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--NextPart

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

       Title     : Compressing IP/UDP/RTP Headers for Low-Speed Serial 
                   Links                                                   
       Author(s) : S. Casner, V. Jacobson
       Filename  : draft-ietf-avt-crtp-01.txt
       Pages     : 16
       Date      : 11/27/1996

This document describes a method for compressing the headers of IP/UDP/RTP 
datagrams to reduce overhead on low-speed serial links. In many cases, all 
three headers can be compressed to 2-4 bytes.         
                     
Comments are solicited and should be addressed to the working group mailing
list rem-conf@es.net and/or the author(s).  This draft updates 
draft-casner-jacobson-crtp-00.txt which was sent to the rem-conf list but 
was never officially posted as an Internet-Draft due to a mail lossage that
then left it out of date.  At the Montreal IETF meeting in June 1996, this 
proposal was accepted as a work item of the the Audio/Video Transport 
working group, hence the draft name change.                                

Internet-Drafts are available by anonymous FTP.  Login with the username
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From rem-conf-request@es.net Mon Dec 02 14:53:23 1996 
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--NextPart

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

       Title     : RTP Payload Format for H.263 Video Stream               
       Author(s) : C. Zhu
       Filename  : draft-ietf-avt-rtp-payload-02.txt
       Pages     : 11
       Date      : 11/26/1996

This document specifies the RTP payload format for encapsulating H.263 
bitstreams in the Real-Time Transport Protocol (RTP). Three modes are 
defined for RTP H.263 payload header. An RTP packet can use one of the 
three modes for H.263 video streams depending on the desired performance 
characteristics and H.263 encoding options employed. The shortest header 
mode (Mode A) supports fragmentation at Group of Block (GOB) boundaries. 
The long header modes (Mode B and C) support fragmentation at Macroblock 
(MB) boundaries.                                                           

Internet-Drafts are available by anonymous FTP.  Login with the username
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NOTE: The mail server at ds.internic.net can return the document in
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      feature, insert the command "ENCODING mime" before the "FILE"
      command.  To decode the response(s), you will need "munpack" or
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      "multipart" MIME messages (i.e., documents which have been split
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From rem-conf-request@es.net Mon Dec 02 20:04:59 1996 
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subscribe

From rem-conf-request@es.net Tue Dec 03 10:13:47 1996 
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Date: Tue, 3 Dec 1996 10:13:16 -0400
To: rem-conf@es.net
From: macker@itd.nrl.navy.mil (Joseph P. Macker)
Subject: I-D ACTION:draft-macker-mdp-framework-01.txt

FYI:
> A Revised Internet-Draft is available from the on-line Internet-Drafts 
> directories.                                                              
>
>       Title     : The Multicast Dissemination Protocol (MDP) Framework    
>       Author(s) : J. Macker, W. Dang
>       Filename  : draft-macker-mdp-framework-01.txt
>       Pages     : 14
>       Date      : 11/27/1996
>
>This document outlines a simple protocol framework for reliable multicast 
>dissemination of data files. The general framework was originally developed
>and used by the Image Multicaster (IMM) application within the Internet 
>MBone for dissemination of satellite imagery.  This document describes the 
>potential for more general use of the protocol framework, its operational 
>modes, some performance issues, and the basic application data units (ADUs)
>presently used.  This is not intended to be a detailed protocol 
>specification document, but rather a broad description of the basic 
>architectural approach.   Further detailed description of the protocol 
>implementation may be provided in future documents.                        
>



From rem-conf-request@es.net Tue Dec 03 14:02:17 1996 
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To: 298-list@bmrc.Berkeley.EDU
From: Jerrlyn Iwata <jerrlyn@postgres.berkeley.edu>
Subject: Berkeley Multimedia & Graphics Seminar

                Berkeley Multimedia and Graphics Seminar 
       (Wednesday December 4, 1996 12:30-2:00 PDT 405 Soda Hall) 

      "The Architecture and Operation of the Berkeley Campus Network" 

                              Cliff Frost 
                 Data Communications & Network Services 
                             U.C. Berkeley 

The Internet has become an extremely well-known entity, appearing almost
routinely on the front pages of such major newspapers as the New York Times,
the Wall Street Journal, and the Daily Californian. As the Internet and the
campus network become more visible, more and more people are interested in
how it is designed and put together. 

This talk will cover how the Berkeley network has grown up over the last few
years, including various design decisions along the way and the current
major issues facing us, from technical problems to funding models to hiring
woes to logistics. 
--------------------------------------------------------------------------------
This seminar will be broadcast on the Internet MBONE.  The broadcast will
begin at 12:40PST (GMT - 8 hrs).  See sd or http://bmrc.berkeley.edu/298 for
instructions on setting up, connecting to, and operating the MBONE tools.


From rem-conf-request@es.net Tue Dec 03 15:46:29 1996 
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Message-ID: <32A52BA8.6718@bitfield.fi>
Date: Tue, 03 Dec 1996 23:43:36 -0800
From: Esa Vitikainen <Esa.Vitikainen@bitfield.fi>
Organization: Bitfield Oy
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Newsgroups: alt.education.distance,comp.human-factors,comp.multimedia,comp.dcom.videoconf
CC: t120-interest@world.std.com, videophone@es.net, vidconf@pulver.com, 
    rem-conf@es.net
Subject: Seeking visions/experiences of multipoint data apps
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Hi all,

as part of my studies I am planning to find out just what are 
users / developers / service providers / VARs / ...
doing / going to do / would like to do 
with data in multipoint multimedia conferencing context. 
In other words, I'm chasing visions and experiences of 
applications adding value to voice/video. 

There are the apparent chat, file transfer, whiteboard, 
and application sharing. But how are they really utilized? 
Where do they fall short? What else is needed out there? 

Are there studies or something about this available somewhere? 
Places to ask around? Willing to reveal your own great idea? 

An example: features I imagine needed for remote presentation:
- slideshow operated by the presenter
- "hands raised" electronically (with text description?)
- slides independently leafed through / edited locally
  (like having paper copies)
- others taking over the slideshow control, with own remarks
- dialogs sent to ask for comments / test understanding, 
  with reply statistics shown (for large audience)

Regards,
Esa

From rem-conf-request@es.net Wed Dec 04 08:54:21 1996 
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From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
Message-Id: <961204085100.ZM10958@arrakis>
Date: Wed, 4 Dec 1996 08:50:54 -0700
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[I apologize if anyone gets this multiple times - I have been having 
some problems sending to rem-conf]

AVT group:

Henning Schulzrinne and I have been investigating the issue of voice 
aggregation within RTP. The premise is to place voice data from many 
users within a single RTP session (and packet). Such aggregation is 
possible in scenarios where you are connecting PBX's, telephone 
switches, or any other device which handles multiple voice users, via 
Internet. We have prepared a draft which discusses the advantages of 
such aggregation, and then looks at defining a new RTP profile for this 
application.

If you are interested, you can find a copy of the draft in html form at 
http://www.cs.columbia.edu/~hgs/rtp/aggregation. Please feel free to 
address any comments or questions to the authors.

Thanks,

Jonathan Rosenberg

-- 
Jonathan Rosenberg			Lucent Technologies
Member of Technical Staff		Bell Laboratories
High Speed Networks Research		101 Crawfords Corner Rd.
PHONE: (908) 949-6418			Holmdel, NJ 07733
FAX:   (908) 949-9118			Rm. 4G-530
EMAIL: jdrosen@bell-labs.com

From rem-conf-request@es.net Wed Dec 04 12:00:44 1996 
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From: Toerless Eckert <Toerless.Eckert@Informatik.Uni-Erlangen.de>
Message-Id: <199612041659.RAA26760@faui40.informatik.uni-erlangen.de>
Subject: vic tweaking question
To: rem-conf@es.net
Date: Wed, 4 Dec 1996 17:59:51 +0100 (MET)
Organisation: CSD IMMD IV, University of Erlangen, Germany
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Hi

I seem to be a tcl ignorant: 
I am trying to modify the vic startup file so that a the stamp-size
window by default is muted. This effectively increases the sender rate
by some frames if the stamp-size picture is remote.

I only failed to get it done.

Any help here ?

Thanks
	Toerless

From rem-conf-request@es.net Wed Dec 04 13:53:44 1996 
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Date: Wed, 4 Dec 1996 13:52:06 -0500 (EST)
From: kevin@cc.gatech.edu (Kevin C. Almeroth)
Message-Id: <199612041852.NAA19416@flora.cc.gatech.edu>
To: rem-conf@es.net
Subject: Re: vic tweaking question

>>I seem to be a tcl ignorant: 
>>I am trying to modify the vic startup file so that a the stamp-size
>>window by default is muted. This effectively increases the sender rate
>>by some frames if the stamp-size picture is remote.

Or a more general question...  is the Tcl interface designed to this
type of thing?

For example, I'd like to start a certain instance of vic with the
stamp-size window muted but with an automatic 1/4 NTSC window showing.

-Kevin Almeroth

From rem-conf-request@es.net Wed Dec 04 14:50:03 1996 
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To: Jerrlyn Iwata <jerrlyn@postgres.berkeley.edu>
cc: 298-list@bmrc.Berkeley.EDU, dap@aber.ac.uk
Subject: Re: Berkeley Multimedia & Graphics Seminar
In-reply-to: Your message of Tue, 03 Dec 1996 10:55:02 -0800. <199612031855.KAA03457@nobozo.CS.Berkeley.EDU>
Date: Wed, 04 Dec 1996 19:32:30 +0000
Message-ID: <22330.849727950@mailhost.aber.ac.uk>
From: D E PRICE <dap@mailsun.aber.ac.uk>

Dear All,
	I am sitting hoping to watch the UCB seminar a little later,
when I just noticed that the sdr announcement is still referring
to the last seminar and my sdr has last seen an announcement
on 29nov at 19:34. Do I need to worry ? or is it normally
like this with new announcements only being seen just
before the broadcast starts?

Thanks,

Dave Price, UW, Aberystwyth. Wales. UK.

From rem-conf-request@es.net Wed Dec 04 17:36:38 1996 
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To: 298-list@bugs-bunny.CS.Berkeley.EDU
Subject: UCB Multimedia Seminar update
From: David Simpson <davesimp@cs.berkeley.edu>
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I have been getting reports from a few people that they are not
receiving the seminar announcement.  I'm not sure if this is a problem on
their end, on our end, or the fact that I'm using the sapAnnouncer to make
the announcements instead of sdr.  At any rate, here is the correct 
address/ports for the seminar:

video 224.2.238.31/51900
audio 224.2.142.37/24882
wb    224.2.158.81/48748

And the IPTV Program Guide at Precept seems to be receiving the announcement
fine, so if you are not receiving an annoucement try checking 
http://www.precept.com/cgi-bin/iptv/iptvmain.pl

-Dave


-- 
----------------------------------------------------------------------------
	         David Simpson, davesimp@cs.berkeley.edu
	    Graduate Student - Computer Science - UC Berkeley
	       http://www-plateau.cs.berkeley.edu/~davesimp



From rem-conf-request@es.net Wed Dec 04 21:26:27 1996 
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To: David Simpson <davesimp@cs.berkeley.edu>
cc: 298-list@bugs-bunny.CS.Berkeley.EDU
Subject: Re: UCB Multimedia Seminar update
In-reply-to: Your message of "Wed, 04 Dec 96 12:12:04 PST." <199612042012.MAA16579@dilbert.CS.Berkeley.EDU>
Date: Wed, 04 Dec 96 21:11:28 -0500
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Sender: mjh@mercury.lcs.mit.edu
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>I have been getting reports from a few people that they are not
>receiving the seminar announcement.  I'm not sure if this is a problem on
>their end, on our end, or the fact that I'm using the sapAnnouncer to make
>the announcements instead of sdr.  At any rate, here is the correct 

This is probably mrouted rate-limiter overload.  For historic reasons
sdr announces sessions in the low-priority range, and recently we've
seen a lot more mbone traffic than we have historically, so
low-priority traffic is often getting lost.  Basically 512Kbps is
not sufficient anymore and that;'s the default rate-limiter in mrouted.

Mark

From rem-conf-request@es.net Wed Dec 04 22:08:20 1996 
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To: Toerless Eckert <Toerless.Eckert@informatik.uni-erlangen.de>
cc: rem-conf@es.net
Subject: Re: vic tweaking question
In-reply-to: Your message of "Wed, 04 Dec 96 08:59:51 PST." <199612041659.RAA26760@faui40.informatik.uni-erlangen.de>
Date: Wed, 4 Dec 1996 19:07:23 PST
Sender: Bill Fenner <fenner@parc.xerox.com>
From: Bill Fenner <fenner@parc.xerox.com>
Message-Id: <96Dec4.190724pst.177711@crevenia.parc.xerox.com>

Have you tried the X resource Vic*muteNewSources: true?

  Bill

From rem-conf-request@es.net Wed Dec 04 22:30:39 1996 
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To: kevin@cc.gatech.edu (Kevin C. Almeroth)
cc: rem-conf@es.net
Subject: Re: vic tweaking question
In-reply-to: Your message of "Wed, 04 Dec 96 10:52:06 PST." <199612041852.NAA19416@flora.cc.gatech.edu>
Date: Wed, 4 Dec 1996 19:29:27 PST
Sender: Bill Fenner <fenner@parc.xerox.com>
From: Bill Fenner <fenner@parc.xerox.com>
Message-Id: <96Dec4.192940pst.177711@crevenia.parc.xerox.com>

In message <199612041852.NAA19416@flora.cc.gatech.edu> you write:
>For example, I'd like to start a certain instance of vic with the
>stamp-size window muted but with an automatic 1/4 NTSC window showing.

Try something like this.  I don't know how kosher the detach_window
call is, you might prefer to just set the X resource Vic.stampInterval
to something high.

rename build.src build.src.real

proc build.src { w src color } {
        set r [build.src.real $w $src $color]

	detach_window $src $w.stamp.video

        global userwin_x userwin_y userwin_size

        # Or wherever you want the window to pop up
        set userwin_x($src) 0
        set userwin_y($src) 0
        set userwin_size($src) "320x240"

        open_window $src

        return $r
}

If you don't want the main vic window hanging around, you could add
something like

proc user_hook {} {
	wm iconify .
}

  Bill

From rem-conf-request@es.net Thu Dec 05 05:01:03 1996 
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From: Toerless Eckert <Toerless.Eckert@Informatik.Uni-Erlangen.de>
Message-Id: <199612051000.LAA16278@faui40.informatik.uni-erlangen.de>
Subject: Re: vic tweaking question
To: fenner@parc.xerox.com (Bill Fenner)
Date: Thu, 5 Dec 1996 11:00:16 +0100 (MET)
Cc: Toerless.Eckert@informatik.uni-erlangen.de, rem-conf@es.net
In-Reply-To: <96Dec4.190724pst.177711@crevenia.parc.xerox.com> from "Bill Fenner" at Dec 4, 96 07:07:23 pm
Organisation: CSD IMMD IV, University of Erlangen, Germany
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> Have you tried the X resource Vic*muteNewSources: true?

Yes. But it is my own source, so maybe that's the reason why it's not
working..

From rem-conf-request@es.net Thu Dec 05 06:44:05 1996 
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From: apiszcz@vector2.mitre.org (Al Piszcz 'peesh')
Message-Id: <199612051144.GAA12155@vector2.>
Reply-To: Alan Piszcz <apiszcz@mitre.org>
To: rem-conf@es.net
Subject: VIC/nv file play option
X-Sun-Charset: US-ASCII


Is there a method available for playing an MPEG file/stream into
nv or vic?


      :-----------------------------------------------------------:
      : Al Piszcz                               apiszcz@mitre.org :
      : MITRE Corporation    Distributed Systems and Technologies :
      : 703.883.7124                             FAX 703.883.3308 :
      :-----------------------------------------------------------:


From rem-conf-request@es.net Thu Dec 05 08:48:20 1996 
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Date: Thu, 5 Dec 1996 08:44:47 -0700
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To: rem-conf@es.net
Subject: Printing draft-rosenberg-itg-00.ps
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My apologies to those people who had trouble printing the postscript 
version of draft-rosenberg-itg.00.ps. For some reason, some 40kB of data 
was truncated from the files when they were emailed to the ietf. The 
problem is fixed. A correct ps (and txt) version are now available from 
the internet-drafts directory (http://www.ietf.org/ids.by.wg/none.html).

Thanks,

Jonathan Rosenberg

-- 
Jonathan Rosenberg			Lucent Technologies
Member of Technical Staff		Bell Laboratories
High Speed Networks Research		101 Crawfords Corner Rd.
PHONE: (908) 949-6418			Holmdel, NJ 07733
FAX:   (908) 949-9118			Rm. 4G-530
EMAIL: jdrosen@bell-labs.com

From rem-conf-request@es.net Thu Dec 05 11:28:45 1996 
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Subject: DMS97 CFP
To: rem-conf@es.net
Date: Thu, 5 Dec 1996 11:27:52 -0500 (EST)
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                               CALL FOR PAPERS

          1997 Pacific Workshop on Distributed Multimedia Systems

             University of British Columbia, Vancouver, Canada

                              July 24-25, 1997

The last few years have seen an explosive growth of multimedia computing,
communication and applications. This revolution is transforming the way
people live, work, and interact with each other, and is impacting the way
businesses, government services, education, entertainment, and health care
are operating. It is safe to say that the multimedia revolution is underway.
Yet, several issues related to modeling, specification, analysis and design
of distributed multimedia systems and applications are still challenging to
both researchers and practitioners.

The purpose of this workshop is to serve as a forum for the exchange of
ideas among practicing engineers and researchers from around the world, as
well as highlight current activities and important topics in the field of
distributed multimedia systems and technology. The paper session is designed
to promote discussion of concepts, methodologies, and results between the
authors and the audience, and provide for a degree of collegiality and
continuity in the discussions of these various topics among the
participants. In addition, tutorials will be held on June 25 and 26 to
provide participants with the opportunity to interact with experts in
various fields of distributed multimedia systems.

The workshop organizers seek contributions of high quality papers, panels or
tutorials, addressing various aspects of distributed multimedia systems and
applications, for presentation at the conference and publication in the
workshop proceedings. Topics of interest include, but are not limited to:

     [*] Distributed Multimedia Databases and Computing
     [*] Multi-paradigmatic Information Retrieval
     [*] Modeling and Analysis of Distributed multimedia systems
     [*] OS Support for Distributed Multimedia systems
     [*] Multimedia Communications and Networking
     [*] Multimedia Digital Libraries and Mail Systems
     [*] Multimedia Human-Computer Interaction
     [*] Visual and Multidimensional Languages for multimedia applications
     [*] Multimedia Workflows
     [*] Multimedia Stream Synchronization
     [*] Virtual Reality, Virtual Environment, and their Applications

The use of prototypes and demonstration video for presentations is
encouraged.

Workshop Site:
The workshop will be held at the University of British Columbia, Vancouver,
Canaca. Special arrangement will be made with local hotels for a limited
number of rooms at the special workshop rate.

Information for Authors:
Submit four copies of panel or tutorial proposals or manuscripts (maximum of
20 double-spaced pages) including abstract, keywords and complete
information of a contact person to one of the following program co-chairs:

                            Professor Taieb Znati
                          Dept. of Computer Science
                          University of Pittsburgh
                          Pittsburgh, PA 15260 USA
                            Tel: +1 412-624-8417
                            Fax: +1 412-624-8854
                          Email: znati@cs.pitt.edu

                            Professor An-Chi Liu
                        Dean, College of Engineering
                            Fong-Chia University
                              Tai-Chung, Taiwan
                            Tel: +886-4-252-7110
                            Fax: +886-4-323-9903
                        Email: liu@fcusqnt.fcu.edu.tw

Papers, tutorials and panels can also be directly deposited by anonymous ftp
at ftp.cs.pitt.edu:/pub/DMS97/Paper-Deposit. Submitted files must be in
PostScript format with all figures and tables included.

Important Dates:

   * January 15, 1996:       Paper\Proposal submission deadline
   * March 15, 1997:         Notification of acceptance
   * April 15, 1997:         Final manuscript due

Sponsored by:

   * Knowledge Systems Institute, USA
   * University of British Columbia, Canada
   * University of Pittsburgh, USA
   * Fong-Chia University, Taiwan

Workshop Co-Chairs:

   * Son T. Vuong, University of British Columbia, Canada
   * S. K. Chang, University of Pittsburgh, USA

Program Committee Co-Chairs:

   * Taieb Znati Univ. of Pittsburgh, USA
   * An-Chi Liu Fong-Chia University, Taiwan

Local Arrangement Chair:

   * Panos Nasiopoulos University of British Columbia, Canada

Program Committee:

   * Hong-Mei Chen Univ. of Hawaii, USA
   * Wen-Tsuen Chen National Tsing Hua University
   * Yanghee Choi Seoul National University
   * Jean-Pierre Courtiat Centre National de Recherche Scientifique, France
   * Gerald Neufeld University of British Columbia
   * Dieu D. Phan National Program on Information Technology, Vietnam
   * Anindya Datta University of Arizona, Tucson AZ
   * David H. C. Du Univ. of Minnesota, USA
   * Nicolas D. Georganas Depart. of Elec. and Comp. Eng., Univ. of Ottawa
   * Norm Hutchinson University of British Columbia BC
   * William Grosky Wayne State Univ., USA
   * Venkat N. Gudivada University of Missouri-Rolla
   * Arding Hsu Siemens Corporate Research, USA
   * Oscar Ibarra Univ. of Cal., Santa Barbara, USA
   * Stephen Y. Itoga University of Hawaii
   * Paul Lin CCL, Taiwan
   * Tom D.C. Little<\b> University of Boston , USA
   * Ralph Martinez ECE, Radiology, & Pathology Dept. Univ. of Arizona
   * J. Mizusawa NTT Multi-media Network Laboratories
   * Ming-Chien Shan Hewlett Packard Labs
   * Olivia R. Liu Sheng HKUST, Hong Kong
   * Jaideep Srivastava Univ. of Minnesota, USA
   * Kazuo Sugihara Univ. of Hawaii, USA
   * Joseph E. Urban Arizona State Univ., USA
   * Chun-Chia Wang Tamkang University, Taiwan, R.O.C.
   * Don-Lin Yang Feng Chia University, Taichung, Taiwan

For additional information, please send mail to znati@cs.pitt.edu or
liu@fcusqnt.fcu.edu.tw. Updated conference announcement and information are
accessible by anonymous ftp at ftp.cs.pitt.edu:/pub/DMS97 or by directly
accessing the conference home page at http://www.cs.pitt.edu/DMS97.







From rem-conf-request@es.net Thu Dec 05 14:24:38 1996 
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To: Mark Handley <mjh@isi.edu>
cc: David Simpson <davesimp@cs.berkeley.edu>, 
    298-list@bugs-bunny.CS.Berkeley.EDU, hopper@cs.ucsd.edu
Subject: Re: UCB Multimedia Seminar update
In-reply-to: Your message of "Wed, 04 Dec 1996 21:11:28 EST." <16463.849751888@mercury.lcs.mit.edu>
Date: Thu, 05 Dec 1996 11:09:24 -0800
From: Steve Hopper <hopper@cs.ucsd.edu>


This probably explains why SDR displays fewer entries during the day,
which then appear again in the evening.  The next day the entries again 
diminish to ~ half the number.


Steve

--------
In reply to the following message:
--------
 >
 >>I have been getting reports from a few people that they are not
 >>receiving the seminar announcement.  I'm not sure if this is a problem on
 >>their end, on our end, or the fact that I'm using the sapAnnouncer to make
 >>the announcements instead of sdr.  At any rate, here is the correct 
 >
 >This is probably mrouted rate-limiter overload.  For historic reasons
 >sdr announces sessions in the low-priority range, and recently we've
 >seen a lot more mbone traffic than we have historically, so
 >low-priority traffic is often getting lost.  Basically 512Kbps is
 >not sufficient anymore and that;'s the default rate-limiter in mrouted.
 >
 >Mark
--------

From rem-conf-request@es.net Thu Dec 05 16:22:22 1996 
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From: brutzman@cs.nps.navy.mil (Don Brutzman)
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Subject: ANNOUNCE: VRML 97
To: ietf@cnri.reston.va.us (IETF "big Internet" list), 
    mbone@isi.edu (Multicast Backbone mail list), 
    rem-conf@es.net (Remote Conferencing mail list), 
    lsma@gmu.edu (Large Scale Multicast Application WG)
Date: Thu, 5 Dec 1996 13:16:31 -0800 (PST)
Cc: ljohnson@redshift.com (Lynn Johnson)
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         [standard apologies if you receive duplicate copies]


You are cordially invited to attend the second annual Virtual Reality 
Modeling Language (VRML) Symposium.  VRML is where 3D graphics meets
the Web.  Internetworking considerations are thus just as important as
interactive 3D in this context.  A schedule and list of papers/courses/panels 
follows the press release.  Queries welcome.  Thanks!    - Don Brutzman


FOR MORE INFORMATION CONTACT:
Lynn Johnson
Evans & Johnson
408 655 9924
408 372 0846 fax
ljohnson@redshift.com

FOR IMMEDIATE RELEASE

VIRTUAL REALITY MODELING LANGUAGE (VRML) SYMPOSIUM SET FOR FEBRUARY 1997

Monterey, California, November 25, 1996 - The Second Annual Virtual Reality
Modeling Language (VRML) Symposium is scheduled for February 24-26, 1997 at 
the Hyatt Regency in Monterey, California.  VRML 97 is an academic and 
technical symposium sponsored by ACM SIGGRAPH and SIGCOMM, in cooperation 
with the VRML Consortium.

This year's VRML Symposium provides a wide range of technical papers, 
educational opportunities, cutting-edge demonstrations and exhibits 
featuring the latest in VRML technology.  The VRML Symposium home page is 
http://www.sdsc.edu/vrml97

Who and What.  300-500 attendees from across the globe are expected to attend 
this powerful 3-day technical symposium.  Attendees can attend half-day or 
full-day tutorials, which are designed for both intermediate and advanced 
VRML users and developers.  The Symposium also includes working groups,
panels, academic papers and in-depth exhibitor demonstrations.  Topics 
include 3D graphics, content, behaviors, interfaces, modeling, simulation, 
multi-user environments, networking and standards. 

Why.  "VRML 97 is the future of 3D on the World Wide Web," says VRML 97 chair 
Don Brutzman. "All of the most exciting work in internetworked 3D graphics
will be shown, discussed and argued over by key players in this 
fast-moving community.  Don't miss it!" 

Courses.  The first day (Monday, Feb 24) will feature both full-day and 
half-day  tutorial courses.  The first full-day course is "Introduction to 
VRML 2.0" presented by Dave Nadeau of the San Diego Supercomputer Center.  
This course, designed for beginning graphics users,  will teach attendees how 
to use VRML to author their own 3D virtual worlds on the World Wide Web.  
"Authoring Compelling, Efficient VRML 2.0 Worlds" presented by Dave Story, 
Delle Maxwell and David Marsland, all of Silicon Graphics, is another full day 
tutorial course.  This course provides authors with a concrete toolset for 
overcoming the limitations and exploring the unique capabilities of VRML 2.0.  
Attendees will learn new approaches to solve their creative challenges. 

More Courses.  Half-day tutorials include "The VRML 2.0 Compressed Binary 
Format" presented by Gabriel Taubin, Bill Horn and Francis Lazarus, all with 
IBM.  This course will familiarize attendees with the fundamentals of the 
compression technologies proposed for the VRML 2.0 standard.  Another half-day
tutorial is "Sound Bytes VRML Authoring for Noisy Worlds" presented by 
Geoff Brown of Silicon Graphics Inc.  This course features demonstrations 
that show effective use of sound in large and small worlds and are presented 
using both PC and SGI platforms.  This tutorial is suitable for beginners and
intermediate developers.  "Networking Services for Shared Virtual Worlds" 
presented by Eric Hoffman is offered as a beginning networking tutorial.
This course will be extremely helpful for those who are considering the 
design of distributed virtual worlds, but are unfamiliar with recent work
in transport protocols. 

Exhibitors.  More than 30 companies will demonstrate the latest tools, 
technology and software at the exhibitor's showcase on Tuesday and Wednesday 
(Feb 25-26).  The exhibitor's showcase serves as a primary resource to learn 
about the latest VRML products.  Not an exhaustive trade show, this group 
includes only the best and brightest VRML developers giving one-on-one demos 
of new and forthcoming work to the most influential people working in VRML.  
An exhibitors' planning guide including detailed descriptions of products and
services will be distributed to all attendees. 

Technical Program.  Tuesday and Wednesday's program includes paper 
presentations and panels.  A solid and highly select set of academic papers 
show research work and results that may affect the future of the VRML standard.
Industry experts will explain and debate controversial topics from a variety 
of viewpoints.  A complete listing of sessions and panels is available at 
http://www.sdsc.edu/vrml97/schedule.html

Event:  Demo SIG.  All Symposium attendees are invited to the no-holds-barred 
Demo SIG hosted by the Virtual Reality Education Foundation (VeRGe).  This 
exciting two-hour presentation, scheduled 5pm Monday (Feb 24), is an intense 
series of live 5-minute internetworked 3D demonstrations.  This event was one 
of the highlights of SIGGRAPH in New Orleans August 1996.  The Demo SIG is 
followed by a dinner and reception hosted by Microsoft Corporation.

Event:  Aquarium Dinner.  Tuesday evening's activities include dinner in front
of the Monterey Bay Aquarium's new 1.2 million gallon Outer Bay exhibit.  
Attendees will enjoy a strolling dinner in front of the largest viewing window 
in the world.  This spectacular and intense exhibit is 35 feet deep, 90 feet 
long, holds over 1 million gallons of water and houses animals rarely viewed,
such as the giant tunas, ocean sunfish, a green sea turtle and soupfin sharks. 
Attendees also will have private access to the world's largest exhibit of 
jellyfish, including transparent crystal jellies and bioluminescent
comb jellies.  This exciting evening is hosted by Silicon Graphics Inc.

VRML Consortium.  The long-term future of VRML has always been a key question.  
AT VRML 97, the newly elected VRML Consortium Board of Directors will report 
on efforts to make VRML an integral part of the Internet infrastructure.  The 
VRML Consortium is a non-profit group made up of corporate and academic 
organizations, all working together to promote VRML standards, outreach and 
education. 

VRML 97 Committee.  Chair Don Brutzman, papers chairs Rikk 
Carey and Paul Strauss, Mitra, Adrian Scott, Glenn Crocker, Stephen Spencer, 
Bill Cockayne, Erkan Akyuz, Dan Ancona, Clark Dodsworth, D.J. LeGoff,
Kent Watsen, Robert Saint John, Lisa Goldman, Mark Meadows, Mark Pesce, 
David Nadeau, Timothy Childs, Gray Schlichting, Gavin Bell, Jean-Francis 
Balaguer, Jan Hardenbergh, David Kerlick, Stephen Matsuba, Tamara Munzner, 
Tom Meyer, Tony Parisi, Erika Whiteway, Dick Puk and Val Watson.  Individual 
committee positions & e-mail addresses of these expert volunteers are 
available at http://www.sdsc.edu/vrml97/committee.html

Lodging and Registration.  The Hyatt Regency Monterey is headquarters hotel 
for VRML 97.  For registration info, please refer to the VRML 97 home page 
http://www.sdsc.edu/vrml97/register.html
or contact Lynn Johnson at 408-655-9924 or ljohnson@redshift.com 
Early-bird registration discounts are available through December 15 1996,
prices jump sharply thereafter.

END RELEASE



  VRML 97
  
  FEBRUARY 24-26, 1997
  HYATT REGENCY HOTEL
  MONTEREY CALIFORNIA USA
  
    http://www.sdsc.edu/vrml97/schedule.html
    
  SCHEDULE
  
  SUNDAY, FEBRUARY 23, 1997
     * Working group meetings by request (Hyatt or NPS)
     * 12:00 PM Exhibitor setup open (prior request only, until 5)
     * 13:00 PM VRML Consortium Board of Directors (private)
     * 5:00 PM Registration open (refreshments available)
     * 6:00 PM Sponsors reception (private)
     * 9:00 PM Registration closed
       
  MONDAY, FEBRUARY 24, 1997
     * 7:00 AM Registration opens
     * 7:30 AM Continental breakfast (if sponsored)
     * 8:00 AM Exhibitor & AV setup open
     * 8:30 AM Full-Day Courses
          + Introduction to VRML 2.0, Dave Nadeau, San Diego
            Supercomputer Center
          + Authoring Compelling, Efficient VRML 2.0 Worlds, Dave Story,
            Delle Maxwell and David Marsland, Silicon Graphics Inc.
     * 8:30 AM Half-Day Courses 
          + The VRML 2.0 Compressed Binary Format, Gabriel Taubin, Bill
            Horn and Francis Lazarus, IBM
     * 8:30 AM Working groups (TBD - to be determined)
     * 10:00 AM Break
     * 10:30 AM Courses & working groups resume
     * 12:00 PM Lunch (box lunch or buffet TBD)
     * 1:00 PM Half-Day Courses
          + Networking Services for Shared Virtual Worlds, Eric Hoffman,
            National Laboratory for Applied Network Research / Ipsilon
          + Sound Bytes VRML Authoring for Noisy Worlds, Geoff Brown,
            Silicon Graphics Inc.
     * 1:00 PM Working groups (TBD)
     * 2:30 PM Break
     * 3:00 PM Courses & working groups resume
     * 4:00 PM Demo SIG and AV rehearsal (ballroom)
     * 4:30 PM Break
     * 5:00 PM Demo SIG (ballroom)
     * 7:00 PM Dinner & Reception at the Hyatt (sponsored)
       
  TUESDAY, FEBRUARY 25, 1997
     * 7:00 AM Registration opens
     * 7:15 AM Continental breakfast (if sponsored)
     * 8:15 AM Welcome
     * 8:30 AM Papers: Implementation I, Val Watson moderator
          + Using spatial techniques to decrease message passing in a
            distributed VE system, Olof Hagsand, Rodger Lea and Marten
            Stenius, Swedish Institute of Computer Science (SICS) and
            Sony Computer Science Lab
          + An Object Oriented Approach to VRML Development, Curtis A.
            Beeson, Silicon Graphics Inc.
          + Object-Oriented VRML for Multi-user Environments, Sungwoo
            Park
     * 10:00 AM Break
     * 10:00 AM Exhibits open
     * 10:30 AM Papers: Multi-user, Dave Nadeau moderator
          + Populating the Internet: Supporting Multiple Users and Shared
            Applications with VRML, Wolfgang Broll, German National
            Institute for Information Technology
          + Community Place: architecture and performance, Rodger Lea,
            Yasuaki Honda, Kouichi Matsuda and Satoru Matsuda, Sony
            Computer Science Lab
          + SpaceFusion: A multi-server architecture for shared virtual
            environments, Hiroyasu Sugano, Fujitsu
     * 12:00 PM Consortium Report: Elected VRML Consortium Inc. Board of
       Directors
     * 12:30 PM Lunch
     * 2:00 PM Papers: Navigation and User Interface, Dave Kerlick
       moderator
          + QOTA: A Fast, Multi-Purpose Algorithm For Terrain Following
            in Virtual Environments, John W. Barrus and Richard Waters,
            Mitsubishi Electric Research Laboratory
          + MaPS: Movement and Planning Support for Navigation in an
            Immersive VRML Browser, John Edwards and Chris Hand, De
            Montfort University, Leicester UK
          + Adapting VRML 2.0 for Immersive Use, Randy Stiles, Sandeep
            Tewari and Mihir Mehta, Lockheed Martin Advanced Technology
            Center
     * 3:30 PM Break
     * 4:00 PM Panel: Avatar and Multi-user Standards, Moderator Bernie
       Roehl - University of Waterloo, Mitra - ParaGraph, David Anderson
       - MERL, Yasuaki Honda - Sony Computer Science Lab, et al.
     * 5:00 PM Panel: Using Java as a VRML development platform,
       Moderator: Michael Stein - Protozoa, Chris Marrin - SGI, Chris
       Laurel - Dimension X, Rodger Lea - Sony Computer Research
       Laboratory, et al.
     * 6:00 PM Break
     * 6:00 PM Exhibits close
     * 6:45 PM Motorcoaches depart for Monterey Bay Aquarium
     * 7:00 PM Monterey Bay Aquarium Dinner & Entertainment (sponsored)
     * 11:00 PM Motorcoach shuttle back to hotel
       
  WEDNESDAY, FEBRUARY 26, 1997
     * 7:00 AM Registration opens
     * 7:30 AM Continental breakfast (if sponsored)
     * 8:30 AM Plenary speaker (TBD)
     * 9:15 AM Working group reports
          + ISO Standardization and Conformance
          + Multi-user / Avatars
          + Databases
          + TBD
     * 10:00 AM Break
     * 10:00 AM Exhibits open
     * 10:30 AM Papers: Case Studies, Tamara Munzner moderator
          + The Out of Box Experience: Lessons learned creating
            compelling VRML 2.0 content, Sam Chen, Rob Myers and Rick
            Pasetto, Silicon Graphics Inc.
          + Networked VR System: Kitchen Layout Design for Customers,
            Tomohiro Fukuda, Ryuichiro Nagahama and Junji Nomura,
            Matsusita Electric Works Ltd.
          + Animation for Performance Debugging of Parallel Computing
            Systems, Noritaka Osawa, Hisaya Morita and Toshitsugu Yuba,
            The University of Electro-Communications, Japan
          + Using VRML to Access Manufacturing Data, Sandy Ressler,
            Qiming Wang, Scott Bodarky, Charles Sheppard and Gregory
            Seidman, National Institute of Standards and Technology
     * 12:30 PM Lunch
     * 2:00 PM Papers: Implementation II, Dick Puk moderator
          + V-COLLIDE: Accelerated Collision Detection for VRML, Tom
            Hudson, Ming C. Lin, Jon Cohen, Stefan Gottschalk and Dinesh
            Manocha, University of North Carolina
          + Lodestar: An Octree-Based Level of Detail Generator for VRML,
            Dieter Schmalstieg, Vienna University of Technology, Austria
          + Wired for Speed: Efficient Routes for VRML 2.0, Daniel Woods,
            Alan Norton and Gavin Bell, Silicon Graphics Inc. and Wazabi
     * 3:00 PM Exhibits close, tear down
     * 3:30 PM Break
     * 4:00 PM Panel: Dynamic and Personalized Worlds: Database
       Integration, Daniel Lipkin - Oracle, Clay Graham - Big Book,
       Adrian Scott - VRMLSite, et al.
     * 5:00 PM Panel: User Interface Design for Digital Space: 3d
       Navigation and Manipulation, Moderator: Bernie Roehl - University
       of Waterloo, Maclen Marvit - Worlds Inc., James Waldrop -
       Construct, Mark Pesce, et al.
     * 6:00 PM Symposium complete
     * 7:00 PM Dinner on own
       
  THURSDAY, FEBRUARY 27, 1997
     * Working group meetings by request (Hyatt or NPS)
       
   
   
   Additional questions? Please contact:
   
   Lynn Johnson
   Evans & Johnson
   P.O. Box 51621
   Pacific Grove, California 93950 USA
   408.655.9924 voice, 408.372.0846 fax
   ljohnson@redshift.com
   

all the best, Don
-- 
Don Brutzman  Naval Postgraduate School, Code UW/Br Root 200  work 408.656.2149
              Monterey California 93943-5000 USA              fax  408.656.3679
Virtual worlds/underwater robots/Internet http://www.stl.nps.navy.mil/~brutzman

From rem-conf-request@es.net Thu Dec 05 17:58:19 1996 
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To: Steve Hopper <hopper@cs.ucsd.edu>
cc: Mark Handley <mjh@isi.edu>, David Simpson <davesimp@cs.berkeley.edu>, 
    298-list@bugs-bunny.CS.Berkeley.EDU
Subject: Re: UCB Multimedia Seminar update
In-reply-to: Your message of "Thu, 05 Dec 1996 11:09:24 PST." <199612051909.LAA15016@empyrean.ucsd.edu>
Date: Fri, 06 Dec 1996 08:39:07 +1000
Message-ID: <22914.849825547@connect.com.au>
From: George Michaelson <ggm@connect.com.au>



If observable fact is that 512k is saturated, and sdr is using low-priority
grade, this introduces a 2x2 matrix of choices.

Why should the mbone remain capped at an informal 512k? I think there are
good reasons to do this: Its still an appropriate level of b/w for the
IETF class of event, and there are still a lot of constrained b/w links in
the backbones worldwide.

Others might disagree.

Why should sdr remain in the low-priority grade? I think this is wrong, and
it should be treated as a higher priority packetflow. I've also stated many
times that I regard the re-announcement/backoff curve as being inappropriate
and I've been told the model encompasses a tiered tree of broadcasts where
local horizons might be more frequent, but I think there are still problems
in terms of pre-announcing sessions, and the need to increase the announce
rate markedly in a window surrounding the start-time of the session when
human factors indicate the vast mass of casual users actually start sdr, and
need to see sessions announced.

I don't see problems with both changing at once, but It seems to me changing
SDR now is simpler than changing the 512k rate limit default, and probably
more sensible in this specific context. Saturation will exist regardless of
bandwidth somewhere, and I think session announcements should have more
priority.

cheers
	-George

From rem-conf-request@es.net Thu Dec 05 21:07:00 1996 
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Date: Thu, 5 Dec 1996 18:06:27 -0800 (PST)
From: Stephen Casner <casner@precept.com>
To: rem-conf@es.net
Subject: AVT Agenda
Message-Id: <Pine.SOL.3.95.961205180439.462A-100000@little-bear.precept.com>
Mime-Version: 1.0
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AVT Working Group:

Enclosed below is my proposed agenda for the two AVT sessions at IETF
next week in San Jose.  I think I have included all the requested
topics, except for those related to reliable multicast since the Area
Directors will be separately addressing that problem in the Open
Transport Services Area Directorate meeting (Wednesday at 13:00).

Comments on the agenda continue to be welcome.  We can also make
adjustments at the start of the session as appropriate.

                                                -- Steve Casner



                       Audio/Video Transport WG
                                   
			     A G E N D A

Tuesday, December 10, 15:30-17:30

  - Introduction and status				 5

  - RTCP scaling to large groups			60

      - Summary of issues [Aboba]
      - Simulation results, proposals [Schulzrinne]
      - Discussion
	- Changes needed to protocol specs
	- Experimentation needed

  - Request for liaison with T1A1.5 [Klenke]		 5

  - IP/UDP/RTP header compression			30

      - Protocol changes since Montreal [Casner]
      - Implementation results [Leelanivas]

  - Transport aspects of RTSP Proposal			20


Wednesday, December 11, 19:30-22:00

  - RTP payload formats

      - Redundant encodings to Prop. Std. [Perkins]	 5
      - Revisions to payload format for G.723 [Zhu]	10
      - Problems with MPEG2 payload format [Civanlar]	20
      - Adding G.728 to Profile [Casner]		 5

  - New applications of RTP

      - RTP and reliable multicast [Casner]		10
        (reference to OTSV session)
	- Payload format for HTTP
	- Adding SRM to RTP
      - Using RTP with caching [Kermode]		20
      - Aggregation of media streams [Rosenberg]	20

  - Status report on RTP MIB [Naudus]			 5

  - Advancing RTP to Draft Standard [Casner]		15

      - Any changes for RTCP scaling?
      - Changes proposed for layered encodings
      - Edits to loop/collision algorithm
      - Proof of interoperability requirements

  - Advancing RTP Profile to Draft Standard [Casner]	15

      - Should A/V profile specify CNAME format?

  - Future work						10


From rem-conf-request@es.net Fri Dec 06 07:46:01 1996 
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X-Mailer: exmh version 1.6.7 5/3/96
To: rem-conf@es.net
cc: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
Subject: Conference bus
X-url: http://www.inria.fr/rodeo/avega.html
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Date: Fri, 06 Dec 1996 13:44:49 +0100
From: Andres Vega Garcia <Andres.Vega_Garcia@sophia.inria.fr>


	Hello, I want to find out if there have being any further work about 
defining a control bus to allow applications in a given host to 
exchange control information?

	The only paper I have is that describing "CCCP: Conference Control 
Channel Protocol" from Handley and Wakeman, but the trail in CCCP 
seems to be stoped there.

	Is somebody else working on this?

	Is there any technical report/article available from the people at 
LBL about the multicast bus implemented in vat and vic?

: Colin Perkins <C.Perkins@cs.ucl.ac.uk> wrote:

>Not sure about the LBL folks, but here at UCL we're working on 
adding 
>conference bus support to RAT, with the aim of allowing easy 
integration 
>of the tool into a unified user interface (son of the ReLaTe project 
[1]),
>and to allow remote control of the tool. It is our desire to make 
this
>compatible with future versions of vic/vat/etc...

	Colin, do you have a specification of your "conference bus"?

	I want to add that kind of capability to Free Phone, and stay 
compatible with others applications.

	Thank you.

-Andres


From rem-conf-request@es.net Fri Dec 06 07:54:34 1996 
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To: Andres Vega Garcia <Andres.Vega_Garcia@sophia.inria.fr>
cc: rem-conf@es.net, mjh@isi.edu, mccanne@eecs.berkeley.edu
Subject: Re: Conference bus
In-reply-to: Your message of "Fri, 06 Dec 1996 13:44:49 +0100." <199612061244.NAA08965@fun.inria.fr>
Date: Fri, 06 Dec 1996 12:51:59 +0000
Message-ID: <1117.849876719@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>

--> Andres Vega Garcia writes:
>	Hello, I want to find out if there have being any further work about 
>defining a control bus to allow applications in a given host to 
>exchange control information?

There has been some discussion between UCL, Mark Handley and Steve
McCanne's group about a successor to CCCP and the LBL conference bus. 
At present this is in a very preliminary state, and not yet ready for
presentation to a wider audience, however I expect further discussion 
will take place next week, and hopefully we can come up with something
more concrete.

Colin

From rem-conf-request@es.net Fri Dec 06 13:38:49 1996 
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To: Andres Vega Garcia <Andres.Vega_Garcia@sophia.inria.fr>
cc: rem-conf@es.net, Colin Perkins <C.Perkins@cs.ucl.ac.uk>
Subject: Re: Conference bus
In-reply-to: Your message of "Fri, 06 Dec 96 04:44:49 PST." <199612061244.NAA08965@fun.inria.fr>
Date: Fri, 6 Dec 1996 10:37:18 PST
Sender: Bill Fenner <fenner@parc.xerox.com>
From: Bill Fenner <fenner@parc.xerox.com>
Message-Id: <96Dec6.103722pst.177711@crevenia.parc.xerox.com>

I was curious to see what messages got passed on the conference bus, so
I wrote the following application.  It might help you some (and, of course,
reading the vic/vat source code doesn't hurt either).

"confbus" by itself will print the messages on the common bus,
"confbus N" will print the messages on IPC channel N (e.g. vic -I N)

  Bill

/*
 * confbus.c
 *
 * usage:
 * confbus
 *   prints out messages on common IPC channel
 * confbus N
 *   prints out messages on IPC channel #N (e.g. vic -I N)
 */

#include <stdio.h>
#include <ctype.h>
#include <sys/types.h>
#include <sys/socket.h>
#include <netinet/in.h>

#define GROUP_IPC_ADDR 0xe0ffdeef
#define GROUP_IPC_PORT 0xdeaf
#define GIPC_MAGIC 0x0ef5ee0a

struct gipc {
	int	magic;
	short	type;
	short	pid;
	char	buf[1];
};

int
main(argc, argv)
	int argc;
	char **argv;
{
	int s;
	int on = 1;
	struct ip_mreq mr;
	struct sockaddr_in sin;
	char buf[1024];
	struct gipc *g = (struct gipc *)buf;
	int i;

	if ((s = socket(AF_INET, SOCK_DGRAM, 0)) < 0) {
		perror("socket");
		exit(1);
	}
	if (setsockopt(s, SOL_SOCKET, SO_REUSEADDR, &on, sizeof(on)) < 0) {
		perror("SO_REUSEADDR");
		exit(1);
	}
	
	mr.imr_multiaddr.s_addr = htonl(GROUP_IPC_ADDR);
	mr.imr_interface.s_addr = INADDR_ANY;
	if (setsockopt(s, IPPROTO_IP, IP_ADD_MEMBERSHIP, &mr, sizeof(mr)) < 0) {
		perror("IP_ADD_MEMBERSHIP");
		exit(1);
	}

	sin.sin_family = AF_INET;
	sin.sin_addr.s_addr = htonl(GROUP_IPC_ADDR);
	sin.sin_port = GROUP_IPC_PORT;
	if (argc == 2) {
		sin.sin_port += atoi(argv[1]);
	}
	sin.sin_port = htons(sin.sin_port);

	if (bind(s, &sin, sizeof(sin)) < 0) {
		perror("bind to confbus");
		exit(1);
	}
	while(1) {
		i = recv(s, buf, sizeof(buf) - 1, 0);
		if (i < 0) {
			perror("recv");
			exit(1);
		}
		if (ntohl(g->magic) != GIPC_MAGIC) {
			printf("non-GIPC msg\n");
			continue;
		}
/*
		for (j = 0, p = buf; j < i; j++) {
			printf("%02X ", (unsigned char)*p++);
		}
		printf("\n");
*/
		if (i == 12) {
			/* backwards compatibility with old vat */
			struct in_addr a;

			a.s_addr = *((int *)(g->buf));
			sprintf(g->buf, "focus %s", inet_ntoa(a));
		}
		printf("%d %d %s\n",ntohs(g->type), ntohs(g->pid), g->buf);
	}
}

From rem-conf-request@es.net Fri Dec 06 14:01:00 1996 
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Date: Fri, 6 Dec 96 14:00:17 EST
From: fruit salad <pineaple@eden.rutgers.edu>
To: rem-conf@es.net
Message-ID: <CMM-RU.1.5.849898817.pineaple@er5.rutgers.edu>

subscribe M-BONE Remote Conferencing Rosa Costanzo

From rem-conf-request@es.net Sat Dec 07 18:42:36 1996 
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Date: Sat, 7 Dec 1996 15:42:04 -0800 (PST)
From: Stephen Casner <casner@precept.com>
To: rem-conf@es.net
Subject: Updated AVT agenda
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AVT Working Group:

Here's an updated agenda.  Due to a brain short I wrote G.723 when I
meant H.263, and I've added an item under Draft Standard transition
about allowing separate addresses for RTP and RTCP prompted by a
question from Deborah Estrin.  I also added the associated reading
which I forgot the first time.
                                                -- Steve Casner



                       Audio/Video Transport WG
                                   
			     A G E N D A

Tuesday, December 10, 15:30-17:30

  - Introduction and status				 5

  - RTCP scaling to large groups			60
	draft-aboba-rtpscale-00.txt & rem-conf list

      - Summary of issues [Aboba]
      - Simulation results, proposals [Schulzrinne]
      - Discussion
	- Changes needed to protocol specs
	- Experimentation needed

  - Request for liaison with T1A1.5 [Klenke]		 5

  - IP/UDP/RTP header compression			30
	draft-ietf-avt-crtp-01.txt

      - Protocol changes since Montreal [Casner]
      - Implementation results [Leelanivas]

  - Transport aspects of RTSP Proposal			20
	draft-ietf-mmusic-rtsp-00.txt
	draft-ietf-mmusic-stream-00.txt,ps


Wednesday, December 11, 19:30-22:00

  - RTP payload formats

      - Redundant encodings to Prop. Std. [Perkins]	 5
	    draft-perkins-rtp-redundancy-01.txt
      - Revisions to payload format for H.263 [Zhu]	10
	    draft-ietf-avt-rtp-payload-02.txt
      - Problems with MPEG2 payload format [Civanlar]	20
	    RFC 2038, draft-civanlar-bmpeg-00.txt
      - Adding G.728 to Profile [Casner]		 5

  - New applications of RTP

      - RTP and reliable multicast [Casner]		10
        (reference to OTSV session)
	- Payload format for HTTP
	    draft-aboba-rtp-http-01.txt
	- Adding SRM to RTP
	    draft-parnes-rtp-ext-srm-01.txt 
      - Using RTP with caching [Kermode]		20
      - Aggregation of media streams [Rosenberg]	20
	    draft-rosenberg-itg-00.txt,ps

  - Status report on RTP MIB [Naudus]			 5

  - Advancing RTP to Draft Standard [Casner]		30
	RFC 1889

      - Any changes for RTCP scaling?
      - Changes proposed for layered encodings
	    draft-speer-avt-layered-video-01.txt
      - Allowing separate addresses for RTP/RTCP
      - Edits to loop/collision algorithm
      - Proof of interoperability requirements

  - Advancing RTP Profile to Draft Standard [Casner]	15
	RFC 1890

      - Should A/V profile specify CNAME format?

  - Future work						10


From rem-conf-request@es.net Sun Dec 08 00:54:00 1996 
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Date: Sun, 8 Dec 1996 00:53:20 -0500
From: MsMegan@aol.com
Message-ID: <961208005319_707848430@emout15.mail.aol.com>
To: freixg@wwgv.com, Videophone@es.net, rem-conf@es.net
Subject: Re: Videoconferencing/Video Telephones

Greg,

There are two books on videoconferencing that I would highly recommend to
anyone that is doing research on videoconferencing:

"Videoconferencing the Whole Picture" by Toby Trowt-Bayard 
" Personal Videoconferencing" by Evan Rosen

Both books can be purchased through Global Videoconferencing Solutions at a
discount. They can be reached at 1-800-909-4874 or via the internet at
GVSI@aol.com. I believe the have a web site somewhere, but I don't have the
URL for it.

Megan



In a message dated 96-12-06 09:01:31 EST, freixg@wwgv.com (Gregory D. Freix)
writes:

<< Subj:	Videoconferencing/Video Telephones
 Date:	96-12-06 09:01:31 EST
 From:	freixg@wwgv.com (Gregory D. Freix)
 To:	Videophone@es.net
 
 I am doing some preliminary research on these subjects.  I would be most
 appreciative of any information you might have on them....or related topics,
 such a various compression technologies, etc.
 
 Thanks very much.
 
 Greg Freix >>


From rem-conf-request@es.net Sun Dec 08 02:34:35 1996 
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Date: Sun, 08 Dec 1996 15:26:21 +0800
From: WANG Jian <fyzhang@netra.nju.edu.cn>
Reply-To: fyzhang@netra.nju.edu.cn
Organization: NJU
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Cc: rem-conf@es.net, Videophone@es.net
Subject: Videoconferencing
Content-Type: text/plain; charset=us-ascii
Content-Transfer-Encoding: 7bit

Hi, everyone in this list:
I'm a new one, and doing some research work on multipoint 
videoconferencing. I hope someone help me some technical materials.
I want to know how or where(Internet Web) I can get the two book:

"Videoconferencing the Whole Picture" by Toby Trowt-Bayard
" Personal Videoconferencing" by Evan Rosen

Thanks a lot.
WANG Jian

From rem-conf-request@es.net Sun Dec 08 16:26:56 1996 
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Date: Sun, 8 Dec 1996 13:26:19 -0800 (PST)
From: Stephen Casner <casner@precept.com>
To: rem-conf@es.net
Subject: Questions/issues for AVT discussion
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I have a few additional questions/issues to consider for discussion
in the AVT working group meeting.  First, I have received a few
comments regarding the IP/UDP/RTP header compression draft:

  - I added several references to the IPv6 header compression draft,
    but I neglected to mention where the RANDOM fields from
    encapsulating headers would be inserted into the COMPRESSED_RTP
    and COMPRESSED_UDP headers.  It is proposed that these fields
    should follow immediately after the UDP checksum just as they
    follow the TCP checksum for TCP header compression with IPv6.

  - The encoding for number deltas does not include any provision for
    negative numbers, so out-of-order packets cause the RTP header to
    be sent uncompressed (IP and UDP can still be compressed).  There
    are some holes in the encoding space where it would be possible
    to insert encodings for negative numbers, but that would make the
    code path longer with a traditional implementation.  Should this
    be done?  Van Jacobson suggests a table-based implementation,
    which would allow the encoding to evolve in the future to full
    entropy encoding, even with dynamically tables downloaded from
    the compressor.

  - This compression scheme is targeted primarily at low-bandwidth
    links for which 256 contexts should be sufficient.  It has been
    suggested, however, that the context ID be changed to a variable
    length to allow a larger space for high-speed links.  Again, this
    is a complexity/flexibility tradeoff.  Comments?

Second, in the update to the agenda I added an item under Draft
Standard transition about the possibility of changing the RTP spec to
allow separate multicast addresses for RTP and RTCP and perhaps even
different multicast addresses for RTCP sender and receiver reports.
This question was posed by Deborah Estrin.  The motivation for using
different addresses is to allow source-specific multicast pruning to
control delivery of RTP data and the corresponding RTCP sender
reports, while allowing RTCP receiver reports to still get through.

The RTP spec attempts to be a general-enough framework to allow
variations for different applications.  This would be a further
generalization requiring only a small change to the document.  The
choice of using one address or multiple addresses would usually be
constrained by a Profile since a consistent choice is required for
interoperation.  So, the question is, are there negative
ramifications that outweigh the potential benefits?

Third, is question whether the A/V Profile should specify a
particular RTCP SDES CNAME format.  Again to be general over a
variety of applications, the RTP specification only states a set of
properties as requirements for the CNAME.  The spec suggests that
either the fully-qualified domain name or numeric address of the host
be used, with the latter preferred because in many environments it is
difficult or impossible to reliably get a FQDN all the time.

If the CNAME is to be used, for example, to pair up audio and video
sources to be synchronized, then the format must be the same for both
media.  If those media come from separate tools, the CNAMEs may not
match, as is the case when the RAT audio tool is paired with the vic
video tool.  Should the profile constrain the format?

							-- Steve


From rem-conf-request@es.net Mon Dec 09 03:29:25 1996 
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From: ShouJen <sjn@walsin.com.my>
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Date: Mon, 9 Dec 1996 15:48:12 +0000
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Hi there,

If anyone has anyone any information regarding those two books, I 
would appreciate if you could copy an email to me.  Thanks!

> Hi, everyone in this list:
> I'm a new one, and doing some research work on multipoint 
> videoconferencing. I hope someone help me some technical materials.
> I want to know how or where(Internet Web) I can get the two book:
> 
> "Videoconferencing the Whole Picture" by Toby Trowt-Bayard
> " Personal Videoconferencing" by Evan Rosen
> 
> Thanks a lot.
> WANG Jian
> 

--------------------------------------------------------------
Shou-Jen Ng
Walsin Technology Corporation (M) Sdn. Bhd.
Voice: 603-9857800
Fax  : 603-9857011                    email : sjn@walsin.com.my



From rem-conf-request@es.net Mon Dec 09 06:54:53 1996 
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Date: Mon, 9 Dec 1996 06:53:57 -0500 (EST)
From: Jeff Pulver <jeff@Pulver.COM>
To: rem-conf@es.net
Subject: 1996 Vote for Top VON Products of 1996 (www.von.com/vonvote.htm)
Message-ID: <Pine.SUN.3.91.961209065334.22394J-100000@enterprise.pulver.com>
MIME-Version: 1.0
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Hi There,

I've opened up voting for the top VON products of 1996 at 
http://www.von.com/vonvote.htm

The results will be posted the first week of January, 1997.

This year there are two different categories - Internet Telephony and
Streaming Audio/Video products.  

If there is a VON product you would like to vote on which isn't listed - 
please enter it in the "write-in" section and I will add that product to the
general product lists.


Best Regards,


Jeff Pulver                                               Tel. 516.487.1424
Publisher                                                 Fax. 516.487.7269
The Pulver Report                                     http://www.pulver.com



From rem-conf-request@es.net Mon Dec 09 19:20:19 1996 
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Date: Tue, 10 Dec 1996 08:19:38 +0800
From: cheng@moon.bjnet.edu.cn (Yan Cheng)
Message-Id: <9612100019.AA05752@moon>
To: rem-conf@es.net
Subject: Why did Voice over Internet boom just now?


Hi Friends,

During recent weeks I have read some books and other materials about tranmitting
voice over packetized networks. After reading, I feel puzzled. From this 
materials, we can know that the research and experiments related with voice
over packet networks or aimed to achieve real-time communication of voice over
packet networks has been done since the 70's, while more enough work has been
done. But why just now, the late of 90's, this field began to cause researchers
 to pay serious attention to and to boom? What was short of in the 70's? What
was the most crucial question?

Regards,
12/10/96


From rem-conf-request@es.net Mon Dec 09 20:19:13 1996 
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Date: Mon, 9 Dec 1996 17:18:30 -0800 (PST)
From: Stephane Chatre <chatre@ENGR.ORST.EDU>
To: Yan Cheng <cheng@moon.bjnet.edu.cn>
Cc: rem-conf@es.net
Subject: Re: Why did Voice over Internet boom just now?
In-Reply-To: <9612100019.AA05752@moon>
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Mime-Version: 1.0
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On Tue, 10 Dec 1996, Yan Cheng wrote:

> 
> During recent weeks I have read some books and other materials about tranmitting
> voice over packetized networks. After reading, I feel puzzled. From this 
> materials, we can know that the research and experiments related with voice
> over packet networks or aimed to achieve real-time communication of voice over
> packet networks has been done since the 70's, while more enough work has been
> done. But why just now, the late of 90's, this field began to cause researchers
>  to pay serious attention to and to boom? What was short of in the 70's? What
> was the most crucial question?

My guess is that there is much more money to get out of it, now that the
Internet has become so popular, than in the 70's.
Traditionally, audio transmission over long distances has been done by
phone companies so far.
Now, imagine doing long-distance or international calls virtuslly for free
with a better quality than phone... This kind of thing can run phone
companies out of business faster than anything.
We are not talking about a million dollar market. We're talking about a
multi-billion dollar market. This itself explains the recent boom you are
talking about.

 ========================
Stephane Chatre                   Email: chatre@cs.orst.edu          
Department of Computer Science    Phone: (541) 737-5583  (office)     
Oregon State University			 (541) 757-3376   (home)      
                            
Life is what happens to you while you're busy making other plans.
                                        - John Lennon            


From rem-conf-request@es.net Mon Dec 09 22:03:40 1996 
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To: Stephane Chatre <chatre@engr.orst.edu>
cc: Yan Cheng <cheng@moon.bjnet.edu.cn>, rem-conf@es.net
Subject: Re: Why did Voice over Internet boom just now?
In-reply-to: Your message of "Mon, 09 Dec 1996 17:18:30 PST." <Pine.HPP.3.95.961209171120.3650H-100000@ia.ENGR.ORST.EDU>
Mime-Version: 1.0
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Date: Tue, 10 Dec 1996 13:01:32 +1000
Message-ID: <26640.850186892@connect.com.au>
From: George Michaelson <ggm@connect.com.au>


Coding technology caught up with the clueless. this permits people outside
the labs to slaver over nice ideas and sell packaged drool to the masses.

	PS profits in telecoms are very very high. Declining, but still
	higher than the margins selling PC's or soap. Not for ISP's but
	for telephone companies you understand...

PTT's have come onboard. Its official: real honest-to-god ITU sanctioned
national carriers are going to do voice-over-ip as core business. Maybe only
5-15% this year, but its going to grow. No more "my s/w doesn't talk to your
s/w" and full bakalite handset compatible 2-way telephone calling.

	You can't argue with ubiquity. Its compellingly awful.

Enough non-IP systems are out there for owners to realize the low-quality
issue is a sales trick to con you into things you don't need. Desktop is
marginal cost for bums-on-seats. Soundcards are now viable and not noddy.

Really nice people are now out there too btw, I don't mean to imply precept
or any of the RTP/RTCP/H.xxx aware people are in the slime-sellers regions.

Convergeance is a human-factors idea as well as a statement of technology, and
for this application, the simplest reason is:

	Its time.

-George


From rem-conf-request@es.net Tue Dec 10 02:22:53 1996 
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          Mon, 9 Dec 1996 23:20:29 -0800
Date: Mon, 9 Dec 1996 23:20:29 -0800 (PST)
From: Stephen Casner <casner@precept.com>
To: George Michaelson <ggm@connect.com.au>
Cc: Stephane Chatre <chatre@engr.orst.edu>, Yan Cheng <cheng@moon.bjnet.edu.cn>, 
    rem-conf@es.net
Subject: Re: Why did Voice over Internet boom just now?
In-Reply-To: <26640.850186892@connect.com.au>
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Mime-Version: 1.0
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In the early 70's when we sent packet audio over the ARPAnet, we used
PDP11 computers with attached array processors to do the compression
and network protocols, and we could only fit one or maybe two voice
streams over ARPAnet's 50Kb/s links.  We just had to wait for
technology to catch up.

Of course, we have learned a few things about network protocols since
then, but there is a lot that is the same, too.
							-- Steve


From rem-conf-request@es.net Tue Dec 10 04:21:11 1996 
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          Tue, 10 Dec 1996 01:20:10 -0800
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          Tue, 10 Dec 1996 10:19:49 +0100 (MET)
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X-Mailer: exmh version 1.6.7 5/3/96
To: rem-conf@es.net
Subject: IETF retransmission -- very bad audio quality
Mime-Version: 1.0
Content-Type: text/plain; charset=us-ascii
Date: Tue, 10 Dec 1996 10:19:46 +0100
From: Thierry Turletti <Thierry.Turletti@sophia.inria.fr>


Hi,

The audio quality is very bad at INRIA. Is it possible to add some
redundancy to the audio flow ? Several tools already implement this
option (e.g. rat, fphone). It's a pity we cannot take advantage of these
improvements for such retransmissions.

Thierry 

PS: my mtrace for the IETF Channel 2 at 10:10am (UTC+1)

Results after 10 seconds:

  Source        Response Dest    Packet Statistics For     Only For Traffic
   * * *        138.96.24.37     All Multicast Traffic     From 205.180.223.4
     v       __/  rtt  843 ms    Lost/Sent = Pct  Rate       To 224.0.1.14
4.0.20.19      
4.0.0.34        sanjose1-mbone1.bbnplanet.net 
     v     ^      ttl    0      730/1349 = 54% 134 pps     0/262  =  0%  26 pps
192.203.230.241 mbone.nsi.nasa.gov 
     v     ^      ttl   64      274/1305 = 21% 130 pps     1/262  =  0%  26 pps
207.12.55.99    stk-mbone-1.sprintlink.net 
     v     ^      ttl   65       36/1027 =  4% 102 pps    10/261  =  4%  26 pps
206.61.106.99   fw-mbone-1.sprintlink.net 
     v     ^      ttl   66       12/947  =  1%  94 pps     2/251  =  1%  25 pps
206.229.87.99   dc-mbone-1.sprintlink.net 
     v     ^      ttl   67       21/829  =  3%  82 pps     9/249  =  4%  24 pps
207.41.200.99   pen-mbone-1.sprintlink.net 
     v     ^      ttl   68      250/1139 = 22% 113 pps    49/240  = 20%  24 pps
192.35.82.97    MBONE.CIT.CORNELL.EDU 
     v     ^      ttl   69      126/615  = 20%   0 pps    37/191  = 19%   0 pps
193.51.208.8   
138.96.152.17   sobone3.inria.fr 
     v      \__   ttl   70          680         68 pps       154         15 pps
138.96.24.37    138.96.24.37
  Receiver      Query Source



From rem-conf-request@es.net Tue Dec 10 10:16:21 1996 
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Date: Tue, 10 Dec 1996 10:15:22 -0500
From: rodgers@lhc2.nlm.nih.gov (R. P. C. Rodgers, M.D.)
Message-Id: <199612101515.KAA03181@billings.csb>
To: cheng@moon.bjnet.edu.cn, chatre@ENGR.ORST.EDU
Subject: Re: Why did Voice over Internet boom just now?
Cc: rem-conf@es.net
X-Sun-Charset: US-ASCII

Woah, wait just a minute here...

> From chatre@ENGR.ORST.EDU Mon Dec  9 22:47:19 1996
> Date: Mon, 9 Dec 1996 17:18:30 -0800 (PST)
> From: Stephane Chatre <chatre@ENGR.ORST.EDU>
> To: Yan Cheng <cheng@moon.bjnet.edu.cn>
> Cc: rem-conf@es.net
> Subject: Re: Why did Voice over Internet boom just now?
> Mime-Version: 1.0
> 
> On Tue, 10 Dec 1996, Yan Cheng wrote:
> 
> > 
> > During recent weeks I have read some books and other materials about tranmitting
> > voice over packetized networks. [...] But why just now, the late of 
> > 90's, this field began [...] to boom?> 
 What was short of in the 70's? What
> > was the most crucial question?

> Traditionally, audio transmission over long distances has been done by
> phone companies so far.

Most of the IP traffic we are talking about is *also* being carried by
"phone" companies, one way or another.

> Now, imagine doing long-distance or international calls virtuslly for free
> with a better quality than phone... This kind of thing can run phone
> companies out of business faster than anything.

If there is one thing that is clear from developments in the past two
years, it is that Internet services are not going to be free.  As for
putting "phone companies out of business," it's not that simple.  There
is a reason why some of the big international carriers did not join in
the recently squashed consortium trying to compell the U.S. government
to prohibit voice communications over the Internet -- as with every
technological paradigm shift, there are winners and losers.  Some of
the telephone carriers concerned with long distance network traffic
have every reason to expect that they will *benefit*, other firms
will lose out.  It certainly seems likely, though, that in the coming
few years we will see dramatic drops in the cost to the consumer of
long-distance voice service, and continued integration of voice with
other information services, in interesting and useful ways.  There will
be the usual tension between cost and quality, so it may not always
be the case that the resulting services will be "better than phone."
We should not extrapolate from today's experience with a lightly loaded
experimental ATM line and assume that everyone will have such quality
of service as firms scale up such services to the mass market.  There
will also be a fair amount of painful thrashing as the various
competitors experiment with different pricing and business models
until someone hits the right combination...

Cheerio, Rick Rodgers

>  ========================
> Stephane Chatre                   Email: chatre@cs.orst.edu          
> Department of Computer Science    Phone: (541) 737-5583  (office)     
> Oregon State University			 (541) 757-3376   (home)--------------------------------------------------------------------------------
   R. P. C. Rodgers, M.D. * rodgers@nlm.nih.gov * (301)496-9305 (voice, fax)
   CSB, LHNCBC, National Library of Medicine, NIH
   8600 Rockville Pike, Bethesda MD 20894 USA
   http://www.nlm.nih.gov/about_nlm/organization/lister_hill/
      computer_science/roster/rodgers.html

From rem-conf-request@es.net Tue Dec 10 10:18:50 1996 
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Date: Tue, 10 Dec 1996 19:45:39 +0100
To: Esa Vitikainen <Esa.Vitikainen@bitfield.fi>
From: Gunnar Hellstrom <gunnar.hellstrom@omnitor.se>
Subject: Re: Seeking visions/experiences of multipoint data apps
Cc: t120-interest@world.std.com, videophone@es.net, vidconf@pulver.com, 
    rem-conf@es.net

Data conferencing application.

One apparent need is for a written interpretation of the voice part of a
videoconference. To be used by hearing impaired people and language
minorities (who could also be interested in a second sound channel to get
voice interpretation instead.)=20

The requirement is character by character or ord by word display in a window
with automatic new line at end of line.=20
The display could be overlaid on video (like subtitling) or separate.

The requirements are very similar to chat requirements, but there might be a
need for chairman control.

Regards
Gunnar Hellstr=F6m

At 23:43 1996-12-03 -0800, Esa Vitikainen wrote:
>Hi all,
>
>as part of my studies I am planning to find out just what are=20
>users / developers / service providers / VARs / ...
>doing / going to do / would like to do=20
>with data in multipoint multimedia conferencing context.=20
>In other words, I'm chasing visions and experiences of=20
>applications adding value to voice/video.=20
>
>There are the apparent chat, file transfer, whiteboard,=20
>and application sharing. But how are they really utilized?=20
>Where do they fall short? What else is needed out there?=20
>
>Are there studies or something about this available somewhere?=20
>Places to ask around? Willing to reveal your own great idea?=20
>
>An example: features I imagine needed for remote presentation:
>- slideshow operated by the presenter
>- "hands raised" electronically (with text description?)
>- slides independently leafed through / edited locally
>  (like having paper copies)
>- others taking over the slideshow control, with own remarks
>- dialogs sent to ask for comments / test understanding,=20
>  with reply statistics shown (for large audience)
>
>Regards,
>Esa
>
>
_____________________________________
Gunnar Hellstr=F6m
Omnitor
Alsn=F6gatan 7, 4 tr
S-116 41 Stockholm
SWEDEN
Tel +46 701 100 501
Fax +46 8 448 27 51
e-mail gunnar.hellstrom@omnitor.se
www: http://public-www.pi.se/~omnitor
-------------------------------------


From rem-conf-request@es.net Tue Dec 10 10:40:54 1996 
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Date: Tue, 10 Dec 1996 07:35:30 -0800
To: Stephen Casner <casner@precept.com>
From: Norm Johnson <njohnson@qualcomm.com>
Subject: Re: Why did Voice over Internet boom just now?
Cc: George Michaelson <ggm@connect.com.au>, 
    Stephane Chatre <chatre@engr.orst.edu>, Yan Cheng <cheng@moon.bjnet.edu.cn>, 
    rem-conf@es.net

The number one reason now is the right time for voice over packet networks
is simply the big carriers of the 70's didn't have the bandwidth or the
computing power.  Long distance backbones were microwave with only analog
circuits.  Digital came along in the late eighties but still no bandwidth.
Most transcontental circuits were at best a dozen or so DS-3's and they were
maxerd out with analog voice calls.  Finally when fiber arrived in the early
nineties, the bandwidth got big enough and the number of computers in the
world increased demand for digital backbones and broad bandwidth data
networks.  It wasn't until 1992 that the Internet backbone finally went to a
full DS-3.  It takes demand from customers to make the big guys gear up and
get moving.  As was pointed out, the idea that now data networks are cheaper
for carrying voice than the old analog circuits has just arrived on the
scene in the middle nineties.  From here though, things are going to get
real crazy and every conceivable communications company is going to be
offering every conceivable service and the market will shake out the
successful.  Voice over data networks is only the beginning.  
It often takes decades to move an idea or invention into the relm of
market-place reality.  Fiber was known since the 60's but only deployed
large scale in the nineties.  Digital modulation was known in the 40's but
was only deployed in the 80's.  However, a fundamental growth pattern has
changed and now it may only be a matter of a few years to see inventions go
>from the design stage to the market.  Hence, dense WDM fiber was developed
in 1993 and is already appearing in the backbones of the major carriers.
The pace of deployment has increased so such techno-delays of the past are
probably no longer going to be seen.


At 11:20 PM 12/9/96 -0800, you wrote:
>In the early 70's when we sent packet audio over the ARPAnet, we used
>PDP11 computers with attached array processors to do the compression
>and network protocols, and we could only fit one or maybe two voice
>streams over ARPAnet's 50Kb/s links.  We just had to wait for
>technology to catch up.
>
>Of course, we have learned a few things about network protocols since
>then, but there is a lot that is the same, too.
>							-- Steve
>
>
>
-----------------------------------------------
Dr. Norman P. Johnson
2727 DeAnza Rd., Box N-14
San Diego, CA. 92109
Phone: Home (619) 581-0447
       Ofc  (619) 651-1491
Check out my web page:
http://www.qualcomm.com/~njohnson 
-----------------------------------------------
"When the going gets weird, the weird turn Pro"
                      -- Dr. Hunter S. Thompson


From rem-conf-request@es.net Tue Dec 10 18:35:27 1996 
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          Tue, 10 Dec 1996 11:34:04 -0800
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          by coral.llnl.gov (8.8.4/8.7.3/LLNL-Jun96) with ESMTP id LAA23747;
          Tue, 10 Dec 1996 11:33:50 -0800 (PST)
Date: Tue, 10 Dec 1996 11:33:50 -0800 (PST)
X-Sender: jed@coral.llnl.gov
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To: cheng@moon.bjnet.edu.cn (Yan Cheng)
From: "James E. [Jed] Donnelley" <jed@llnl.gov>
Subject: Re: Why did Voice over Internet boom just now?
Cc: rem-conf@es.net

>Hi Friends,
>
>During recent weeks I have read some books and other materials about
>tranmitting
>voice over packetized networks. After reading, I feel puzzled. From this
>materials, we can know that the research and experiments related with voice
>over packet networks or aimed to achieve real-time communication of voice over
>packet networks has been done since the 70's, while more enough work has been
>done. But why just now, the late of 90's, this field began to cause
>researchers
> to pay serious attention to and to boom? What was short of in the 70's? What
>was the most crucial question?
>
>Regards,
>12/10/96

The answer is the Web.  You might say that the growth in the Internet
in general eventuly made the Web inevitable, but it was the Web
that created the market.  Before that voice over packet networks
was a research exercise.  Now the potential is there to reach essentially
everybody.  That makes it serious financially.  Hence the "boom."

That's my opinion.


--Jed   http://www-atp.llnl.gov/atp/jed-signature.html



From rem-conf-request@es.net Tue Dec 10 19:01:45 1996 
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Date: Tue, 10 Dec 1996 16:02:57 -0800
To: "James E. [Jed] Donnelley" <jed@llnl.gov>, 
    cheng@moon.bjnet.edu.cn (Yan Cheng)
From: tim@boxtop.com (Tim Dorcey)
Subject: Re: Why did Voice over Internet boom just now?
Cc: rem-conf@es.net

At 11:33 AM 12/10/96, James E. [Jed] Donnelley wrote:
>The answer is the Web.  You might say that the growth in the Internet
>in general eventuly made the Web inevitable, but it was the Web
>that created the market.

Let's not forget a number of other developments that contributed to the
explosion of interest in the Internet:

1.  Availability of dial-up IP
2.  Commercialization of the ISP business
3.  Standardization of e-mail and widespread acceptance of its utility

Sure, the www was important, but I think it would be a mistake to
over-emphasize it.  5 years ago, Internet access was almost exclusively via
LANs at universities and research institutes; commercial activities were
expressly forbidden; and you could make a living figuring out how to
translate between Bitnet, Decnet, UUCP, JANet, QuickMail, etc.  A lot of
things happened at once.  Of course, The Media find the web most
interesting, because the web looks like The Media and The Media loves The
Media.





From rem-conf-request@es.net Tue Dec 10 19:04:49 1996 
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Date: Tue, 10 Dec 1996 19:03:40 +0500
From: devi@sol.genie.uottawa.ca (Sridevi Palacharla)
Message-Id: <9612110003.AA03213@sol>
To: rem-conf@es.net
Subject: RTP timestamps and inter-media synchronization
Content-Length: 663


In the RTP specification, "timestamp" in RTP header corresponds to
the media clock and depends on the payload type. 8 Khz for PCM audio,
90 khz for Jpeg video etc.

In case of pre-recorded video or audio, tranmitted using RTP at a 
later time or on-demand, how should the timestamp in RTP header be used?

Also RTCP SR has NTP timestamp to associate with "timestamp" in RTP header,
if this is to be used for inter-media synchronization? What should be the
frequency at which RTCP SR should be generated? 
If this is so crucial for inter-media sync. should this be sent on TCP
rather than over UDP?

Any clarifications are most helpful.

thanx in advance
sridevi

From rem-conf-request@es.net Wed Dec 11 13:12:48 1996 
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          Wed, 11 Dec 1996 10:03:41 -0800
To: Thierry Turletti <Thierry.Turletti@sophia.inria.fr>
cc: rem-conf@es.net
Subject: Re: IETF retransmission -- very bad audio quality
In-reply-to: Your message of "Tue, 10 Dec 96 01:19:46 PST." <199612100919.KAA11775@merlot.inria.fr>
Date: Wed, 11 Dec 1996 10:03:37 PST
Sender: Bill Fenner <fenner@parc.xerox.com>
From: Bill Fenner <fenner@parc.xerox.com>
Message-Id: <96Dec11.100341pst.177711@crevenia.parc.xerox.com>

In message <199612100919.KAA11775@merlot.inria.fr> you write:
>207.41.200.99   pen-mbone-1.sprintlink.net 
>     v     ^      ttl   68      250/1139 = 22% 113 pps    49/240  = 20%  24 pp
>192.35.82.97    MBONE.CIT.CORNELL.EDU 

It's probably appropriate to ask Sprint or Cornell to try to isolate and
fix this loss.

>192.35.82.97    MBONE.CIT.CORNELL.EDU 
>     v     ^      ttl   69      126/615  = 20%   0 pps    37/191  = 19%   0 pp
>193.51.208.8   
>138.96.152.17   sobone3.inria.fr 

This was one of the first pathological tunnels that we found when we
started doing the Multicast Planet visualization.  I thought that we
worked at the time to get it removed, but I can't find any copies of
such email so I must be wrong.  This appears to be a backup US<>Europe
link that made sense a long time ago, but probably doesn't now unless
Inria and Cornell have a private line that this tunnel travels over.

  Bill

From rem-conf-request@es.net Wed Dec 11 17:00:31 1996 
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Date: Wed, 11 Dec 1996 13:57:47 -0800
To: rem-conf@es.net
From: Brian Bulkowski <brianb@navio.com>
Subject: Group size estimation
Mime-Version: 1.0
Content-Type: text/plain; charset="us-ascii"

Hello,

It was proposed yesterday that group size could be tracked without keeping
each SSRC value. The necessity of keeping every SSRC, or a table large
enough to have indexed all SSRC values, is onerous in a million-node
receiver group. Just so we know the section we're talking about:

6.2.1 Maintaining the number of session members

   Calculation of the RTCP packet interval depends upon an estimate of
   the number of sites participating in the session. New sites are added
   to the count when they are heard, and an entry for each is created in
   a table indexed by the SSRC or CSRC identifier (see Section 8.2) to
   keep track of them. New entries may not be considered valid until
   multiple packets carrying the new SSRC have been received (see
   Appendix A.1). Entries may be deleted from the table when an RTCP BYE
   packet with the corresponding SSRC identifier is received.

   A participant may mark another site inactive, or delete it if not yet
   valid, if no RTP or RTCP packet has been received for a small number
   of RTCP report intervals (5 is suggested). This provides some
   robustness against packet loss. All sites must calculate roughly the
   same value for the RTCP report interval in order for this timeout to
   work properly.

   Once a site has been validated, then if it is later marked inactive
   the state for that site should still be retained and the site should
   continue to be counted in the total number of sites sharing RTCP
   bandwidth for a period long enough to span typical network
   partitions.  This is to avoid excessive traffic, when the partition
   heals, due to an RTCP report interval that is too small. A timeout of
   30 minutes is suggested. Note that this is still larger than 5 times
   the largest value to which the RTCP report interval is expected to
   usefully scale, about 2 to 5 minutes.

Notice that there are a large number of "mays" and few "musts". The client
is allowed to implement whatever it chooses, however, it is clear what the
writers of the RFC intended. One interesting point of the wording is that
it assumes an estimate of what the report interval is, but I see nowhere
else that suggests a good method for getting that estimate.

In most algorythms, you need an estimate of the report interval, including
the spec'd algorythm; here's one that comes to mind. Maintain an estimate
of the RR interval by tracking a small number of specific SSRC RR timings.
Make a table including the last time heard, number of times heard from, and
SSRC value. Whenever you receive an SSRC BYE packet, invalidate that timing
entry. When you have received a certain number of SSRC packets (5?), add
the interval to the running average, then clear the entry. On every RR
receipt, check for an empty entry, if you find one, use that SSRC value for
estimation. Have a timer run though the interval estimation table to throw
out entries that it hasn't heard from for a while. Suggested time is three
times the current report interval.

Is there another way of estimating the report interval besides watching
everyone else? If you have a size-of-group estimate, and a stream size
estimate, you can calculate the report interval. This can be used instead,
except in initial cases. There was a bit of discussion at the IETF about
initial wait intervals before sending RR packets.

I haven't figured out the best way to implement the intial case. In the
table-driven case it's much easier, since you can just wait until you've
got a couple of reports from a particular SSRC, then assume all the
estimates you've got are good.

In terms of getting group size, once you have a good estimate of the report
interval, you need only keep track of the number of RR packets received
(say, per second, making sure you use a different timer than all others).
Whenever you desire the size of the receiver group, multiply the current
report interval by the number of RRs received, and you get the size of the
receiver group.

This algorythm seems to converge slowly, but has a nice attribute that if
you are losing RR packets, you get a larger size estimation, which leads to
the sending of fewer RR packets, which means that the algorythm is
congestion friendly.

Another algorythm is as follows.

Keep a hashtable that has only two states (a bitfield would work):
received, and not received. Use as a hash algorythm masking out the high
bits of the SSRC. Whenever you receive an RR packet, hash the SSRC value,
and set the state of the table entry to received.

The table is valid only after one report interval. Therefore, the report
interval must still be calculated using some method (such as the one
above). After one report interval, sum the number of states set to
received, and set that as your estimate. Use that estimate until the next
report interval has expired. Note that variation in estimates of report
interval will cause either undersizing or oversizing. Unfortunatly, I
suspect the system doesn't converge. If you have too small of an interval,
then you will think the group is smaller than it is, and recalculate an
even smaller interval. If you think the interval is larger than it really
is you'll be fine, since the algorythm caps out.

Note that this algorythm still requires memory dependant on the number of
other receivers, but at least it is only one bit. (128K for a
million-receiver session). However, if there is a hashtable collision,
there will be underreporting, which leads to greater RR sending then there
should be, which is bad. The exact frequency of probable collision depends
on the exact random number generator used for the SSRC values, the size of
the session, the hashtable size chosen, and the hash algorythm. I suspect
that there might be a positive feedback cycle if everyone underestimates.
On the other hand, if everyone underestimates by a fixed amount, you might
just get a constant level of underreporting.

In order to dynamically resize the hash table, you could keep track of a
few marker buckets, and whenever there is a collision in one of your marker
buckets, realloc your table and start again. Or, you could make an estimate
at the amount of collision you are getting, and multiply your size estimate
by that factor. This allows people to pick certain table sizes and not
reallocate. In the "marker buckets" you need to keep track of the exact
SSRCs received, so you can detect true collisions vs. bogus collisions due
to inaccurate estimation of the report interval.

I'm sure that the RTP designers thought over these issues. It is clear that
the current set of "mays" in section 6.2.1 will not work with large
receiver groups. Would it be better to add suggested algorythms and words
for larger groups, or would it be better to spec one of the other
estimation algorythms (like the polling one) which doesn't require the
receiver's calculation of receiver groupsize? Or would it be better to turn
off RRs in large groups? What's the thinking?

Regarding what Van said about multicast being the fastest path to
convergance, the client side memory must be considered as well. If an
algorythm has a particular network charisteric, but takes memory linearly
depending on groupsize, it is not scaleable. Which might be OK; the points
made in the WG meeting about the initial design goals are well taken. I am
working on a list of my specific design requirements. RTP might not be the
protocol for me.

There must be a suggested implementation which is constant size in memory,
and also has good convergance characteristics, or a statement about the
limits of scalebility of the protocol. Granted, exact supportable groupsize
before scalebility runs out will depend on many things, including the
client resources, but a statement needs to be made.

In any case, I'd like to have wording suggesting methods for estimating the
report interval, since it is required for section 6.2.1.

Regards,
brianb

From rem-conf-request@es.net Thu Dec 12 08:07:22 1996 
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From: ag@eragny.rce.fr
To: rem-conf@es.net
Date: Thu, 12 Dec 1996 14:06 GMT
Subject: packetized voice characteristics
Content-Length: 951
Content-Type: text/plain
Message-ID: <32b011700.46d5@eragny.rce.fr>

Hi,

Given the traffic characterisation for real-time data, from TENET at Berkeley
University (Zhang, Ferrari), in which a source specifies its characteristics by:
	(Xmin, Xave, I, P)
where:
Xmin=minimum packet inter-arrival time over time period I
Xave=average packet inter-arrival time over time period I
I=time period
P=max packet size

and ask to the network some performance requirments (e.g. Delay, Delay-Jitter, ...),
does anybody knows some typical values of Xmin, Xave, I, P at the ouput of a
packetization machine ?
In the litterature, we often read:
. an uncompressed voice source needs 64kbps of bandwidth and a max delay of 200ms
. a compress voice source needs16kbps or 8kbps (depending on algorithm used) and a max
delay of 30ms
I don't guess it is enough to conclude on the Xmin, Xave, I, P characteritics of the source.
Can anybody help ?

Thanks.

Anouchiravan Gharagozlou
Software Development Engineer
CS Telecom/RCE
anouchg@rce.fr

From rem-conf-request@es.net Thu Dec 12 10:29:39 1996 
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          with ESnet SMTP (PP); Thu, 12 Dec 1996 07:29:00 -0800
From: ag@eragny.rce.fr
To: mbc@attitw.attmail.com, rem-conf@es.net, support@bayarea.net, 
    daemon@usenet.ucs.indiana.edu, dec@dfv.rwth-aachen.de, 
    comp.dcom.cell-relay@indiana.edu
Date: Thu, 12 Dec 1996 16:28 GMT
Subject: voice packet characterizations
Content-Length: 1116
Content-Type: text/plain
Message-ID: <32b032ab0.48c2@eragny.rce.fr>

Hi,

Given the traffic characterization modele for real-time data, from TENET at
Berkeley University (Zhang, Ferrari), in which a source specifies its characteristic by:
	(Xmin, Xave, I, P)
where:
Xmin=minimum packet inter-arrival time over time period I
Xave=average packet inter-arrival time over time period I
I=time period
P=max packet size

and asks to the network some performance requirment (e.g. Delay,
Delay-Jitter, ...), what are the typical values of these parameters
for compressed and uncompressed voice packets ?
 If it depends on the packetization strategie, please provide some numerical examples.

We often read in the litterature that an uncompressed voice source needs
64kbps of bandwidth and a maximum delay of 200ms while a compressed voice
source needs 16kbps or 8kbps (depending on the algorithm used) of bandwidth
and a maximum delay of 30ms.

This doesn't tell how often voice packets are generated by the packetization
machine and what are the length of the packets... (i.e. Xmin, Xave, I, P).

Thanks,


Anouchiravan Gharagozlou
Software Development Engineer
CS Telecom/RCE
anouchg@rce.fr

From rem-conf-request@es.net Fri Dec 13 11:31:18 1996 
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To: Stephen Casner <casner@precept.com>
cc: rem-conf@es.net
Subject: Re: IP/UDP/RTP header compression
In-reply-to: Your message of "Sun, 24 Nov 1996 15:15:44 PST." <Pine.PCW.3.95.961124150032.7183F-100000@port4.precept.com>
Date: Fri, 13 Dec 1996 16:26:18 +0000
Message-ID: <2370.850494378@cs.ucl.ac.uk>
From: Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>



the realy neat thing about cRTP is that you could envisage using 
RTP+UDP+IP+AAL5+ATM, but the full header would really kill things....

but with cRTP, you can do voice on ATM, with transparent interworking
to the Mbone/Internet/H323....as long as people use the right voice
sample size, with a single cell (and no echo cacnellors unless the IP
on NOT ATM path incurs too much delay...)

of course, as usual, this is assuming ATM as a low bandwidth
technology.....which it appears usually to be:-)
thus we win again....

not entirely joking!

cheers
jon

From rem-conf-request@es.net Fri Dec 13 21:02:48 1996 
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                          protocol
From: Mark Handley <mjh@isi.edu>
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X-Phone: +1 617 253 6011
To: Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>
cc: Stephen Casner <casner@precept.com>, rem-conf@es.net
Subject: Re: IP/UDP/RTP header compression
In-reply-to: Your message of "Fri, 13 Dec 96 16:26:18 GMT." <2370.850494378@cs.ucl.ac.uk>
Date: Fri, 13 Dec 96 21:01:58 -0500
Message-ID: <25389.850528918@mercury.lcs.mit.edu>
Sender: mjh@mercury.lcs.mit.edu
X-Mts: smtp


>
>
>the realy neat thing about cRTP is that you could envisage using 
>RTP+UDP+IP+AAL5+ATM, but the full header would really kill things....
>
>but with cRTP, you can do voice on ATM, with transparent interworking
>to the Mbone/Internet/H323....as long as people use the right voice
>sample size, with a single cell (and no echo cacnellors unless the IP
>on NOT ATM path incurs too much delay...)

Of course then we can apply the header compression techniques to
combined ATM/AAL5/IP/UDP/RTP header compression.  But the first two
parts compress to nothing, so we already have a draft that covers this
case...

Mark

From rem-conf-request@es.net Fri Dec 13 21:49:48 1996 
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Date: Fri, 13 Dec 1996 18:48:56 -0800 (PST)
From: Stephen Casner <casner@precept.com>
To: rem-conf@es.net
Subject: IETF AVT meeting summary
Message-ID: <Pine.SOL.3.95.961213184721.28308B-100000@little-bear.precept.com>
MIME-Version: 1.0
Content-Type: TEXT/PLAIN; charset=US-ASCII

This is the "one paragraph" summary that the WG chair is required to
send to the Area Director by the end of IETF week.  Full minutes of
the meeting will be forthcoming soon.
							-- Steve


The AVT working group met for two sessions at the San Jose '96 IETF.
The most significant topic was RTCP scaling for large groups and
scenarios beyond those currently described in the RTP spec.  Bernard
Aboba and Henning Schulzrinne gave presentations on the problems that
arise and a variety of possible solutions.  In the discussion that
followed, there was not one solution that was the clear answer in all
cases.  However, it was agreed that the wording in the main RTP spec
should be modified to allow more flexibility in how RTCP may be used,
and that the means by which different modes should be defined and
selected is through the specification of (a small number of)
additional RTP Profiles.  Proposals for additional profiles are
solicited.

Steve Casner presented an update on the proposal developed with Van
Jacobson for hop-by-hop compression of IP/UDP/RTP headers for
applications running over low-speed lines that was introduced at the
previous meeting in Montreal.  The significant changes were to better
integrate this compression scheme into the IPv6 header compression
scheme (which also supports IPv4 and encapsulation in multiple
headers) and to replace the temporary use of the delta encoding
scheme from RFC 1144 TCP/IP header compression with one designed to
match the deltas encoutered in typical RTP applications.  Manoj
Leelanivas presented a report on his implementation of the
compression algorithm and its performance.  The working group agreed
that this proposal should go to Last Call.  In a related
presentation, the authors of the RTSP proposal in the MMUSIC working
group reported that the means of specifying the underlying stream
transport protocol would be made more flexible, and that the
reduced-size variant of RTP included as one transport option in that
proposal would be extracted as a non-standards-track interim proposal
and would be named CUSH rather than "compressed RTP".
 
The second session began with updates on the RTP Payload Formats for
Redundant Audio and H.263 video.  Both of these should be ready for
Last Call after a few minor edits.  Reha Civanlar gave a presentation
of a problem with the packet loss resilience mechanism for the
Proposed Standard MPEG2 payload format.  He proposed a modification
to the header to solve that problem, and was asked by the chair to
write up that proposed change to be made to the document before its
transition to Draft Standard.

Several proposals for new applications of RTP have been recently
submitted to the working group.  Some fall into the general area of
reliable multicast which the Transport Area Directors want to
organize as a separate IRTF Research Group and later IETF Working
Groups.  The group discussed this plan and the fit of new work into
the AVT charter.

Two presentations were made on new applications.  The first by Roger
Kermode on the idea of layering audio and video streams in time as
well as quality, then combining this layering with caching to reduce
latency and bandwidth requirements for on-demand playback.  The
second was by Jonathan Rosenberg on multiplexing several RTP audio
sessions into one packet stream.  This could be used, for example, to
increase packet efficiency substantially between Internet Telephony
gateways.  Stan Naudus gave a status report on the development of an
RTP MIB.

The group discussed changes required to advance the RTP and Profile
specs to Draft Standard.  In addition to the RTCP scaling mentioned
earlier, the other significant point of discussion was on
generalizing the specification of an RTP session to allow different
addresses to be used for RTP and RTCP.  This would allow RTCP
delivery to be constrained for improved scalability in scenarios.
The consensus of the group was that we should take the conservative
approach and not make this change at this time.  The group also
agreed that the RFC 1890 A/V Profile should recommend the use of only
the numeric address form of the RTCP SDES CNAME item in common across
all tools so that audio and video streams can be matched for
synchronization.  The Profile will also have added some
specifications for simple payload formats such as G.728.

The AVT meeting ended with a discussion of how much work remains,
since the original charter has been completed.  The group agreed to
meet again in Memphis, with the intention that work on new Profiles
for RTCP scaling and the new Payload Formats currently in progress
can be finished then.


From rem-conf-request@es.net Sun Dec 15 23:41:08 1996 
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From: Scott Petrack <Scott_Petrack@vocaltec.com>
To: J.Crowcroft@cs.ucl.ac.uk
cc: casner@precept.com, rem-conf@es.net
Message-ID: <C2256402.0017DFD5.00@USNT2.vocaltec.com>
Date: Mon, 16 Dec 1996 06:37:16 +0300
Subject: Re: IP/UDP/RTP header compression
Mime-Version: 1.0
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Scott Petrack
12/16/96 06:37 AM

Sorry to break the hilarity, but I can't help but ask a more serious question
about
C/RTP:

It does seem that there is some interest in compressing RTP in a high bandwidth
point to point link.
Steve pointed out that if the pipe is long or fat there is a problem, because
there may not be enough
bits for sequence numbers or contexts. We mentioned the possibility of using
more bits for more
contexts, but not for more sequence numbers.

It would be nice if somehow a future hook could be made for allowing an extra
byte for long
sequence numbers and/or an extra byte for lots of contexts, perhaps as a message
to be sent
when compression is negotiated.

On a related note, it was suggested that it might be more efficient if there
could be a compression
table per context, rather than a single table for all contexts. Surely on a high
bandwidth line, at least
a few different tables would be nice to have, even if not one per context.

Unless I'm mistaken, it is a pure complexity/bandwidth tradeoff (maybe with
memory if there are many
tables). In the near future CPU may be VERY cheap compared to bandwidth, and one
might like
to use RTP compression on a higher speed link. It would be sad it this will be
impossible because there are only 4 bits for sequence
numbers and no way to negotiate more.

This would not be hard to do in a way which preserves backward compatibility
with the boxes that
Cisco is going to ship next week ;-)

Scott

-------- On 12/13/96 04:26 PM GMT  J.Crowcroft @ cs.ucl.ac.uk  wrote:





the realy neat thing about cRTP is that you could envisage using
RTP+UDP+IP+AAL5+ATM, but the full header would really kill things....

but with cRTP, you can do voice on ATM, with transparent interworking
to the Mbone/Internet/H323....as long as people use the right voice
sample size, with a single cell (and no echo cacnellors unless the IP
on NOT ATM path incurs too much delay...)

of course, as usual, this is assuming ATM as a low bandwidth
technology.....which it appears usually to be:-)
thus we win again....

not entirely joking!

cheers
jon



From rem-conf-request@es.net Mon Dec 16 13:34:28 1996 
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          id TAA13375; Mon, 16 Dec 1996 19:33:34 +0100
Message-Id: <199612161833.TAA13375@cismsun.univ-lyon1.fr>
Subject: some strange nt error messages
To: mjh@ISI.EDU (Mark Handley)
Date: Mon, 16 Dec 1996 19:33:34 +0100 (MET)
From: Lucia Gradinariu <lucia@univ-lyon1.fr>
Cc: rem-conf@es.net
In-Reply-To: <2976.849552357@mercury.lcs.mit.edu> from "Mark Handley" at Dec 2, 96 01:45:57 pm
X-Mailer: ELM [version 2.4 PL23]
Content-Type: text
Content-Length: 987

hi,

while using nt between 4 stations as a shared program editor we've got these 
error messages (as associated behaviour that is all nt applications blocked)
--------------------------------------------------------------
Got a bug in my recent list code
Paranoia check in ds_fns.c - got a duplicated block
 t86d665f433d70006 != t86d665f433d70006
ttl=3 multicast address: 224.5.6.7  port: 8500
--------------------------------------------------------------
on 2SGI+1SS20 we have v1.5a23 and on a PC we have v1.5a20

any idea how can we have two different blocks with the same
identifier? 

thanks,

--------------------------------------------------------------------------------
Lucia GRADINARIU, Eng.
CISM-Univ. Cl. Bernard Lyon1			voice:(33).72.43.13.69
Bat. 101 Bd. du 11 Novembre 1918		fax:(33).72.44.84.10
69622 Villeurbanne FRANCE			email:lucia@univ-lyon1.fr
		http://www.univ-lyon1.fr/CISM/lucia
---------------------------------------------------------------------------------

From rem-conf-request@es.net Mon Dec 16 23:21:08 1996 
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          Mon, 16 Dec 1996 20:11:50 -0800
Date: Mon, 16 Dec 1996 20:12:24 -0800 ()
From: Stephen Casner <casner@precept.com>
To: Scott Petrack <Scott_Petrack@vocaltec.com>
cc: J.Crowcroft@cs.ucl.ac.uk, rem-conf@es.net
Subject: Re: IP/UDP/RTP header compression
In-Reply-To: <C2256402.0017DFD5.00@USNT2.vocaltec.com>
Message-ID: <Pine.WNT.3.95.961216195153.-141721K-100000@oak.precept.com>
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> It does seem that there is some interest in compressing RTP in a high
> bandwidth point to point link.  Steve pointed out that if the pipe is
> long or fat there is a problem, because there may not be enough bits
> for sequence numbers or contexts. We mentioned the possibility of
> using more bits for more contexts, but not for more sequence numbers.

I think it would be straightforward to provide the option for
two-byte context identifiers using the techniques already defined in
the ipv6-hc draft.

However, it seems to me that the size of the sequence number is
unrelated to the length or obesity of the pipe.  The sequence number
just needs to allow detection of loss or misordering.  If the pipe
misorders, then the delta coding is not going to work.  If the
probability of losing sequence-space-sized groups of packets is high,
then the pipe is not a good candidate for compression either.

> On a related note, it was suggested that it might be more efficient
> if there could be a compression table per context, rather than a
> single table for all contexts. Surely on a high bandwidth line, at
> least a few different tables would be nice to have, even if not one
> per context.

Yes, the compression table could be per context.  I guess that just
requires that the yet-to-be-defined message for negotiating
compression tables needs to be able to indicate to which context(s)
the table applies, probably with a way to say "all".  I'm thinking
that this negotiation should be left to PPP rather than making it
part of CRTP itself and requiring the definition of more packet
types.  The negative consequence is that the negotiation scheme would
have to be defined by each link protocol.

If a full entropy encoding is used, then one might want to reduce the
smallest number of bits to less than 8.  That would mean we'd
probably want to re-align to a byte boundary after the encoded
values.  Would it be reasonable to just have that re-alignment be
implicit?
							-- Steve


From rem-conf-request@es.net Tue Dec 17 00:10:44 1996 
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From: Kyungran Kang <krkang@cosmos.kaist.ac.kr>
Message-Id: <199612170506.OAA09016@cosmos.kaist.ac.kr>
Subject: (Q) mbone media on demand service
To: mbone@isi.edu
Date: Tue, 17 Dec 1996 14:06:37 +0900 (KST)
Cc: rem-conf@es.net
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Hello, everyone.

I'm planning to build a mbone archive site.
And I'd like to have a sample site. Is anyone operating a mbone
media on demand service?

I've visited mice server but it seemed to have stopped.

If you are oprating a mbone media on demand service or if you are
planning to, please let me know. It will be great help to me.

Thanks very much in advance.

- krkang

From rem-conf-request@es.net Tue Dec 17 10:30:59 1996 
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To: rem-conf@es.net
Subject: question about CBC session.
Date: Wed, 18 Dec 1996 00:30:32 +0900
From: Yozo Toda <yozo@aohakobe.ipc.chiba-u.ac.jp>


Hi, I have two questions about "CBC newsworld Online".

joining the CBC session, "Incoming Call" window pops up. 
the window has no buttons, only three lines; 

                Incoming Call
          You have an incoming call  from inviting you to
          join the following session

what's this? I wonder a few lines missed?
some message appears to the console;

  can't read  "sip_invite_status(131.211.35.48/850866683)": no such variable
      while executing
  "switch $sip_invite_status($id)..."
      invoked from within
  bluh bluh bluh

I use sdr2.2a23, is it a problem?

another question, I tried to join the video session of CBC, but
vic2.7a29 core-dumped.  is it also old-version problem?

btw, vat4.0a2 works fine for audio session.

-- yozo.


From rem-conf-request@es.net Tue Dec 17 17:11:59 1996 
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Date: Tue, 17 Dec 1996 17:11:04 -0500
To: rem-conf@es.net
From: David P Pleva <dpleva@popserve.lerc.nasa.gov>
Subject: Mbone Seminar: "Hybrid Communications Networks" (12/18/96)
Cc: M#u#Baldizzi@qmgate.lerc.nasa.gov

Mbone  Seminar  Announcement:

The Satellite Networks and Architectures Branch and the 
Computer Services Division of NASA Lewis Research Center 
plans to broadcast a seminar on

        "Hybrid Communications Networks:
          Architectures, Management and Services"

by    Professor John S. Baras Director
        Center for Satellite and Hybrid Communication Networks 
        University of Maryland.

Date:          Wednesday, December 18, 1996
Time:           10:30am -11:30 AM (EST)

About the Topic:

Satellite and terrestrial networks are becoming increasingly critical in the
hybrid architecture of the NII/GII.  Hybrid networks, in our terminology, are 
seamlessly interoperable terrestrial wireless and wireline and satellite 
networks.  In this seminar, we will describe the research program of the 
Center in hybrid networking.  We will focus on:  hybrid network architectures, 
asymmetric Internet via satellite, network modeling and simulation, broadband 
satellite constellations, network management and information dissemination
over hybrid networks.  We will also describe several commercial products in 
Internet services and network management that have recently resulted from 
the research of the Center.

About the Speaker:

Professor John S. Baras is the Director of the NASA Center for Hybrid and 
Satellite Communication Networks and the Lockheed Martin Chair in
Systems Engineering at the University of Maryland.  He received his Ph.D.
degree from Harvard University.  He is a Fellow of the IEEE and has received 
several awards for his research activities.
----------------------------------------------------------------------------
----------
David Pleva (Sterling Software) - CSD-7190  Networking and Telecom.             
NASA/Lewis Research Center  MS142-1                 (216) 433-9061
email:  dpleva@popserve.lerc.nasa.gov        FAX:  (216) 433-8000
----------------------------------------------------------------------------
----------


From rem-conf-request@es.net Wed Dec 18 03:49:20 1996 
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Date: Wed, 18 Dec 1996 09:43:14 +0100 (MET)
From: Beatrice Chambet <Beatrice.Chambet@dassault-elec.fr>
To: REMCONF <rem-conf@es.net>
Subject: Videoconference sofware under windows NT
Message-ID: <Pine.SOL.3.95.961218094059.3197A-100000@viviane.dassault-elec.fr>
MIME-Version: 1.0
Content-Type: TEXT/PLAIN; charset=US-ASCII


Hello,


 Is there any videoconference software for windows NT 3.51 or 4.00 with
the quickcam color camera.


Thanks in advance.

bye



From rem-conf-request@es.net Wed Dec 18 04:45:07 1996 
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                          process doing -bs
Date: Wed, 18 Dec 1996 10:43:23 +0100 (MET)
From: Wieland Holfelder <whd@pi4.informatik.uni-mannheim.de>
X-Sender: whd@eratosthenes
To: Kyungran Kang <krkang@cosmos.kaist.ac.kr>
Cc: mbone@isi.edu, rem-conf@es.net
Subject: Re: (Q) mbone media on demand service
In-Reply-To: <199612170506.OAA09016@cosmos.kaist.ac.kr>
Message-Id: <Pine.OSF.3.95.961218102949.14511B-100000@eratosthenes>
Mime-Version: 1.0
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On Tue, 17 Dec 1996, Kyungran Kang wrote:
> I'm planning to build a mbone archive site.
> And I'd like to have a sample site. Is anyone operating a mbone
> media on demand service?

Seems as there is quite some development going on in this area...

yes, we are also working on a media on demand service for the MBone,
we call it MBone VCR on demand service. It is the followup project
of the MBone VCR that I implemented some time ago at ICSI.
The system will consist of Java based clients (they may run as
java application as well as java applets) and multithreaded servers
implemented in java and native C++ code.

They communicate through ROJ (remote objects for java) which is
a Corba compatible implementation of SUN written in Java.

Clients can request recordings and playbacks of MBone sessios 
at the servers, there will be access control and of course, clients
have full VCR control during playback and recording.
The server has an interface to SAP (Version 0 and 1) and SDP (Version 0
as described in the sdp draft 03.2). The server makes announcments
available to clients and announces playbacks over these protocols.

We hope to have a first alpha version ready early next year.
Stay tuned.

-- Wieland
---------------------------------------------------------------------
Wieland Holfelder                              University of Mannheim
Praktische Informatik IV                      Phone: +49-621-292-5679
L 15,16                                       Fax  : +49-621-292-5745
68131 Mannheim                     whd@pi4.informatik.uni-mannheim.de
Germany                    http://www.informatik.uni-mannheim.de/~whd
---------------------------------------------------------------------


From rem-conf-request@es.net Wed Dec 18 06:42:22 1996 
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          Wed, 18 Dec 1996 03:41:23 -0800
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          id <g.06394-0@bells.cs.ucl.ac.uk>; Wed, 18 Dec 1996 11:40:37 +0000
From: Isidor Kouvelas <I.Kouvelas@cs.ucl.ac.uk>
X-Organisation: University College London, CS Dept.
X-Phone: +44 171 419 3670
To: Beatrice Chambet <Beatrice.Chambet@dassault-elec.fr>
cc: REMCONF <rem-conf@es.net>
Subject: Re: Videoconference sofware under windows NT
In-reply-to: Your message of "Wed, 18 Dec 1996 09:43:14 +0100." <Pine.SOL.3.95.961218094059.3197A-100000@viviane.dassault-elec.fr>
Date: Wed, 18 Dec 1996 11:40:25 +0000
Message-ID: <2911.850909225@cs.ucl.ac.uk>
Sender: I.Kouvelas@cs.ucl.ac.uk

> Is there any videoconference software for windows NT 3.51 or 4.00 with
>the quickcam color camera.

The VFW vic port will work. For information look at:
http://www.cs.ucl.ac.uk/staff/I.Kouvelas/vic

Isidor

From rem-conf-request@es.net Wed Dec 18 10:05:18 1996 
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Date: Wed, 18 Dec 96 10:04:21 EST
From: andrew@calvin.dgbt.doc.ca (Andrew Patrick)
Message-Id: <9612181504.AA13348@calvin.dgbt.doc.ca>
In-Reply-To: Yozo Toda <yozo@aohakobe.ipc.chiba-u.ac.jp> "question about CBC session." (Dec 18, 12:30am)
Reply-To: andrew@calvin.dgbt.doc.ca
X-Mailer: Mail User's Shell (7.2.5 10/14/92)
To: Yozo Toda <yozo@aohakobe.ipc.chiba-u.ac.jp>, rem-conf@es.net
Subject: Re: question about CBC session.


On Dec 18, 12:30am, Yozo Toda wrote:
} Subject: question about CBC session.
| 
| Hi, I have two questions about "CBC newsworld Online".
| 

I am the orginator of the CBC Newsworld session.

| joining the CBC session, "Incoming Call" window pops up. 
| the window has no buttons, only three lines; 
| 
|                 Incoming Call
|           You have an incoming call  from inviting you to
|           join the following session
| 
| what's this? I wonder a few lines missed?
| some message appears to the console;
| 
|   can't read  "sip_invite_status(131.211.35.48/850866683)": no such variable
|       while executing
|   "switch $sip_invite_status($id)..."
|       invoked from within
|   bluh bluh bluh
| 
| I use sdr2.2a23, is it a problem?

This is strange, as there should not be any SIP requests attached to
this session.  I have not heard of any previous problems like this.

| another question, I tried to join the video session of CBC, but
| vic2.7a29 core-dumped.  is it also old-version problem?

I don't know, but that version of vic is out of date so you may want
to update it on principle.

You did not mention the platform you are running on.


-- 
Andrew Patrick, Ph.D. <andrew@calvin.dgbt.doc.ca>
1996 Psychic Report Card: http://www.csicop.org/articles/psychic-1996.html

From rem-conf-request@es.net Wed Dec 18 10:54:31 1996 
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To: andrew@calvin.dgbt.doc.ca
cc: rem-conf@es.net
Subject: Re: question about CBC session.
In-reply-to: Your message of "Wed, 18 Dec 1996 10:04:21 EST." <9612181504.AA13348@calvin.dgbt.doc.ca>
Date: Thu, 19 Dec 1996 00:52:57 +0900
From: Yozo Toda <yozo@aohakobe.ipc.chiba-u.ac.jp>


thanks for your response, andrew!
today I receive no SIP request.

> I don't know, but that version of vic is out of date so you may want
> to update it on principle.

yeah, I'll update it later.
I compiled vic with X11R6 library, but
today I did "ldd `which vic`" and found that
the vic binary uses openwin library.
it may be the problem...

> You did not mention the platform you are running on.

a ha, sorry. SS20, solaris2.4 with X11R6.0.

I want to try NetBSD on SS20 someday (-:

-- yozo.


From rem-conf-request@es.net Wed Dec 18 13:27:44 1996 
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To: Isidor Kouvelas <I.Kouvelas@cs.ucl.ac.uk>
cc: Beatrice Chambet <Beatrice.Chambet@dassault-elec.fr>, 
    REMCONF <rem-conf@es.net>
Subject: Re: Videoconference sofware under windows NT
In-reply-to: Your message of "Wed, 18 Dec 1996 11:40:25 GMT." <2911.850909225@cs.ucl.ac.uk>
Mime-Version: 1.0
Content-Type: text/plain; charset=us-ascii
Date: Wed, 18 Dec 1996 19:17:16 +0100
From: Frank Lyonnet <Frank.Lyonnet@sophia.inria.fr>

> > Is there any videoconference software for windows NT 3.51 or 4.00 with
> >the quickcam color camera.
> 
> The VFW vic port will work. For information look at:
> http://www.cs.ucl.ac.uk/staff/I.Kouvelas/vic
> 
> Isidor

Quoted from connectix home page :

> Connectix supports Windows 3.1x and Windows 95. WindowsNT is not yet
> supported, but we are close to releasing a driver for the grayscale QuickCam
> (due in Dec).

This would mean that color quickcam cannot be used at all under Windows NT, 
even
with VideoForWindows port of vic. And as far as I know the grayscale driver 
cannot
be used with the color quickcam.

So my answer to the question is : there is no videoconference software for NT
that can work with color quickcam until connectix release a suitable driver.

Cheers,

Frank
Frank.Lyonnet@inria.fr


From rem-conf-request@es.net Wed Dec 18 18:59:18 1996 
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Message-ID: <c=US%a=_%p=msft%l=RED-73-MSG-961218211716Z-14518@INET-01-IMC.microsoft.com>
From: Max Morris <maxm@MICROSOFT.com>
To: 'Beatrice Chambet' <Beatrice.Chambet@dassault-elec.fr>, 
    'Isidor Kouvelas' <I.Kouvelas@cs.ucl.ac.uk>
Cc: 'REMCONF' <rem-conf@es.net>
Subject: RE: Videoconference sofware under windows NT
Date: Wed, 18 Dec 1996 13:17:16 -0800
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Encoding: 28 TEXT

Are there NT drivers for the color Connectix QuickCam available yet?  I
didn't think so.

Microsoft has just released the second beta of NetMeeting 2.0, with
audio and video conferencing support.  It works on NT 4.0.  No problem
using the b/w QuickCam, since there definitely are drivers available for
it.  If the color drivers are also available, then the color QuickCam
should also work.

For information on NetMeeting, see http://microsoft.com/netmeeting

-Max

>----------
>From: 	Isidor Kouvelas[SMTP:I.Kouvelas@cs.ucl.ac.uk]
>Sent: 	Wednesday, December 18, 1996 3:40 AM
>To: 	Beatrice Chambet
>Cc: 	REMCONF
>Subject: 	Re: Videoconference sofware under windows NT
>
>> Is there any videoconference software for windows NT 3.51 or 4.00 with
>>the quickcam color camera.
>
>The VFW vic port will work. For information look at:
>http://www.cs.ucl.ac.uk/staff/I.Kouvelas/vic
>
>Isidor
>

From rem-conf-request@es.net Thu Dec 19 03:33:15 1996 
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Date: Thu, 19 Dec 1996 19:32:22 +1100 (EST)
From: sanjay <sanjay@socs.uts.EDU.AU>
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To: rem-conf@es.net
Subject: Has someone added MPEG to vic?
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Hi,

Has someone added MPEG compression/decomp to Vic ? I would appreciate any
pointer ?

Thx in advance.


  Sanjay Kumar Jha          E-mail: sanjay@socs.uts.edu.au
  ph: +61 2 9514  1858		Fax: +61 2 9514 1807


From rem-conf-request@es.net Thu Dec 19 11:17:01 1996 
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Date: Thu, 19 Dec 1996 08:16:54 -0700
In-Reply-To: sanjay <sanjay@socs.uts.EDU.AU> "Has someone added MPEG to vic?" (Dec 19, 7:32pm)
References: <Pine.SOL.3.95.961219193015.12876A-100000@linus>
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Subject: Re: Has someone added MPEG to vic?
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On Dec 19,  7:32pm, sanjay wrote:
> 
> Has someone added MPEG compression/decomp to Vic ? I would appreciate 
> any pointer ?

I'm also curious to know if anyone has ever tried adding support for Indeo 
video to vic?




-- 
    John Cole, Systems Engineer         Tel: 415.845.5200      
    Precept Software, Inc.              Fax: 415.845.5235
    1072 Arastradero Road               http://www.precept.com
    Palo Alto, CA 94304                 E-Mail: jcole@precept.com





From rem-conf-request@es.net Thu Dec 19 17:50:24 1996 
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From: Bill Fenner <fenner@parc.xerox.com>
To: mbone@isi.edu, rem-conf@es.net
Subject: Announcing the 5.1 release of mtrace, the multicast traceroute client.
Message-Id: <96Dec19.142648pst.177713@crevenia.parc.xerox.com>
Date: Thu, 19 Dec 1996 14:26:33 PST

$Id: README.mtrace,v 5.1 1996/12/19 21:53:24 fenner Exp $

	Multicast Traceroute

	Release 5.1
	December 19, 1996

	available from ftp.parc.xerox.com,
	file pub/net-research/ipmulti/mtrace5.1.tar.Z
	binaries pub/net-research/ipmulti/mtrace5.1-sparc-sunos41x.tar.Z
	         pub/net-research/ipmulti/mtrace5.1-sparc-solaris2.tar.Z
	         pub/net-research/ipmulti/mtrace5.1-alpha-osf1.tar.Z
	         pub/net-research/ipmulti/mtrace5.1-sgi-irix.tar.Z

The 5.1 release represents the divorce of the multicast traceroute tool
>from the mrouted distribution.  To reduce confusion with mrouted,
mtrace needed a version number from a completely different space, so I
"arbitrarily" chose 5.

The 5.1 release fixes the following bugs:

o Workaround for "0 pps" printed for little-endian hops.  There is a
  missing parenthesis in mrouted 3.8's mtrace responder, causing half
  of a calculation to be byte-swapped and half not.  mtrace now uses
  the time the response was received instead of the timestamp in the
  packet if it detects this bug.

o TTL required is now calculated correctly.

o Wrong-type packets are properly ignored.  If multiple mtrace's are
  running on a single machine, it's possible that a DVMRP_NEIGHBORS2
  ("mrinfo") packet could have been misinterpreted as an mtrace packet,
  displaying a long trace with lots of "0.0.0.0" entries.

o "-m" traces that take less hops than requested are now handled properly.

o No longer allows magic address expansion for multicast addresses, so
  "224.100.102" is now a syntax error instead of a synonym for "224.100.102.0".

o If statistics were aborted because the route changed, don't count them
  as having been printed.

o If the last hop forgot to fill in its IP address, fill it in with the
  source address of the returned packet.

o Missing statistics detection is now more robust; it will just print
  blank lines rather than the confusing "58/58=100%" then "-58/0 = --%".

The 5.1 release has the following new features:

o Netmask printing (with -v) and default route notification

o Traces are no longer restarted from scratch when the route changes;
  they are continued from the information already known.

o The new Watch-Path mode (-P) allows an individual path to be monitored
  periodically; mtrace will print the path initially and only when it
  changes.

o Default multicast TTL changed from 64 to 127, to better match reality.

o Per-hop statistics are normally only printed as a packet-per-second count;
  there are so many bugs in the collection of these counters that the packet
  loss that they report is generally misleading.  The -T flag may be used
  to include these numbers anyway.

o Don't print statistics if there is no group specified, since the per-group
  statistics are the only ones that are meaningful.

o Change the default destination when using "-g <gwy>" to <gwy> if it is
  a different subnet than the host running mtrace.  This elminiates the need
  to use "mtrace -g foo bar foo".

o Print "[reset counters]" on a hop that has reset its counters.

o Allow the number of extra hops to be traced after a non-responding
  router to be specified with the "-e <nhops>" flag.

o Statistics collection in passive mode (slightly buggy)

o The -O flag may be used to supress use of the Router Alert IP option.
  Some versions of Cisco's IOS corrupt multicast traceroute replies when
  the request is sent with IP options, so if the last hop is a Cisco you
  may need to use -O.

From rem-conf-request@es.net Thu Dec 19 19:56:44 1996 
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To: Bill Fenner <fenner@parc.xerox.com>
cc: mbone@isi.edu, rem-conf@es.net
Subject: Re: Announcing the 5.1 release of mtrace, the multicast traceroute 
         client.
In-reply-to: Your message of "Thu, 19 Dec 96 14:26:33 PST." <96Dec19.142648pst.177713@crevenia.parc.xerox.com>
Date: Thu, 19 Dec 1996 16:55:45 PST
Sender: Bill Fenner <fenner@parc.xerox.com>
From: Bill Fenner <fenner@parc.xerox.com>
Message-Id: <96Dec19.165547pst.177712@crevenia.parc.xerox.com>

I made a last-minute logic error which made the source un-buildable on
systems with ANSI compilers and header files that define the __P
macro.  I have replaced mtrace5.1.tar.Z on ftp.parc.xerox.com with the
fixed version (mtrace.c will have version 5.1.1.1 in the fixed
version).  Those who fetched the source and are having trouble building
should fetch the fixed version and try again.

  Bill

(pardon me while I go catch a plane... Happy Holidays, everyone!)

From rem-conf-request@es.net Fri Dec 20 02:34:23 1996 
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subscribe
---------------------------------------------
Wang Hui
Department of Systems Engineering and Applied Mathemathics
Changsha Institue of Technology
Changsha, Hunan Province
410073, P.R.China
Tel: +86(731)455-5601 EXT. 87709,87703(Office)
     +86(731)421-1364 (Home)
E-mail: huiwang@csit.edu.cn

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From: "Mr. Hui Wang" <huiwang@csit.edu.cn.>
To: VideoConf Mailing list <rem-conf@es.net>
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subscribe

-------------------------------------------------------------------
King Wang
Multimedia Lab.  
Changsha Institue of Technology
Changsha, Hunan Province
410073, P.R.China
Tel: +86(731)455-2914 (Office)
     +86(731)421-1364 (Home)
E-mail: huiwang@csit.edu.cn



From rem-conf-request@es.net Fri Dec 20 10:10:06 1996 
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Date: Fri, 20 Dec 1996 09:53:43 -0500 (EST)
From: Arif Ghafoor <ghafoor@ecn.purdue.edu>
To: cellular@dfv.rwth-aachen.de, perform@tay1.dec.com, tccc@cs.umass.edu
Cc: announcements.chi@xerox.com, arl@arl1.wustl.edu, atm@bbn.com, 
    ccrc@dworkin.wustl.edu, cip@bbn.com, cnom@maestro.bellcore.com, 
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    f-troup@aurora.cis.upenn.edu, ietf@isi.edu, rem-conf@es.net
Subject: Call for paper on multimedia for IEEE COMPSAC97






IEEE 25-th Annual International Computer Software and Application Conference (COMPSAC97)
will be held in Washington D.C. (August 13 - 15, 1997)
Papers on various topics, including multimedia, can be submitted.
The detailed Call for Papers can be found at the following web site:

	http://www.cs.umn.edu/~tsai/compsac/compsac.html


The deadline for the paper and panel proposal submission is January 15, 1997.

From rem-conf-request@es.net Fri Dec 20 21:20:31 1996 
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From rem-conf-request@es.net Wed Dec 25 08:32:32 1996 
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help subscribe

From rem-conf-request@es.net Wed Dec 25 20:12:38 1996 
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Date: Thu, 26 Dec 1996 09:12:27 +0800 (CST)
From: Niu Junyu <niujy@conger.neu.edu.cn>
Subject: Re: (no subject)
To: Dan Takkunen <takk@dascom-systems.com>
Cc: rem-conf@es.net
In-Reply-To: <32C12C76.2C0E@dascom-systems.com>
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Hi,


send your reguest to 
           
              mbone-request@zephyr.isi.edu



dora.

From rem-conf-request@es.net Wed Dec 25 20:52:58 1996 
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Date: Thu, 26 Dec 1996 09:50:56 +0800
From: haitao@moon.bjnet.edu.cn (Zhang Haitao)
Message-Id: <9612260150.AA29433@moon>
Apparently-To: rem-conf@es.net

hi,

now i am using an receiver-oriented multicast file transfer application,
and the sender sends data packet periodly(eg. 100ms/frame), then how 
can i calculate the total thoroughput of the data transfer rate? 
just the data amount divide elasped time? 

thanks ,
-----------------------------------------------------------------------
Zhang haitao,                           Email:haitao@moon.bjnet.edu.cn
Lab of Multimedia and Networking,       http://koala.ee.tsinghua.edu.cn
Dept. of Electronic Engineering,             
Tsinghua University,                        
Beijing, P.R.China
-----------------------------------------------------------------------



From rem-conf-request@es.net Thu Dec 26 19:52:50 1996 
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Date: Thu, 26 Dec 1996 16:44:27 -0800 ()
From: Stephen Casner <casner@precept.com>
To: Niu Junyu <niujy@conger.neu.edu.cn>
cc: Dan Takkunen <takk@dascom-systems.com>, rem-conf@es.net
Subject: Re: (no subject)
In-Reply-To: <Pine.3.89.9612260925.A5198-0100000@conger>
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This is wrong.  The rem-conf and mbone lists are entirely separate.
To subscribe to the rem-conf list, send a message to
rem-conf-request@es.net.

On Thu, 26 Dec 1996, Niu Junyu wrote:

> Hi,
> 
> 
> send your reguest to 
>            
>               mbone-request@zephyr.isi.edu
> 
> 
> 
> dora.
> 

							-- Steve


From rem-conf-request@es.net Sun Dec 29 03:39:54 1996 
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From: Bernard Aboba <aboba@internaut.com>
To: "'rem-conf@es.net'" <rem-conf@es.net>
Cc: "'casner@precept.com'" <casner@precept.com>, 
    "'louie@uu.net'" <louie@uu.net>, 
    "'thalerd@eecs.umich.edu'" <thalerd@eecs.umich.edu>
Subject: RTP/RTCP scaling
Date: Sun, 29 Dec 1996 00:36:05 -0800
Encoding: 134 TEXT

>> On Tue, 17 Dec 1996, Louis A. Mamakos wrote:
>> 
>> > On one particular network using IP multicast technology, we restrict (using
>> > filters and other mechanisms) the ability to source to particular groups from
>> > the vast majority of multicast participants whose role is generally described
>> > as "receiver".  This also handily solves the RTCP scaling problem :-)
>> 
>> Yes, but it also handily precludes the feedback that RTCP was
>> intended to provide.

I would agree with Steve that RTCP feedback is potentially a useful thing; the
data provided by RTCP is essential not only for engineering but also for
business reasons. However, I would also agree with Louie that RTCP is not currently deployable at
large scale. This is somewhat distressing, since I believe that we are likely
to see large scale deployment of RTP over the next year, and network providers
are making judgements based on current technology, that could have the potential
to shape the future of internet multcast for some time to come. 

The problem is serious enough to warrant both a short term and a long term
approach. In the short term, definition of new RTP profiles can probably get us
through the next 12-18 months, by which time we may have come up with
a better long term solution. I would suggest that dealing with both short
and long term aspects of the problem is an appropriate agenda item for
AVT. 

>In a "radio broadcast" type environment, it's just not at all clear to
>me that the feedback which RTCP supplies is most appropriately
>multicast to the same group as the content.  

As we discussed in the last AVT meeting, there are some cases in which
it is helpful to have the RTCP feedback end up on a *different* group than
RTP. For example, in a sparse mode PIM environment, putting RTP
and RTCP on different groups will result in the RTCP group ending up on
the shared tree, whereas if they were both on the same group, then
if the RTP source rate were high enough, the RTP/RTCP group would be on
to the source-based tree with a potentially high number of forwarding cache
entries. 

However, unless you believe that DVMRP will go away sometime soon (and I
don't see evidence of this), then such schemes will probably
merely result in deployable islands of multicast connectivity unable to connect to the
MBONE for fear of the problems that would result from doing so. Ultimately,
I fear there only one solution: reducing the effective number of RTCP senders
dramatically, via summarization. 

>There a a bunch of
>issues, (including but not limited to privacy) to be considered.
>Having this data be unicast back to some designated monitoring
>device(s), perhaps sampled statistically, may be more appropriate.

While unicasting addresses the issue of multicast forwarding table explosion, as well
as some privacy concerns, it does not address the problem of RTCP 
rate control and population estimate convergence. Thus with unicasting,
it is still possible for the sender to be overwhelmed with RTCP feedback. 

>> I am concerned about implementation of policy at the wrong level or
>> by inflexible means.  For now, it may well be that the policy must be
>> imposed by the network operator administratively rather than by the
>> applications.  However, if the network devices are incapable of
>> sending multicast in the direction from the "receivers", then
>> applications can't make use of RTCP's functionality even when
>> operating at a small scale.

Assuming that RTP and RTCP are placed on separate groups, and that
things work as we believe they will, then I believe that use of local filters
blocking RTCP feedback should not be necessary; however, the need
for administrative scoping or boundary-level filtering will remain until the
RTCP sender proliferation problem can be addressed. 

>I don't seem to have very many policy choices to make right now.  I
>can do what I've done (prohibit the "receivers" from generating
>multicast traffic), and have it work.  Or I can try to replicate the
>environment on the MBONE and hope it scales to the 100000 or more
>customers on my network.  Having the network work less and less well
>as more and more participants use it is to be avoided at *all* costs.
>Especially when my control (as a network operator) over the protocol
>implementations on the end devices is limited.

I think that there is an important lesson embedded in this problem. This
is that Internet growth has reached the point where the time between
deployment of "experimental" services and commercial implementations
is growing smaller and smaller. This means that it is increasingly difficult
to conduct field pilot programs in which early versions can be debugged and
modified based on experimental results. In such environments, it is typical
to see increasing emphasis on simulation techniques, with simulations
validated based on limited trials. 

>Having the state in the network scale anything approaching linearally
>with the number of users of the network is broken.  This is my
>reality.  The only problem is scale.  If it doesn't scale, it's
>broken.

Rest assured, there is a good level of agreement that improvements in
RTCP scaling are needed. Where do not perhaps have as good agreement
is in understanding what requirements a solution will need to fulfill. Your
input is solicited. 

> I would rather see this filtering done by source-specific pruning
> where the receivers, following directions from the session
> originator, ask for pruning of all sources other than the intended
> data sender.  Since not all the mechanisms required to implement this
> are in place yet, we'll have to use other means.  But I hope that the
> alternatives are implemented such that they can be turned off when
> source pruning is ready.

For source-specific pruning to help, RTP receivers would need to not
listen to the RTCP feedback. Is this what is being suggested? Assuming
that the RTP and RTCP groups were separated, and the RTCP group
moved to the shared tree, how could source-specific pruning be employed?
I'm unclear on this. 

>I am deploying a production network, with technology that is available
>today.  When and if these other technologies become available, they
>may be deployed in the network.  I don't see any source specific
>pruning capability available in either "production quality" multicast
>routers or in stacks running in the end systems.  When and if these
>additional features appear in the future, they'll be considered for
>deployment into a production service; but they'll need to make sense
>both from a business, operations and technology perspective.

>Though I hope there's serious progress in a multicast interdomain
>routing infrastructure and a way to specify and enforce policy between
>multicast routing domains.  The fact that the MBONE operates as one
>big, happy "family" is reminiscent of the days when we ran EGP, except
>there's even fewer tools available.  

>From where I sit, this looks like an accurate assessment of current
technology. However, Dave Thaler has recently done some very interesting work on how
to interconnect multicast routing domains. Care to comment, Dave? 


Bernard Aboba
Microsoft



From rem-conf-request@es.net Mon Dec 30 15:10:37 1996 
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To: rem-conf@es.net
Cc: rkennerly@wpine.com
Subject: Is there an "official" UDP port range for A/V data.
Date: Mon, 30 Dec 1996 15:06:42 -0500
From: Brian O'Shea <boshea@wpine.com>


A while back there was mail discussing the ranges of UDP ports to use for
audio, video and whiteboard data:

	[0, 16384) - lowest priority, unclassified
	[16384, 32768) - highest priority, i.e. audio
	[32768, 49152) - medium priority, i.e. whiteboard
	[49152, 65536) - low priority, i.e. video

A quick grep of my RFC's and internet drafts reveals no "official" policy on
this.

Is this published anywhere, or is it simply "tribal knowledge"?  Are the
firewall vendors members of the tribe?

TIA

-bos

+***********************************************************************+
+ Brian O'Shea					White Pine Software	+
+ Reflector Group Leader			542 Amherst St		+
+ bos@wpine.com					Nashua NH, 03063	+
+ Phone: 603-886-9050				Fax: 603-886-9051	+
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From rem-conf-request@es.net Mon Dec 30 20:58:41 1996 
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To: rem-conf@es.net
From: Ross Finlayson <finlayson@lvn.com>
Subject: Re: Is there an "official" UDP port range for A/V data.

>A while back there was mail discussing the ranges of UDP ports to use for
>audio, video and whiteboard data:
>
>	[0, 16384) - lowest priority, unclassified
>	[16384, 32768) - highest priority, i.e. audio
>	[32768, 49152) - medium priority, i.e. whiteboard
>	[49152, 65536) - low priority, i.e. video
>
>A quick grep of my RFC's and internet drafts reveals no "official" policy on
>this.

This reminds me of something that I forgot to bring up when we were
discussing this last:

According to the IANA document
"ftp://ftp.isi.edu/in-notes/iana/assignments/port-numbers", the "dynamic
and/or private" ports are those in the range 49152 through 65535; ports
lower than 49152 are in the "well-known" or "registered" ranges.  This
suggests (to me anyway) that transient multimedia sessions, such as those
created by SDP browsers, should really be using only the range 49152 through
65535: the range that mrouted currently treats as "low priority, i.e., video".

Perhaps it's *this* port range that we should really be partitioning and
prioritizing?

        Ross.


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Date: Tue, 31 Dec 1996 16:24:14 -0800 ()
From: Stephen Casner <casner@precept.com>
To: Brian Bulkowski <brianb@navio.com>
cc: rem-conf@es.net
Subject: Re: Group size estimation
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During the IETF meeting, Brian Bulkowski sent the subject message
describing some algorithms that might be used to calculate the RTCP
interval without having to keep track of all participants when the
number gets very large.  I'll reply to some specific points in
Brian's message below, but I'd like to first comment on the
discussions in the meeting and on the history of the RTCP interval
algorithm.

There was some confusion in the meeting because Henning Schulzrinne
had misinterpreted the RTP spec regarding the initial RTCP interval
being randomized over [0,2.5] seconds rather than [1.25,3.75].  Van
Jacobson pointed out that this difference is important so the
receiver can listen to the incoming report rate before sending its
first report.  However, the RTCP interval calculation algorithm as
currently specified does NOT include the "reconsideration" idea that
Henning was describing.  That is, when the interval expires,
calculate a new estimate of the group size and skip sending the
report if the previous interval was too short.  Without this
reconsideration, waiting doesn't help: if a million receivers were to
auto-start in perfect synchrony for a scheduled event, there would be
a million packets sent over a 2.5 second interval.  To allow enough
time to hear other's reports at low bandwiths so that reconsideration
work properly, I believe we should also make the minimum RTCP
interval scale upward from the current 5 second value for low
bandwidths.  The 5 second value is appropriate for on the order of
100 kb/s.

This notion of reconsideration is particularly important for very
large groups and ties in with Brian's question about algorithms for
estimating group size without keeping track of all the participants.
I don't remember a discussion of the reconsideration idea in past AVT
meetings (I didn't find it in the transcripts or minutes), but we did
talk about estimating group size via sampling, as did Christian
Huitema in the recent meeting.

That brings me to a few specific responses to Brian's message:

Regarding RFC 1889 section 6.2.1 on estimating the number of
participants in order to calculate the RTCP interval:

> One interesting point of the wording is that
> it assumes an estimate of what the report interval is, but I see nowhere
> else that suggests a good method for getting that estimate.

This may appear recursive, but it's not really a problem.  Timeouts
for maintaining the membership table are based on the current value
of the RTCP interval, and the interval will change to converge to the
right value as the membership table grows.  The initial estimate for
the RTCP interval is the minimim value (5 seconds).

> In most algorythms, you need an estimate of the report interval, including
> the spec'd algorythm; here's one that comes to mind. Maintain an estimate
> of the RR interval by tracking a small number of specific SSRC RR timings.
> Make a table including the last time heard, number of times heard from, and
> SSRC value. Whenever you receive an SSRC BYE packet, invalidate that timing
> entry. When you have received a certain number of SSRC packets (5?), add
> the interval to the running average, then clear the entry. On every RR
> receipt, check for an empty entry, if you find one, use that SSRC value for
> estimation. Have a timer run though the interval estimation table to throw
> out entries that it hasn't heard from for a while. Suggested time is three
> times the current report interval.

One problem here is that it would take 5 intervals before one could
send a report.  I think the algorithm needs to be able to respond
after a time on the order of one interval.  A more serious problem is
that if everyone is using this algorithm, you'll either get deadlock
with everyone waiting for someone else to send first so they can
estimate the interval, or one guy will send at the minimum 5 seconds
and everyone else will follow.

> Is there another way of estimating the report interval besides watching
> everyone else? If you have a size-of-group estimate, and a stream size
> estimate, you can calculate the report interval. This can be used instead,
> except in initial cases. There was a bit of discussion at the IETF about
> initial wait intervals before sending RR packets.

As Christian pointed out, you can estimate the group size through
sampling.  You keep track of only a subset of the SSRC id's you hear
by accepting only those that match some criterion with a probability
of 1/(2^n).  Then you multiply the number you count by 2^n to get
your estimate.  As Henning and I have been discussing in some private
email, you may want to start with n=0 for the normal small-scale
case, and then increase n (discarding those SSRCs that don't match
the new lower probability criterion) when you reach the max number
you want to save.

> In terms of getting group size, once you have a good estimate of the report
> interval, you need only keep track of the number of RR packets received
> (say, per second, making sure you use a different timer than all others).
> Whenever you desire the size of the receiver group, multiply the current
> report interval by the number of RRs received, and you get the size of the
> receiver group.

I don't think you would do this; it would be the other way around as
in the previous paragraph.

> This algorythm seems to converge slowly, but has a nice attribute that if
> you are losing RR packets, you get a larger size estimation, which leads to
> the sending of fewer RR packets, which means that the algorythm is
> congestion friendly.

The sampling technique is subject to under-estimation if RR packets
are lost.  However, assuming that the bandwidth is sufficient to
allow 5% for RTCP, then with the reconsideration the rate should be
close to the target value and you should not be losing lots of RR
packets.  If packets arrive faster than the session bandwidth should
allow, this should insure that the reconsideration backs off.

> In order to dynamically resize the hash table, you could keep track of a
> few marker buckets, and whenever there is a collision in one of your marker
> buckets, realloc your table and start again. Or, you could make an estimate
> at the amount of collision you are getting, and multiply your size estimate
> by that factor. This allows people to pick certain table sizes and not
> reallocate. In the "marker buckets" you need to keep track of the exact
> SSRCs received, so you can detect true collisions vs. bogus collisions due
> to inaccurate estimation of the report interval.

This is approaching the sampling algorithm described above.

> I'm sure that the RTP designers thought over these issues. It is clear that
> the current set of "mays" in section 6.2.1 will not work with large
> receiver groups. Would it be better to add suggested algorythms and words
> for larger groups, or would it be better to spec one of the other
> estimation algorythms (like the polling one) which doesn't require the
> receiver's calculation of receiver groupsize? Or would it be better to turn
> off RRs in large groups? What's the thinking?

We deliberately chose to keep the example algorithm in the RTP spec
simple in the hope that we could avoid having implementers decide
that it was too complicated and just put in a fixed number instead.
Or even worse, to abandon RTP entirely.  This is a very real concern
since there is not really any way to "enforce" a standard.  On the
other hand, after more time has elapsed and RTP has gained some level
of acceptance, it is appropriate now to consider specifying a more
complete algorithm.
							-- Steve


