From rem-conf Tue Sep 02 09:15:22 1997 
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Date: Tue, 02 Sep 1997 17:57:20 +0200
From: Peter Hoermann <peha@darmstadt.gmd.de>
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Subject: H.320/H.323-Gateways & Pricing
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Hi all,

looking for H.320/H.323 videoconferencing gateways I came across 3
companies offering these products:

+ RADVision (L2W-323)
+ Picturetel (LiveGateway 3.0)
+ First Virtual Corporation (V-Gate 323)

Are their more companies offering these kind of gateways ?
Very interesting would be their features, the date of availability and
the pricing, too.

Regards

Peter



From rem-conf Tue Sep 02 12:24:26 1997 
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To: confctrl@ISI.EDU, rem-conf@es.net
From: Alagu Periyannan <alagu@apple.com>
Subject: Sending "cliprect" over SDP
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Hi all:

Popular video compressions schemes used with RTP
(H.262, H.263) offer a restricted set of picture sizes.

These are CIF (376x288), QCIF (188x144) and in the
case of H.263, 4CIF, 16CIF. Since these sizes are fixed
many of today's conferencing and broadcast tools
place a gray border around an image captured at a
different size.

This looks fine when the aspect ratio is close to
the CIF ratio as is the case for NTSC. However, if one
needs to send say cinema-style letterbox format the
gray borders will be fat and will not look good.
Letterbox format will be popular for sending movies
over RTP in conjuction with RTSP.

I would like to propose a scheme that can be used to
send the real video size over SDP. This will not solve
the gray border problem for conferencing since each
participant can potential have a different size border.
However, it will solve the problem in broadcast and
unicast (in conjunction with RTSP) situtations.

The "cliprect", i.e. the rectangle containing the useful
part of the picture, will be sent in an "a=" attribute
line within the "m=video" context.

a=cliprect: <top> <left> <bottom> <right>

Examples:

if we want to send 160x120 centered
inside a QCIF RTP session we send,

a=cliprect: 12 14 132 174

if we want to send 160x120 top left justified
inside a QCIF RTP session we send,

a=cliprect: 0 0 120 160

if we want to send letterbox (160x69) top left justified
inside a QCIF RTP session we send,

a=cliprect: 0 0 69 160

It would be useful if we all agree to this and generate/recognize it.

Any suggestions?

Also, how does one formalize an "a=" attribute? (IANA?)





---------------------------------------------------
Alagu Periyannan                   alagu@apple.com

Interactive Multimedia Group
Apple Computer, Inc.





From rem-conf Tue Sep 02 13:04:39 1997 
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From: Mark Handley <mjh@east.isi.edu>
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X-Phone: +1 617 253 6011
To: Alagu Periyannan <alagu@apple.com>
cc: confctrl@ISI.EDU, rem-conf@es.net
Subject: Re: Sending "cliprect" over SDP 
In-reply-to: Your message of "Tue, 02 Sep 1997 12:17:05 PDT."
             <v03020923b031a685ecfd@[17.255.20.120]> 
Date: Tue, 02 Sep 1997 16:01:06 -0400
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>The "cliprect", i.e. the rectangle containing the useful
>part of the picture, will be sent in an "a=" attribute
>line within the "m=video" context.
>
>a=cliprect: <top> <left> <bottom> <right>


>It would be useful if we all agree to this and generate/recognize it.
>
>Any suggestions?

This seems like a reasonable suggestion to me.  It would be even
better if there was a simple way to indicate the same thing for
multi-party conferencing, but short of adding a new RTCP message type
(which isn't really a good idea) I can't think of one.

>Also, how does one formalize an "a=" attribute? (IANA?)

The process isn't formalized yet, but we expect this to happen through
IANA.  Whether we expect people to write a one-page RFC for each
attribute, or whether we'll collect new attributes together
periodically, or whatever is not decided yet, but will I expect be
sorted out when the IESG approve SDP for Proposed Standard.

Mark



From rem-conf Tue Sep 02 14:40:14 1997 
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To: Alagu Periyannan <alagu@apple.com>, confctrl@ISI.EDU, rem-conf@es.net
From: "Dr. Stephan Wenger" <stewe@cs.tu-berlin.de>
Subject: Re: Sending "cliprect" over SDP
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At 12:17 02.09.97 -0700, Alagu Periyannan wrote:
>
>Hi all:
>
>Popular video compressions schemes used with RTP
>(H.262, H.263) offer a restricted set of picture sizes.
>
>These are CIF (376x288), QCIF (188x144) and in the
>case of H.263, 4CIF, 16CIF. Since these sizes are fixed
>many of today's conferencing and broadcast tools
>place a gray border around an image captured at a
>different size.

I think that everyone agrees to the necessity of having a
more flexible choice of picture sizes than the old CIF
model. ITU-T expert groups on video compression realized this
as well. The new version of H.263, currently known as it's
working name H.263+, offers pictures sizes between 4 and 2048
pixels in X and Y dimension, which can be negotiated in an
4 pixel granularity. The signalling of the size of the coded
picture is done within the Picture Header of H.263+. Similar
mechanisms are available in MPEG 2 alias H.262.

H.263+ will be a ITU-T recommendation around January 1998. It
might be worth to wait for this new standard, especially
because the implementation effort for the enhancement of
the current (1995) version of H.263 to the new standard is
not very high, if none of the optional modes are implemented.

However, it might be very useful to think today about the
necessity of having any announcement of this information,
if it is already contained in the bitstream. One reason for
that, which comes in mind, is recource allocation in the receivers.

Opinions?

Regards

Stephan
--
--
Dr. Stephan Wenger   stewe@cs.tu-berlin.de



From rem-conf Wed Sep 03 03:26:55 1997 
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Date: Wed, 3 Sep 1997 10:25:18 +0200
From: Baltzer Science <mailer@ns.baltzer.nl>
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Dear Reader 

Please find below the contents of our journal

    Annals of Software Engineering 3

More information on this journal:

    http://www.baltzer.nl/ansoft/

Sincerely,
Baltzer Science Publishers
mailer@ns.baltzer.nl


_________________________________________

Annals of Software Engineering 3 (1997)

Software Requirements Engineering

Editor: Nancy R. Mead


Nancy R. Mead
Preface 1-3

Michael Jackson
The meaning of requirements 5-21

Colin Potts and Idris Hsi
Abstraction and context in requirements engineering: Toward a synthesis
23-61

Alan M. Davis, Kathleen Jordan and Tsuyoshi Nakajima
Elements underlying the specification of requirements 63-100

Ian Sommerville and Pete Sawyer
Viewpoints: principles, problems and a practical approach to requirements
engineering 101-130

Jawed I. Siddiqi, Ian C. Morrey, Chris R. Roast and Mehmet B. Ozcan
Towards quality requirements via animated formal specifications 131-155

Mark A. Ardis
Formal methods for telecommunication system requirements: A survey of
standardized languages 157-187

P. Ciancarini, S. Cimato and C. Mascolo
Engineering formal requirements: An analysis and testing method for Z
documents 189-219

David L. Coleman and Albert L. Baker
Synthesizing structured analysis and object-based formal specifications
221-253

Fran Lustman
A formal approach to scenario integration 255-271

Sunil Vadera and Farid Meziane
Tools for producing formal specifications: a view of current architectures
and future directions 273-290

Pei Hsia, David Kung and Chris Sell
Software requirements and acceptance testing 291-317

Hermann Kaindl
A practical approach to combining requirements definition and
object-oriented analysis 319-343

F. Javier Lerch, Deborah J. Ballou and Donald E. Harter
Using simulation-based experiments for software requirements engineering
345-366

Christof Ebert
Dealing with nonfunctional requirements in large software systems 367-395

Balasubramaniam Ramesh, Curtis Stubbs, Timothy Powers and Michael Edwards
Requirements traceability: Theory and practice 397-415

Richard E. Fairley and Richard H. Thayer
The concept of operations: The bridge from operational requirements to
technical specifications 417-432

George Spanoudakis and Anthony Finkelstein
Reconciling requirements: a method for managing interference, inconsistency
and conflict 433-457

Robyn R. Lutz and Robert M. Woodhouse
Requirements analysis using forward and backward search 459-475

Robert J. Kosman
A two-step methodology to reduce requirement defects 477-494





From rem-conf Wed Sep 03 04:14:10 1997 
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Subject: Protocol Action: RTP Payload for Redundant Audio Data to Proposed
	 Standard
Date: Wed, 03 Sep 1997 07:11:18 -0400
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  The IESG has approved the Internet-Draft "RTP Payload for Redundant
  Audio Data" <draft-ietf-avt-rtp-redundancy-01.txt, .ps> as a Proposed
  Standard.  This document is the product of the Audio/Video Transport
  Working Group.  The IESG contact person is Allyn Romanow.


Technical Summary

  This document describes a payload format for use with the Real-time
  Transport Protocol (RTP), verson 2, for encoding redundant data.
  Motivation for the protocol is the development of audio conferencing
  tools used in lossy packet networks such as the Internet Mbone.  In a
  packet network, packet loss disrupts speech intelligibility making
  audio applications such as multimedia conferencing over the Internet
  undesirable to users. This protocol specifies a method for
  introducing redundancy into the data stream which allows the message
  to be reconstructed by the receiver when there is packet loss.
  Research referenced in the draft shows that the method works in
  situations found in the internet where simpler solutions are less
  effective.

Working Group Summary

  There were initially two or three approaches to indicate which
  encodings are being carried.  However, joint development between UCL
  and INRIA led to a single approach which is reflected in this draft.
  Bottom line, no WG dissension.

Protocol Quality

This draft was reviewed for the IESG by Allyn Romanow.



From rem-conf Wed Sep 03 04:41:27 1997 
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From: Peter Hoermann <peha@darmstadt.gmd.de>
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> > + RADVision (L2W-323)
> > + Picturetel (LiveGateway 3.0)
> > + First Virtual Corporation (V-Gate 323)
> >
> 
> Do you have URLs for these products?

Hi Merv and all,

in addition to my request of yesterday I'm adding the URLs for the above
gateways (the ones I already know of):

Radvision:
http://www.radvision.com/press.htm

Picturetel:
http://www.pictel.com/livelanp.htm

FVC:
http://www.fvc.com/whatsnew/gate.html

Regards

Peter



From rem-conf Wed Sep 03 15:08:13 1997 
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	MBone Broadcast Announcement
	----------------------------

Title:       
	Advanced Perspective Display Formats for Air Traffic Control 
Date:        
	Oct 01, 1997

Time:        
	12:30 PST8PDT 2 hours

Contact:     
	wu@bmrc.berkeley.edu

URL:         
	http://bmrc.berkeley.edu/298/w6.html

Description:        
	An on-line dynamic method for automated viewing parameters management in perspective displays is presented.









mbone broadcast schedule http://www.msri.org/mbone



From rem-conf Wed Sep 03 21:33:07 1997 
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This broadcast is already in progress.


	MBone Broadcast Announcement
	----------------------------

Title:       
	AUUG97 conference from Brisbane Australia
Date:        
	3-5 September 1997

Time:        
	0900-2330 AEST (GMT+10)

Contact:     
	malcolm@it.ntu.edu.au

URL:         
	http://www.auug.org.au/auug97/

Description:        
        Keynote speakers and the technical stream from the AUUG97 conference.
This event (and MBONE broadcast) is already half over - but there is still 
many things to come, including the conference dinner which will feature Robots
>from the University of Queensland! (Dinner starts at 1900 AEST (GMT+10) don't
miss it!)


-- 
Malcolm Caldwell - Network Supervisor             Email:malcolm@it.ntu.edu.au
Information Technology Support                    Ph:  +61 8 89466631
Northern Territory University,Darwin              Fax: +61 8 89466630
CASUARINA 0909 Australia



From rem-conf Thu Sep 04 11:47:44 1997 
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To: confctrl@ISI.EDU, rem-conf@es.net
From: Alagu Periyannan <alagu@apple.com>
Subject: Re: Sending "cliprect" over SDP
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Hi all,

There was an error in the CIF/QCIF numbers I sent out
in my previous email. Thanks to Kipp (H.A. Kippenhan Jr.) for
pointing this out.

Actually the CIF size is 352x288 and the QCIF size is 176x144.
(I had incorrectly used 376x288 and 188x144)

Also please note that the origin for the purposes of sending the
cliprect is located at the top left corner of the image.

So the examples will read,

Examples:

if we want to send 160x120 centered
inside a QCIF RTP session we send,

a=cliprect: 12 8 132 168

if we want to send 160x120 top left justified
inside a QCIF RTP session we send,

a=cliprect: 0 0 120 160

if we want to send letterbox (160x69) top left justified
inside a QCIF RTP session we send,

a=cliprect: 0 0 69 160





---------------------------------------------------
Alagu Periyannan                   alagu@apple.com

Interactive Multimedia Group
Apple Computer, Inc.





From rem-conf Fri Sep 05 11:13:30 1997 
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Message-ID: <341047C3.9A0EB8CB@dnrc.bell-labs.com>
Date: Fri, 05 Sep 1997 13:56:19 -0400
From: "Jonathan D. Rosenberg" <jdrosen@dnrc.bell-labs.com>
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avt'ers,

Does anyone know of any good MPEG2 encoder/decoder boards for PC's (Win
NT) that come with software that allow me to send the compressed video
over a network? A board with an NTSC output to display the decompressed
video on a TV would also be a big plus.

Thanks,

Jonathan R.
-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
PHONE: (908) 949-6418                       Rm. 4B-505
FAX:   (908) 834-5379
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen



From rem-conf Fri Sep 05 12:26:26 1997 
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Date: Fri, 5 Sep 1997 12:24:22 -0700
To: jdrosen@dnrc.bell-labs.com
From: ari@sprintlabs.com (Ari Ollikainen)
Subject: Re: MPEG Boards
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jdrosen@dnrc.bell-labs.com wrote:
>avt'ers,
>
>Does anyone know of any good MPEG2 encoder/decoder boards for PC's (Win
>NT) that come with software that allow me to send the compressed video
>over a network? A board with an NTSC output to display the decompressed
>video on a TV would also be a big plus.


        You could just go and by the IBM 8300 V(ideo)A(ccess)Node for
        about ~$40K and have a complete, integrated (encoder + decoder+
        ATM25/155 interface + software+ development tools) solution all
        packaged in a 200Mhz Pentium running OS/2!

        Alternatively:

        Check out the "The MPEGXpress 2000...a real-time MPEG-2 encoding and
        playback solution. Similar configurations are available for
        Windows NT and for PCI-based Power Macintosh..."

        more info at: http://www.mpegexpress.com/2000specs.html

        Wired Inc makes a encoder (PCI) board, ButaneII, which is available
        by special order for ~$5K. They also make PCI decoder boards
        that are around ~$.5K.

        more info at: http://www.wiredinc.com

        InnovaCom has an encoder board (PCI), DV2100/2110, that's ~$10K.

        more info at: http:www.innovacom-mpeg2.com

        Expect these prices to plummet by 1Q/2Q98 when single-chip
        encoders will become available to build single-board codecs...




           _/_/   _/_/_/_/    _/  Ari Ollikainen <ari@sprintlabs.com>
        _/  _/   _/     _/   _/ Sprint Advanced Technology Laboratories
     _/_/_/_/   _/_/_/_/    _/ Switching and Interworking Development
   _/     _/   _/     _/   _/ 1 Adrian Court (M/S: CABURS0102)
 _/      _/   _/       _/ _/ Burlingame, CA 94010
~~~~~~~~~~ OLTECO ~~~~~~~~~ Voice: 415.375.4265 FAX: 415.375.4490





From rem-conf Fri Sep 05 13:46:10 1997 
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From: gordon.joly@bbc.co.uk (Dr. Gordon Joly)
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X-Phone: +44 181 752 5454
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The BBC has joined to the MBONE to provide a service to the public,
both within the UK and abroad.  There will be a live multicast of BBC
Radio 5 Live, for part of Saturday, 6th September, 1997, with coverage
of the funeral of Diana, Princess of Wales, in London.

Please see the announcement in sd(r).

-- 
gordon.joly@bbc.co.uk gjoly@acm.org http://www.bbc.co.uk/
BBC White City, Room G374, 201 Wood Lane, LONDON W12 7TS.




From rem-conf Fri Sep 05 14:55:03 1997 
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	MBone Broadcast Announcement
	----------------------------

Title:       
	Pittsburgh Supercomputing Center Workshop
Date:        
	Sep 08, 1997

Time:        
	09:30 GMT 9 hours

Contact:     
	mmadrid@psc.edu

URL:         
	http://www.psc.edu/training/J90_Sept_97/welcome.html

Description:        
	 The Pittsburgh Supercomputing Center  will be broadcasting its CRAY J90/C90 workshop,  September 8-11. Agenda and lecture notes can  be found at   http://www.psc.edu/training/J90_Sept_97/welcome.html  









mbone broadcast schedule http://www.msri.org/mbone



From rem-conf Fri Sep 05 15:10:10 1997 
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Message-ID: <340FDAB7.1887@netlink.co.uk>
Date: Fri, 05 Sep 1997 11:11:05 +0100
From: Dennis Johnstone <dennisj@netlink.co.uk>
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To: Gordon Joly <gordon.joly@bbc.co.uk>, dominik@tecc.co.uk, brandon@bbc.co.uk,
        Simon Lockhart <simonl@rd.bbc.co.uk>, tim.locke@bbc.co.uk,
        mbone@isi.edu, rem-conf@es.net
Subject: Re: BBC Radio 5 Live multicast - Saturday 6th September 1997
References: <199709052043.AA20774@www0.reith.bbc.co.uk>
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Gordon Joly wrote:
>There will be a live multicast of BBC
> Radio 5 Live, for part of Saturday, 6th September, >1997

Actually we're live from 2300 on Friday (right now actually!) until 1300
on Sunday. I'd say that's more than part of Saturday... The mirrors kick
in between 0830 and 0930, and then drop out at 1700.

Dennis Johnstone



From rem-conf Fri Sep 05 21:16:40 1997 
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Message-ID: <3410D76B.2CFC3A59@nc.u-tokyo.ac.jp>
Date: Sat, 06 Sep 1997 13:09:15 +0900
From: Shingo Ichii <ichii@nc.u-tokyo.ac.jp>
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To: jdrosen@dnrc.bell-labs.com
CC: rem-conf@es.net
Subject: Re: MPEG Boards
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Jonathan D. Rosenberg wrote:

> Does anyone know of any good MPEG2 encoder/decoder boards for PC's (Win
> NT) that come with software that allow me to send the compressed video
> over a network? A board with an NTSC output to display the decompressed
> video on a TV would also be a big plus.

Though I don't know if they export, NTT Electronics (www.nel.co.jp) makes
a signle-board (PCI) MPEG2 real-time encoder.=81@=81@ As long as I could =
see, the
quality is very good, but the price is high (~ 4Myen =3D $33K).

Shingo Ichii
Computer Center, University of Tokyo
=81@




From rem-conf Sun Sep 07 21:21:28 1997 
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	id 0x7v3p-0003fG-00; Sun, 7 Sep 1997 21:04:25 -0700
Message-Id: <01BCBBE9.47F146E0.paul@vela.com>
From: Paul Mears <paul@vela.com>
Reply-To: "paul@vela.com" <paul@vela.com>
To: "'ari@sprintlabs.com'" <ari@sprintlabs.com>
Cc: "'jdrosen@dnrc.bell-labs.com'" <jdrosen@dnrc.bell-labs.com>,
        "'rem-conf@es.net'" <rem-conf@es.net>
Subject: Don't forget about Vela Research
Date: Sun, 7 Sep 1997 16:51:27 -0400
Organization: Vela Research
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> From:	Ari Ollikainen <ari@sprintlabs.com>
> To:	jdrosen@dnrc.bell-labs.com
> Cc:	rem-conf@es.net
> Subject:	Re: MPEG Boards
> Date:	Friday, September 05, 1997 3:24 PM
> 
> jdrosen@dnrc.bell-labs.com wrote:
> >avt'ers,
> >
> >Does anyone know of any good MPEG2 encoder/decoder boards for PC's (Win
> >NT) that come with software that allow me to send the compressed video
> >over a network? A board with an NTSC output to display the decompressed
> >video on a TV would also be a big plus.
> 
> 
>         You could just go and by the IBM 8300 V(ideo)A(ccess)Node for
>         about ~$40K and have a complete, integrated (encoder + decoder+
>         ATM25/155 interface + software+ development tools) solution all
>         packaged in a 200Mhz Pentium running OS/2!
> 
>         Alternatively:
> 
>         Check out the "The MPEGXpress 2000...a real-time MPEG-2 encoding
and
>         playback solution. Similar configurations are available for
>         Windows NT and for PCI-based Power Macintosh..."
> 
>         more info at: http://www.mpegexpress.com/2000specs.html
> 
>         Wired Inc makes a encoder (PCI) board, ButaneII, which is
available
>         by special order for ~$5K. They also make PCI decoder boards
>         that are around ~$.5K.
> 
>         more info at: http://www.wiredinc.com
> 
>         InnovaCom has an encoder board (PCI), DV2100/2110, that's ~$10K.
> 
>         more info at: http:www.innovacom-mpeg2.com
> 
>         Expect these prices to plummet by 1Q/2Q98 when single-chip
>         encoders will become available to build single-board codecs...
> 
> 
> 
> 
>            _/_/   _/_/_/_/    _/  Ari Ollikainen <ari@sprintlabs.com>
>         _/  _/   _/     _/   _/ Sprint Advanced Technology Laboratories
>      _/_/_/_/   _/_/_/_/    _/ Switching and Interworking Development
>    _/     _/   _/     _/   _/ 1 Adrian Court (M/S: CABURS0102)
>  _/      _/   _/       _/ _/ Burlingame, CA 94010

Vela Research also designs and sells MPEG-2 encoders and decoders.
Check out our web page at http://www.vela.com/ for our extensive set
of products based upon C-Cube and IBM encoder chip sets, and LSI
Logic and IBM decoder chip sets.

Drop me an email or call if you have questions.

Paul...
 




From rem-conf Mon Sep 08 08:26:49 1997 
From rem-conf-request@es.net Mon Sep 08 08:26:48 1997
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Message-Id: <m0x85aX-001BjCC@csc.com>
Comments: Authenticated sender is <ftroy@explorer.csc.com>
From: "Frank P. Troy" <ftroy@es.net>
To: rem-conf@es.net
Date: Mon, 8 Sep 1997 11:15:27 +0000
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Subject: Mrouted
Reply-to: ftroy@csc.com
Priority: normal
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Hi all

Is there a white paper published somewhere that explains (in detail) 
how Mrouted works?

Thanks
--------------------------------------------
Frank P. Troy          Phone:   703-471-3694
Sr. Network Engineer   Fax:     703-471-1575
AT&T DISC Development  Page:    888-808-0593 
3001 Centreville Rd.   e-mail:  ftroy@csc.com    
Herndon, VA 20171        



From rem-conf Tue Sep 09 08:16:29 1997 
From rem-conf-request@es.net Tue Sep 09 08:16:28 1997
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	id 0x8Rn8-0002QD-00; Tue, 9 Sep 1997 08:01:22 -0700
Sender: helbig@pfa.research.philips.com
Message-Id: <341566D0.1826@pfa.research.philips.com>
Date: Tue, 09 Sep 1997 17:10:08 +0200
From: "Dr. Tobias Helbig" <helbig@pfa.research.philips.com>
Organization: Philips Research Labs Aachen
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Subject: Looking for video conferencing system
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Hi there,
 
I'm looking for a video conferencing system to be installed at our 
site based on Win 95 and/or Win NT PCs. It shall support the feature 
of redirecting on-going video conferencing calls from one room to
another (e.g., when the person is moving to another room).
 
Can anybody help me locating a product or stable prototype that
(a) supports the feature of redirecting on-going calls?
(b) enables to control this feature by external software (i.e., by 
    our code which detects when a person is moving from one room to
    another)?
 
I appreciate any reply to my address given below.
 
Thanks a lot.
 
Tobias Helbig

-- 

 ----------------------------------------------------------------------
| Dr. Tobias Helbig            | Philips GmbH Research Laboratories    |
| Research Scientist           | Weisshausstr. 2                       |
|                              | D-52066 Aachen, Germany               |
|                              | Phone +49-241-6003-515 / Fax -518     |
|                              | Email helbig@pfa.research.philips.com |
 ----------------------------------------------------------------------



From rem-conf Tue Sep 09 19:17:06 1997 
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Date: Tue, 9 Sep 1997 19:08:42 -0700 (PDT)
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To: rem-conf@es.net
From: <jyluke@ew.mimos.my>
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Subject: Multimedia Asia 97 Conference and Plenary Sessions
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	MBone Broadcast Announcement
	----------------------------

Title:       
	Multimedia Asia 97 Conference and Plenary Sessions
Date:        
	Sep 16, 1997

Time:        
	08:00 GMT+8 10 hours

Contact:     
	jyluke@ew.mimos.my

URL:         
	http://www2.jaring.my/nitc/events/mma97/

Description:        
	Multimedia Asia 97 is an annual multimedia event  held in Malaysia for Malaysia's Multimedia Super   Corridor (MSC).  The event will bill held from Sept 16 to 19, 1997.  This broadcast will include key note address by  the Prime Minister of Malaysia and other selected   plenary sessions.









mbone broadcast schedule http://www.msri.org/mbone



From rem-conf Wed Sep 10 21:51:03 1997 
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Message-ID: <34177372.6628@fh-zwickau.de>
Date: Thu, 11 Sep 1997 06:28:34 +0200
From: Rainer Mueller <rainer.mueller@fh-zwickau.de>
Organization: HTW-Zwickau
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MBone Broadcast Announcement
        ----------------------------

Title:
        International ICEM Surf User Meeting 1997

Date:
        September 23-24, 1997

Time:
        10:30 MET; 2 hours

Contact org:    
        lutz.nagel@fh-zwickau.de
        icem.technologies@cdc.com

Contact tech:   
        rainer.mueller@fh-zwickau.de

URL:
        http://www.fh-zwickau.de/veranst/icem/icem.htm


Description:    
        The first two hours of the meeting will be broadcasted



-- 
Name: Rainer Mueller
Org: Westsaechsische Hochschule Zwickau (FH) / HRZ
Phone: +49 375 536-1203 
Fax  : +49 375 536-1202
E-Mail: rainer.mueller@fh-zwickau.de



From rem-conf Thu Sep 11 09:05:46 1997 
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To: rem-conf@es.net
From: <davidwu@bmrc.berkeley.edu>
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Subject: Virtual Prototyping of Mathematically Inspired Sculptures
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	MBone Broadcast Announcement
	----------------------------

Title:       
	Virtual Prototyping of Mathematically Inspired Sculptures
Date:        
	Sep 17, 1997

Time:        
	12:30 PST8PDT 2 hours

Contact:     
	davidwu@bmrc.berkeley.edu

URL:         
	http://bmrc.berkeley.edu/298/w4.html

Description:        
	A computer program has been developed to visualize different configurations  of abstract geometrical sculptures involving knots and saddle surfaces,  and to optimize the parameters that define the final shape.









mbone broadcast schedule http://www.msri.org/mbone



From rem-conf Thu Sep 11 10:22:11 1997 
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	MBone Broadcast Announcement
	----------------------------

Title:       
	Internet2 Campus networking Workshop
Date:        
	Sep 15, 1997

Time:        
	08:30 PST8PDT 3.5 hours

Contact:     
	rdhobby@ucdavis.edu

URL:         
	http://ansa.ucdavis.edu/cenic/ws_agenda.html

Description:        
	The Workshop will discuss campus networking issues  in implementing Internet2 functionality.  See  http://www.internet2.edu for more details on the  Internet2 Project.









mbone broadcast schedule http://www.msri.org/mbone



From rem-conf Thu Sep 11 10:22:11 1997 
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	MBone Broadcast Announcement
	----------------------------

Title:       
	Internet2 Campus networking Workshop
Date:        
	Sep 16, 1997

Time:        
	08:30 PST8PDT 6.5 hours

Contact:     
	rdhobby@ucdavis.edu

URL:         
	http://ansa.ucdavis.edu/cenic/ws_agenda.html

Description:        
	The Workshop will discuss campus networking issues  in implementing Internet2 functionality.  See  http://www.internet2.edu for more details on the  Internet2 Project.









mbone broadcast schedule http://www.msri.org/mbone



From rem-conf Thu Sep 11 12:05:16 1997 
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Subject: Mathematics of Games and Sports by Joseph B. Keller, Stanford University 
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	MBone Broadcast Announcement
	----------------------------

Title:       
	Mathematics of Games and Sports by Joseph B. Keller, Stanford University 
Date:        
	Sep 15, 1997

Time:        
	12:00 GMT+8 1 hours

Contact:     
	mbone@msri.org 

URL:         
	http://www.msri.org/lecturenotes/97/SIAM/keller/

Description:        
	The speaker will present a mathematician's perspective on  some games and sports, and answer questions like these:     How many shuffles to mix a deck? When is a team  eliminated? What is the probability of heads? How should  teams be ranked? What is the probability of a shutout?  How do world records vary with time? How should a  runner vary his speed in a race? Does it pay to exercise?  









mbone broadcast schedule http://www.msri.org/mbone



From rem-conf Thu Sep 11 12:19:22 1997 
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To: rem-conf@es.net
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Subject:  The Baleful Effect of Computer Languages and Benchmarks Upon Applied Mathematics, Physics and Chemistry
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	MBone Broadcast Announcement
	----------------------------

Title:       
	 The Baleful Effect of Computer Languages and Benchmarks Upon Applied Mathematics, Physics and Chemistry
Date:        
	Sep 15, 1997

Time:        
	12:00 GMT+8 1 hours

Contact:     
	mbone@msri.org 

URL:         
	http://www.msri.org/lecturenotes/97/SIAM/kahan/

Description:        
	 









mbone broadcast schedule http://www.msri.org/mbone



From rem-conf Thu Sep 11 12:23:40 1997 
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From: <mbone@msri.org>
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Subject: Multiresolution Algorithms in Computer Graphics by Peter Schroder, California Institute of Technology   
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	MBone Broadcast Announcement
	----------------------------

Title:       
	Multiresolution Algorithms in Computer Graphics by Peter Schroder, California Institute of Technology   
Date:        
	Sep 16, 1997

Time:        
	12:00 GMT+8 1 hours

Contact:     
	mbone@msri.org 

URL:         
	http://msri.org/lecturenotes/97/SIAM/schroder/

Description:        
	Computational techniques based on wavelets and more  generally multiresolution approaches have made  tremendous inroads into computer graphics applications.  Examples include illumination computations, hierarchical  mesh editing, geometric modeling, and 3D scanning. This  development is motivated by the twin challenges of  computer graphics: complexity and interactivity. In order  to generate realistic images of objects and creatures with  realistic motion, accurate simulations need to be  performed for ever more complicated geometry. At the  same time fast updates are necessary for interactive  environments. Multiresolution as a general tool provides  representations and algorithms which scale well to large  problem sizes, are very general and robust, and have  modest storage costs.     The speaker will give an overview of these applications  and algorithms and discuss some of the issues that arise  when attempting to take classical constructions into the  messiness of real !
!
applications.  









mbone broadcast schedule http://www.msri.org/mbone



From rem-conf Thu Sep 11 13:09:53 1997 
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To: rem-conf@es.net
From: <mbone@msri.org>
Reply-to: mbone@msri.org
Subject: High Performance Computer Architecture by John L. Hennessy, Stanford University
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	MBone Broadcast Announcement
	----------------------------

Title:       
	High Performance Computer Architecture by John L. Hennessy, Stanford University
Date:        
	Sep 16, 1997

Time:        
	13:00 GMT+8 1 hours

Contact:     
	mbone@msri.org 

URL:         
	http://www.msri.org/lecturenotes/97/SIAM/hennessy/

Description:        
	Recent directions in high performance computers for  scientific computing have been driven by two key  factors: the advantages of employing commodity  technology as building blocks and the understanding of  how to build scalable multiprocessors that efficiently  support shared-memory programming. The first  capability can lead to high performance computers that  are cost competitive with workstations and servers, while  the second capability allows high performance machines  to be software compatible with small and mid-range  computers that uniformly use shared-memory  multiprocessing. While these trends may increase the  usability and generality of high performance computing,  other technology trends will require users to be more  aware of issues such as locality of access and data  distribution. The challenge in the future will be in  creating hardware and software that helps the user to  achieve high performance without undue programming  effort.    









mbone broadcast schedule http://www.msri.org/mbone



From rem-conf Thu Sep 11 13:09:57 1997 
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Reply-to: mbone@msri.org
Subject: 35 Years of (Linear) Probing, by Donald E. Knuth, Professor Emeritus, Stanford University
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	MBone Broadcast Announcement
	----------------------------

Title:       
	35 Years of (Linear) Probing, by Donald E. Knuth, Professor Emeritus, Stanford University
Date:        
	Sep 17, 1997

Time:        
	12:00 GMT+8 1 hours

Contact:     
	mbone@msri.org 

URL:         
	http://www.msri.org/lecturenotes/97/SIAM/knuth/

Description:        
	  Many computer programs access large tables of data by  using a classical variant of hashing called "linear probing,"  also known to children as the game of "musical chairs."  When the author first studied the characteristics of this  simple method in 1962, he came to understand that the  mathematical analysis of algorithms was a rich subject  suitable for a lifetime of study. This talk surveys the  mathematical analysis of algorithms by focussing on  advances that have been in the analysis of linear probing  from 1962 to 1997, and by noting surprising relations  between this algorithm and other important algorithms and  combinatorial problems, including the study of random  graphs.    









mbone broadcast schedule http://www.msri.org/mbone



From rem-conf Thu Sep 11 13:12:58 1997 
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	MBone Broadcast Announcement
	----------------------------

Title:       
	Clusters and Massively Parallel Computers: Are the Architectures Converging? by Paul C. Messina, California Institute of Technology
Date:        
	Sep 17, 1997

Time:        
	13:00 GMT+8 1 hours

Contact:     
	mbone@msri.org 

URL:         
	http://www.msri.org/lecturenotes/97/SIAM/messina/

Description:        
	    Clusters of powerful workstations or PCs connected by  various network technologies have become a popular,  inexpensive computing resource. Massively parallel  computers are still considered to be the right tool for the  biggest scientific and engineering computations.  However, from an application standpoint, the difference  between the two architectural options seem to be getting  less and less obvious. The cluster approach can scale to  very large configurations effectively, although perhaps  not to equal the level of the very biggest MPPs. Perhaps  the most important issues have to do with how the  system is managed (as one powerful system or as many  workstations), and on the entire system environment,  including the type of job scheduling, system software,  and software tools, and the availability of mass storage  and archival file systems.     This talk will explore these issues and present data from  a variety of applications and configurations in use today.     









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From rem-conf Thu Sep 11 13:18:24 1997 
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	MBone Broadcast Announcement
	----------------------------

Title:       
	IP10 Impact of the Internet on Scientific Computing by Eric Grosse, Bell Laboratories, Lucent Technologies
Date:        
	Sep 18, 1997

Time:        
	12:00 GMT+8 1 hours

Contact:     
	mbone@msri.org 

URL:         
	http://www.msri.org/lecturenotes/97/SIAM/grosse/

Description:        
	    The scientific computing community has long used the  Internet for communication of email, software, and papers.  SIAM's online journal server this year is a welcome further  step. But, with the notable early exception of the Macsyma  server at MIT in the 70s, there has been little use of the  Internet for actual computation. In the past few years, we  have again started to see experimental computational  servers appearing. What are their prospects for success?  What is needed at the client end? What languages and  protocols should be used for maximum interoperability and  efficiency? How can we structure our applications to allow  flexible adaptation to clients and channels of different  strengths?             Slides are available for this talk in pdf format   http:http://cm.bell-labs.com/who/ehg/slides/siam97/talk.pdf      









mbone broadcast schedule http://www.msri.org/mbone



From rem-conf Thu Sep 11 13:22:05 1997 
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	MBone Broadcast Announcement
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Title:       
	Past-President's Address: The Pursuit of Optimality: From the Big Picture to the Gory Details by Margaret H. Wright, Bell Laboratories
Date:        
	Sep 18, 1997

Time:        
	13:00 GMT+8 1 hours

Contact:     
	mbone@msri.org 

URL:         
	http://www.msri.org/lecturenotes/97/SIAM/wright/

Description:        
	A desire to optimize---to do the best thing---is ubiquitous  throughout mathematics, computing, science, and  engineering (not to mention everyday life). Research in  optimization comes in forms ranging from mathematical  theory to practical rules of thumb, from general-purpose  methods to application-specific heuristics. This talk will be a  melange of topics involving recent work in non-derivative  optimization and interior methods, chosen by the speaker to  convey an overview of selected themes in optimization today.  The punch line will be the joy of seeking, and finding,  underlying commonalities and substantive distinctions.      









mbone broadcast schedule http://www.msri.org/mbone



From rem-conf Thu Sep 11 14:54:18 1997 
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Date: Thu, 11 Sep 1997 17:51:55 -0400
From: Debanjan Saha <debanjan@watson.ibm.com>
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To: end2end-interest@ISI.EDU, tccc@cs.umass.edu, ietf@ietf.org,
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Subject: JHSN special issue on Multimedia Network (CFP)
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Please feel free to circulate the CFP by any means you deem
appropriate.  Also, please excuse any multiple copies of this 
CFP you may receive due to your memberships in multiple mailing 
lists.

Thanks and regards,
Debanjan

                            CALL FOR PAPERS
----------------------------------------------------------------------------
                     JOURNAL OF HIGH SPEED NETWORKS 
                            Special Issue On
                 	 Multimedia Networking 
-----------------------------------------------------------------------------

The rapid proliferation of multimedia applications has severely
strained an already overloaded networking infrastructure. In order 
to foster an environment for ubiquitous deployment of these
applications, it is imperative that we design and implement the next 
generation of network protocols and services  that provide scalable
end-to-end  support to networked multimedia applications. The purpose 
of this journal issue is to report on the latest  technological 
advances that will enable the development of such an infrastructure. 
The journal is looking for contributions in the following areas:

	o RSVP and integrated services in the Internet
	o Queue management in routers and switches
	o QoS signaling and routing in IP and ATM networks
 	o Audio/Video streaming, push and pull technologies
        o Multicasting and media scaling, rate control
	o Network conscious/adaptive applications and protocols
	o Application level framing, customizable protocol features
	o New applications, experimental protocols and systems 
	o Multimedia over cable modems/xDSL, wireless/satellite links
	o Traffic analysis, performance modeling and evaluation



Important Dates:
	Paper submission deadline:  	October 31, 1997.
	First Round of Reviews:		February 15, 1998
	Acceptance Notification:	March 30, 1998
	Expected Publication Date:	Summer, 1998
	

Please submit  4 copies of your paper to the editors of the special issue:

	Satish K. Tripathi			Debanjan Saha
	Dean and Johnson Chair Professor	Research Staff Member
	Bourns College of Engineering	        Room# H3-D32	
	University of California		IBM T.J. Watson Research Center
	Riverside, CA 92521-0425		Hawthorne, NY 10532

	Phone: (909) 787-6374			Phone: (914) 784-7194
	Fax:   (909) 787-3188			Fax:   (914) 784-6205
	Email: tripathi@engr.ucr.edu		Email: debanjan@watson.ibm.com
						http://www.research.ibm.com/people/d/debanjan
	



From rem-conf Fri Sep 12 01:44:42 1997 
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From: petri.koskelainen@research.nokia.com (Koskelainen Petri NRC/Tre)
To: confctrl@isi.edu ('confctrl@isi.edu'), rem-conf@es.net ('rem-conf@es.net')
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Date: Fri, 12 Sep 1997 11:36:09 +0200
Subject: SDP format for H.263 options
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Hi,

Currently it is not possible to specify any codec specific options
in SIP/SDP architecture. For example, H.263 has many options and
parameters which are almost mandatory (it makes no sense to say
"lets use H.263" without knowledge of its capabilities).

Best place to include these codec specific parameters in SDP is:
a=fmtp:<format> <format specific parameters>

So, these format speficic parameters must be defined somewhere
and it seems (like Henning proposed) that the best place to do it,
is in corresponding RTP mapping draft as an annex.
Another solution is to make new internet-draft (for every codec).

Below is a proposal for H.263 SDP format specific parameter mapping,
to be included into "RTP Payload Format for H.263 Video Streams" draft.

This proposal assumes:
(needs to be said also in the SIP draft)

In the case of SIP, these codec options mean decoder capabilities.
The idea is, that when making an INVITE, it includes the codec (H.263) and
those codec options (e.g. SAC, PB-frames) which are decodeable by
this decoder.


Example excerpt from SIP INVITE message:
...
m=video 55514 RTP/AVP 34
a=fmtp 34 CIF=4 QCIF=2/MaxBitRate=1000/SAC/AP

Syntax above means that H.263 codec with preferred picture size of CIF (with 
min picture interval of 4) and secondary picture size is QCIF and arithmetic 
coding and advance prediction are in use and max bitrate decoable by decoder 
is 100 kbit/s.

If caller does not want to receive all options (perhaps because
his pc is a slow 486) then he just tells what options he wishes to receive.

Other end, may (or may not) use them in the encoder.


Any comments about mapping or the right place to define these ?

 --------------------------------------- clip clip----------------------


                              Harri Honko
                              Petri Koskelainen
                              Jouni Salonen
                              Nokia Research Center


ANNEX A  (in "RTP payload format for H.263 video")

SDP syntax for H.263 video

This annex defines the SDP [2] syntax for H.263 codec options
and parameters. It is often useful to know beforehand (in call setup phase)
what features of H.263 the other end supports. This annex defines
the <format spefic parameters> for H.263, which exists in
a=fmtp:<format> <format specific parameters> as defined in the SDP document.
Actual usage and negotiation of attributes is specified in SIP [3] and SAP 
[4] documents. These options are especially useful in SIP.  In SAP it is not 
clear whether they have any use.
In SIP, these format specific parameters are decoder properties, and in SAP 
they are encoder capabilities.  The SAP rules are applied also if a 
multicast session is advertized in a web page in SDP format.

Following text assumes that SIP is used. In the end of this annex, there is 
a short chapter which discusses how these options and parameters are used 
with SAP.

A.1 H.263 Codec Options
Following words describe 4 options defined for H.263. Later versions
of H.263 (e.g. so called H.263+, currently a ITU-T draft) defines more 
options and they should be specified later for SDP. These options are 
separated by single slash ("/") between them.

For further description about these parameters, please refer to
ITU documents [1].

Word: URV
Explanation: UnRestricted motion Vector option.

Word: SAC
Explanation: Syntax based Arithmetic Coding.

Word: AP
Explanation: Advanced Prediction.

Word: PB
Explanation: PB Frames.

These words exist only if the sender of this SDP message is able or willing 
to decode those. E.g. If a terminal is capable of decoding SAC and AP 
options, it can put SAC/AP in the beginning of <format specific parameters>. 
Then the other party knows it, and can use those options in its encoder.



A.2 Other H.263 codec parameters

This chapter describes parameters in H.263 to be used with SIP

Supported picture sizes and their corresponding minimum picture interval 
(MPI) information can be combined. All picture sizes can be advertised to 
other party, or only some subset of it.  Terminal announces only those 
picture sizes (with their MPIs) which it is willing to receive.
MPI is an integer value (1..32) and it means that maximum picture (frame) 
rate is  29.97/MPI value/s. MPI value may be bigger, meaning that picture 
rate is slower. For example, MPI=2 means that maximum (decodeable) picture 
rate per sec is about 15.

Parameter occurring first is the preferred picture mode to be received, and 
last is the least preferred (but still supported) one. Following words are 
presented with no slash ("/") symbol between them, and are separated by 
space.

Following words are present in SDP list only if a terminal is willing to 
decode the picture size. If terminal is not willing to receive some size, it 
is not present in the list.

Word: SQCIF=MPI(1..32)
Explanation: SQCIF picture size and its MPI value. Means that sender can 
decode SQCIF resolution with this MPI value.

Word: QCIF=MPI(1..32)
Explanation:  QCIF picture size and its MPI value. Means that sender can 
decode QCIF resolution with this MPI value.

Word: CIF=MPI(1..32)
Explanation: CIF picture size and its MPI value. If present, means that 
sender can decode CIF  resolution with this MPI value.

Word: CIF4=MPI(1..32)
Explanation: CIF4 picture size and its MPI value. If present, means that 
sender can decode this resolution with this MPI value.

Word: CIF16=MPI(1..32)
Explanation: CIF16 picture size and its MPI value. If present, means that 
sender can decode this resolution with this MPI value.

Word: XMAX=maximum_x_size YMAX=maximum_y_size MPI=MPI(1..32)
This list of  words means that terminal is capable and willing to receive 
arbitrary picture sizes. These X and Y values are the maximum of allowed 
picture sizes with corresponding MPI value. All these three words must exist 
together in this order, or none is allowed to exist. Both picture sizes must 
be divisible by 4.

Example of the usage of these words:

CIF=4  QCIF=3 SQCIF=2 XMAX=360 YMAX=240 MPI=2

This means that sender hopes to receive CIF picture size, which it can
decode at MPI=4. If that is not possible, then QCIF with MPI value 3, if 
that is neither possible, then SQCIF with MPI value =2.  It is also allowed 
(but least preferred) to send arbitrary picture sizes (max 360x240) with 
MPI=2.
Note that most encoders support at least QCIF and CIF fixed resolutions and 
they are expected to be available almost in every H.263-based video 
application.


Following two parameters are used only with SIP.

MaxBitRate=INTEGER(1..19200)
Explanation: maximum video stream bitrate receivable by the sending 
terminals H.263 decoder, presented at 100 bits/s.

BitsPerPictureMaxKb=INTEGER(0..65536)
Explanation: Maximum amount of kilobits allowed to represent a single 
picture frame, value is specified by largest supported picture resolution, 
see [1]. If  this parameter is not present, then default value, that is 
based on the maximum supported resolution, is used.


All these parameters must be presented in SDP a=fmtp line exactly in this 
order if they are present.


Example: (in SIP)
a=fmtp 34 CIF=4 QCIF=2/MaxBitRate=1000/SAC/AP

This means that the sender of this message can decode H.263 (RTP payload 
type 34) bitstream with following options and parameters:
Preferred resolution is CIF (its MPI is 4), but if that is not possible then
QCIF size is ok. Maximum receivable bitrate is 100 kbit/s (1000*100 bit/s) 
and SAC and AP options can be used.



A.3 Use of SDP H.263 options and parameters with SAP

SAP announcements are one-way only. H.263 options (like SAC) mean that 
sending terminal (host) is going to use these options in its transmitted 
H.263 stream. It is just an informal message.

Usually only one picture size (with its MPI) exists. However, since it is 
possible for a video source (terminal) to change its picture size during 
session, several picture sizes can exist in the parameter list. First one is 
the original picture size to be used in the beginning of the session.
Other H.263 parameters are not used with SAP (nor when announced in 
web-page).


Example (with SAP):
a=fmtp 34 CIF=2 URV/SAC

The video source is sending an H.263 bitstream and picture size is CIF, 
MPI=2 and URV and SAC options are used. This kind of announcement can be 
used, e.g., in the MBone.


Authors' Addresses

Harri Honko, Petri Koskelainen, Jouni Salonen
Nokia Research Center
P.O.Box 100
FIN-33721 Tampere
Finland
e-mail:
harri.honko@research.nokia.com,
petri koskelainen@research.nokia.com,
jouni.salonen@research.nokia.com



 References:

 [1]  International Telecommunication Union.
 Video Coding for Low Bitrate Communication, ITU-T Recommendation  H.263, 
1996

 [2]  Handley et al, ``SDP - Session Description Protocol'', INTERNET-DRAFT, 
 IETF 1997.

 [3]  Handley et al, ``SIP - Session Initiation  Protocol'',
 INTERNET-DRAFT, IETF 1997.

[4]  ``SAP - Session Announcement  Protocol'',
 INTERNET-DRAFT, IETF 1997.







From rem-conf Sun Sep 14 00:15:59 1997 
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From rem-conf Mon Sep 15 05:41:08 1997 
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	id 0xAZV4-0000N8-00; Mon, 15 Sep 1997 04:39:30 -0700
Date: Mon, 15 Sep 1997 13:39:00 +0200 (MET DST)
Message-Id: <199709151139.NAA06521@chouette.inria.fr>
From: Emmanuel Duros <Emmanuel.Duros@sophia.inria.fr>
To: udlr@sophia.inria.fr
CC: rem-conf@es.net, mbone@ISI.EDU, end2end-interest@ISI.EDU,
        merci@cs.ucl.ac.uk, dabbous@chouette.inria.fr, bolot@chouette.inria.fr,
        cdiot@chouette.inria.fr, flyonnet@chouette.inria.fr,
        sfosse@chouette.inria.fr
Subject: SIGCOMM 97 - Mbone event
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The following SIGCOMM events will be retransmitted on-line on the Mbone:

Sigcomm award talk by Jon Postel
Tuesday September 16 from 9:00 am to 10:30 am GMT+2

Sigcomm award talk by Louis Pouzin
Wednesday September 17 from 9:00 am to 10:30 am GMT+2

These two talks will be multicast from INRIA using a satellite link
offered by EUTELSAT. UCL (London, England) and GMD (Darmstadt, Germany)
are equipped with reception hardware and will forward the sessions to the
rest of the MBone.


Technical note :
----------------
The sending tool will be the integrated audio and video MBone tool
Rendez-Vous (http://www.inria.fr/rodeo/rv). Rendez-Vous is vic and vat
compatible. The sdr session entry is SIGCOMM '97.

High quality audio will also be broadcast separately on its own channel
(see SIGCOMM '97 HQ Audio) with the FreePhone audio MBone tool
(http://www.inria.fr/rodeo/fphone). This HQ transmission will only be
available to FreePhone users.

Please use wb for transmission feedback.

For Rendez-Vous users : as sd/sdr do not take into account integrated
audio and video tools, Rendez-Vous users will have to launch it manually
by typing the following command :

./ivstng.sh 224.6.224.2 31034 224.8.224.4 51488 255


Emmanuel



From rem-conf Mon Sep 15 19:00:30 1997 
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	id 0xAmnG-0001Bf-00; Mon, 15 Sep 1997 18:51:10 -0700
Date:    Mon, 15 Sep 97 18:51 PDT
To:      mbone@ISI.EDU,rem-conf@ES.NET
From:    Denis DeLaRoca                       <CSP1DWD@MVS.OAC.UCLA.EDU>
Subject: SDR 2.4a6 calendar bug: syntax err in expression "09/3"
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The calendar of sdr 2.4a6 is failing with the above syntax error --
most likely a bad session announcement. Any workaround for this?

-- Denis




From rem-conf Tue Sep 16 17:06:51 1997 
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Date: Tue, 16 Sep 1997 16:51:15 -0700 ()
From: Stephen Casner <casner@precept.com>
To: Koskelainen Petri NRC/Tre <petri.koskelainen@research.nokia.com>
cc: "'confctrl@isi.edu'" <confctrl@isi.edu>,
        "'rem-conf@es.net'" <rem-conf@es.net>
Subject: Re: SDP format for H.263 options
In-Reply-To: <1997Sep12.112945.1751.1063168@nrchub01he.research.nokia.com>
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On Fri, 12 Sep 1997, Koskelainen Petri NRC/Tre wrote:

> Currently it is not possible to specify any codec specific options
> in SIP/SDP architecture. For example, H.263 has many options and
> parameters which are almost mandatory (it makes no sense to say
> "lets use H.263" without knowledge of its capabilities).
> 
> Best place to include these codec specific parameters in SDP is:
> a=fmtp:<format> <format specific parameters>

I agree that this is the right place to convey additional H.263
parameters for a session.  I presume that these parameter are
instructions to the encoder; that is, except for performance or
implementation constraints, the decoder can tell from the headers what
functions are in use and can decode can decode the stream without
knowing these option settings.

> So, these format speficic parameters must be defined somewhere
> and it seems (like Henning proposed) that the best place to do it,
> is in corresponding RTP mapping draft as an annex.
> Another solution is to make new internet-draft (for every codec).

Not sure what you mean by "mapping draft".  The only example use of
a=fmtp I know of so far is in the Redundant Audio payload format,
recently published as RFC 2198.

> Below is a proposal for H.263 SDP format specific parameter mapping,
> to be included into "RTP Payload Format for H.263 Video Streams" draft.

Since the H.263 payload format spec has also been published now as RFC
2190, it would be necessary to produce a revised RFC to add the format
mapping info.  The way to start that process would be to write an
Internet-Draft specifying the details of the format parameters.  That
could be published as a separate RFC, but more likely it would be
better to publish a new combined RFC.  Since the additional
information would be specifying part of the standard, it would
probably not be an annex or appendix but rather an additional section
of the document.

> m=video 55514 RTP/AVP 34
> a=fmtp 34 CIF=4 QCIF=2/MaxBitRate=1000/SAC/AP
> 
> Syntax above means that H.263 codec with preferred picture size of CIF (with 
> min picture interval of 4) and secondary picture size is QCIF and arithmetic 
> coding and advance prediction are in use and max bitrate decoable by decoder 
> is 100 kbit/s.

There are already SDP fields to specify maximum bandwidth, so I
suspect that should not be included as a codec parameter.  I don't
know enough about the options in H.263 to say whether what you have
proposed is appropriate/sufficient/etc.  Those who know more about
H.263 should chime in.

One simple comment on the text structure is that I think it would be
better to define the options independent of the protocol in which they
would be used (SIP, SAP or something else) and then state the
applicability to particular scenarios.

The SIP spec should address the capability negotiation (or capability
statement if negotiation does not really occur) in an
encoding-independent manner.
							-- Steve




From rem-conf Wed Sep 17 10:51:42 1997 
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	MBone Broadcast Announcement
	----------------------------

Title:       
	Disney's Aladdin: First Steps Towards Storytelling in Virtual Reality
Date:        
	Sep 24, 1997

Time:        
	12:30 PST8PDT 2 hours

Contact:     
	davidwu@bmrc.berkeley.edu

URL:         
	http://bmrc.berkeley.edu/298/w5.html

Description:        
	Walt Disney Imagineering has developed a high-fidelity virtual reality (VR) attraction where guest fly a magic carpet through a virtual world based on the animated film "Aladdin."  









mbone broadcast schedule http://www.msri.org/mbone



From rem-conf Thu Sep 18 14:39:24 1997 
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	MBone Broadcast Announcement
	----------------------------

Title:       
	BayLISA: Sendmail and DNS
Date:        
	Sep 18, 1997

Time:        
	19:00 PST8PDT 2 hours

Contact:     
	blw@baylisa.org

URL:         
	http://www.baylisa.org

Description:        
	Hal Pomeranz speaks about setting up split DNS and  SMAP/Sendmail on an Internet firewall.









mbone broadcast schedule http://www.msri.org/mbone



From rem-conf Fri Sep 19 18:58:25 1997 
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Date: Fri, 19 Sep 1997 18:47:33 -0700
To: rem-conf@es.net, confctrl@isi.edu
From: Rob Lanphier <robla@prognet.com>
Subject: ASF Specification
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Hi folks,

I just wanted to solicit comments on the "Active Streaming Format (ASF)"
specification that Microsoft announced with the support of Progressive
Networks, Intel, Adobe, and Vivo, and a whole buncha other companies who
are interested in putting the file format compatibility problem to bed.

I'm sure most folks here are more interested in wire protocols, but having
a common file format that all tools can use to record to and servers can
use to playback from is a very handy thing, so I'd really encourage you to
look at it.  The folks at MS have put together a very good web site over at:
http://www.microsoft.com/asf

The spec, open issues, and instructions for joining the mailing list are
all on the MS web site (e-mail to Listserv@listserv.msn.com, no subject,
body: "subscribe ASF your name" (no quotes))

This is *really* on the fast track (period for public comment ends
September 30).  After this, tool and server vendors will plow ahead with
the current version, and the spec will be submitted to a standards body for
future versions (destination TBD).

Now's really the time to get your two cents in on this.  There's a lot of
really knowledgeable people on this list who could usefully contribute to
this spec, and we'd love to hear what y'all have to say.

Thanks
Rob

---
Rob Lanphier               Voice: (206)674-2322         Fax: (206)674-2699
Program Manager-Protocols                         Email: robla@prognet.com
Progressive Networks-Home of RealAudio            Web: http://www.real.com
For more information on firewalls:       http://www.real.com/help/firewall
For more information on RTSP:                     http://www.real.com/rtsp



From rem-conf Sat Sep 20 06:01:00 1997 
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Date: Sat, 20 Sep 1997 08:53:12 -0400
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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Attached are some of my quick observations on the current ASF draft.
-- 
Henning Schulzrinne        email: schulzrinne@cs.columbia.edu
Dept. of Computer Science  phone: +1 212 939-7042 (@Bell Labs: 908 949
8344)
Columbia University        fax:   +1 212 666-0140
New York, NY 10027         URL:   http://www.cs.columbia.edu/~hgs
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Comments on ASF Spec (version of Sept. 8, 1997)

These remarks are addressing the basic need of ASF to be efficient and
suitable for streaming data using the Internet standard or
to-be-standardized protocols for these purposes (RTP and RTSP,
respectively).

(x.y) refers to section x.y

(1.2) ASF is not clear:  it says it is a file format, but then says "it
supports data delivery over a wide variety of networks".  The latter
implies (or may be read to imply) that ASF would be transported in
real-time (streamed) across networks.  This seems unlikely and
inappropriate. Adding a disclaimer would avoid confusion, e.g., with
the role of RTP payload type definitions.

(1.3) Design Goals should list "operating system independence". Or am I
right in suspecting that this was not a goal :-)?

(1.3) Simplicity is also notably absent. The spec should at least
outline what objects a minimal implementation should understand. (See
SIP and RTSP specs for examples on this...)

(3.1) The GUID is not described in a level of detail that makes this
document self-contained.  The GUID does not solve any real problems: 
There is no way (at least not described within the document) that I can
find out who "owns" a particular GUID and who I might ask what this
object type is that I just found in an ASF file.  If there is a
registry, I don't need 16-byte GUIDs.  Internet MIME types, for example,
allow me to do this by consulting the IANA web site. Collisions are not
a real problem for file format objects, finding descriptions is.

The spec cannot be implemented as is, since the values of the GUIDs are
not provided. 

(3.1) Non-network byte order:  I suppose this explains the omission in
(1.3) :-) This is very inconvenient; all systems offering socket or
Winsock services have htonl() and ntohl() to allow to write byte-order
independent code, with one of these functions being a null op.  There
are no standard routines for the reverse-byte order suggested here.

(4.1) Major problem:  ASF assumes that a single stream has a single
clock and media type.  When streaming or recording RTP data, this is not
true.  The media type (RTP payload type) within a single stream may
change frequently, with each payload type having its own "natural"
sampling and clock frequency.  The number and type of these payload
types within a stream is not known ahead of time if I record data from a
live session (e.g., a conference).  Thus, for this reason and the
reasons listed below, ASF as it currently stands is useless for
efficient and general RTP recording and playback. 

The format assumes, by having a fixed number of stream property objects,
that the number of streams is known ahead of time. For recording
conferences, for example, this is not the case. To record a conference,
I would have to rewrite the whole data file after the conference has
concluded - very inefficient. This also makes it impossible to allow
simultaneous recording and playback, useful for sessions with "late
joiners".

(5) The separation of material in Sections 5 and 6 is only confusing.
The only information not contained in Section 5 is the data type and
field length (the former should imply the latter in any event). Thus, I
suggest moving the second column of the tables in Section 6 into Section
5. The first column in the tables in Section 5 is useless and should be
removed to avoid confusion. Having two sections simply means that I need
to flip back and forth; plus, there is the danger of misalignment
between the two.

The data types UINT, UINT8, FILETIME and others are not defined.  UINT
refers to datatypes ranging from 16 to 32 bits - why not have UINT16 and
UINT32?

No values are given for 'invalid' fields. According to Internet protocol
custom, should be zero for writing and ignored on reading.

(5.2): The time field contains no indication of the time zone to be used.

(5.3): The explanation of 'preroll' is confusing. What's a delay *factor*?
(A factor is a multiplier - what is being multiplied?) What does it mean
to 'start' a stream?

Extension System and Extension Data Size is not explained.

Recordable: recording by whom? Is caching ok - in the network, at the
end system?

(5.4) Mainy of the MARC fields are inappropriate for multimedia content
or not defined. What for, example, should 'physical description' stand
for - the size and color of the bits? This makes sense for books, not
for ASF. I have no idea what 'System Details' or 'Leader' should be in
this context.

Instead (or at least in addition), the Dublin core elements should be
used, as these are generic:
http://purl.org/metadata/dublin_core_elements
ftp://ds.internic.net/internet-drafts/draft-kunze-dc-01.txt

(6.2) The File Properties Object should contain the standard IETF
IntServ Tspec parameters.  From the parameters given, I cannot compute
this and thus not make an RSVP reservation.

(7.1) The ASF Packet Definition has, I believe, too many variants (two
packet length, three presentation time, offset, and size each, for a
total of 54 combinations), making it very difficult to overlay a
structure onto the data read from disk.  The space savings are marginal.
A common format (16 bit long size, with pres. time, no fragment) should
be defined. All variants should be listed and should have fixed-length
headers for ease and speed of implementation.

The 'clean point' should be mapped into RTP marker bits (possibly in an
'Use of RTP with ASF appendix'), if they are indeed equivalent.

Major problem: As is, the Packet Definition cannot be used for RTP
packets, since it does not contain a Payload type field. CSRC and SSRC
values cannot be stored. Since the size of these fields is not known,
the Stream Properties Object Extension Data size is not known. Since
only a single extension is specified, this would be "used up" for RTP
immediately, with no ability to add per-media extensions.

(9) This should refer and use the existing RTP payload definitions, as
these define channel orderings, byte order, etc. The data given in the
spec is not sufficient for interoperability. Trivial example: PCM can
mean mu-law or A-law and the channel ordering for multiple channels
needs to be specified. The RTP Profile (RFC 1890) does this at great
detail.

Other remarks:

- No provision to store RTCP SDES information for conferences and
streaming media. (See RFC 1889)

General nits:

- e.g. should always be followed by a comma.

- When referring to data types, the section # should be given for easy
reference.

- 9.1 -> typo: UNIT -> UINT

- It is, I believe, generally useful to add motivation to spec
decisions. (This seems to be, at least, the consensus within the
Internet RFC community. See "Guide to Standards Writers".)


--------------006277611CED184687564E48--




From rem-conf Mon Sep 22 02:50:00 1997 
From rem-conf-request@es.net Mon Sep 22 02:49:59 1997
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Message-ID: <34263A3F.5620B116@KEGO.COM>
Date: Mon, 22 Sep 1997 02:28:31 -0700
From: Cameron Elliott <cam@elliott.net>
Organization: KEGO - Software & Datacommunications
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Hello,

I'm wondering if anyone is continuing work on VIC?

I sent a message to the developers, but didn't receive
a reply.

Thanks,
Cameron Elliott






--
Cameron Elliott

Kego - Software & Datacommuncations
  Unix, Embedded Software, TCP/IP, H.323

888-FOR-KEGO
Cam@Kego.com
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From rem-conf Mon Sep 22 09:12:59 1997 
From rem-conf-request@es.net Mon Sep 22 09:12:59 1997
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To: Cameron Elliott <cam@elliott.net>
cc: rem-conf@es.net
Subject: Re: Vic work continuing? 
In-reply-to: Your message of "Mon, 22 Sep 1997 02:28:31 PDT."
             <34263A3F.5620B116@KEGO.COM> 
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In message <34263A3F.5620B116@KEGO.COM>, Cameron Elliott writes:
> This is a multi-part message in MIME format.
> --------------3AE62E6F399114FB77A5E4E3
> Content-Type: text/plain; charset=3Dus-ascii
> Content-Transfer-Encoding: 7bit
>=20
> Hello,
>=20
> I'm wondering if anyone is continuing work on VIC?
>=20
> I sent a message to the developers, but didn't receive
> a reply.
>=20
> Thanks,
> Cameron Elliott

Well, I guess that we are quite a few people eagerly awaiting the 2.8.1
(memory leak-free ?) release. Last rumor I heard said "September".

/H=E5kan




From rem-conf Tue Sep 23 00:00:25 1997 
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Date: Mon, 22 Sep 1997 23:52:46 -0700 (PDT)
From: Stephen Casner <casner@precept.com>
To: rem-conf@es.net
Subject: Minutes of Munich AVT meeting
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Enclosed are the minutes I have prepared for the AVT meeting held last
month at IETF in Munich.  Comments are welcome.
							-- Steve


Minutes of the Audio/Video Transport Working Group

Reported by Steve Casner

1.  Introduction and status

The AVT working group had not planned to meet in Munich during a
period of dormancy while implementation and testing of RTCP scaling
mechanisms was underway.  However, in response to several requests,
one AVT session was scheduled to discuss a new problem with the RTCP
scaling plus open issues for several new payload formats that were
submitted since the last meeting.

The next goal for AVT is to get RFCs 1889 and 1890, the Real-time
Transport Protocol and the companion RTP profile for audio/video
conferencing, revised for advancement to Draft Standard by year's end.
The profile has been revised in draft-ietf-avt-new-profile-01.txt and
.ps.  The main RTP specification needs to incorporate RTCP scaling
changes that are still being refined; however, an interim first draft
of the other revisions, including extensions for layered encodings,
should be published as soon as posible.

As of the Munich meeting, the RTP payload format for H.263 video had
been approved for publication as a Proposed Standard, and the payload
format for redundant audio was close to approval.  (Since the meeting,
both have been published as RFCs 2190 and 2198.)  The drafts on
compression of IP/UDP/RTP headers and a revision to the RTP payload
format for MPEG2 in RFC 2038 have been posted for IESG Last Call
before publication as Proposed Standards.


2.  RTCP "timer reconsideration"

Jonathan Rosenberg gave a brief review of the problem that a flood of
RTCP packets can occur if a large number of participants
simultaneously join a session.  This problem and the solution of RTCP
"timer reconsideration" have been discussed at the previous two IETF
meetings and are explained in the recently posted Internet-Draft
draft-ietf-avt-reconsider-00.ps.  However, a concern raised at the
last meeting was that the timer reconsideration algorithm exhibits a
"plateau effect" wherein no RTCP packets are sent for a period after
the algorithm stops the flood.  New simulations show that the plateau
effect is significantly reduced as the joins are spread out in time
and disappears when the join rate is no more than 500 users/sec.
Since perfectly synchronized joins are very unlikely, the plateau
effect is not considered to be a problem.

There is an analogous problem of an RTCP BYE packet flood on
simultaneous leaves.  Unlike the initial RTCP packets at the time of
joining, BYE packets can't be delayed because the application is
terminating.  Some partial remedies were discussed at the last
meeting.

The newly discovered problem is a relatively minor one related to the
BYE flood: if many participants leave a session at once, other
participants that remain in the session may be falsely timed out.
Some of the remaining participants will have their RTCP interval
expire earlier than others.  These participants will reduce their
estimate of the number of participant and consequently also reduce
their RTCP interval.  Since the remaining participants may still have
a long time to wait before their previously calculated RTCP interval
expires, they might not send any RTCP packets while multiple shorter
RTCP intervals elapse for the particpants that have noticed the drop.
The long-waiting participants will therefore be timed out (considered
inactive) by those that have adjusted to shorter intervals.  Even if
the BYE flood is slowed to the normal 5% RTCP bandwidth, simulations
show that a single missed packet can cause a timeout.

An extension to the timer reconsideration algorithm, dubbed "reverse
reconsideration", avoids this timeout.  Whenever the group size
estimate decreases (due to a BYE), the RTCP timer is recomputed.
There is still a problem that the group size estimate drops to zero.
This problem can be eliminated, as suggested by Sharma and Jacobson in
an INFOCOM97 paper on a similar problem, using a filter to slow the
rate of decrease of the estimate.  Further analysis to determine the
right kind of filter is underway.

Jon Crowcroft suggested keeping a bit of history from interval to
interval and continuously estimating the group size to use as a
predictor.  However, a predictor might not work well with unexpected
sharp transitions in group size.  He also suggested that the BYE flood
might be managed by having participants send BYEs with a limited scope
and then having the participants that remain act as proxies to
retransmit the BYEs spread out in time.  A proxy election protocol
would be needed.  This starts to get pretty complicated.

Dave Oran suggested that there could be a server thread that takes
responsibility for sending a delayed BYE after the application is
closed, just as TCP implementations must save some state when a
connection closes to shut down gracefully.  Steve Casner responded
that if some implementations can do this, great, but we can't assume
all will.  Similarly, it is important to note that although we want to
define a timer reconsideration algorithm to go in the appendix of the
RTP spec, different implementations can implement different timer
reconsideration algorithms and the whole system will still work.

The group was asked if there were any objections to the proposal to
add the timer reconsideration algorithm to the RTP spec.  There were
none.


2.1 Large-scale tests of the RTCP scaling mechanisms

The timer reconsideration algorithm has been simulated, but we would
have more confidence if it was tested on a large scale with real
implementations.  If that is not practical, testing on a large-scale
simulation/emulation testbed network would be a good backup for the
initial simulations.  The audience was queried about any planned
tests, but no plans were reported at this time.

Jonthan Rosenberg is implementing a test program called rtpbomb that
can emulate a large number of participants.  It may be used in
combination real participants to test timer reconsideration and other
aspects of RTCP scaling.  His analysis suggests that most of the
phenomena in the algorithm are linear with group size, so a smaller
test should still be useful to predict behavior of a larger group.


3. Other RTP Issues/Questions

Steve Casner brought up three other issues for the group to consider,
but there were few comments:

  - During the discussion of the global multicast address allocation
    scheme in MBONED working group, Van Jacobson asserted that
    applications should not depend upon sequential multicast addresses
    being allocated for multiple groups carrying the different layers
    of a layered coding scheme.  The layered encoding extension to
    RTP proposed by Speer and McCanne does assume sequential
    addresses, so a revision may be necessary.

  - AVT has received a Request for Proposals from the DAVIC group.  It
    would be difficult for the group as a whole to respond, but
    individual participants are invited to do so.

  - Should AVT set a policy of allocating no more static payload
    types such that dynamic payload types should be used instead?

Scott Petrack responded to the last point to say that we should
consider none of the assigned payload types to be static, but rather
default assignments.  When a session control protocol is in use, if
there is a need for more than the 32 dynamic payload types already set
aside, the default values could be reassigned.  The group seemed to be
in general agreement with this proposal, but it should be discussed on
the mailing list as well.


4. New RTP payload format proposals

It is expected that additional payload formats for RTP will continue
to be developed as new encodings are developed and as RTP is employed
in new applications.  At this meeting, new payload formats were
introduced for H.263+, MPEG4, BT-656 and QuickTime video, and DTMF
tone signalling in audio.  In addition, "meta" payload formats that
specify methods for repair of packet losses were discussed, including
the question of how much error correction is appropriate.

4.1 H.263+ video payload format

As noted in the introduction, the payload format specification for
H.263 video has been published as a Proposed Standard.  However,
enhancement of the encoding itself continues under the label H.263+
for improved loss resilience and to utilize increased processing power
to achieve higher quality.  As agreed in previous AVT meetings, a
separate RTP payload format is to be devised to support the enhanced
encoding.  In fact, two different approaches were proposed in draft
payload format specifications sent to the AVT mailing list just before
the meeting.  Stephen Wenger, who is the primary author of one and a
co-author on the other, gave an overview of the H.263+ encoding
enhancements and the tradeoffs between the two payload formats.

The first approach, from TU-Berlin and U-Bremen, uses a very simple
payload header but includes the H.263+ picture header in each packet.
This allows for the use of almost all of the optional modes defined in
the H.263+ encoding scheme.  The second approach, from Intel, follows
more the design of the existing H.263 payload format in which selected
fields from the picture header are incorporated into a set of payload
format headers for different modes.  This approach reduces overhead to
support small packet sizes but precludes use of some encoding options
deemed to be rarely used.

Wenger stated the intention of the authors of both approaches to
resolve the differences and produce one merged proposal.  He suggested
that the merged proposal might include both approaches.  Steve Casner
countered that this might not be a good result because it increases
complexity and reduces interoperability -- putting in all the options
when a decision can't be made is often the result of a committee
design.  Christian Huitema commented that some encoding functions were
left out when the H.261 payload format was designed because they were
not applicable to packet transport, and asked if a similar analysis
could be made here.  Wenger replied that in H.263+ the enhancements
were expressly designed to work with packet loss.

Wenger requested input soon from the working group regarding the
tradeoffs between the two approaches because the recommendations for
mode combinations to use are being prepared now by the ad-hoc group he
leads.  The two specifications should be posted as Internet-Drafts for
wider exposure.

4.2 MPEG4 payload format

There is no draft payload format specification for MPEG4 yet because
Gerard Fernando would like feedback on some design questions first.
He gave a brief overview of the scope and structure of MPEG4, which is
a framework for integrating different kinds of natural and synthetic
media streams built around audio/visual objects called AVIOs.  There
are two primary questions:

  - How can MPEG4 payload format refer to existing and forthcoming
    payload formats for the encodings of individual media streams
    (AVIOs) rather than trying to redo them all?

  - How should the "scene description data", which composes the
    individual AVIOs into the complete presentation, be transmitted?
    If this data is lost, the whole scene is lost, so adequate loss
    resilience mechanisms must be employed.  Can the scene description
    data be decomposed over both time and "space" -- time because a
    transmission may be joined after the start, and space because
    media are sent in separate streams and some receivers might not
    tune into all of them?

MPEG4 is still being developed, with standardization due in January
1999.  The MPEG4 committee would like to work with AVT to ensure that
packetization considerations are included in the design; they view
RTP/UDP multiplexing as an appropriate means of multiplexing
elementary streams.  This is clearly beneficial from AVT's point of
view as well.  One aspect of the use of separate media streams in RTP
is the ability to apply different network QoS and reliability
mechanisms as needed; the VMIF part of MPEG4 is trying to formalize
the selection of stream-specific QoS in a transport-independent and
network-independent way.  Fernando and Casner are to explore what
formal or informal liaison procedures should be followed in this case,
both for information transfer and to enable reference to RTP in the
MPEG4 specification.

4.3 BT-656 video payload format

Dermot Tynan presented a proposal, draft-tynan-rtp-bt656-00.txt, for
carrying ITU-R BT.656-3 uncompressed video over RTP.  BT656 is
studio-quality digital video sampled according to BT.601-5 (formerly
CCIR601) at 13.5 or 18 MHz.  At the normal, lower rate, each scan line
contains 720 samples occupying 1440 bytes in the 4:2:2 chrominance
encoding.  At the "high definition" rate, each line contains 1144
samples for NTSC or 1152 samples for PAL.  The payload format consists
of a simple header with bit fields to indicate NTSC/PAL, sampling
rate, framing and scan line information, followed by one scan line of
samples.

Steve Casner expressed the concern that the packet size might exceed
the MTU for some networks.  Although at the lower sampling rate an
IP/UDP/RTP packet will fit within a 1500-byte MTU, at the higher
sampling it would not.  Don Hoffman pointed out that if the packet
size does exceed the MTU such that IP fragmentation occurs, integrated
services packet classification won't work on fragments other than the
first so those packets won't get the desired QoS.  Christian Huitema
claimed it is important to have application-level fragmentation so
that services such as forward error correction will apply when a
fragment is lost.  It should not cost much to be able to indicate that
a packet contained part of a scan line, perhaps by giving the starting
sample number.  Tynan agreed to consider this addition.

4.4 QuickTime payload format

Alagu Periyannan just asked to call the working group's attention to
the recently posted draft "RTP Payload Format for QuickTime Media
Streams", draft-ietf-avt-qt-rtp-00.txt, and in particular to the list
of open issues in section 4 of that draft.  The motivation in
developing this format was to carry all the payloads in QuickTime
without having to define an RTP payload format for each.

Steve Casner pointed out that the tradeoff of this technique is that
there is some amount of additional constant description info that must
be carried in each packet but would not be needed with separate
formats.  Philipp Hoschka asked if it wouldn't be better to define
individual payload formats because the encodings might also be used
outside of QuickTime as well.  Periyannan replied that where
individual payload formats are defined, they should be used in
preference to this format.  However, several of the encodings used
with QuickTime are not standardized, such as Apple Video, Cinepak, and
several proprietary codecs.

4.5 DTMF audio payload format

Jonathan Rosenberg presented Henning Schulzrinne's proposed audio
payload format to carry DTMF (tone dialing) signals as defined in
draft-ietf-avt-dtmf-00.txt and .ps.  This payload format might be used
for calling across the Internet through IP telephony gateways to
control an answering machine or other device.  Low data rate speech
codecs may not reproduce the DTMF tones faithfully enough to work
properly at the far end.  This payload format essentially provides a
very low rate encoding specialized for DTMF tones.

The draft defines a primary format in which each DTMF digit is
represented by 32 bits to control frequency, amplitude and duration.
Redundancy can be provided using the mechanism specified in RFC 2198.
This introduces 64 bits of overhead, but the data rate is so low that
this probably does not matter.  Alternatively, a more compact
representation of each digit would allow adding redundancy within the
DTMF payload format and still fitting each digit into 32 bits.  The
selection between these techniques is an open issue on which input
>from the working group is sought.  Steve Casner noted that the more
compact format would require the RTP clock rate to be different than
that of the normal audio payload format within which the DTMF packets
are interspersed, and suggested that this is a significant enough
disadvantage to prefer the larger format.  Scott Petrack agreed that
the size difference was probably not significant.

4.6 Forward Error Correction payload format

Two "meta" payload formats on forward error correction (that is,
independent of media type and format) have recently been posted:
draft-budge-media-error-correction-00.txt by Budge, et al., and
draft-ietf-avt-fec-00.txt by Rosenberg and Schulzrinne.  The authors
of the first draft did not attend, but Jonathan Rosenberg presented
the second draft which builds on ideas from the first, and compared
the two drafts.  Both schemes are based on the idea of sending
additional FEC packets which are the XOR of multiple packets from the
original packet stream.

It should be noted that the scheme presented by Rosenberg differs from
that described in the second draft.  An RTP header extension (X bit)
is no longer used; instead, the payload type is changed to indicate an
FEC packet, and an additional format-specific header is inserted
before the XOR of the covered payloads, as in the Budge scheme.

One drawback of the Budge draft was that the timestamp and marker bit
of lost packets could not always be recovered.  The proposed remedy is
for these fields in the FEC packets to be the XOR of the corresponding
fields from the original packets covered by the FEC packet.  This has
its own drawback that the timestamp will vary erratically, perturbing
the jitter feedback calculation.  Carsten Bormann also pointed out
that when RTP header fields are perturbed from their usual increments,
it can have a negative impact on the efficiency of RTP header
compression.

A second drawback of the Budge scheme was that the payload type was
changed in the original packets to be the FEC payload type which means
that a receivers without FEC capability could not receive just the
original packets and ignore the FEC packets.  In Rosenberg's scheme,
the original packets are unmodified, including the payload type.

To extend the range of packet patterns that could be covered by an FEC
without having to predefine specific scemes, Rosenberg proposes to
carry a bit mask to indicate the pattern explicitly.  The RTP sequence
number of FEC packets in Rosenberg's proposal is the minimum of the
sequence numbers of the packets covered by the FEC to serve as a base
for the mask.  Steve Casner claimed that this is not acceptable
because it will prevent the FEC packets from passing header validation
and duplicate suppression algorithms in RTP packet processing.
Rosenberg said alternative sequence number schemes had been
considered, but the cost is additional overhead.

Christian Huitema argued that if a general FEC payload format is to be
defined, it should support other schemes with better performance and
only a marginal increase in complexity compared to that of "n+1"
schemes like XOR.  For example, it is possible to specify a scheme
that adds two FEC packets to eight original packets which will allow
recovery from a loss of any two of the ten packets.  This is important
because simulations have shown that one packet of redundancy was not
enough.  Rosenberg agreed that the FEC proposals need further
refinement.

4.7 Applicability of error correction

In addition to the redundant audio and FEC payload formats already
mentioned, retransmission and interleaving are two more loss
resilience schemes that might be employed with RTP.  Colin Perkins
reviewed these four methods to compare the tradeoffs in latency,
bandwidth overhead and processing overhead (details are available at
http://www.cs.ucl.ac.uk/staff/c.perkins/slides/).  It is an important
question to consider when these schemes are applicable to particular
applications or to particular network conditions.

Colin presented some network loss statistics showing that single
packet losses predominated by a factor of four over two-packet loss
bursts.  The probability drops rapidly such that long burst losses are
rare.  On the other hand, in a large conference, most packets will be
lost by at least one receiver, therefore a retransmission scheme may
need to resend almost every packet.

The redundant transmission scheme has low latency and low bandwidth
overhead, but potentially high processor overhead if the low-rate
redundant encoding is computationally complex.  A retransmission
scheme is likely to impose much longer delay and will incur the
overhead of control traffic in addition to the duplicate data, but
provides exact repair and can correct more than single-packet losses.
There is a synergy between redundancy and retransmission in that
redundancy can cover most of the errors, leaving retransmission to
pick up the remainder for those receivers that care more about low
loss than low delay.

Interleaving disperses the effects of loss, but does not eliminate
them.  It has low overhead, but high latency.  FEC may have a
similarly high latency or a high bandwidth overhead, depending on the
size of the pattern covered, but its major advantage is media and
format independence.  For interactive applications, either redundant
transmission or a low-latency FEC would seem most suitable, while for
broadcast-style applications interleaving works well.

An important open question is what constitutes a sensible operating
point for real-time media transmission?  How much loss should
applications try to cope with before declaring that the user should
try again at some less congested time?  There is little congestion
control in many real-time media applications, and extreme loss
resilience measures would just worsen congestion.  That would not be
network friendly.  If fair congestion avoidance mechanisms are
deployed in routers, then applications that don't implement congestion
control may be penalized.  Therefore, any loss resilience schemes that
are defined for RTP should consider not just loss performance but also
impact on the network.

5. Revision of RTP MIB specification

The final topic of the meeting was an update by Mark Baugher on the
RTP MIB design and implementation plans.  An interim draft of the MIB
was sent to the mailing list before the meeting to provide an
opportunity for feedback; a real Internet-Draft will be posted in
October.  A more complete description of the MIB was given in Memphis;
the changes since then were to fix errors identified by Fred Baker in
his review of the MIB, to add reporting of receiver feedback, to
support RTP operation over different underlying protocols, and to
remove from the specification those parts that won't be included in
initial implementions and hence won't be validated yet.  In
particular, support for RTP translators was removed.

Receiver feedback is reported through additional tables that are
created only upon request from the network manager as a side-effect of
creating an entry in the Session Table.  This avoids the state
explosion that might occur if all receiver feedback was always
monitored for all sessions.

There are a few details remaining to be worked out, but implementation
of a network management application using the RTP MIB is underway to
evaluate how well this MIB works for managing real-time applications
on networks that weren't designed for them.  To make this really
workable, the management application needs to handle multicast routing
in addition to RTP.  The next stage of MIB development will be based
on the results of the implementation.

6.  Next meeting

Steve Casner closed the meeting saying that AVT will meet again at the
next IETF in December.  As stated above, a goal for the working group
is to get the RTP spec revised for Draft Standard by then.  In
addition, there is outstanding work on several topics presented at
this meeting which should be discussed on the mailing list so that
completed drafts are ready by the next meeting.




From rem-conf Tue Sep 23 06:05:40 1997 
From rem-conf-request@es.net Tue Sep 23 06:05:39 1997
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	id 0xDULy-000386-00; Tue, 23 Sep 1997 05:46:10 -0700
Message-ID: <3427B9BC.7EB00284@dnrc.bell-labs.com>
Date: Tue, 23 Sep 1997 08:44:44 -0400
From: "Jonathan D. Rosenberg" <jdrosen@dnrc.bell-labs.com>
Reply-To: jdrosen@dnrc.bell-labs.com
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Stephen Casner wrote:
> 
> An extension to the timer reconsideration algorithm, dubbed "reverse
> reconsideration", avoids this timeout.  Whenever the group size
> estimate decreases (due to a BYE), the RTCP timer is recomputed.
> There is still a problem that the group size estimate drops to zero.
> This problem can be eliminated, as suggested by Sharma and Jacobson in
> an INFOCOM97 paper on a similar problem, using a filter to slow the
> rate of decrease of the estimate.  Further analysis to determine the
> right kind of filter is underway.
> 

A minor clarification here. With reverse reconsideration, you can still
get premature timeouts (that is why the group size drops to zero, even
when members remain). The effect of reverse reconsideration is to make
the group size estimate return to the correct value more rapidly.
Without it, the amount of time for which the estimate is wrong can be as
large as the RTCP interval before the sudden drop in membership. With
reverse reconsideration, the amount of time can only be as large (in the
worst case) as the RTCP interval after the drop in membership. 


> 4.6 Forward Error Correction payload format

> One drawback of the Budge draft was that the timestamp and marker bit
> of lost packets could not always be recovered.  The proposed remedy is
> for these fields in the FEC packets to be the XOR of the corresponding
> fields from the original packets covered by the FEC packet.  This has
> its own drawback that the timestamp will vary erratically, perturbing
> the jitter feedback calculation.  

This is only true if you include the FEC packet in the jitter
computation. 


Thanks,

Jonathan R.

-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
PHONE: (908) 949-6418                       Rm. 4B-505
FAX:   (908) 834-5379
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen



From rem-conf Tue Sep 23 12:57:31 1997 
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Date: Tue, 23 Sep 1997 12:46:09 -0700 ()
From: Stephen Casner <casner@precept.com>
To: rem-conf@es.net
Subject: Re: Minutes of Munich AVT meeting
In-Reply-To: <Pine.SOL.3.95.970922234946.15855B-100000@little-bear.precept.com>
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Oops.  I owe an apology to Stephan Wenger for misspelling his name in
the AVT meeting minutes I sent last evening.  I should be more
sensitive to the spelling of that particular name!

I'll make edits for the comments I have received and send that to the
IETF minutes editor.
							-- Steve




From rem-conf Tue Sep 23 20:17:58 1997 
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Date: Tue, 23 Sep 1997 22:02:26 -0500 (CDT)
From: BEZALEL GAVISH <GAVISHB@ctrvax.Vanderbilt.Edu>
Subject: CFP - 6th International Conference on Telecommunication Systems
To: LIST3: ;
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----------------------------------------------------------------------------
			    Call for Papers
      6th International Conference on Telecommunication Systems
			 Modeling and Analysis
			    March 5-8, 1998
			     Nashville, TN

The 6th International Conference on Telecommunication Systems - Modeling
and Analysis will be held in Nashville on March 5-8, 1998. The conference
will build on the tradition of the earlier conferences with a few changes
in format due to the new conference location. The general idea is to
limit the number of participants, concentrate on a few topics, present
new problems and problem areas, and to encourage informal interaction and
exchanges of ideas. The objective is to advance the state of the modeling
and analysis in telecommunications by stimulating research activity on
new and important problems.

The conference will be divided into segments with each segment devoted to
a specific topic. This will allow for little conflict between segments.
All papers will be screened by the Program Committee to ensure the
quality of presentations. Social and cultural activities will be included
in the 1998 agenda.

The deadline for submission of papers and extended abstracts is October
1, 1997. Additional information about the conference can be obtained from
the conference web page at URL http://munin.utdallas.edu/telsys/cfp6.htm

Listed below are some of the potential conference segments:

   - Configuration and Management of ATM networks
   - Internet and its Impact on Commerce and Organizational Structures
   - Internet and Intranet
   - Mobility and Nomadicity
   - Topological Design and Network Configuration Problems
   - Design and Analysis of Local Access Networks and Outside Plant Problems
   - Low and Medium Earth Orbit Satellite Communication Systems
   - Cellular Systems and PCS Modeling and Configuration
   - Time Dependent Expansion of Telecommunication Systems
   - Designing Networks for Reliability and Availability
   - Network Design Problems in Gigabit and Terabit Networks
   - LAN, WAN Global Network Interconnection
   - ADSL and its Impact
   - ATM, ISDN, BISDN Modeling and Analysis Issues
   - Telecommunication Standards
   - Artificial Intelligence/Heuristics in Telecommunication Systems
   - Group Decision Support Systems
   - Quantitative Methods in Network Management
   - Pricing and Economic Analysis of Telecommunications
   - Handling Investments in High Risk Technologies
   - Impact of Telecommunications on Industrial Organization
   - Performance Evaluation of Telecommunication Systems
   - Distributed Computing and Distributed Data Bases
   - Security and Privacy Issues in Telecommunications
   - Virtual Reality, Multimedia and their Impact

I look forward to a very successful conference.

Sincerely yours,
Bezalel Gavish

-------------------------------------------------------------------------------
Bezalel Gavish
Owen Graduate School of Management
Vanderbilt University                     Tel : (615) 322-3659
Nashville, TN, 37203                      FAX : (615) 343-7177
Internet: GAVISHB@CTRVAX.VANDERBILT.EDU   Home: (615) 370-0813
-------------------------------------------------------------------------------



From rem-conf Wed Sep 24 08:04:07 1997 
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Message-Id: <199709241350.JAA15486@east.isi.edu>
To: rem-conf@es.net
Subject: MBONE Event: Deering on IPv6, 9-11 AM EDT, Sep 25
Reply-To: mankin@EAST.ISI.EDU
Date: Wed, 24 Sep 1997 09:50:33 EDT
From: Allison Mankin <mankin@ISI.EDU>
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A seminar on IPv6 by Steve Deering will be presented 
on the MBONE 9-11 AM EDT, tomorrow, Thursday Sep 25.
See the session advertisement for more information.



From rem-conf Wed Sep 24 09:37:42 1997 
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	id 0xDuEK-0001Z3-00; Wed, 24 Sep 1997 09:24:00 -0700
Date: Wed, 24 Sep 1997 18:22:43 +0200
From: likavec@igd.fhg.de (Jaromir Likavec)
Message-Id: <199709241622.SAA11360@owl.igd.fhg.de>
To: rem-conf@es.net
Subject: vic2.8/Parallax_Xtra_Problem
Cc: likavec@igd.fhg.de
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Hi,

I've changed SUN Video Card with Parallax XVideo Xtra overlay card on the
SUN Ultra 1 with creator 3D under Solaris 2.6.

Parallax tools VideoTool, MovieTool are working without any problem
Vic 2.8 brings the message vic:can't open parallax capture device.

Any idea what to do?

Regards

Jaromir


********************************************************************************
Jaromir Likavec	        Computer Graphics Center   |  A journey of a thousand
(likavec@igd.fhg.de)    Rundeturmstr. 6            |  miles begins with  a
                        D-64283 Darmstadt, Germany |  single step.
                        Phone: +49/6151/155-314    |  Lao-Tsu Dao  De Sing
                        Fax: +49/6151/155-399      |
********************************************************************************



From rem-conf Wed Sep 24 09:48:53 1997 
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Date: Wed, 24 Sep 1997 09:46:27 -0700
Message-Id: <199709241646.AA18484@zephyr.isi.edu>
From: estrin@usc.edu (Deborah Estrin)
Sender: estrin@ISI.EDU
To: rem-conf@es.net
Subject:  Murray Campbell, IBM,  Deep Blue:  The IBM Chess Machine
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As part of the USC COMPUTER SCIENCE AND COMPUTER ENGINEERING
                      DISTINGUISHED LECTURE SERIES
                             1997-1998

                         Murray Campbell
                              IBM
              Deep Blue:  The IBM Chess Machine
                   Wednesday, Sept. 24, 1997
                            1:00 PM
                     Host: Steven Minton
                     
Today on the mbone (we hope). Join us on USCLectureSeries now being
advertised in sdr. 
Sorry for the late notice. I will try to get these out earlier for 
future seminars. 


(FYI Upcoming events in the series) 
		        Butler Lampson
                           Microsoft
                    Title: (to be announced)
                   Wednesday, Nov. 19, 1997, 12 noon
                     Host: Deborah Estrin

                        Ernest S. Kuh
            University of California at Berkeley
  Interconnect modeling and simulation for deep-submicron design
                 Wednesday, January 28, 1998, 12 noon
                    Host: Massoud Pedram

                         Yale Patt
                  University of Michigan
            The Microprocessor of the Year 2008
                Thursday, February 19, 1998, 12 noon
                  Host: Jean-Luc Gaudiot

                        Barbara Grosz
                     Harvard University
                    Title: (To be announced)
                    Thursday, March 19, 12 noon
                     Host: Milind Tambe

                          David Dill
                     Stanford University
     Towards Formal Verification of Microprocessor Designs
                    Wednesday, April 29, 12 noon
                     Host: Peter Beerel



NOTE: we also have a more or less regular wed noon seminar
approximately three times a month which we also plan to mbone. 

Next week Wed Oct 1, 12 noon, Vern Paxson will be speaking





From rem-conf Wed Sep 24 10:31:43 1997 
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Date: Wed, 24 Sep 1997 10:00:29 -0700
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 <Pine.SOL.3.95.970922234946.15855B-100000@little-bear.precept.com>
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From: Alagu Periyannan <alagu@apple.com>
Subject: Re: Minutes of Munich AVT meeting
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At 11:52 PM -0700 9/22/97, Stephen Casner wrote:
>
>
>4. New RTP payload format proposals
>
>It is expected that additional payload formats for RTP will continue
>to be developed as new encodings are developed and as RTP is employed
>in new applications.  At this meeting, new payload formats were
>introduced for H.263+, MPEG4, BT-656 and QuickTime video, and DTMF
>tone signalling in audio.  In addition, "meta" payload formats that
>specify methods for repair of packet losses were discussed, including
>the question of how much error correction is appropriate.
>

Hi all,

I would like to point put that the QuickTime payload format
covers not just video but all the QuickTime track types.

This includes audio, video, text, MIDI, etc.




---------------------------------------------------
Alagu Periyannan                   alagu@apple.com

Interactive Multimedia Group
Apple Computer, Inc.





From rem-conf Thu Sep 25 05:45:30 1997 
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Date: Thu, 25 Sep 1997 14:30:06 +0200
From: wangd@sebb.bel.alcatel.be
Message-Id: <199709251230.OAA26892@btmplq.god.bel.alcatel.be>
To: rem-conf@es.net
Subject: How to attend multicast meeting?
X-Sun-Charset: US-ASCII
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Hi,

	I have a SUN workstation connected to Internet. I also downloaded
"vat", and ran "vat" in our LAN, it works. Now I have following questions:

	Can I attend multicast meetings? and how? What else do I need?
	My workstation does not have a video card, can I run "vic" to
		receive and play video session information?




Thanks!

David Wang


e-mail: wangd@sebb.bel.alcatel.be



From rem-conf Thu Sep 25 10:52:12 1997 
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From: "Glenn L. Cash" <glenn@research.att.com>
To: <rem-conf@es.net>
Subject: RTP JPEG payload and interlaced video
Date: Thu, 25 Sep 1997 13:40:19 -0400
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Has anyone tried to use RFC2035 for JPEG payloads
with hardware codecs that code/decode 2 fields per
frame? If so, how are you indicating which field is
contained in the payload?

		Thanks,

			Glenn Cash
--
Glenn L. Cash
AT&T Labs - Research
100 Schulz Drive, Room 3-221,
Red Bank, NJ  07701-7033
tel:  732 345 3303 
fax: 732 345 3033
E-mail: glenn@research.att.com
WWW:  http://www.research.att.com/info/glenn



From rem-conf Thu Sep 25 17:20:26 1997 
From rem-conf-request@es.net Thu Sep 25 17:20:25 1997
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Reply-To: stewart@parc.xerox.com
To: "Glenn L. Cash" <glenn@research.att.com>
cc: rem-conf@es.net
Subject: Re: RTP JPEG payload and interlaced video 
In-reply-to: Your message of "Thu, 25 Sep 1997 10:40:19 PDT."
             <199709251744.NAA01983@surfcity.research.att.com> 
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Date: Thu, 25 Sep 1997 16:47:39 PDT
From: Paul Stewart <stewart@parc.xerox.com>
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In message <199709251744.NAA01983@surfcity.research.att.com>you write:
>Has anyone tried to use RFC2035 for JPEG payloads
>with hardware codecs that code/decode 2 fields per
>frame? If so, how are you indicating which field is
>contained in the payload?

In previous versions of the payload specification, there were bits in 
the type field that specified interlace information, but that was 
dropped in later revisions.  I've recently been meeting with Ron 
Frederick and Bill Fenner to work out a number of issues related to 
this standard, so hopefully in an upcoming revision you will be able 
to specify this sort of information without need for non-standard 
extensions.  I'll post a bit more detail about some of these changes 
in an upcoming message to rem-conf.

--
Paul





From rem-conf Thu Sep 25 17:38:29 1997 
From rem-conf-request@es.net Thu Sep 25 17:38:28 1997
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Subject: RTP JPEG payload format
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Date: Thu, 25 Sep 1997 17:35:26 PDT
From: Paul Stewart <stewart@parc.xerox.com>
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  Recently Ron Frederick, Bill Fenner and I have been spending a 
little time working on some of the issues related to the RTP Payload 
Format for JPEG-compressed Video (RFC 2035).  We'd like to petition 
the public for their opinions on some of the topics we are addressing.


    * Interlace support

    We'd like to one again resurrect a 4:2:2 interlaced video format 
type.  Even and odd video fields can specified using the "Type 
specific" data field.  A separate type can be used for video where 
only one of the two fields are encoded so that decoders know they 
need to line double video in order to display the content at the 
correct aspect ratio.


    * Restart markers

    Restart markers as they are currently specified are only 
partially useful.  Firstly, they are a separate type, and as such, 
cannot be added orthogonally to a additional JPEG types.  Secondly, 
the optional DRI segment is 6 bytes, causing the real payload data 
not to be 4 byte aligned.  In light of these issues, we'd like to 
move the restart marker information to being a single high order bit 
in the type field, and encoding both a 16 bit RCOUNT and the 16 bits 
of pertinent info in a single 4 byte block immediately after the 
RTP/JPEG header, leaving the "Type specific" data field free for use 
by the specific format specified in the low order bits of the type 
field.  How many software and hardware codecs currently support types 
2-5?


    * Q-Factor support

    Allow for a special Q-factor value that specifies that the 
Q-table is specified at the beginning of the payload data.  This can 
be used with codecs for which the table for the encoder can be 
queried but cannot be set.

--
Paul





From rem-conf Thu Sep 25 22:49:47 1997 
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To: Jim Lowe <james@miller.cs.uwm.edu>, rem-conf@es.net
Reply-To: mankin@EAST.ISI.EDU
Subject: Re: MBONE Event: Deering on IPv6, 9-11 AM EDT, Sep 25 
In-reply-to: Your message of Thu, 25 Sep 1997 09:32:51 -0500.
             <199709251432.JAA21676@miller.cs.uwm.edu> 
Date: Fri, 26 Sep 1997 01:44:05 EDT
From: Allison Mankin <mankin@ISI.EDU>
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This event was taped and as soon as we check that
the tape came out ok, we will announce a rebroadcast time.

> It looks like part of the mbone was down during this transmission.
> I was unable to receive it.  Do you know if it was recorded anywhere?



From rem-conf Fri Sep 26 00:38:08 1997 
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From: petri.koskelainen@research.nokia.com (Koskelainen Petri NRC/Tre)
To: confctrl@isi.edu ('confctrl@isi.edu'), rem-conf@es.net ('rem-conf@es.net')
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Date: Fri, 26 Sep 1997 10:35:18 +0200
Subject: SDP syntax fo H.263 options (draft)
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There is new  internet-draft coming out soon describing the SDP syntax
for H.263 options and parameters, and their relation to SIP and SAP.
For those who can't wait, see it in:
http://www.cs.tut.fi/~petkos/draft-koskelainen-sdp263-00.txt

We have made most changes suggested by Stephen Casner, except that
MaxBitRate is still used since we think that SDP b attribute is not the same
as MaxBitRate in SDP H.263 decoder capabilities.

Moreover, SIP spec would need minor changes to say that unicast
codec / codec options are not always symmetric. To make asymmetric
usage possible, it is necessary to define that SDP codecs and codec options
are only (advertized)  decoder capabilities. Both ends (in INVITE, and its 
OK reply
messages), define their decoders. Current SIP spec is not saying anything 
about
this.


 --
Petri





From rem-conf Fri Sep 26 09:22:01 1997 
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Message-Id: <342BDEC8.83622A6F@watson.ibm.com>
Date: Fri, 26 Sep 1997 12:11:52 -0400
From: Ping Pan <pan@watson.ibm.com>
Organization: IBM Research Center
X-Mailer: Mozilla 4.03 [en] (Win95; U)
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Hi,

Where can I find wb for NT or win95? 

Many thanks!

--
Ping Pan
pan@watson.ibm.com



From rem-conf Fri Sep 26 14:59:56 1997 
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http://www.hispavista.com/





From rem-conf Fri Sep 26 18:40:53 1997 
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From: Mark Allard <mallard@mail.arc.nasa.gov>
Subject: Re: MBONE Event: Deering on IPv6, 9-11 AM EDT, Sep 25 
In-Reply-To: <199709260543.BAA28610@east.isi.edu>
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Please restart your tools and try again.
Thanks,
M

At 01:44 AM 9/26/97 EDT, Allison Mankin wrote:
>This event was taped and as soon as we check that
>the tape came out ok, we will announce a rebroadcast time.
>
>> It looks like part of the mbone was down during this transmission.
>> I was unable to receive it.  Do you know if it was recorded anywhere?
>
>



From rem-conf Sat Sep 27 00:16:17 1997 
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Date: Sat, 27 Sep 1997 00:07:40 -0700 (PDT)
From: Stephen Casner <casner@precept.com>
To: rem-conf@es.net
Subject: Revised AVT Munich minutes
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I've revised the minutes according to the comments received both on
this list and privately.  Thanks for the feedback.
							-- Steve

Minutes of the Audio/Video Transport Working Group

Reported by Steve Casner

1.  Introduction and status

The AVT working group had not planned to meet in Munich during a
period of dormancy while implementation and testing of RTCP scaling
mechanisms was underway.  However, in response to several requests,
one AVT session was scheduled to discuss a new problem with the RTCP
scaling plus open issues for several new payload formats that were
submitted since the last meeting.

The next goal for AVT is to get RFCs 1889 and 1890, the Real-time
Transport Protocol and the companion RTP profile for audio/video
conferencing, revised for advancement to Draft Standard by year's end.
The profile has been revised in draft-ietf-avt-new-profile-01.txt and
.ps.  The main RTP specification needs to incorporate RTCP scaling
changes that are still being refined; however, an interim first draft
of the other revisions, including extensions for layered encodings,
should be published as soon as posible.

As of the Munich meeting, the RTP payload format for H.263 video had
been approved for publication as a Proposed Standard, and the payload
format for redundant audio was close to approval.  (Since the meeting,
both have been published as RFCs 2190 and 2198.)  The drafts on
compression of IP/UDP/RTP headers and a revision to the RTP payload
format for MPEG2 in RFC 2038 have been posted for IESG Last Call
before publication as Proposed Standards.


2.  RTCP "timer reconsideration"

Jonathan Rosenberg gave a brief review of the problem that a flood of
RTCP packets can occur if a large number of participants
simultaneously join a session.  This problem and the solution of RTCP
"timer reconsideration" have been discussed at the previous two IETF
meetings and are explained in the recently posted Internet-Draft
draft-ietf-avt-reconsider-00.ps.  However, a concern raised at the
last meeting was that the timer reconsideration algorithm exhibits a
"plateau effect" wherein no RTCP packets are sent for a period after
the algorithm stops the flood.  New simulations show that the plateau
effect is significantly reduced as the joins are spread out in time
and disappears when the join rate is no more than 500 users/sec.
Since perfectly synchronized joins are very unlikely, the plateau
effect is not considered to be a problem.

There is an analogous problem of an RTCP BYE packet flood on
simultaneous leaves.  Unlike the initial RTCP packets at the time of
joining, BYE packets can't be delayed because the application is
terminating.  Some partial remedies were discussed at the last
meeting.

The newly discovered problem is a relatively minor one related to the
BYE flood: if many participants leave a session at once, other
participants that remain in the session may be falsely timed out.
Some of the remaining participants will have their RTCP interval
expire earlier than others.  These participants will reduce their
estimate of the number of participant and consequently also reduce
their RTCP interval.  Since the remaining participants may still have
a long time to wait before their previously calculated RTCP interval
expires, they might not send any RTCP packets while multiple shorter
RTCP intervals elapse for the particpants that have noticed the drop.
The long-waiting participants will therefore be timed out (considered
inactive) by those that have adjusted to shorter intervals.  Even if
the BYE flood is slowed to the normal 5% RTCP bandwidth, simulations
show that a single missed packet can cause a timeout.

An extension to the timer reconsideration algorithm, dubbed "reverse
reconsideration", recomputes the RTCP timer whenever the group size
estimate decreases (due to a BYE).  This significantly reduces the
amount of time during which the group size estimate may be wrong, but
there is still a problem that the estimate can drop to zero.  This
problem can be eliminated, as suggested in an INFOCOM97 paper by
Sharma, Estrin, Floyd and Jacobson on a similar problem, using a
filter to slow the rate of decrease of the estimate.  Further analysis
to determine the right kind of filter is underway.

Jon Crowcroft suggested keeping a bit of history from interval to
interval and continuously estimating the group size to use as a
predictor.  However, a predictor might not work well with unexpected
sharp transitions in group size.  He also suggested that the BYE flood
might be managed by having participants send BYEs with a limited scope
and then having the participants that remain act as proxies to
retransmit the BYEs spread out in time.  A proxy election protocol
would be needed.  This starts to get pretty complicated.

Dave Oran suggested that there could be a server thread that takes
responsibility for sending a delayed BYE after the application is
closed, just as TCP implementations must save some state when a
connection closes to shut down gracefully.  Steve Casner responded
that if some implementations can do this, great, but we can't assume
all will.  Similarly, it is important to note that although we want to
define a timer reconsideration algorithm to go in the appendix of the
RTP spec, different implementations can implement different timer
reconsideration algorithms and the whole system will still work.

The group was asked if there were any objections to the proposal to
add the timer reconsideration algorithm to the RTP spec.  There were
none.


2.1 Large-scale tests of the RTCP scaling mechanisms

The timer reconsideration algorithm has been simulated, but we would
have more confidence if it was tested on a large scale with real
implementations.  If that is not practical, testing on a large-scale
simulation/emulation testbed network would be a good backup for the
initial simulations.  The audience was queried about any planned
tests, but no plans were reported at this time.

Jonthan Rosenberg is implementing a test program called rtpbomb that
can emulate a large number of participants.  It may be used in
combination real participants to test timer reconsideration and other
aspects of RTCP scaling.  His analysis suggests that most of the
phenomena in the algorithm are linear with group size, so a smaller
test should still be useful to predict behavior of a larger group.


3. Other RTP Issues/Questions

Steve Casner brought up three other issues for the group to consider,
but there were few comments:

  - During the discussion of the global multicast address allocation
    scheme in MBONED working group, Van Jacobson asserted that
    applications should not depend upon sequential multicast addresses
    being allocated for multiple groups carrying the different layers
    of a layered coding scheme.  The layered encoding extension to
    RTP proposed by Speer and McCanne does assume sequential
    addresses, so a revision may be necessary.

  - AVT has received a Request for Proposals from the DAVIC group.  It
    would be difficult for the group as a whole to respond, but
    individual participants are invited to do so.

  - Should AVT set a policy of allocating no more static payload
    types such that dynamic payload types should be used instead?

Scott Petrack responded to the last point to say that we should
consider none of the assigned payload types to be static, but rather
default assignments.  When a session control protocol is in use, if
there is a need for more than the 32 dynamic payload types already set
aside, the default values could be reassigned.  The group seemed to be
in general agreement with this proposal, but it should be discussed on
the mailing list as well.


4. New RTP payload format proposals

It is expected that additional payload formats for RTP will continue
to be developed as new encodings are developed and as RTP is employed
in new applications.  At this meeting, payload formats were introduced
for H.263+ and BT-656 video, MPEG4 and QuickTime multimedia, and DTMF
tone signalling in audio.  In addition, "meta" payload formats that
specify methods for repair of packet losses were discussed, including
the question of how much error correction is appropriate.

4.1 H.263+ video payload format

As noted in the introduction, the payload format specification for
H.263 video has been published as a Proposed Standard.  However,
enhancement of the encoding itself continues under the label H.263+
for improved loss resilience and to utilize increased processing power
to achieve higher quality.  As agreed in previous AVT meetings, a
separate RTP payload format is to be devised to support the enhanced
encoding.  In fact, two different approaches were proposed in draft
payload format specifications sent to the AVT mailing list just before
the meeting.  Stephan Wenger, who is the primary author of one and a
co-author on the other, gave an overview of the H.263+ encoding
enhancements and the tradeoffs between the two payload formats.

The first approach, from TU-Berlin and U-Bremen, uses a very simple
payload header but includes the H.263+ picture header in each packet
except for "follow-on packets" when app-level fragmentation is done.
This allows for the use of almost all of the optional modes defined in
the H.263+ encoding scheme.  The second approach, from Intel, follows
more the design of the existing H.263 payload format in which selected
fields from the picture header are incorporated into a set of payload
format headers for different modes.  This approach reduces overhead to
support small packet sizes but precludes use of some encoding options
deemed to be rarely used.

Wenger stated the intention of the authors of both approaches to
resolve the differences and produce one merged proposal.  He suggested
that the merged proposal might include both approaches.  Steve Casner
countered that this might not be a good result because it increases
complexity and reduces interoperability -- putting in all the options
when a decision can't be made is often the result of a committee
design.  Christian Huitema commented that some encoding functions were
left out when the H.261 payload format was designed because they were
not applicable to packet transport, and asked if a similar analysis
could be made here.  Wenger replied that H.263+ has been designed to
work with packet loss and that some of the optional modes were
added expressly to deal with packet loss.

Wenger requested input soon from the working group regarding the
tradeoffs between the two approaches because the recommendations for
mode combinations to use are being prepared now by the ad-hoc group he
leads.  The two specifications should be posted as Internet-Drafts for
wider exposure.

4.2 MPEG4 payload format

There is no draft payload format specification for MPEG4 yet because
Gerard Fernando would like feedback on some design questions first.
He gave a brief overview of the scope and structure of MPEG4, which is
a framework for integrating different kinds of natural and synthetic
media streams built around audio/visual objects called AVIOs.  There
are two primary questions:

  - How can MPEG4 payload format refer to existing and forthcoming
    payload formats for the encodings of individual media streams
    (AVIOs) rather than trying to redo them all?

  - How should the "scene description data", which composes the
    individual AVIOs into the complete presentation, be transmitted?
    If this data is lost, the whole scene is lost, so adequate loss
    resilience mechanisms must be employed.  Can the scene description
    data be decomposed over both time and "space" -- time because a
    transmission may be joined after the start, and space because
    media are sent in separate streams and some receivers might not
    tune into all of them?

MPEG4 is still being developed, with standardization due in January
1999.  The MPEG4 committee would like to work with AVT to ensure that
packetization considerations are included in the design; they view
RTP/UDP multiplexing as an appropriate means of multiplexing
elementary streams.  This is clearly beneficial from AVT's point of
view as well.  One aspect of the use of separate media streams in RTP
is the ability to apply different network QoS and reliability
mechanisms as needed; the VMIF part of MPEG4 is trying to formalize
the selection of stream-specific QoS in a transport-independent and
network-independent way.  Fernando and Casner are to explore what
formal or informal liaison procedures should be followed in this case,
both for information transfer and to enable reference to RTP in the
MPEG4 specification.

4.3 BT-656 video payload format

Dermot Tynan presented a proposal, draft-tynan-rtp-bt656-00.txt, for
carrying ITU-R BT.656-3 uncompressed video over RTP.  BT656 is
studio-quality digital video sampled according to BT.601-5 (formerly
CCIR601) at 13.5 or 18 MHz.  At the normal, lower rate, each scan line
contains 720 samples occupying 1440 bytes in the 4:2:2 chrominance
encoding.  At the "high definition" rate, each line contains 1144
samples for NTSC or 1152 samples for PAL.  The payload format consists
of a simple header with bit fields to indicate NTSC/PAL, sampling
rate, framing and scan line information, followed by one scan line of
samples.

Steve Casner expressed the concern that the packet size might exceed
the MTU for some networks.  Although at the lower sampling rate an
IP/UDP/RTP packet will fit within a 1500-byte MTU, at the higher
sampling it would not.  Don Hoffman pointed out that if the packet
size does exceed the MTU such that IP fragmentation occurs, integrated
services packet classification won't work on fragments other than the
first so those packets won't get the desired QoS.  Christian Huitema
claimed it is important to have application-level fragmentation so
that services such as forward error correction will apply when a
fragment is lost.  It should not cost much to be able to indicate that
a packet contained part of a scan line, perhaps by giving the starting
sample number.  Tynan agreed to consider this addition.

4.4 QuickTime payload format

Alagu Periyannan just asked to call the working group's attention to
the recently posted draft "RTP Payload Format for QuickTime Media
Streams", draft-ietf-avt-qt-rtp-00.txt, and in particular to the list
of open issues in section 4 of that draft.  The motivation in
developing this format was to carry all the payloads in QuickTime
without having to define an RTP payload format for each.

Steve Casner pointed out that the tradeoff of this technique is that
there is some amount of additional constant description info that must
be carried in each packet but would not be needed with separate
formats.  Philipp Hoschka asked if it wouldn't be better to define
individual payload formats because the encodings might also be used
outside of QuickTime as well.  Periyannan replied that where
individual payload formats are defined, they should be used in
preference to this format.  However, several of the encodings used
with QuickTime are not standardized, such as Apple Video, Cinepak, and
several proprietary codecs.

4.5 DTMF audio payload format

Jonathan Rosenberg presented Henning Schulzrinne's proposed audio
payload format to carry DTMF (tone dialing) signals as defined in
draft-ietf-avt-dtmf-00.txt and .ps.  This payload format might be used
for calling across the Internet through IP telephony gateways to
control an answering machine or other device.  Low data rate speech
codecs may not reproduce the DTMF tones faithfully enough to work
properly at the far end.  This payload format essentially provides a
very low rate encoding specialized for DTMF tones.

The draft defines a primary format in which each DTMF digit is
represented by 32 bits to control frequency, amplitude and duration.
Redundancy can be provided using the mechanism specified in RFC 2198.
This introduces 64 bits of overhead, but the data rate is so low that
this probably does not matter.  Alternatively, a more compact
representation of each digit would allow adding redundancy within the
DTMF payload format and still fitting each digit into 32 bits.  The
selection between these techniques is an open issue on which input
>from the working group is sought.  Steve Casner noted that the more
compact format would require the RTP clock rate to be different than
that of the normal audio payload format within which the DTMF packets
are interspersed, and suggested that this is a significant enough
disadvantage to prefer the larger format.  Scott Petrack agreed that
the size difference was probably not significant.

4.6 Forward Error Correction payload format

Two "meta" payload formats on forward error correction (that is,
independent of media type and format) have recently been posted:
draft-budge-media-error-correction-00.txt by Budge, et al., and
draft-ietf-avt-fec-00.txt by Rosenberg and Schulzrinne.  The authors
of the first draft did not attend, but Jonathan Rosenberg presented
the second draft which builds on ideas from the first, and compared
the two drafts.  Both schemes are based on the idea of sending
additional FEC packets which are the XOR of multiple packets from the
original packet stream.

It should be noted that the scheme presented by Rosenberg differs from
that described in the second draft.  An RTP header extension (X bit)
is no longer used; instead, the payload type is changed to indicate an
FEC packet, and an additional format-specific header is inserted
before the XOR of the covered payloads, as in the Budge scheme.

One drawback of the Budge draft was that the timestamp and marker bit
of lost packets could not always be recovered.  The proposed remedy is
for these fields in the FEC packets to be the XOR of the corresponding
fields from the original packets covered by the FEC packet.  This has
its own drawback that the timestamp will vary erratically, perturbing
the jitter feedback calculation unless the FEC packets are excluded.
Carsten Bormann also pointed out that when RTP header fields are
perturbed from their usual increments, it can have a negative impact
on the efficiency of RTP header compression.

A second drawback of the Budge scheme was that the payload type was
changed in the original packets to be the FEC payload type which means
that a receivers without FEC capability could not receive just the
original packets and ignore the FEC packets.  In Rosenberg's scheme,
the original packets are unmodified, including the payload type.

To extend the range of packet patterns that could be covered by an FEC
without having to predefine specific scemes, Rosenberg proposes to
carry a bit mask to indicate the pattern explicitly.  The RTP sequence
number of FEC packets in Rosenberg's proposal is the minimum of the
sequence numbers of the packets covered by the FEC to serve as a base
for the mask.  Steve Casner claimed that this is not acceptable
because it will prevent the FEC packets from passing header validation
and duplicate suppression algorithms in RTP packet processing.
Rosenberg said alternative sequence number schemes had been
considered, but the cost is additional overhead.

Christian Huitema argued that if a general FEC payload format is to be
defined, it should support other schemes with better performance and
only a marginal increase in complexity compared to that of "n+1"
schemes like XOR.  For example, it is possible to specify a scheme
that adds two FEC packets to eight original packets which will allow
recovery from a loss of any two of the ten packets.  This is important
because simulations have shown that one packet of redundancy was not
enough.  Rosenberg agreed that the FEC proposals need further
refinement.

4.7 Applicability of error correction

In addition to the redundant audio and FEC payload formats already
mentioned, retransmission and interleaving are two more loss
resilience schemes that might be employed with RTP.  Colin Perkins
reviewed these four methods to compare the tradeoffs in latency,
bandwidth overhead and processing overhead (details are available at
http://www.cs.ucl.ac.uk/staff/c.perkins/slides/).  It is an important
question to consider when these schemes are applicable to particular
applications or to particular network conditions.

Colin presented some network loss statistics showing that single
packet losses predominated by a factor of four over two-packet loss
bursts.  The probability drops rapidly such that long burst losses are
rare.  On the other hand, in a large conference, most packets will be
lost by at least one receiver, therefore a retransmission scheme may
need to resend almost every packet.

The redundant transmission scheme has low latency and low bandwidth
overhead, but potentially high processor overhead if the low-rate
redundant encoding is computationally complex.  A retransmission
scheme is likely to impose much longer delay and will incur the
overhead of control traffic in addition to the duplicate data, but
provides exact repair and can correct more than single-packet losses.
There is a synergy between redundancy and retransmission in that
redundancy can cover most of the errors, leaving retransmission to
pick up the remainder for those receivers that care more about low
loss than low delay.

Interleaving disperses the effects of loss, but does not eliminate
them.  It has low overhead, but high latency.  FEC may have a
similarly high latency or a high bandwidth overhead, depending on the
size of the pattern covered, but its major advantage is media and
format independence.  For interactive applications, either redundant
transmission or a low-latency FEC would seem most suitable, while for
broadcast-style applications interleaving works well.

An important open question is what constitutes a sensible operating
point for real-time media transmission?  How much loss should
applications try to cope with before declaring that the user should
try again at some less congested time?  There is little congestion
control in many real-time media applications, and extreme loss
resilience measures would just worsen congestion.  That would not be
network friendly.  If fair congestion avoidance mechanisms are
deployed in routers, then applications that don't implement congestion
control may be penalized.  Therefore, any loss resilience schemes that
are defined for RTP should consider not just loss performance but also
impact on the network.

5. Revision of RTP MIB specification

The final topic of the meeting was an update by Mark Baugher on the
RTP MIB design and implementation plans.  An interim draft of the MIB
was sent to the mailing list before the meeting to provide an
opportunity for feedback; a real Internet-Draft will be posted in
October.  A more complete description of the MIB was given in Memphis;
the changes since then were to fix errors identified by Fred Baker in
his review of the MIB, to add reporting of receiver feedback, to
support RTP operation over different underlying protocols, and to
remove from the specification those parts that won't be included in
initial implementions and hence won't be validated yet.  In
particular, support for RTP translators was removed.

Receiver feedback is reported through additional tables that are
created only upon request from the network manager as a side-effect of
creating an entry in the Session Table.  This avoids the state
explosion that might occur if all receiver feedback was always
monitored for all sessions.

There are a few details remaining to be worked out, but implementation
of a network management application using the RTP MIB is underway to
evaluate how well this MIB works for managing real-time applications
on networks that weren't designed for them.  To make this really
workable, the management application needs to handle multicast routing
in addition to RTP.  The next stage of MIB development will be based
on the results of the implementation.

6.  Next meeting

Steve Casner closed the meeting saying that AVT will meet again at the
next IETF in December.  As stated above, a goal for the working group
is to get the RTP spec revised for Draft Standard by then.  In
addition, there is outstanding work on several topics presented at
this meeting which should be discussed on the mailing list so that
completed drafts are ready by the next meeting.




From rem-conf Sat Sep 27 02:02:38 1997 
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From rem-conf Mon Sep 29 06:06:03 1997 
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Subject: Any RTP RTSP libraries available?
Organisation: Multi-media group, CWI, Kruislaan 413, Amsterdam
Phone: +31 20 5924098(work), +31 20 5924199 (fax), +31 20 6160335(home)
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Does anyone know of libraries to handle the RTP and RTSP protocols? I've come 
across PN's RTSP package, but the architecture of that is rather integrated 
and I would prefer something that was designed as a library to drop into other 
applications.
--
Jack Jansen             | ++++ stop the execution of Mumia Abu-Jamal ++++
Jack.Jansen@cwi.nl      | ++++ if you agree copy these lines to your sig ++++
http://www.cwi.nl/~jack | see http://www.xs4all.nl/~tank/spg-l/sigaction.htm 





From rem-conf Mon Sep 29 09:37:37 1997 
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From: Rob Lanphier <robla@real.com>
Subject: Re: Any RTP RTSP libraries available?
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At 02:55 PM 9/29/97 +0200, Jack Jansen wrote:
>Does anyone know of libraries to handle the RTP and RTSP protocols? I've
come 
>across PN's RTSP package, but the architecture of that is rather integrated 
>and I would prefer something that was designed as a library to drop into
other 
>applications.

Just to make things clear, RealNetworks (formerly Progresive Networks,
"formerly" being 4 days ago) offers two different implementations of RTSP.
The RTSP Reference kit was released pretty early on in the standardization
of RTSP primarily as a test bed, and has the properties you describe.  

The RealMedia SDK, released for developer beta earlier this year, is also
an implementation of RTSP, and is available for free download now.  It's
much more modular than the RTSP reference, and has a lot more features.
The downside is that it doesn't include source code, and that the
redistribution licensing is slightly trickier (but we are a pretty
reasonable bunch of folks to work with, IMHO :).  It also may not give you
the exact entry point into the system that you want.  For example, you can
create new datatypes and file formats, and it has hooks for adding new
headers on the DESCRIBE message, but doesn't allow you "down to the metal"
access to RTSP.

Other than that, it's a complete system, and is where we are putting the
bulk of our engineering effort (and have for the past year), so it's very
well-designed and is only going to get better.  A lot of our efforts have
been geared squarely at folks making tools (such as yourselves), so I think
it's worth looking at.

Rob

---
Rob Lanphier (robla@real.com)    Voice: (206)674-2322   Fax: (206)674-2699
RealNetworks:  http://www.real.com
RTSP Info:     http://www.real.com/rtsp
Firewall Info: http://www.real.com/help/firewall  




