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From: Christoph Fleck <fleck@Rcs1.urz.tu-dresden.de>
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Reply-To: Multimedia Referenzzentrum <mmt-ref@tu-dresden.de>
To: Ross Finlayson <finlayson@live.com>
cc: Patrick De Muynck <Patrick.DeMuynck@belnet.be>, Rex Xi Xu <rexx@cs.cmu.edu>, 
    mbone@ISI.EDU, rem-conf@es.net, 
    Multimedia Referenzzentrum <mmt-ref@tu-dresden.de>
Subject: Re: "liveCaster": MP3 streaming via multicast
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> >I have a related question, which is if there exists a generic program
> >which can handle the "application/sdp" MIME type, i.e. it should be
> >able to accept an .sdp file, parse the SDP announcement, and then
> >startup the proper audio/video tools
> [...]
> >Would "multikit" be able to do this?
> 
> Not at present, although it could be modified to do this (thanks for the
> suggestion :-)  Right now multikit, like "sdr", can launch SDP sessions
> from its own, built-in directory GUI - but cannot launch SDP files that
> come from someone else's directory GUI (e.g., from a separate web browser).

Try a perl script (written by Rex Xi Xu rexx@cs.cmu.edu) that can do that.
Thats the only one, I have fond to do that, on Windows. It needs still 
some extensions.
My description (sorry only in german):
http://www-mm.urz.tu-dresden.de/mbone/sessions/

the script for win95/98 (using start.exe for background):
ftp://ftp-mm.urz.tu-dresden.de/pub/mbone/mmrz/sdp.pl/win95/sdp.pl

the script for unix:
ftp://ftp-mm.urz.tu-dresden.de/pub/mbone/mmrz/sdp.pl/unix/sdp.pl

If anybody makes some improvements or has a better soultion, 
please let me know.

Rgds,
Christoph Fleck

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




From rem-conf Mon Mar 01 16:17:23 1999 
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From: Virginia Thomas <virginia@real.com>
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CC: mbone@ISI.EDU, rem-conf@es.net, Ross Finlayson <rsf@CS.STANFORD.EDU>
Subject: Re: "liveCaster": MP3 streaming via multicast
References: <199902281721.SAA22268@vivaldi.belnet.be>
Content-Type: multipart/mixed; boundary="------------143F257DE31EF856AF0FDFA"
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This is a multi-part message in MIME format.

--------------143F257DE31EF856AF0FDFA
Content-Type: text/plain; charset=us-ascii
Content-Transfer-Encoding: 7bit

Hi Patrick,

Yes, the RealPlayer G2 does support local and network playback of MP3.
You were able to play the mp3 file locally which means you must have 
the Digital Bitcast Mpeg DLLs on your system. 

The Player will play network based MP3 content as long as it is streamed
using standard RTP and the payload is specified according to what the
avt wg has documented to date. For instance, the RealPlayer can be used
to playback content served by vic/vat and the ICast server, as you
mentionned below. I've attached a NasaTV SDP file which you can load
directly into your RealPlayer. I believe NasaTV is being streamed by an
ICast server. Note: the player will  grab the H.261 and PCM plugins from
the update server.

Can you send me the SDP file that you tried loading into the RealPlayer?

THanks,

Virginia




Patrick De Muynck wrote:
> 
> I thought that RealPlayer G2 should be able to do the job, because:
> 
> - with the new so called "scalable multicast" support it's able to
>   play an RTP/RCP multicasted stream and furthermore to parse the
>   SDP announcement that describes the stream
> 
> - it has Codec support for playing MP3
> 
> It didn't work for me however.
> 
> Here's what I tried: I started broadcasting with the liveCaster,
> sending SDP announcements.  I launched the RealPlayer by using Peter
> Parnes' WEB-based multicast Session Directory
> (http://www.cdt.luth.se/~peppar/progs/mSD/) and clicking on the
> liveCaster announcement (alternatively you can just save the SDP
> announcement on disk as a file with extension ".sdp"
> and open that file in RealPlayer).  Note: if you install RealPlayer G2
> on a PC it becomes the default program for the MIME type "application/sdp"
> MIME type, file extension = .sdp .
> 
> The result was that it receives the stream but can't play it: actually
> it goes to the RealNetworks WEB site to find an "unknown codec"
> plugin.  Then it reports that no appropriate plugin was found :-(
> 
> That's rather strange because the RealPlayer streamed the MP3 file
> from disk all right...  So why?
> 
> FYI: using RealPlayer G2 / scalable multicast this way works for
> playback of one-way MBone sessions which use H261/DVI/PCM...
> 
> I have a related question, which is if there exists a generic program
> which can handle the "application/sdp" MIME type, i.e. it should be
> able to accept an .sdp file, parse the SDP announcement, and then
> startup the proper audio/video tools like MBone or RealPlayer G2.  So
> something similar as mLaunch (http://www.cdt.luth.se/~peppar/progs/mlaunch/)
> but it should also support Windows and it should be more configurable
> concerning the choice of the tools (not only MBone).  Would "multikit"
> be able to do this?
> 
> Patrick De Muynck
> 
> BELNET, the Belgian National Research Network
> 
> Patrick.DeMuynck@belnet.be  |   DWTC - BELNET
> Tel : +32/2/238.36.76       |   Wetenschapsstraat, 4
> Fax : +32/2/513.57.30       |   B-1000 Brussel
> 
> In message <3.0.5.16.19990214154327.20975616@shell7.ba.best.com>,
> Ross Finlayson writes:
> >FYI, a quick update on the multicast MP3 audio tools that I first announced
> >a couple of weeks ago:
> >
> >"liveCaster" (the multicast MP3 audio source):
> >       <http://www.live.com/liveCaster/>
> >This can now take its MP3 input not only from ".mp3" files, but also from
> >stdin, or from a HTTP stream (such as a 'Shoutcast' server, or just from a
> >MP3 file on the web).  Also, it will now automatically make an SDP
> >announcements for its session (although you can choose not to do so).
> >
> >"playRTPMPEG" (the MP3/RTP/UDP multicast receiver tool):
> >       <http://www.live.com/multikit/playRTPMPEG.html>
> >Unix versions (Linux, FreeBSD, Solaris/SPARC and IRIX) are now available.
> >These will write the received MP3 frames to 'stdout', so you can use this
> >tool to feed a separate MP3 player program (such as "X11Amp").  (The
> >Windows version feeds the "Winamp" MP3 player, but by using a hack where it
> >acts as a special-purpose HTTP server.)  This tool doesn't do any buffering
> >of the incoming MP3/RTP/UDP data; instead, it assumes that whatever
> >application it feeds (via 'stdout') will have sufficient buffering.
> >
> >(I repeat my plea for someone to add MP3 decoding to an existing multicast
> >audio tool, such as "rat".  This would ultimately be a better solution.)
> >
> >Finally, both tools now do RTCP, although currently only at a basic level.
> >(CNAME is the only item sent in a SDES, and the RRs are 'empty'.)
> >
> >       Ross.

-- 

-----------------------------------------------------------
Virginia Thomas, Program Manager               206.674.2267
                                          virginia@real.com
RealNetworks, Inc.                                         
1111 Third Ave #2900           Home of RealAudio, RealVideo
Seattle, WA    98101                          and RealMedia
http://www.real.com

--------------143F257DE31EF856AF0FDFA
Content-Type: text/plain; charset=us-ascii; name="nasa.sdp"
Content-Transfer-Encoding: 7bit
Content-Disposition: inline; filename="nasa.sdp"

v=0
o=-   IN IP4 
s=<No title>
i=<No author> <No copyright>
t=0 0
m=audio 22840 RTP/AVP 0
c=IN IP4 224.2.135.86/127
m=video 57914 RTP/AVP 31
c=IN IP4 224.2.135.86/127

--------------143F257DE31EF856AF0FDFA--




From rem-conf Mon Mar 01 16:17:23 1999 
From rem-conf-request@es.net Mon Mar 01 16:17:22 1999
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Date: Mon, 01 Mar 1999 15:49:42 -0800
To: rem-conf@es.net
From: Ryan Thurman <lobosdelmar@worldnet.att.net>
Subject: video training info
Cc: drmm@mindspring.com
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Hello,

My name is Ryan Thurman and I am researching some information for a resource
guide on the application of videoconferencing in telehealth for Dr. Marlene
Maheu.  
Your answer to the following questions would be very helpful and your company
will be cited appropriately.

I am wondering if you have any information on the type of training that is
packaged with your equipment.  
What is the normal procedure for training individuals after they have
purchased
the equipment?
Could you describe some the important points involved in training people to
use
the videoconferencing equipment?
What can you suggest for people who want more training with the equipment?

Also, if your company is interested re-seller arrangements with Dr. Maheu
please write at <drm@telehealth.net> and put your vendor information at this
address for us: 
<http://cybertowers.com/cgibin/prof.cgi>http://cybertowers.com/cgibin/prof.cgi


Thank you for your consideration,
Ryan Thurman
Research Assistant





From rem-conf Tue Mar 02 02:42:08 1999 
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To: Virginia Thomas <virginia@real.com>
cc: mbone@ISI.EDU, rem-conf@es.net, Ross Finlayson <rsf@CS.STANFORD.EDU>
Subject: Re: "liveCaster": MP3 streaming via multicast 
In-reply-to: Your message of "Mon, 01 Mar 1999 15:57:13 PST."
             <36DB2959.76D59F7E@real.com> 
Date: Tue, 02 Mar 1999 11:33:06 +0100
From: Patrick De Muynck <Patrick.DeMuynck@belnet.be>
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If memory serves me well, the files tmp.sdp-1 and tmp.sdp-2
included below are the SDP announcements as they were captured
respectively by the "multicast Session Directory" and by 
the Cisco router on our LAN.  I didn't capture the cache files
of sdr though, but I can try again.

Do they look OK?

::::::::::::::
tmp.sdp-1
::::::::::::::
v=0
o=- 920054817 3129043744 IN IP4 172.24.198.32
s=tmp
i=MP3 audio
e=patrick@belnet.be
b=AS:320
t=3129043744 3129045364
a=type:broadcast
a=tool:liveCaster v1.0a26
m=audio 49334 RTP/AVP 14
c=IN IP4 235.246.133.254/127
::::::::::::::
tmp.sdp-2
::::::::::::::
Session Name: tmp
  Description: MP3 audio
  Group: 0.0.0.0, ttl: 0, Contiguous allocation: 1
  Lifetime: from 20:08:24 MET Feb 26 1999C until 20:35:24 MET Feb 26 1999C
  Uptime: 3d15h, Last Heard: 3d15h
  Announcement source: 172.24.198.32, destination: 224.2.127.254
  Created by: - 920054817 3129044904 IN IP4 172.24.198.32
  Phone number: 
  Email: patrick@belnet.be
  URL: 
  Media: audio 49334 RTP/AVP 14
    Media group: 235.246.133.254, ttl: 127
::::::::::::::


In message <36DB2959.76D59F7E@real.com>,
Virginia Thomas writes:
>
>Hi Patrick,
>
>Yes, the RealPlayer G2 does support local and network playback of MP3.
>You were able to play the mp3 file locally which means you must have 
>the Digital Bitcast Mpeg DLLs on your system. 
>
>The Player will play network based MP3 content as long as it is streamed
>using standard RTP and the payload is specified according to what the
>avt wg has documented to date. For instance, the RealPlayer can be used
>to playback content served by vic/vat and the ICast server, as you
>mentionned below. I've attached a NasaTV SDP file which you can load
>directly into your RealPlayer. I believe NasaTV is being streamed by an
>ICast server. Note: the player will  grab the H.261 and PCM plugins from
>the update server.
>
>Can you send me the SDP file that you tried loading into the RealPlayer?
>
>THanks,
>
>Virginia
>
>
>
>
>Patrick De Muynck wrote:
>> 
>> I thought that RealPlayer G2 should be able to do the job, because:
>> 
>> - with the new so called "scalable multicast" support it's able to
>>   play an RTP/RCP multicasted stream and furthermore to parse the
>>   SDP announcement that describes the stream
>> 
>> - it has Codec support for playing MP3
>> 
>> It didn't work for me however.
>> 
>> Here's what I tried: I started broadcasting with the liveCaster,
>> sending SDP announcements.  I launched the RealPlayer by using Peter
>> Parnes' WEB-based multicast Session Directory
>> (http://www.cdt.luth.se/~peppar/progs/mSD/) and clicking on the
>> liveCaster announcement (alternatively you can just save the SDP
>> announcement on disk as a file with extension ".sdp"
>> and open that file in RealPlayer).  Note: if you install RealPlayer G2
>> on a PC it becomes the default program for the MIME type "application/sdp"
>> MIME type, file extension = .sdp .
>> 
>> The result was that it receives the stream but can't play it: actually
>> it goes to the RealNetworks WEB site to find an "unknown codec"
>> plugin.  Then it reports that no appropriate plugin was found :-(
>> 
>> That's rather strange because the RealPlayer streamed the MP3 file
>> from disk all right...  So why?
>> 
>> FYI: using RealPlayer G2 / scalable multicast this way works for
>> playback of one-way MBone sessions which use H261/DVI/PCM...
>> 
>> I have a related question, which is if there exists a generic program
>> which can handle the "application/sdp" MIME type, i.e. it should be
>> able to accept an .sdp file, parse the SDP announcement, and then
>> startup the proper audio/video tools like MBone or RealPlayer G2.  So
>> something similar as mLaunch (http://www.cdt.luth.se/~peppar/progs/mlaunch/)
>> but it should also support Windows and it should be more configurable
>> concerning the choice of the tools (not only MBone).  Would "multikit"
>> be able to do this?
>> 
>> Patrick De Muynck
>> 
>> BELNET, the Belgian National Research Network
>> 
>> Patrick.DeMuynck@belnet.be  |   DWTC - BELNET
>> Tel : +32/2/238.36.76       |   Wetenschapsstraat, 4
>> Fax : +32/2/513.57.30       |   B-1000 Brussel
>> 
>> In message <3.0.5.16.19990214154327.20975616@shell7.ba.best.com>,
>> Ross Finlayson writes:
>> >FYI, a quick update on the multicast MP3 audio tools that I first announced
>> >a couple of weeks ago:
>> >
>> >"liveCaster" (the multicast MP3 audio source):
>> >       <http://www.live.com/liveCaster/>
>> >This can now take its MP3 input not only from ".mp3" files, but also from
>> >stdin, or from a HTTP stream (such as a 'Shoutcast' server, or just from a
>> >MP3 file on the web).  Also, it will now automatically make an SDP
>> >announcements for its session (although you can choose not to do so).
>> >
>> >"playRTPMPEG" (the MP3/RTP/UDP multicast receiver tool):
>> >       <http://www.live.com/multikit/playRTPMPEG.html>
>> >Unix versions (Linux, FreeBSD, Solaris/SPARC and IRIX) are now available.
>> >These will write the received MP3 frames to 'stdout', so you can use this
>> >tool to feed a separate MP3 player program (such as "X11Amp").  (The
>> >Windows version feeds the "Winamp" MP3 player, but by using a hack where it
>> >acts as a special-purpose HTTP server.)  This tool doesn't do any buffering
>> >of the incoming MP3/RTP/UDP data; instead, it assumes that whatever
>> >application it feeds (via 'stdout') will have sufficient buffering.
>> >
>> >(I repeat my plea for someone to add MP3 decoding to an existing multicast
>> >audio tool, such as "rat".  This would ultimately be a better solution.)
>> >
>> >Finally, both tools now do RTCP, although currently only at a basic level.
>> >(CNAME is the only item sent in a SDES, and the RRs are 'empty'.)
>> >
>> >       Ross.
>
>-- 
>
>-----------------------------------------------------------
>Virginia Thomas, Program Manager               206.674.2267
>                                          virginia@real.com
>RealNetworks, Inc.                                         
>1111 Third Ave #2900           Home of RealAudio, RealVideo
>Seattle, WA    98101                          and RealMedia
>http://www.real.com



From rem-conf Tue Mar 02 09:58:49 1999 
From rem-conf-request@es.net Tue Mar 02 09:58:48 1999
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From: Orion Hodson <O.Hodson@cs.ucl.ac.uk>
X-Organisation: University College London, CS Dept.
X-Phone: +44 171 419 3704
To: Henning Schulzrinne <hgs@cs.columbia.edu>, rem-conf@es.net
Subject: G726/727 RTP Profile comments
Date: Tue, 02 Mar 1999 17:47:28 +0100
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Quick question on the RTP profile for G726 / G727.

* Section 4.5.6 G726-16/24/32/30 

The payload format in draft-ietf-avt-profile-new-04.txt for G726 only
specifies the packing order for G726-32 (4 bits per sample).  The
specified scheme aligns to byte boundaries as:

	+----+----+----+----+---
	| S1 | S0 | S3 | S2 | ...
	+----+----+----+----+---

Looking at the ITU's G726 spec there does not appear to be a bitstream
specified either.  An obvious approach for all encodings is pack to
the samples in the order they are produced.  It makes the
implementation marginally easier and means the sample time is
monotonically increasing.

For compatibility reasons it may be best to leave G726-32 alone.  

How many implementations implement this now?  

Are there implementations using 2,3,5 bits per sample?

What have they elected to do?

thanks
/orion

Orion Hodson                                 [Research Student]
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
Networked Multimedia Research Group      Tel: ++(0)171 419 3704
Department of Computer Science           Fax: ++(0)171 419 1397
University College London, Gower Street
London, WC1E 6BT, UK.     
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-



From rem-conf Tue Mar 02 11:21:10 1999 
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From: Florissa Colina <florissa@bmrc.berkeley.edu>
Subject: Reminder 3/3 Berkeley Multimedia, Interfaces and Graphics
  Seminar
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Berkeley Multimedia, Interfaces and Graphics Seminar

Parallel Software-Only Video Effects Processing

Wednesday, March 3, 1999, 1:00-2:30 p.m. 405 Soda Hall 

Ketan Meyer-Patel 
Computer Science Division - EECS, UC Berkeley 

Experience in the television and film industries shows that video effects
like fades, transitions, and titling improve the perceived quality of video
material. Traditional production methods for video effects, however, are
not well matched to the needs of Internet video streaming. In this talk, I
describe a parallel software-only video effects processing system designed
specifically for streaming Internet video. I will describe mechanisms for
supporting different types of parallelism and show how their design is
constrained by properties of streaming video formats and protocols. I will
also describe a novel control mechanism built on Scalable Reliable
Multicast (SRM). 

The seminar is broadcast on the Internet Mbone. The addresses are video
224.2.163.7/57076 and audio 224.2.147.61/27202. 

Sponsored by the Berkeley Multimedia Research Center 


____________________________________________

Florissa Colina
Berkeley Multimedia Research Center (BMRC)
http://bmrc.berkeley.edu/
626 Soda Hall #1776
University of California
Berkeley, CA 94720-1776
Phone: (510) 643-0800   FAX: (510) 642-5615  
Email: florissac@bmrc.berkeley.edu



From rem-conf Tue Mar 02 22:43:10 1999 
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Date: Tue, 02 Mar 1999 21:14:41 -0500
From: "John Weiler, CTO and Founder" <john_weiler@TheOTG.com>
Organization: The OBJECTive Technology Group
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Dear Colleagues,

The Interoperability Clearinghouse is formalizing as a govt. sanctioned 501C6
non-profit organization this quarter.  We will be establishing several
categories of membership to help map key information elements into this internet
based knowledge repository of standards, "validated & interoperable COTS product
suites" and associated implementation services.  

Those interested in providing support for this joint government/industry effort,
and help major industries shorten their technology adoption process, please
respond to the following questionnaire by COB March 24 if your organization is
interested in becoming one of the "founding fathers".  This is not a statement
of commitment, but a means of assessing what industry groups would be best
represented.  Please indicate your interest areas;

_ Am interested in participating in the IC Coalition
_ Am interested in seeing validated standards or product profiles
_ Am willing to sharing testing, implementation configuration results
_ Would be willing to pay a small fee to access an internet based configurator
for finding optimal set of "interoperable product suites"
_ Am interested in using the IC List server for RFI/RFP notifications
_ Am willing to share lessons learned in exchange of accessing all others
_ Am interested in some of the more interactive support services provided to its
members (_IT architecture support, _technology studies, _tools reports, _RFP
development, _technology education)

Am interested in participating the upcoming Industry@OOTS (object oriented
technology solutions) executive forums where these validated architecture
frameworks are outlined by industry (healthcare, finance, government,
manufacturing, telecommunications) in the following manner;
_ speaking/presenting
_ sponsoring
_ attending
_ chairing

_ Please call to set up an onsite brief for our organization.

On March 25th, the AFCEA MD chapter will be hosting a breakfast meeting on the
Interoperability Clearinghouse and want to know if your organization should be
included in the list of potential participants.  Afterwards, we will be
conducting the IC Work Group meeting.  Time and place will be provided to all
respondents.

For more information on this effort, please visit the following 
Interoperability Clearinghouse URLs;
http://www.omg.org/techprocess/meetings/ic.html
http://www.infoworld.com/cgi-bin/displayStory.pl?981113.whsoft.htm
http://www.fcw.com/pubs/fcw/1999/0208/fcw-newsinterop-2-8-99.html
http://www.gcn.com/gcn/1999/February8/1c.htm

With another major article in next months Healthcare Informatics.  We look
forward to working with you to "making IT happen".
Or give a call if you have any questions.....


*********************************************
"From Architectures to Reality"

John A. Weiler
CTO & Founder, The OBJECTive Technology Group
Ombudsman, Interoperability Clearinghouse Coalition
Member; OMG, IEEE, ACM, AFCEA
8011 Washington Ave.
Alexandria, VA 22308
703-768-0400(v) 703-765-9295(f)
http://www.TheOTG.com (Industry@OOTS Forums)
john_weiler@TheOTG.com

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



From rem-conf Wed Mar 03 00:31:56 1999 
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Date: Wed, 3 Mar 1999 00:24:46 -0800 (Pacific Standard Time)
From: Stephen Casner <casner@cisco.com>
To: rem-conf@es.net
Subject: Requesting agenda items for AVT
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Colin and I are collecting requests for agenda items at the AVT
meeting two weeks hence.  We have received the following requests
already:

Jon Crowcroft, RTCP location reports
Ross Finlayson, A More Loss-Tolerant RTP Payload Format for MP3 Audio
Bill Strahm, RTP MIB draft
Katsushi Kobayashi, RTP Payload Format for DV Format Video

							-- Steve




From rem-conf Thu Mar 04 08:29:51 1999 
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From: Chinmay Padhye <padhye@eng.usf.edu>
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Subject: Voice Packet Traces.
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Hello:

	I am a Computer Science Graduate student at the University Of
South Florida. My thesis topic deals with transporting voice over IP. I am
looking for trace files for my simulations. I am interested in traces with
different sampling rates. i.e sampling at 20ms, 40ms , etc.. 

	I would appreciate it if someone could point me to where I can get
these.

	Your help is appreciated.

Yours Truly,

Chinmay V. P.
University Of South Florida.




From rem-conf Thu Mar 04 10:09:27 1999 
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From: Dave Singer <singer@apple.com>
Subject: payload name and type for H263+ (RFC2429)
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RFC 2429 says that "Payload Type (PT): The Payload Type shall specify the
H.263+ video payload format."  Presumably somewhere the Mime name is
defined for this.  What is it? (Where is it?).  I missed it.

Thanks!
David Singer
Apple Computer/QuickTime



From rem-conf Thu Mar 04 17:44:38 1999 
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--NextPart

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

	Title		: Real-Time Transport Protocol Management Information Base
	Author(s)	: M. Baugher, B. Strahm, I. Suconick
	Filename	: draft-ietf-avt-rtp-mib-04.txt
	Pages		: 29
	Date		: 03-Mar-99
	
This memo defines an experimental  Management Information Base
(MIB) for use with network management protocols in
TCP/IP-based internets.  In particular, it defines objects for
managing Real-Time Transport Protocol systems [1].  Comments
should be made to the IETF Audio/Video Transport Working Group
at rem-conf@es.net.This memo does not specify a standard for the Internet
community.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-mib-04.txt

Internet-Drafts are also available by anonymous FTP. Login with the username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-ietf-avt-rtp-mib-04.txt".

A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html 
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
	mailserv@ietf.org.
In the body type:
	"FILE /internet-drafts/draft-ietf-avt-rtp-mib-04.txt".
	
NOTE:	The mail server at ietf.org can return the document in
	MIME-encoded form by using the "mpack" utility.  To use this
	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
	a MIME-compliant mail reader.  Different MIME-compliant mail readers
	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
	up into multiple messages), so check your local documentation on
	how to manipulate these messages.
		
		
Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
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Content-Type: text/plain
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ENCODING mime
FILE /internet-drafts/draft-ietf-avt-rtp-mib-04.txt

--OtherAccess
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	directory="internet-drafts"

Content-Type: text/plain
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--NextPart--





From rem-conf Fri Mar 05 00:29:53 1999 
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Date: Fri, 5 Mar 1999 00:22:04 -0800 (Pacific Standard Time)
From: Stephen Casner <casner@cisco.com>
To: Dave Singer <singer@apple.com>
cc: cabo@tzi.org, rem-conf@es.net
Subject: Re: payload name and type for H263+ (RFC2429)
In-Reply-To: <v04020a29b304798564d6@[17.255.20.102]>
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On Thu, 4 Mar 1999, Dave Singer wrote:
> RFC 2429 says that "Payload Type (PT): The Payload Type shall specify the
> H.263+ video payload format."  Presumably somewhere the Mime name is
> defined for this.  What is it? (Where is it?).  I missed it.

I have made another revision to both the RTP spec and the profile.
(See my next message on that topic.)  In the profile, I have added
"H263-1998", which is the name suggested by Joerg or Carsten, I
believe.

Also, please note that Philipp Hoschka has done a rough draft on the
MIME registration of names from the RTP profile.
							-- Steve




From rem-conf Fri Mar 05 00:39:49 1999 
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Date: Fri, 5 Mar 1999 00:34:16 -0800 (Pacific Standard Time)
From: Stephen Casner <casner@cisco.com>
To: rem-conf@es.net
Subject: Revised RTP spec and profile drafts
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I have revised the Real-time Transport Protocol (RTP) specification
and the RTP Audio/Video Profile drafts.  Both of these may now be
ready for working group Last Call -- that is a question I will ask at
the IETF meeting.  The spec includes the corrections that Jonathan
Rosenberg provided to the code in the appendix, and some improvements
he suggested in the wording of sections 6.2 and 6.3 which contain most
of the new work since RFC 1889.  Recall that I had asked everyone to
review those sections at the last meeting, and I appreciate Jonathan's
careful review.

The Profile draft includes the additional payload formats to date, and
refers to a separate draft for MIME registration of the encoding /
payload format names.  You may have noticed the I-D announcement for
that separate draft (currently at the rough draft stage) authored by
Philipp Hoschka: draft-hoschka-rtp-mime-00.txt

I am not sure if the revised RTP drafts are going to be officially
posted or not.  I submitted the request before the deadline, but I did
not notice that the boilerplate text in the .txt version was not
updated and so I got a rejection.  (I look at the .ps version while
editing because it has the changebars.)  Henning Schulzrinne created a
set of scripts and programs that allow a LaTeX source to generate both
the .ps and .txt versions, but I had forgotten that the I-D
boilerplate for the .txt version comes from a Tcl script off in
another directory rather than from the one in the source used by
LaTeX.  I am hoping that Cynthia will at least post the .ps versions,
or even better, accept the corrected .txt versions I sent.

In any case, you can pick up the drafts from

ftp://ftp.isi.edu/mbone/avt/draft-ietf-avt-rtp-new-03.txt
ftp://ftp.isi.edu/mbone/avt/draft-ietf-avt-rtp-new-03.ps
ftp://ftp.isi.edu/mbone/avt/draft-ietf-avt-profile-new-05.txt
ftp://ftp.isi.edu/mbone/avt/draft-ietf-avt-profile-new-05.ps

The .ps versions are recommended because of the changebars.

							-- Steve




From rem-conf Fri Mar 05 08:52:12 1999 
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If you have a spare minute of two, i'd appreciate if a couple of you could
join the NASA Medical Surveillance Discussion for a few minutes and email a
sentence or two about the quality of the audio and video you receive.

                          thanks,
                                o-> el

Session information:
--------------------
Title: NASA Medical Surveillance Discussion
Audio: 224.2.193.25:24478
Video: 224.2.193.25:52544


ps.  Anyone know exactly how to do something like:

Click <rtp addr="224.2.193.25:24478" type="audio">here</rtp> to hear a live
conversation.

...and have it bring the appropriate mbone util on your system?



From rem-conf Fri Mar 05 08:52:13 1999 
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To: itc@ieee.org, comsoc-chapters@ieee.org, multicomm@cc.bellcore.com,
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Subject: ICNP'99 Call for Papers
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Please pass this along to anyone who might be interested. Note that the
ICNP'99 web site is now active.

Thanks,
Terry Todd

++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++



			   CALL FOR PAPERS

	  7th INTERNATIONAL CONFERENCE on NETWORK PROTOCOLS

		The Royal York Hotel, Toronto, Canada

		    October 31 - November 3, 1999

		 http://boa.crl.mcmaster.ca/~icnp99/


In just 6 years, ICNP has established itself as one of the premier
conferences in the computer networking field. ICNP deals with all
aspects of communication protocols, from design and specification, to
verification, testing, performance analysis, implementation, and
performance tuning.  Protocol functions of interest include (but are
not limited to) network access, switching, routing, flow and
congestion control, multimedia transport, wireless protocols, network
security, Web protocols and applications, electronic commerce, network
management, interoperability, and internetworking.

ICNP uses a single-track format which provides an intimate setting for
discussion and debate. The conference is known for it's high quality
papers from the research communities of IEEE COMSOC, IEEE Computer
Society, and ACM SIGCOMM.  To be considered for acceptance, a
submission should present a significant contribution in its topic
area. Selected papers will be forwarded for "fast track" consideration
to IEEE/ACM Transactions on Networking. ICNP also includes panel
sessions in which experts in a specific area offer their observations
and opinions about a current topic.

ICNP'99 will be held in Toronto, whose name is a Huron Indian word
meaning "place of meeting".  Toronto is Canada's largest city, the
capital of the province of Ontario, and one of the most exciting,
vibrant, and progressive cities in the world! It's attractions are far
too numerous to list. The United Nations has designated Toronto as the
world's "most ethnically-diverse city".  It has the world's tallest
building, the CN Tower, is the third-largest theatre centre in the
English-speaking world, and contains over 5,000 restaurants and
eateries. Toronto is also a very safe city with many family
attractions. ICNP'99 will be held at the famous Royal York Hotel. The
Royal York has been in operation since 1929 and is one of the grand
hotels of Canada.  It is located in the centre of downtown Toronto, a
focal point for shopping, culture and nightlife.


Important Dates
---------------

Paper Submission Deadline:      May 3, 1999
Notification of Acceptance:     August 2, 1999

Camera Ready Due:               August 25, 1999

Tutorials:                      October 31, 1999
Conference:                     November 1-3, 1999


Submission Information
----------------------

Submissions will be made via email. Details of the procedure are given
at the ICNP'99 web site: http://boa.crl.mcmaster.ca/~icnp99/


General Chair
-------------
Johnny Wong, University of Waterloo
jwwong@bcr.uwaterloo.ca

Technical Program Chairs
------------------------
Joe Bannister, USC Information Sciences Institute
Email: joseph@isi.edu

Terry Todd, McMaster University
Email: todd@mcmaster.ca

Tutorial Chair
--------------
Kevin Almeroth, UC Santa Barbara
Email: almeroth@cs.ucsb.edu

Local Arrangements Chair
------------------------
Bart Domzy, Trent University

ICNP Steering Committee
-----------------------
Mostafa Ammar, Georgia Insitute of Technology
Mohamed Gouda, University of Texas at Austin
Simon Lam, University of Texas at Austin
David Lee, Bell Labs
Ming T. (Mike) Liu, Ohio State University
Raymond Miller, University of Maryland, College Park
Krishan Sabnani, Bell Labs

Program Committee
-----------------
Sudhir Aggarwal, SUNY at Binghamton
Kevin Almeroth, UCSB
Mostafa Ammar, Georgia Tech
Anish Arora, Ohio State Univ.
Harmen van As, Vienna Technical Univ.
Anindo Banerjea, USC ISI
Jose Brustoloni, Bell Labs, Lucent
Ken Calvert, Univ. of Kentucky
Imrich Chlamtac, Univ. of Texas, Dallas
Jorge Cobb, Univ. of Texas, Dallas
Jon Crowcroft, University College, London
Michael Dahlin, Univ. of Texas, Austin
Christophe Diot, Sprint Labs
Chris Edmondson-Yurkanan, Univ. of Texas, Austin
Mario Gerla, UCLA
Li Gong, Sun Microsystems
Mohamed Gouda, Univ. of Texas, Austin
Mark Handley, AT&T Center for Internet Research at ICSI
Teruo Higashino, Osaka Univ.
Chao-Ju Jennifer Hou, Ohio State Univ.
Allison Mankin, USC ISI-East
Ibrahim Matta, Northeastern Univ.
Melody Moh, San Jose State Univ.
Mart Molle, Univ. of California, Riverside
Sanjoy Paul, Bell Labs, Lucent
Chunming Qiao, SUNY at Buffalo
Luigi Rizzo, Univ. of Pisa
Marco Schneider, SBC Technology Resources
Gurdip Singh, Kansas State Univ.
Martha Steenstrup, BBN
David Su, NIST
Kenji Suzuki, Kokusai Denshin Denwa Co.
Ljiljana Trajkovic, Simon Fraser Univ.
Brett Vickers, Rutgers Univ.
David Yau, Purdue Univ.
Geoffrey Xie, Naval Postgraduate School
Ellen Zegura, Georgia Tech
Hui Zhang, Carnegie Mellon Univ.

ICNP is sponsored by the IEEE Computer Society










From rem-conf Fri Mar 05 10:13:24 1999 
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Unfortunately the discussion has come to an end sooner than i had expected.
 However, this was the first in a series of such discussions and it is
expected that each subsequent discussion will be made available to the
public via the mbone.  (If anyone has a special interest in viewing these
sessions, please send a simple email message to me so that i can present
your names to the discussion coordinators.  This will increase the
likelihood that future sessions will be made available via the mbone).
Many thanks to those of you who took the time to view the session and
respond; i appreciate your help.

                                 o-> el  



From rem-conf Fri Mar 05 11:00:12 1999 
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Subject: Re: Live: NASA Medical Surveillance Discussion 
In-reply-to: Your message of "Fri, 05 Mar 99 08:43:54 PST."
             <3.0.32.19990305114353.009aba00@mail.hq.nasa.gov> 
Date: Fri, 5 Mar 1999 10:55:04 PST
Sender: Bill Fenner <fenner@parc.xerox.com>
From: Bill Fenner <fenner@parc.xerox.com>
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In message <3.0.32.19990305114353.009aba00@mail.hq.nasa.gov> you write:
>ps.  Anyone know exactly how to do something like:
>
>Click <rtp addr="224.2.193.25:24478" type="audio">here</rtp> to hear a live
>conversation.
>
>...and have it bring the appropriate mbone util on your system?

You want:

Click <A HREF="data:application/sdp,v=0%0Ao=me%201%201%20IN%20IP4%20my.host%0As=here%0Am=audio%201234%20RTP/AVP%200%0Ac=IN%20IP4%20224.2.17.12/127">here</A>.

i.e. create an SDP description (RFC2327) and encode it using a data:
URL (RFC2397).  (And then find a launcher that handles application/sdp;
there are one or two floating around.)

  Bill



From rem-conf Fri Mar 05 12:29:14 1999 
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Date: Fri, 5 Mar 1999 12:25:56 -0800 (Pacific Standard Time)
From: Stephen Casner <casner@cisco.com>
To: Orion Hodson <O.Hodson@cs.ucl.ac.uk>
cc: rem-conf@es.net
Subject: Re: G726/727 RTP Profile comments
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Orion,

> Quick question on the RTP profile for G726 / G727.

As I was editing the RTP profile, I came upon the same question you
ask.  I also sent a message to Henning to ask if anything had been
established.

> Looking at the ITU's G726 spec there does not appear to be a bitstream
> specified either.  An obvious approach for all encodings is pack to
> the samples in the order they are produced.  It makes the
> implementation marginally easier and means the sample time is
> monotonically increasing.

Isn't this merely a question of big-endian or little-endian bit
packing within octets?  Note that we have a mix of the two ends across
the collection of payload formats because some have come out of the
Internet world and some have come out of the telecom world.

One could define a consistent set of little-endian bit packings for
the other sample sizes as well.  People writing code for little-endian
machines would find the samples nicely aligned when they picked them
up as words.  Folks writing on big-endian machines will have to swap
bytes first, then pick up samples from the bottom of a word.  Nothing
unusual, no lack of timestamp monotonicity.
							-- Steve




From rem-conf Fri Mar 05 14:16:11 1999 
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From: anand@ece.iisc.ernet.in (SVR Anand)
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Subject: codecs
To: rem-conf@es.net
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Hi,

	I am wondering if it is ever possible for the sender to send the
implementation of the software codec he is going to use to the recipients in
some portable byte code. This byte code that implements the codec 
functionality can either be sent on-the-fly or off-line depending on the
feasibility. There should be some way of making the application use the
codec supplied in the mentioned scenarios, I am not sure how to go about this.
	If what I am saying can be realised, then the receiver need not have
to implement all those varieties of codecs the senders might use for, say,
bandwidth efficiency. The sender is now free to choose any codec he has and
can expect the receiver to have the same. 
	I think as many sophisticated codecs get developed, it can take
lot of time for the deployment on all the receivers before the sender can 
really use. Of course, the question of, should I send the codec implementions
for everything ? should be addressed alongwith perhaps many.
	Please pardon me if I sound too vague and whimsical :) 

Regards
Anand



From rem-conf Fri Mar 05 14:16:11 1999 
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--NextPart

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

	Title		: Sampling of the Group Membership in RTP
	Author(s)	: J. Rosenberg, H. Schulzrinne
	Filename	: draft-ietf-avt-rtpsample-02.txt
	Pages		: 10
	Date		: 04-Mar-99
	
   In large multicast groups, the size of the group membership table
   maintained by RTP (Real Time Transport Protocol) participants may
   become unwieldy, particularly for embedded devices with limited
   memory and processing power. This document discusses mechanisms for
   sampling of this group membership table in order to reduce the memory
   requirements. Several mechanisms are proposed, and the performance of
   each is considered.
 
 
   NOTE:
 
   This is to inform you that Lucent Technologies has applied for and/or
   has patent(s) that relates to the binning algorithm which is a part
   of the specification.
 
   This declaration is being made pursuant to the provisions of IETF IPR
   Policy , Sections 10.3.1 and 10.3.2.
 
   Lucent Technologies Inc. will offer patent licenses for submissions
   made by it which are adopted as a standard by your organization as
   follows:
 
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   license on such included part(s) on reasonable, non-discriminatory
   terms and conditions.

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--NextPart

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

	Title		: RTP Profile for Audio and Video Conferences 
                          with Minimal Control
	Author(s)	: H. Schulzrinne
	Filename	: draft-ietf-avt-profile-new-05.txt
	Pages		: 38
	Date		: 04-Mar-99
	
         This memorandum is a revision of RFC 1890 in preparation
         for advancement from Proposed Standard to Draft Standard
         status. Readers are encouraged to use the PostScript form
         of this draft to see where changes from RFC 1890 are
         marked by change bars.
 
         This document describes a profile called 'RTP/AVP' for
         the use of the real-time transport protocol (RTP),
         version 2, and the associated control protocol, RTCP,
         within audio and video multiparticipant conferences with
         minimal control. It provides interpretations of generic
         fields within the RTP specification suitable for audio
         and video conferences. In particular, this document
         defines a set of default mappings from payload type
         numbers to encodings.
 
         This document also describes how audio and video data may
         be carried within RTP. It defines a set of standard
         encodings and their names when used within RTP. The
         descriptions provide pointers to reference
         implementations and the detailed standards. This document
         is meant as an aid for implementors of audio, video and
         other real-time multimedia applications.

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--NextPart

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

	Title		: RTP: A Transport Protocol for Real-Time Applications
	Author(s)	: H. Schulzrinne,  S. Casner, R. Frederick, V. Jacobson
	Filename	: draft-ietf-avt-rtp-new-03.txt
	Pages		: 95
	Date		: 04-Mar-99
	
         This memorandum is a revision of RFC 1889 in preparation
         for advancement from Proposed Standard to Draft Standard
         status. Readers are encouraged to use the PostScript form
         of this draft to see where changes from RFC 1889 are
         marked by change bars.
 
         This memorandum describes RTP, the real-time transport
         protocol. RTP provides end-to-end network transport
         functions suitable for applications transmitting real-
         time data, such as audio, video or simulation data, over
         multicast or unicast network services. RTP does not
         address resource reservation and does not guarantee
         quality-of-service for real-time services. The data
         transport is augmented by a control protocol (RTCP) to
         allow monitoring of the data delivery in a manner
         scalable to large multicast networks, and to provide
         minimal control and identification functionality. RTP and
         RTCP are designed to be independent of the underlying
         transport and network layers. The protocol supports the
         use of RTP-level translators and mixers.
 
 
   This specification is a product of the Audio/Video Transport working
   group within the Internet Engineering Task Force. Comments are
   solicited and should be addressed to the working group's mailing list
   at rem-conf@es.net and/or the authors.

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From rem-conf Fri Mar 05 15:06:39 1999 
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To: anand@ece.iisc.ernet.in (SVR Anand)
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Subject: Re: codecs 
In-reply-to: Your message of "Thu, 04 Mar 1999 18:52:02 GMT."
             <9903041322.AA24060@ece.iisc.ernet.in> 
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Date: Fri, 05 Mar 1999 23:58:45 +0100
From: Peter Parnes <peppar@cdt.luth.se>
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Sure, why not? It would be very simple to realize if you solve the usual 
security problems (or just ignore them :-).

On demand downloading of codecs is already done by e.g. RealPlayer. 

/P




From rem-conf Fri Mar 05 16:01:31 1999 
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From: "Neal Schneider" <nschneid@hotmail.com>
To: anand@ece.iisc.ernet.in, rem-conf@es.net
Subject: Re: codecs
Date: Fri, 05 Mar 1999 15:55:45 PST
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>Hi,
>
>	I am wondering if it is ever possible for the sender to send the
>implementation of the software codec he is going to use to the 
recipients in
>some portable byte code. 

General-purpose DSPs can use a shared memory to support various standard 
algorithms, loading only those needed to program memory. This approach 
only requires one set of codecs per DSP farm.  If one were to forward a 
proprietary algorithm, it would have to be ported to whatever DSP the 
receiver was using, creating dependancy issues.

Regards,
Neal 

______________________________________________________
Get Your Private, Free Email at http://www.hotmail.com



From rem-conf Fri Mar 05 16:45:50 1999 
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Cc: rem-conf@es.net
From: "Ted Brunner 503.627.1317" <ted.brunner@tek.com>
Reply-to: ted.brunner@tek.com
Subject: Re: codecs 
In-reply-to: Your message of Thu, 04 Mar 1999 18:52:02 +0000.
             <9903041322.AA24060@ece.iisc.ernet.in> 
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> 	I am wondering if it is ever possible for the sender to send the
> implementation of the software codec he is going to use to the recipients in
> some portable byte code. This byte code that implements the codec 

Netscape plug-in?



Ted Brunner			VideoTele.com
				Tektronix
				MS 50-475
ted.brunner@tek.com		14150 SW Karl Braun
503.627.1317			Beaverton OR 97005	
		



From rem-conf Fri Mar 05 20:29:05 1999 
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From: "Ram Kordale" <kordale@relic.engr.sgi.com>
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Date: Fri, 5 Mar 1999 15:20:26 -0800
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        "I-D ACTION:draft-ietf-avt-rtp-new-03.txt" (Mar  5,  5:10pm)
References: <199903052210.RAA24688@ietf.org>
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On Mar 5,  5:10pm, Internet-Drafts@ietf.org wrote:
> Subject: I-D ACTION:draft-ietf-avt-rtp-new-03.txt
>
> A New Internet-Draft is available from the on-line Internet-Drafts
directories.
> This draft is a work item of the Audio/Video Transport Working Group of the
IETF.
>
> 	Title		: RTP: A Transport Protocol for Real-Time Applications
> 	Author(s)	: H. Schulzrinne,  S. Casner, R. Frederick, V. Jacobson
> 	Filename	: draft-ietf-avt-rtp-new-03.txt
> 	Pages		: 95
> 	Date		: 04-Mar-99
>
>          This memorandum is a revision of RFC 1889 in preparation
>          for advancement from Proposed Standard to Draft Standard
>          status. Readers are encouraged to use the PostScript form
>          of this draft to see where changes from RFC 1889 are
>          marked by change bars.
>

Is there a standard place at the IETF web site where one can find the
postscript version (referred to above and in many other ID summaries)?

A search at the IETF web site only resulted in ASCII versions.

Thanks.
Ram

-- 
Ram Kordale
Advanced Media Products



From rem-conf Sat Mar 06 09:52:25 1999 
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Subject: Re: I-D ACTION:draft-ietf-avt-rtp-new-03.txt
In-reply-to: Your message of "Fri, 05 Mar 1999 15:20:26 PST." <9903051520.ZM28871@relic.engr.sgi.com>
Date: Sat, 06 Mar 1999 17:41:19 +0000
Message-ID: <1267.920742079@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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--> Ram Kordale writes:
>On Mar 5,  5:10pm, Internet-Drafts@ietf.org wrote:
>> Subject: I-D ACTION:draft-ietf-avt-rtp-new-03.txt
>>
>> A New Internet-Draft is available from the on-line Internet-Drafts
>directories.
>> This draft is a work item of the Audio/Video Transport Working Group of the
>IETF.
>>
>> 	Title		: RTP: A Transport Protocol for Real-Time Applications
>> 	Author(s)	: H. Schulzrinne,  S. Casner, R. Frederick, V. Jacobson
>> 	Filename	: draft-ietf-avt-rtp-new-03.txt
>> 	Pages		: 95
>> 	Date		: 04-Mar-99
>>
>>          This memorandum is a revision of RFC 1889 in preparation
>>          for advancement from Proposed Standard to Draft Standard
>>          status. Readers are encouraged to use the PostScript form
>>          of this draft to see where changes from RFC 1889 are
>>          marked by change bars.
>>
>
>Is there a standard place at the IETF web site where one can find the
>postscript version (referred to above and in many other ID summaries)?
>
>A search at the IETF web site only resulted in ASCII versions.

They normally appear on the AVT charter page, I don't know why this didn't
happen this time. Alternatively, see

ftp://ftp.isi.edu/mbone/avt/draft-ietf-avt-rtp-new-03.txt
ftp://ftp.isi.edu/mbone/avt/draft-ietf-avt-rtp-new-03.ps
ftp://ftp.isi.edu/mbone/avt/draft-ietf-avt-profile-new-05.txt
ftp://ftp.isi.edu/mbone/avt/draft-ietf-avt-profile-new-05.ps

Colin



From rem-conf Sat Mar 06 11:55:24 1999 
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From: anand@ece.iisc.ernet.in (SVR Anand)
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Subject: Re: codecs
To: peppar@cdt.luth.se (Peter Parnes)
Date: Sat, 6 Mar 99 10:26:17 GMT+5:30
Cc: rem-conf@es.net
In-Reply-To: <199903052258.XAA17929@basil.cdt.luth.se>; from "Peter Parnes" at Mar 5, 99 11:58 pm
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Peter Parnes

	Oh! It means what I am saying is not all that fancy. Does this also
mean that the RTP payload type management can get somewhat simplified ? The
receiver need not necessarily concerned about how to interpret the incoming
audio packet payload type for properly playing back! 

Regards
Anand

> 
> Sure, why not? It would be very simple to realize if you solve the usual 
> security problems (or just ignore them :-).
> 
> On demand downloading of codecs is already done by e.g. RealPlayer. 
> 
> /P
> 
> 




From rem-conf Sun Mar 07 13:04:19 1999 
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Date: Sun, 07 Mar 1999 14:18:34 -0600
To: "ISIMADE7@Baden-Baden.Germany":;
From: Mahbubur Syed <msyed@badlands.nodak.edu>
Subject: ISIMADE'99 call for papers
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--=====================_920859514==_
Content-Type: text/plain; charset="us-ascii"

Dear Colleague:

Please find attached a call for paper. 

I am also attching a pdf version of the CFP.

Please let me know if you or some one from your organisation would be
willing to submit a paper.
 
It will be highly appreciated, if you take the trouble also to circulate
among interested faculty and your network.

My apologies if you received this email more than once.

Best Regards.

Thanks.


Syed M Rahman



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


INTERNATIONAL SYMPOSIUM ON INTELLIGENT MULTIMEDIA AND DISTANCE EDUCATION

In Conjunction with
The 11th International Conference on Systems Research, Informatics and
Cybernetics

2-7 August 1999 in Baden-Baden, Germany 

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

More Details at URL: 
http://venus.ee.ndsu.nodak.edu/ee/research/conferences/isimade/
==========================================

Symposium Co-Chairs
Dr. Mahbubur Rahman Syed, North Dakota State University, USA
Dr. Orlando R. Baiocchi, North Dakota State University, USA

Conference Chair
Dr. G. E. Lasker I.I.A.S, Scool of Computer Science, University of
Windsor,
canada


=======  IMPORTANT DATES ======

Extended abstract/full paper to symposium chair		    25 April 1999

Notification of acceptance		                    12 May 1999

Receipt of camera-ready papers by the conference chair      1 June 1999


======  MAIN TOPICS =======

STREAMS: MULTIMEDIA stream and DISTANCE EDUCATION stream.

Topics include, but not limited to, following.


In Multimedia Stream:

- Internet Applications
- Applications in visualization, human computer interactions 
- Applications in virtual environments
- Industrial, medical and other applications
- Multimedia in electronic commerce applications
- Video, audio & image  processing and retrieval
- Vision and Image Processing
- Image data structures and databases
- video, audio and image compression techniques
- Neural network and AI applications
- Next generation applications
- Multimedia communications and databases
- Multimedia standards
- Intelligent multimedia
- Document image understanding
- Animation
- Tools for multimedia production and services
- Networked Multimedia
- Digital Signal Processing
- Distributed Intelligent Systems
- Multimedia and Human Computer Interaction
- Integration of MM Information
- Interactive MM
- Hypermedia
- Representation of MM Information
- Integration of Graphics & Vision
- Synchronization of MM Information
- MM and Virtual Reality
- Education and Applications of MM
- Cooperative Information Systems

In Distance Education (DE) Stream:

- DE delivery, experience and tools development at secondary schools, 
  universities, business and industries and in all academic areas such
  as computer science, arts, engineering, business, medicine etc.
- DE, the Internet and Internet-based visualization
- Using computer graphics; visualization for education; video- and 
  Multimedia-based Learning
- Cognitive aspects of visual teaching and learning
- New Educational paradigms such as interactive classrooms,
tele-immersion,
  networked tele-collaboration and WebTV etc.
- Internet-based visualization
- Hardware and software tools for educational applications
- Instructional Management, student assessment, student support services
- Virtual Education environment; Virtual university - concepts and
futures
- Needs Assessments and Perspectives including National Policies and
Regional Programs
- Organizational Models of Distance Education Institutions and case
studies


======  SUBMISSIONS  =======

Submit electronic version (Word, postscript or rtf format) of the
original 
research work, not submitted elsewhere, as an extended abstract
(limited to
4 pages) or full paper (not exceeding 15 double-spaced pages) to the 
following email address:

msyed@badlands.nodak.edu

Alternatively, submit three copies in hard copy form to the symposium
Co-chair 
at the following address.

Dr. Mahbubur Rahman Syed
Electrical Engineering Department, EERoom 101
North Dakota State University
Fargo, ND 58105, USA

Email: msyed@badlands.nodak.edu
Phone: (701) 231 7689 
Fax:   (701) 231 8677

A cover sheet should include address, phone/fax numbers, email
address of the 
corresponding author.

All papers must be in English. All submitted papers will be refereed. 

Authors should indicate, at the time of submission, their preferences to 
present their papers either in the oral or poster sessions as
detailed below.


SESSIONS:

There will be oral sessions and poster sessions for presentation of
papers in 
the symposium. Authors should indicate their preferred mode of
presentation
at the time of submission of their papers. The paper review and
acceptance
process will be same for both type of presentation options and all
accepted 
papers will be published in the proceedings. At least one of the
authors must 
present their paper in the accepted presentation mode. However, poster 
session authors may arrange for their papers to be presented by a
symposium 
attendee. Authors of accepted papers for the poster presentation must
submit 
their posters along with the camera-ready papers to the conference
chair. 
The authors who can not attend the symposium due to special
circumstances 
must mention at the time of submission and may be considered for poster 
presentations only.

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


--=====================_920859514==_
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Content-Transfer-Encoding: base64
Content-Disposition: attachment; filename="ISIMADE99.pdf"

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--=====================_920859514==_
Content-Type: text/plain; charset="us-ascii"


********************************
Dr. Mahbubur Rahman Syed
Department of Electrical Engineering, EERoom 101
North Dakota State University
Fargo, ND 58105, USA

Email: msyed@badlands.nodak.edu
Phone: (701) 231 7689 
Fax:   (701) 231 8677
******************************** 
--=====================_920859514==_--




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From: pierre de saqui sannes DFR/MI <pdss@ensica.fr>
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Subject: IDMS99: extended deadline March 21st, 1999
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=======>
=======> IDMS99	extended deadline ===> March 21st, 1999 <===
=======>

			C a l l    f o r    p a p e r

				  IDMS'99
		  6th International Workshop on Interactive
		     Distributed Multimedia Systems and 
		         Telecommunication Services

		    12-15 October 1999, Toulouse, France


                                Organized by
                            LAAS-CNRS and ENSICA
                         http://www.ensica.fr/idms99
                          email : idms99@ensica.fr

The Sixth International Workshop on Interactive Distributed Multimedia 
Systems and Telecommunication Services will be held in 1999 at ENSICA, 
Toulouse, France. The purpose of this workshop is to bring together 
researchers, developers, and practitioners from academia and industry. 
The workshop serves as a forum for discussion, presentation, and 
exploration of technologies and their advances in the broad field of 
interactive distributed multimedia systems and telecommunication 
services - ranging from basic system technologies such as networking 
and operating system support to all kinds of teleservices and distributed 
multimedia applications. Case studies and papers describing experimental 
work are especially welcome. Relevant topics include, but are not 
limited to: 

* High-speed/ATM networks 
* Mobile multimedia systems 
* Multimedia over satellite 
* Multimedia middleware 
* Quality of service issues 
* Media scaling 
* Resource management 
* Protocol design and implementation 
* Development tools for distributed multimedia applications 
* Distributed multimedia database systems
* Multimedia-specific intelligent agents 
* Computer supported collaborative work 
* Distributed virtual reality systems 
* Distance education 
* Conferencing 
* Digital libraries 
* Interactive television 
* Video-on-demand systems 
* Compression algorithms 


IDMS'99 will consist of a three day technical program, a full day of 
tutorials, and demonstrations during the workshop. In order to keep the 
flavor of a workshop, the number of participants will be restricted. 
Furthermore, we encourage contributions in form of full papers and 
position papers. Full papers are expected to describe innovative and 
significant work. The purpose of position papers is to provide a seed 
for debate and discussion. Position papers enable researchers to 
present exciting ongoing work in early stages, suggestions for 
future directions, and concerns about current developments. Both types 
of papers will be reviewed by the program committee and printed in the 
workshop proceedings edited by Springer Verlag in the Lecture Notes in 
Computer Science series.


Information for authors: Authors are invited to submit full papers and 
position papers for review. Submitted manuscripts must describe original 
work (not submitted or published elsewhere). Full papers must not be 
longer than 20 double spaced pages and position papers must not be longer 
than 8 double spaced pages. Both types of papers should contain an abstract 
of approximately 300 words, and include title, authors and affiliations. 
The submission process of papers will be handled electronically. Detailed 
description of the electronic submission procedures is available on the 
IDMS'99 web page: http://www.ensica.fr/idms99/. Authors without web 
access may email to owe@laas.fr for requesting electronic submission 
information. Authors unable to submit electronically are invited to 
send 5 copies of their contribution to Dr Philippe Owezarski.

Important Dates
Submission due:			March 21, 1999		NEW !!!!!!!
Notification acceptance:	April 30, 1999
Camera ready version:		June 14, 1999
Workshop:			October 12 - 15, 1999


Co-chairs: Michel Diaz, Philippe Owezarski, Patrick Sénac

LAAS-CNRS, 7 Avenue du Colonel Roche, 31077 Toulouse cedex 4, France
e-mail: {diaz, owe}@laas.fr, Phone: +33 5 61 33 63 17, Fax: +33 5 61 33 64 11

ENSICA, 1 Place Emile Blouin, 31056 Toulouse cedex, France
e-mail: senac@ensica.fr, Phone: +33 5 61 61 86 81, Fax: +33 5 61 61 86 86



Program Committee Chairmen:		Michel Diaz, Philippe Owezarski
Organization Chairman:			Patrick Sénac
Tutorial Chairman: 			Laurent Dairaine
Publicity and Demonstration Chairman: 	Pierre de Saqui-Sannes
Conference electronic support Chairman:	Marc Boyer

Program Committee

H. Affifi, ENST Bretagne, France
P.D. Amer, U. Delaware, USA
E. Biersack, Institut Eurécom, France
G. v. Bochmann, U. Ottawa, Canada
B. Butscher, GMD FOKUS, Germany
A.T. Campbell, Columbia U., USA
T.S. Chua, U. Singapore, SG
J.-P. Courtiat, LAAS-CNRS, France
J. Crowcroft, UCL, UK
L. Delgrossi, U. Piacenza, Italy
C. Diot, Sprint ATL, USA
R. Dssouli, U. Montréal, Canada
M. Dudet, CNET France Télécom, France
W. Effelsberg, U. Mannheim, Germany
F. Eliassen, U. Oslo, Norway
S. Fdida, LIP6, France
D. Ferrari, U. Piacenza, Italy
V. Goebel, UNIK, Norway
T. Helbig, Philips, Germany
J.-P. Hubaux, EPFL, Switzerland
D. Hutchison, Lancaster U., UK
W. Kalfa, TU Chemnitz, Germany
T.D.C. Little, Boston U., USA
E. Moeller, GMD FOKUS, Germany
K. Nahrstedt, U. Illinois, USA
G. Neufeld, UBC Vancouver, Canada
B. Pehrson, KTH Stockholm, Sweden
T. Plagemann, UNIK, Norway
B. Plattner, ETH Zurich, Switzerland
L.A. Rowe, U. California Berkeley, USA
H. Scholten, U. Twente, Netherlands
A. Seneviratne, UTS, Australia
R. Steinmetz, GMD, Germany
J.B. Stéfani, CNET France Télécom, France
L. Wolf, TU Darmstadt, Germany
M. Zitterbart, TU Braunschweig, Germany
Pierre de Saqui-Sannes		pdss@ensica.fr
ENSICA				http://www.ensica.fr/~pdss
1 Place Emile Blouin		tel. +33 5 61.61.86.81
31056 Toulouse Cedex 5		fax. +33 5 61.61.86.86
	



From rem-conf Mon Mar 08 09:41:57 1999 
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To: "Marketing Ireland" <marketing@irelandconferences.com>
Subject: Fw: 30% Higher Attendance at your Conference
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Subject: 30% Higher Attendance at your Conference


>Please forgive our interruption but this will help you.
>
>
>We have a new Web Site
>
>http://www.irelandconferences.com
>
>You are a busy professional so we have geared ourselves to be of the most
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>If you do not want to receive further information please write REMOVE and
return this email.




From rem-conf Mon Mar 08 09:46:08 1999 
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To: rem-conf@es.net
Subject: REMINDER: AVT agenda items due
Date: Mon, 08 Mar 1999 17:42:03 +0000
Message-ID: <4043.920914923@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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Just a reminder that requests for agenda time in AVT should be sent to
Steve and I as soon as possible. We hope to have a draft agenda out
tomorrow.

Colin



From rem-conf Mon Mar 08 12:24:54 1999 
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Subject: Question: Interleaving
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Dear all,

i am a newbie to avt and would like to know whether there are publications/research
on interleaving of audio packets to reduce the effect of burst loss, especially for
frame-based codecs (J. Rosenberg stated in "G.729 Error Recovery" that frame-based codecs
are quite sensitive against burst loss).

Thank you,
-- long



From rem-conf Tue Mar 09 05:44:23 1999 
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I notice most of the RTP mux drafts are
	expired or expiring. Does this mean we can
	forget all about this (good).

[\] Robert Wood   

The St. Lawrence river - fresh, warm, visible diving.

mailto:robert_wood@mitel.com http://www.magma.ca/~rgwood/



From rem-conf Tue Mar 09 07:08:04 1999 
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Subject: Call for Papers (IEEE Network Magazine -- Special Issue on Multicasting)
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Please excuse me if you receive multiple copies of this announcement.
Sorry to bother you if you shouldn't have received this e-mail.

- Sanjoy Paul



From rem-conf Tue Mar 09 07:48:21 1999 
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Subject: Forgot the attachment for Call for Papers -- Here it is!
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This is a multi-part message in MIME format.
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--------------20DE0035C4B94C2E44EDD670
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IEEE Network Magazine - Special Issue on Multicasting 

Guest Editor:

Sanjoy Paul
Room 4G-532, Bell Laboratories
101 Crawfords Corner Road
Holmdel, NJ 07733
Phone: (732) 949-1637
Fax: (732) 949-9118
WWW: http://www.bell-labs.com/user/sanjoy
email: sanjoy@research.bell-labs.com

Scope:

Multicasting is a fundamental communication paradigm for 
supporting one-to-many communications. There are several 
key applications, such as, real-time distribution of market 
data, website replication, push-based applications, 
distributed database synchronization, multimedia streaming, 
multi-party conferencing, distributed interactive 
simulations (DIS), distributed multi-party games etc. which 
cannot be supported in a scalable way on the Internet without 
multicasting. This special issue covers the technology needed 
to make multi-party communication a reality and the challenges 
that need to be addressed.

Topics of interest for technical papers include but are not 
limited to:

	Multicast routing algorithms and protocols
	Multicast transport protocols
	Multicast applications
	Multicast Traffic Engineering
	Quality-of-service support for multicast communication
	Mobile multicast
	Charging and billing for multicast 
	Novel architectures to support multicast 
	Role of satellites in multicast
	Novel applications 
	Multicast security and privacy issues

We are especially interested in research papers that describe 
real systems and experiments and also good tutorial papers on 
any one or more of the above-mentioned topics. Work-in-progress 
papers describing the state of on-going research projects in 
multicasting are also encouraged. 

Research papers should demonstrate the feasibility of the approach 
and describe the state of realization. Case studies and applied 
papers should discuss the key factors that made the system work 
and should also mention the problems encountered and their potential 
solutions. Tutorial papers must be based on a detailed survey of the 
literature and written in a simple-to-understand manner.

Submission, Approval, Review, and Acceptance:

Authors are requested to send e-mail to the guest-editor, Sanjoy Paul
(sanjoy@research.bell-labs.com), giving a URL from where a postscript 
or PDF file can be downloaded by the guest-editor.

Upon approval by the guest-editor, all feature articles will then 
undergo a technical peer review consistent with other archival 
publications. Potential authors may wish to consult the 
sections on author information and guidelines for reviewers, which 
are given at http://www.comsoc.org/socstr/techcom/ntwrk.

More details on the submission guidelines can be found in
http://www.computer.org/internet/edguide.htm and
http://www.comsoc.org/socstr/techcom/ntwrk

Schedule:

Call for Papers: March 1, 1999
Submission Deadline: July 1, 1999
Acceptance Notification: September 1, 1999
Final Manuscripts Due: October 1, 1999
Publication of Completed Special Issue: January/February 2000



--------------20DE0035C4B94C2E44EDD670--




From rem-conf Tue Mar 09 10:10:36 1999 
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From: Florissa Colina <florissac@bmrc.berkeley.edu>
Subject: 3/10 Berkeley Multimedia, Interfaces and Graphics Seminar
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Berkeley Multimedia, Interfaces and Graphics Seminar

Technology and Accessibility 

Wednesday, March 10, 1999, 1:00-2:30 p.m. 405 Soda Hall 

Gary Robson 
VITAC 

As new technologies have invaded everyday life, they have created cycles of
increased and decreased accessibility for people with hearing and vision
impairments. This seminar focuses on multimedia and deafness, and where we
are on the curve now. I'll discuss specific accessibility tools and
technologies, and point out resources for those who want to explore the
subject in greater depth.  

The seminar is broadcast on the Internet Mbone. The addresses are video
224.2.163.7/57076 and audio 224.2.147.61/27202. 

Sponsored by the Berkeley Multimedia Research Center 


____________________________________________

Florissa Colina
Berkeley Multimedia Research Center (BMRC)
http://bmrc.berkeley.edu/
626 Soda Hall #1776
University of California
Berkeley, CA 94720-1776
Phone: (510) 643-0800   FAX: (510) 642-5615  
Email: florissac@bmrc.berkeley.edu



From rem-conf Tue Mar 09 11:47:15 1999 
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To: rem-conf@es.net
Subject: DRAFT AVT agenda for Minneapolis
cc: casner@cisco.com
Date: Tue, 09 Mar 1999 19:40:47 +0000
Message-ID: <2613.921008447@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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Enclosed is a draft agenda for the forthcoming AVT meeting in Minneapolis.
Please send comments/corrections to Steve and I as soon as possible.
Colin





                         Audio/Video Transport WG
	                                   
                               A G E N D A
 

Wednesday, March 17 at 15:30-17:30
==================================

- Introduction

- Documents published since the last meeting
	RFC2508 - IP/UDP/RTP header compresssion

- Update to RTP
	draft-ietf-avt-rtp-new-03.txt			(Casner)
	draft-ietf-avt-rtpsample-02.txt			  		  *
	draft-ietf-avt-rtcptest-00.txt			(Rosenberg)	15

- Update to RTP audio/video profile
	draft-ietf-avt-profile-new-05.txt		(Casner)
	draft-hoschka-rtp-mime-00.txt
	draft-ietf-avt-rtp-format-guidelines-01.txt     		  *

- RTP MIB 
	draft-ietf-avt-rtp-mib-04.txt			(Strahm)	10*

- A More Loss-Tolerant RTP Payload Format for MP3 Audio
	draft-finlayson-rtp-mp3-00.txt			(Finlayson) 	10

- RTP Payload Format for DV Format Video
	draft-kobayashi-dv-video-00.txt			(Kobayashi)	10

- RTP Payload Format for Generic FEC		
	draft-ietf-avt-fec-05.txt			(Rosenberg)	 5*

- RTP Payload Format for Interleaved Media
	draft-ietf-avt-interleaving-01.txt		(Perkins)	 5

- RTP Payload Format for PureVoice (QCELP) audio			 5
	draft-mckay-qcelp-02.txt

- RTCP location reports
	draft-crowcroft-rtcp-latlong-00.txt		(Crowcroft)	10


Thursday, March 18 at 15:30-17:30
=================================

- RTP Payload Format for MPEG-4 Streams					15
	 draft-ietf-avt-rtp-mpeg4-01.txt		(Civanlar)

- RTP Multiplexing							30

Items marked with a * are considered ready for last call.




From rem-conf Tue Mar 09 12:15:51 1999 
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Date: Tue, 9 Mar 1999 15:10:24 -0500 (EST)
From: Chinmay Padhye <padhye@eng.usf.edu>
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Subject: Voice Traces.
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Hello:

	I am computer science grad student at the University Of South
Florida. My thesis area deals with transporting voice over IP. I had a
couple of questions regarding research in this field.

1)	Is there any standard set of traces that can be used for
simulation. I think a standard set of traces will help when comparing two
algorithms. 

2)	Is there any site where I can download any voice traces.

3)	What is the tolerable end to end delay limit in interactive voice
conversations. I have seen different tolerance limits taken in the
literature. What is the general acceptable limit. 

	It will be a great help if someone can let me have a few traces
for my simulation work. 


Thanks in advance,

Yours Truly,

Chinmay V. P.
University Of South Florida.




From rem-conf Wed Mar 10 07:21:50 1999 
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http://www.patents.ibm.com/patlist?icnt=US&patent_number=5870412&x=37&y=9

On cursory inspection, this may be directly related to some of the
redundancy and FEC work in rem-conf.
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



From rem-conf Wed Mar 10 08:14:11 1999 
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Subject: Re: US Patent 5870412: Forward error correction system for packet 
         based real time media
In-reply-to: Your message of "Wed, 10 Mar 1999 10:07:43 EST." <36E68ABF.D9CFA5A4@cs.columbia.edu>
Date: Wed, 10 Mar 1999 16:06:44 +0100
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From: Orion Hodson <O.Hodson@cs.ucl.ac.uk>
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<36E68ABF.D9CFA5A4@cs.columbia.edu>Henning Schulzrinne writes:
> http://www.patents.ibm.com/patlist?icnt=US&patent_number=5870412&x=37&y=9
> 
> On cursory inspection, this may be directly related to some of the
> redundancy and FEC work in rem-conf.

Prior art ?

Audio-Video Transport Working Group                            D. Budge
INTERNET-DRAFT                                              R. McKenzie
                                                               W. Mills
                                                                W. Diss
                                                                P. Long
                                             Smith Micro Software, Inc.
                                                               May 1997
                                               Expires: December 4 1997


             Media-independent Error Correction using RTP
               draft-budge-media-error-correction-00.txt

- Orion



From rem-conf Wed Mar 10 08:15:33 1999 
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        Bulent Yener <yener@research.bell-labs.com>
Subject: Call For Papers  IEEE Network Magazine
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Please excuse me if you receive multiple copies of this announcement
and/or if you shouldn't have received it at all.

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

               Call For Papers 
             
               IEEE Network Magazine

             Special Issue on Network Security

      Guest Editors:

  Bulent Yener                                Patrick Dowd 
  Bell Labs, Lucent                           Laboratory for
  Technologies                                Telecommunications
  Murray Hill, NJ 07974                       Sciences
  Tel: +1 908 582 7087                        United States Department
  Fax: +1 908 582 1239                        of Defense
  Email: yener@research.bell-labs.com         Ft. Meade MD 20755-6512
                                              Tel: +1 301 688 0347
                                              Fax: +1 301 688 0588 
                                              Email: p.dowd@ieee.org 

       Scope: 

  Network and Internet security has become a  crucial requirement for both
users and service providers. The Internet is a commercial infrastructure
where sensitive and confidential personal and business data are carried
over public networks. Although security is often treated as an
after-thought, this attitude is changing. Security within an application
needs to be considered as a fundamental element of the application,
treated analogously to Quality of Service (QoS) considerations.
 Security is often viewed as a one-size-fits-all paradigm, but this is
difficult to sustain due to the eclectic collection of communications
mediums that compose the Internet infrastructure. The danger of a
cookie-cutter strategy is that security will contend with performance
since it is not suited to the environment. As the QoS
requirements of applications and the physical layer properties
internetworking become more diverse, agile but robust  and consistent
security solutions are needed. This is difficult, since custom solutions
typically have difficulty surviving in a mass market, yet flexibility is
needed for security use to become ubiquitous. 
 
 We are interested in tutorial-oriented research papers that describe real 
services, software systems and experiments. Work-in-progress papers
describing the state of on-going research projects in Internet security
are encouraged. Research papers should demonstrate the feasibility of the
approach and describe the state of realization. Case studies and applied
papers should discuss the key factors that made  the system work and
should also mention the pitfalls and problems encountered and how they may
be overcome.


     Topics of interest include:

                      Intrusion detection 
                      Authentication 
                      Mobile code and agent security 
                      Privacy and anonymity 
                      Key management 
                      Access control and Firewalls 
                      Wireless, mobile network security 
                      Secure multicasting 
                      Data integrity 
                      Security verification 
                      Security protocols 
                      Policy modeling 
                      Commercial security 
                      Electronic commerce 
                      Security management 


If you are unsure if your work falls within the scope of this special
issue, please send an abstract to one of the guest editors. We would be
happy to review  it and provide feedback.


                       Manuscript Submission

 This special issue will only consider electronic submissions in the
format of postscript, PDF, or MS WORD. To submit a paper for
consideration, authors  should 
1.Send your paper to one of the guest editors via email. The paper should
be included as an email attachment, or the author may provide a URL where
the file can be downloaded. 
2.Indicate which author is to serve as the primary correspondence contact. 
3.Provide a contact list for all of the other authors. Please list
affiliations, mailing addresses, phone/fax numbers, and email addresses. 

We will provide an acknowledgement of receipt of the paper within 24 hours
of submission. 


                     Schedule:

                     Paper Submission Deadline: June 1, 1999
                     Feedback to Authors: August 1, 1999
                     Final Manuscripts to Publisher: September 1, 1999
                     Publication of Special Issue: Nov/Dec 1999
                            

_____________________________________________________ 
Bulent Yener
Information Sciences Research Center
Bell Laboratories, Lucent Technologies
600 Mountain Avenue,  Room 2T-314
Murray Hill, NJ 07974

email: yener@research.bell-labs.com
phone: (908) 582 7087
fax:   (908) 582 1239
_____________________________________________________ 




From rem-conf Wed Mar 10 08:38:03 1999 
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From: "Strahm, Bill" <bill.strahm@intel.com>
To: "'rem-conf@es.net'" <rem-conf@es.net>
Subject: RE: US Patent 5870412: Forward error correction system for packet
	  based real time media
Date: Wed, 10 Mar 1999 08:33:03 -0800
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Barely prior art <G>...

One thing I am wondering about, do we think the IETF should start putting
dates of first drafts on documents.  For example the patent mention was
awarded this year, but its date of filing was Dec 12 97.  This document was
published as an RFC Dec 4 1997, however I bet there were drafts around for a
year or more that would help move the prior art date back quite a bit.
There is no record of this available as far as I know (Drafts expire you
know)

Bill

-----Original Message-----
From: Orion Hodson [mailto:O.Hodson@cs.ucl.ac.uk]
Sent: Wednesday, March 10, 1999 7:07 AM
To: Henning Schulzrinne
Cc: rem-conf@es.net
Subject: Re: US Patent 5870412: Forward error correction system for
packet based real time media


<36E68ABF.D9CFA5A4@cs.columbia.edu>Henning Schulzrinne writes:
> http://www.patents.ibm.com/patlist?icnt=US&patent_number=5870412&x=37&y=9
> 
> On cursory inspection, this may be directly related to some of the
> redundancy and FEC work in rem-conf.

Prior art ?

Audio-Video Transport Working Group                            D. Budge
INTERNET-DRAFT                                              R. McKenzie
                                                               W. Mills
                                                                W. Diss
                                                                P. Long
                                             Smith Micro Software, Inc.
                                                               May 1997
                                               Expires: December 4 1997


             Media-independent Error Correction using RTP
               draft-budge-media-error-correction-00.txt

- Orion



From rem-conf Wed Mar 10 09:53:40 1999 
From rem-conf-request@es.net Wed Mar 10 09:53:39 1999
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Date: Wed, 10 Mar 1999 18:50:38 +0200
To: rem-conf@es.net
From: wolf@informatik.uni-ulm.de (Klaus H. Wolf)
Subject: RE: US Patent 5870412: Forward error correction system for packet based real time
 media
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I know of a publication from March '93. This paper calls the recent IBM
'invention' the state of the art and proposes higher order FEC codes.

-Klaus H. Wolf

----
Distributed Systems Dept.                    voice: +49 731 502 4145
University of Ulm                        ethernet: 08:00:20:12:2a:01
89069 Ulm                             Cobrow: http://www.cobrow.com/
Germany           Live: http://www.cobrow.com/pages/people/wolf.html





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From rem-conf Thu Mar 11 02:13:19 1999 
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        "winsock2 mailing list" <winsock-2@mailbag.cps.intel.com>
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Hello all

If anyone has/is/plans on working with elemedia's RTP implementation =
please contact me so we can cooperate. I am testing it and i have a lot =
f problems.

Thanks in advance

Nikos Laoutaris
Network Operations Center
National and Kapodestrian University of Athens, Greece
e-mail : laoutaris@noc.uoa.gr

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<META content=3D'"MSHTML 4.72.3110.7"' name=3DGENERATOR>
</HEAD>
<BODY bgColor=3D#c0c0c0>
<DIV><FONT size=3D2>Hello all</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>If anyone has/is/plans on working with elemedia's =
RTP=20
implementation please contact me so we can cooperate. I am testing it =
and i have=20
a lot f problems.</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>Thanks in advance</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT color=3D#000000 size=3D2>Nikos Laoutaris<BR>Network =
Operations=20
Center<BR>National and Kapodestrian University of Athens, =
Greece<BR>e-mail : <A=20
href=3D"mailto:laoutaris@noc.uoa.gr">laoutaris@noc.uoa.gr</A></FONT></DIV=
></BODY></HTML>

------=_NextPart_000_0017_01BE6BB7.8DD3AE80--




From rem-conf Thu Mar 11 09:15:53 1999 
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From: Florissa Colina <florissa@bmrc.berkeley.edu>
Subject: 3/17 Berkeley Multimedia, Interfaces and Graphics Seminar
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Berkeley Multimedia, Interfaces and Graphics Seminar


An Investigation into Web Site Design Practice


Wednesday March 17, 1999, 

1:00-2:30 p.m. 405 Soda Hall 


<italic>Please note the time of the seminar has changed!!

</italic>

Mark Newman

Computer Science Division - EECS 

University of California at Berkeley 


How can computer systems better support creative processes? Many
computer-based tools currently exist to support art and design tasks, but
the technology and the need exist to push these capabilities of these
tools farther. 


In the Group for User Interface Research at UC Berkeley, we are
undertaking a research project to understand how technology can be used
to further assist the web design process, with the ultimate goal of
developing tools to support designers. During the first phase of our
project, we conducted an informal study involving interviews with
professional designers and examination of works-in-progress to improve
our understanding of the web design practice in professional settings. 


In this talk we present the results of our initial investigation,
including observations about common and divergent design practices among
the organizations studied, and the existence of multiple design
specialties which confuse the definition of "web design". 


The seminar is broadcast on the Internet Mbone. The addresses are video
224.2.163.7/57076 and audio 224.2.147.61/27202. 


Sponsored by the Berkeley Multimedia Research Center 




____________________________________________


Florissa Colina

Berkeley Multimedia Research Center (BMRC)

http://bmrc.berkeley.edu/

626 Soda Hall #1776

University of California

Berkeley, CA 94720-1776

Phone: (510) 643-0800   FAX: (510) 642-5615  

Email: florissac@bmrc.berkeley.edu



From rem-conf Thu Mar 11 10:01:23 1999 
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Dear Colleagues,

I am looking for information about rfc 2508 for RTP/UDP/IP header 
compression over low capacity links   :

- Products that support this RFC (I have seen Cisco Routers so far).
- SW implementations available on the market or shareware.
- Performance measures of the algorithm in real scenarios.
- etc.

I would appreciate any information on this subject.

Thanks a lot,


Pedro Reviriego

 
Get Your Private, Free Email at http://www.hotmail.com



From rem-conf Fri Mar 12 16:07:24 1999 
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To: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
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Subject: Re: Reed Solomon Draft 
In-reply-to: Your message of Tue, 10 Nov 1998 14:01:19 EST.
             <36488D7F.F5F490C1@dnrc.bell-labs.com> 
From: yuzo@bmrc.berkeley.edu
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Dear All,

I have some comments on draft-ietf-avt-reedsolomon-00.txt.

1) What do you think about adding a flavor of interleaving?  If the
length of burst packet loss is proportional to bitrate, the protection
becomes weak at high bitrate.  Increasing of N is a way to strengthen
the protection, but complexity also increases.  Another way is
interleaving, which disperses the consecutive lost packets.  By
introducing a new parameter S, we easily lengthen a media block while
using small numbers of N and K.  The step of media packets is defined
as 2**S.  Thus a media block contains K media packets numbered
SN+(2**S)*i where 0<=i<K.  The behavior with S=0 is exactly the same
as the draft.

2) In the Reed-Solomon FEC header, representations of (K-1,N-K-1,i)
seems to be better than (N,K,i), because the ranges of N-K-1 and i are
same.

I recently joined AVT.  Please forgive me if these comments were
already discussed or are beyond the scope of the draft.

----
Yuzo Senda
Berkeley Multimedia Research Center
University of California, Berkeley
Berkeley, CA 94720-1776
Fax: 510-642-5615



From rem-conf Fri Mar 12 17:07:10 1999 
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From: "Dr. Injong Rhee" <rhee@eos.ncsu.edu>
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Date: Fri, 12 Mar 1999 19:42:26 -0500
In-Reply-To: Henning Schulzrinne <hgs@cs.columbia.edu>
        "US Patent 5870412: Forward error correction system for packet based real time media" (Mar 10, 10:07am)
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We have also recently put together
a techreport on FEC-based recovery scheme for interactive video  transmission
(http://www.csc.ncsu.edu/faculty/rhee/pub/fec2.ps.gz).
An extension on my sigcomm'98 paper.

The abstract for the paper (for your cursory reading pleasure
before you download the monster file -- it is ~1MB, sorry)
is as follows:

-----------------------------------
Real-time interactive video transmission in the current Internet has mediocre
quality because of high packet loss rates. Loss of packets belonging to a
video frame manifests itself not only in the reduced quality of that frame
but also in the propagation of that distortion to successive frames.
This error propagation problem is inherent in any motion-based
video codec because of the interdependence of encoded video frames.
Since packet losses in the best-effort Internet environment
cannot be prevented, minimizing the impact of these packet
losses to the final video quality is important. In this
paper, we present a new forward error correction (FEC) technique
that effectively alleviates error propagation in the transmission
of interactive video. The technique is based on our recently
developed error recovery scheme called Recovery from Error
Spread using Continuous Updates (RESCU).  RESCU allows transport
level recovery techniques previously known to be infeasible
for interactive video transmission applications
to be successfully used in such applications. The proposed FEC
technique can be
very useful when the feedback channel from the receiver
is highly limited, or transmission delay is high.
Both simulation and Internet experiments indicate that our proposed
FEC technique  effectively alleviates the error spread
problem and is able to sustain much better video quality
than H.261 or other conventional FEC schemes
under various packet loss rates.

Key words: Packet Loss Recovery, Interactive Video,
                 Error Propagation, Forward Error Correction,
                 Video conferencing.


On Mar 10, 10:07am, Henning Schulzrinne wrote:
> Subject: US Patent 5870412: Forward error correction system for packet bas
> http://www.patents.ibm.com/patlist?icnt=US&patent_number=5870412&x=37&y=9
>
> On cursory inspection, this may be directly related to some of the
> redundancy and FEC work in rem-conf.
> --
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
>
>-- End of excerpt from Henning Schulzrinne



-- 
Dr. Injong Rhee
Assistant Professor
North Carolina State University
Raleigh, NC 27695
Email: rhee@csc.ncsu.edu
Phone: 919-515-3305
Fax: 919-515-7925




From rem-conf Fri Mar 12 18:17:18 1999 
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From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
Organization: Bell Laboratories
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yuzo@bmrc.berkeley.edu wrote:
> 
> Dear All,
> 
> I have some comments on draft-ietf-avt-reedsolomon-00.txt.
> 
> 1) What do you think about adding a flavor of interleaving?  If the
> length of burst packet loss is proportional to bitrate, the protection
> becomes weak at high bitrate.  Increasing of N is a way to strengthen
> the protection, but complexity also increases.  Another way is
> interleaving, which disperses the consecutive lost packets.  By
> introducing a new parameter S, we easily lengthen a media block while
> using small numbers of N and K.  The step of media packets is defined
> as 2**S.  Thus a media block contains K media packets numbered
> SN+(2**S)*i where 0<=i<K.  The behavior with S=0 is exactly the same
> as the draft.

For interactive apps, this would move the latency for FEC from bad to
worse.  Of course, for streaming media, its not so bad. Its worth noting
that this kind of interleaving is possible in the parity draft, because
of the use of the bitmask to indicate the packets that are being
covered. I wonder if its perhaps not a bad idea to apply that concept
here as well. This would enable the interleaving Yuzo describes for the
Reed Solomon as a "freebie".

There is also a draft on interleaving of packets
(draft-ietf-avt-interleaving-01), but that is something different. It
interleaves the regular media packets, and has nothing to do with how
FEC is generated.



> 
> 2) In the Reed-Solomon FEC header, representations of (K-1,N-K-1,i)
> seems to be better than (N,K,i), because the ranges of N-K-1 and i are
> same.

Good idea. We could possibly get away with fewer bits this way.

-Jonathan R.


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



From rem-conf Sat Mar 13 01:53:01 1999 
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Date: 12/15/98 5:08:53 PM Pacific Daylight Time
From: breath@angelfire.com
Reply-To:breath@angelfire.com
To: breath@angelfire.com
Subject: Personal Alcohol Testers (Breathalyzers)
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Personal Alcohol Testers (Breathalyzers) 

We are pleased to introduce our line of Personal Alcohol Testers 
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in your subject heading. (Only serious inquiries)

Thank You








From rem-conf Sat Mar 13 06:29:41 1999 
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From: rhee@unity.ncsu.edu
Message-Id: <199903131424.JAA02287@cc00du.unity.ncsu.edu>
To: jdrosen@dnrc.bell-labs.com, rem-conf@es.net
Subject: Re: Reed Solomon Draft
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> For interactive apps, this would move the latency for FEC from bad to
> worse.  Of course, for streaming media, its not so bad. Its worth noting
> that this kind of interleaving is possible in the parity draft, because
> of the use of the bitmask to indicate the packets that are being
> covered. I wonder if its perhaps not a bad idea to apply that concept
> here as well. This would enable the interleaving Yuzo describes for the
> Reed Solomon as a "freebie".
> -- 
> Jonathan D. Rosenberg                       Lucent Technologies
> Member of Technical Staff                   101 Crawfords Corner Rd.
> High Speed Networks Research                Holmdel, NJ 07733
> FAX: (732) 834-5379                         Rm. 4C-526
> EMAIL: jdrosen@bell-labs.com
> URL: http://www.cs.columbia.edu/~jdrosen
> 

Well, believe it or not, even for interactive  video, 
interleaving (even over the period of 100 - 1000 ms) can be
very effective. And this does not introduce any playout latency.

It might be very hard to believe.... In our technical report
(http://www.csc.ncsu.edu/faculty/rhee/pub/fec2.ps.gz), we present  this
feasibility. The main idea is that repair FEC packets arriving after the
display of their (damaged) protecting frames are still useful to remove 
errors propagating from those damaged frames. We can use these "late"
packets to restore those damaged frames. Since they are used as 
reference frames for succeeding frames, restoring them removes 
error spread from them. By doing this, we can improve the received video
quality by 3-6dB over the conventional schemes. We performed experiments
using real Internet traces and MPEG-4 test video sequences.

The similar idea can be
applied to retransmission as well. We discuss the retransmission idea in
sigcomm98. You can find some more related papers from my web site. We are
currently working on making these schemes adaptive to network congestion
levels, loss rate, and latency. So, stay tuned..... 
As soon as we finish writing about it (hopefully within a week),
we will put together another techreport.

Injong

--
Dr. Injong Rhee
Assistant Professor
North Carolina State University
Raleigh, NC 27695
Home page: www.csc.ncsu.edu/faculty/rhee
Email: rhee@csc.ncsu.edu
Phone: 919-515-3305
Fax: 919-515-7925





From rem-conf Sun Mar 14 01:34:52 1999 
From rem-conf-request@es.net Sun Mar 14 01:34:51 1999
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Reply-To: "Yuzo Senda" <yuzo@bmrc.berkeley.edu>
From: "Yuzo Senda" <yuzo@mub.biglobe.ne.jp>
To: <rhee@unity.ncsu.edu>
Cc: <jdrosen@dnrc.bell-labs.com>, <rem-conf@es.net>
Subject: RE: Reed Solomon Draft
Date: Sun, 14 Mar 1999 01:24:38 -0800
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>Well, believe it or not, even for interactive  video,
>interleaving (even over the period of 100 - 1000 ms) can be
>very effective. And this does not introduce any playout latency.

I understand what you said.  That is a case of relatively low bitrate.
Current PCs have plenty of both processing power and memory to decode
multiple low-bitrate streams.  Since the Reed-Solomon stream is sent
separately, the interleaving I said does not affect playout latency
directly.  As you described, advanced decoders may utilize the FEC stream.

At high bitrate, the latency induced by the interleaving is negligible.  I
guess within a few years it will be physically possible to send a program at
1Mbps or more.

Yuzo





From rem-conf Sun Mar 14 03:55:41 1999 
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To: Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp>
cc: Christine Guillemot <Christine.Guillemot@irisa.fr>,
        Gerald Kuehne <kuehne@informatik.uni-mannheim.de>,
        4onIP-sys <4onIP-sys@fzi.de>, christ <christ@rus.uni-stuttgart.de>,
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Subject: Re: Ad hoc report + contribution on generic multiplexing 
In-reply-to: Your message of "Sat, 13 Mar 1999 21:18:38 JST."
             <01f201be6d4b$dbeefd00$17c77885@kaiji> 
Date: Sun, 14 Mar 1999 20:50:53 +0900
From: Jiro Katto <katto@dsp.cl.nec.co.jp>
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Dear all, 

I know most people are away from this reflector until next week, but
let me put my personal views. I am sorry if I missed important points
discussed on the reflector (especially of Cristine and Gerald). 

(1) Header duplication (re-synchronization)
I don't have any technical objections about where the header
duplication is specified, in HEC or in RTP payload format. But let me 
repeat again, when specified in HEC, it becomes non IP-specific. 

As Gerald mentioned, I also appreciate the RTP payload format for H261
(quite a great and pioneering standard). But, notice that it can be
applied to any error-prone networks (if they are referred to by other
network standards). So, I think the header duplication may be
specified in a coding standard, if available and avoidable. 

(2) Channel encoding (network adaptation)
I am not sure header duplication is a part of channel encoding. But,
generally speaking, I completely agree that the channel encoding
should be done by network standards (TransMux, in MPEG4 word). 

In my humble knowledge, e.g., error correction codes seem to be quite
differently specified according to TransMuxes (RTP, TS, H223, etc.).
So, I agree with you to let them remain in network standards.  

(3) Pre-encoding or real-time
I am not sure why the pre-encoding becomes a serious problem (although
your insight is quite interesting and impressive).  

It seems free for receivers to save the received streams in a format
with or without HECs as long as the stream syntax complies to MPEG4
standard. It seems free for servers to transcode them or insert/remove
HECs adaptively. Am I missing something ?

(4) Generic payload format
I am not yet sure what is a main purpose of the generic payload
format. Is it something like a container into which everybody can
specify payload formats of any coding standards out of RFC 1890 ?
Or something else ?

Anyway, I am quite impressed and educated by the discussion on this
reflector. Thank you very much for all and I hope both the meetings
(MPEG and IETF) will be successfull. 

Best regards,
--
Jiro Katto
NEC Corporation.

--------------------------------------------------------------------
Kikuchi-san wrote:
>
> Apologize for my late reply. I hope this not too late.
> 
> I add "mpeg4_robust" to the CC field because I think this issue is
> much relevant to the VIDEO error resiliency.
> 
> > > To rely only on the source encoder for resilience against packet
> > > losses, may not be sufficient.  Also, if this may work with
> > > real-time encoding when the encoder can adjust video packets,
> > > HEC field content to network conditions, how can this work with
> > > pre-encoded streams?  Shouldn't this type  of application be
> > > considered as well? 
> > >
> > > It seems to us that separating source encoding from 'channel
> > > encoding' (header duplication) would allow to provide protection
> > > (adaptive to network conditions) in both (real-time or non-real
> > > time encoding) scenarios.
> > >
> > > So, please, if there is something wrong with these technical
> > > concerns and technical arguments, we would be grateful to hear
> > > the TECHNICAL reasons.
> >
> > I did not read your contribution (M4469 Proposal for a normative ESI)
> > because I have no access to the MPEG-ftp yet (Could you send it to me
> > by e-mail?)). Maybe you covered in the document some items I am
> > going to mention in the following.
> >
> > I agree with you that it would be the best approach to split error
> > resilience information and header duplication from the compression
> > layer. 
> > That means, ideally we would use an RTP payload like the one for
> > H.261, where all necessary information for independent decoding
> > are given in the RTP header.
> >
> > However, to achieve this for MPEG-4 video ES poses several problems.
> > In contrast to H.261, MPEG-4 video uses enhanced predictive coding.
> > Consider for example the coding of the DC coefficients in intra
> > macroblocks as it is described in the FDIS:
> >
> >        B C D     A, B, C, D, X, Y = 8x8 blocks
> >        A X Y
> >
> > The DC coefficient in block X depends on a predictor calculated from
> > neighbouring blocks. Block Y depends on a predictor, too. This
> > predictor cannot be calculated from X's predictor. The predictor
> > concept is also used in AC prediction, motion vector coding and
> > shape coding. 
> 
> Yes.
> 
> > Therefore, if we want independently decodable units (e.g. a collection
> > of macroblocks), we have to provide all necessary predictors (and
> > there are many predictors) in an RTP header not to mention
> > duplicated header information.
> 
> Yes, this is correct. Is it realy good to have such BIG fields in an RTP
> header?
> 
> > It was my understanding that these advanced predictive modes were
> > notallowed (or switched off) when we were using the error
> > resilient coding modes, is that correct?
> 
> No. Please read the ISO14496-2 carefully.
> 
> >So, can't we imagine that an application (or the application producer/
> >designer) knows that its streams are to be transported on networks
> >with non guaranteed QoS. In this case the error resilient modes should
> >be used, and the advanced prediction modes switched off, right?
> >My point in the previous messages was to say that the nice features
> >of error resilient modes, incorporated in the spec., may be in certain
> >network conditions not sufficient.   Complementary - and definitely
> >not conflicting - mechanisms, know in the litterature under the
> >name of 'error control' may be necessary.  But these error control
> >mechanisms are beyond the scope of a source encoder, this is
> >the scope of a network adaptation layer or of a channel encoder.
> 
> > In this context the video packet concept is a useful compromise - all
> > predictively encoded data is confined within the packet. However, you
> > are right, it mixes error resilience with compression aspects.
> 
> >The concept of a video packet is nice, especially the idea of having
> >sync makersevery M bits instead of every M macroblocks. Only the
> >HEC field incorporates some 'channel coding' in the sense of
> >redundant or duplicated data.
> >This is ok when having real-time encoding, but may not be the most
> >efficient way in the case of off-line encoding :-(.
> 
> I cannot understand why HEC field is a 'channel coding'.
> 
> Just an addition of the duplicated information of the VOP header to the
> video bitstream generated without aware of the video packet boundary
> does NOT help the decoder at all because there are lots of prediction
> signals needed for video decoding, as explained by Gerald.
> 
> That's why the duplication tool is treated in conjunction with Video
> coding in MPEG-4. This may be an different manner from the existing
> standards, say H.261, but the video coding scheme itself is different.
> 
> I hope that when defining a RTP format, this should really be a media
> aware manner. I'm not saying that duplicate information should not be
> place in RTP header. But,  just a duplication of headers without aware
> of video coding will not help anything.
> 
> 
> Regards,
> Yoshihiro
> 
> ----
>         Yoshihiro Kikuchi
> 
> E-mail: kiku@eel.rdc.toshiba.co.jp
> Research and Development Center     Toshiba Corp.
> TEL: +81 +44 549 2151    FAX: +81 +44 520 1308




From rem-conf Sun Mar 14 10:49:23 1999 
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From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
Organization: Bell Laboratories
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rhee@unity.ncsu.edu wrote:
> 
> It might be very hard to believe.... In our technical report
> (http://www.csc.ncsu.edu/faculty/rhee/pub/fec2.ps.gz), we present  this
> feasibility. The main idea is that repair FEC packets arriving after the
> display of their (damaged) protecting frames are still useful to remove
> errors propagating from those damaged frames. We can use these "late"
> packets to restore those damaged frames. Since they are used as
> reference frames for succeeding frames, restoring them removes
> error spread from them. By doing this, we can improve the received video
> quality by 3-6dB over the conventional schemes. We performed experiments
> using real Internet traces and MPEG-4 test video sequences.

Neat idea. The benefit will depend a lot on how good the codec itself is
at recovering "naturally". For example, a video sequence with mostly
intra macroblocks probably won't benefit much from your approach.
There's a cost to this approach too - a significant increase in decoder
complexity, since you will need to re-decode the old frames with the
updated information. Have you done any studies with speech codecs? I
have a report on error recovery for G.729. It basically concluded that
error propagation due to encoder/decoder de-synchronization was more
significant than the loss of a frame itself, based on both SNR and
subjective evaluation. This would be an argument for your approach for
speech. You can find a copy of the report at:

http://www.cs.columbia.edu/~jdrosen/e6880/index.html



-Jonathan R.


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



From rem-conf Sun Mar 14 11:07:12 1999 
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        "Re: Reed Solomon Draft" (Mar 14,  1:42pm)
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On Mar 14,  1:42pm, Jonathan Rosenberg wrote:
> Subject: Re: Reed Solomon Draft
> rhee@unity.ncsu.edu wrote:
> >
> > It might be very hard to believe.... In our technical report
> > (http://www.csc.ncsu.edu/faculty/rhee/pub/fec2.ps.gz), we present  this
> > feasibility. The main idea is that repair FEC packets arriving after the
> > display of their (damaged) protecting frames are still useful to remove
> > errors propagating from those damaged frames. We can use these "late"
> > packets to restore those damaged frames. Since they are used as
> > reference frames for succeeding frames, restoring them removes
> > error spread from them. By doing this, we can improve the received video
> > quality by 3-6dB over the conventional schemes. We performed experiments
> > using real Internet traces and MPEG-4 test video sequences.
>
> Neat idea. The benefit will depend a lot on how good the codec itself is
> at recovering "naturally". For example, a video sequence with mostly
> intra macroblocks probably won't benefit much from your approach.

You are absolutely right.
 If the  input video contains fast mocing objects or complete
scene change, my RESCU technique does not work well because motion
estimation and compensation is not simply effective in this case.

However, in a telephony or conferencing case, this would pay off big time.
In general, when the video sequence is ameanable to motion estimation,
RESCU can work well.


> There's a cost to this approach too - a significant increase in decoder
> complexity, since you will need to re-decode the old frames with the
> updated information.

Yes. However, increase in the complexity of decoder part may not
be so significant because it has to decode only the missing portion
by keeping the old frames in the buffer.


> Have you done any studies with speech codecs? I
> have a report on error recovery for G.729. It basically concluded that
> error propagation due to encoder/decoder de-synchronization was more
> significant than the loss of a frame itself, based on both SNR and
> subjective evaluation. This would be an argument for your approach for
> speech. You can find a copy of the report at:
>
> http://www.cs.columbia.edu/~jdrosen/e6880/index.html

That's interesting. I always thought it would be difficult to apply the idea
to speech. I will certianly look into it.

Thanks for pointing that out to me.

Injong


-- 
Injong Rhee
Assistant Professor
North Carolina State University
Raleigh, NC 27695
Home page: http://www.csc.ncsu.edu/faculty/rhee
Email: rhee@csc.ncsu.edu
Phone: 919-515-3305
Fax: 919-515-7925




From rem-conf Sun Mar 14 23:41:12 1999 
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I don't consider forgetting about RTP mux as 'good' from the following
reason:

Ignoring standardisation for RTP muxing (mainly for purposes of reducing
waste
of bandwidth) will not avoid non-standardized implementations by different
vendors (since efficient bandwidth requirements are requested by our
customers).

Ignoring the issue will not affect the existing demand for it (demand that
is not
going to be reduced - to say the least (at least from my point of view)).

I believe that numerous vendor-specific (most likely not interoperable)
solutions
would be worse than having one standardised solution. A standardised spec
will
result in a more stable and network-friendly solution, than most of the
initiatives
by 3rd party vendors will. As a side-kick a standardised spec will allow
interoperability which is always a good thing to have.

-- Gal.

=================================
        Gal Ashour
     ashour@il.ibm.com

  Audio/Video Technologies Group
   IBM HRL - Haifa Research Lab
=================================


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

Robert Wood wrote:

I notice most of the RTP mux drafts are
     expired or expiring. Does this mean we can
     forget all about this (good).

[\] Robert Wood

The St. Lawrence river - fresh, warm, visible diving.

mailto:robert_wood@mitel.com http://www.magma.ca/~rgwood/






From rem-conf Mon Mar 15 09:56:04 1999 
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From: =?iso-8859-1?Q?kav=E9?= Salamatian <salamat@lri.fr>
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> Dear All,
>
> I have some comments on draft-ietf-avt-reedsolomon-00.txt.
>
> 1) What do you think about adding a flavor of interleaving?  If the
> length of burst packet loss is proportional to bitrate, the protection

> becomes weak at high bitrate.  Increasing of N is a way to strengthen
> the protection, but complexity also increases.  Another way is
> interleaving, which disperses the consecutive lost packets.  By
> introducing a new parameter S, we easily lengthen a media block while
> using small numbers of N and K.  The step of media packets is defined
> as 2**S.  Thus a media block contains K media packets numbered
> SN+(2**S)*i where 0<=i<K.  The behavior with S=0 is exactly the same
> as the draft.

Hello,

interleaving basically reduce the memory of the loss process. But its
performance fade out fast. After a interleaving order of 4 to 5 your
interleaved process loss all its memory and became as a memoryless
process. If you want to have higher performance you should use stronger
MDS codes (by stronger codes I means more lenghty codes). About the
complexity of the decoding process it is linear on the number of lost
packets, squared on the length of the block and squared on size of the
underlying field of the Reed Solomon code. Some faster implementation
using discrete Fourier Transform exist who reduce the complexity to Nlog
N where N is the size of the code. But there are interesting only for
large values of N.

Interleaving increase highly the decoding latency but some means can be
used to reduce this latency. The idea is to interleave shifted versions
of a long block.  For example

Interleaving matrix
1   2   3   4   5
3   4   5   6   7
5   6   7   8   9
10 11 12 13 14
R1 R3 R5 R7 R9        Redundant packets
R2 R4 R6 R8 R10

Sent sequence
1 3 5 10 R1 R2 2 4 6 11 R3 R5 7 12 R6 R7  8 13 R8 R9 9 14

The latency is reduced but the redondance is added. You can easily
construct more intelligent interleaving patterns with interesting
properties.About interleaving performance and globally about general
performance of MDS codes (as Reed Solomon are) over erasure channels
with memory we have a report (which is unfortunately in french). You can
get it from www.lri.fr/~bouchero/MARS/main.ps.gz

Kave Salamatian






From rem-conf Mon Mar 15 21:47:45 1999 
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From: "John Weiler, CTO and Founder" <john_weiler@TheOTG.com>
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Dear Colleagues,

The Interoperability Clearinghouse Working Group (ICWG), is holding several
industry meetings in the upcoming weeks, hosted by OMG and AFCEA. Those
interested in engaging this joint government/ industry initiative should may
attend one of the planned ICWG meetings;

1) March 22, OMG Technical Committee Meeting, Wyndham Hotel, Philadelphia
(215-448-2000).
Open session for all OMG participants :12:00-1:00pm includes lunch and
overview briefing.
Meeting continues 1:00-4:00pm for working discussions.
Evening Session for prospective IC sponsor 5:00-7:00pm Cocktail hour follows.
Please register at the OMG registration desk.

2) March 23, 08:00am-12noon.  AFCEA Gaithersburg Breakfast Meeting, Hampton Inn,
off Rt. 118.  Cost $20. Payable to AFCEA.  Please RSVP. Limited seating.

3) On-site Brief. If your organization has an large IT shop who would
be interested in this effort, the principles would will be glad to set up an
on-site brief at your convenience.

The primary goal of the Interoperability Clearinghouse will be establish
an internet based knowledge repository of IT standards, "validated &
interoperable COTS product suites" and associated implementation services.
Integration, testing and implementation successes will help validate vendor
claims, and enable IT users to configure and validate enterprise information
solutions.

For more information on this effort, please visit the following
Interoperability Clearinghouse URLs;
http://www.omg.org/techprocess/meetings/ic.html
http://www.infoworld.com/cgi-bin/displayStory.pl?981113.whsoft.htm
http://www.fcw.com/pubs/fcw/1999/0208/fcw-newsinterop-2-8-99.html
http://www.gcn.com/gcn/1999/February8/1c.htm

We look forward to working with you to "making IT happen" for all.  Details of
how all of this will unfold will be discussed at these meetings. us know if your
organization is interested in participating.

Best Regards,

John

******************************************************
"From Architectures to Reality"

703-768-0400(v)
703-765-9295(f)

http://www.TheOTG.com
john_weiler@TheOTG.com
*********************************************



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Hello=20

    In my previous mail i had stated that i am facing problems with the =
stack in windows 98. I finally figured out why, the stack needs windows =
NT to operate correctly.

Thanks=20

Nikos Laoutaris
Network Operations Center
National and Kapodestrian University of Athens, Greece
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<DIV><FONT color=3D#000000 size=3D2>Hello </FONT></DIV>
<DIV><FONT color=3D#000000 size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#000000 size=3D2>&nbsp;&nbsp;&nbsp; In my previous =
mail i had=20
stated that i am facing problems with the stack in windows 98. I finally =
figured=20
out why, the stack needs windows NT to operate correctly.</FONT></DIV>
<DIV><FONT color=3D#000000 size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#000000 size=3D2>Thanks&nbsp;</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT color=3D#000000 size=3D2>Nikos Laoutaris<BR>Network =
Operations=20
Center<BR>National and Kapodestrian University of Athens, =
Greece<BR>e-mail : <A=20
href=3D"mailto:laoutaris@noc.uoa.gr">laoutaris@noc.uoa.gr</A></FONT></DIV=
></BODY></HTML>

------=_NextPart_000_0025_01BE6FA5.B3378600--




From rem-conf Tue Mar 16 09:14:15 1999 
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From: Florissa Colina <florissa@bmrc.berkeley.edu>
Subject: 3/17 Berkeley Multimedia, Interfaces and Graphics Seminar
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Berkeley Multimedia, Interfaces and Graphics Seminar


An Investigation into Web Site Design Practice


Wednesday March 17, 1999, 

1:00-2:30 p.m. 405 Soda Hall 


<italic>Please note the time of the seminar has changed!!

</italic>

Mark Newman

Computer Science Division - EECS 

University of California at Berkeley 


How can computer systems better support creative processes? Many
computer-based tools currently exist to support art and design tasks, but
the technology and the need exist to push these capabilities of these
tools farther. 


In the Group for User Interface Research at UC Berkeley, we are
undertaking a research project to understand how technology can be used
to further assist the web design process, with the ultimate goal of
developing tools to support designers. During the first phase of our
project, we conducted an informal study involving interviews with
professional designers and examination of works-in-progress to improve
our understanding of the web design practice in professional settings. 


In this talk we present the results of our initial investigation,
including observations about common and divergent design practices among
the organizations studied, and the existence of multiple design
specialties which confuse the definition of "web design". 


The seminar is broadcast on the Internet Mbone. The addresses are video
224.2.163.7/57076 and audio 224.2.147.61/27202. 


Sponsored by the Berkeley Multimedia Research Center 




____________________________________________


Florissa Colina

Berkeley Multimedia Research Center (BMRC)

http://bmrc.berkeley.edu/

626 Soda Hall #1776

University of California

Berkeley, CA 94720-1776

Phone: (510) 643-0800   FAX: (510) 642-5615  

Email: florissac@bmrc.berkeley.edu



From rem-conf Tue Mar 16 13:55:58 1999 
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From: srodawa <srodawa@oakland.edu>
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Subject: DVI audio player location.
To: rem-conf@es.net
Date: Tue, 16 Mar 1999 16:48:12 -0500 (EST)
Cc: srodawa@cliff.acs.oakland.edu (Ron Srodawa)
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I am trying to attend the IETF MBONE sessions.  VIC and WB reception is just 
fine.  AUdio is being sent in DVI format and VAT won't accept it.  Is there
an easy solution to this--either a "plugin" to VAT or another audio player?
Thanks for your help.  Ron.

| Ronald J. Srodawa               | Internet: srodawa@oakland.edu |
| School of Engineering and CS    | Voice:    (248) 370-2247      |
| Oakland University              | FAX:      (248) 370-4625      |
| Rochester, Michigan  48309-4478 |                               |



From rem-conf Tue Mar 16 18:42:30 1999 
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cc: rem-conf@es.net, srodawa@cliff.acs.oakland.edu (Ron Srodawa)
Subject: Re: DVI audio player location.
In-reply-to: Your message of "Tue, 16 Mar 1999 16:48:12 EST." <199903162148.QAA14462@jupiter.acs.oakland.edu>
Date: Wed, 17 Mar 1999 02:35:41 +0000
Message-ID: <1594.921638141@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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--> srodawa writes:
>I am trying to attend the IETF MBONE sessions.  VIC and WB reception is just 
>fine.  AUdio is being sent in DVI format and VAT won't accept it.  Is there
>an easy solution to this--either a "plugin" to VAT or another audio player?

Both vat and rat accept DVI format audio. If you're using vat, ensure that
the -r option is specified on the command line to make it understand RTP.

Colin



From rem-conf Tue Mar 16 20:57:01 1999 
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To: srodawa <srodawa@oakland.edu>
cc: rem-conf@es.net, srodawa@cliff.acs.oakland.edu (Ron Srodawa)
Subject: Re: DVI audio player location. 
Date: Tue, 16 Mar 1999 21:51:39 -0700
From: Evi Nemeth <evi@anchor.cs.colorado.edu>
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thats wierd, vat is generating the dvi.
dont know why he wouldnt accept it.  i
guess we could switch to pcm if it would
help you.

-evi



From rem-conf Wed Mar 17 12:15:50 1999 
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From: TECemail@aol.com
Message-ID: <9224984.36f008cc@aol.com>
Date: Wed, 17 Mar 1999 14:55:56 EST
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Subject: The Education Coalition (TEC) Newsletter!
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Welcome to The Education Coalition (TEC) Newsletter! We are very excited a=
bout
initiating an update to keep our business associates, and partners informe=
d
about our activities in educational technology and distributed learning. F=
or
further information visit our website at TECWeb.org or call us at
949-369-3867. 


TEC Newsletter  March 1999

Contents: 

  TEC headquarters relocates to Southern California
The move is completed !  TEC=92s corporate headquarters is unpacked and se=
ttled
in beautiful San Clemente, California, on the coast midpoint between San D=
iego
and Los Angeles.
 ( Address:  31 Segovia, San Clemente, California  tel: 949-369-3867) 
 
 You will hear the friendly voice of Kathleen Bail, office administrator,
answering the phones in San Clemente; Donna LoBue, operating TEC in the Ba=
y
area, can be reached at 925-691-5650.
 
  VTEC selected as developer for TEC Personal Learning Model 

Providing course material in a student=92s preferred learning style offers
quicker, better learning with a higher retention rate.  The question becom=
es
how to ascertain learning styles for online courses. To meet this need, TE=
C is
developing the Personal Learning Model, (PLM), an online learning styles/
multiple intelligences (LS/MI) assessment that will determine a student=92=
s
preferred learning style and secondary learning modes.

 The PLM is not a "paper-and-pencil" type test; rather it will be a series=
 of
tasks and problems to solve in a virtual environment. The  responses will =
be
tracked and built into a LS/MI profile.  The profile is tagged, stored in =
a
data base, and passed to an online course so material can be presented in =
the
preferred learning style. 

  After issuing an RFP, TEC has selected the Virtual Technology Education
Center (VTEC) from the South Orange County Community College District (SOC=
CCD)
in California to develop the PLM using Asymetrix software.   PLM adheres t=
o
the EDUCAUSE/Information Management System (IMS) standards, and utilizes
object-oriented, reusable learning objects. 

TEC is currently accepting applicants to be BETA testers for the software.=
 If
you are interested in participating in this milestone project, call the TE=
C
office for more details. 

 
  Training =9299 in Chicago
 
 TEC, represented by Dr. Carla Lane and Joanne Carle-Accornero, presented =
to
approximately 125 people in the Windy City on February 3rd. Topics of
discussion were  an overview of learning styles, addressing learning style=
s
through technology, the Advanced Distributed Learning (ADL) initiative, an=
d
the how to assess learning styles online.  
 
  The San Bernardino Student and Teacher Excellence Project (STEP)  
 San Bernardino is one of the few areas in the US chosen to offer its stud=
ents
accelerated science and math programs based on material from NASA. This
engaging program enables students to communicate with astronauts and learn
complex math and science concepts  
 by solving problems based on actual NASA space travel issues. 
 
 San Bernardino is committed to providing  total educational improvement
through the integration of staff development, curriculum development, home
access, evaluation and technology/ infrastructure support. TEC is delighte=
d to
have been selected as the evaluator of this innovative program with its
winning use of technology in the classroom.
 
  TeleCon East and TeleCon West
 TEC has again been selected to produce the TeleCon East (Washington, DC,
March) and TeleCon West (Anahaim, CA, Oct ) conferences. These are super-
charged shows that feature bleeding-edge technologies, new as well as
sophisticated users, educators, business people and government
representatives, all focused on using technology to enhance learning. Tele=
Con
East is being held from March 17 through March 19th at the Marriott Wardma=
n
Park Hotel in Washington, DC.
 
 We hope to see you at the TEC Breakout sessions on Thursday, March 18th ,=
 in
the Virginia C Conference Room from noon through 5:15 PM !
 
  
  Web-based instruction offered through TEC 
With an already FULL schedule, how can one keep up with the constant and r=
apid
changes in the distributed learning field? Try a TEC online course
specifically designed for the busy professional! 

Facilitated by Dr. Lane and Joanne Carle-Accornero, developers and instruc=
tors
for numerous online programs including UCLA and UC Berkeley,  the followin=
g
courses are currently being offered:
  Introduction to Distributed Learning - 15 hours - asynchronous online an=
d
audio conferencing - $59
  Using Technology to Address Learning Styles - 15 hours  - asynchronous
online and audio-conferencing- $59  

The textbook for the courses is Guide to Teleconferencing and Distance
Learning, 3rd Edition, co-authored by Dr. Carla Lane, which is available
through Amazon.com. 
 

Course sections will have limited enrollments! For more information and to
enroll, call 949-369-3867.  



TECWeb.org              Tel:  949-369-3867     Fax: 949-369-3865




From rem-conf Wed Mar 17 16:24:38 1999 
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Message-ID: <36EFDFDD.2375A698@TheOTG.com>
Date: Wed, 17 Mar 1999 12:01:17 -0500
From: "John Weiler, CTO and Founder" <john_weiler@TheOTG.com>
Organization: The OBJECTive Technology Group
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>>>>>>>>>>>>>>>>>Please note the corrected dates below:

Dear Colleagues,

The Interoperability Clearinghouse Working Group (ICWG), is holding several
industry meetings in the upcoming weeks, hosted by OMG and AFCEA. Those
interested in engaging this joint government/ industry initiative should may
attend one of the planned ICWG meetings;

1) Wednesday, March 24, OMG Technical Committee Meeting, Wyndham Hotel,
Philadelphia (215-448-2000).
Open session for all OMG participants :12:00-1:00pm includes lunch and
overview briefing.
Meeting continues 1:00-4:00pm for working discussions.
Evening Session for prospective IC sponsors 5:00-7:00pm Cocktail hour follows.
Please register at the OMG registration desk. (Guest price for OMG meeting day
is $50).

2) Thursday, March 25, 08:00am-12noon.  AFCEA Gaithersburg Breakfast Meeting,
Hampton Inn,
off Rt. 118.  Cost $20. Payable to AFCEA.   Limited seating. Please RSVP to
James M. Robinette, james.robinette@hq.doe.gov

3) On-site Brief. If your organization has an large IT shop who would
be interested in this effort, the principles would will be glad to set up an
on-site brief at your convenience.

The primary goal of the Interoperability Clearinghouse will be establish
an internet based knowledge repository of IT standards, "validated &
interoperable COTS product suites" and associated implementation services.
Integration, testing and implementation successes will help validate vendor
claims, and enable IT users to configure and validate enterprise information
solutions.

For more information on this effort, please visit the following
Interoperability Clearinghouse URLs;
http://www.omg.org/techprocess/meetings/ic.html
http://www.infoworld.com/cgi-bin/displayStory.pl?981113.whsoft.htm
http://www.fcw.com/pubs/fcw/1999/0208/fcw-newsinterop-2-8-99.html
http://www.gcn.com/gcn/1999/February8/1c.htm

We look forward to working with you to "making IT happen" for all.  Details of
how all of this will unfold will be discussed at these meetings. us know if your

organization is interested in participating.

Best Regards,

John

******************************************************
"From Architectures to Reality"

703-768-0400(v)
703-765-9295(f)

http://www.TheOTG.com
john_weiler@TheOTG.com
*********************************************


--

*********************************************
"From Architectures to Reality"

John A. Weiler
CTO & Founder, The OBJECTive Technology Group
Ombudsman, Interoperability Clearinghouse Coalition
Member; OMG, IEEE, ACM, AFCEA
8011 Washington Ave.
Alexandria, VA 22308
703-768-0400(v) 703-765-9295(f)
http://www.TheOTG.com (Industry@OOTS Forums)
Interoperability Clearinghouse URLs;
http://www.omg.org/techprocess/meetings/ic.html
http://www.infoworld.com/cgi-bin/displayStory.pl?981113.whsoft.htm
http://www.fcw.com/pubs/fcw/1999/0208/fcw-newsinterop-2-8-99.html
http://www.gcn.com/gcn/1999/February8/1c.htm

john_weiler@TheOTG.com

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





From rem-conf Thu Mar 18 03:08:33 1999 
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To: abctech@gte.net
Subject: You have been invited
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You are invited!

Now for the first time ever you have the opportunity to join the most 
extraordinary and most powerful wealth-building program in the world!

Because of your desire to succeed, you have been given the opportunity to
take a closer look at this program.

If you're skeptical, that's ok. Just make the call and see for yourself.
My job is to inform you. Your job is to make your own decision.

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You owe it to yourself and to your family to give our program 
Serious Consideration!

Also, when you start making this kind of money within weeks, after joining
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I invite you to call me for more details at 1-800-636-6773 Ext 4848. 
This is a free 2 minute recording, so call right now!

Prosperous regards,


Mary Shearer

This is Not Multi-level Marketing. Please, Serious Inquiries Only!

This is a one time mailing. When you visited one of our web pages you indicated
that you would be interested in this information If not, please excuse the intrusion.
Thank you.

Under Bill s.1618 TITLE III passed by the 105th U.S. Congress this letter can not
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All future mailings from other e-mail addresses must be dealt with separetely.



From rem-conf Thu Mar 18 15:00:14 1999 
From rem-conf-request@es.net Thu Mar 18 15:00:13 1999
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From: "Neal Schneider" <nschneid@hotmail.com>
To: rem-conf@es.net
Cc: nschneid@hotmail.com
Subject: Unique RTP Sessions
Date: Thu, 18 Mar 1999 14:48:56 PST
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In a situation with gateways there can be many unique RTP sessions 
established between them.  In general, are RTP sessions uniquely 
identified via distinct UDP ports?  Or is there a single UDP port opened 
for all RTP traffic, with other unique identifiers?  I realize this is 
application specific, but am curious as to what is currently popular.

Regards, 
Neal A. Schneider

 

______________________________________________________
Get Your Private, Free Email at http://www.hotmail.com



From rem-conf Thu Mar 18 19:38:27 1999 
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Date: Thu, 18 Mar 1999 22:39:37 -0500
To: mbone@ISI.EDU, rem-conf@es.net
From: "Ellery D. Coleman" <Ellery.Coleman@hq.nasa.gov>
Subject: NASA - Workplace Health Surveillance/Maintenance
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	NASA HQ will be mulitcasting the 2nd of a series of discussions on the
topic of "Workplace Health Surveillance/Maintenance" tomorrow (3.19.99)
>from 9:30am-12:00pm EST.  The session will feature speakers Vladimir
Subbotin and Albert Nechaev (Moscow, Russia) and Jun-Yao Li (Beijing,
China) via the NASA HQ Video Teleconferencing facility.  I would greatly
appreciate if some of you could tune in and send email with reguards to the
quality of video and audio you're receiving. 

                       Thanks in Advance!,
                                    o-> el


Session Info:
Title:  NASA - Workplace Health Surveillance/Maintenance
Video:  224.2.235.72:59348
Audio:  224.2.235.72:17274



From rem-conf Thu Mar 18 20:38:21 1999 
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From: "Dean Willis" <Dean.Willis@MCI.COM>
To: "Neal Schneider" <nschneid@hotmail.com>, <rem-conf@es.net>
Subject: RE: Unique RTP Sessions
Date: Thu, 18 Mar 1999 22:31:47 -0600
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I would say that in practice RTP sessions are uniquely identified by the
"socket", that is, the double tuple of the host IP Addresses and
associated port numbers at each end, or {(IP1,PN1),(IP2,PN2)}

So, the session from 10.1.1.1 port 3456 to host 172.16.1.2 port 4692
would be identified by the socket {(10.1.1.1,3456),(172.16.1.2)}.

That way, the IP demux layer can hand incoming packets to the right
process. For reference, I like Richard Steven's "TCP/IP Illustrated,
Volume 1". It's getting a bit dated, but is a good read, and Stevens is
a personable fellow who used to be quite handy at answering questions on
usenet group comp.protocols.tcpip.

Quoting from RFC1889 (RTP spec)
---
Port: The "abstraction that transport protocols use to distinguish among
multiple destinations within a given host computer. TCP/IP protocols
identify ports using small positive integers." [3] The transport
selectors (TSEL) used by the OSI transport layer are equivalent to
ports.  RTP depends upon the lower-layer protocol to provide some
mechanism such as ports to multiplex the RTP and RTCP packets of a
session.
---

And also:
---
In RTP, multiplexing is provided by the destination transport address
(network address and port number) which define an RTP session. For
example, in a teleconference composed of audio and video media encoded
separately, each medium should be carried in a separate RTP session with
its own destination transport address. It is not intended that the audio
and video be carried in a single RTP session and demultiplexed based on
the payload type or SSRC fields. Interleaving packets with different
payload types but using the same SSRC would introduce several problems.
. .
---

Note that there is in general also an associated RTCP socket, occupying
by default the next higher-numbered port on each machine.

Also note that there is an ongoing discussion on how to combine multiple
RTP streams onto one socket using a additional multiplexing protocol
that allows extraction of the individual streams.

--
Dean

> -----Original Message-----
> From: Neal Schneider [mailto:nschneid@hotmail.com]
> Sent: Thursday, March 18, 1999 4:49 PM
> To: rem-conf@es.net
> Cc: nschneid@hotmail.com
> Subject: Unique RTP Sessions
>
>
> In a situation with gateways there can be many unique RTP sessions
> established between them.  In general, are RTP sessions uniquely
> identified via distinct UDP ports?  Or is there a single UDP
> port opened
> for all RTP traffic, with other unique identifiers?  I
> realize this is
> application specific, but am curious as to what is currently popular.
>
> Regards,
> Neal A. Schneider
>
>
>
> ______________________________________________________
> Get Your Private, Free Email at http://www.hotmail.com
>




From rem-conf Thu Mar 18 21:57:44 1999 
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From: "Dean Willis" <Dean.Willis@MCI.COM>
To: "Dean Willis" <Dean.Willis@MCI.COM>,
        "Neal Schneider" <nschneid@hotmail.com>, <rem-conf@es.net>
Subject: RE: Unique RTP Sessions, oops
Date: Thu, 18 Mar 1999 23:49:45 -0600
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You know, it's really embarassing when I screw up while lecturing.
Especially when the audience has already forgotten more than I will ever
know.

The:
> would be identified by the socket {(10.1.1.1,3456),(172.16.1.2)}.

should look more like:
{(10.1.1.1,3456),(172.16.1.2,4692)}

The port number of the destination end is rather important . . .

--
Dean

> -----Original Message-----
> From: Dean Willis [mailto:Dean.Willis@mci.com]
> Sent: Thursday, March 18, 1999 10:32 PM
> To: Neal Schneider; rem-conf@es.net
> Subject: RE: Unique RTP Sessions
>
>
>
> I would say that in practice RTP sessions are uniquely
> identified by the
> "socket", that is, the double tuple of the host IP Addresses and
> associated port numbers at each end, or {(IP1,PN1),(IP2,PN2)}
>
> So, the session from 10.1.1.1 port 3456 to host 172.16.1.2 port 4692
> would be identified by the socket {(10.1.1.1,3456),(172.16.1.2)}.
>
> That way, the IP demux layer can hand incoming packets to the right
> process. For reference, I like Richard Steven's "TCP/IP Illustrated,
> Volume 1". It's getting a bit dated, but is a good read, and
> Stevens is
> a personable fellow who used to be quite handy at answering
> questions on
> usenet group comp.protocols.tcpip.
>
> Quoting from RFC1889 (RTP spec)
> ---
> Port: The "abstraction that transport protocols use to
> distinguish among
> multiple destinations within a given host computer. TCP/IP protocols
> identify ports using small positive integers." [3] The transport
> selectors (TSEL) used by the OSI transport layer are equivalent to
> ports.  RTP depends upon the lower-layer protocol to provide some
> mechanism such as ports to multiplex the RTP and RTCP packets of a
> session.
> ---
>
> And also:
> ---
> In RTP, multiplexing is provided by the destination transport address
> (network address and port number) which define an RTP session. For
> example, in a teleconference composed of audio and video media encoded
> separately, each medium should be carried in a separate RTP
> session with
> its own destination transport address. It is not intended
> that the audio
> and video be carried in a single RTP session and
> demultiplexed based on
> the payload type or SSRC fields. Interleaving packets with different
> payload types but using the same SSRC would introduce several
> problems.
> . .
> ---
>
> Note that there is in general also an associated RTCP socket,
> occupying
> by default the next higher-numbered port on each machine.
>
> Also note that there is an ongoing discussion on how to
> combine multiple
> RTP streams onto one socket using a additional multiplexing protocol
> that allows extraction of the individual streams.
>
> --
> Dean
>
> > -----Original Message-----
> > From: Neal Schneider [mailto:nschneid@hotmail.com]
> > Sent: Thursday, March 18, 1999 4:49 PM
> > To: rem-conf@es.net
> > Cc: nschneid@hotmail.com
> > Subject: Unique RTP Sessions
> >
> >
> > In a situation with gateways there can be many unique RTP sessions
> > established between them.  In general, are RTP sessions uniquely
> > identified via distinct UDP ports?  Or is there a single UDP
> > port opened
> > for all RTP traffic, with other unique identifiers?  I
> > realize this is
> > application specific, but am curious as to what is
> currently popular.
> >
> > Regards,
> > Neal A. Schneider
> >
> >
> >
> > ______________________________________________________
> > Get Your Private, Free Email at http://www.hotmail.com
> >
>
>




From rem-conf Fri Mar 19 01:47:31 1999 
From rem-conf-request@es.net Fri Mar 19 01:47:30 1999
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To: rem-conf@es.net
Subject: clarification on IPR issues
Cc: sob@harvard.edu
Date: Fri, 19 Mar 1999 01:42:37 PST
From: Vern Paxson <vern@ee.lbl.gov>
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My comments at this week's meeting regarding IPR issues weren't quite right.
In particular, it's possible for a document to advance on the standards
track if no reply is received from an alleged rights holder regarding
licensing.  I've appended the relevant section from RFC 2026.

		Vern


10.3.2. Standards Track Documents

   (A)  Where any patents, patent applications, or other proprietary
      rights are known, or claimed, with respect to any specification on
      the standards track, and brought to the attention of the IESG, the
      IESG shall not advance the specification without including in the
      document a note indicating the existence of such rights, or
      claimed rights.  Where implementations are required before
      advancement of a specification, only implementations that have, by
      statement of the implementors, taken adequate steps to comply with
      any such rights, or claimed rights, shall be considered for the
      purpose of showing the adequacy of the specification.
   (B)  The IESG disclaims any responsibility for identifying the
      existence of or for evaluating the applicability of any claimed
      copyrights, patents, patent applications, or other rights in the
      fulfilling of the its obligations under (A), and will take no
      position on the validity or scope of any such rights.
   (C)  Where the IESG knows of rights, or claimed rights under (A), the
      IETF Executive Director shall attempt to obtain from the claimant
      of such rights, a written assurance that upon approval by the IESG
      of the relevant Internet standards track specification(s), any
      party will be able to obtain the right to implement, use and
      distribute the technology or works when implementing, using or
      distributing technology based upon the specific specification(s)
      under openly specified, reasonable, non-discriminatory terms.
      The Working Group proposing the use of the technology with respect
      to which the proprietary rights are claimed may assist the IETF
      Executive Director in this effort.  The results of this procedure
      shall not affect advancement of a specification along the
      standards track, except that the IESG may defer approval where a
      delay may facilitate the obtaining of such assurances.  The
      results will, however, be recorded by the IETF Executive Director,
      and made available.  The IESG may also direct that a summary of
      the results be included in any RFC published containing the
      specification.



From rem-conf Fri Mar 19 08:32:27 1999 
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To: mbone@ISI.EDU, rem-conf@es.net
From: "Ellery D. Coleman" <Ellery.Coleman@hq.nasa.gov>
Subject: LIVE:  NASA - Workplace Health Surveillance/Maintenance
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==============
Short version:
==============

All, 
       please tune in to the session listed below:

Title:  NASA - Workplace Health Surveillance/Maintenance
Video:  224.2.235.72:59348
Audio:  224.2.235.72:17274

Any comments on quality of reception will be greatly appreciated.

                             o-> el




==============
Long version:
==============   
NASA HQ is currently mulitcasting the 2nd of a series of discussions on the
topic of "Workplace Health Surveillance/Maintenance".  This discussion is
estimated to run 
>from 9:30am-12:00pm EST.  The session features speakers Vladimir
Subbotin and Albert Nechaev (Moscow, Russia) and Jun-Yao Li (Beijing,
China) via the NASA HQ Video Teleconferencing facility.  I would greatly
appreciate if some of you could tune in and send email with reguards to the
quality of video and audio you're receiving. 

                       Thanks in Advance!,
                                    o-> el


Session Info:
Title:  NASA - Workplace Health Surveillance/Maintenance
Video:  224.2.235.72:59348
Audio:  224.2.235.72:17274





From rem-conf Fri Mar 19 10:17:33 1999 
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Sender: "Mark Baugher" <mbaugher@intel.com>
To: "'Neal Schneider'" <nschneid@hotmail.com>, <rem-conf@es.net>
Subject: RE: Unique RTP Sessions
Date: Fri, 19 Mar 1999 10:10:40 -0800
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hi
>From RFC 1889:
   RTP session: The association among a set of participants
        communicating with RTP. For each participant, the session is
        defined by a particular pair of destination transport addresses
        (one network address plus a port pair for RTP and RTCP). The
        destination transport address pair may be common for all
        participants, as in the case of IP multicast, or may be
        different for each, as in the case of individual unicast network
        addresses plus a common port pair.  In a multimedia session,
        each medium is carried in a separate RTP session with its own
        RTCP packets. The multiple RTP sessions are distinguished by
        different port number pairs and/or different multicast
        addresses.

   Synchronization source (SSRC): The source of a stream of RTP packets,
        identified by a 32-bit numeric SSRC identifier carried in the
        RTP header so as not to be dependent upon the network address.
        All packets from a synchronization source form part of the same
        timing and sequence number space, so a receiver groups packets
        by synchronization source for playback. Examples of
        synchronization sources include the sender of a stream of
        packets derived from a signal source such as a microphone or a
        camera, or an RTP mixer (see below). A synchronization source
        may change its data format, e.g., audio encoding, over time. The
        SSRC identifier is a randomly chosen value meant to be globally
        unique within a particular RTP session (see Section 8). A
        participant need not use the same SSRC identifier for all the
        RTP sessions in a multimedia session; the binding of the SSRC
        identifiers is provided through RTCP (see Section 6.4.1).  If a
        participant generates multiple streams in one RTP session, for
        example from separate video cameras, each must be identified as
        a different SSRC.

Mark

> -----Original Message-----
> From: Neal Schneider [mailto:nschneid@hotmail.com]
> Sent: Thursday, March 18, 1999 2:49 PM
> To: rem-conf@es.net
> Cc: nschneid@hotmail.com
> Subject: Unique RTP Sessions
>
>
> In a situation with gateways there can be many unique RTP sessions
> established between them.  In general, are RTP sessions uniquely
> identified via distinct UDP ports?  Or is there a single UDP
> port opened
> for all RTP traffic, with other unique identifiers?  I
> realize this is
> application specific, but am curious as to what is currently popular.
>
> Regards,
> Neal A. Schneider
>
>
>
> ______________________________________________________
> Get Your Private, Free Email at http://www.hotmail.com
>





From rem-conf Fri Mar 19 10:42:28 1999 
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From: Christian Isnard - CERN IT/IA <Christian.Isnard@cern.ch>
To: rem-conf MBONE List <rem-conf@es.net>
Subject: CERN MBone Announcement: Next LEPC 
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           CERN is pleased to announce the MBONE broadcast of the

                             LEPC Open Session
                             =================
                               24 March 1999

Place: CERN Main Auditorium

*** Times are UTC+1 ***

                LEP machine report:
09:00 - 09:30   Summary of the 1999 Chamonix Workshop (Roger Bailey)
                LEP2 physics jamboree: 
09:30 - 10:10   DELPHI (Niels Kjaer)
10:10 - 10:50   L3 (Gerjan Bobbink)
10:50 - 11:20   Coffee Break
11:20 - 12:00   OPAL (Douglas Glenzinski)
12:00 - 12:40   ALEPH (Fabiola Gianotti)
                LEP2 workshop report: 
12:40 - 13:00   Summary of the LEP2 Monte Carlo Workshop (Roberto Pittau)

The broadcast is announced via sdr as "CERN LEPC". vat and vic applications
will be used with a ttl of 127.

In case of questions or problems please contact: multicast@noc.cern.ch







From rem-conf Fri Mar 19 10:44:55 1999 
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From: Christian Isnard - CERN IT/IA <Christian.Isnard@cern.ch>
To: rem-conf MBONE List <rem-conf@es.net>
Subject: CERN MBone Announcement: Next LHCC 
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           CERN is pleased to announce the MBONE broadcast of the

                             LHCC Open Session
                             =================
                               25 March 1999

 Place: CERN Main Auditorium

*** Times are UTC+1 ***

09.00-09.40     TOTEM Proposal: Total Cross Section, Elastic Scattering and
Diffraction Dissociation at the LHC (G. Matthiae)

09.40-10.20     MOEDAL Letter of Intent: A search for highly ionizing
particles and slow exotic decays at the LHC (J. Pinfold)

10.20-10.45     COFFEE BREAK

10.45-12.00     ATLAS Technical Co-ordination TDR (F. Butin, M. Hatch,
B. Nicquevert)

12.00-12.45     ALICE Photon Spectrometer TDR (V. Manko)

12.45-13.15     ALICE Zero-Degree Calorimeter TDR (M. Gallio)


The broadcast is announced via sdr as "CERN LHCC". vat and vic applications
will be used with a ttl of 127.

In case of questions or problems please contact: multicast@noc.cern.ch






From rem-conf Fri Mar 19 11:16:40 1999 
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Date: Fri, 19 Mar 1999 16:09:48 -0300
From: Gillermo Rigotti <grigotti@exa.unicen.edu.ar>
Organization: UNICEN
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Hi,

I have a question about the functionality required in routers to support

mixers and translators.
Some RTCP packets are received and processed by mixers and translators,
this packets can
be modified and send. The original ones must not be propagated (for
example a translator converting
encodings). If we are using multicast, the router directly connected to
the host where translator resides must not propagate that packets.
Is necessary  to modify router  functionality to avoid propagation?
(assuming the case translators
are placed dynamically in funcion of characteristics of sets of
receivers, so they can't be statically  configured).

Many Thanks

M Guillermo Rigotti
ISISTAN Research Institute
Universidad Nacional  del Centro de la Provincia de Buenos Aires
Campus Universitario - Paraje Arroyo Seco - Tandil - 7000
Buenos Aires -  Argentina
Home: +54 (2293) 440824  Work: +54 (2293) 440363
e-mail: grigotti@exa.unicen.edu.ar








From rem-conf Fri Mar 19 15:36:09 1999 
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Subject: Re: Mixers, translator and routers
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I think the assumption is that the mixer/translator is involved in two
different sessions. For a translator, media comes in from one session,
gets translated, and goes into another session. Similar for a mixer.
Since these are different sessions, the RTCP packets from one won't be
part of the other. If the different sessions are on the same group, but
different ports, the packets will be forwarded but not processed by
hosts in the other session. No explicit support is needed in routers for
translators or mixers.

-Jonathan R.

Gillermo Rigotti wrote:
> 
> Hi,
> 
> I have a question about the functionality required in routers to support
> 
> mixers and translators.
> Some RTCP packets are received and processed by mixers and translators,
> this packets can
> be modified and send. The original ones must not be propagated (for
> example a translator converting
> encodings). If we are using multicast, the router directly connected to
> the host where translator resides must not propagate that packets.
> Is necessary  to modify router  functionality to avoid propagation?
> (assuming the case translators
> are placed dynamically in funcion of characteristics of sets of
> receivers, so they can't be statically  configured).
> 
> Many Thanks
> 
> M Guillermo Rigotti
> ISISTAN Research Institute
> Universidad Nacional  del Centro de la Provincia de Buenos Aires
> Campus Universitario - Paraje Arroyo Seco - Tandil - 7000
> Buenos Aires -  Argentina
> Home: +54 (2293) 440824  Work: +54 (2293) 440363
> e-mail: grigotti@exa.unicen.edu.ar

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



From rem-conf Sat Mar 20 22:59:58 1999 
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From: <print2live@angelfire.com>
Subject: AD:Family Reunion T Shirts & More
Date: Sun, 21 Mar 1999 01:45:19
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Message sent by:  Kuppler Graphics, 32 West Main Street, Maple Shade, New Jersey, 08052,
1-800-810-4330.   This list will NOT be sold.  All addresses 
are automatically added to our remove list.

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From rem-conf Mon Mar 22 02:40:51 1999 
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Date: Mon, 22 Mar 1999 20:01:05 +0930
From: Malcolm Caldwell <malcolm@it.ntu.edu.au>
To: rem-conf@es.net
Subject: mrouted 3.8 for Sparc Linux
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Hello,

I have a redhat 5.2 system running on a Sparc box with a Linux 2.2.1 kernel.

I am having heaps of problems compiling mrouted for this box.

I have tried many ways to get around the problem.

I have tried the patches for 3.8 as mentioned in
/usr/src/linux/Documentation/networking 

I have tried following instructions on the page (somewhere) that was supposed
to help. (It mentions using headers from tcpdump)

I keep getting to the point where it says that
vif.c: In function `dump_vifs':
vif.c:1373: `SIOCGETVIFCNT' undeclared (first use in this function)
vif.c:1373: (Each undeclared identifier is reported only once
vif.c:1373: for each function it appears in.)
make: *** [vif.o] Error 1

This is an ioctl:
...
   if (ioctl(igmp_socket, SIOCGETVIFCNT, (char *)&v_req) < 0) {
...

I can not find this defined anywhere in /usr/include

I am at my wits end - can anyone help??




From rem-conf Mon Mar 22 13:53:38 1999 
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Subject: 3/31 Berkeley Multimedia, Interfaces and Graphics Seminar
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Berkeley Multimedia, Interfaces and Graphics Seminar


How did IP Multicast Get So Complicated? 


Wednesday March 31, 1999

1:00-2:30 p.m. 405 Soda Hall 

<italic>Please note the time frame of the seminar series has changed!!

</italic>

Mark Handley

AT&T Center for Internet Research at ISCI (ACIRI)


To understand where IP Multicast looks like it's going, you need to have
some understanding of PIM-DM, PIM-SM, MSDP, BGMP, MBGP, MZAP, MADCAP, AAP
and MASC. And that's before you even consider anything at the transport
or application layers. It is not surprising then that people have started
to question the underlying assumptions about what problem we are
attempting to solve, and alternative proposals such as Express and
"Simple Multicast" have recently been proposed. 


In this talk I will try to go through the evolution of IP multicast,
explaining briefly how these protocols work and how they fit together.
I'll also touch on the two proposed alternatives and explain why I
believe the "traditional" approach is still desirable despite the
additional mechanism needed to support it. No prior knowledge of
multicast routing or address allocation will be assumed. 


The seminar is broadcast on the Internet Mbone. The addresses are video
224.2.163.7/57076 and audio 224.2.147.61/27202. 


Sponsored by the Berkeley Multimedia Research Center 




____________________________________________


Florissa Colina

Berkeley Multimedia Research Center (BMRC)

http://bmrc.berkeley.edu/

626 Soda Hall #1776

University of California

Berkeley, CA 94720-1776

Phone: (510) 643-0800   FAX: (510) 642-5615  

Email: florissac@bmrc.berkeley.edu



From rem-conf Mon Mar 22 14:11:16 1999 
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--NextPart

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

	Title		: An RTP Payload Format for Generic Forward Error Correction
	Author(s)	: J. Rosenberg, H. Schulzrinne
	Filename	: draft-ietf-avt-fec-05.txt
	Pages		: 19
	Date		: 19-Mar-99
	
   This document specifies a payload format for generic forward error
   correction of media encapsulated in RTP. It is engineered for FEC
   algorithms based on the exclusive-or (parity) operation. The payload
   format allows end systems to transmit using arbitrary block lengths
   and parity schemes. It also allows for the recovery of both the pay-
   load and critical RTP header fields. Since FEC is sent as a separate
   stream, it is backwards compatible with non-FEC capable hosts, so
   that receivers which do not wish to implement FEC can just ignore the
   extensions.

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From rem-conf Mon Mar 22 19:09:33 1999 
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To: rem-conf@es.net
Subject: Re: mrouted 3.8 for Sparc Linux 
In-reply-to: Your message of "Mon, 22 Mar 1999 20:01:05 +0930."
             <19990322200105.A6259@hakea.cs.ntu.edu.au> 
Date: Tue, 23 Mar 1999 12:05:31 +0900
From: Yozo Toda <yozo@aohakobe.ipc.chiba-u.ac.jp>
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> vif.c: In function `dump_vifs':
> vif.c:1373: `SIOCGETVIFCNT' undeclared (first use in this function)
> vif.c:1373: (Each undeclared identifier is reported only once
> vif.c:1373: for each function it appears in.)
> make: *** [vif.o] Error 1
> 
> This is an ioctl:
> ...
>    if (ioctl(igmp_socket, SIOCGETVIFCNT, (char *)&v_req) < 0) {
> ...
> 
> I can not find this defined anywhere in /usr/include

on RedHat-5.2/intel I can find /usr/include/linux/mroute.h and
it contains "#define SIOCGETVIFCNT ...".
I suppose RedHat-5.2/sparc is the same.

BTW is your mrouted 3.8?  try compile mrouted-3.9-beta3.

and there is mrouted-3.9-beta3-snmp (-:
see
  <ftp://ftp.kyoto.wide.ad.jp/
           multicast/mrouted/snmp-mrouted/snmp-mrouted-3.9-beta3/>
or
  <http://aohakobe.ipc.chiba-u.ac.jp/
            misc/JP-MBone/tmp/mrouted-3.9-beta3-snmp-working/>.

cmu-snmp library used in this version (taken from mrouted-3.8-snmp)
looks very old and not support linux,
but adjusting the sources not difficult, I think, if you have enough time.
or how about importing newer cmu-snmp or ucsd-snmp library?

-- yozo.




From rem-conf Tue Mar 23 05:34:18 1999 
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Date: Tue, 23 Mar 1999 19:02:38 +0530
From: Venkata Narasimha Murthy Nanduri <nvnmurthy@seeyes.co.in>
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Organization: SeeYes
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Hai,

Can you please let me know whether any sample code for RTP header
compression/decompression exists anywhere in the net?
Any algorithm available for this purpose?

I have gone through the RFC2508 and RFC1144.
Any suggestions/help is appreciated..

Thanks in advance
Murthy NVN


--
-------------------------------------
Venkata Narasimha Murthy Nanduri,
Software Engineer,
SeeYes Network Technologies Pvt. Ltd
H-4, DD Colony, Bagh Amberpet,
Hyderabad, Andhra Pradesh,
India - 500 013

Office : 91-40-760 7842
Home   : 91-40-712 3437
Fax    : 91-40-761 2860
-------------------------------------


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From rem-conf Tue Mar 23 05:34:18 1999 
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Message-ID: <36F798B0.8168C2C@seeyes.co.in>
Date: Tue, 23 Mar 1999 19:05:44 +0530
From: Venkata Narasimha Murthy Nanduri <nvnmurthy@seeyes.co.in>
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Hai,

In the RFC2508, it has been mentioned that RFC1144 has to be
taken as the base for RTP Header compression/decompression.

Can any of U please tell me how I can proceed/modify the code
given in RFC1144?

Thanks in advance
Murthy NVN


--
-------------------------------------
Venkata Narasimha Murthy Nanduri,
Software Engineer,
SeeYes Network Technologies Pvt. Ltd
H-4, DD Colony, Bagh Amberpet,
Hyderabad, Andhra Pradesh,
India - 500 013

Office : 91-40-760 7842
Home   : 91-40-712 3437
Fax    : 91-40-761 2860
-------------------------------------


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From rem-conf Tue Mar 23 09:41:55 1999 
From rem-conf-request@es.net Tue Mar 23 09:41:54 1999
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To: rem-conf@es.net
Subject: Slides from the Minneapolis AVT meeting
Date: Tue, 23 Mar 1999 17:36:41 +0000
Message-ID: <29034.922210601@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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I've started to make a collection of the presentation slides from the AVT
meeting in Minneapolis which is available on the web at
	http://www-mice.cs.ucl.ac.uk/multimedia/misc/avt/
Could all those who presented please send me a copy of their slides so I
can include them here and in the minutes.

Cheers,
Colin



From rem-conf Thu Mar 25 08:10:48 1999 
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Date: Thu, 25 Mar 1999 23:59:49 +0800 (HKT)
From: Ngai Tsz Man <h9608112@hkusua.hku.hk>
To: rem-conf@es.net
Subject: Vic problem 
In-Reply-To: <199902230934.KAA24956@chico.rediris.es>
Message-ID: <Pine.GSO.4.05.9903252355130.7093-100000@hkusua.hku.hk>
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Dear all,

	I have installed Linux RedHat in my PC and I have a capture card
with chipset Bt878 and I want to do video conferencing with them using
Vic. Is it possible ?
	I know there is a project called Video4Linux that has driver for
my captuer card but what else do I need ?
	Thank you very much !

Tom





From rem-conf Thu Mar 25 08:10:52 1999 
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Date: Thu, 25 Mar 1999 08:03:58 -0800 (PST)
From: almeroth@cs.ucsb.edu (Kevin C. Almeroth)
Message-Id: <199903251603.IAA09874@jackson.cs.ucsb.edu>
To: almeroth@jackson.cs.ucsb.edu
Cc: ksarac@jackson.cs.ucsb.edu
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Based on some of the feedback we received at the IETF we have modified
the sdr-monitor WWW pages to add a new look for the GLOBAL sessions.

http://imj.ucsb.edu/sdr-monitor/global/

This page has a matrix of red/green showing reachability for GLOBAL 
sessions and sdr-monitor participants.  The original idea was suggested 
by Bill Fenner.

This type of format has proven quite valuable.  For example, if a whole
COLUMN is red then the sdr-monitor participant is likely having local 
problems.  If a whole ROW is red (there has to be at least one green
since someone has to see the session) this suggests the sdr-monitor 
participant is advertising a session but the outside world cannot see 
it.  

In any case, take a look at the page and send us any suggestions.

And as always, if you are interested in participating in the effort,
send me email...  it is very simple.  You'll be expected to add some 
tcl/tk code to your sdr.tcl file.  BTW, if you aren't running sdr,
you can't participate at this time...  that's the only mechanism we
have to collecting session information.

-Kevin Almeroth & Kamil Sarac



From rem-conf Thu Mar 25 09:47:47 1999 
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Date: Thu, 25 Mar 1999 09:43:36 -0800
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From: Florissa Colina <florissa@bmrc.berkeley.edu>
Subject: 3/31 Berkeley Multimedia, Interfaces and Graphics Seminar
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Berkeley Multimedia, Interfaces and Graphics Seminar


How did IP Multicast Get So Complicated? 


Wednesday March 31, 1999

1:00-2:30 p.m. 405 Soda Hall 

<italic>Please note the time frame of the seminar series has changed!!

</italic>

Mark Handley

AT&T Center for Internet Research at ISCI (ACIRI)


To understand where IP Multicast looks like it's going, you need to have
some understanding of PIM-DM, PIM-SM, MSDP, BGMP, MBGP, MZAP, MADCAP, AAP
and MASC. And that's before you even consider anything at the transport
or application layers. It is not surprising then that people have started
to question the underlying assumptions about what problem we are
attempting to solve, and alternative proposals such as Express and
"Simple Multicast" have recently been proposed. 


In this talk I will try to go through the evolution of IP multicast,
explaining briefly how these protocols work and how they fit together.
I'll also touch on the two proposed alternatives and explain why I
believe the "traditional" approach is still desirable despite the
additional mechanism needed to support it. No prior knowledge of
multicast routing or address allocation will be assumed. 


The seminar is broadcast on the Internet Mbone. The addresses are video
224.2.163.7/57076 and audio 224.2.147.61/27202. 


Sponsored by the Berkeley Multimedia Research Center 




____________________________________________


Florissa Colina

Berkeley Multimedia Research Center (BMRC)

http://bmrc.berkeley.edu/

626 Soda Hall #1776

University of California

Berkeley, CA 94720-1776

Phone: (510) 643-0800   FAX: (510) 642-5615  

Email: florissac@bmrc.berkeley.edu



From rem-conf Fri Mar 26 03:18:31 1999 
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          id <g.11554-0@bells.cs.ucl.ac.uk>; Fri, 26 Mar 1999 11:13:49 +0000
To: rem-conf@es.net
Subject: Re: Slides from the Minneapolis AVT meeting
In-reply-to: Your message of "Tue, 23 Mar 1999 17:36:41 GMT." <29034.922210601@cs.ucl.ac.uk>
Date: Fri, 26 Mar 1999 11:13:47 +0000
Message-ID: <1047.922446827@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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--> Colin Perkins writes:
>I've started to make a collection of the presentation slides from the AVT
>meeting in Minneapolis which is available on the web at
>	http://www-mice.cs.ucl.ac.uk/multimedia/misc/avt/
>Could all those who presented please send me a copy of their slides so I
>can include them here and in the minutes.

All the slides presented in AVT are now online at the above URL, with the
exception of Jon Crowcroft's presentation where there doesn't exist an
electronic copy.

Colin



From rem-conf Fri Mar 26 18:50:23 1999 
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From: <coolguy74@eastmail.com>
Subject: HUMAN GROWTH HORMONES AD
Date: Sun, 4 Apr 1999 09:30:54
Message-Id: <E10Qj5S-0001ZY-00@mail1.es.net>
Bcc:
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Human Growth Hormones

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From rem-conf Tue Mar 30 09:42:20 1999 
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Date: Tue, 30 Mar 1999 09:25:43 -0800
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From: Florissa Colina <florissa@bmrc.berkeley.edu>
Subject: 3/31 Berkeley Multimedia, Interfaces and Graphics Seminar
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Berkeley Multimedia, Interfaces and Graphics Seminar


How did IP Multicast Get So Complicated? 


Wednesday March 31, 1999

1:00-2:30 p.m. 405 Soda Hall 

<italic>Please note the time frame of the seminar series has changed!!

</italic>

Mark Handley

AT&T Center for Internet Research at ISCI (ACIRI)


To understand where IP Multicast looks like it's going, you need to have
some understanding of PIM-DM, PIM-SM, MSDP, BGMP, MBGP, MZAP, MADCAP, AAP
and MASC. And that's before you even consider anything at the transport
or application layers. It is not surprising then that people have started
to question the underlying assumptions about what problem we are
attempting to solve, and alternative proposals such as Express and
"Simple Multicast" have recently been proposed. 


In this talk I will try to go through the evolution of IP multicast,
explaining briefly how these protocols work and how they fit together.
I'll also touch on the two proposed alternatives and explain why I
believe the "traditional" approach is still desirable despite the
additional mechanism needed to support it. No prior knowledge of
multicast routing or address allocation will be assumed. 


The seminar is broadcast on the Internet Mbone. The addresses are video
224.2.163.7/57076 and audio 224.2.147.61/27202. 


Sponsored by the Berkeley Multimedia Research Center 




____________________________________________


Florissa Colina

Berkeley Multimedia Research Center (BMRC)

http://bmrc.berkeley.edu/

626 Soda Hall #1776

University of California

Berkeley, CA 94720-1776

Phone: (510) 643-0800   FAX: (510) 642-5615  

Email: florissac@bmrc.berkeley.edu



From rem-conf Tue Mar 30 12:07:59 1999 
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Date: Tue, 30 Mar 1999 14:02:16 -0600 (CST)
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To: atm@sun.COM, members@farnet.ORG, nren-discuss@psi.COM, rem-conf@es.NET,
        wireless@tandem.COM
Subject: ICCCN99 CFP
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------------------------------------------------------------------------------
(This cfp appeared in the CACM and IEEE Computer magazines, Feb. 1999 issues.
 Note that conference dates have been changed slightly - by one day!!)
------------------------------------------------------------------------------

                      IEEE IC3N'99  CALL FOR PAPERS
                    EIGHTH INTERNATIONAL CONFERENCE ON
                   COMPUTER COMMUNICATIONS AND NETWORKS
                 
                October 11 -- 13, 1999, Crowne Plaza Hotel
                    Boston - Natick, Massachusetts, USA 

         Sponsored by IEEE Communication Society (tech. co-sponsor),
                      NOKIA and Army Research Office.
     In Cooperation with The Center For Telecommunications Studies-USL 
                and Computer Science Telecommunications-UMKC
             (pending approval of sponsorships/in cooperations)

ICCCN is a major international forum to present original and fundamental 
advances in the field of Computer Communications and Networks. It also serves
to foster communication among researchers and practitioners working in a wide 
variety of scientific areas with a common interest in improving Computer 
Communications and Networks.

SCOPE: 

The primary focus of the conference is on new and original research results
in the areas of design, implementation and applications of Computer 
Communications and Networks. We invite you to submit papers that address 
novel, challenging, and innovative results. The topics include, but are not 
limited to:

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

SUBMISSION:

Authors are invited to submit complete and original papers. Papers that may 
be submitted should not have been previously published in another forum, or 
are not currently under review by another journal or conference. All submitted
papers will be refereed for quality, correctness, originality and relevance.  
The program committee reserves the right to accept a submission as long, 
short, or poster presentation. Of particular interest are papers which address
experiences with concrete Computer Communications and applications. All 
accepted papers will be published in the conference proceedings. Special 
issues of journals containing outstanding papers from the conference are being
planned. Manuscripts should include an abstract and be limited to 5000 words.
Submissions should include the title, author(s), author's affiliation, e-mail 
address, fax number and postal address.  In case of multiple authors, an 
indication of which author  is responsible for correspondence and preparing 
the camera ready paper for the proceedings should also be included. Electronic
submission is strongly encouraged (go to the end of this CFP for more 
information; ps or pdf format is preferred). Six copies of the manuscript 
should be submitted by Friday, May 14, 1999 to either of the program co-chairs:

  Dr. Sudhir S. Dixit                 Professor Arun K. Somani
  Nokia Research Center               Dept of Electrical and Computer Eng
  3 Burlington Woods Drive            Iowa State University 
  Burlington, MA 01803-4514, USA      3227 Coover Hall, Ames, IA, 50011-3060
  sudhir.dixit@research.nokia.com     arun@iastate.edu  
  (781)238-4915; (781)359-5196(fax)   (515) 294-0442; (515) 294-8432(fax)

IMPORTANT DATES:
                 Paper submission deadline :  May  14, 1999 
                 Notification of acceptance:  July 16, 1999
                 Camera ready papers due   :  August 16, 1999 

WORKSHOPS and TUTORIALS:

Proposals are solicited for organizing workshops and tutorials. Please send 
your proposals by April 24, 1999 to one of the General Chairs, Prof. Niki 
Pissinou}, Center For Advanced Computer Studies, University of Southwestern
Louisiana, Lafayette, LA 70504, USA, pissinou@cacs.usl.edu, Tel: (318) 482-
6604, Fax:(318) 482-6604 or Prof. Kia Makki, Center for Telecommunications 
Studies, University of Southwestern Louisiana, Lafayette, LA 70504, USA,
kia@usl.edu}, Tel: (318) 482-5872, Fax: (318) 482-6687. For all other 
information please contact Prof. E. K. Park, Computer Science and 
Telecommunications Program, University of Missouri, 4747 Troost Ave, Room 207,
Kansas City, MO 64110 USA, ekpark@cstp.umkc.edu, Tel: (816) 235-1497, Fax: 
(816) 235-5159 or send email to ic3n@cacs.usl.edu.

General Chairs: Kia Makki, USL and N. Pissinou, USL
Program Chairs: S. Dixit, NOKIA and A. Somani, ISU
Registration Chair: E. K. Park, U. of Missouri
More information about the conference: www.icccn99.cstp.umkc.edu 

We are currently setting up electronic paper submission procedures (will be 
ready soon), so please visit ICCCN99 web site www.icccn99.cstp.umkc.edu for 
more up-to-date information !!









From rem-conf Tue Mar 30 12:46:47 1999 
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Date: Tue, 30 Mar 1999 14:34:33 -0600 (CST)
From: Eun Kyo Park <ekpark@cstp.umkc.edu>
Message-Id: <199903302034.OAA06946@helios.cstp.umkc.edu>
To: alg@comm.toronto.edu, apc@ee.nthu.edu.tw, apc_members@hornbill.ee.nus.sg,
        cabernet-general@newcastle.ac.uk, ccrc@dworkin.wustl.edu,
        cellular@comnets.rwth-aachen.de, cnom@maestro.bellcore.com,
        commsoft@cc.bellcore.com, comsoc-chapters@ieee.org,
        comsoc-gicb@ieee.org, comsoc.tac@tab.ieee.org, comswtc@gmu.edu,
        cost237-transport@comp.lancs.ac.uk, ctc-members@redbank.tinac.com,
        dbworld@cs.wisc.edu, enternet@bbn.com, f-troup@codex.cis.upenn.edu,
        fokus-user@fokus.gmd.de, g-troup@ccrc.wustl.edu, giga@tele.pitt.edu,
        hipparch@sophia.inria.fr, ieee_rtc_list@cs.tamu.edu,
        ieeetcpc@ccvm.sunysb.edu, isadsoc@fokus.gmd.de, itc@fokus.gmd.de,
        kia@usl.edu, multicomm@cc.bellcore.com, osimcast@bbn.com,
        performance@haven.epm.ornl.gov, rem-conf@es.net, reres@laas.fr,
        sb.all@ieee.org, sc6wg4@ntd.comsat.com, sigmedia@bellcore.com,
        tccc@ieee.org, xtp-relay@cs.concordia.ca
Subject: ICCCN99 CFP
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------------------------------------------------------------------------------
(This cfp appeared in the CACM and IEEE Computer magazines, Feb. 1999 issues.
 Note that conference dates have been changed slightly - by one day!!)
------------------------------------------------------------------------------

                      IEEE IC3N'99  CALL FOR PAPERS
                    EIGHTH INTERNATIONAL CONFERENCE ON
                   COMPUTER COMMUNICATIONS AND NETWORKS
                 
                October 11 -- 13, 1999, Crowne Plaza Hotel
                    Boston - Natick, Massachusetts, USA 

         Sponsored by IEEE Communication Society (tech. co-sponsor),
                      NOKIA and Army Research Office.
     In Cooperation with The Center For Telecommunications Studies-USL 
                and Computer Science Telecommunications-UMKC
             (pending approval of sponsorships/in cooperations)

ICCCN is a major international forum to present original and fundamental 
advances in the field of Computer Communications and Networks. It also serves
to foster communication among researchers and practitioners working in a wide 
variety of scientific areas with a common interest in improving Computer 
Communications and Networks.

SCOPE: 

The primary focus of the conference is on new and original research results
in the areas of design, implementation and applications of Computer 
Communications and Networks. We invite you to submit papers that address 
novel, challenging, and innovative results. The topics include, but are not 
limited to:

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

SUBMISSION:

Authors are invited to submit complete and original papers. Papers that may 
be submitted should not have been previously published in another forum, or 
are not currently under review by another journal or conference. All submitted
papers will be refereed for quality, correctness, originality and relevance.  
The program committee reserves the right to accept a submission as long, 
short, or poster presentation. Of particular interest are papers which address
experiences with concrete Computer Communications and applications. All 
accepted papers will be published in the conference proceedings. Special 
issues of journals containing outstanding papers from the conference are being
planned. Manuscripts should include an abstract and be limited to 5000 words.
Submissions should include the title, author(s), author's affiliation, e-mail 
address, fax number and postal address.  In case of multiple authors, an 
indication of which author  is responsible for correspondence and preparing 
the camera ready paper for the proceedings should also be included. Electronic
submission is strongly encouraged (go to the end of this CFP for more 
information; ps or pdf format is preferred). Six copies of the manuscript 
should be submitted by Friday, May 14, 1999 to either of the program co-chairs:

  Dr. Sudhir S. Dixit                 Professor Arun K. Somani
  Nokia Research Center               Dept of Electrical and Computer Eng
  3 Burlington Woods Drive            Iowa State University 
  Burlington, MA 01803-4514, USA      3227 Coover Hall, Ames, IA, 50011-3060
  sudhir.dixit@research.nokia.com     arun@iastate.edu  
  (781)238-4915; (781)359-5196(fax)   (515) 294-0442; (515) 294-8432(fax)

IMPORTANT DATES:
                 Paper submission deadline :  May  14, 1999 
                 Notification of acceptance:  July 16, 1999
                 Camera ready papers due   :  August 16, 1999 

WORKSHOPS and TUTORIALS:

Proposals are solicited for organizing workshops and tutorials. Please send 
your proposals by April 24, 1999 to one of the General Chairs, Prof. Niki 
Pissinou}, Center For Advanced Computer Studies, University of Southwestern
Louisiana, Lafayette, LA 70504, USA, pissinou@cacs.usl.edu, Tel: (318) 482-
6604, Fax:(318) 482-6604 or Prof. Kia Makki, Center for Telecommunications 
Studies, University of Southwestern Louisiana, Lafayette, LA 70504, USA,
kia@usl.edu}, Tel: (318) 482-5872, Fax: (318) 482-6687. For all other 
information please contact Prof. E. K. Park, Computer Science and 
Telecommunications Program, University of Missouri, 4747 Troost Ave, Room 207,
Kansas City, MO 64110 USA, ekpark@cstp.umkc.edu, Tel: (816) 235-1497, Fax: 
(816) 235-5159 or send email to ic3n@cacs.usl.edu.

General Chairs: Kia Makki, USL and N. Pissinou, USL
Program Chairs: S. Dixit, NOKIA and A. Somani, ISU
Registration Chair: E. K. Park, U. of Missouri
More information about the conference: www.icccn99.cstp.umkc.edu 

We are currently setting up electronic paper submission procedures (will be 
ready soon), so please visit ICCCN99 web site www.icccn99.cstp.umkc.edu for 
more up-to-date information !!









From rem-conf Wed Mar 31 04:32:23 1999 
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From: "Andrea Basso" <basso@research.att.com>
To: <mp4-sys@fzi.de>, <gen-sys@fzi.de>, <confctrl@isi.edu>,
	<rem-conf@es.net>
Subject: Packet Video '99 Workshop - New York 26-27 April 1999
Date: Wed, 31 Mar 1999 07:27:59 -0500
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Registration for the 

       Packet Video '99 Workshop  
New York 26-27 April 1999 is now open.


Please visit

http://www.research.att.com/~mrc/PacketVideo99.html

for more information.


Andrea Basso, AT&T Labs Research
Room 3-219
100 Shultz Drive, Red Bank,NJ 07701
ph:  (732) 345-3302
fax  (732) 345-3033
basso@research.att.com






From rem-conf Wed Mar 31 08:15:32 1999 
From rem-conf-request@es.net Wed Mar 31 08:15:31 1999
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From: "Graeme Gibbs" <gibbs@nortelnetworks.com>
To: "'rem-conf@es.net'" <rem-conf@es.net>
Cc: "Rade Gvozdanovic" <rade@nortelnetworks.com>, 
    "Rajakulasingam Raviraj" <raviraj@nortelnetworks.com>, 
    "'Chaoxin.Qiu@mail.sprint.com'" <Chaoxin.Qiu@mail.sprint.com>
Subject: Additional Named Signal Events
Date: Wed, 31 Mar 1999 17:02:51 +0100
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To whoever is editting/championing "RTP Payloads for Telephone Signal
Events"

Firstly I have not been following the mailing list so apologies if this
topic is old & cold.

Basically I believe this draft would benefit from the addition of 2 extra
Named Signal Events ie 
2100Hz tones (ITU G.164)&
2100Hz tone with Phase reversals (ITU G.165)

Rationale:
These tones, while not used by traditional TDM switches per se, are used by
echo cancellers within the TDM network to turn off echo cancellation for
calls carrying modems & faxes.
Certain carriers may wish to build telephony over IP transit networks,
where, at least initially, modem/faxes are carried transparently as G.711
(64kbit/s) packets over IP
(yes I know this amounts to data over voice over data, but it may be some
period before all gateways support all modulation techniques, and carriers
must provide equivalence to current Public Switched Telephone Network from
Day 1). Modems/faxes sending such tones would expect both the intervening IP
network and prior/subsequent echo cancellers in PSTN to moderate their
behaviour accordingly.

Take  a scenario where the VOIP gateways are running voice calls as
compressed to G.729, with silence suppression and echo cancellation, and
such a 2100Hz tone is detected
The local GW must: Upspeed to 64 kbit/s & turn off silence suppression, and
echo cancellation
The remote GW must also: Upspeed to 64 kbit/s & turn off silence
suppression, and echo cancellation
The tone must be propagated back into the PSTN by remote GW

2100Hz tones will not reliably propagate through G.729, therefore if the
system  relied on passing  this tone in-band, the near-end modem would have
to detect tone , decide to upspeed (maybe local CAC decisions), and then
execute upspeeding before far end GW can begin to detect tones in the stream
>from the IP network- this may take too long. This method also relies on
codec that can receive one codec type and transmit in another (desirable BUT
not always the case)
However if the near-end GW were to send a Named Signal Event once it had
detected the tone, this would allow CAC and upspeeding to occur in parallel,
together with a correct tone being enforced to the PSTN

Any replies, please mail direct to me as I do not subscribe to avt list







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From: "Graeme Gibbs" <gibbs@nortelnetworks.com>
To: "'rem-conf@es.net'" <rem-conf@es.net>
Subject: RE: Additional Named Signal Events
Date: Wed, 31 Mar 1999 19:38:09 +0100
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		-----Original Message-----
		From:	Gibbs, Graeme [HAL02:HQ20-I:EXCH] 
		Sent:	Wednesday, March 31, 1999 5:03 PM
		To:	'rem-conf@es.net'
		Cc:	Gvozdanovic, Rade [HAL02:HQ20-M:EXCH]; Raviraj,
Rajakulasingam [HAL02:HF10-I:EXCH]; 'Chaoxin.Qiu@mail.sprint.com'
		Subject:	Additional Named Signal Events

		To whoever is editting/championing "RTP Payloads for
Telephone Signal Events"

		Firstly I have not been following the mailing list so
apologies if this topic is old & cold.

		Basically I believe this draft would benefit from the
addition of 2 extra Named Signal Events ie 
		2100Hz tones (ITU G.164)&
		2100Hz tone with Phase reversals (ITU G.165)

		Rationale:
		These tones, while not used by traditional TDM switches per
se, are used by echo cancellers within the TDM network to turn off echo
cancellation for calls carrying modems & faxes.
		Certain carriers may wish to build telephony over IP transit
networks, where, at least initially, modem/faxes are carried transparently
as G.711 (64kbit/s) packets over IP
		(yes I know this amounts to data over voice over data, but
it may be some period before all gateways support all modulation techniques,
and carriers must provide equivalence to current Public Switched Telephone
Network from Day 1). Modems/faxes sending such tones would expect both the
intervening IP network and prior/subsequent echo cancellers in PSTN to
moderate their behaviour accordingly.

		Take  a scenario where the VOIP gateways are running voice
calls as compressed to G.729, with silence suppression and echo
cancellation, and such a 2100Hz tone is detected
		The local GW must: Upspeed to 64 kbit/s & turn off silence
suppression, and echo cancellation
		The remote GW must also: Upspeed to 64 kbit/s & turn off
silence suppression, and echo cancellation
		The tone must be propagated back into the PSTN by remote GW

		2100Hz tones will not reliably propagate through G.729,
therefore if the system  relied on passing  this tone in-band, the near-end
modem would have to detect tone , decide to upspeed (maybe local CAC
decisions), and then execute upspeeding before far end GW can begin to
detect tones in the stream from the IP network- this may take too long. This
method also relies on codec that can receive one codec type and transmit in
another (desirable BUT not always the case)
		However if the near-end GW were to send a Named Signal Event
once it had detected the tone, this would allow CAC and upspeeding to occur
in parallel, together with a correct tone being enforced to the PSTN

		Any replies, please mail direct to me as I do not subscribe
to avt list



		



From rem-conf Wed Mar 31 21:37:06 1999 
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From: vishwas <vishwas@future.futsoft.com>
To: "'Joseph Serra'" <joeserra@WORLDNET.ATT.NET>,
        "IPMULTICAST@stardust.com" <IPMULTICAST@stardust.com>
Cc: mbone <mbone@ISI.EDU>, rem-conf <rem-conf@es.net>
Subject: RE: Who is offering multicast commercially?
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Hi All,
 Great News for me. I have been long discussing this issue with my ISP   
"PSI Net" based in US.  I am not aware wether Huges can provide "native   
multicast", if atleast they can provide us with a DVMRP tunnel it is just   
great for us.

we care connected to PSI net like this,


FS---(fire wall)--64kbps leased line--(fire wall)--FCS--PSI Net-->Other   
Public/Private Peers.

What we want is a tunnel to a CISCO router in FS from a peering point   
>from PSI Net,
As I made sure PSI Net itself doesent provide Multicast Services.

Please check out this image and let me know any body else out there can   
help

http://www.arl.mil/ARL-Directorates/ASHPC/HPCD/MBONE/mbone-topology.gif

Thanks
vishu

 -----Original Message-----
From: Joseph Serra [SMTP:joeserra@WORLDNET.ATT.NET]
Sent: Thursday, April 01, 1999 3:56 AM
To: IPMULTICAST@stardust.com
Subject: Re: Who is offering multicast commercially?

Hughes Network Systems Probably is the Best Multicasting service provider
out there.

They'rr in Germantown, MD and can provide service anywhere in the USA.

Joe Serra



 ----- Original Message -----
From: Alejandro Avella <aavella@ADVTECH.USWEST.COM>
To: <IPMULTICAST@stardust.com>
Sent: Wednesday, March 31, 1999 10:49 AM
Subject: Re: Who is offering multicast commercially?


> Bernhard Friedl wrote:
>
> >  Does anyone know other
> > network providers that offer multicasting as a commercial service?
>
> It seems that only 8 ISPs do it today. See here:
>
http://www.ipmulticast.com/ipmi_dir//dc_indexes//category_to_product_names  
/6
.html
>
> For more info look here:
> http://www.ipmulticast.com/ipmi_dir/
>
> Alejandro  



