =
                             =20
>=20
>    * Indian Economics                   * Online Resources         =20
>=20
>                                         * Information Services     =20
>=20
>                                         * Education                =20
>=20
> =20
>=20
> 2.  Network Technologies               5. Policies and Guidelines for  =
     =20
>=20
>     * Networking  Products              Cyberlaws                      =
      =20
>=20
>     * LAN/WAN Technology                * Ethics and Legalities        =
          =20
>=20
>     * Server Technologies               * Intellectual Property =
Rights(IPRs
>=20
>     * Network Issues                    * Freedom vs abuse of Internet =
   =20
>=20
>                                                                        =
                                          =20
>=20
> 3. Software Tools                       6. Internet Issues             =
     =20
>=20
>    * Multimedia                         * Security                     =
     =20
>=20
>    * Internet Tools                     * Authenticity of information  =
     =20
>=20
>    * Web Browser and Search Engines     * Forgery, Frauds              =
     =20
>=20
>    * Audio/Video on Internet            * Privacy                      =
     =20
>=20
> =20
>=20
>                                         7.  Future Developments        =
   =20
>=20
>     =20
>=20
> Besides   the papers  on the main theme, papers on any field   of =20
>=20
> interest  covered  by any IEEE Society/Council are also welcome.       =
       =20
>=20
> =20
>=20
> Associated   with   the  Conference  will be a  prestigious   All =20
>=20
> India   Best  Web  Page  Award. Companies registered in India  or=20
>=20
> individuals engaged in web design can compete for the Award.  For=20
>=20
> competing  for  the  award,  please send the  URL  along  with  a=20
>=20
> Certificate from the client.  A disk containing the  Web Page may =20
>=20
> also be  provided.
>=20
> =20
>=20
> CONFERENCE PROGRAM :
>=20
> =20
>=20
> 9   Dec, 1998   Tutorial on INERNET
>=20
> 10  Dec, 1998   Tutorials on Software Technologies for Networked=20
>=20
>                 & Internet-connected systems.
>=20
> 11-12 Dec,1998  The Conference
>=20
> 12  Dec, 1998   Presentation of the All India Web Page Award
>=20
> =20
>=20
> =20
>=20
> Participate in the Manufacturer's Presentation  :
>=20
> =20
>=20
> The  ACE-98  December  Seminar will  be a  gathering  of  leading =20
>=20
> professionals and decision makers. These are people who will   be=20
>=20
> actual  users  and buyers of your products and  will  decide  the =20
>=20
> suitability  of  your products for their  applications  in  their=20
>=20
> respective  organisation.
>=20
> =20
>=20
> Keeping  this  in  mind,   Special  Presentation  slots  will  be=20
>=20
> allotted to the manufacturers of Networking Products and Services=20
>=20
> during the conference.
>=20
> =20
>=20
> So  if  your   industry has anything to offer for  use  by  these=20
>=20
> computer  professionals, don't miss this opportunity. Book   your=20
>=20
> time  slots now. Please note  that these are limited and will  be=20
>=20
> booked on first-come first-served basis.
>=20
> =20
>=20
> The rate is Rs. 15,000 (US$ 400) for a 30 minutes slot.
>=20
>            =20
>=20
> DEADLINES
>=20
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***********************************************************************
Claudio Cabano                               Phone     +41 31 342-59-26
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Subject: Question about IP/UDP/RTP headers compression
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Hi,
I'm investigating the Casner-Van Jacobson draft in order to produce
an implementation of it. Still some doubts remain about it...
I apologize whether this subject has been already faced on the mailing list
(it should be indeed), but I'm still quite new with it :(

In the draft an important part of the compression scheme it's based on the 
assumption that the link-level protocol provides for a packet length 
indication.
I've checked 'PPP' and 'PPP in HDLC-like framing' RFCs and I haven't yet
figured out where (or how) the packet length indication comes from
(or is deduced).

Any hint?

Thanks in advance

Valter

================================================================
|            Valter Mazzaro        | E-mail:                   |
| ---   ---  SW Engineer           |  teima@kk.etx.ericsson.se |
|    \|/     ERICSSON ITALIA R&D   |  mazzaro@tei.ericsson.se  |
| SW- O --L  c/o SwitchLab         |                           |
|    /|\     KK/ETX/DN/SL 2204:083 | Phone : +46 8 7191607     |
| ---   ---  Dialoggatan 1, 12625  | Ecn   : 850 91607         |
|            Stockholm (Sweden)    | Fax   : +46 8 7196677     |
================================================================



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Date: Tue, 26 May 1998 14:10:05 -0700
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Wednesday, May 27, 1998, 7pm
San Diego Supercomputer Center (SDSC) Auditorium

We have scheduled two very interesting speakers this month:

Jim Holscher
Symantec Corporation

Symantec will demonstrate their most current Java Tool, Visual Cafe Database
Development Edition 2.5. Come see why Visual Cafe is the hottest Java tool
and enter your name in a raffle to win a free evaluation copy of Visual
Cafe! 


Lane Sharman
GoodTimes
"Time and Business Management. Anywhere, Anytime."

Lane Sharman will present GoodTimes, a new Java based application for time
and business management. 

GoodTimes contains a new and intuitive interface. It is a multi-tiered
application using Oracle or a JDBC compliant database on the backend. For
this presentation, Lane will discuss some of the engineering challenges and
approaches to building a client-server application that runs exclusively
over the Internet. GoodTimes provides a managed set of essential business
and developmental functions to any Java capable workstation.  The delivery
and continuous
maintenance of the software is transparent. Application Distribution and
Management is a core ingredient allowing customers to enjoy
zero-administration benefits on the desktop. With time permitting, Lane will
also touch on some new technical developments including mutable algorithms
in Java. 


For directions and parking information, see
the Southern California Web Programmers Forum
home page at  http://www.sdsc.edu/scwpf




From rem-conf Wed May 27 05:16:04 1998 
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From: The IESG <iesg-secretary@ietf.org>
Subject: Document Action: Options for Repair of Streaming Media to
	 Informational
Date: Wed, 27 May 1998 08:06:47 -0400
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The IESG has approved the Internet-Draft 'Options for Repair of
Streaming Media' <draft-ietf-avt-info-repair-03.txt> as a Informational
RFC.  This document is the product of the Audio/Video Transport Working
Group.  The IESG contact persons are Scott Bradner and Vern Paxson.



From rem-conf Thu May 28 00:33:34 1998 
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Date: Thu, 28 May 1998 00:21:49 -0700 (PDT)
From: Stephen Casner <casner@precept.com>
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To: Valter Mazzaro <teima@kk.etx.ericsson.se>
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Subject: Re: Question about IP/UDP/RTP headers compression
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On Tue, 26 May 1998, Valter Mazzaro wrote:

> I've checked 'PPP' and 'PPP in HDLC-like framing' RFCs and I haven't yet
> figured out where (or how) the packet length indication comes from
> (or is deduced).

There is not an explicit length field.  HDLC framing uses a starting
flag character and ending flag character to delimit the frame.  The
packet length is determined from the number of bytes in between (after
subtracting overhead bytes such as checksum).
							-- Steve




From rem-conf Fri May 29 21:50:58 1998 
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Message-ID: <356F8B9D.17B9@icast.com>
Date: Fri, 29 May 1998 21:31:25 -0700
From: Vinay Kumar <vinay@icast.com>
Reply-To: vinay@icast.com
Organization: ICAST Corporation
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Subject: icast c:tv 3.0 beta release
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the icast c:tv software is fully compatible with vat/vic/sdr and works
on Win95, Win NT 4.0, and
MacOS8. "icast c:tv 3.0" includes:

- viewer & guide 3.0
     software for viewing sdp, rtp, rtcp. includes video codecs (h.263,
h.261).

- iss (icast server suite) 3.0 includes:
     recorder server: records and playsback any ip multicast rtp stream.
     relay server: converts multicast to unicast rtp streams.
     meter server: enables rtcp monitoring.
iss runs as an NT service.

- streamsMeter client 3.0
    viewer to connect to icast meter server and monitor rtcp stats.

- broadcaster, 2.2.1
    server to transmit h.261, h.263 live streams; or pre-recorded AVI
files.

as you can see, we are fully committed to the advancement of open
standards based IP Multicast protocols (rtp, rtcp, sap, sdp et.al.) and
applications.

however we are a commercial company and have to pay our own salaries
therefore we do recommend everyone to eventually buy the product(s).
this is the only way to fix and stabilize the MBone and make it a
household name for internet broadcasting. 

Download site: http://www.icast.com/evalform.html
submit bug-reports to: support@icast.com
ICAST website: http://www.icast.com/

we will do our best to fix outstanding bugs, problems and
interoperability issues in our products. please report bugs to:
support@icast.com. we look forward to working with you.

happiness and regards
---
  vinay
http://www.icast.com/evalform.html




From rem-conf Sat May 30 15:44:47 1998 
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From: E0jrO9dS1@1step6.za
DATE: 30 May 98 6:29:41 PM
Message-ID: <lc5s9udE2A4g5yi>
TO: magnli4y.ur@mail.ats.it
SUBJECT: In the market for a new car?
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Union Nissan of Lake County can show you over 500 new, almost new program cars, and fully 
certified used cars of all makes and models. Whether you're looking for a great buy on a used vehicle, 
would like to save thousands on a program car, or you really have your heart set on a new car; you 
owe it to yourself to see our selection and talk to our people. Our service department has evening 
and Saturday hours and every new Union Nissan comes with Gold Card privileges. Your Gold Card 
will entitle you to $9.95 lifetime oil changes, free service loaners, free towing and more! We invite you 
to e-mail our Internet Sales Manager Aaron Segal at unionnissan@iconnect.net or e-mail directly 
>from our website, www.unionnissan.com., or call (847) 244-8000 and ask for Aaron. 
ALL INTERNET CUSTOMERS WILL RECEIVE A FREE TIRE ROTATION JUST FOR COMING IN. 
We look forward to hearing from you!






From rem-conf Sun May 31 23:16:09 1998 
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To: rem-conf@es.net
Reply-To: anoop@sasi.com
Subject: H.323 protocol stack.
Date: Mon, 01 Jun 1998 11:27:46 +0530
From: Anoop Kulkarni <anoop@sasi.com>
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Hello,

Is there any public domain H.323 protocol stack implementation available?
If not, then what are the companies that sell the development version of
H.323 (alongwith source code that is)? If it *has* to be bought, then 
what is the typical price range?

Thanks a lot for your responses,

Best regards,

anoop
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