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Date: Fri, 31 Jan 1997 23:11:07 -0800
Message-Id: <199702010711.XAA28679@stilton.cisco.com>
From: Dino Farinacci <dino@cisco.com>
To: mleelani@cisco.com
CC: casner@precept.com, mleelani@cisco.com, rem-conf@es.net
In-reply-to: <199702010327.TAA03048@chow.cisco.com> (message from Manoj Leelanivas on Fri, 31 Jan 1997 19:27:30 +73600 (PST))
Subject: Re: Minutes for AVT WG at San Jose IETF

>> Yes, it has to keep a small source cache(S, G and the rate) to
>> implement the way the protocol specifies it. Still the baggage is much
>> lesser than the (S, G) entry which needs to keep track of the outgoing
>> interfaces and such.

    Just to be clear, that is one way to implement it. We (cisco) did not
    implement it this way.

Dino

From rem-conf-request@es.net Sat Feb 01 02:12:44 1997 
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From: Dino Farinacci <dino@cisco.com>
To: casner@precept.com
CC: mleelani@cisco.com, rem-conf@es.net
In-reply-to: <Pine.WNT.3.95.970131193514.-230513J-100000@oak.precept.com> (message from Stephen Casner on Fri, 31 Jan 1997 19:48:16 -0800 ())
Subject: Re: Minutes for AVT WG at San Jose IETF

>> And, it's important to make the distinction that it is only the leaf
>> router which keeps this cache, and only for the sources sending
>> through that leaf router.  This compares to (S,G) routing state that
>> would have to be kept in all the routers involved in the source-rooted
>> trees, some of which would have to have state for all 20000 sources.

    And the RP for the same reasons.

Dino

From rem-conf-request@es.net Sat Feb 01 11:48:16 1997 
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From: Videoduck1@aol.com
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To: rem-conf@es.net
Subject: UK ISDN Videoconferencing Demonstration Service

Useful UK ISDN Videoconferencing Test/Demo Numbers and information can be
found at:
http://members.aol.com/videoduck1/

Hope it's useful.

From rem-conf-request@es.net Sat Feb 01 11:55:11 1997 
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[This is a courtesy copy of an article posted to Usenet via Deja News]

For info check out the UK ISDN Videoconferencing Demonstration Service Website at:
http://members.aol.com/videoduck1

-------------------==== Posted via Deja News ====-----------------------
      http://www.dejanews.com/     Search, Read, Post to Usenet

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From rem-conf-request@es.net Mon Feb 03 01:49:42 1997 
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From: Manoj Leelanivas <mleelani@cisco.com>
Message-Id: <199702030648.WAA29947@chow.cisco.com>
Subject: Re: Minutes for AVT WG at San Jose IETF
To: casner@precept.com (Stephen Casner)
Date: Sun, 2 Feb 1997 22:48:56 -0800 (PST)
Cc: mleelani@cisco.com, rem-conf@es.net
In-Reply-To: <Pine.WNT.3.95.970131193514.-230513J-100000@oak.precept.com> from "Stephen Casner" at Jan 31, 97 07:48:16 pm
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=>    
=>    > Yes, it has to keep a small source cache(S, G and the rate) to
=>    > implement the way the protocol specifies it. Still the baggage is much
=>    > lesser than the (S, G) entry which needs to keep track of the outgoing
=>    > interfaces and such.
=>    
=>    And, it's important to make the distinction that it is only the leaf
=>    router which keeps this cache, and only for the sources sending
=>    through that leaf router.  This compares to (S,G) routing state that
=>    would have to be kept in all the routers involved in the source-rooted
=>    trees, some of which would have to have state for all 20000 sources.
=>  

Yes, and also at the RP as Dino mentioned earlier.
  

=>    > The RP which is switching RTP traffic also may have sources which are
=>    > sending RTP streams interrmittently. Even otherwise, it has to switch
=>    > the traffic for the active source(ie, it is indeed loaded to an extent though
=>    > switching path may be much faster).
=>    
=>    But doesn't that load get moved off to a source-rooted tree?
=>    


Since one of the receivers may have a higher threshold, the RP may be
still forwarding packets through the shared tree. Or there could be a
second lower rate source, which could contribute to the load at the
RP. But, as you observed there is no concrete justification yet for
separate groups instead of 1.

-Manoj
 

From rem-conf-request@es.net Mon Feb 03 11:22:00 1997 
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From: Dino Farinacci <dino@cisco.com>
To: mleelani@cisco.com
CC: casner@precept.com, mleelani@cisco.com, rem-conf@es.net
In-reply-to: <199702030648.WAA29947@chow.cisco.com> (message from Manoj Leelanivas on Sun, 2 Feb 1997 22:48:56 -0800 (PST))
Subject: Re: Minutes for AVT WG at San Jose IETF

>> Since one of the receivers may have a higher threshold, the RP may be
>> still forwarding packets through the shared tree. Or there could be a
>> second lower rate source, which could contribute to the load at the
>> RP. But, as you observed there is no concrete justification yet for
>> separate groups instead of 1.

    Or a branch of the source distribution tree flows through the RP.

Dino

From rem-conf-request@es.net Mon Feb 03 12:40:41 1997 
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Date: Mon, 3 Feb 1997 09:39:59 -0800
To: rem-conf@es.net
From: alagu@mailman.apple.com (Alagu Periyannan)
Subject: Re: MBone teleconf tools for Macs?

At 6:31 PM 1/30/97, Steve Deering wrote:
>
>> There is a tool called "Maven" available on the internet
>> that will interoperate with VAT.
>
>Last I knew, Maven supported point-to-point (i.e., unicast) operation only.
>

Yes. You're probably right. I think it uses the old MacTCP API.
If it is moved to use the OpenTransport API then multicast support
can be added very easily. I think parts of Maven were incorporated
into CUSeeMe. I'm not sure who maintains the standalone Maven nowadays.
Anyone knows?




Alagu Periyannan                   alagu@mailman.apple.com

Communications Products and Technology
Apple Computer, Inc.



From rem-conf-request@es.net Mon Feb 03 14:07:30 1997 
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Subject: [Reminder] Berkeley Multimedia & Graphics Seminar

          Berkeley Multimedia and Graphics Seminar 
   (Wednesday February 5, 1997 12:30-2:00 PDT 405 Soda Hall) 

"The Impact of Future Developments in Flat Panel and Projection Display" 

                        Dave Mentley 
                Stanford Resources, Inc. 
                        San Jose, CA 

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

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


From rem-conf-request@es.net Mon Feb 03 14:11:11 1997 
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From: Kathy Lynn Hewitt <klhewitt@eos.ncsu.edu>
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Date: Mon, 3 Feb 1997 14:10:18 -0500
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Subject: wb crashes
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Hi, I've got a new SUN Ultra running Solaris 2.5 to test out.  I've been
working with the mbone tools on it and am running into problems with the wb
program.  When wb is started from the command line or through sdr for any
multicast session, it exits out and crashes with a segmentation fault when the
"receive only" box is unchecked.  You can view a session just fine, but you
can't add to the whiteboard.  I've seen this problem on another SUN machine we
have running Solaris 2.4, but then again on my SUN (2.4), wb works just fine -
no segmentation faults or anything.  Also, on the "faulty" machines, I can
start a unicast session and I don't run into any problems when I check the
"receive only" button off.

When I try running wbimport on one of the 2.4 machines, it appears to be locked
up - none of the buttons work and you can't import anything.  Then again, it
works fine on a different 2.4 machine.

Has anybody experienced this before or know of anything that I might try?

Thanks for your help!

--Kathy Hewitt

-- 
-------------------------------------------------
Kathy Hewitt
Dept. of Electrical and Computer Engineering
North Carolina State University
<klhewitt@eos.ncsu.edu>
919-515-5604
-------------------------------------------------


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Subject: Topics in Geometry and Physics

	MBone Broadcast Announcement
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To: rem-conf@es.net
From: mbone <mbone@euclid.ucsd.edu>
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Subject: UCSD, Mathematics - Lecture Series

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

Title:       
	UCSD, Mathematics - Lecture Series
Date:        
	Feb 04, 1997

Time:        
	15:00 PST8PDT 2 hours

Contact:     
	mbone@math.ucsd.edu

URL:         
	http://math.ucsd.edu/mbone/

Description:        
	Special Topics  Prof. Harvey Friedman  "The Formalization of Mathematics"  Lecture notes for this presentation are available at  http://math.ucsd.edu/mbone/freidman/talk1.html









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

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Reply-to: mbone@euclid.ucsd.edu
Subject: UCSD, Mathematics - Lecture Series

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

Title:       
	UCSD, Mathematics - Lecture Series
Date:        
	Feb 05, 1997

Time:        
	11:00 PST8PDT 2 hours

Contact:     
	mbone@math.ucsd.edu

URL:         
	http://math.ucsd.edu/mbone/

Description:        
	Special Topics  Prof. Harvey Friedman  "The Formalization of Mathematics"  Lecture notes for this presentation are available at  http://math.ucsd.edu/mbone/freidman/talk1.html









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

From rem-conf-request@es.net Tue Feb 04 15:03:33 1997 
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From: John H Wilson <John_H_Wilson@ccm.jf.intel.com>
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To: rem-conf@es.net
Subject: Encryption info in SDP

draft-ietf-mmusic-sdp-02.txt defines a key field in a session 
announcement. For "public" announcements of conferences that 
require some form of registration, the field allows the 
specification of a URI at which to register. The "private" 
announcement that may be received as a result of this 
registration contains the key(s) that will be used on the 
RTP streams.

However, I cannot find any definition of how the encryption 
algorithm(s) are specified. 

Clearly, it would be possible to define an application extension 
field to so specify, but do the authors have a better way? 

john_h_wilson@ccm.jf.intel.com

From rem-conf-request@es.net Tue Feb 04 18:02:28 1997 
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Subject: Re: Encryption info in SDP
In-reply-to: Your message of "Tue, 04 Feb 1997 09:23:00 PST." <Tue, 04 Feb 97 12:01:38 PST_1@ccm.hf.intel.com>
Date: Tue, 04 Feb 1997 17:51:44 -0500
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>draft-ietf-mmusic-sdp-02.txt defines a key field in a session 
>announcement. For "public" announcements of conferences that 
>require some form of registration, the field allows the 
>specification of a URI at which to register. The "private" 
>announcement that may be received as a result of this 
>registration contains the key(s) that will be used on the 
>RTP streams.
>
>However, I cannot find any definition of how the encryption 
>algorithm(s) are specified. 

Do you mean how the HTTP server specifies an encryption key in an
HTTP reply?  This should be specified, but I haven't given the issue
any thought.  It shouldn't be in the SDP spec though, because it is
more general than just the context of SDP.  

My first thoughts are that a MIME content-type of application/key,
with a payload as specified in the RTP a/V profile for user-entered
pass phrases would be appropriate.

Mark



From rem-conf-request@es.net Wed Feb 05 13:57:28 1997 
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    humcomp@info.sims.berkeley.edu
From: Diane Harley <dianeh@uclink2.berkeley.edu>
Subject: From Wagner to Virtual Reality: A History of Multimedia/Randall Packer
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<bigger>A CyberSemester Lecture 


>From Wagner to Virtual Reality: A History of Multimedia

Randall Packer, Visiting Lecturer, Art Practice


Wednesday, February 5, 7:00-7:30PM

2050 Valley Life Sciences Building


</bigger><bigger>(http://socrates.berkeley.edu/~cybersem)



</bigger>*****************************************************


>From Wagner to Virtual Reality: A History of Multimedia, is a 
historical

account of the avant-garde artists and scientists who have pushed the

envelope of their respective disciplines to bring about the dissolution
of

boundaries that traditionally exist between the artistic and
technological

media. In the past twenty-five years, the intersection of art and

human-computer interactivity has emerged as a mass medium, triggering
new

forms of artistic, entertainment, and educational content for 
laserdisc,

CD-ROM, and the World Wide Web. This lecture follows the evolution of
the

various artistic and scientific disciplines that converge as multimedia,
a

critical overview of the past that provides a conceptual understanding
of

the digital medium, as well as a forum for informed discussion about 
the

implications for the future.


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


Randall Packer is one of San Francisco's pioneering multimedia artists,

composers and producers. Founder of Zakros InterArts / New Music Theatre
in

1988, he has produced seminal interdisciplinary performance artworks,
new

music events, and lectures by such 20th century groundbreakers as John

Cage, Karlheinz Stockhausen, Brian Eno, Laurie Anderson and Morton

Subotnick. He has also created original works exploring new forms of

interaction between live performance and the electronic media. Recently
he

completed two CD-ROMs of computer films and interactive media that were

premiered this year at the San Francisco International Film Festival 
and

the Mill Valley Film Festival.


As an educator, Randall Packer was the first instructor and former
director

of San Francisco State University's acclaimed Multimedia Studies
Program,

one of the most comprehensive multimedia programs in the nation. For 
the

past fifteen years, his work as an artist and historian has taken him

around the world where he has held residencies at several institutes

including Xerox PARC (Palo Alto Research Center), IRCAM (Institute for
the

Research and Coordination of Music) at the Centre Pompidou in Paris, 
and

CCRMA (Center for the Coordination and Research in Music and Acoustics)
at

Stanford University. Packer lectures extensively on multimedia theory
and

aesthetics. He is currently a visiting lecturer at the University of

California, Davis and the University of California, Berkeley, where he
is

teaching interdisciplinary art practices and the history of multimedia.

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




From rem-conf-request@es.net Wed Feb 05 14:36:16 1997 
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To: rem-conf@es.net
From: mbone-support <mbone-support@lists.stanford.edu>
Reply-to: mbone-support@lists.stanford.edu
Subject: The Stanford Channel

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

Title:       
	The Stanford Channel
Date:        
	Feb 05, 1997

Time:        
	14:00 GMT 3 hours

Contact:     
	mbone-support@lists.stanford.edu

URL:         
	http://www-51.stanford.edu/ch51/home.html

Description:        
	In order to support multicast testing and to   provide a consistent, high quality source of   audio and video Stanford University will be   transmitting the Stanford Channel over the mbone   M-F 2-5pm PST.









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

From rem-conf-request@es.net Wed Feb 05 17:02:13 1997 
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Date: Wed, 5 Feb 1997 17:00:51 -0500 (EST)
From: kevin@cc.gatech.edu (Kevin C. Almeroth)
Message-Id: <199702052200.RAA23761@flora.cc.gatech.edu>
To: rem-conf@es.net
Subject: MBone Community Size?


Okay, I've asked this question before...  how big is the MBone in terms of
actual numbers of users?

Well, here is my guess...

At least since the beginning of 1995 I've been collecting statistics
using Mlisten, a tool I wrote to capture the "control" packets sent by
vic and vat.  (See http://www.cc.gatech.edu/computing/Telecomm/mbone/
for more info)

So I've been collecting this data on and off for about 2 years...  sometimes
for individual sessions, sometimes for all sessions, sometime just for
audio, sometimes just for video.  All told, I've got several 100s of
Megabytes of data.

I've taken this data, processed it, sorted it, uniqued it, etc. and came
up with a list of 10,600 IP addresses.  So over the past couple of years
I've received at least one packet, via the MBone, from about this many
IP addresses.

Now there are obvious limitations to this sort of estimate, but it is the
best I've seen.  FYI.

Any comments?

-Kevin Almeroth


From rem-conf-request@es.net Wed Feb 05 18:54:10 1997 
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From: Deb Agarwal <deba@george.lbl.gov>
Message-Id: <199702052351.PAA03720@george.lbl.gov>
Subject: MBone session announcement
To: rem-conf@es.net
Date: Wed, 5 Feb 1997 15:51:19 -0800 (PST)
Cc: tonner@csd.uwm.edu (Brian Tonner)
Reply-To: DAAgarwal@lbl.gov
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        MBone Broadcast Announcement
        ----------------------------
 
Title:       
        LabWeb - The Spectro-Microscopy Collaboratory
Dates:        
        beginning - Feb 05, 1997
	ending    - indefinite (can be negotiated)
 
Contacts:     
	tonner@csd.uwm.edu
        DAAgarwal@lbl.gov
 
URL:         
        http://www-itg.lbl.gov/BL7Collab.html
 
Description:        

We have built a prototype collaboratory testbed with the University of
Wisconsin-Milwaukee that is allowing remote operation of a
sophisticated synchrotron-radiation beamline in the Advanced Light
Source (ALS) of Lawrence Berkeley National Laboratory. The Advanced
Light Source is one of the brightest source of soft X-ray beams in the
world today. The Spectro-Microscopy Facility consists of experimental
analytical instruments attached to the first undulator beamline of the
ALS, Beamline 7.0. Because of the unique capabilities of these
instruments the Spectro Microscopy project is a large collaboration
involving many institutions.
 
Telepresence is a significant component of the collaboratory
environment.  Telepresence includes remotely controllable robotic
cameras, a video switcher, and a videoconferencing session to transmit
the video feed from the cameras. We have three cameras located
throughout the ALS beamline floor space.  We will be transmitting vic
using h.261 format and vat using PCM format.  The transmission
bandwidth will be kept at a low rate when not in use for remote
experimentation.  We will be using a ttl of 127 to transmit this
session.

-- 
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Deb Agarwal                                  e-mail:DAAgarwal@lbl.gov
MS50B-2239                                   phone :(510)486-7078
Lawrence Berkeley National Lab  
Berkeley, CA 94530
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

From rem-conf-request@es.net Wed Feb 05 18:57:39 1997 
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From: "Carl Malamud [IMS]" <carl@also.media.org>
Message-Id: <199702052357.SAA22317@also.media.org>
Subject: Re: MBone Community Size?
To: kevin@cc.gatech.edu (Kevin C. Almeroth)
Date: Wed, 5 Feb 1997 18:57:01 -0500 (EST)
Cc: rem-conf@es.net
In-Reply-To: <199702052200.RAA23761@flora.cc.gatech.edu> from "Kevin C. Almeroth" at Feb 5, 97 05:00:51 pm
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> I've taken this data, processed it, sorted it, uniqued it, etc. and came
> up with a list of 10,600 IP addresses.  So over the past couple of years
> I've received at least one packet, via the MBone, from about this many
> IP addresses.
> 

That number seems about right.  To give some perspective, our
main machines at IMS saw 720,000 unique IP addresses during 1996.
Our live broadcasts would get a typical audience of 25-100 people
and I believe NASA peaks at ~1000.  Our typical download audience
for ftp'able files is something like 1000-2000 downloads per week
for our more popular programs, with that rate sustained pretty
well over the course of the year.

The numbers I've seen for RealAudio/Xing/.... are peak server
capacities of 10k-15k streams for the very largest sites.  I
don't believe any of them are close to that in terms of real
audiences and my guess is that the largest sites are looking at 
audiences of 1,000-5,000 streams at peak periods.

The discussion in the last week or so on the lists has been
very interesting.  On the one hand, the ISP's say they'll support
this when the demand is ready.  On the other hand, mbone 
developers tend to be clinging to the "this is an experiment
which must be carefully controlled" theory of software deployment.

I'm not sure I believe the mbone is really all that fragile
and distributed control seems to have worked very well for
things like the web, despite very real fears of backbone collapse.
My biggest fear for the mbone is that we control it so carefully
that the demand never happens, the ISP's never support it
properly, and the whole thing turns into a technically-correct
solution that is buried by sub-optimal unicast streaming
solutions.

Carl


From rem-conf-request@es.net Wed Feb 05 21:15:03 1997 
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Date: Wed, 5 Feb 1997 21:14:17 -0500 (EST)
From: kevin@cc.gatech.edu (Kevin C. Almeroth)
Message-Id: <199702060214.VAA11721@morticia.cc.gatech.edu>
To: carl@also.media.org
Subject: Re: MBone Community Size?
Cc: rem-conf@es.net

>>> I've taken this data, processed it, sorted it, uniqued it, etc. and came
>>> up with a list of 10,600 IP addresses.  So over the past couple of years
>>> I've received at least one packet, via the MBone, from about this many
>>> IP addresses.

>>That number seems about right.  To give some perspective, our
>>main machines at IMS saw 720,000 unique IP addresses during 1996.
>>Our live broadcasts would get a typical audience of 25-100 people
>>and I believe NASA peaks at ~1000.  

I think 1000 is a bit high.  I've tried to do a pretty accurate max
and the most I ever saw on one of the shuttle sessions was about
450 people.  Keep in mind that not all the people in the vat window
are actually group members.  Some who leave and don't send the final
vat-appl-layer "group leave" msg (or if it is lost) hang around in
the session list for another 15 mins or so.  Of course, I have
also missed a few missions so I'm not saying you're wrong

Also, the most I've every seen on the MBone (as a whole) at any one
time was during the LA IETF last year when the shuttle was also
going on and I think there was one other "interesting" session.  At

-Kevin Almeroth
that time I recall seeing just more than 600 total people.

From rem-conf-request@es.net Wed Feb 05 21:40:27 1997 
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Date: Wed, 05 Feb 1997 21:39:02 -0500
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
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I wonder how many of the 10,600 are "this looks cool, let's try this -
this is not ready for prime time and there's nothing interesting on -
let me go back to RealAudio or VRML or other neat toy". In other words,
how many addresses can be considered "regulars", who actually use this
on a more or less daily or at least regular basis? I have the suspicion
that Mbone "churn" may be pretty high; the core user community may be
closer to the 600 mentioned...

Part of the problem may also be that some of tools are not quite as
robust on the Wintel platform or not fully available on the
second/third-most common platform (Win 3.1/Apple) as on Unix.
-- 
Henning Schulzrinne         email: schulzrinne@cs.columbia.edu
Dept. of Computer Science   phone: +1 212 939-7042
Columbia University         fax:   +1 212 666-0140
New York, NY 10027          URL:   http://www.cs.columbia.edu/~hgs

From rem-conf-request@es.net Thu Feb 06 06:43:29 1997 
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To: kevin@cc.gatech.edu (Kevin C. Almeroth)
cc: rem-conf@es.net
Subject: Re: MBone Community Size?
In-reply-to: Your message of "Wed, 05 Feb 1997 17:00:51 EST." <199702052200.RAA23761@flora.cc.gatech.edu>
Date: Thu, 06 Feb 1997 11:41:40 +0000
Message-ID: <4396.855229300@cs.ucl.ac.uk>
From: Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>



we had 900 people listening to globecom......admittedley a lot via a
ntt server based vidceo distribution system but nevertheless on the internet....

on the mbone, a plot of mbone events against time:
 
(i may not have logged ALL mbone announcements) - starting from
aeround sep 93 - grepping out dates, and then plotting frequecny of
announce by month....over the last 50 or so months:

ftp://cs.ucl.ac.uk/darpa/mbone-event-grow.ps

From rem-conf-request@es.net Thu Feb 06 08:31:01 1997 
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To: rem-conf@es.net
Subject: Question on IP/UDP/RTP header compression ID
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If have a littel problem with the understanding of one section in
the ID for IP/UDP/RTP header compression. On page 3, section
3.1, the "FULL_HEADER" packet size ist explained:
	".. The 4 bit sequence numer defined in section 3.3 ...
	is carried in the data field ... as defined in section 
	5.3.2 ... of [3]"

Im not familar with IPv6 [3], so I do not understand, where to put 
the sequence number to :-) Does this mean, that it will be 
appended to the data following the IP header? If so, have the header
to be changed in any way ("no" in my understanding, because the 
receiver will remove it).

Any comments are welcome,

Chris


From rem-conf-request@es.net Thu Feb 06 13:34:26 1997 
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Date: Thu, 06 Feb 1997 18:57:46 +0100
To: congres@prism.uvsq.fr
From: ICCC'97 <iccc@prism.uvsq.fr>
Subject: ICCC'97

          Announcement and Call for Papers

International Conference for Computer Communications - ICCC'97

THEME: "Keys to a mature information society"

November 19-21, 1997

Cannes, France

The 13th biennal international conference organized by the International
Council for Computer Communications.

AIMS
The conference aims is to provide an integrated perspective and a timely
spectrum of computer and communications technologies in the context of
present and future applications.

TOPICS

NEW TECHNOLOGIES
Internetworking, Mobile Communications, Personal Wireless Communications,
High Performance Computing, ATM, Broadband Networks, Architecture and
Protocols

NEW APPLICATIONS
Desktop Video Conferencing, Broadband Applications e.g. Telemedicine,
Telecommuting, Distance learning, Video on Demand, Multimedia, Electronic
Commerce

NEW SERVICES
Intelligent Networks, Network Management, Virtual networks

TRENDS AND STRATEGIES
Information Highway, Information Filtering, Data Warehouse/Data Mining
Security/ Privacy/ Legal Issues, User Acceptance/Billing, Commercialising
Information

POLICY AND GOVERNMENTAL ISSUES
Information & communications policy, Control/Censorship Of Information,
Deregulation/Liberalisation, Security/Privacy/Legal Issues, Role Of
Government

TARGET PARTICIPANTS
Academics, accountants, advisors, auditors, consultants, doctors,
engineers, investors, lawyers, planners, managers, policy makers,
practitioners, researchers, scientists and others interested in
communications and computer networks.

CONFERENCE FORMAT
The conference will last three days, November 19-21, 1977.
Tutorials will be offered on November 17-18.

LOCATION
Cannes, a major mediterranean city on the French Riviera.

SPONSORS
ICCC (International Council for Computer Communications) and IFIP-TC.6
(International Federation of Information Processing, Technical Committee on
Data Communications)

ORGANIZER
Telecom Valley, with the support of CNET (Centre National d'Etudes des
T=E9l=E9communications), EURECOM, INRIA (Institut National de Recherche en
Informatique et Automatique)

Organizing Committee
A. Andr=E9 (TV)
E. Bibal (TV)
P Conruyt (FT-DGSA)
C. Gueguen (EURECOM)
C. Juncker (INRIA)
F. Jutand (CNET)
T. Laurent (TV)
G. Leb=E8gue (TV)
L. Pouzin (ICCC)

CONFERENCE CHAIR
Samir Tohm=E9 (F)
ENST
46 rue Barrault, 75634 Paris Cedex 13, France

PROGRAMME CHAIR
Guy Pujolle (F)
Lab. PRiSM, University of Versailles
45 avenue des Etats-Unis, 78 035 Versailles Cedex, France

PROGRAMME COMMITTEE

J. Arnbak, Delft Technical Univ. (NL)
F. Bar, Stanford Univ. (USA)
C. Blondia, University of Antwerp (B)
P. Brown, France Telecom - CNET (F)
K. Boyanov, CICT (BG)
A. Casaca, INESC (P)
O. Casals, Universitat Politecnica de Catalunya (SP)
P. Chemouil, France Telecom - CNET (F)
S. J. Chung, ETRI (KR)
J-P. Coudreuse, Mitsubishi (F)
W. Dabbous, INRIA (F)
Dang. N'Guyen, ENSTB (F)
S. Fdida, University Paris VI (F)
L. Fratta, Politecnico di Milano (I)
D. Gaiti, Technical University of Troyes (F)
C. Galand, IBM (F)
N.D. Georganas, University of Ottawa (CA)
A. Gravey, France Telecom - CNET (F)
G. Hebuterne, INT (F)
T. Housley, Housley Computer Communications (AU)
C. Huitema, Bellcore (USA)
P. Humblet, EURECOM (F)
P. Jackson, Qualcomm (USA)
F. Kamoun, ENSI (TU)
M. Kato, Xerox (J)
J. Kella, NET (IL)
D. Khakhar, Lunds Univ. (S)
P. K=FChn, Stuttgart Univ. (D)
R. Kung, France Telecom - CNET (F)
J. Labetoulle, EURECOM (F)
J-Y. Le Boudec, EPFL (SW)
O. Martikainen, Telecom Finland (FI)
N. Naffah, Sabre Group (F)
H. Perros, NCSU, (USA)
S. Ramani, National Center for Software Technology (IN)
J. Roberts, France Telecom - CNET (F)
P. Rolin, ENSTB (F)
O. Spaniol, Aachen Technical Univ. (D)
E. Stefferud, NMA, USA
Y. Takahashi, Aist-Nara (J)
L. Tarouco, Univ. Federale de Rio Grande del Sul (BR)
F. Tobagi, Stanford University (USA)
C. Wellekens, EURECOM (F)

KEY DATES IN 1997

* April  1st, papers received
* June 1st, papers accepted /rejected
* Sep 1st, papers received in camera ready form
* Nov 17-18, Tutorials
* Nov 19-21, ICCC'97

SUBMISSION OF PAPERS
Paper can be submitted by mail or email
* by mail: submit five copies of a full paper by April 1st, 1997 to
ICCC97 Conference
Lab PRiSM - University of Versailles
45 av. des Etats-Unis
78035 Versailles Cedex - France
fax +33 (0) 139 254 057
http://www.prism.uvsq.fr

* by email: poscript version of the paper to be sent to:=
 ICCC97@prism.uvsq.fr

Papers are invited on the conference theme and related topics. Authors must
state that their paper have neither been published before nor currently
being submitted elsewhere.

Full papers should be no longer than 15 pages, including tables, diagrams
and pictures. The font size must be 12 points or larger. The cover page
must contain an abstract of about 150 words, name and affiliation of
author(s) as well as the lead author's postal address, telephone number,
fax number and e-mail.

Papers will be evaluated on the basis of relevance, originality and
clarity. To ensure acceptance of high quality papers, each paper will be
double and blind refereed.

Accepted papers will be included in the conference proceedings and will be
distributed to the attendees at the conference.

The language of the conference is English and papers must be in this=
 language.

For more information, fill and return the following form:
ICCC97 Conference
905, rue Albert Einstein
BP 261
06 921 Sophia Antipolis Cedex
France

----------------------------------------------------------------------------
--------------
ICCC'97
1st Name: ...........................Last
Name:......................................
email:......................................................................
.........
Title:......................................................................
.........
Organization:...............................................................
.........
Postal
address:....................................................................=
..
............................................................................
..........
City:.......................................Postal
code:..............................
Country:....................................................................
..........
Telephone:..................................................................
...........
Fax:........................................................................
...........

Topics I am interested in :
...........................................................
O Topics I am interested in
:...........................................................
O I wish to attend the ICCC=9297 conference.
O I wish to present a paper and will submit a paper by Apil 15, 1997.=20
Title of the paper :
...................................................................
............................................................................
............
............................................................................
............
_____________________________________________________________________
_____________________________________________________________________


From rem-conf-request@es.net Thu Feb 06 14:36:37 1997 
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From: Christian Zahl <zahl@fokus.gmd.de>
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Date: Thu, 6 Feb 1997 20:35:13 +0100
Message-Id: <199702061935.UAA06419@tao.fokus.gmd.de >
X-Mailer: exmh version 1.6.1 5/23/95
To: rem-conf@es.net
Subject: Question on IP/UDP/RTP compression
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Another question: see the following scenario (for one session):
	o RTP packets were sent as COMPRESSED_RTP, so that a context
	  for this session exists.
	o the saved delta fields have been changed from the initial
	  values (e.g. IP ID delta = 5, bacause of loss)
	o one RTP packet was sent as FULL_HEADERS or COMPRESSED_UDP
	o the next packet should be send as COMPRESSED_RTP

The question is if the values in the context should be reset to their
initial values, or if the values remain unchanged? So if the delta
of the IP ID is 1 for the next packet to be send, do we have to
indicate the change (set the I bit and include the new delta) or
not? 

I think it makes sense to reset the values in the context to their 
initial values. This seems to be a "clean" resynchronisation...

Thanks,
Chris


From rem-conf-request@es.net Thu Feb 06 15:16:17 1997 
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From: Christian Zahl <zahl@fokus.gmd.de>
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To: rem-conf@es.net
Subject: IP/UDP/RTP compression and RTP payload-type
Mime-Version: 1.0
Content-Type: text/plain

1. PT changes compression

We are currently developing a method for transfering multiple
RTP streams over a low-bandwidth PPP connection. The implementaion
consists of an RTP mixer on both ends of the PPP link, which
adjust the payload-type field according to the bandwidth
available. So it can happen very often, that the payload (and so the
PT field also) changes to adjust to the current bandwidth available
(will measured while having by PPP link). Our major concern are
modem connections.

The current ID for IP/UDP/RTP compression doesn't have any compression
thechnique for changes of the PT field. This means, that every change
in the coding will result in sending an uncompressed RTP header
(i.e. >=12 bytes instead of 2+1). Because we have a mixer, we also
can assume to have a list of several CSRC entries, resulting in a
even larger packet!

Well it seems to be a little difficult to get a bit to indicate any
change in the PT field, but it would be very helpfull to it!
Otherwise it can happen, that the occupied bandwidth will be greater
when adjusting the payload, than if we don't do so! Not very helpfull.

I think it makes more sense to shorten the session context id to
7 bit and to add a P bit, which indicates a new octet with the
changed PT value.

2. CSRC change compression
Well as described above we use a mixer and expect several sources for
the outgoing media stream. The current ID says, that, when the CSRC
list changes, then the whole list has to be transmitted. Because there
are several sources and we can assume that they also change more 
frequenctly, I think that sending the whole list ist very poor!

I think if would be much better to send ONLY the CSRC entries which
have been changed, i.e. added or removed. Because the other side also
have knowledge of the current set of CSRC entries, it can decided
if the entry as to be add or the be removed. This results in much less
overhead than in the current ID. 

Only when the sum of the leaving and comming CSRC entries is larger
than the current CSRC list, there will be send more than in the
current ID.


Any comments are welcome,
Chris




From rem-conf-request@es.net Thu Feb 06 16:57:52 1997 
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Date: Thu, 6 Feb 1997 13:49:26 -0800 ()
From: Stephen Casner <casner@precept.com>
To: Christian Zahl <zahl@fokus.gmd.de>
cc: rem-conf@es.net
Subject: Re: several messages
In-Reply-To: <199702061935.UAA06419@tao.fokus.gmd.de >
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Christian,

Thanks for your comments on the IP/UDP/RTP header compression spec.

> If have a littel problem with the understanding of one section in
> the ID for IP/UDP/RTP header compression. On page 3, section
> 3.1, the "FULL_HEADER" packet size ist explained:
> 	".. The 4 bit sequence numer defined in section 3.3 ...
> 	is carried in the data field ... as defined in section 
> 	5.3.2 ... of [3]"
> 
> Im not familar with IPv6 [3], so I do not understand, where to put 
> the sequence number to :-) Does this mean, that it will be 
> appended to the data following the IP header? If so, have the header
> to be changed in any way ("no" in my understanding, because the 
> receiver will remove it).

I guess I need to make this more clear.  Section 5.3.2 of [3] shows
some diagrams indicating where the "Data" field is.  This 8-bit field
is a function recently added to the ipv6-hc compression draft as a
"hook" for additional header compression schemes such as the RTP
compression to pass data from the compressor to the decompressor.
When using an 8-bit context ID, this field will be the low byte of the
UDP length field of the FULL_HEADER packet.  I should probably add the
previous sentence to the draft.

Perhaps this confusion about the specific Data field versus the
general data part of the packet the follows the header stems from the
difference in capitalization rules between English and German? :)


> Another question: see the following scenario (for one session):
> 	o RTP packets were sent as COMPRESSED_RTP, so that a context
> 	  for this session exists.
> 	o the saved delta fields have been changed from the initial
> 	  values (e.g. IP ID delta = 5, bacause of loss)
> 	o one RTP packet was sent as FULL_HEADERS or COMPRESSED_UDP
> 	o the next packet should be send as COMPRESSED_RTP
> 
> The question is if the values in the context should be reset to their
> initial values, or if the values remain unchanged? So if the delta
> of the IP ID is 1 for the next packet to be send, do we have to
> indicate the change (set the I bit and include the new delta) or
> not? 

The delta values MUST be reset by the FULL_HEADER packet since the
first FULL_HEADER packet that the decompressor hears is what
establishes the context.  The purpose of the FULL_HEADER packet is to
re-establish the context when the decompressor and the compressor get
out of sync so you can't trust the any of the context values.

Section 3.3 explains that the delta for the RTP timestamp field is
reset whenever a FULL_HEADER, COMPRESSED_NON_TCP, or COMPRESSED_UDP
packet is transmitted.  However, I see that the rules for resetting
the IP ID delta are not clear enough.  The general idea is that when
the absolute value of a field is transmitted, the delta is also reset.
Therefore, the COMPRESSED_NON_TCP packet does reset the IP ID delta,
but the COMPRESSED_UDP packet does not.  In fact, the COMPRESSED_UDP
packet makes use of the current value of the IP ID delta if the I bit
is 0, or sets the value of the IP ID delta if the I bit is 1.

I have a question about a possible misunderstanding that may be
implied by your description of the scenario above.  The loss which
caused the IP ID to be set to 5 would have been upstream from the
compressor; that is, it is not due to a loss between the compressor
and decompressor.  So, that by itself does not imply that a
FULL_HEADER or COMPRESSED_UDP packet will be sent next.  Therefore,
did you mean that between the second and third bullets a packet was
lost between the compressor and decompressor?
							-- Steve


From rem-conf-request@es.net Fri Feb 07 05:14:53 1997 
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To: Stephen Casner <casner@precept.com>
Cc: rem-conf@es.net
Subject: Re: several messages
In-reply-to: Your message of "Thu, 06 Feb 1997 13:49:26 PST." <Pine.WNT.3.95.970206124414.-258053G-100000@oak.precept.com>
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Stephen,

> I guess I need to make this more clear.  Section 5.3.2 of [3] shows
> some diagrams indicating where the "Data" field is.  This 8-bit field
> is a function recently added to the ipv6-hc compression draft as a
> "hook" for additional header compression schemes such as the RTP
> compression to pass data from the compressor to the decompressor.
> When using an 8-bit context ID, this field will be the low byte of the
> UDP length field of the FULL_HEADER packet.  I should probably add the
> previous sentence to the draft.

Ok I see. BTW, a very stupid question: how about the high byte
of the IPv4 header? Why isn't it used like the low byte for the
context-id?

> Perhaps this confusion about the specific Data field versus the
> general data part of the packet the follows the header stems from the
> difference in capitalization rules between English and German? :)

Hm, perhaps; I don't see any difference between "Data field" and
"data field" ?! Anyway, because I'm not familar with IPv6, I didn't
know what the "Data field" is.
 
> The delta values MUST be reset by the FULL_HEADER packet since the
> first FULL_HEADER packet that the decompressor hears is what
> establishes the context.  The purpose of the FULL_HEADER packet is to
> re-establish the context when the decompressor and the compressor get
> out of sync so you can't trust the any of the context values.

OK, that's what I've expected.

> Section 3.3 explains that the delta for the RTP timestamp field is
> reset whenever a FULL_HEADER, COMPRESSED_NON_TCP, or COMPRESSED_UDP
> packet is transmitted.  However, I see that the rules for resetting
> the IP ID delta are not clear enough.  The general idea is that when
> the absolute value of a field is transmitted, the delta is also reset.
> Therefore, the COMPRESSED_NON_TCP packet does reset the IP ID delta,
> but the COMPRESSED_UDP packet does not.  In fact, the COMPRESSED_UDP
> packet makes use of the current value of the IP ID delta if the I bit
> is 0, or sets the value of the IP ID delta if the I bit is 1.

OK.

> I have a question about a possible misunderstanding that may be
> implied by your description of the scenario above.  The loss which
> caused the IP ID to be set to 5 would have been upstream from the
> compressor; that is, it is not due to a loss between the compressor
> and decompressor.  So, that by itself does not imply that a
> FULL_HEADER or COMPRESSED_UDP packet will be sent next.  Therefore,

That's right, so the scenareo wasn't the best one :-/

> did you mean that between the second and third bullets a packet was
> lost between the compressor and decompressor?

Either a loss of packets or any other change in the IP header (like
the change of the TTL e.g.). But as you wrote in the ID, the IP ID
field NORMALY increments by one, but it can also be used any other
algorithm or by the upper layer (as defined in RFC791). So when
the ID delta > 0x3fffff or <0, the FULL_HEADER packet HAVE to be
transmitted.

> 							-- Steve

Thanks a lot for your statements, they helped me to understand
some points much better and to validate my understanding.

Bye,
Chris

P.S: any comments regarding my thought about PT and CSRC compression?


From rem-conf-request@es.net Fri Feb 07 06:57:23 1997 
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To: ICCC'97 <iccc@prism.uvsq.fr>, congres@prism.uvsq.fr
From: Martin Bryan <mtbryan@sgml.u-net.com>
Subject: Re: ICCC'97

At 18:57 6/2/97 +0100, ICCC'97 wrote:
>FYI.  Last year we were still talking about the emerging IS.
>Now we are talking about the mature IS.  1998 - postmortem 
>on the IS?

Surely senile dementia sets in before the postmortem!
----
Martin Bryan, The SGML Centre, Churchdown, Glos. GL3 2PU, UK 
Phone/Fax: +44 1452 714029   WWW home page: http://www.u-net.com/~sgml/



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  address Martin Fangmeyer/Werner-von-Siemens-Str. 12/Erlangen/Germany
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From: Christian Zahl <zahl@fokus.gmd.de>
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Date: Fri, 7 Feb 1997 13:07:10 +0100
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To: Stephen Casner <casner@precept.com>
Cc: rem-conf@es.net
Subject: Re: several messages (IP/UDP/RTP compr)
In-reply-to: Your message of "Thu, 06 Feb 1997 13:49:26 PST." <Pine.WNT.3.95.970206124414.-258053G-100000@oak.precept.com>
Mime-Version: 1.0
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Stephen,

> ...
> but the COMPRESSED_UDP packet does not.  In fact, the COMPRESSED_UDP
> packet makes use of the current value of the IP ID delta if the I bit
> is 0, or sets the value of the IP ID delta if the I bit is 1.
> 
> I have a question about a possible misunderstanding that may be
> implied by your description of the scenario above.  The loss which
> caused the IP ID to be set to 5 would have been upstream from the
> compressor; that is, it is not due to a loss between the compressor
> and decompressor.  So, that by itself does not imply that a
> FULL_HEADER or COMPRESSED_UDP packet will be sent next.  Therefore,
> did you mean that between the second and third bullets a packet was
> lost between the compressor and decompressor?

Here is another real example: Linux 2-x seems to generate the IP-ID
a little bit different. The ID inclements by 1 but is send in
little-endian encoding, so the following sequence of IP-IDs can occur
without any loss of packages (I found this while looking into my
PPP trace some minutes ago):
	IP-ID field 
	fe 00
	ff 00
	00 01	- FULL_HEADER needed
	01 01

You can see that in this case the FULL_HEADER type is needed, because
the absolute value decreases when using the network-byte-order, 
without any kind of packet loss.

Perhaps we should think about using negative values for the delta
values? BTW, the same can theoretically happen when two IP packets
arrive in the wrong order (packet N+1 first, then packet N). If this
happens often, then the compression has NO gain! So how about of
using INTs for the delta fields instead of UNSIGNEDs. Ok, this
will reduce the maximum transferable delta value in one byte to
+-63. If you think this is too less, then we can say:
	o the 1 byte form is UNSIGNED
	o the 2 and 3 byte forms are SIGNED


What do you think about this idea?

Thanks,
Chris


From rem-conf-request@es.net Fri Feb 07 08:57:00 1997 
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Date: Fri, 7 Feb 1997 08:22:44 -0500 (EST)
From: "Donald E. Eastlake 3rd" <dee@cybercash.com>
To: Martin Bryan <mtbryan@sgml.u-net.com>, postmaster@prism.uvsq.fr, 
    congres@prism.uvsq.fr
Subject: PLEASE DELETE THIS MAILING LIST (Re: ICCC'97 (fwd))
Message-Id: <Pine.SUN.3.91.970207081921.1542A-100000@cybercash.com>
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It's bad enough that you were spamming everyone you could think of with 
the original message, which I got more than three copies of, now we has a 
damn coverstation being blasted out on a zillion mailing lists.

Please stop this network abuse immeediately.

Unsolicited mass email is not an appropriat way to advertise.  Use web 
pages, appropriate news groups, and your own organizations mailing 
lists.  Don't spam other mailing lists.

Donald

=====================================================================
Donald E. Eastlake 3rd     +1 508-287-4877(tel)     dee@cybercash.com
   318 Acton Street        +1 508-371-7148(fax)     dee@world.std.com
Carlisle, MA 01741 USA     +1 703-620-4200(main office, Reston, VA)
http://www.cybercash.com           http://www.eff.org/blueribbon.html

---------- Forwarded message ----------
Date: Fri, 07 Feb 1997 11:07:17 +0000
From: Martin Bryan <mtbryan@sgml.u-net.com>
To: ICCC'97 <iccc@prism.uvsq.fr>, congres@prism.uvsq.fr
Subject: Re: ICCC'97

At 18:57 6/2/97 +0100, ICCC'97 wrote:
>FYI.  Last year we were still talking about the emerging IS.
>Now we are talking about the mature IS.  1998 - postmortem 
>on the IS?

Surely senile dementia sets in before the postmortem!
----
Martin Bryan, The SGML Centre, Churchdown, Glos. GL3 2PU, UK 
Phone/Fax: +44 1452 714029   WWW home page: http://www.u-net.com/~sgml/




From rem-conf-request@es.net Fri Feb 07 09:51:17 1997 
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Date: Fri, 07 Feb 1997 09:50:26 -0500
From: Allen Robel <robelr@stone51.netlab.indiana.edu>

Hi,

Due to insurmountable logistical problems, this event has been cancelled.  
Please accept our apologies.

regards,

-- 
Allen Robel                                             Indiana University
mailto:robelr@netlab.indiana.edu               Area Code + Prefix: 812-855
http://www.netlab.indiana.edu/~robelr   Office 0962, NetLab 3697, Fax 8299



From rem-conf-request@es.net Fri Feb 07 11:03:01 1997 
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Martin Fangmeyer wrote:
> 
> unsubscribe
> --
>   address Martin Fangmeyer/Werner-von-Siemens-Str. 12/Erlangen/Germany
>  homepage http://www.rrze.uni-erlangen.de/~unrzg5
> motorbike 1988-94 Suzuki GS 500 '80 oldie but goldie
>           1994-?  Yamaha XJ900 '90 ... ten years after
>       car Renault R4 can't beat _this_ feeling (now sold)

You should send your un-subscribe request to:

rem-conf-request@es.net
-- 
++++++++++++++++++++++++++++++++++++++++++++++

Giuseppe Fabio Campanale
 
c/o CEFRIEL - Politecnico di Milano         
    Via L. Emanueli, 15                            
    20126 MILANO - Italy                             

mailto:campanal@mailer.cefriel.it
fax   : +39-2-66-100-448
URL   : http://www.cefriel.it/

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

From rem-conf-request@es.net Fri Feb 07 12:37:32 1997 
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References: <199702052200.RAA23761@flora.cc.gatech.edu> from "Kevin C. Almeroth" at Feb 5,
            97 05:00:51 pm
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To: "Carl Malamud [IMS]" <carl@also.media.org>
From: "James E. [Jed] Donnelley" <jed@llnl.gov>
Subject: Re: MBone Community Size?, fragility of the MBone
Cc: rem-conf@es.net, Petri Helenius <pete@sms.fi>


>I'm not sure I believe the mbone is really all that fragile
>and distributed control seems to have worked very well for
>things like the web, despite very real fears of backbone collapse.
>My biggest fear for the mbone is that we control it so carefully
>that the demand never happens, the ISP's never support it
>properly, and the whole thing turns into a technically-correct
>solution that is buried by sub-optimal unicast streaming
>solutions.

I see more differences between the MBone and the Web that
make the Web commercially viable and the MBone not (yet).

Mainly with the Web there is much more tolerance for packet
losses.  This results from several factors:

1.  TCP and the ability to wait (and wait and wait and ...)

2.  If things are so bad that a connection times out on
a Web transfer, what have I lost?  Generally not much.
No matter what it was, I can always try again (and do...).
If I have been listening to a real time MBone multicast
and it has my attention and then a crutial section is
dropped, what have I lost?  Potentially (and actually in
my experience) a lot!  Such lost sections can ruin an
hours engagement.  I got so frustrated with it that I
stopped using it for serious interactions, and I am
MUCH more tolerant of such things than most of my
colleagues.

I think the IP Multicast technology and the whole MBone
approach as it stands have serious technical difficulties.
You seem to belittle unicast streaming like "Real Audio",
but from a user perspective it generally works.  The
bandwidth demand is low enough, it runs over TCP (takes
care of packet losses) and you can even repeat it
(not real time).

I will mention for your possible interest a paper I wrote:

http://www-atp.llnl.gov/atp/papers/HRM/HRM.html

that develops what might be called TCP multicast.  It is
a different approach to reliable multicast than the
"over IP multicast" crowd (e.g. RMP).  It uses retransmission
on a link by link basis (e.g. with TCP, but not necessarily)
to achieve reliable end-to-end multicast.

I think a technology like that with a fair amount of delay
tolerance might make something like multicast "Real Audio"
(even realtime) work.  You would probably need a "catch
up" technology (e.g. time compressed audio) for real time,
but such an approach might give commercialization an opportunity.

There is still the problem of allocating bandwidth.
I don't really know why, but I don't see how the typical
Internet laize faire "transmit when you can/dare" approach
can work with real time multimedia.  Perhaps it can.  Maybe
when the load gets too large the least interested people
will begin to drop off and bring the load down.  Something
like that might work, but it doesn't seem likely to me.
Perhaps it is just that I would be an early casualty,
willing to pay the telephone charge for clarity.

I think the simpliest test of this is "Internet Phone."
As the bandwidth available on the Internet grows, why
won't people just start doing more Internet telephoning
to save long distance costs?  Most noise tolerant first?

I can't answer that one.  Do you know Carl?


Jed Donnelley    jed@llnl.gov   Phone: (510) 422-4309
http://www.webstart.com/jed/    FAX: (510) 423-1355



From rem-conf-request@es.net Fri Feb 07 13:12:58 1997 
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Date: Fri, 7 Feb 1997 13:11:18 -0500 (EST)
From: kevin@cc.gatech.edu (Kevin C. Almeroth)
Message-Id: <199702071811.NAA29392@morticia.cc.gatech.edu>
To: schulzrinne@cs.columbia.edu
Subject: Re: MBone Community Size?
Cc: rem-conf@es.net

>>I wonder how many of the 10,600 are "this looks cool, let's try this -
>>this is not ready for prime time and there's nothing interesting on -
>>let me go back to RealAudio or VRML or other neat toy". In other words,
>>how many addresses can be considered "regulars", who actually use this
>>on a more or less daily or at least regular basis? I have the suspicion
>>that Mbone "churn" may be pretty high; the core user community may be
>>closer to the 600 mentioned...

Okay, I've thrown a few more statistics together and here's some additional
info (I've also done a more careful search for IP addresses):


10,995 unique IP addresses
531,728 observed "connections" that I have data for (all over about a 2 year period)

mean:   477 connections per IP address
median: 261 connections per IP address
max:    21,791 connections for one particular IP address (wow!)

25% of the IP addresses connected 126 times or less
50% of the IP addresses connected 261 times or less
75% of the IP addresses connected 473 times or less

The least active 1% of IP addresses connected (on average) 6 times

-Kevin

From rem-conf-request@es.net Fri Feb 07 15:40:02 1997 
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To: rem-conf@es.net
Cc: Stephen Casner <casner@precept.com>
Subject: Inconsistence in IP/UDP/RTP compression ID
Mime-Version: 1.0
Content-Type: text/plain

Well if I'm right then there seems to be a problem in the ID. I'm currently
implementing a compessor according to the mentioned ID and have
encountered a problem, if I'm right; any corrections are welcome :-)

So imagine the following scenareo:
	o there are a RTP translator T, sitting on a host.
	o the translater passes the SSRC field of the incomming RTP
	  streams unmodified (that's it's jobs, of course).
	o all IP packets are "owned" by the translator, i.e. the IP
	  SRC and DST address are allways the same, also the UDP
	  SRC and DST ports.
	o this results in a context for EACH sender the tranlater
	  forwards, because of the different SSRC for each sender
	  (which belongs to the session-context-indicator)

Now think of two senders, which the translater receives and forwards.
The IP/UDP pair will be generated out of from the translater, so they
can be expected to be in sequence. We have two established contexts,
one for sender A and one for sender B:

	+---+ IP(A,T) UDP(1,2) RTP(...,A,..)			CXT cA
	| A | -------->
	+---+		+---+
			| T | ------> IP(T,D) UDP(4,5) RTP(...,A|B,...)
	+---+		+---+
	| B | --------->
	+---+ IP(B,T) UDP(3,2) RTP(...,B,...)			CXT cB

Now sender A changes the payload-type. This forces the translator T to
send the packet as COMPRESSED_UDP instead of COMPRESSED_RTP. There
will arrive several questions:
	o to which context do the outgoing packet belongs, cA or cB?
	  Do we have to create a new one?
	o will all other contexts belonging to this source (the
	  tanslator) will be invalidated (cA and cB) ?

On page 5 you can read: "COMPRESSED_UDP ... It redefines the SSRC field
of the (potential) RTP header...". So the SSRC field has no meaning
(in my understanding), because we cannot assume that we have an
RTP packet. So we also cannot save any RTP header informations in the
context (right?). Therefore, we have lost any kind of synchronization
>from the compressor to the decompressor regarding this RTP stream!
If so, this would result in sending a FULL_HEADER next, to resync.

Or perhaps I have misunderstood one importand think: are contexts
for the COMPRESSED_RTP and COMPRESSED_UDP are totally indepndent?
>From one point of view this seems to make sense, but on the other
side, it seems to be impossible to have two different applications
using the same SRC and DST addresses and ports, so when we have an
RTP context, the UDP context is a subset of it, isn't it?

Stephen, perhaps you have an description what have to be sent in the
scenareo shown above. I think in this point the ID is a little bit
confusing :-)

Well, I have to admit that I'm a little bit confused now :-/ So,
any comments are welcome!

Thanks,
Chris



From rem-conf-request@es.net Fri Feb 07 16:14:42 1997 
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Date: Fri, 7 Feb 1997 13:00:01 -0800
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From: Jerrlyn Iwata <jerrlyn@postgres.berkeley.edu>
Subject: Berkeley Multimedia & Graphics Seminar

                Berkeley Multimedia and Graphics Seminar 
        (Wednesday February 5, 1997 12:30-2:00 PDT 405 Soda Hall) 

  "The Impact of Future Developments in Flat Panel and Projection Display" 

                              Dave Mentley 
                       Stanford Resources, Inc. 
                              San Jose, CA 

The search for a useful and economical flat panel display began in the
1960s. The problem has been much harder to solve than anyone imagined, but
the progress in recent years has also been astounding.  Likewise, electronic
projection technology, after many stagnant years, has accelerated to the
point where a new generation of projector is arriving every year. Despite
the progress, many issues remain to be resolved, particularly regarding
performance, price and reliability. The talk will discuss the technology
market and performance issues for LCDs, color plasma, projection displays
and some of the advanced displays expected to make an impact within the next
5 years. 

Stanford Resources is interested in hiring folks to work on Electronic
Display Market Research.  Interested students should contact Dave Mentley. 

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

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


From rem-conf-request@es.net Fri Feb 07 16:17:20 1997 
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Subject: Request for any comments: SDP compression
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During the last month I worked on the integration of low-bandwidth
PPP users to the mbone. One of point of interest is the IP/UDP/RTP
compression, for which Stephen Casner and Van Jacobson already set up 
an ID. Another issue of interest is the access to SDP session
announcements. Currently, we have an (very alpha based) draft on how
to compress SDP announcements. Another draft (currently in process)
handles an efficient way to transfer informations of changes in the
global SDP "cache".

The current version of the SDP compression can be examined at:
	http://www.fokus.gmd.de/step/cz/da-cz/sdpCompr.html
This doc also included some comparisons between gzip and the new
compression technique. A draft for transfering SDP informations
over an PPP link can be expected soon.

Any comments are welcome,
Chris


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To: 298-list@bugs-bunny.CS.Berkeley.EDU
From: Jerrlyn Iwata <jerrlyn@postgres.berkeley.edu>
Subject: [correction] Berkeley Multimedia & Graphics Seminar

                Berkeley Multimedia and Graphics Seminar 
        (Wednesday February 12, 1997 12:30-2:00 PDT 405 Soda Hall) 

         "Client-Server Architectures for PDA-based Applications" 

                              Wayne Citrin 
                   University of Colorado, Boulder 

Personal Digital Assistants [PDAs] are an attractive computational platform
for mobile users, particularly due to their small size and weight and the
ability to directly enter and interact with data on the display using a pen.
However, the fact that PDAs are poor in computational resources generally
precludes many useful applications. We have developed a client-server
architecture that supports computationally intensive PDA-based applications
by distributing functionality between a resource-poor PDA front end that
gathers, displays, and stores data and performs basic recognition, and more
computationally powerful host-based back ends that perform higher-level
recognition tasks and database access. Communication between the front and
back ends is over wired or wireless
connections. We discuss the design of the architecture and its associated
communications protocol, as well as related communications and performance
issues. We also describe a prototype implementation based on the
architecture that supports collaborative design and diagram recognition. 

This work was done in collaboration with Mark Gross of the College of
Architecture and Design at the University of Colorado, and was supported in
part by the Colorado Advanced Software Institute and US West Advanced
Technologies. 

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

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


From rem-conf-request@es.net Fri Feb 07 18:30:11 1997 
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Date: Fri, 7 Feb 1997 15:24:57 -0800 (PST)
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To: "James E. [Jed] Donnelley" <jed@llnl.gov>
From: Ross Finlayson <finlayson@lvn.com>
Subject: Re: MBone Community Size?, fragility of the MBone
Cc: "Carl Malamud [IMS]" <carl@also.media.org>, rem-conf@es.net, 
    Petri Helenius <pete@sms.fi>

>I think the IP Multicast technology and the whole MBone
>approach as it stands have serious technical difficulties.
>You seem to belittle unicast streaming like "Real Audio",
>but from a user perspective it generally works.  The
>bandwidth demand is low enough, it runs over TCP (takes
>care of packet losses) and you can even repeat it
>(not real time).

Jed, I think you're being too pessimistic here.  BTW, if you haven't already
done so, you might want to try listening to the experimental 28.8 kbps MBone
feed that the RealAudio folks set up recently.  The quality is remarkable (&
I think it actually uses something more like 20 kbps of bandwidth!).  You
can launch this from somewhere on RealAudio's web site, I think.  (I also
have it announced in multikit's "Public Grab Bag" directory, so if you're
running multikit you can launch it from there.)

Of course this is a live feed, so you don't get to repeat it.  But I think
there's some redundancy against packet loss built in.

Now if only the RealAudio encoding were open (or alternatively, if only
someone started demonstrating low-bandwidth feeds of similar quality using
open standard encodings)...

        Ross.


From rem-conf-request@es.net Fri Feb 07 18:36:08 1997 
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To: rem-conf@es.net
From: lee <lee@math.washington.edu>
Reply-to: lee@math.washington.edu
Subject: Pacific Northwest Geometry Seminar

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

Title:       
	Pacific Northwest Geometry Seminar
Date:        
	Feb 08, 1997

Time:        
	09:00 PST8PDT 9 hours

Contact:     
	lee@math.washington.edu

URL:         
	http://www.math.washington.edu/~lee/PNGS/97-winter/

Description:        
	The talks by Marsden, Thurston, and Schoen will be broadcast live on the MBone (the Internet Multicast  Backbone). They will also be rebroadcast on Tuesday, Wednesday, and Thursday, February 18-20, at noon  PST.









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

From rem-conf-request@es.net Fri Feb 07 22:19:10 1997 
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From: Deb Agarwal <deba@george.lbl.gov>
Message-Id: <199702080316.TAA14408@george.lbl.gov>
Subject: Length of sdr announcement warning
To: rem-conf@es.net
Date: Fri, 7 Feb 1997 19:16:23 -0800 (PST)
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Hi all,

I just ran up against a problem that many of you may not
be aware of so I thought I'd write the list.  My original
sdr announcement of the LabWeb session was only going out
to a small portion of the MBone.  It turns out that I had
placed so much descriptive text in the announcement that
the multicast sdr packet containing the announcement was
being fragmented.  This caused it to be dropped by some
routers in the MBone.

Lesson learned!  Don't make your session announcement overly
verbose because that may keep it from going out.

Deb Agarwal
PS.  There are no warnings in sdr or anywhere else telling you
when an announcement is overly verbose.
-- 
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Deb Agarwal                                  e-mail:DAAgarwal@lbl.gov
MS50B-2239                                   phone :(510)486-7078
Lawrence Berkeley National Lab  
Berkeley, CA 94720
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

From rem-conf-request@es.net Sat Feb 08 02:47:04 1997 
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Date: Sat, 8 Feb 1997 09:46:23 +0200 (EET)
Message-Id: <199702080746.JAA03407@silver.sms.fi>
From: Petri Helenius <pete@sms.fi>
To: Ross Finlayson <finlayson@lvn.com>
Cc: "James E. [Jed] Donnelley" <jed@llnl.gov>, 
    "Carl Malamud [IMS]" <carl@also.media.org>, rem-conf@es.net
Subject: Re: MBone Community Size?, fragility of the MBone
In-Reply-To: <1.5.4.16.19970207161130.4f374efe@pop.best.com>
References: <1.5.4.16.19970207161130.4f374efe@pop.best.com>

Ross Finlayson writes:

 > Now if only the RealAudio encoding were open (or alternatively, if only
 > someone started demonstrating low-bandwidth feeds of similar quality using
 > open standard encodings)...
 > 
Haven't unfortunately done that but there are two similar feeds
running at addresses 224.42.42.2 and 224.42.42.4. First is at 112kbps
and second at 24kbps. They are running using Xing technology MPEG
audio encoders/decoders. Pick up the software from Xing or from
http://www.ml.tele.fi and tune in!

Pete

From rem-conf-request@es.net Sat Feb 08 12:54:42 1997 
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From: Mark Handley <mjh@isi.edu>
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To: DAAgarwal@lbl.gov
cc: rem-conf@es.net
Subject: Re: Length of sdr announcement warning
In-reply-to: Your message of "Fri, 07 Feb 1997 19:16:23 PST." <199702080316.TAA14408@george.lbl.gov>
Date: Sat, 08 Feb 1997 12:52:52 -0500
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Sender: mjh@buttle.lcs.mit.edu


>I just ran up against a problem that many of you may not
>be aware of so I thought I'd write the list.  My original
>sdr announcement of the LabWeb session was only going out
>to a small portion of the MBone.  It turns out that I had
>placed so much descriptive text in the announcement that
>the multicast sdr packet containing the announcement was
>being fragmented.  This caused it to be dropped by some
>routers in the MBone.
>
>Lesson learned!  Don't make your session announcement overly
>verbose because that may keep it from going out.

Yes, that's what the URL is for.  The GUI makes it difficult to add so
much descriptive text that fragmentation is caused, but won't prohibit
it.  However sdr will reject announcements that are greater than 2K
bytes (a somewhat arbitrary limit that we really never want to reach.)

Point taken - a warning would be good.

Mark

From rem-conf-request@es.net Sat Feb 08 17:52:46 1997 
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Date: Sat, 8 Feb 1997 09:28:38 -0500 (EST)
From: kevin@cc.gatech.edu (Kevin C. Almeroth)
Message-Id: <199702081428.JAA03441@flora.cc.gatech.edu>
To: rem-conf@es.net
Subject: Re: MBone Community Size?, fragility of the MBone

>>I see more differences between the MBone and the Web that
>>make the Web commercially viable and the MBone not (yet).

The web is commercially viable today?  I certainly disagree.
Sure you can point to a few businesses but they are limping
along.  The web's commercial viability suffers from its own
set of (security) problems.

In your comparison, the web is not much better than the MBone.

>>2.  If things are so bad that a connection times out on
>>a Web transfer, what have I lost?  Generally not much.
>>No matter what it was, I can always try again (and do...).
>>If I have been listening to a real time MBone multicast
>>and it has my attention and then a crutial section is
>>dropped, what have I lost?  Potentially (and actually in
>>my experience) a lot!  Such lost sections can ruin an
>>hours engagement.  I got so frustrated with it that I
>>stopped using it for serious interactions, and I am
>>MUCH more tolerant of such things than most of my
>>colleagues.

If you use the web and don't get what you want, you've lost
whatever you were trying to get.  I've stopped using some
sites because I can't tolerate the extremely long wait times.
There went their commercial viability.

In your comparison, the web is not much better than the MBone.

>>I think the IP Multicast technology and the whole MBone
>>approach as it stands have serious technical difficulties.

No doubt.  So what do you suggest we do with the MBone?
Throw it away and go to TCP multicast?  You'll have a hard time
getting a majority of people buy into the premise that the
MBone has failed and then get those same people to believe a
better way is to use TCP multicast.

But that's just my opinion.  The MBone is continually being
improved and offers a somewhat orthogonal solution to reliable
multicast.  Applications exist which are better supported by
one or the other.  Independent of the problem of available 
bandwidth of course.

-Kevin Almeroth

From rem-conf-request@es.net Sun Feb 09 17:42:34 1997 
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From: Arlie Davis <arlie@thepoint.net>
To: "'kevin@cc.gatech.edu'" <kevin@cc.gatech.edu>
Cc: "'rem-conf@es.net'" <rem-conf@es.net>
Subject: RE: MBone Community Size?, fragility of the MBone
Date: Sun, 9 Feb 1997 17:34:34 -0500
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The web is commercially viable in some ways, if not all.  One of the
real selling points of the web is that it allows companies to provide a
great deal of very detailed information to (potential and real)
customers.  The revenue gained from this is very real.

True, not many people are successfully selling products on the web.  At
least the transaction is not occurring there; many people do a lot of
feature-comparing and shopping on the web, then call to place the order,
or other hybrids.

This has almost nothing to do with why the MBONE is not commercially
viable.  It's a completely different set of problems.  First and
foremost, the major stumbling block for multicast applications
(commercial and otherwise) is the lack of a fast, robust,
well-administered multicast routing infrastructure.  Many ISPs don't
understand multicast; many think that it will increase their link
utilization, when in all likelihood it will decrease it due to
elimination of redundant feeds.

Second, security for MBONE technologies is not much of a problem, or at
least it is not a central requirement.  The applications that need it
support it, usually in the form of encryption.  Also, I doubt that
you'll ever have a multipoint financial transaction -- that's a unicast
thing.

-- arlie

>-----Original Message-----
>From:	kevin@cc.gatech.edu [SMTP:kevin@cc.gatech.edu]
>Sent:	Saturday, February 08, 1997 9:29 AM
>To:	rem-conf@es.net
>Subject:	Re: MBone Community Size?, fragility of the MBone
>
>>>I see more differences between the MBone and the Web that
>>>make the Web commercially viable and the MBone not (yet).
>
>The web is commercially viable today?  I certainly disagree.
>Sure you can point to a few businesses but they are limping
>along.  The web's commercial viability suffers from its own
>set of (security) problems.
>
>In your comparison, the web is not much better than the MBone.

From rem-conf-request@es.net Sun Feb 09 20:13:53 1997 
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To: Arlie Davis <arlie@thepoint.net>
cc: "'kevin@cc.gatech.edu'" <kevin@cc.gatech.edu>, 
    "'rem-conf@es.net'" <rem-conf@es.net>
Subject: Re: MBone Community Size?, fragility of the MBone
In-reply-to: Your message of "Sun, 09 Feb 1997 17:34:34 EST." <c=US%a=_%p=ThePoint_Interne%l=TIS_MAIL-970209223434Z-227@tis-mail.thepoint.net>
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Date: Mon, 10 Feb 1997 11:11:33 +1000
Message-ID: <9033.855537093@connect.com.au>
From: George Michaelson <ggm@connect.com.au>


I take a more cynical view Arlie.

The reason MBONE is not commercially viable is that nobody has worked out
how to make an awsome profit from it, and nobody at the PC/Mac cutting
edge has put that into a free app people want to use over any other toy.

Being commercially viable doesn't mean its well engineered or even works,
and by jingo a LOT of internet productization right now is about selling
snake oil in broken glass bottles.

MBONE is hard financials. Its about making some amount of b/w available
and selling it to people. If you understand MBONE then you probably understand
that somebody else is also getting it, and paying for it too, and pretty soon
recipients start asking how much of the shared cost they should be paying and
what the profit-points are like. In other words the more you sell the better
the margin, but the more you sell the bigger the pressure to sell it cheaper.

The 'defensive' reasons we all cite for why MBONE is preferable to pt-to-pt
multi-way are not sells. Nothing is being driven by ISP's being smart right
now, its all about end-users being smarter and code-releasing people getting
stuff into their hands. 

ISPs are the lag-point, not the innovators. Innovation comes from people like
Van, who seems to have written vat for three or four core reasons:

	he wanted to
	his users wanted a tool like that
	his funding body wanted to save money on travel and vat helped

I sure am glad he's not working in the ISP business. If he was he'd be stuck
writing financials packages and re-working the billing system around a
version of tcpdump from satan.

MBONE is the same. Its not cisco's in-house killer protocol they went out
and sold en-masse to the SNA users of the world (like GRE and STUN) its
something nifty, but its not the bottom line.

-George

--
George Michaelson         |  connect.com.au pty/ltd
Email: ggm@connect.com.au |  c/o AAPT,
Phone: +61 7 3834 9976    |  level 8, the Riverside Centre,
  Fax: +61 7 3834 9908    |  123 Eagle St, Brisbane QLD 4000



From rem-conf-request@es.net Sun Feb 09 21:29:05 1997 
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From: "Carl Malamud [IMS]" <carl@also.media.org>
Message-Id: <199702100226.VAA03830@also.media.org>
Subject: Re: MBone Community Size?, fragility of the MBone
To: ggm@connect.com.au (George Michaelson)
Date: Sun, 9 Feb 1997 21:26:55 -0500 (EST)
Cc: arlie@thepoint.net, kevin@cc.gatech.edu, rem-conf@es.net
In-Reply-To: <9033.855537093@connect.com.au> from "George Michaelson" at Feb 10, 97 11:11:33 am
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According to George Michaelson:
> 
> 
> I take a more cynical view Arlie.
> 
> The reason MBONE is not commercially viable is that nobody has worked out
> how to make an awsome profit from it, and nobody at the PC/Mac cutting
> edge has put that into a free app people want to use over any other toy.
> 

My view of the mbone was that is was simply one of many different
ways to get our content out.  For the National Press Club, for
example, we would send the data live out via mbone, but also
use multicast to record the data on disk, then turn it into various
other "streaming" formats (e.g., .au, .gsm, .ra) and then serve
that back out via a web interface.

To me the mbone was a tool, not a product.  Nobody is going to make
a living out of "multicasting content" but they might be able
to use multicasting as one of the (many) tools to package the content
in an attractive way.  


From rem-conf-request@es.net Sun Feb 09 21:45:34 1997 
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To: "Carl Malamud [IMS]" <carl@also.media.org>
cc: arlie@thepoint.net, kevin@cc.gatech.edu, rem-conf@es.net
Subject: Re: MBone Community Size?, fragility of the MBone
In-reply-to: Your message of "Sun, 09 Feb 1997 21:26:55 EST." <199702100226.VAA03830@also.media.org>
Date: Mon, 10 Feb 1997 12:42:50 +1000
Message-ID: <10666.855542570@connect.com.au>
From: George Michaelson <ggm@connect.com.au>


  My view of the mbone was that is was simply one of many different
  ways to get our content out.  For the National Press Club, for
  example, we would send the data live out via mbone, but also
  use multicast to record the data on disk, then turn it into various
  other "streaming" formats (e.g., .au, .gsm, .ra) and then serve
  that back out via a web interface.
  

This is a good view. Its a view that says delivery method is less exciting
than the content. Its under-the-cover stuff.

  To me the mbone was a tool, not a product.  Nobody is going to make
  a living out of "multicasting content" but they might be able
  to use multicasting as one of the (many) tools to package the content
  in an attractive way.  
  
I totally agree. Its like asking why home video editing suites can't do
SMPTE timestamp based edit and don't come with jog-shuttle. Its not because
they aren't useful tools, and if your 'market' is professional video editors
you can expect these to be checkbox items. If your prime interest is
getting the material into peoples hands, who cares anyway?

I think mbone suffers from the confusion/blurred destinction of infrastructure
and end-use. 

Carl, did you find mbone actually 'made a difference' there? Was the pain worth
the benefits? did the final .ra and other forms sound acceptable even after
filtering through MBONE as a delivery path?

-George

--
George Michaelson         |  connect.com.au pty/ltd
Email: ggm@connect.com.au |  c/o AAPT,
Phone: +61 7 3834 9976    |  level 8, the Riverside Centre,
  Fax: +61 7 3834 9908    |  123 Eagle St, Brisbane QLD 4000

From rem-conf-request@es.net Mon Feb 10 01:54:28 1997 
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To: George Michaelson <ggm@connect.com.au>
cc: "Carl Malamud [IMS]" <carl@also.media.org>, arlie@thepoint.net, 
    kevin@cc.gatech.edu, rem-conf@es.net
Subject: Re: MBone Community Size?, fragility of the MBone
In-reply-to: Your message of "Mon, 10 Feb 1997 12:42:50 +1000." <10666.855542570@connect.com.au>
Date: Mon, 10 Feb 1997 06:51:13 +0000
Message-ID: <691.855557473@cs.ucl.ac.uk>
From: Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>



 >  To me the mbone was a tool, not a product.  Nobody is going to make
 >  a living out of "multicasting content" but they might be able
 >  to use multicasting as one of the (many) tools to package the content
 >  in an attractive way.  

according to a recent seminar here by mike lesk, very few people are
making much money out of the web either, so this is a spurious
distinction.....

most commercial people  (as per the doonesbury cartoon on the subject) are on 
the web because the hype has made them "scared they will look stupid
if they are not".

sure there are some positive cases, but selling information is not big
business (except share prices and people just dont do that yet there)

video/audio content IS big business, but has stringent quality
expectations which the mbone cannot do...

actually, the quality issue is the same for the web - as soon as the
underlying net can deliver guaratees (e.g. ipsec, int-serv, type
guarantees) then the share delaers might think about serious use of
the public internet - just the same as the tv and radio people 

but for now, disney, reuters alike, just keep a watching brief

 jon


From rem-conf-request@es.net Mon Feb 10 02:14:00 1997 
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cc: "Carl Malamud [IMS]" <carl@also.media.org>, arlie@thepoint.net, 
    kevin@cc.gatech.edu, rem-conf@es.net
Subject: Re: MBone Community Size?, fragility of the MBone
In-reply-to: Your message of "Mon, 10 Feb 1997 06:51:13 GMT." <691.855557473@cs.ucl.ac.uk>
Date: Mon, 10 Feb 1997 17:11:06 +1000
Message-ID: <15904.855558666@connect.com.au>
From: George Michaelson <ggm@connect.com.au>



I think quite a lot of people are making money from the web. Little of it
is justified and little of it looks to be good longterm business.

I suspect that the ability of existing broadcast media to send one instance
of VERY expensive production to around 200,000,000 people simultaneously
changes the quality pressure somewhat. Even if only 1% dislike the soundmix
you can expect a LOT of calls onto a helpdesk to complain 

 (people who don't have hearing loss will wonder on that, but I know
 from my own experience with mild loss, and older peoples war-induced
 major loss, that soundmix is the number-1 complaint about TV shows after
 excessive (or insufficient) flesh content. Words and MUSAK do not mix.)

Right now MBONE leverages off willingness of recipients (because when its
being used many-to-many it still typically one speaker many listener modes
for a variant set of speaker, listener pair/tuples) to put up with the
quality, and it also leverages off the higher quality sourcing for some
feeds. If it was all done using quickcam or ISA-bus video onto 8bit-cards
it would be even more dire.

I think the main uses of MBONE haven't even begun to be clear. When it looked
like NNTP was going to be a winner under RMP I thought that was it because
traffic efficient porn has to be a good choice, but it didn't work. Likewise
when people doing layered codings I assumed we'd see multi-lingual soundflow
over the slowscan image for IETF (surely with that many francophones on the
councils they would push for this :-) but I don't hear much about that or
title-options, or any of a number of things that digital TV is touted as
providing.

So is there some chance that Digital TV is also doomed to failure? Do people
really not want the features?

-George

From rem-conf-request@es.net Mon Feb 10 05:16:11 1997 
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To: J.Crowcroft@cs.ucl.ac.uk
cc: ggm@connect.com.au, carl@also.media.org, arlie@thepoint.net, 
    kevin@cc.gatech.edu, rem-conf@es.net
Message-ID: <4225643A.00370288.00@maskit1.vocaltec.co.il>
Date: Mon, 10 Feb 1997 12:13:30 +0200
Subject: Re: MBone Community Size?, fragility of the MBone
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Scott Petrack
02/10/97 12:13 PM


-------- On 02/10/97 06:51 AM GMT  J.Crowcroft @ cs.ucl.ac.uk  wrote:
>video/audio content IS big business, but has stringent quality
>expectations which the mbone cannot do...

>actually, the quality issue is the same for the web - as soon as the
underlying net can >deliver guaratees (e.g. ipsec, int-serv, type
>guarantees) then the share delaers might think about serious use of
>the public internet - just the same as the tv and radio people

What is needed are not guarantees, just lots of bandwidth of decent
quality.
Television gives you no gurantees, neither do international Telephone
calls, nor radio.
What it does give you is enough decent-quality bandwidth that to be useful
for the purpose.
No gurantees.

The Internet and multicast promise to offer the sorts of services that
news, radio,
television, etc. can offer, but on a global scale rather than the current
local reach
which is possible. For telephony and mail, it promises richer, more
flexible, cheaper function.

That's "all."

The reasons that these things are not taking off is simply that there isn't
 enough
bandwidth of the right quality.

Of course, it might well be that the guarantess are an essential thing so
that businesses
can charge to get the money and incentive to upgrade the bandwidth. So it
may be that we
need guarantees to put in place the *financial* infrastructure to build up
the
internet. But it is not needed to insure usefulness of the service.

Scott



From rem-conf-request@es.net Mon Feb 10 13:26:10 1997 
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To: rem-conf@es.net
Subject: Re: MBone Community and fragility ...


    Maybe we need to stop thinking of the web / MBONE as a product and realize
they are really more a mode of transport.  There may exist "better" transport
mechanisims to convey the content of "the web" and MBone".  In the process we
may even improve the Internet's performance by using it for what it was designed
for,  two-way data transport.

                                    ......victor barajas


From rem-conf-request@es.net Mon Feb 10 13:50:54 1997 
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To: rem-conf@es.net
From: m1 <m1@msri.org>
Reply-to: m1@msri.org
Subject: Jerry Marsden "Geometry and Classical Mechanics"

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

Title:       
	Jerry Marsden "Geometry and Classical Mechanics"
Date:        
	Feb 18, 1997

Time:        
	12:00 PST8PDT 1 hours

Contact:     
	m1@msri.org

URL:         
	http://www.math.washington.edu/~lee/PNGS/97-winter/marsden.html 

Description:        
	One of the many problems that control theory addresses is that of feedback stabilization. A simple example  illustrating what is meant is how humans learn to balance (that is, stabilize) when walking, riding bicycles,  or to balance inverted broomsticks, or whirling a lasso. Related examples are the problem of stabilizing a  spinning satellite and the reorientation problem for a springboard diver. This talk presents a general  stabilization technique introduced by Bloch, Krishnaprasad, Marsden and Sanchez in the context of a rigid  body with internal rotors. This technique is now known to result from a general Kaluza-Klein type of  construction, familiar to geometers and physicists. Other examples, such as stabilizing an inverted pendulum  on a cart, the dynamics of an underwater vehicle and the motion of a double spherical pendulum will also be  discussed.









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

From rem-conf-request@es.net Mon Feb 10 13:52:50 1997 
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To: rem-conf@es.net
From: m1 <m1@msri.org>
Reply-to: m1@msri.org
Subject: Bill Thurston "Foliations and Geometry"

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

Title:       
	Bill Thurston "Foliations and Geometry"
Date:        
	Feb 19, 1997

Time:        
	12:00 PST8PDT 1 hours

Contact:     
	m1@msri.org

URL:         
	http://www.math.washington.edu/~lee/PNGS/97-winter/thurston.html

Description:        
	I will discuss progress in understanding the geometry of foliations, particularly in dimension 3, in connection  with the program of trying to reconcile and combine several approaches toward three-dimensional topology.  A main goal of the program is to find a construction for a geometric decomposition for a 3-manifold based  on a taut foliation, an essential lamination, or a tight contact structure. An important related goal is to  analyze how these structures relate to a hyperbolic structure.









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

From rem-conf-request@es.net Mon Feb 10 14:04:41 1997 
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From: jcole@precept.com
Message-Id: <970210102839.ZM3414@little-bear>
Date: Mon, 10 Feb 1997 10:28:37 -0800
Reply-To: jcole@precept.com
X-Mailer: Z-Mail 4.0 (4.0.0 Aug 21 1995)
To: rem-conf@es.net
Subject: MPEG Support In MBONE Tools
Cc: vic@ee.lbl.gov
Mime-Version: 1.0
Content-Type: text/plain; charset=us-ascii

I work for Precept software, and the next version of our IP/TV application
will support serving and receiving MPEG audio/video using RTP MPEG
elementary streams payload format (RFC 2038). Since IP/TV can interoperate
with vic/vat and Apple's MBONE TV app I'm curious to know if anyone has
ever attempted to roll support for MPEG video and audio into vic and vat?

Currently the intersection is H261 video and 8kHz (PCM, DVI, GSM) audio. I'm
not thinking of the public MBONE context, (where bandwidth and multicast
support are significant hurdles to widesrpead deployment), but rather
intranet environments at government, academic and corporate sites where a
cross platform networked video solution is high on the "desired" list, and
perhaps best served by different "standards-based" tools on the different
platforms which can interoperate.

We've been talking to folks in Apple's QuickTime TV engineering group, and 
they're considering adding support for MPEG to a future version of QuickTime 
TV which will interoperate with the MBONE tools. Anyone care to comment on 
the UNIX vic/vat front?

Also am curious if H263 video will likely be added as a supported video type 
to vic anytime soon?

-- 
    John Cole, Systems Engineer         Tel: 415.845.5217      
    Precept Software, Inc.              Fax: 415.845.5235
    1072 Arastradero Road               http://www.precept.com
    Palo Alto, CA 94304                 E-Mail: jcole@precept.com






From rem-conf-request@es.net Mon Feb 10 14:07:57 1997 
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Date: Mon, 10 Feb 1997 11:06:29 -0800 (PST)
From: Stephane Chatre <chatre@ENGR.ORST.EDU>
Cc: rem-conf@es.net
Subject: Question about RTCP in RFC1889
In-Reply-To: <4225643A.00370288.00@maskit1.vocaltec.co.il>
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In RFC1889, on page 17:

"The alignment requirement and a length field in the fixed
   part are included to make RTCP packets "stackable". Multiple RTCP
   packets may be concatenated without any intervening separators to
   form a compound RTCP packet that is sent in a single packet of the
   lower layer protocol, for example UDP. There is no explicit count of
   individual RTCP packets in the compound packet since the lower layer
   protocols are expected to provide an overall length to determine the
   end of the compound packet."

This paragraph is a little ambiguous to me.
Does the length field in the fixed part indicate how many RTCP packets
have been concatenated to form the current one?
Now, if as said, there is no explicit count, what does this length field
stand for? If there is no explicit count, how can we know how many packets
have been concatenated? By trying to parse what we receive?

Thanks.

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


From rem-conf-request@es.net Mon Feb 10 15:55:57 1997 
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X-Mailer: exmh version 1.6.9 8/22/96
To: jcole@precept.com
cc: rem-conf@es.net, vic@ee.lbl.gov
Subject: Re: MPEG Support In MBONE Tools
In-reply-to: Your message of "Mon, 10 Feb 1997 10:28:37 PST." <970210102839.ZM3414@little-bear>
Mime-Version: 1.0
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Date: Mon, 10 Feb 1997 12:54:44 -0800
From: Amancio Hasty <hasty@rah.star-gate.com>

>From The Desk Of jcole@precept.com :
> Also am curious if H263 video will likely be added as a supported video type 
> to vic anytime soon?

Is there a publicly available H263 codec? My understanding is that H263
is a propieratory format which if used in a public domain tool one
may have to pay royalties. The same goes for G.763 .

	Amancio





From rem-conf-request@es.net Mon Feb 10 19:57:27 1997 
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Date: Mon, 10 Feb 97 16:57:57 PST
From: braden@isi.edu
Posted-Date: Mon, 10 Feb 97 16:57:57 PST
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To: Scott_Petrack@vocaltec.com
Subject: Re: MBone Community Size?, fragility of the MBone
Cc: ggm@connect.com.au, carl@also.media.org, arlie@thepoint.net, 
    kevin@cc.gatech.edu, rem-conf@es.net


  *> What is needed are not guarantees, just lots of bandwidth of decent
  *> quality.  Television gives you no gurantees, neither do international
  *> Telephone calls, nor radio.  What it does give you is enough
  *> decent-quality bandwidth that to be useful for the purpose.  No
  *> gurantees.
  *> 
Scott,

Huh?  That seems like a surprising statement.  In fact, TV and radio
operate with very rigid service guarantees.  Each channel ("application")
gets a carefully crafted and strictly guaranteed slice of the bandwidth.
That seems to map pretty exactly into an int-serv type guarantee.  What
do you mean when you say "guarantee"??

Bob Braden

  *> The Internet and multicast promise to offer the sorts of services that
  *> news, radio, television, etc. can offer, but on a global scale rather
  *> than the current local reach which is possible. For telephony and mail,
  *> it promises richer, more flexible, cheaper function.
  *> 
  *> That's "all."
  *> 
  *> The reasons that these things are not taking off is simply that there
  *> isn't enough bandwidth of the right quality.
  *> 
  *> Of course, it might well be that the guarantess are an essential thing
  *> so that businesses can charge to get the money and incentive to upgrade
  *> the bandwidth. So it may be that we need guarantees to put in place the
  *> *financial* infrastructure to build up the internet. But it is not
  *> needed to insure usefulness of the service.
  *> 
  *> Scott
  *> 
  *> 
  *> 

From rem-conf-request@es.net Mon Feb 10 20:06:43 1997 
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To: braden@isi.edu
cc: Scott_Petrack@vocaltec.com, carl@also.media.org, arlie@thepoint.net, 
    kevin@cc.gatech.edu, rem-conf@es.net
Subject: Re: MBone Community Size?, fragility of the MBone
In-reply-to: Your message of "Mon, 10 Feb 1997 16:57:57 PST." <9702110057.AA14526@can.isi.edu>
Date: Tue, 11 Feb 1997 11:02:27 +1000
Message-ID: <4869.855622947@connect.com.au>
From: George Michaelson <ggm@connect.com.au>


  

Also interesting is the "QDU" concept which states what level of degredation
can be introduced by compression or alternate delivery methods beyond which
voice doesn't meet service level guarantees. 

I can't speak for how well it fits reality, but as a model of 'acceptable'
distortion its interesting to think how this could fit into an IP model,
perhaps layered codings could be exploited here to select "QDU-appropriate'
versions for injection down thinner and thinner pipes?

TV broadcasts certainly aim to present a dynamic range in audio which as
near as possible "flat" and this presents interesting apparent shifts in
volume as you alter from movie to ads to movie. 

Its quite plain when CNN and other offshore news sources are being sent
down compressed channels, there is clear pixelization on the image when
its zoomed in on by post-production, and the sound quality drop is very
marked.
 
I remember seeing TV engineers fiddling with old style orthicon cameras
at BBC to get colour balance right, they were very prone to thermal shifts
and there was a huge array of standard colourcards and adjustments to suit
going on all the time. Clearly they regarded some level of self-consistency
as important.

I think all this goes to re-inforce Bobs comment: in the main TV sticks
to very clear norms of message quality (ignoring meaning of content :-)

cheers

-George

From rem-conf-request@es.net Mon Feb 10 20:14:48 1997 
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From: Albert Chen <albert@julia.gcast.com>
To: "'Scott_Petrack@vocaltec.com'" <Scott_Petrack@vocaltec.com>, 
    "J.Crowcroft@cs.ucl.ac.uk" <J.Crowcroft@cs.ucl.ac.uk>
Cc: "ggm@connect.com.au" <ggm@connect.com.au>, 
    "carl@also.media.org" <carl@also.media.org>, 
    "arlie@thepoint.net" <arlie@thepoint.net>, 
    "kevin@cc.gatech.edu" <kevin@cc.gatech.edu>, 
    "rem-conf@es.net" <rem-conf@es.net>
Subject: RE: MBone Community Size?, fragility of the MBone
Date: Mon, 10 Feb 1997 17:09:23 -0800
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Content-Transfer-Encoding: quoted-printable

Scott,

Television, radio and the telephone differ from the Internet and Mbone, =
because they are networks dedicated to a specified number of sessions.  =
These sessions are the frequencies, channels or stations that are =
administered to only one provider at a time.  The Internet and Mbone are =
packet switched networks, on which many session providers are =
simultaneously vying for the same network resources.  Guarantees on =
delivery, or resource reservation, are necessary to provide high quality =
presentation.  In essence, there is bandwidth reservation on television =
and radio, because there are no two stations sharing the same frequency =
at the same time.  Same with telephone, there is not more than one =
conversation on any given circuit.  The same needs to be done for the =
Internet and Mbone.  Not having enough bandwidth is just one part of the =
problem.

Regards,

Albert Chen

GlobalCast Communications



-----Original Message-----
From:	Scott_Petrack@vocaltec.com [SMTP:Scott_Petrack@vocaltec.com]
Sent:	Monday, February 10, 1997 2:14 AM
To:	J.Crowcroft@cs.ucl.ac.uk
Cc:	ggm@connect.com.au; carl@also.media.org; arlie@thepoint.net; =
kevin@cc.gatech.edu; rem-conf@es.net
Subject:	Re: MBone Community Size?, fragility of the MBone






Scott Petrack
02/10/97 12:13 PM


-------- On 02/10/97 06:51 AM GMT  J.Crowcroft @ cs.ucl.ac.uk  wrote:
>video/audio content IS big business, but has stringent quality
>expectations which the mbone cannot do...

>actually, the quality issue is the same for the web - as soon as the
underlying net can >deliver guaratees (e.g. ipsec, int-serv, type
>guarantees) then the share delaers might think about serious use of
>the public internet - just the same as the tv and radio people

What is needed are not guarantees, just lots of bandwidth of decent
quality.
Television gives you no gurantees, neither do international Telephone
calls, nor radio.
What it does give you is enough decent-quality bandwidth that to be =
useful
for the purpose.
No gurantees.

The Internet and multicast promise to offer the sorts of services that
news, radio,
television, etc. can offer, but on a global scale rather than the =
current
local reach
which is possible. For telephony and mail, it promises richer, more
flexible, cheaper function.

That's "all."

The reasons that these things are not taking off is simply that there =
isn't
 enough
bandwidth of the right quality.

Of course, it might well be that the guarantess are an essential thing =
so
that businesses
can charge to get the money and incentive to upgrade the bandwidth. So =
it
may be that we
need guarantees to put in place the *financial* infrastructure to build =
up
the
internet. But it is not needed to insure usefulness of the service.

Scott




From rem-conf-request@es.net Mon Feb 10 20:38:59 1997 
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Date: Mon, 10 Feb 1997 17:37:36 -0800
To: rem-conf@es.net
From: alagu@mailman.apple.com (Alagu Periyannan)
Subject: Re: Question about RTCP in RFC1889

At 11:06 AM 2/10/97, Stephane Chatre wrote:
>In RFC1889, on page 17:
>
>"The alignment requirement and a length field in the fixed
>   part are included to make RTCP packets "stackable". Multiple RTCP
>   packets may be concatenated without any intervening separators to
>   form a compound RTCP packet that is sent in a single packet of the
>   lower layer protocol, for example UDP. There is no explicit count of
>   individual RTCP packets in the compound packet since the lower layer
>   protocols are expected to provide an overall length to determine the
>   end of the compound packet."
>
>This paragraph is a little ambiguous to me.
>Does the length field in the fixed part indicate how many RTCP packets
>have been concatenated to form the current one?
>Now, if as said, there is no explicit count, what does this length field
>stand for? If there is no explicit count, how can we know how many packets
>have been concatenated? By trying to parse what we receive?
>
>Thanks.
>

My understanding is that the length field specifies the length
of each RTCP packet.

The sum of these lengths must add up to the UDP data length
(after taking into account whether headers are included in
the lengths?).

The above paragraph essentially says that there is already enough
information in the packet. Hence it is not necessary to send an
explicit count of RTCP packets in a UDP packet.




Alagu Periyannan                   alagu@mailman.apple.com

Communications Products and Technology
Apple Computer, Inc.



From rem-conf-request@es.net Mon Feb 10 21:06:27 1997 
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From: Stephane Chatre <chatre@ENGR.ORST.EDU>
Reply-To: Stephane Chatre <chatre@ENGR.ORST.EDU>
To: Alagu Periyannan <alagu@mailman.apple.com>
Cc: rem-conf@es.net
Subject: Re: Question about RTCP in RFC1889
In-Reply-To: <v02110112af257bbee353@[17.255.9.131]>
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On Mon, 10 Feb 1997, Alagu Periyannan wrote:

> At 11:06 AM 2/10/97, Stephane Chatre wrote:
> >In RFC1889, on page 17:
> >
> >"The alignment requirement and a length field in the fixed
> >   part are included to make RTCP packets "stackable". Multiple RTCP
> >   packets may be concatenated without any intervening separators to
> >   form a compound RTCP packet that is sent in a single packet of the
> >   lower layer protocol, for example UDP. There is no explicit count of
> >   individual RTCP packets in the compound packet since the lower layer
> >   protocols are expected to provide an overall length to determine the
> >   end of the compound packet."
> >
> >This paragraph is a little ambiguous to me.
> >Does the length field in the fixed part indicate how many RTCP packets
> >have been concatenated to form the current one?
> >Now, if as said, there is no explicit count, what does this length field
> >stand for? If there is no explicit count, how can we know how many packets
> >have been concatenated? By trying to parse what we receive?
> >
> >Thanks.
> >
> 
> My understanding is that the length field specifies the length
> of each RTCP packet.
> 
> The sum of these lengths must add up to the UDP data length
> (after taking into account whether headers are included in
> the lengths?).
> 
> The above paragraph essentially says that there is already enough
> information in the packet. Hence it is not necessary to send an
> explicit count of RTCP packets in a UDP packet.
> 
> 
> 
> 
> Alagu Periyannan                   alagu@mailman.apple.com
> 
> Communications Products and Technology
> Apple Computer, Inc.
> 
> 
> 
Thanks to all those who answered! Everything is clear now.
The excerpt was already self-explanatory.

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



From rem-conf-request@es.net Mon Feb 10 23:58:16 1997 
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Date: Mon, 10 Feb 1997 20:29:13 -0800 ()
From: Stephen Casner <casner@precept.com>
To: Christian Zahl <zahl@fokus.gmd.de>
cc: rem-conf@es.net
Subject: Re: Inconsistence in IP/UDP/RTP compression ID
In-Reply-To: <199702072038.VAA07013@tao.fokus.gmd.de >
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Christian,

I'll try to catch up with your collection of questions...

> We are currently developing a method for transfering multiple
> RTP streams over a low-bandwidth PPP connection. The implementaion
> consists of an RTP mixer on both ends of the PPP link, which
> adjust the payload-type field according to the bandwidth
> available. So it can happen very often,

How often?  It seems to me that the rate will need to be limited in
order to maintain stability (avoid oscillation).  In particular,
changes in the direction of increasing the data rate should not be
done too quickly.  How long an integration time is needed to measure
the available bandwidth?  Considering those factors, how many packets
would be transmitted with one payload type before switching to
another, and for what size packets?  Putting those numbers together,
then you could figure out the percentage overhead caused by the need
to send the full RTP header, including what your expected CSRC list
size would be, on a payload switch.  If you have done this analysis,
please tell us about the numbers.

> Well it seems to be a little difficult to get a bit to indicate any
> change in the PT field, but it would be very helpfull to it!
> Otherwise it can happen, that the occupied bandwidth will be greater
> when adjusting the payload, than if we don't do so! Not very helpfull.
> 
> I think it makes more sense to shorten the session context id to
> 7 bit and to add a P bit, which indicates a new octet with the
> changed PT value.

I think there would be a number of people uncomfortable about reducing
the size of the context ID.

You could use multiple contexts which differ only by the payload type
value so that when the payload type changes you only have to switch to
a different context ID rather than send the whold header.  However, as
the sequence number and timestamp will likely have large changes since
the last time the new context was used, you'd have to send large
deltas for them.


> 2. CSRC change compression
> Well as described above we use a mixer and expect several sources for
> the outgoing media stream. The current ID says, that, when the CSRC
> list changes, then the whole list has to be transmitted. Because there
> are several sources and we can assume that they also change more 
> frequenctly, I think that sending the whole list ist very poor!

Do you expect to have several sources active at once?  Part of the
motivation for the current CSRC design is the expectation that, for
example in the cause of mixing an audio conference, most of the time
there will be only one person talking so the CSRC list will contain
only that one identifier.

Now, your application might be quite different such that many sources
are transmitting at once.  If so, would they stop and start
frequently, or would they be transmitting all the time such that the
CSRC list wouldn't change?  In particular, would the number of packets
sent in whatever is equivalent to a "talkspurt" be small?  If the
number of packets is large enough, then the cost of sending the CSRC
list at the beginning/end of a "talkspurt" is amortized.

> I think if would be much better to send ONLY the CSRC entries which
> have been changed, i.e. added or removed. Because the other side also
> have knowledge of the current set of CSRC entries, it can decided
> if the entry as to be add or the be removed. This results in much less
> overhead than in the current ID. 

What you have proposed would work, although it would require adding
one bit to the CC count in the compressed header so that, if the whole
list changes, there is room to include all the old identifiers being
removed and all the new ones being installed.  However, I am concerned
about a protocol mechanism that means "if you have this, delete it,
but if you don't have it, keep it".  The compressor could also wind up
feeding the decompressor more than 15 identifiers, which it would not
be able to forward in the decompressed packets.

In the conferencing example I mentioned earlier, when one person stops
talking and another person starts talking, your proposal would require
a CSRC list with two entries whereas the current spec would require
only one.  I expect this case to be pretty common for mixers.


> Ok I see. BTW, a very stupid question: how about the high byte
> of the IPv4 header? Why isn't it used like the low byte for the
> context-id?

If you mean the IPv4 length field, it contains the "Generation"
number.  See section 5.3.2 of the IPv6 Header Compression spec.

> Hm, perhaps; I don't see any difference between "Data field" and
> "data field" ?! Anyway, because I'm not familar with IPv6, I didn't
> know what the "Data field" is.

What I meant was, the IPv6 Header Compression spec defines a specific
field named "Data".  Please note that the reference here is not to
IPv6, but to the IPv6 Header Compression spec.

> But as you wrote in the ID, the IP ID
> field NORMALY increments by one, but it can also be used any other
> algorithm or by the upper layer (as defined in RFC791). So when
> the ID delta > 0x3fffff or <0, the FULL_HEADER packet HAVE to be
> transmitted.

The ID is only a 16-bit number, so the delta can't be > 0x3fffff or
<0.  Or, more precisely, any delta value can be encoded into 1 to 3
bytes by the specified encoding table.


> Here is another real example: Linux 2-x seems to generate the IP-ID
> a little bit different. The ID inclements by 1 but is send in
> little-endian encoding, so the following sequence of IP-IDs can occur
> without any loss of packages (I found this while looking into my
> PPP trace some minutes ago):
> 	IP-ID field 
> 	fe 00
> 	ff 00
> 	00 01	- FULL_HEADER needed
> 	01 01

That's unfortunate, although legal, I guess.  Can the purveyors of
Linux be convinced to htonl() the IP ID field?

> You can see that in this case the FULL_HEADER type is needed, because
> the absolute value decreases when using the network-byte-order, 
> without any kind of packet loss.

No, again, any 16-bit delta can be encoded.

> Perhaps we should think about using negative values for the delta
> values? BTW, the same can theoretically happen when two IP packets
> arrive in the wrong order (packet N+1 first, then packet N). If this
> happens often, then the compression has NO gain! So how about of
> using INTs for the delta fields instead of UNSIGNEDs. Ok, this
> will reduce the maximum transferable delta value in one byte to
> +-63. If you think this is too less, then we can say:
> 	o the 1 byte form is UNSIGNED
> 	o the 2 and 3 byte forms are SIGNED

At the IETF meeting, it was suggested that the delta encoding be
specified as a table rather than a function, with the default value of
the table being set according to the encoding currently in the spec.
(This is as a precursor to a future capability to download a new table
on a per-context basis for full entropy encoding.)  Since there are
some holes in the current encoding, I suggest using those for negative
offsets.  This keeps the encoding tuned for the common case of
positive offsets, but handles a portion of the negative offsets with
reasonable efficiency as well:

      Decimal  Hex

       -16384  C0 00 00
            :  :
         -129  C0 3F 7F
         -128  80 00
            :  :
           -1  80 7F
            0  00
            :  :
          127  7F
          128  80 80
            :  :
        16383  BF FF
        16384  C0 40 00
            :  :
      4194303  FF FF FF

Any delta < -16384 or > 4194303 would require a FULL_HEADER,
COMPRESSED_NON_TCP or COMPRESSED_UDP packet type.

This encoding still leaves one hole of size 128 from C0 3F 80 to
C0 3F FF just to keep the logical (vs. arithmetic) relationship in the
table.  We could get 128 more negative numbers by sliding down the
three-byte values instead.  I don't know if that makes a significant
difference.  (Should be none if the algorithm is table-driven.)
Comments, anyone?


> Now sender A changes the payload-type. This forces the translator T to
> send the packet as COMPRESSED_UDP instead of COMPRESSED_RTP. There
> will arrive several questions:
> 	o to which context do the outgoing packet belongs, cA or cB?
> 	  Do we have to create a new one?

cA, because that is the matching context.

> 	o will all other contexts belonging to this source (the
> 	  tanslator) will be invalidated (cA and cB) ?

No, the contexts are all independent even though they have the same
network address and ports.

> On page 5 you can read: "COMPRESSED_UDP ... It redefines the SSRC field
> of the (potential) RTP header...". So the SSRC field has no meaning
> (in my understanding), because we cannot assume that we have an
> RTP packet.

No, the context is set assuming there is an RTP header following the
UDP header.  That is how the SSRC is set.  In this example, the SSRC
in cA will be set to the same value that it already had.

> So we also cannot save any RTP header informations in the
> context (right?). Therefore, we have lost any kind of synchronization
> from the compressor to the decompressor regarding this RTP stream!
> If so, this would result in sending a FULL_HEADER next, to resync.

No, that's wrong.  The COMPRESSED_UDP packet sets the RTP part of the
context is set just the way the FULL_HEADER would have set it.  The
only difference is that FULL_HEADER also sets the IP and UDP portions
of the context.

> Or perhaps I have misunderstood one importand think: are contexts
> for the COMPRESSED_RTP and COMPRESSED_UDP are totally indepndent?

No.  There are just 256 contexts (assuming one-byte context ID, which
is all the November version of the draft specifies).

> From one point of view this seems to make sense, but on the other
> side, it seems to be impossible to have two different applications
> using the same SRC and DST addresses and ports, so when we have an
> RTP context, the UDP context is a subset of it, isn't it?

Yes.  When a UDP packet arrives, the compressor selects a context
based on the SRC and DST addresses and ports plus the bytes that are
in the SSRC position of what might be an RTP header.  The context is
saved at both the compressor and decompressor is the IP header, UDP
header, and what might be an RTP header.  If the protocol is not RTP,
then the bytes in what would be the SSRC field will probably change
all the time, which would gobble up all the contexts.  When that
happens, that set of SRC and DST addresses and ports is added to the
negative cache so compression is not attempted on those packets.  In
the case of the translator, the SRC and DST addresses and ports may
always be the same, but the SSRC field would be different for the
different sources.  If there are more than 256 sources, the context
cache will have some amount of churn depending upon how often the
sources trade off.  Or, you can use 16-bit context IDs as described in
the ipv6-hc draft (details to be added to the compressed RTP spec
soon).

> Stephen, perhaps you have an description what have to be sent in the
> scenareo shown above. I think in this point the ID is a little bit
> confusing :-)

I'd appreciate any comments you might have on what additional
information I should add to the document and where.
							-- Steve


From rem-conf-request@es.net Tue Feb 11 00:55:00 1997 
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Scott Petrack
02/11/97 07:52 AM

Let me lump together parts of a few of the replies to clarify what I mean.
Please forgive me for not attributing each thing to its original poster.

-------------------------------------------
>In fact, TV and radio operate with >very rigid service guarantees.
>Each channel ("application") gets a carefully crafted and strictly
>guaranteed slice of the bandwidth.
>That seems to map pretty exactly into an int-serv type guarantee.  What
>do you mean when you say "guarantee"??

>In essence, there is bandwidth reservation on television and radio,
because there
>are no two stations sharing the same frequency at the same time.

Apportioning the spectrum is more like UDP port numbers than int-serv
definitions
or the end-to-end *guarantees* of RSVP.
Spectrum is apportioned because
without it the whole system wouldn't work at all - there would be no way to
separate the different application streams.

By guarantee I mean "end to end guarantee of service", of the
kind that int-serv and RSVP together are going to enable. Because of many
many
factors both within and beyond human control (neighbouring buildings,
aircraft,
weather, the cheapo digitization and editing schemes used by my network,
etc.), the
thing that comes out of my box has usually pretty lousy quality. Of course
I
live in the third world and don't have cable, but even if I had cable,
I don't have a *guarantee* of quality. I can't sue CNN if they use some
horrible
digitization scheme, and I can't sue anybody if because of poor weather I
get
bad reception on my radio, or because I like classical music and the
transmitters
are much weaker than the rock stations, so if I drive too far out of the
city
then I receive classical music much worse than rock music.

There are many market pressures which give incentive to the providers to
produce
a decent image or sound in my house, but I get no *guarantees* at all.

I have nothing against int-serv and friends (well, I do, but that's not
the point here ;-)), but I do believe that guarantees and RSVP
will not solve anything at all
until there is enough bandwidth of high-enough quality.

Please read on (especially about telephony ....)
-------------------------------------------------------------

>Its quite plain when CNN and other offshore news sources are being sent
>down compressed channels, there is clear pixelization on the image when
>its zoomed in on by post-production, and the sound quality drop is very
marked.

And you certainly can't sue CNN or anybody else because of the quality
drop.
You have no guarantee at all. You just have some guys who are "trying their
best", and who have serious market pressure to keep the quality up. No
guarantees. BTW, I agree, you see and hear more and more awful digital
imagery and audio on TV, precisely because it's cheaper to produce and they
have no obligation to give you anything better

>I remember seeing TV engineers fiddling with old style orthicon cameras
>at BBC to get colour balance right, they were very prone to thermal shifts
and >there was a huge array of standard colourcards and adjustments to suit
going on >all the time. Clearly they regarded some level of
self-consistency as important.

I would even say, "they regarded the quality of end
image and audio that came out your endpoint as important." They did their
best.
But they did not use any guarantees to force themselves to deliver that
quality.

>I think all this goes to re-inforce Bobs comment: in the main TV sticks
>to very clear norms of message quality (ignoring meaning of content :-)

I think that it does the opposite: it shows that the infrastructure has
developed to ensure that the quality is "good enough" - but there are no
*guarantees* that what will come out of your box has any delay/noise or
other measureable quality characteristics.

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

>Same with telephone, there is not more than one conversation on any given
circuit.

I still get cross talk occasionally, but more important, I claim that
telephony gives absolutely no guarantees of ANY kind. You can (and do) get
long delays, lousy coders, high packet loss. Try making an international
phone call and get music-on-hold, you'll hear about 20-30% packet loss
because
of the ECI boxes and silence deletion. This makes it difficult or
impossible to
make an international call over a modem.

Competition has driven the prices and quality way down, and there is no
carrier
that I know that offers an phone call with a guaranteed delay or audio
quality.
(If there were I would actually use it, because it does happen that I need
to make a modem call from the US to Israel occasionally for something
sensitive).

I note with interest that in the telephone world, the fastest growing
segment
is *mobile* stuff, which has terrible quality, and definitely offers a
"best-effort" service only. No one is suggesting that we need guarantees
there -
we just need more bandwidth of higher quality so that best-effort will be
"good enough."

>Not having enough bandwidth is just one part of the problem.

I agree, that's why in original note I said we need lots of
"decent-quality bandwidth".
The stuff has to be "good enough", like television or radio or telephone.
What I don't need are guarantees (in the sense that I defined above).

If int-serv were to be used as a way of measuring and comparing internet
infrastructure, so that we can quantify and then improve quality of
service,
this would be a great thing. What worries me is that
people are going to insist on *guarantees* for things that are not
necessary
and impossible to produce, and therefore they will decide that the whole
internet thing isn't very useful.

Scott (Petrack)

P.S. Question for Miss Manners:

Do I really need to send replies individually like everyone else does?
Or is it enough to reply to rem-conf? Surely everyone who wrote subscribes
to rem-conf, so a single reply to rem-conf is enough?



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Scott Petrack
02/11/97 08:27 AM

Let me lump together parts of a few of the replies to clarify what I mean.
Please forgive me for not attributing each thing to its original poster.

-------------------------------------------
>In fact, TV and radio operate with >very rigid service guarantees.
>Each channel ("application") gets a carefully crafted and strictly
>guaranteed slice of the bandwidth.
>That seems to map pretty exactly into an int-serv type guarantee.  What
>do you mean when you say "guarantee"??

>In essence, there is bandwidth reservation on television and radio,
because there
>are no two stations sharing the same frequency at the same time.

Apportioning the spectrum is more like UDP port numbers than int-serv
definitions
or the end-to-end *guarantees* of RSVP.
Spectrum is apportioned because
without it the whole system wouldn't work at all - there would be no way to
separate the different application streams.

By guarantee I mean "end to end guarantee of service", of the
kind that int-serv and RSVP together are going to enable. Because of many
many
factors both within and beyond human control (neighbouring buildings,
aircraft,
weather, the cheapo digitization and editing schemes used by my network,
etc.), the
thing that comes out of my box has usually pretty lousy quality. Of course
I
live in the third world and don't have cable, but even if I had cable,
I don't have a *guarantee* of quality. I can't sue CNN if they use some
horrible
digitization scheme, and I can't sue anybody if because of poor weather I
get
bad reception on my radio, or because I like classical music and the
transmitters
are much weaker than the rock stations, so if I drive too far out of the
city
then I receive classical music much worse than rock music.

There are many market pressures which give incentive to the providers to
produce
a decent image or sound in my house, but I get no *guarantees* at all.

I have nothing against int-serv and friends (well, I do, but that's not
the point here ;-)), but I do believe that guarantees and RSVP
will not solve anything at all
until there is enough bandwidth of high-enough quality.

Please read on (especially about telephony ....)
-------------------------------------------------------------

>Its quite plain when CNN and other offshore news sources are being sent
>down compressed channels, there is clear pixelization on the image when
>its zoomed in on by post-production, and the sound quality drop is very
marked.

And you certainly can't sue CNN or anybody else because of the quality
drop.
You have no guarantee at all. You just have some guys who are "trying their
best", and who have serious market pressure to keep the quality up. No
guarantees. BTW, I agree, you see and hear more and more awful digital
imagery and audio on TV, precisely because it's cheaper to produce and they
have no obligation to give you anything better

>I remember seeing TV engineers fiddling with old style orthicon cameras
>at BBC to get colour balance right, they were very prone to thermal shifts
and >there was a huge array of standard colourcards and adjustments to suit
going on >all the time. Clearly they regarded some level of
self-consistency as important.

I would even say, "they regarded the quality of end
image and audio that came out your endpoint as important." They did their
best.
But they did not use any guarantees to force themselves to deliver that
quality.

>I think all this goes to re-inforce Bobs comment: in the main TV sticks
>to very clear norms of message quality (ignoring meaning of content :-)

I think that it does the opposite: it shows that the infrastructure has
developed to ensure that the quality is "good enough" - but there are no
*guarantees* that what will come out of your box has any delay/noise or
other measureable quality characteristics.

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

>Same with telephone, there is not more than one conversation on any given
circuit.

I still get cross talk occasionally, but more important, I claim that
telephony gives absolutely no guarantees of ANY kind. You can (and do) get
long delays, lousy coders, high packet loss. Try making an international
phone call and get music-on-hold, you'll hear about 20-30% packet loss
because
of the ECI boxes and silence deletion. This makes it difficult or
impossible to
make an international call over a modem.

Competition has driven the prices and quality way down, and there is no
carrier
that I know that offers an phone call with a guaranteed delay or audio
quality.
(If there were I would actually use it, because it does happen that I need
to make a modem call from the US to Israel occasionally for something
sensitive).

I note with interest that in the telephone world, the fastest growing
segment
is *mobile* stuff, which has terrible quality, and definitely offers a
"best-effort" service only. No one is suggesting that we need guarantees
there -
we just need more bandwidth of higher quality so that best-effort will be
"good enough."

>Not having enough bandwidth is just one part of the problem.

I agree, that's why in original note I said we need lots of
"decent-quality bandwidth".
The stuff has to be "good enough", like television or radio or telephone.
What I don't need are guarantees (in the sense that I defined above).

If int-serv were to be used as a way of measuring and comparing internet
infrastructure, so that we can quantify and then improve quality of
service,
this would be a great thing. What worries me is that
people are going to insist on *guarantees* for things that are not
necessary
and impossible to produce, and therefore they will decide that the whole
internet thing isn't very useful.

Scott (Petrack)

P.S. Question for Miss Manners:

Do I really need to send replies individually like everyone else does?
Or is it enough to reply to rem-conf? Surely everyone who wrote subscribes
to rem-conf, so a single reply to rem-conf is enough?



From rem-conf-request@es.net Tue Feb 11 10:13:02 1997 
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To: mbone@isi.edu
Cc: rem-conf@es.net, mbone-seminars@cl.cam.ac.uk
Subject: Any *FIRST HAND* use of Linux Boxes to source MBone video ?
X-uri: <URL:http://www.cl.cam.ac.uk/users/pb>
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From: Piete Brooks <mbone-seminars@cl.cam.ac.uk>
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Date: Tue, 11 Feb 1997 15:11:51 +0000
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We have a very old and slow dec3000/300lx running osf1_3.2D with a J300 board 
which we use to transmit our Security Seminars (using nv, as it won't work 
unver vic) which we'd like to replace with a linux box and frame grabber.

A member of the dept bought a Matrox Millenium so that they could source 
video, but so far they've been unable to get it to work as Matrox will not 
release the necessary info to write a Linux driver :-(((

SO: is anyone using a Linux box to transmit video to the MBone using available
    hardware and an external video source (Quick Cam is not sifficient !) ?
    If so, PLEASE let me know !


From rem-conf-request@es.net Tue Feb 11 10:31:20 1997 
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          Tue, 11 Feb 1997 07:29:54 -0800
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Date: Tue, 11 Feb 1997 16:26:50 +0100 (MET)
Message-Id: <199702111526.QAA15917@maillol.inria.fr>
From: Tie Liao <Tie.Liao@inria.fr>
To: java-networking@cdt.luth.se, rem-conf@es.net
Cc: Bernard.Martin@inria.fr, Jean-Francois.Abramatic@inria.fr, 
    Yves.Peynaud@inria.fr, Catherine.Chat@inria.fr, Patrick.Duval@inria.fr, 
    Philipp.Hoschka@inria.fr, rodgers@nlm.nih.gov, jct@edelweb.fr, 
    Benoit.Boute@enst.fr, likavec@igd.fhg.de, Christian.Donot@inria.fr, 
    Tie.Liao@inria.fr
Subject: Announce: WebCanal - a Mutlicast Web Application
Reply-to: Tie.Liao@inria.fr

Sorry if you receive two times the same message.

======================================================================
We are pleased to announce that a multicast web application - WebCanal is
made available on our ftp site. This application, developed by INRIA,
is still in a beta test version. It is aimed at sharing your web pages on
the MBONE, either they are maintained on a Web server or just stored as
local files. The main features include:

  o broadcast documents in a multicast channel, for example, broadcast
    latest news or update common files. This can be performed as a
    background task.

  o interactively exchange documents with other users in a multicast
    session using a web browser.

  o all received documents are cached and can be reviewed.

  o document transfer is optimized so that unnecessary document transfer
    is generally avoided.

WebCanal is designed to be network friendly, it can work at low bit rate for
most Web documents including small inlined images.

WebCanal is based on the Light-weight Reliable Multicast Protocol (LRMP)
which makes use of the NACK mechanism described in SRM. But it is not a
simple copy of SRM, it also addresses the problems related to packet
ordering and rate based flow control.

WebCanal is implemented in Java and requires JDK1.1 to run.

To download the package, please go to:

  ftp://ftp.inria.fr/INRIA/Actions/webcanal

This directory includes mainly the following files:

  webcanal-1.0b.tar.gz - the binary files, documentation, etc. It should be
      enough for most users.

  webcanal-1.0b.src.tar.gz - the source package.

See the copyright notice included in the package for copyright information.
For other information, please look at:

  http://monet.inria.fr/

Suggestions and questions could be sent to Tie.Liao@inria.fr


Date: Tue Feb 11 14:49:39 MET 1997

From rem-conf-request@es.net Tue Feb 11 11:23:21 1997 
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Date: Tue, 11 Feb 1997 18:22:06 +0200 (EET)
Message-Id: <199702111622.SAA07044@silver.sms.fi>
From: Petri Helenius <pete@sms.fi>
To: mbone-seminars@cl.cam.ac.uk
Cc: mbone@isi.edu, rem-conf@es.net
Subject: Any *FIRST HAND* use of Linux Boxes to source MBone video ?
In-Reply-To: <E0vuJsA-00039W-00@heaton.cl.cam.ac.uk>
References: <E0vuJsA-00039W-00@heaton.cl.cam.ac.uk>

Piete Brooks writes:
 > We have a very old and slow dec3000/300lx running osf1_3.2D with a J300 board 
 > which we use to transmit our Security Seminars (using nv, as it won't work 
 > unver vic) which we'd like to replace with a linux box and frame grabber.
 > 
 > A member of the dept bought a Matrox Millenium so that they could source 
 > video, but so far they've been unable to get it to work as Matrox will not 
 > release the necessary info to write a Linux driver :-(((

Matrox Millenium is a display board, you don't get it to source video
even if you would have all the information on the card. So I assume
you must be talking about Matrox Meteor.
 > 
 > SO: is anyone using a Linux box to transmit video to the MBone using available
 >     hardware and an external video source (Quick Cam is not sifficient !) ?
 >     If so, PLEASE let me know !
 > 
So since you have the hardware only thing you have to do is to upgrade
the software. Install FreeBSD and the Meteor support will come with
it.

Pete

From rem-conf-request@es.net Tue Feb 11 11:34:14 1997 
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From: Arlie Davis <arlie@thepoint.net>
To: "'Scott_Petrack@vocaltec.com'" <Scott_Petrack@vocaltec.com>
Cc: "'rem-conf@es.net'" <rem-conf@es.net>
Subject: RE:
Date: Tue, 11 Feb 1997 11:29:58 -0500
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>----------
>From: 	Scott_Petrack@vocaltec.com
>Sent: 	Tuesday, February 11, 1997 12:52 AM
>To: 	rem-conf@es.net
>Cc: 	Arlie Davis; albert@julia.gcast.com; ggm@connect.com.au;
>carl@also.media.org; kevin@cc.gatech.edu; J.Crowcroft@cs.ucl.ac.uk
>
>Apportioning the spectrum is more like UDP port numbers than int-serv
>definitions
>or the end-to-end *guarantees* of RSVP.

I strongly disagree.  True, spectrum allocation serves to distinguish
the services / channels, like UDP port numbers.  However, one station
can send as much "traffic" as its wants without ever interfering with
any other station at all.  This stands in stark contrast to UDP, where
one source can easily overwhelm (or render useless) all other sources.


>

From rem-conf-request@es.net Tue Feb 11 11:36:19 1997 
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Date: Tue, 11 Feb 1997 16:32:38 +0100 (MET)
From: Oliver Frommel <oliver@AEC.at>
To: Petri Helenius <pete@sms.fi>
cc: mbone-seminars@cl.cam.ac.uk, mbone@isi.edu, rem-conf@es.net
Subject: Re: Any *FIRST HAND* use of Linux Boxes to source MBone video ?
In-Reply-To: <199702111622.SAA07044@silver.sms.fi>
Message-ID: <Pine.LNX.3.91.970211163121.31321B-100000@web.aec.at>
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On Tue, 11 Feb 1997, Petri Helenius wrote:

> Piete Brooks writes:
>  > We have a very old and slow dec3000/300lx running osf1_3.2D with a J300 board 
>  > which we use to transmit our Security Seminars (using nv, as it won't work 
>  > unver vic) which we'd like to replace with a linux box and frame grabber.
>  > 
>  > A member of the dept bought a Matrox Millenium so that they could source 
>  > video, but so far they've been unable to get it to work as Matrox will not 
>  > release the necessary info to write a Linux driver :-(((
> 
> Matrox Millenium is a display board, you don't get it to source video
> even if you would have all the information on the card. So I assume
> you must be talking about Matrox Meteor.
>  > 
>  > SO: is anyone using a Linux box to transmit video to the MBone using available
>  >     hardware and an external video source (Quick Cam is not sifficient !) ?
>  >     If so, PLEASE let me know !
>  > 
> So since you have the hardware only thing you have to do is to upgrade
> the software. Install FreeBSD and the Meteor support will come with
> it.

Linux does also support the MAtrox Meteor ..
take a look at sunsite.unc.edu's Linux archive directory apps/video:

meteor-1.4a.tgz 
     driver for the matrox meteor frame grabber card 


oliver


From rem-conf-request@es.net Tue Feb 11 12:02:45 1997 
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Subject: Re: Any *FIRST HAND* use of Linux Boxes to source MBone video ?
From: Matti Aarnio <mea@utu.fi>
To: pete@sms.fi (Petri Helenius)
Date: Tue, 11 Feb 1997 19:00:39 +0200 (EET)
Cc: mbone-seminars@cl.cam.ac.uk, mbone@isi.edu, rem-conf@es.net
In-Reply-To: <199702111622.SAA07044@silver.sms.fi> from "Petri Helenius" at Feb 11, 97 06:22:06 pm
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Petri Helenius writes: 
> Piete Brooks writes:
...
> Matrox Millenium is a display board, you don't get it to source video
> even if you would have all the information on the card. So I assume
> you must be talking about Matrox Meteor.

>  > SO: is anyone using a Linux box to transmit video to the MBone using available
>  >     hardware and an external video source (Quick Cam is not sifficient !) ?
>  >     If so, PLEASE let me know !
>  > 
> So since you have the hardware only thing you have to do is to upgrade
> the software. Install FreeBSD and the Meteor support will come with it

Or pick proper driver:
---- /pub/mirrors/sunsite.unc.edu/pub/Linux/apps/video/meteor-1.4a.lsm ----
Begin3
Title:          meteor
Version:        1.4a
Entered-date:   05SEP96
Description:    A driver for the matrox meteor frame grabber card, as a
                kernel module. This is a port from the FreeBSD driver.
Keywords:       meteor, driver, video, frame grabber, matrox, module
Author:         (Jim Lowe, Mark Tinguely, Jim Bray)
Maintained-by:  ian@robots.ox.ac.uk
Primary-site:   ftp.rwii.com /pub/linux/system/Meteor
                52959 meteor-1.4a.tgz
Alternate-site: sunsite.unc.edu /pub/Linux/apps/video
                52959 meteor-1.4a.tgz
Original-site:  joy.cs.ndsu.nodak.edu /pub
                39621 meteor.1.11.tgz
Platforms:      kernel 1.3.72 and later with the (provided) bigphysarea
                patch by Matt Welsh
                Obviously a Matrox Meteor frame grabber card.
Copying-policy: (C)
End

> Pete

	I recall a friend of mine told me on friday that he has integrated
	Meteor driver into vic 2.8.  I must get it for storing into
	  ftp://ftp.funet.fi/pub/networking/services/multicast/LINUX/
	subtree..

	/Matti Aarnio <mea@utu.fi>

From rem-conf-request@es.net Tue Feb 11 17:02:24 1997 
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Date: Tue, 11 Feb 1997 17:01:20 -0500
From: Raymond Liao <liao@ctr.columbia.edu>
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CC: mnl@ctr.columbia.edu
Subject: Seminar: H. Shulzrinne, "RTP Scaling over Very Large Groups"
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MBone Broadcast Announcement

Title:
    RTP SCALING FOR VERY LARGE GROUPS

Speaker:
    Prof. Henning Schulzrinne
    Columbia University

Date:
    Friday, January 24, 1997

Time:
    10:00 AM - 11:00 AM EST

Multicast Address:
    audio:  DVI, RTP, 224.2.243.158/30528/127
    video:  H.261, RTP, 224.2.154.154/59442/127

Contact:
    Andrew T. Campbell (campbell@ctr.columbia.edu)

URL:
    http://comet.ctr.columbia.edu/activities/seminars/

Place:
    Multimedia Network Laboratory
    8LE1 Schapiro Research Building
    Center for Telecommunications Research
    Columbia University
    New York City, USA

Abstract:

We describe, analyze and simulate algorithms that allow scaling of
feedback in very large multicast groups with potentially millions of
members, both for any-to-any feedback and any-to-one feedback.  While
the techniques generalize, our particular focus is on feedback
mechanisms suitable for the real-time transport protocol (RTP).  To
suppress the initial flood of packets occuring when a large number of
users simultaneously join a session, we propose a new, but
backward-compatible algorithm, called reconsideration, which minimizes
the initial transient.  Another algorithm quickly ``teaches'' new
arrivals to a group the current group size.  We then describe polling
(any-to-one) algorithms that quickly estimate multicast populations.
Adaptive statistical sampling may be used to reduce the state storage
overhead incurred by tracking large user populations for either polling
or the reconsideration algorithms.


-- 
Raymond R.-F. Liao
(212) 939-7158
http://www.ctr.columbia.edu/~liao
Center for Telecommunications Research, Columbia University

From rem-conf-request@es.net Tue Feb 11 17:10:29 1997 
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To: mbone-seminars@cl.cam.ac.uk
cc: mbone@isi.edu, rem-conf@es.net
Subject: Re: Any *FIRST HAND* use of Linux Boxes to source MBone video ?
In-reply-to: Your message of "Tue, 11 Feb 1997 15:11:51 GMT." <E0vuJsA-00039W-00@heaton.cl.cam.ac.uk>
Mime-Version: 1.0
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Date: Tue, 11 Feb 1997 14:09:15 -0800
From: Amancio Hasty <hasty@rah.star-gate.com>

>From The Desk Of Piete Brooks :
> We have a very old and slow dec3000/300lx running osf1_3.2D with a J300 board
 
> which we use to transmit our Security Seminars (using nv, as it won't work 
> unver vic) which we'd like to replace with a linux box and frame grabber.

Well, there is a matrox meteor driver that comes with FreeBSD;additionally,
the FreeBSD Matrox Meteor has been ported to Linux. It is kind of nice
that the driver is available for both platforms because we can benefit
>from bug fixes and enhancements. 

As for programming info, the Matrox Meteor is a simple video capture
which has to chipsets : Philips SAA 7116 and SAA 7196 . So to learn
how to program the Meteor you can download the docs for those chipsets
>from Philips Web Site.

I wrote a device driver for the Intel Smart Video Recorder III with 
sufficient functionality to support vic as well as my simple video
tool "tv" which I use to watch TV on my system.  The driver is 
based upon the Matrox Meteor one so it uses the exact same interface;however,
I must warn you that the SAA 7196 and the Bt848 are very different
chipsets.

There is no PAL support however it should not be difficult to add PAL 
support. The readme file has contact info for getting the programming docs for the card.
Architectually, the driver is very simple and should not be difficult
to port to Linux.

The driver is really for the BrookTree Bt848 chipset which means that
it should work with any Bt848 chipset based boards such as the Hauppauge's
Wincast/tv. One of the FreeBSD users reported great success with 
the WinCast/tv card on the first try.

The driver is available at : ftp://rah.star-gate.com/pub/bt848.tar.gz

The driver so far works best with Triton based motherboards and to a lesser
extent with Natoma chipset based motherboards. For instance, I can
get smooth video at 640x480 32bits on my P100 / Triton system but 
a few more dma errors on my PPRO 200Mhz /Natoma system. It is fairly
safe to assume that the Natoma chipset is having problems with
substain high video data rate.

Last but not least the primary motivation for writing the Bt848 driver
is that the SAA 7116 is not PCI 2.1 compliant and in certain instances
can lock up solid a PPRO / Natoma system. Yes, I know the work around
that some have found however independent tests on my PPRO systems
shows that it can still crash a PPRO 200Mhz system.

	Amancio



From rem-conf-request@es.net Tue Feb 11 17:13:25 1997 
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Sorry for the resent, the date was wrong.

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

MBone Broadcast Announcement

Title:
    RTP SCALING FOR VERY LARGE GROUPS

Speaker:
    Prof. Henning Schulzrinne
    Columbia University

Date:
    Thursday, Feb. 13, 1997

Time:
    10:00 AM - 11:00 AM EST

Multicast Address:
    audio:  DVI, RTP, 224.2.243.158/30528/127
    video:  H.261, RTP, 224.2.154.154/59442/127

Contact:
    Andrew T. Campbell (campbell@ctr.columbia.edu)

URL:
    http://comet.ctr.columbia.edu/activities/seminars/

Place:
    Multimedia Network Laboratory
    8LE1 Schapiro Research Building
    Center for Telecommunications Research
    Columbia University
    New York City, USA

Abstract:

We describe, analyze and simulate algorithms that allow scaling of
feedback in very large multicast groups with potentially millions of
members, both for any-to-any feedback and any-to-one feedback.  While
the techniques generalize, our particular focus is on feedback
mechanisms suitable for the real-time transport protocol (RTP).  To
suppress the initial flood of packets occuring when a large number of
users simultaneously join a session, we propose a new, but
backward-compatible algorithm, called reconsideration, which minimizes
the initial transient.  Another algorithm quickly ``teaches'' new
arrivals to a group the current group size.  We then describe polling
(any-to-one) algorithms that quickly estimate multicast populations.
Adaptive statistical sampling may be used to reduce the state storage
overhead incurred by tracking large user populations for either polling
or the reconsideration algorithms.


-- 
Raymond R.-F. Liao
(212) 939-7158
http://www.ctr.columbia.edu/~liao
Center for Telecommunications Research, Columbia University

From rem-conf-request@es.net Tue Feb 11 19:32:16 1997 
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  *> 
  *> Apportioning the spectrum is more like UDP port numbers than int-serv
  *> definitions or the end-to-end *guarantees* of RSVP.  Spectrum is
  *> apportioned because without it the whole system wouldn't work at all -
  *> there would be no way to separate the different application streams.
  *> 
Scott,

That is a good point.  In analog transmission spectrum dedication
serves TWO functions:  "addressing" as well as guarantee of
non-interference from other signals.


  *> By guarantee I mean "end to end guarantee of service", of the kind that
  *> int-serv and RSVP together are going to enable. Because of many many
  *> factors both within and beyond human control (neighbouring buildings,
  *> aircraft, weather, the cheapo digitization and editing schemes used by
  *> my network, etc.), the thing that comes out of my box has usually
  *> pretty lousy quality. Of course I live in the third world and don't
  *> have cable, but even if I had cable, I don't have a *guarantee* of
  *> quality. I can't sue CNN if they use some horrible digitization scheme,
  *> and I can't sue anybody if because of poor weather I get bad reception
  *> on my radio, or because I like classical music and the transmitters are
  *> much weaker than the rock stations, so if I drive too far out of the
  *> city then I receive classical music much worse than rock music.

Integrated service guarantees are essentially of dedicated bandwidth,
though it is packaged as a bound on queueing delay.  They give no
guarantee of reliability (bits can still be dropped) or of quality of
the data (the sender can be doing bad things).  So the parallel is
still pretty strong to radio or TV.  [I am pursuing this point because
there has been a tendency to assume too much about the "guarantees" of
integrated service; another version of this confusion is a huge
overloading of the term QoS.]

  *> 
  *> >I think all this goes to re-inforce Bobs comment: in the main TV sticks
  *> >to very clear norms of message quality (ignoring meaning of content :-)
  *> 
  *> I think that it does the opposite: it shows that the infrastructure has
  *> developed to ensure that the quality is "good enough" - but there are no
  *> *guarantees* that what will come out of your box has any delay/noise or
  *> other measureable quality characteristics.
  *> 

I guess we are at a stand-still here, but I will say it again: "Good
enough" is exactly what the inventors of integrated services hope to
provide you on the Internet.


  *> 
  *> P.S. Question for Miss Manners:
  *> 
  *> Do I really need to send replies individually like everyone else does?
  *> Or is it enough to reply to rem-conf? Surely everyone who wrote subscribes
  *> to rem-conf, so a single reply to rem-conf is enough?
  *> 

Do what works best for you.

Bob Braden


From rem-conf-request@es.net Tue Feb 11 19:39:00 1997 
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To: rem-conf@es.net
Subject: Re: MBone Community Size?, fragility of the MBone
In-reply-to: Your message of "Tue, 11 Feb 1997 08:27:24 +0200." <4225643B.001F43BF.00@maskit1.vocaltec.co.il>
Date: Wed, 12 Feb 1997 10:38:18 +1000
Message-ID: <29586.855707898@connect.com.au>
From: George Michaelson <ggm@connect.com.au>



Ok. I'd like to revise my comments a bit.

In the case of TV, quality is a mixed meaning word. At the source end, before
transmission is introduced, I believe TV signals and aspects of that signal
such as colour balance, soundmix, relative signal strength etc etc is managed
under pretty rigorous specs. This is as much a function of history as anything
else since signal loss under digital method should be much smaller than the
requirements to control this kind of quality, but allowing for tape based
methods of storage and edit, there is a quality loss issue in creating the
final image, and they *need* that kind of QA to ensure the final product
is what they loosely call 'broadcast quality'. Broadly speaking this is
the same as the QDU issue for telecomms, but its pre-transmission stuff.

When I've talked to the next rung down, the professional video production
teams doing private video production for corporates and the like, the issue
is the same but the compromises are a bit more lax. The discussion is usually
about how S-VHS is now viable compared to betacamPRO, and that non-linear
edit on S-VHS is as good as traditional edit on a betacam edit suite. The
broadcasters would not regard S-VHS as being what they call 'broadcast quality'
yet (I'd be very happy to be proven wrong)

Once transmission is introduced, there is still a QA issue. The transmission
engineers are pretty active in terms of monitoring signal quality and in
attempting adjustments of antenna design and signal injection to overcome
physical limits. In the UK, this often used to take the form of helping
communities achieve a viable group aerial to overcome local problems. In
Australia, its about regional upgrades to transform VHF to UHF, or to increase
availablility beyond one merged channel of commercial plus the national public
broadcaster. 

Its interesting to think about how they do this given the lack
of a real feedback channel in the broadcast method. Perhaps they don't 
formalize it, but do in fact have a TV picking up the real signal and not
showing foldback from the source. I sure hope so.

However we *are* talking about a unidirectional protocol, with no flow control
or feedback. Its not plausible to say there are going to be the quality issues
you get from having something like RTCP in the loop. I suspect that the UDLR
people are likely to say similar things, that the encodings used to send
material should attempt to provide redundancy and alternate codings to permit
the final product to get as close as possible to the sent content, but that
for some people, reception is always an approximation.

Jon comments that you can't say there is consistent quality comparing PAL to
SECAM to NTSC. I think that within each standard, you'd find the standard
attempted  to define what (hah) gracefull failure each should display under
signal loss, and what were considered the key content. In the case of NTSC
clearly motion is favoured over chroma for instance. In the case of PAL
I believe the encoding was designed to permit b&w signal to exist as a parasite
under colour encoding so the existing user-base didn't have a major upgrade
hassle (excluding the VHF-UHF transition)

Does anybody expect that under digital TV we'll see reception reports come
in as a feature, to permit the broadcasters to monitor real reception quality
and make live adjustment? One imagines that MPEG can display highly 
differential quality under the variety of motion which is subject/edit 
dependant, and that you could fiddle the knobs on the compression box to
attempt to preserve motion aspects at the cost of total bitrate for sports
but pull it back for other forms of media.

-George

From rem-conf-request@es.net Wed Feb 12 02:14:34 1997 
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Date: Wed, 12 Feb 1997 08:13:40 +0100 (MET)
From: Dietrich Thie <dietrich.thie@informatik.tu-chemnitz.de>
To: Piete Brooks <mbone-seminars@cl.cam.ac.uk>
cc: mbone@isi.edu, rem-conf@es.net
Subject: Re: Any *FIRST HAND* use of Linux Boxes to source MBone video ?
In-Reply-To: <E0vuJsA-00039W-00@heaton.cl.cam.ac.uk>
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On Tue, 11 Feb 1997, Piete Brooks wrote:

> We have a very old and slow dec3000/300lx running osf1_3.2D with a J300 board 
> which we use to transmit our Security Seminars (using nv, as it won't work 
> unver vic) which we'd like to replace with a linux box and frame grabber.
> 
> A member of the dept bought a Matrox Millenium so that they could source 
> video, but so far they've been unable to get it to work as Matrox will not 
> release the necessary info to write a Linux driver :-(((

As far as I know, XFree86 3.2 supports Matrox Millenium, see
http://www.bf.rmit.edu.au/~ajv/xf86-matrox.html

if it is a Millenium with frame grabber modul, you are right, the modul is
not supported yet. But why you don't use the Matrox Meteor frame grabber.
In my mind the MBone tools from LBL should play with it.
> 
> SO: is anyone using a Linux box to transmit video to the MBone using available
>     hardware and an external video source (Quick Cam is not sifficient !) ?
>     If so, PLEASE let me know !
> 
> 

Dietrich Thie

phone   :      +49 371 531-1532
fax     :      +49 371 531-1629
$HOME   :      Lungwitzer Str. 12, 09337 Hohenstein-Er./Germany
phone   :      +49 3723 48506 | mobile +49 177 3331626

>>  Und aus dem Chaos sprach eine Stimme: "Sei froh und laechle, 
    denn es koennte schlimmer kommen ... "
    ... Und ich laechelte und war froh und es kam schlimmer !




From rem-conf-request@es.net Wed Feb 12 02:26:32 1997 
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From: Greg Kulosa <greg@Synopsys.COM>
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Subject: Re: Any *FIRST HAND* use of Linux Boxes to source MBone video ?
To: mbone-seminars@cl.cam.ac.uk
Date: Tue, 11 Feb 1997 23:24:57 -0800 (PST)
Cc: mbone@isi.edu, rem-conf@es.net
In-Reply-To: <E0vuJsA-00039W-00@heaton.cl.cam.ac.uk> from "Piete Brooks" at Feb 11, 97 03:11:51 pm
Content-Type: text



> We have a very old and slow dec3000/300lx running osf1_3.2D with a J300 board 
> which we use to transmit our Security Seminars (using nv, as it won't work 
> unver vic) which we'd like to replace with a linux box and frame grabber.
> 
> A member of the dept bought a Matrox Millenium so that they could source 
> video, but so far they've been unable to get it to work as Matrox will not 
> release the necessary info to write a Linux driver :-(((
> 
> SO: is anyone using a Linux box to transmit video to the MBone using available
>     hardware and an external video source (Quick Cam is not sifficient !) ?
>     If so, PLEASE let me know !

I have used the QuickCAM with success.  However, you say that it won't
work for you.

I am working on support for the "AITech AIGotcha" Parallel-port
frame grabber.  It claims to do "real-time" video, and costs ~$149 US.

It probably won't be able to do 30 frames/second, but I don't need that.
5-8 frames/second or so is _plenty_ for me and the MBONE.

I wrote an E-mail message to info@aitech.com, and they said they would be
releasing a developers kit "in a month".  When I receive that information,
I will work on a driver.  My main focus is to get "vic" support ASAP.

-- 
Greg A. Kulosa          | "The avalanche has already started, it is too
Systems Administrator   |  late for the pebbles to vote." - Ambassador Kosh
Synopsys, Inc.          |___________________________________________________
greg@synopsys.com       700 E. Middlefield Rd, Mountain View, CA 94043

From rem-conf-request@es.net Wed Feb 12 04:59:12 1997 
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Date: Wed, 12 Feb 1997 10:56:55 +0100
From: Hans Mayer <Hans.Mayer@gmd.de>
Message-Id: <199702120956.AA27107@mail.gmd.de>
To: hasty@rah.star-gate.com, jcole@precept.com
Subject: Re: MPEG Support In MBONE Tools
Cc: rem-conf@es.net, vic@ee.lbl.gov

>>Also am curious if H263 video will likely be added as a supported video type 
>>to vic anytime soon?
 
>Is there a publicly available H263 codec? My understanding is that H263
>is a propieratory format which if used in a public domain tool one
>may have to pay royalties. The same goes for G.763 .

There is a sample codec implementation available at telenor, the Norwegian
Telecom Research Institute. I have it and am looking into vic to find a
way to interface it (well, actually, at the moment I'm struggling with
native ATM support :-( ). If someone else already has made some progress in
either writing their own codec or interfaceing the telenor one I would be
very interested. There should be no licensing fees required if you don't
make a commercial product out of it.

Hans

From rem-conf-request@es.net Wed Feb 12 05:12:15 1997 
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X-Mailer: exmh version 1.6.9 8/22/96
To: Greg Kulosa <greg@Synopsys.COM>
cc: mbone-seminars@cl.cam.ac.uk, mbone@isi.edu, rem-conf@es.net
Subject: Re: Any *FIRST HAND* use of Linux Boxes to source MBone video ?
In-reply-to: Your message of "Tue, 11 Feb 1997 23:24:57 PST." <199702120724.XAA04109@jaxom.synopsys.com>
Mime-Version: 1.0
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Date: Wed, 12 Feb 1997 02:11:28 -0800
From: Amancio Hasty <hasty@rah.star-gate.com>

>From The Desk Of Greg Kulosa :
> 
> It probably won't be able to do 30 frames/second, but I don't need that.
> 5-8 frames/second or so is _plenty_ for me and the MBONE.

Well, the price of fast frame grabbers are dropping for instance
Hauppage's WinCast/Tv retails for about $149(US) . This is a PCI 
card with a Bt848 chipset and a tuner .

I bought a CCD camera for about $200 from Computer Modules
(http://www.compmod.com) . The camera has a builtin mic.
So for about $349 you can have a fairly decent video subsystem.

The fast video transfer comes in handy to watch TV 8)

	Enjoy,
	Amancio



From rem-conf-request@es.net Wed Feb 12 14:28:18 1997 
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Date: Wed, 12 Feb 1997 13:28:19 -0600
To: rem-conf@es.net, mbone@anl.gov
From: Chris Marshall <clmarshall@anl.gov>
Subject: Mbone Broadcast of Seminar
Cc: mike_petrick@qmgate.anl.gov (Michael Petrick/ANL), 
    klein@che.udel.edu (Michael Klein at U of Delaware), 
    bertolac@che.udel.edu (Ralph Bertolacini/Amoco/retired)
Mime-Version: 1.0
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On Thursday, February 20, 1997 the Argonne Petroleum Seminar Series will
broadcast the following seminar live.  The broadcast will begin at 10:30 am
CST.  Visit the Argonne Petroleum Seminar Series web site at
(http://www.anl.gov/Petroleum) for further information.

		Computer-Assisted Generation of Detailed Kinetic Models
				(ABSTRACT BELOW)
                                         Professor Michael T. Klein
                                    Department of Chemical Engineering
                                           University of Delaware
                                          Newark, Delaware 19716
____________________________________________

As a preliminary test to this live broadcast we will be broadcasting two of
the 1996 talks from 10 am to 2 pm on February 18 & 19, 1997.  These talks are:

	RECENT AND FUTURE DEVELOPMENTS IN FLUIDIZED CATALYTIC CRACKING (FCC)
                                      				Hartley Owen
                               Sr. Engineering Consultant (retired)
                              Mobil Research & Development Corp.

	THE IMPACT OF PRODUCT SPECIFICATIONS ON OIL REFINING
                                          William H. Keesom
                                        UOP Research Center

________________________________________________

Computer-Assisted Generation of Detailed Kinetic Models

ABSTRACT

The industrially relevant problem of kinetics assisted catalyst analysis
and design motivated the development of a kinetic model generation
technique with on-the-fly computational quantum chemistry calculations.
This integrated system for the computer generation of kinetic models begins
by modeling the structure of the reactants. Required reaction information
includes the rules by which the reactant and product species react and the
parameters of a structure/property kinetics correlation. This structural
and reaction information is transformed into reactant/product
relationships, i.e., the reaction mechanism, species properties, rate
constants, and the FORTRAN or C code corresponding to the governing species
balance equations. 

The advantage of computer generated reaction mechanisms is the enumeration
of what can be thousands of elementary reactions and the strict accounting
of what can be hundreds of species without the tedious manual effort that
is prone to errors. A set of reactions has no quantitative value, however,
without the associated rate parameters. The elementary reactions are
therefore categorized in terms of reaction families, the reactivity within
each reaction family being characterized in terms of a linear free energy
relationship, e.g., Eq. (1). 

                                            log ki = a + b RIi...........(1) 

The use of linear free energy relationships transforms the problem of
estimating reaction rate constants into the problem of estimating the
reactivity index (RI) for each reaction. The appropriate reactivity index
might require estimation of a wide range of species properties, such as
heat of reaction, electron affinity, electron density, proton affinity,
etc. Computational quantum chemistry is a robust, growing technique for
obtaining this information, and it is therefore interfaced directly with
the reaction mechanism generator to accomplish the on-the-fly calculation
of properties. 

This kinetic model building approach is illustrated using the topics of
hydrocarbon catalysis, pyrolysis and oxidation. 
===============================================
Christopher L. Marshall, Research Scientist          
Chemistry Division/Coal Chemistry Group
Argonne National Laboratory, CHM/200
9700 South Cass Avenue
Argonne, IL 60439-4831
===============================================
E-mail: CLMARSHALL@ANL.GOV
Phone: (630)252-4310
Fax: (630)252-9288      
=============================================== 
 Current Issues in Petroleum Processing Seminar Series at Argonne
< http://www.anl.gov/Petroleum >
 15th North American Catalysis Society Meeting, Chicago, 1997
< http://www.anl.gov/NAM >
===============================================

From rem-conf-request@es.net Wed Feb 12 18:10:21 1997 
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Subject: Re: Any *FIRST HAND* use of Linux Boxes to source MBone video ?
To: hasty@rah.star-gate.com (Amancio Hasty)
Date: Wed, 12 Feb 1997 15:08:20 -0800 (PST)
Cc: greg@Synopsys.COM, mbone-seminars@cl.cam.ac.uk, mbone@isi.edu, 
    rem-conf@es.net
In-Reply-To: <199702121011.CAA28997@rah.star-gate.com> from "Amancio Hasty" at Feb 12, 97 02:11:28 am
Content-Type: text



> >From The Desk Of Greg Kulosa :
> > 
> > It probably won't be able to do 30 frames/second, but I don't need that.
> > 5-8 frames/second or so is _plenty_ for me and the MBONE.
> 
> Well, the price of fast frame grabbers are dropping for instance
> Hauppage's WinCast/Tv retails for about $149(US) . This is a PCI 
> card with a Bt848 chipset and a tuner .

I should have mentioned that I am working on a Parallel-port frame
grabber because I want to do MBONE from a laptop.  There are no PCI
slots on any laptop I have found :-)

BTW, I also know that there is "vic" support for the "IBM PCMCIA Smart
Capture Card", but I have been unable to find this device.  It is not
on IBM's web page.

> 	Enjoy,
> 	Amancio

-- 
Greg A. Kulosa          | "The avalanche has already started, it is too
Systems Administrator   |  late for the pebbles to vote." - Ambassador Kosh
Synopsys, Inc.          |___________________________________________________
greg@synopsys.com       700 E. Middlefield Rd, Mountain View, CA 94043

From rem-conf-request@es.net Wed Feb 12 21:57:25 1997 
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Date: Wed, 12 Feb 1997 18:53:08 -0800 (PST)
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X-Priority: 1 (Highest)
To: rem-conf@es.net
From: mallard@mail.arc.nasa.gov (Mark Allard)
Subject: NASA-Shuttle Mission (STS-82) MBone Coverage Change

Those currently logged into the separate video and audio sessions of
"NASA-Shuttle Mission STS-82" MBone coverage need to disconnect from those
sessions and reconnect to the "NASA-Shuttle Mission (Video/Audio) STS-82"
combined session ASAP.

This is required due to significant problems identified in some combined
viewing tools and new integrated products that are having difficulty
dealing with the separate streams.

I would like to get specific EMail feedback from those who have difficulty
running both video and audio simultaneously to better understand the extent
of the issue, what the specific limitations are, and how we can best
respond to these needs; send to <mallard@mail.arc.nasa.gov>.

For those with limited connectivity desiring access to Shuttle mission
video and audio, I might suggest, as an immediate alternative, the use of
the CU-SeeMe reflector services provided by NASA; <quark.lerc.nasa.gov
(139.88.27.43)> is a good site to try.

Apologies for the inconvenience and confusion this has caused and hope you
will enjoy the remainder of the Shuttle mission on our combined MBone
session.

M






From rem-conf-request@es.net Thu Feb 13 00:54:41 1997 
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Subject: MBone/sdr-"phonebook"
To: rem-conf@es.net
Date: Thu, 13 Feb 1997 06:52:57 +0100 (MET)
Reply-To: wsb@rrzn.uni-hannover.de
From: wsb@rrzn.uni-hannover.de (Wolfgang Sander-Beuermann)
X-Mailer: ELM [version 2.4 PL25]
Content-Type: text

Hi!

We have made an MBone/sdr-Directory-Service doing more than just a
simple listing:

    http://meta.rrzn.uni-hannover.de/mbone.html

A public directory service for Mbone/sdr users: every potential user can
list her/himself here. The directory service checks automatically at
several random times per 24 hours the sdr port. After an inactivity of
more than 2 month the entry will be deleted. The entries can also be
checked manually at any time for current sdr activity.

In short words: some kind of "phonebook", which checks whether the phone
is actually connected.

Comments/Entries/etc. more than welcome,

Wolfgang
--
Wolfgang Sander-Beuermann     Tel.: use email   wsb@rrzn.uni-hannover.de
WebAdmin, Univ. of Hannover, Germany            http://www.uni-hannover.de/
----> MetaGer, the Meta-Search Enginge for German search engines
      http://meta.rrzn.uni-hannover.de/

From rem-conf-request@es.net Thu Feb 13 07:57:51 1997 
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          for <rem-conf@es.net>; Thu, 13 Feb 1997 14:55:47 +0200 (EET)
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Date: Thu, 13 Feb 1997 14:57:46 +0200
To: rem-conf@es.net
From: Jan Lucenius <jan.lucenius@vtt.fi>
Subject: video overlay cards for Windows NT
Mime-Version: 1.0
Content-Type: text/plain; charset="us-ascii"

Which video overlay cards are there which supports Windows NT (please
mention also which version(s) of NT) on the PC ?


_________________________________________________________________________
    _/       _/    _/   _/_/_    _/_/_   _/_/_/_/    Jan Lucenius
   _/       _/    _/  _/       _/       _/           phone +358 9 4566511
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From rem-conf-request@es.net Thu Feb 13 13:52:26 1997 
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From rem-conf-request@es.net Thu Feb 13 14:29:32 1997 
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To: 298-list@bmrc.Berkeley.EDU, rem-conf@es.net
From: Jerrlyn Iwata <jerrlyn@postgres.Berkeley.EDU>
Subject: Berkeley Multimedia & Graphics Seminar

                Berkeley Multimedia and Graphics Seminar 
        (Wednesday February 19, 1997 12:30-2:00 PDT 405 Soda Hall) 

      "EColabor: Collaborative Elaboration of Requirements Documents" 

                                Jeff Smith 
                             NTT Laboratories 

EColabor is an active hypermedia for collaboratively elaborating system
requirements. EColabor is designed to address problems in communication,
agreement, change management in requirements analysis, which are reportedly
significant but rarely addressed. Based on the Inquiry Cycle model, EColabor
supports analysts in systematically managing their requirements. EColabor
records all the processes of elaborating requirements in shared hypermedia
and provides comprehensive support for utilizing these records. 

Our goal is to apply multimedia and CSCW technologies to the requirements
elicitation process. The main technical challenges with which we are faced
are: (1) to establish feasible multimedia communication technology by using
the Internet, (2) to integrate synchronous and asynchronous CSCW
technologies, which have been independently developed, and (3) to develop a
model and tools for multimedia note-taking. Conventionally, research on
multimedia tends towards presentation. We emphasize support for authoring
(or note-taking) multimedia information instead. 

The talk will provide a background on EColabor, report on the current status
of the project, difficulties faced so far - including problems with
MBONE/Internet multimedia, and future directions.

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

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

Please direct all technical questions to davesimp@cs.berkeley.edu


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From rem-conf-request@es.net Thu Feb 13 15:35:49 1997 
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Date: Thu, 13 Feb 1997 12:10:31 -0800 ()
From: Stephen Casner <casner@hydra.precept.com>
To: Mark Allard <mallard@mail.arc.nasa.gov>
cc: rem-conf@es.net
Subject: Re: NASA-Shuttle Mission (STS-82) MBone Coverage Change
In-Reply-To: <v02140b19af27bb91ae65@[128.102.80.28]>
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On Wed, 12 Feb 1997, Mark Allard wrote:

> This is required due to significant problems identified in some combined
> viewing tools and new integrated products that are having difficulty
> dealing with the separate streams.

Precept's IP/TV was an example, and we appreciate Mark's making the
change to a combined session.

I'd like to clarify a couple of points, though:  IP/TV does _not_
require that the audio and video be sent to the same multicast
address, as has been done in the new combined session announcement.
sdr does allow separate addresses when both media are defined in one
session (in fact this is the default).  Using separate addresses is a
good idea to allow sites on low-bandwidth links to tune in to only one
of the media.

The problem for us was that we don't have a way to tune in to separate
audio and video sessions and synchronize them.  Perhaps this is a
feature we need to add if sessions are likely to be defined in
multiple parts.
							-- Steve


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From rem-conf-request@es.net Thu Feb 13 19:48:39 1997 
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To: Stephen Casner <casner@precept.com>
Cc: rem-conf@es.net
Subject: Re: Inconsistence in IP/UDP/RTP compression ID
In-reply-to: Your message of "Mon, 10 Feb 1997 20:29:13 PST." <Pine.WNT.3.95.970210202818.-106697A-100000@oak.precept.com>
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Stephen,

first of all many thanks for the detaild answers you have given.

> > We are currently developing a method for transfering multiple
> > RTP streams over a low-bandwidth PPP connection. The implementaion
> > consists of an RTP mixer on both ends of the PPP link, which
> > adjust the payload-type field according to the bandwidth
> > available. So it can happen very often,
> 
> How often?  It seems to me that the rate will need to be limited in
> order to maintain stability (avoid oscillation).  In particular,
> changes in the direction of increasing the data rate should not be
> done too quickly.  How long an integration time is needed to measure
> the available bandwidth?  Considering those factors, how many packets
> would be transmitted with one payload type before switching to
> another, and for what size packets?  Putting those numbers together,
> then you could figure out the percentage overhead caused by the need
> to send the full RTP header, including what your expected CSRC list
> size would be, on a payload switch.  If you have done this analysis,
> please tell us about the numbers.

Of course, the switch of the payload should not be made too often. 
Otherwise the coding related contexts as for GSM or LPC are totally
out of date. I expected not to switch the PT more than once in
about 30 seconds. So you're right, the overhead to send a full RTP
header (12 bytes) instead of 2 (first 2 bytes) +1 (for the PT) bytes
isn't worth to be mentioned! 

> > 2. CSRC change compression
> > Well as described above we use a mixer and expect several sources for
> > the outgoing media stream. The current ID says, that, when the CSRC
> > list changes, then the whole list has to be transmitted. Because there
> > are several sources and we can assume that they also change more 
> > frequenctly, I think that sending the whole list ist very poor!
> 
> Do you expect to have several sources active at once?  Part of the
> motivation for the current CSRC design is the expectation that, for
> example in the cause of mixing an audio conference, most of the time
> there will be only one person talking so the CSRC list will contain
> only that one identifier.

Well I think this is theory. In pratice, most players will have
silence detection enabled. So when you have a conference or
seminar, and one or two interrupts the speaker, you will have several
sources at the same time. Because of the silence detection, the
packets will oscillate, and because of the delay, the orginal
speaker notices this some time after. (Well I haven't meassured
this, so this statement is also theory :-)

> Now, your application might be quite different such that many sources
> are transmitting at once.  If so, would they stop and start
> frequently, or would they be transmitting all the time such that the
> CSRC list wouldn't change?  In particular, would the number of packets
> sent in whatever is equivalent to a "talkspurt" be small?  If the
> ...
> 
> In the conferencing example I mentioned earlier, when one person stops
> talking and another person starts talking, your proposal would require
> a CSRC list with two entries whereas the current spec would require
> only one.  I expect this case to be pretty common for mixers.

You're right I think. It was only an idea of mine, because while trying to
implement the compression protocol, this question arrose. My problem
is that I don't have so much experiences of how often the changes
occurs. But in any way, 1) my mentioned protocol isn't very fine and
stable, 2) the gain seems to be very low. So let's forget this idea!

> > Hm, perhaps; I don't see any difference between "Data field" and
> > "data field" ?! Anyway, because I'm not familar with IPv6, I didn't
> > know what the "Data field" is.
> 
> What I meant was, the IPv6 Header Compression spec defines a specific
> field named "Data".  Please note that the reference here is not to
> IPv6, but to the IPv6 Header Compression spec.

I understand, I haven't looked into the spec up to now, but your
explaination helped me. (I think it would be good to also explain
this in the ID?)

> The ID is only a 16-bit number, so the delta can't be > 0x3fffff or
> <0.  Or, more precisely, any delta value can be encoded into 1 to 3
> bytes by the specified encoding table.

Oh dear me, I'm an idiot, of course! %%$%&/!@#

> That's unfortunate, although legal, I guess.  Can the purveyors of
> Linux be convinced to htonl() the IP ID field?

Because of the mistake above, there is no need. BTW, W95 seems to 
do the same...

> > Perhaps we should think about using negative values for the delta
> > values? BTW, the same can theoretically happen when two IP packets
> > arrive in the wrong order (packet N+1 first, then packet N). If this
> > happens often, then the compression has NO gain! So how about of
> > using INTs for the delta fields instead of UNSIGNEDs. Ok, this
> > will reduce the maximum transferable delta value in one byte to
> > +-63. If you think this is too less, then we can say:
> > 	o the 1 byte form is UNSIGNED
> > 	o the 2 and 3 byte forms are SIGNED
> 
> At the IETF meeting, it was suggested that the delta encoding be
> specified as a table rather than a function, with the default value of
> the table being set according to the encoding currently in the spec.
> (This is as a precursor to a future capability to download a new table
> on a per-context basis for full entropy encoding.)  Since there are
> some holes in the current encoding, I suggest using those for negative
> offsets.  This keeps the encoding tuned for the common case of
> positive offsets, but handles a portion of the negative offsets with
> reasonable efficiency as well:
> 
>       Decimal  Hex
> 
>        -16384  C0 00 00
>             :  :
>          -129  C0 3F 7F
>          -128  80 00
>             :  :
>            -1  80 7F
>             0  00
>             :  :
>           127  7F
>           128  80 80
>             :  :
>         16383  BF FF
>         16384  C0 40 00
>             :  :
>       4194303  FF FF FF
> 
> Any delta < -16384 or > 4194303 would require a FULL_HEADER,
> COMPRESSED_NON_TCP or COMPRESSED_UDP packet type.
> 
> This encoding still leaves one hole of size 128 from C0 3F 80 to
> C0 3F FF just to keep the logical (vs. arithmetic) relationship in the
> table.  We could get 128 more negative numbers by sliding down the
> three-byte values instead.  I don't know if that makes a significant
> difference.  (Should be none if the algorithm is table-driven.)
> Comments, anyone?

That's very interesting, I think. First, PLEASE mention in the ID
that "7F" is different to "807F". In my prior understanding I assumed
that both are the same (127) and that the first form is just some
kind of "compression". I didn't undersood that the second form is a
"whole" in the definition.

Anyway, I think that using the holes for negative values is a very
interesting aspect. Also the use of a downloadable table or an 
dynamical changing table is very interesting. When you think of an
RTP stream comming from one sender, you can expect discrete values
for the deltas. E.g. in most cases the deltas in the context
changes only when a packet was lost (for audio). So when you 
can expect that the new delta is a multiple of the smallest delta
in the context (i.e. RTP-ID = 1 and RTP-timestamp = #samples in
frame, e.g. 320 for vat's PCM2), you only need those values. While
listening to the NASA STS-82 session, I noticed that the packet loss
(here in Germany) is often between 1 and 10. It also occurs that
the packets gets out of order (i.e. negative timestamp). So we need only,
let's say 20 *2 different deltas, which can be encoded in on octet,
even if the delta (for the timestamp) is up to 3200 (320*10).
So using a table sounds to be very efficient for me.

> > Now sender A changes the payload-type. This forces the translator T to
> > send the packet as COMPRESSED_UDP instead of COMPRESSED_RTP. There
> > will arrive several questions:
> > 	o to which context do the outgoing packet belongs, cA or cB?
> ....
> sources trade off.  Or, you can use 16-bit context IDs as described in
> the ipv6-hc draft (details to be added to the compressed RTP spec
> soon).

Thanks a lot for these very detailed informations! They helped me
in understanding the real process and idea much better.

> I'd appreciate any comments you might have on what additional
> information I should add to the document and where.

Ok, here are some comments:
	o fields like "Data" field should be explicitely explained 
	  on not only by reference to the IPv6 hc draft.
	o the holes should be explixit mentioned (see above).
	o the fact that the RTP fields also HAVE TO be saved
	  for the COMPRESSED_UDP should be described a little bit
	  better. Ok, after your comments and after rereading pages
	  4 and 5, it seems to be clear, but it wasn't before :-/
	o I also think that it would be better to explicitly 
	  describe each "packet type" (FULL_HEADER, ...) in it's
	  own section, and to describe what happens with which 
	  fields in the context. This will make the understanding
	  much easier, I hope. One problem in the current ID is,
	  that you have read (nearly) the whole ID to understand
	  one part (that's only my opinion!)

Anyway, it's an interesting ID and I'm happy to play with it :--)

Once again, thanks a lot for your very detailed help you have given
me.

CU,
Chris




From rem-conf-request@es.net Fri Feb 14 01:32:26 1997 
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From: Rainer Kranz <kranz@rcs.urz.tu-dresden.de>
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To: Amancio Hasty <hasty@rah.star-gate.com>
Cc: mbone-seminars@cl.cam.ac.uk, mbone@isi.edu, rem-conf@es.net
Subject: Re: Any *FIRST HAND* use of Linux Boxes to source MBone video ?
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How can I use the WinCast/Tv in a Linux System ?

Rainer

,-----------------------------------------------------------------.
| Referenzzentrum f=FCr multimediale Teledienste (MMRZ), TU Dresden |
|         Dr. Klaus K=F6hler, Christoph Fleck, Rainer Kranz         |
| e-mail: mmt-ref@tu-dresden.de            Tel.: 0351 / 463 5653  |
|    WWW: http://www-mm.urz.tu-dresden.de              (GERMANY)  |
`-----------------------------------------------------------------'




From rem-conf-request@es.net Fri Feb 14 02:38:06 1997 
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To: Rainer Kranz <kranz@Rcs1.urz.tu-dresden.de>
cc: mbone-seminars@cl.cam.ac.uk, mbone@isi.edu, rem-conf@es.net
Subject: Re: Any *FIRST HAND* use of Linux Boxes to source MBone video ?
In-reply-to: Your message of "Fri, 14 Feb 1997 07:31:18 +0100." <Pine.OSF.3.95.970214072911.5985D-100000@rcs12.urz.tu-dresden.de>
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Date: Thu, 13 Feb 1997 23:37:01 -0800
From: Amancio Hasty <hasty@rah.star-gate.com>

I have no idea ;however, if you are interested in porting the FreeBSD 
driver to linux I can provide the driver.

	Amancio


>From The Desk Of Rainer Kranz :
> 
> How can I use the WinCast/Tv in a Linux System ?
> 
> Rainer
> 
> ,-----------------------------------------------------------------.
> | Referenzzentrum f=FCr multimediale Teledienste (MMRZ), TU Dresden |
> |         Dr. Klaus K=F6hler, Christoph Fleck, Rainer Kranz         |
> | e-mail: mmt-ref@tu-dresden.de            Tel.: 0351 / 463 5653  |
> |    WWW: http://www-mm.urz.tu-dresden.de              (GERMANY)  |
> `-----------------------------------------------------------------'
> 
> 
> 



From rem-conf-request@es.net Fri Feb 14 06:25:39 1997 
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To: rem-conf@es.net
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Subject: MERCI Seminar
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Date: Fri, 14 Feb 1997 11:13:59 +0000
Message-ID: <12893.855918839@cs.ucl.ac.uk>
From: Roy Bennett <R.Bennett@cs.ucl.ac.uk>

The MERCI (Multimedia European Research Conferencing Integration)
project announces the next in an on-going series of seminars on
subjects in the fields of Multimedia, Communications, Networks,
Distributed Systems and CSCW

Time :   18 February 14:00 UTC 

Title:   Network Measurement on the Internet
Chair:   Peter Kirstein 
Speaker: Rob Cole
         HP Labs, Bristol, UK
 
An announcement will be made in sdr later.

The seminar has been booked in the MBone Session Agenda
(http://www.cilea.it/MBone/browse.html). 
If there are still problems with overlap, please contact me.

Roy
-----------------------------------------------------------------------
Roy Bennett, Project Manager, MERCI http://www-mice.cs.ucl.ac.uk/merci/
Computer Science                           Phone: +44 171 380 7934
University College London                  Fax:   +44 171 387 1397
Gower Street, LONDON WC1E 6BT              Email: rbennett@cs.ucl.ac.uk
----------- http://www-dept.cs.ucl.ac.uk/staff/R.Bennett --------------



From rem-conf-request@es.net Sat Feb 15 03:07:16 1997 
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Date: Sat, 15 Feb 1997 00:05:44 -0800 (PST)
From: Stephen Casner <casner@precept.com>
To: Christian Zahl <zahl@fokus.gmd.de>
cc: rem-conf@es.net
Subject: Re: Inconsistence in IP/UDP/RTP compression ID
In-Reply-To: <199702132155.WAA04708@tao.fokus.gmd.de >
Message-ID: <Pine.SOL.3.95.970214235135.2592G-100000@little-bear.precept.com>
MIME-Version: 1.0
Content-Type: TEXT/PLAIN; charset=US-ASCII

> Well I think this is theory. In pratice, most players will have
> silence detection enabled. So when you have a conference or
> seminar, and one or two interrupts the speaker, you will have several
> sources at the same time. Because of the silence detection, the
> packets will oscillate, and because of the delay, the orginal
> speaker notices this some time after.

Yes, there will be some periods of talk-over.  But it should not be
necessary to update the CSRC list on a packet-by-packet basis.  When
someone interrupts the speaker, there will be a period of several
packets when there is sound from both sources at once.  The CSRC list
will need to be updated when the interruption starts and when it
ends.  If the first speaker stops upon hearing the interruption, and
then perhaps starts again, there may be some oscillation, but it will
be on the scale of units of seconds not packets.  So there should be
several packets transmitted to amortize each CSRC list update.

> > That's unfortunate, although legal, I guess.  Can the purveyors of
> > Linux be convinced to htonl() the IP ID field?
> 
> Because of the mistake above, there is no need. BTW, W95 seems to 
> do the same...

Even more unfortunate.  I use W95 (not necessarily by choice :-) but I
had not noticed this.  Even though the byte-swapped update can be
accommodated by the delta encoding, it will take more bytes.  It would
be nice if this htonl() could be introduced.

> That's very interesting, I think. First, PLEASE mention in the ID
> that "7F" is different to "807F". In my prior understanding I assumed
> that both are the same (127) and that the first form is just some
> kind of "compression". I didn't undersood that the second form is a
> "whole" in the definition.

You are (or were) right.  The spec as currently written does not
include the encoding of the negative deltas.  This is a new proposal
as of the IETF meeting, and I need to update the draft to include it.

> Ok, here are some comments:

Thanks for your suggestions.
							-- Steve


From rem-conf-request@es.net Sun Feb 16 21:24:53 1997 
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Date: Sun, 16 Feb 1997 21:26:03 -0800
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To: rem-conf@es.net
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Subject: HPN'97 Advance Program + Registration Information
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H P N '  9 7

                 A D V A N C E    P R O G R A M

                   SEVENTH IFIP CONFERENCE ON

              HIGH PERFORMANCE NETWORKING (HPN'97)

          White Plains, New York, April 28-May 2, 1997

                http://www.ctr.columbia.edu/hpn97


THE PROGRAM AT A GLANCE:

Monday April 28, 1997

    9:00 - 12:30  Tutorial A:
                  Wireless ATM and Mobile Computing
                  Upkar Varshney, Washburn University, USA

   13:30 - 17:00  Tutorial B:
                  Scheduling Algorithms for Integrated Services Networks
                  Hui Zhang, Carnegie Mellon University, USA

Tuesday April 29, 1997:

    9:00 - 12:30  Tutorial C:
                  The Next Generation Internet Protocol Suite
                  Doug Montgomery, NIST, USA

   13:30 - 17:00  Tutorial D:
                  Protocol Support for End-to-End QoS Guarantees across
                  the Internet
                  Raj Yavatkar, Intel, USA

Wednesday April 30, 1997:

    8:45 -  9:45  Conference Opening and Keynote Session
    9:45 - 10:45  Experience with Standards
   11:15 - 12:30  Multicast Implementation Issues
   12:30 - 13:30  Conference Luncheon
   13:30 - 15:00  Experimental Results
   15:30 - 17:00  Multimedia Traffic

Thursday May 1, 1997:

    9:00 - 10:30  Quality of Service
   11:00 - 12:30  Fundamental Concepts
   12:30 - 13:30  Conference Luncheon
   13:30 - 15:00  Architectural Issues
   15:30 - 16:45  Bandwidth Allocation

   18:00 - 22:00  CONFERENCE BANQUET (Dinner Cruise around NYC)

Friday May 2, 1997:

    9:00 - 10:30  Invited Presentations: Use of HPNs
   11:00 - 12:30  Panel Discussion: Trends and Future of HPNs


For details of the full program and registration information, please
visit
http://www.ctr.columbia.edu/hpn97

From rem-conf-request@es.net Sun Feb 16 23:57:04 1997 
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Date: Sun, 16 Feb 1997 23:56:01 -0500 (EST)
From: Jeff Pulver <jeff@Pulver.COM>
To: rem-conf@es.net
cc: rhaber@von.com
Subject: Spring '97 Voice on the Net: Advance Program + Registration Information
In-Reply-To: <3307EBEB.527C@ctr.columbia.edu>
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V O N'  9 7

                A D V A N C E    P R O G R A M

                 Spring '97 Voice on the Net Conference 

             Telecommunications and Streaming Media on the Net
	
                    San Francisco, April 1-3, 1997

                   http://www.pulver.com/von97


THE PROGRAM AT A GLANCE:

Tuesday April 1, 1997:

    8:00 - 12:30  Keynotes / General Sessions
                   (University of Pennslvania, Cisco, MCI, WorldCom,
                    Intel, Netscape, elemedia) 

   13:45 - 15:45  Keynotes / General Sessions
                   (Compaq, Lucent, Pacific Bell, IBM)

   15:45 - 18:45  Breakout Sessions

Wednesday April 2, 1997:

   8:00 -   8:30  General Session - Standards Update
                  (IMTC)
                  
   8:30 -  12:30  Breakout Sessions

  13:45 -  14:45  Scott Adams (columnist Dilbert;Author the Dilbert Principle)

  14:45 -  17:30  General Sessions  
                  (NTIA, MCI, AT&T, Sprint, Bell Atlantic) 

Thrusday April 3, 1997:

   9:00 -  17:00  Internet Telephony Gateway Workshop


For details of the full program and registration information, please
visit-  http://www.pulver.com/von97

From rem-conf-request@es.net Mon Feb 17 10:36:51 1997 
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From: Bernard Aboba <aboba@internaut.com>
To: "'mboned@ns.uoregon.edu'" <mboned@network-services.uoregon.edu>
Cc: "'rem-conf@es.net'" <rem-conf@es.net>
Subject: Multicast diagnostic tools I-D
Date: Mon, 17 Feb 1997 07:32:05 -0800
Encoding: 53 TEXT

For the Memphis IETF meeting, it would be useful to pull together an 
Internet Draft describing publicly distributable multicast diagnostic tools. 
The idea is to provide an overview of what tools are available and what 
problems they can be used to solve. Hope is that this will be a useful
reference for those looking to deploy multicast on their networks. 

Assistance from tool authors and volunteers is solicited. I will act
as editor. 

Information to be included on each tool:

Author (name and e-mail address)
Description (a paragraph or two describing what the tool does)
Example (an example of the output of the tool)
Applicability (domain of applicability i.e. uses SNMP, depends on DVMRP nearest neighbor, etc.)
Availability (Platform availability, pointers to source and binary distributions) 

Tools that I'd like to include:

Tools based on RTCP reports
  RTPmon
  RTPqual

Tools based on SNMP
  mstat
  mconfig
  snmpoidget
  mrtree

Tools based on DVMRP nearest neighbor messages
  mrdebug
  mrinfo
  mrmap
  map-mbone

Tools based on IGMP
  mtrace
  mrbranch
  mbranch
  mcollect
  mlisten

Network analysis tools
  mdump
  mpacket
  MVIEW

AS mapping tools
  asn
  asname





From rem-conf-request@es.net Tue Feb 18 02:31:12 1997 
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Date: Mon, 17 Feb 1997 23:29:52 -0800 (PST)
From: Stephane Chatre <chatre@ENGR.ORST.EDU>
Reply-To: Stephane Chatre <chatre@ENGR.ORST.EDU>
To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>, rem-conf@es.net
cc: Stephane Chatre <chatre@ENGR.ORST.EDU>
Subject: rtptools' file format.
In-Reply-To: <32F72B71.3818@cs.columbia.edu>
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Hi!

As you probably noticed, several sites allowing playback on demand of
recorded MBone sessions poped up here and there on the web, using for most
of them rtpdump and rtpplay.

IMHO, the file once dumped doesn't contain enough information regarding
the content of the session: what is it about? When was it held and where?
etc...

I think this is time to decide for a standard format in which to record
sessions, before it gets too much spread.

I am thinking about adding a header which would be sort of a copy of the
SDP packet used for the announcement of the session, because:
- SDP was designed to contain almost all the informations needed about a
session.
- SDP info is easy to parse.
- It would be easily applicable for recording a session in advance: a
recorder could "listen" announcements and record interesting sessions,
thus knowing the SDP content already.
- On-demand Servers could Multicast SDP announcements of recordings
available on a dedicated address/port, as well as the URL of "where" these
sessions are available to playback.

A drawback I see is that SDP packets contain information about all the
streams of the session, which we not necessarly need to know about for a
certain stream. But this might be useful, for example to know "part of
which" the recorded stream was.

 If the session was not announced or the SDP packets were not
received (session recorded on the fly), its fields could be left
blank.

Do you know if the IETF is working on defining a format for recorded
sessions? If yes, has there been a draft about it? If not, what do you
think?

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



From rem-conf-request@es.net Tue Feb 18 12:46:01 1997 
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Date: Tue, 18 Feb 1997 18:31:30 +0100
From: pierre de saqui sannes DFR/MI <pdss@ensica.fr>
Message-Id: <199702181731.SAA11198@magic>
To: rem-conf@es.net
Subject: Workshop on Multimedia and Concurrency

Please accept our apologies if you receive this email more than once.


                      +++++++++++++++++++++++
                          CALL FOR PAPERS
                      +++++++++++++++++++++++

     FIRST INTERNATIONAL WORKSHOP ON MULTIMEDIA AND CONCURRENCY 

                      A Workshop within

      the XVIII International Conference on Applications and
           Theory of Petri Nets (June 23-27, 1997)

                  Toulouse, June 24, 1997

   Organizers :
   Michel Diaz, LAAS CNRS, Toulouse, France 
   Thomas Little, Boston University, Boston, USA  
   Patrick Senac, ENSICA, Toulouse, France  


Petri nets have been applied to the design of multimedia systems for
a few years for different purposes : modeling, semantics, scheduling,
resource allocation, protocols, etc. By multimedia systems it is meant 
general distributed systems that have new and complex constraints.   
Indeed distributed multimedia systems have to transfer and synchronise
large volumes of data and to enforce strong time requirements on top
of general Local area and Wide area networks.  
For instance, a remote multimedia presentation including one sequence of
High Definition video, at a rate of 25 images per second, needs, only
for this sequence, a time constrained flow of tens of megabits 
per second of transfert rate. 

This first workshop on this subject will be organized
within the well-known conference "International 
Conference on Application and Theory of Petri nets".
A set of papers plus one or two surveys will be presented. 
The scope of the workshop is focused on the application and theory of Petri 
nets applied to multimedia systems. Nevertheless, other models will
be considered. Case studies describing applications will be welcome.  

Some topics are (not an exhaustive list):

  - modeling techniques (hierarchical, modular, high level, etc)
  - modeling time in multimedia information 
  - performance evaluation and simulation
  - scheduling (off-line and real time) and optimization
  - time synchronisation and timed constraints
  - hypermedia reactive multimedia models
  - integration of different models of concurrency  
  - user and user interaction modelling
  - PNs in Multimedia-Hypermedia communication protocols
  - PNs and the Web 
  - applications and case studies 

Researchers interested are invited to submit a position paper 
(15 pages max, with a small abstract) or an extended abstract (some 5-8 pages) 
indicating the key ideas and results.  
The workshop will include time for presentations and discussions.


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

Submissions: Monday, March 31, 1997 
             * three hardcopies to: 
                 Michel Diaz 
                 LAAS-CNRS
                 7 Av. du Colonel Roche
                 F-31077 Toulouse Cedex 4, FRANCE
                 
             * or postscript files to the three following email address:
                 diaz@laas.fr
                 tdcl@BU.EDU 
                 senac@ensica.fr  
       
Notification of acceptance: May 5, 1997

Final copy of the paper, 15 pages, is due: May 30,1997
  A proceedings of the papers will be given to the participants
 




Pierre de Saqui-Sannes		pdss@ensica.fr
ENSICA
1 Place Emile Blouin		tel. +33 5 61.61.86.81
31056 Toulouse Cedex		fax. +33 5 61.61.86.86
	

From rem-conf-request@es.net Tue Feb 18 14:50:32 1997 
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From: Jerrlyn Iwata <jerrlyn@postgres.Berkeley.EDU>
Subject: [Reminder] Berkeley Multimedia & Graphics Seminar

                Berkeley Multimedia and Graphics Seminar 
        (Wednesday February 19, 1997 12:30-2:00 PDT 405 Soda Hall) 

      "EColabor: Collaborative Elaboration of Requirements Documents" 

                                Jeff Smith 
                             NTT Laboratories 

                MBONE BROADCAST BEGINS AT 12.40 PDT (GMT - 8 HOURS)


From rem-conf-request@es.net Tue Feb 18 18:18:08 1997 
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From: Michael.Speer@Eng.Sun.COM (Michael F. Speer)
Message-Id: <199702182316.PAA04350@valathar.eng.sun.com>
To: rem-conf@es.net
Subject: MBONE RTP session
Cc: Michael.Speer@Eng.Sun.COM


Folks:

What happened to the "MBONE RTP Audio" session?  It doesn't seem to announced
anymore.

Michael

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From: m1 <m1@msri.org>
Reply-to: m1@msri.org
Subject: Gil Kalai "Flag f-vectors of polytopes"

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

Title:       
	Gil Kalai "Flag f-vectors of polytopes"
Date:        
	Feb 25, 1997

Time:        
	12:00 PST8PDT 1 hours

Contact:     
	m1@msri.org

URL:         
	http://www.msri.org/lecturenotes/97/kalai/

Description:        
	   Flag numbers of polytopes count chains of faces of prescribed  dimensions. We will discuss what is known about them and  especially the following:     1. The Bayer-Billera theorem on the affine dimension of flag  numbers of $d$-polytopes, and various proofs and extensions of  this theorem. 2. The notion of $h$-vector coming from  intersection homology and its combinatorial significance for  linear inequalities among flag numbers. 3. Combinatorial  applications of the linear inequalities that were derived by  Meisinger's program: FLAGTOOL. 4. A recent monotonicity  theorem of Braden and MacPherson and some of the applications  of the theorem and of the path to its proof. 5. The recent  attempts by Jonathan Fine to extend the $h$ numbers into a  complete set of invariants. 









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

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To: rem-conf@es.net
From: m1 <m1@msri.org>
Reply-to: m1@msri.org
Subject: Bernd Sturmfels "The Geometry of A-graded Algebras"

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

Title:       
	Bernd Sturmfels "The Geometry of A-graded Algebras" 
Date:        
	Feb 26, 1997

Time:        
	12:00 PST8PDT 1 hours

Contact:     
	m1@msri.org

URL:         
	http://www.msri.org/lecturenotes/97/sturmfels/

Description:        
	  Let A be a configuration of n lattice vectors in d-space. An  algebra over a field is called A-graded if it is graded by the  semigroup spanned by A and each graded component is  one-dimensional. This notion was introduced by V.I.Arnold who  showed (jointly with Korkina and Post) that for d = 1, n < 4 and A  fixed there are only finitely many isomorphism types of A-graded  algebras, and they are indexed by the faces of the state polytope  of A.     Arnold's finiteness theorem is conjectured to hold for general d  and n < d+4; this would extend Carl Lee's result that all  triangulations of a d-polytope with d+3 vertices are regular. For n  >= d+4 there are moduli of A-graded algebras, and the distinct  algebra are parametrized by a binomial scheme which refines the  Baues poset of all polyhedral subdivisions of A.     This talk gives an introduction to this subject, along the lines of  Chapter 10 in my book "Groebner Bases and Convex Polytopes".  Emphasis will be placed on c!
!
onnections to polytopes and integer  programming. 









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

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To: rem-conf@es.net
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Reply-to: m1@msri.org
Subject: Juergen Richter-Gebert "Moving and proving for configurations of 
         points, lines, and conics"

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

Title:       
	Juergen Richter-Gebert "Moving and proving for configurations of points, lines, and conics"
Date:        
	Feb 27, 1997

Time:        
	12:00 PST8PDT 1 hours

Contact:     
	m1@msri.org

URL:         
	http://www.msri.org/lecturenotes/97/richter-gebert/

Description:        
	  This talk illustrates the close interrelation of problems concerning    <ul>  <li>configuration spaces,  <li>invariant theoretic methods for  proving geometric theorems, and   <li>the concrete implementation  of interactive geometry software.   </ul>    Here the influence of Abstract mathematics and concrete  implementation work is bi-directional. On the one side interactive  geometry programs can be used to explore Abstract geometric  structures (like configuration spaces). Conversely, it often  requires deep mathematical insights to implement interactive  geometry programs that behave in a mathematical consistent  way. The talk will be dedicated to an actual selection of  interesting aspects of these interrelations. 









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

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To: Stephane Chatre <chatre@ENGR.ORST.EDU>
From: Ross Finlayson <finlayson@lvn.com>
Subject: Re: rtptools' file format.
Cc: rem-conf@es.net

>I think this is time to decide for a standard format in which to record
>sessions, before it gets too much spread.
>
>I am thinking about adding a header which would be sort of a copy of the
>SDP packet used for the announcement of the session, because:
>- SDP was designed to contain almost all the informations needed about a
>session.
>- SDP info is easy to parse.
>- It would be easily applicable for recording a session in advance: a
>recorder could "listen" announcements and record interesting sessions,
>thus knowing the SDP content already.
>- On-demand Servers could Multicast SDP announcements of recordings
>available on a dedicated address/port, as well as the URL of "where" these
>sessions are available to playback.

Stephane,

If you find that SDP contains all of the attributes that you need, then you
should probably use it as-is.  However, if there are additional attributes -
not defined in SDP - that you need to represent, then you might consider
something like MAFP (see http://www.lvn.com/multikit/mafp-list-info.html).

MAFP can be thought of as a generalization of SDP that (among other things)
allows an arbitrary set of attributes to be specified in a multicast-related
announcement.  Also, there is already a mapping defined (in code; not yet
documented) from SDP to MAFP.

        Ross.


From rem-conf-request@es.net Tue Feb 18 21:12:46 1997 
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          with ESnet SMTP (PP); Tue, 18 Feb 1997 18:11:22 -0800
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          id VAA16484; Tue, 18 Feb 1997 21:11:18 -0500 (EST)
From: John Hawkinson (local) <ljhawk@bbnplanet.com>
Message-Id: <199702190211.VAA16484@all-purpose-gunk.near.net>
Subject: Re: MBONE RTP session
To: Michael.Speer@eng.sun.com (Michael F. Speer)
Date: Tue, 18 Feb 1997 21:11:17 -0500 (EST)
Cc: rem-conf@es.net
In-Reply-To: <199702182316.PAA04350@valathar.eng.sun.com> from "Michael F. Speer" at Feb 18, 97 03:16:03 pm
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> What happened to the "MBONE RTP Audio" session?  It doesn't seem to announced
> anymore.

My workstation is still happily announcing it.

It appears there's some issue with
routing to me from you:

[all-purpose-gunk!jhawk] ~> mtrace -g mbone.sun.com ap-gunk.near.net sap.mcast.net
* Mtrace from 199.94.220.184 to 192.9.9.71 via group 224.2.127.254
Querying full reverse path... * switching to hop-by-hop:
  0  mbone.Sun.COM (192.9.9.71)
 -1  * * * Timed out receiving responses
[all-purpose-gunk!jhawk] ~> 

I suggest we take this up offline.

Multicast in my neighborhood has been a bit more flakey than
normal, though, so it may be somewhat local.

--jhawk

From rem-conf-request@es.net Wed Feb 19 15:16:29 1997 
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From: Eric Fleischman <ericfl@MICROSOFT.com>
To: "'rem-conf@es.net'" <rem-conf@es.net>
Subject: Problems with RTP Payload Types
Date: Wed, 19 Feb 1997 12:13:37 -0800
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This note addresses my concern that the payload type field of RTP (RFC 
1889) is too small to accomplish its purpose over time. This note first 
examines our problem with dynamic payload type entries and extrapolates 
those problems to the general case.

Table 2 in RFC 1890 defines the initial payload types for RTP. We are using 
the dynamic payload entries (values 96-127) of Table 2 in our current 
product. I have not noticed an official description of how this community 
desires these dynamic values to be used. Thus, I would appreciate pointers 
to the relevant documents, if any.

What we are doing is using the dynamic values to specify specific codec and 
encoding values. Since our application has control over both the client and 
server "pieces" of the communication stream, we have arbitrarily defined 
that "Dynamic value x means codec Y at a specific sample size, sampling 
frequency and average bytes/sec." This, anyway, is how we have defined the 
meaning of "dynamic" -- subject to correction from this list.

What we have found is that the 32 available dynamic payload values are 
inadequate for our needs. The problem which we are finding, and I suspect 
the industry as a whole can corroborate, is that a tremendous amount of 
work by a great many companies is going into defining new codecs and 
encoding/decoding approaches. Thus our product must necessarily support 
scads of codecs each with different possible sample sizes, sample 
frequencies, avg bytes/sec, etc.

Users seem to demand the best quality encoding possible. Codec x may 
provide the best quality today but codec y may do so next week. Thus, we 
need a provision to keep adding new codecs with new encodings into our 
product. Thus, even if the 32 dynamic payload values were fully adequate 
for us today, it is only a matter of time before we run out of dynamic 
payload types that we can use within our application.

Since the payload type (PT) field of RFC 1889 is only 7 bits long, the 128 
available values thus are subject to the same problems. A glance at Table 2 
of RFC 1890 shows that many of the listed codecs are now obsolete. Mind 
you, they are not obsolete in the sense that their specifications have 
somehow "gone away" or that those types are no longer used within some 
product somewhere. Rather, they are obsolete in the sense that their 
encoding quality is so poor that no "modern commercial product" would be 
willing to use them in the general case. Similarly, our product needs to 
maintain a certain number of "obsolete dynamic PTs" simply due to the 
versioning problem. My point is that it is impractical to "flush away" 
obsolete entries so that their place may be taken by more viable entries.

If we assume all of today's most viable approaches will eventually become 
registered with IANA, it is improbable that there will be room to register 
some-future-date's most viable approaches if we assume that today's rapid 
codec development rate continues to persist over time.

A solution scenario which expands the RTP header size to support a larger 
PT field would resolve this problem to an unspecified future time. But it 
would also increase the overhead of RTP. This would be very undesirable for 
slow modem rates which are very susceptible to such overheads. Please feel 
free to contact me if you are interested in discussing alternative solution 
scenarios. Yet before we initiate such a conversation, I'd first like to 
determine if the above observations resonate with this group as a whole or 
not.


From rem-conf-request@es.net Wed Feb 19 16:11:15 1997 
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Date: Wed, 19 Feb 1997 16:10:15 -0500
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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To: Eric Fleischman <ericfl@MICROSOFT.com>
CC: rem-conf@es.net
Subject: Re: Problems with RTP Payload Types
References: <c=US%a=_%p=msft%l=RED-68-MSG-970219201337Z-4059@INET-04-IMC.microsoft.com>
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Dynamic is meant to mean that these values can be re-assigned on a (say)
per-session basis, so unless you have more than 32 truly different
codecs (not just a 16 kb/s version and a 32 kb/s version of the same
video format, i.e., decodable with the same codec) active during the
same session, this should not be a problem. Unfortunately, except for
the case of session announcements with a limited set of pre-assigned
codecs, there is no generally accepted way of negotiating the list of
encodings of interest to the receiver. One might imagine a protocol
where the client indicates to the server the list of codecs it
understands/prefers and then have the server set up dynamic PTs for that
list (which, within each stream, should be easily manageable with 32).
SIP has facilities for things along this line.

If there are codecs which are closely related (e.g., same codec, but
different sample rate or channel count or bit depth or image size), the
appropriate approach is to define the class an an RTP PT and define the
specific subclass as part of the payload-specific encapsulation, unless
you are so bit-constrained that this is not feasible.

So, is it likely that a single server for a single media type (audio or
video, say) will need to support 32 different encoding classes?

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

From rem-conf-request@es.net Wed Feb 19 16:14:35 1997 
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          Wed, 19 Feb 1997 16:13:44 -0500 (EST)
Sender: liao@ctr.columbia.edu
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Date: Wed, 19 Feb 1997 16:13:43 -0500
From: Raymond Liao <liao@ctr.columbia.edu>
Organization: CTR, Columbia University
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To: rem-conf@es.net
CC: mnl@ctr.columbia.edu
Subject: Seminar: C. Huitema "The Internet Architecture - End-to-End Principle"
Content-Type: text/plain; charset=us-ascii
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MBone Broadcast Announcement

Title:
    The Internet Architecture - End to End Principle

Speaker:
    Christian Huitema
    Bellcore

Date:
    Thursday, Feb. 20, 1997

Time:
    12:00 PM - 1:00 PM EST

Multicast Address:
    audio:  DVI, RTP, 224.2.243.158/30528/127
    video:  H.261, RTP, 224.2.154.154/59442/127

Contact:
    Andrew T. Campbell (campbell@ctr.columbia.edu)

URL:
    http://comet.ctr.columbia.edu/activities/seminars/

Place:
    Multimedia Network Laboratory
    8LE1 Schapiro Research Building
    Center for Telecommunications Research
    Columbia University
    New York City, USA
-- 
Raymond R.-F. Liao
(212) 939-7158
http://www.ctr.columbia.edu/~liao
Center for Telecommunications Research, Columbia University

From rem-conf-request@es.net Wed Feb 19 17:10:30 1997 
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          Wed, 19 Feb 1997 14:10:49 -0800
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From: Eric Fleischman <ericfl@MICROSOFT.com>
To: 'Henning Schulzrinne' <schulzrinne@cs.columbia.edu>
Cc: "'rem-conf@es.net'" <rem-conf@es.net>
Subject: RE: Problems with RTP Payload Types
Date: Wed, 19 Feb 1997 14:05:53 -0800
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Thank you, Henning, for your very helpful reply. You pointed out some ideas 
which I hadn't considered which very likely may end my concern about 
dynamic PTs. If not, I'll get back to you once I have thought more about 
how things work if done the way you intended.

Now I will try to answer your question:
I have not encountered a real-life situation where a server needs to 
support 32 different encoding classes for a single media type when 
sub-classes are in use.

Because the PT contains both audio and video entries, I originally imagined 
that both spaces had to be satisfied within that same PT class space and in 
that case the 32 limitation is a problem (see below). We also support 
several other media types in addition to audio and video. Thus, the problem 
which I envisioned was the following
1)	our particular content may or may not be pre-announced via SIP or a 
similar capability. If it is pre-announced then we can benefit from the PT 
values being constrained to the values indicated within the announcement as 
you indicated. If it is not pre-announced, but rather initiated in an "on 
demand" fashion by user(s), then the PT values need to have already been 
agreed to by the sender-receiver and thus should include all of the 
encodings currently supported on that server. An example of "on-demand" use 
is when one or more users request that a pre-produced multimedia production 
be streamed to them.
2)	In the "on demand" case, the total number of needed PT values definitely 
exceeds 32 if we lump all of the encodings for all of the supported media 
types on that server. If we segregate the encodings by media type and use 
your sub-class delineations, then 32 PT values should work for us today.

Unfortunately, I can't speak for the future (as indicated by my original 
message) because the future may entail different codec innovation rates 
than today. If the rate of innovation increases, then we may or may not 
have a problem. Another relevant factor is how persistent the "older 
encodings" remain. (If they don't disappear then they take up space.) In 
any case, I can easily imagine the non-dynamic PT values quickly filling up 
with entries with an unknown shelf life viability. Thus I suspect that 
long-term only dynamic PT values will remain viable. It is thus desirable 
that we address how these dynamic values can best be used in "on-demand" 
scenarios.



-----Original Message-----
From:	Henning Schulzrinne [SMTP:schulzrinne@cs.columbia.edu]
Sent:	Wednesday, February 19, 1997 1:10 PM
To:	Eric Fleischman
Cc:	rem-conf@es.net
Subject:	Re: Problems with RTP Payload Types

Dynamic is meant to mean that these values can be re-assigned on a (say)
per-session basis, so unless you have more than 32 truly different
codecs (not just a 16 kb/s version and a 32 kb/s version of the same
video format, i.e., decodable with the same codec) active during the
same session, this should not be a problem. Unfortunately, except for
the case of session announcements with a limited set of pre-assigned
codecs, there is no generally accepted way of negotiating the list of
encodings of interest to the receiver. One might imagine a protocol
where the client indicates to the server the list of codecs it
understands/prefers and then have the server set up dynamic PTs for that
list (which, within each stream, should be easily manageable with 32).
SIP has facilities for things along this line.

If there are codecs which are closely related (e.g., same codec, but
different sample rate or channel count or bit depth or image size), the
appropriate approach is to define the class an an RTP PT and define the
specific subclass as part of the payload-specific encapsulation, unless
you are so bit-constrained that this is not feasible.

So, is it likely that a single server for a single media type (audio or
video, say) will need to support 32 different encoding classes?

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


From rem-conf-request@es.net Wed Feb 19 18:45:38 1997 
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Date: Wed, 19 Feb 1997 15:27:05 -0800 ()
From: Stephen Casner <casner@precept.com>
To: Eric Fleischman <ericfl@MICROSOFT.com>
cc: "'rem-conf@es.net'" <rem-conf@es.net>
Subject: Re: Problems with RTP Payload Types
In-Reply-To: <c=US%a=_%p=msft%l=RED-68-MSG-970219201337Z-4059@INET-04-IMC.microsoft.com>
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Eric,

> What we are doing is using the dynamic values to specify specific codec and 
> encoding values. Since our application has control over both the client and 
> server "pieces" of the communication stream, we have arbitrarily defined 
> that "Dynamic value x means codec Y at a specific sample size, sampling 
> frequency and average bytes/sec." This, anyway, is how we have defined the 
> meaning of "dynamic" -- subject to correction from this list.

The root of the problem is that you have interpreted "dynamic" as
"private".  The point of dynamic payload types is that a given number
will indicate a different encoding in different sessions.  The mapping
for a given session is specified outside RTP; for example, SDP defines
a means to specify RTP dynamic payload type mappings.  The allocation
of a space of 32 type codes was based on the assumption that in a
given session no more than 32 distinct encodings would be needed.

							-- Steve


From rem-conf-request@es.net Wed Feb 19 18:55:23 1997 
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From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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Eric Fleischman wrote:
> 
> It is thus desirable
> that we address how these dynamic values can best be used in "on-demand"
> scenarios.

That's where RTSP comes in. Clearly, RTSP can use SDP or other session
description formats, so the server can advertise only those media types
*for each stream* that are actually supported at any given time and
assign the minimal set of PTs for those, without the client having to
"remember" any of them.

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

From rem-conf-request@es.net Wed Feb 19 20:45:06 1997 
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Date: Wed, 19 Feb 1997 17:40:02 -0800 ()
From: Stephen Casner <casner@precept.com>
To: Eric Fleischman <ericfl@MICROSOFT.com>
cc: 'Henning Schulzrinne' <schulzrinne@cs.columbia.edu>, 
    "'rem-conf@es.net'" <rem-conf@es.net>
Subject: RE: Problems with RTP Payload Types
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> Because the PT contains both audio and video entries, I originally imagined 
> that both spaces had to be satisfied within that same PT class space

No, the dynamic PT space is separate for each RTP session.  In
addition to the spaces being separate for audio and video sessions, a
multimedia session which defined, for example, three audio RTP
sessions could use different dynamic PT assignments for each of the
audio sessions.

> 1)	our particular content may or may not be pre-announced via SIP or a 
> similar capability. If it is pre-announced then we can benefit from the PT 
> values being constrained to the values indicated within the announcement as 
> you indicated. If it is not pre-announced, but rather initiated in an "on 
> demand" fashion by user(s), then the PT values need to have already been 
> agreed to by the sender-receiver and thus should include all of the 
> encodings currently supported on that server. An example of "on-demand" use 
> is when one or more users request that a pre-produced multimedia production 
> be streamed to them.

It is likely that there is some information that needs to be
communicated between the server and client if you are going to use
RTP.  At a minimum, the port numbers to be used.  You should be able
to communicate the PT mapping along with the other info.  If the
server supports multiple encodings, then there may even have been a
negotiation between client and server.  As Henning mentioned, RTSP is
applicable here.

> Unfortunately, I can't speak for the future (as indicated by my original 
> message) because the future may entail different codec innovation rates 
> than today. If the rate of innovation increases, then we may or may not 
> have a problem. Another relevant factor is how persistent the "older 
> encodings" remain. (If they don't disappear then they take up space.) In 
> any case, I can easily imagine the non-dynamic PT values quickly filling up 
> with entries with an unknown shelf life viability. Thus I suspect that 
> long-term only dynamic PT values will remain viable. It is thus desirable 
> that we address how these dynamic values can best be used in "on-demand" 
> scenarios.

It is essential that the dynamic PT values remain dynamic, or as you
say, we will run out.  It is just like we are now realizing with IP
addresses.  So, this has to be per session so it is independent of
innovation rate.

Regarding the non-dynamic part of the space, it seems likely that many
new codecs will be used only with dynamic payload types throughout
their lifetime.  Perhaps only (a subset of) those standardized by the
ITU will get static PT assignments.
							-- Steve


From rem-conf-request@es.net Wed Feb 19 23:36:41 1997 
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From: bill@wizard.gsfc.nasa.gov (Bill Fink)
Message-Id: <9702200435.AA25163@wizard.gsfc.nasa.gov>
Subject: Re: MBONE RTP session
To: ljhawk@bbnplanet.com (John Hawkinson)
Date: Wed, 19 Feb 1997 23:35:39 -0500 (EST)
Cc: Michael.Speer@eng.sun.com, rem-conf@es.net, 
    bill@wizard.gsfc.nasa.gov (Bill Fink)
In-Reply-To: <199702190211.VAA16484@all-purpose-gunk.near.net> from "John Hawkinson" at Feb 18, 97 09:11:17 pm
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> > What happened to the "MBONE RTP Audio" session?  It doesn't seem to announced
> > anymore.
> 
> My workstation is still happily announcing it.
> 
> --jhawk

I also no longer see the "MBONE RTP Audio" session.

				-Bill



wizard# mtrace ap-gunk.near.net sap.mcast.net
Mtrace from 199.94.220.184 to 128.183.115.17 via group 224.2.127.254
Querying full reverse path... * switching to hop-by-hop:
  0  wizard.gsfc.nasa.gov (128.183.115.17)
 -1  rtr-sno-e1.gsfc.nasa.gov (128.183.115.1)  DVMRP  thresh^ 0  
 -2  mbone2-f.gsfc.nasa.gov (128.183.242.17)  DVMRP  thresh^ 16  
 -3  * * mbone-cf.gsfc.nasa.gov (128.183.251.129)  DVMRP  thresh^ 16  
 -4  * * mbone.nsi.nasa.gov (192.203.230.241)  DVMRP  thresh^ 32  
 -5  sanjose1-mbone1.bbnplanet.net (4.0.0.34)  PIM/Special  thresh^ 0  
 -6  paloalto-mbone1.bbnplanet.net (131.119.0.197)  PIM/Special  thresh^ 32  
 -7  collegepk-mbone1.bbnplanet.net (128.167.252.196)  PIM/Special  thresh^ 64  
 -8  cambridge1-mbone1.bbnplanet.net (199.94.207.2)  PIM/Special  thresh^ 32  
 -9  * * * * frobozz-magic-robot-e6-5.bbnplanet.com (192.52.71.34) didn't respond
-10  * * * 
-11  * * * 
-12  * * * ...giving up
Round trip time 196 ms; total ttl of 66 required.


Results after 39 seconds:

  Source        Response Dest    Overall     Packet Statistics For Traffic From
   * * *        224.0.1.32       Packet      199.94.220.184 To 224.2.127.254
     v       __/  rtt  215 ms     Rate       Lost/Sent = Pct  Rate
199.94.207.2    cambridge1-mbone1.bbnplanet.net 
     v     ^      ttl   33                            0/0    = --%   0 pps
128.167.252.196 collegepk-mbone1.bbnplanet.net 
     v     ^      ttl   66                            0/0    = --%   0 pps
131.119.0.197   paloalto-mbone1.bbnplanet.net 
     v     ^      ttl   66                            0/0    = --%   0 pps
4.0.0.34        sanjose1-mbone1.bbnplanet.net 
     v     ^      ttl   66     v  51 pps        0/0    = --%   0 pps
192.203.230.241 mbone.nsi.nasa.gov 
     v     ^      ttl   66       164 pps        0/0    = --%   0 pps
128.183.251.129 mbone-cf.gsfc.nasa.gov 
     v     ^      ttl   66       105 pps        0/0    = --%   0 pps
128.183.242.17  mbone2-f.gsfc.nasa.gov 
     v     ^      ttl   66         5 pps        0/0    = --%   0 pps
128.183.251.28 
128.183.115.1   rtr-sno-e1.gsfc.nasa.gov 
     v      \__   ttl   66         0 pps
128.183.115.17  128.183.115.17
  Receiver      Query Source

From rem-conf-request@es.net Thu Feb 20 11:27:50 1997 
Received: from all-purpose-gunk.near.net by osi-west.es.net 
          with ESnet SMTP (PP); Thu, 20 Feb 1997 08:27:10 -0800
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          id LAA26181; Thu, 20 Feb 1997 11:27:07 -0500 (EST)
Date: Thu, 20 Feb 1997 11:27:07 -0500 (EST)
Message-Id: <199702201627.LAA26181@all-purpose-gunk.near.net>
To: rem-conf@es.net
Subject: Re: MBONE RTP session
From: John Hawkinson <jhawk@bbnplanet.com>


So, in case anyone cares, it turns out there was indeed something
broken in my local topology (in addition to whatever confusion caused
mtraces from Sun to myself to break) resulting in the MBone RTP Audio
announcement failing to get out.

That's been fixed, sorry for any confusion.

--jhawk


From rem-conf-request@es.net Thu Feb 20 12:10:31 1997 
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          Thu, 20 Feb 1997 11:09:40 -0600
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          id LAA08584; Thu, 20 Feb 1997 11:09:39 -0600
Date: Thu, 20 Feb 1997 11:09:39 -0600
From: Matt Crawford <crawdad@FNAL.GOV>
Subject: Re: Seminar: C. Huitema "The Internet Architecture - End-to-End 
         Principle"
In-reply-to: "19 Feb 1997 16:13:43 EST." <"330B6D07.3ED4"@ctr.columbia.edu>
Sender: crawdad@gungnir.fnal.gov
To: Raymond Liao <liao@ctr.columbia.edu>
Cc: rem-conf@es.net, mnl@ctr.columbia.edu, campbell@ctr.columbia.edu
Message-id: <199702201709.LAA08584@gungnir.fnal.gov>
Content-transfer-encoding: 7BIT
X-Face: /RKQi"kntyd}7l)d8n%'Dum<~(aMW3,5g&'NiH5I4Jj|wT:j;Qa$!@A<~/*C:{:MmAQ:o%S 
        /KKi}G4_.||4I[9!{%3]Hd"a*E{<k&QF?d6L7o&zLqb%kXn!!]ykXMKtTiy9#20]$EKP/^Z$T]'P6, 
        8L#r&mH4PB<ljN,_.=iCpv#N:HIcy5t7{HV:<=g=V?^;-d,J*xkq0r

I'm getting really weird looking video (although I can recognize
Christian) and no audio using the parameters in Raymond Liao's
message:

> Multicast Address:
>     audio:  DVI, RTP, 224.2.243.158/30528/127
>     video:  H.261, RTP, 224.2.154.154/59442/127

From rem-conf-request@es.net Thu Feb 20 14:46:31 1997 
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          Thu, 20 Feb 1997 11:45:09 -0800
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          id AA13421; Thu, 20 Feb 97 11:44:57 PST
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          Thu, 20 Feb 1997 11:44:56 -0800
From: Michael mcCarthy <mccarthy@cs.nps.navy.mil>
Message-Id: <9702201144.ZM22242@grant.cs.nps.navy.mil>
Date: Thu, 20 Feb 1997 11:44:56 -0800
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--PART-BOUNDARY=.19702201144.ZM22242.cs.nps.navy.mil
X-Zm-Content-Name: vrml
Content-Description: Text
Content-Type: text/plain ; name="vrml" ; charset=us-ascii

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

         VIRTUAL REALITY MODELING LANGUAGE (VRML) SYMPOSIUM 

            Monterey, California, February 24-26, 1997

                   http://www.sdsc.edu/vrml97/


The Second Annual Virtual Reality Modeling Language (VRML) Symposium is 
scheduled for February 24-26, 1997 at the Hyatt Regency in Monterey, 
California.  VRML97 is an academic and technical symposium sponsored by 
ACM SIGGRAPH and SIGCOMM, in cooperation with the VRML Consortium.

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

400-500 attendees from across the globe are expected to attend 
this powerful 3-day technical symposium.  Attendees can attend half-day or 
full-day tutorials, which are designed for both intermediate and advanced 
VRML users and developers.  The Symposium also includes working groups,
panels, academic papers and in-depth exhibitor demonstrations.  Topics 
include 3D graphics, content, behaviors, interfaces, modeling, simulation, 
multi-user environments, networking and standards. 

The Hyatt Regency Monterey is headquarters hotel for VRML 97.  For 
registration info, please refer to the VRML 97 home page 
http://www.sdsc.edu/vrml97/register.html or contact Lynn Johnson at 
408-655-9924 or ljohnson@redshift.com.

MBone production comments to vrlm97@cs.nps.navy.mil,
scheduling conflicts to mccarthy@cs.nps.navy.mil 

Symposium MBone Broadcast Schedule:
(Demo SIG, papers and panels to be broacasted live via Mbone)

Monday, February 24, 1997 

     5:00 PM Demo SIG at the Hyatt Regency Ballroom - Timothy Childs (Sponsored 
         by VeRGe and Microsoft) 
     
Tuesday, February 25, 1997 

     8:30 AM Papers: Implementation I, Val Watson session chair 
         Using spatial techniques to decrease message passing in a distributed VE 
         system, Olof Hagsand, Rodger Lea and Marten Stenius, Swedish Institute of 
         Computer Science (SICS) and Sony Computer Science Lab 
         An Object Oriented Approach to VRML Development, Curtis A. Beeson, Silicon
         Graphics Inc. 
         Object-Oriented VRML for Multi-user Environments, Sungwoo Park 
     10:00 AM Break 
     10:00 AM Exhibits open 
     10:30 AM Papers: Multi-user, Dave Nadeau session chair 
         Populating the Internet: Supporting Multiple Users and Shared Applications 
         with VRML, Wolfgang Broll, German National Institute for Information 
         Technology 
         Community Place: architecture and performance, Rodger Lea, Yasuaki Honda,
         Kouichi Matsuda and Satoru Matsuda, Sony Computer Science Lab 
         SpaceFusion: A multi-server architecture for shared virtual environments, 
         Hiroyasu Sugano, Fujitsu 
     12:00 PM VRML Consortium Report: Elected VRML Consortium Inc. Board of 
         Directors
     2:00 PM Papers: Navigation and User Interface, Dave Kerlick session chair 
         QOTA: A Fast, Multi-Purpose Algorithm For Terrain Following in Virtual
         Environments, John W. Barrus and Richard Waters, Mitsubishi Electric 
         Research Laboratory 
         MaPS: Movement and Planning Support for Navigation in an Immersive VRML
         Browser, John Edwards and Chris Hand, De Montfort University, Leicester UK 
         Adapting VRML 2.0 for Immersive Use, Randy Stiles, Sandeep Tewari and Mihir
         Mehta, Lockheed Martin Advanced Technology Center 
     3:30 PM Break 
     4:00 PM Panel: Avatar and Multi-user Standards, Moderator Bernie Roehl - 
         University of Waterloo, Mitra - ParaGraph, David Anderson - 
         MERL, Yasuaki Honda - Sony Computer Science Lab 
     5:00 PM Panel: Using Java as a VRML development platform, Moderator: Jeff 
             Close, Chris Marrin - SGI, Chris Laurel - Dimension X, John DeCuir - 
             Sony Computer Research Laboratory 
     6:00 PM Break 
   
Wednesday, February 26, 1997 

      8:30 AM Plenary speaker: Tony Parisi, Intervista 
      9:00 AM Working group and workshop reports 
     10:00 AM Break 
     10:00 AM Exhibits open 
     10:30 AM Papers: Case Studies, Tamara Munzner session chair 
         The Out of Box Experience: Lessons learned creating compelling VRML 2.0 
         content,Sam Chen, Rob Myers and Rick Pasetto, Silicon Graphics Inc. 
         Networked VR System: Kitchen Layout Design for Customers, Tomohiro Fukuda,
         Ryuichiro Nagahama and Junji Nomura, Matsusita Electric Works Ltd. 
         Animation for Performance Debugging of Parallel Computing Systems, 
         Noritaka Osawa, Hisaya Morita and Toshitsugu Yuba, The University of
         Electro-Communications, Japan 
         Using VRML to Access Manufacturing Data, Sandy Ressler, Qiming Wang, 
         Scott Bodarky, Charles Sheppard and Gregory Seidman, National Institute of 
         Standards and Technology 
    
     2:00 PM Papers: Implementation II, Dick Puk moderator 
         V-COLLIDE: Accelerated Collision Detection for VRML, Tom Hudson, 
         Ming C. Lin, Jon Cohen, Stefan Gottschalk and Dinesh Manocha, 
         University of North Carolina Lodestar: An Octree-Based Level of Detail 
         Generator for VRML, Dieter Schmalstieg, Vienna University of Technology, 
         Austria 
         Wired for Speed: Efficient Routes for VRML 2.0, Daniel Woods, Alan 
         Norton and Gavin Bell, Silicon Graphics Inc. and Wazabi 
     3:00 PM Exhibits close, tear down 
     3:30 PM Break 
     4:00 PM Panel: Expanding Cyberspace: Persistence, Scalability, and Security in 
          VRML, Moderator: Tony Parisi - Intervista Software, Clay Graham - 
          tinySystems, Dan Lipkin - Oracle, Bob Rockwell - Black Sun Interactive 
     5:00 PM Panel: User Interface Design for Digital Space: 3d Navigation and 
         Manipulation,
         Moderator: Bernie Roehl - University of Waterloo, Maclen Marvit - Worlds 
         Inc., James Waldrop - Construct, Mark Pesce - Author 
     6:00 PM Symposium complete 
     



--PART-BOUNDARY=.19702201144.ZM22242.cs.nps.navy.mil--


From rem-conf-request@es.net Thu Feb 20 14:47:58 1997 
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          Thu, 20 Feb 1997 11:46:47 -0800
From: Michael mcCarthy <mccarthy@cs.nps.navy.mil>
Message-Id: <9702201146.ZM22259@grant.cs.nps.navy.mil>
Date: Thu, 20 Feb 1997 11:46:47 -0800
X-Mailer: Z-Mail (3.2.3 08feb96 MediaMail)
To: rem-conf@es.net
Subject: VRML 97 MBone Broadcast Announcement
Mime-Version: 1.0
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              boundary="PART-BOUNDARY=.19702201146.ZM22259.cs.nps.navy.mil"


--PART-BOUNDARY=.19702201146.ZM22259.cs.nps.navy.mil
X-Zm-Content-Name: vrml
Content-Description: Text
Content-Type: text/plain ; name="vrml" ; charset=us-ascii

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

         VIRTUAL REALITY MODELING LANGUAGE (VRML) SYMPOSIUM 

            Monterey, California, February 24-26, 1997

                   http://www.sdsc.edu/vrml97/


The Second Annual Virtual Reality Modeling Language (VRML) Symposium is 
scheduled for February 24-26, 1997 at the Hyatt Regency in Monterey, 
California.  VRML97 is an academic and technical symposium sponsored by 
ACM SIGGRAPH and SIGCOMM, in cooperation with the VRML Consortium.

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

400-500 attendees from across the globe are expected to attend 
this powerful 3-day technical symposium.  Attendees can attend half-day or 
full-day tutorials, which are designed for both intermediate and advanced 
VRML users and developers.  The Symposium also includes working groups,
panels, academic papers and in-depth exhibitor demonstrations.  Topics 
include 3D graphics, content, behaviors, interfaces, modeling, simulation, 
multi-user environments, networking and standards. 

The Hyatt Regency Monterey is headquarters hotel for VRML 97.  For 
registration info, please refer to the VRML 97 home page 
http://www.sdsc.edu/vrml97/register.html or contact Lynn Johnson at 
408-655-9924 or ljohnson@redshift.com.

MBone production comments to vrlm97@cs.nps.navy.mil,
scheduling conflicts to mccarthy@cs.nps.navy.mil 

Symposium MBone Broadcast Schedule:
(Demo SIG, papers and panels to be broacasted live via Mbone)

Monday, February 24, 1997 

     5:00 PM Demo SIG at the Hyatt Regency Ballroom - Timothy Childs (Sponsored 
         by VeRGe and Microsoft) 
     
Tuesday, February 25, 1997 

     8:30 AM Papers: Implementation I, Val Watson session chair 
         Using spatial techniques to decrease message passing in a distributed VE 
         system, Olof Hagsand, Rodger Lea and Marten Stenius, Swedish Institute of 
         Computer Science (SICS) and Sony Computer Science Lab 
         An Object Oriented Approach to VRML Development, Curtis A. Beeson, Silicon
         Graphics Inc. 
         Object-Oriented VRML for Multi-user Environments, Sungwoo Park 
     10:00 AM Break 
     10:00 AM Exhibits open 
     10:30 AM Papers: Multi-user, Dave Nadeau session chair 
         Populating the Internet: Supporting Multiple Users and Shared Applications 
         with VRML, Wolfgang Broll, German National Institute for Information 
         Technology 
         Community Place: architecture and performance, Rodger Lea, Yasuaki Honda,
         Kouichi Matsuda and Satoru Matsuda, Sony Computer Science Lab 
         SpaceFusion: A multi-server architecture for shared virtual environments, 
         Hiroyasu Sugano, Fujitsu 
     12:00 PM VRML Consortium Report: Elected VRML Consortium Inc. Board of 
         Directors
     2:00 PM Papers: Navigation and User Interface, Dave Kerlick session chair 
         QOTA: A Fast, Multi-Purpose Algorithm For Terrain Following in Virtual
         Environments, John W. Barrus and Richard Waters, Mitsubishi Electric 
         Research Laboratory 
         MaPS: Movement and Planning Support for Navigation in an Immersive VRML
         Browser, John Edwards and Chris Hand, De Montfort University, Leicester UK 
         Adapting VRML 2.0 for Immersive Use, Randy Stiles, Sandeep Tewari and Mihir
         Mehta, Lockheed Martin Advanced Technology Center 
     3:30 PM Break 
     4:00 PM Panel: Avatar and Multi-user Standards, Moderator Bernie Roehl - 
         University of Waterloo, Mitra - ParaGraph, David Anderson - 
         MERL, Yasuaki Honda - Sony Computer Science Lab 
     5:00 PM Panel: Using Java as a VRML development platform, Moderator: Jeff 
             Close, Chris Marrin - SGI, Chris Laurel - Dimension X, John DeCuir - 
             Sony Computer Research Laboratory 
     6:00 PM Break 
   
Wednesday, February 26, 1997 

      8:30 AM Plenary speaker: Tony Parisi, Intervista 
      9:00 AM Working group and workshop reports 
     10:00 AM Break 
     10:00 AM Exhibits open 
     10:30 AM Papers: Case Studies, Tamara Munzner session chair 
         The Out of Box Experience: Lessons learned creating compelling VRML 2.0 
         content,Sam Chen, Rob Myers and Rick Pasetto, Silicon Graphics Inc. 
         Networked VR System: Kitchen Layout Design for Customers, Tomohiro Fukuda,
         Ryuichiro Nagahama and Junji Nomura, Matsusita Electric Works Ltd. 
         Animation for Performance Debugging of Parallel Computing Systems, 
         Noritaka Osawa, Hisaya Morita and Toshitsugu Yuba, The University of
         Electro-Communications, Japan 
         Using VRML to Access Manufacturing Data, Sandy Ressler, Qiming Wang, 
         Scott Bodarky, Charles Sheppard and Gregory Seidman, National Institute of 
         Standards and Technology 
    
     2:00 PM Papers: Implementation II, Dick Puk moderator 
         V-COLLIDE: Accelerated Collision Detection for VRML, Tom Hudson, 
         Ming C. Lin, Jon Cohen, Stefan Gottschalk and Dinesh Manocha, 
         University of North Carolina Lodestar: An Octree-Based Level of Detail 
         Generator for VRML, Dieter Schmalstieg, Vienna University of Technology, 
         Austria 
         Wired for Speed: Efficient Routes for VRML 2.0, Daniel Woods, Alan 
         Norton and Gavin Bell, Silicon Graphics Inc. and Wazabi 
     3:00 PM Exhibits close, tear down 
     3:30 PM Break 
     4:00 PM Panel: Expanding Cyberspace: Persistence, Scalability, and Security in 
          VRML, Moderator: Tony Parisi - Intervista Software, Clay Graham - 
          tinySystems, Dan Lipkin - Oracle, Bob Rockwell - Black Sun Interactive 
     5:00 PM Panel: User Interface Design for Digital Space: 3d Navigation and 
         Manipulation,
         Moderator: Bernie Roehl - University of Waterloo, Maclen Marvit - Worlds 
         Inc., James Waldrop - Construct, Mark Pesce - Author 
     6:00 PM Symposium complete 
     



--PART-BOUNDARY=.19702201146.ZM22259.cs.nps.navy.mil--


From rem-conf-request@es.net Thu Feb 20 16:14:05 1997 
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          Thu, 20 Feb 1997 13:13:08 -0800
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          id QAA04718; Thu, 20 Feb 1997 16:12:06 -0500 (EST)
From: Dave Thaler <thalerd@eecs.umich.edu>
Message-Id: <199702202112.QAA04718@dip.eecs.umich.edu>
Subject: Re: Seminar: C. Huitema "The Internet Architecture - End-to-End
To: crawdad@FNAL.GOV (Matt Crawford)
Date: Thu, 20 Feb 1997 16:12:06 -0500 (EST)
Cc: liao@ctr.columbia.edu, rem-conf@es.net, mnl@ctr.columbia.edu, 
    campbell@ctr.columbia.edu
In-Reply-To: <199702201709.LAA08584@gungnir.fnal.gov> from "Matt Crawford" at Feb 20, 97 11:09:39 am
X-Mailer: ELM [version 2.4 PL24 PGP1]
MIME-Version: 1.0
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Content-Transfer-Encoding: 7bit

> I'm getting really weird looking video (although I can recognize
> Christian) and no audio using the parameters in Raymond Liao's
> message:
> 
> > Multicast Address:
> >     audio:  DVI, RTP, 224.2.243.158/30528/127
> >     video:  H.261, RTP, 224.2.154.154/59442/127

I experienced the same thing.  I looked at "global stats" in vat-4.0b2
and it showed a bunch of wrong RTP version errors I think.

I was able to listen in with rat-2.6a14 though.

-Dave

From rem-conf-request@es.net Fri Feb 21 08:44:09 1997 
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          Fri, 21 Feb 1997 05:43:28 -0800
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          Fri, 21 Feb 1997 13:43:01 GMT
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From: Markus Wiesinger <M.Wiesinger1@student.derby.ac.uk>
Organization: University of Derby
To: rem-conf@es.net
Date: Fri, 21 Feb 1997 13:40:32 +0000
Subject: videoconferencing
Return-receipt-to: "Markus Wiesinger" <M.Wiesinger1@student.derby.ac.uk>
Priority: normal
X-mailer: Pegasus Mail for Windows (v2.23)
Message-ID: <4FAD1E8377C@sdk3.derby.ac.uk>

I'm an exchange  student from Austria studying at the University of Derby (Uk)
 and I currently work on a project dealing with videoconferencing. I am 
especially interested in desktop videoconferencing over ISDN and Internet.

If you can give me further information on this topic please do not 
hesitate to do so. Scientific research documents would please me 
most.

Thanks a lot,
Markus Wiesinger

e-mail: M.Wiesinger1@student.derby.ac.uk

From rem-conf-request@es.net Fri Feb 21 16:22:01 1997 
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          Fri, 21 Feb 1997 13:21:12 -0800
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          Fri Feb 21 16:19:28 EST 1997
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          Fri, 21 Feb 1997 16:17:05 -0500 (EST)
Message-ID: <330E1103.2D28@research.att.com>
Date: Fri, 21 Feb 1997 16:17:56 -0500
From: "M. Reha Civanlar" <civanlar@research.att.com>
Reply-To: civanlar@research.att.com
Organization: AT&T Labs - Research
X-Mailer: Mozilla 3.0Gold (Win95; I)
MIME-Version: 1.0
To: Stephen Casner <casner@precept.com>
CC: rem-conf@es.net
Subject: Re: Problems with RTP Payload Types
References: <Pine.WNT.3.95.970219170925.-101443Q-100000@oak.precept.com>
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I guess, one remaining question is how will one define new RTP payloads
(not PT assignments) for these new codecs so that they can be archived
as IETF documents. Is it viable that the AVT group continues to monitor
this process for the foreseeable future? 

Stephen Casner wrote:
> 
> Regarding the non-dynamic part of the space, it seems likely that many
> new codecs will be used only with dynamic payload types throughout
> their lifetime.  Perhaps only (a subset of) those standardized by the
> ITU will get static PT assignments.
>                                                         -- Steve

-- 
M. Reha Civanlar
Principal Technical Staff Member, AT&T Labs - Research
101 Crawfords Corner Road, 4C520, Holmdel, NJ 07733-3030
(908) 949 6705
(908) 949 3697 (fax)
civanlar@research.att.com
http://www.research.att.com/info/mrc

From rem-conf-request@es.net Fri Feb 21 18:03:48 1997 
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          with ESnet SMTP (PP); Fri, 21 Feb 1997 15:03:09 -0800
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          Fri Feb 21 18:02:04 EST 1997
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          Fri, 21 Feb 1997 17:58:47 -0500 (EST)
Sender: hgs@dnrc.bell-labs.com
Message-ID: <330E28A7.1FC6@cs.columbia.edu>
Date: Fri, 21 Feb 1997 17:58:47 -0500
From: "Henning Schulzrinne (BL)" <hgs@cs.columbia.edu>
Organization: Columbia University
X-Mailer: Mozilla 3.0Gold (X11; I; SunOS 5.4 sun4m)
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To: rem-conf@es.net, confctrl@isi.edu
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A new draft of the RTSP specification is now available as PostScript at
http://www.cs.columbia.edu/~hgs/rtsp/draft/rtsp.ps

This will be submitted as an I-D in the next few days (after
ASCIIfication).

The draft raises a number of questions for discussion. The authors look
forward to discussion and comments (on the confctrl mailing list or
privately, if needed).

Henning


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

From rem-conf-request@es.net Mon Feb 24 07:48:45 1997 
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From: multicast <multicast@dxcoms.cern.ch>
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Subject: Seminar on physics results from the LEP run at 172 GeV

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

Title:       
	Seminar on physics results from the LEP run at 172 GeV
Date:        
	Feb 25, 1997

Time:        
	14:00 GMT+1 2 hours

Contact:     
	multicast@noc.cern.ch   

URL:         
	http://sunmed2.cern.ch:8888/Seminar_25_02_97.html

Description:        
	 









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

From rem-conf-request@es.net Mon Feb 24 18:30:05 1997 
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Date: Mon, 24 Feb 1997 12:31:22 -0500
From: rodgers@nlm.nih.gov (R. P. C. Rodgers, M.D.)
Message-Id: <199702241731.MAA25215@billings.csb>
To: dem@fnal.gov, evi@boulder.Colorado.EDU, likavec@igd.fhg.de, mbone@isi.edu, 
    rem-conf@es.net
Subject: WWW6 Multicasting Plans

To: rem-conf-request@es.net, mbone-request@isi.edu
Subject: WWW6 Multicasting 7-11 April 1997

Dear MBONE/Net Colleagues,

We would like to reserve the capability to multicast audio/video/wb/nt
>from the Sixth International World Wide Conference, to be held 7-11 April 1997
at the Santa Clara Convention Center in Nothern California, USA.  We realize
that the IETF is occurring the same week, so will take extra care to
coordinate our activities with our colleagues at the IETF to avoid generation
of excessive bandwidth (hi there, Evi! 8^).  We are tentatively announcing
four channels via sdr (2 for live presentation, 2 for tape delay rebroadcast),
though we would never have more than 2 running concurrently, and very likely
not even that most of the time.  We are not certain at this point if the
tape delay rebroadcast will occur at all.  Sdr announcments will be updated
as plans firm up.  In the tradition of recent conferences,
we also anticipate the possibility of using one of the new webcasting tools
to experimentally multicast some of the presenter's documents in html.

The Web server for the Conf. is at: http://www6conf.slac.stanford.edu/,
though it does not currently contain any information about multicasting
schedules.

For further information, please contact any of:

   David Martin (dem@fnal.gov)
   Jaromir Likavec (likavec@igd.fhg.de)
   Rick Rodgers (rodgers@.nlm.nih.gov)
   
Best Regards, Rick Rodgers

From rem-conf-request@es.net Mon Feb 24 18:44:10 1997 
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To: rem-conf@es.net, 298-list@bmrc.Berkeley.EDU
From: Jerrlyn Iwata <jerrlyn@postgres.Berkeley.EDU>
Subject: [Reminder] Berkeley Multimedia & Graphics Seminar

                Berkeley Multimedia and Graphics Seminar 
        (Wednesday February 26, 1997 12:30-2:00 PDT 405 Soda Hall) 

        "Why the Future of the Internet is not Multimedia and Other Assorted
High Heresies" 

                                 Mike O'Dell 
                              UUNET Technologies 

        MBONE BROADCAST BEGINS AT 12.40 PDT (GMT - 8 HOURS)



From rem-conf-request@es.net Tue Feb 25 10:06:50 1997 
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To: rem-conf@es.net
From: PC Center <pchelp@ucop.edu>
Subject: Windows mbone clients
Mime-Version: 1.0
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I'd appreciate hearing about any free Windows based mbone clients I might try. 

From rem-conf-request@es.net Tue Feb 25 10:41:38 1997 
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> I'd appreciate hearing about any free Windows based mbone clients I might try. 
> 
> 

I'm interested in it too. Please make a pubblic reply.

Thank you

From rem-conf-request@es.net Tue Feb 25 11:14:01 1997 
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Date: Tue, 25 Feb 97 11:13:20 EST
From: andrew@calvin.dgbt.doc.ca (Andrew Patrick)
Message-Id: <9702251613.AA15521@calvin.dgbt.doc.ca>
Reply-To: andrew@calvin.dgbt.doc.ca
X-Mailer: Mail User's Shell (7.2.5 10/14/92)
To: rem-conf@es.net, mbone@ISI.EDU, multicast@canarie.ca
Subject: "mpoll" - a new multicast polling tool

As part of the MERCI project, I have been working on a new multicast
application for real-time polling of opinions and ratings. "MPOLL"
was developed with the following uses in mind:

   - collecting quality ratings during multicast sessions
   
   - collecting opinions and votes during multicast collaborative
      work sessions
   
   - collecting opinions on general topics or current events

MPOLL may be useful during multicast sessions to get opinions from
all the participants quickly and efficiently.  It may also be useful
for collecting quality and reception ratings during MBONE events.  I
am releasing MPOLL now to get some initial feedback, and will also be
announcing an MPOLL test session via SDR (you will need a plugin).

You can pick-up the current (very early Alpha) version of MPOLL at

   ftp://debra.dgbt.doc.ca/pub/mbone/mpoll/

source code and binaries for SunOS 4.1.4 and Solaris 2.5 are
available.

Please try the program and tell me what you think.  Any and all
feedback is welcome.

Below are some more details about this program.  You can also find
more information at:

   http://debra.dgbt.doc.ca/mbone/mpoll/mpoll.html


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

                                 MPOLL
                           Version Alpha 2.*
                                                   
             Andrew Patrick, Communications Research Centre
                      <andrew@calvin.dgbt.doc.ca>
                              
Introduction
------------

"MPOLL" is a real-time opinion polling and rating collection tool
that uses multicasting to distribute questions and responses to all
multicast session participants. 

"MPOLL" was developed by Andrew Patrick at the Communications
Research Centre (CRC) in Ottawa, Canada, as part of the MERCI project
on Collaborative Work Tools.

MPOLL was developed with the following uses in mind:

   - collecting quality ratings during multicast sessions
   
   - collecting opinions and votes during multicast collaborative
      work sessions
   
   - collecting opinions on general topics or current events


Using MPOLL
-----------

   Usage: mpoll [-t ttl] [-C "title"] address/port
   
    OPTIONS:
      -C "title" : the theme title for the questions
      -t ttl     : the multicast TTL (default = 15)

MPOLL requires specifying a multicast address and port.   MPOLL can
be used by itself, or with other multicast tools such as VAT and VIC
in an MBONE session. MPOLL can be started from SDR once an
appropriate "plugin" is installed (see below), or from the command
line.  Command options that MPOLL accepts include the multicast TTL
(-t), which defaults to 15, and a session title (-C), which defaults
to "Unknown Theme".  These are the same command options used by other
multicast tools.

Once MPOLL is started, there is built-in help for each of the windows
that explains the features and functions.


How MPOLL Works
---------------

MPOLL works by sending question and response packets to the multicast
address, and listening for packets from other hosts.  The users can
create a new question, and this will become available to all the
users who have joined that multicast session.  As users answer the
question, the responses are distributed to all the participants,
where they are tallied and displayed in a graphical histogram.  MPOLL
packets are redistributed periodically as long as the program is
running, so that all current users will see all the results.

   Normally, MPOLL will be left running continuously during a polling
   period or multicast session.

MPOLL uses a simple plain-text protocol for constructing multicast
packets that is similar to the Session Description Protocol (SDP)
used in SDR.  There are currently 3 types of packets generated by
MPOLL: questions, remove question instructions, and user responses.

MPOLL creates a cache file in your home directory ("~/.mpoll_cache")
for storing the questions from your last MPOLL session.  This allows
you to quit and restart MPOLL and have the questions for a session,
yours and other users, reloaded.


Installing MPOLL
----------------

   Binary Distribution
   -------------------
   
   The MPOLL binary file is a stand-alone C program that does not
   require any extra libraries in order to run.  Simply copy the
   MPOLL program to a directory that is in your path.

   Currently, binary distributions of MPOLL are available for:

      - Solaris 2.5
      - SunOS 4.1.4

   A WIN32 port is planned.  Anyone who can help to compile MPOLL for
   other platforms is asked to contact the author.


   Source Distribution
   -------------------
   
   MPOLL is written in C and Tcl/Tk.  The development environment is:

      - SPARC Solaris 2.5
      - gcc version 2.7.2
      - Tcl 7.6 / Tk 4.2

   In addition, MPOLL uses the Embedded Tk (ET) toolkit by Richard
   Hipp (http://users.vnet.net/drh/ET.html) for integrating C and
   Tcl/Tk programming.  Version 1.7.1 of ET is included in the MPOLL
   source distribution, and it must be installed before MPOLL is
   compiled.
   
   No configuration procedure is available yet for MPOLL.  MPOLL has
   been developed under Solaris 2.5, so that is the default platform
   for make.  Possible make commands are currently: 
   
   For Solaris 2.5:  "make"
   For SunOS 4.1.x:  "make sunos" 
   
   I would love to hear about any problems in building and running
   the program.  Also, if you port MPOLL to another platform, please
   contact me with any information about changes necessary, and if
   you can supply a binary for my archive site.


   The SDR Plugin
   --------------
   
   Since MPOLL is a new program, and polling is a new MBONE
   application, an appropriate media has not been defined in SDR. 
   So, in order to use MPOLL from SDR you must install a "plugin".  
   The plugin provided in the distribution is
   
      sdr2.plugin.S53.poll.udp.mpoll.mpoll
   
   and you should copy this to a "~/.sdr.plugins" dirctory on your
   system.  See the SDR documentation for more information about
   plugins.

   
Limitations
-----------

- MPOLL currently has no unicast support

- MPOLL currently only supports multiple-choice questions with 1 to 5
   possible responses.  Questions can be created that allow one or
   many responses from the users.
   
- MPOLL is currently limited to 100 questions per session, and up to
   5 possible answers per question.  The question limit should be
   configurable be editing the MAX_TOPICS parameter in "mpoll.h", but
   changes to the MAX_RESPONSES parameter will probably not work.

- MPOLL should someday read the ~/.RTPdefaults file, and perhaps some
   other X resources

- a function to visibily remind users to update their responses
   should be added

- we may need a function to edit an existing question


Known Problems
--------------

- this program is in an early Alpha stage of development.  There are
   probably lots of features missing, and many bugs.

- MPOLL has only been tested on Sun systems.  No work on
   portability has been done yet.
   
- no attempt is made to do robust font selection, so MPOLL will
   probably fail if the fonts I use are not on your system.  See
   "mpoll.h" for the font definitions.

- the command line args "-h or -help" dump core, use "--help" instead


Acknowledgements
----------------

The protocol used in MPOLL is fashioned after the SDP protocol used
in SDR, and I thank Mark Handley, Van Jacobson, and other
contributors for their work on SDP and SDR.

I also used the SDR source code as an example while working on MPOLL,
and that code was useful for techniques like multicast socket
management and periodic redistribution of packets.  I learned a lot
>from viewing other people's code, and I am sure that I made lots of
mistakes, for which I am wholely responsible.


Licence Terms
-------------

   Copyright (c) Communications Research Centre, Government of Canada
   1997.  All rights reserved.
   
   License is granted to copy and use this software, for research and
   evaluation purposes, provided that the Communications Research
   Centre is acknowledged in all documentation pertaining to any such
   copy or use. The Communications Research Centre grants no other
   licenses expressed or implied. Title, ownership rights, and
   intellectual property rights in the software shall remain with the
   Communications Research Centre.  The Centre's name and the
   Government of Canada should not be used in any advertising without
   written permission.
   
   THE COMMUNICATIONS RESEARCH CENTRE MAKES NO REPRESENTATIONS
   CONCERNING EITHER THE MERCHANTABILITY OF THIS SOFTWARE OR THE
   SUITABILITY OF THIS SOFTWARE FOR ANY PARTICULAR PURPOSE.  The
   software is provided "as is" without express or implied warranty
   of any kind.
   
   These notices must be retained in any copies of any part of this
   software.
   
  
Author
------

      Dr. Andrew Patrick
      Network Services & Interface Design Laboratory
      Communications Research Centre
      Industry Canada
      3701 Carling Ave.
      P.O. Box 11490, Station 'H'
      Ottawa, ON CANADA
      K2H 8S2

      Phone: (613) 990-4675
      FAX:   (613) 998-9648
      E-Mail: andrew@calvin.dgbt.doc.ca
      WWW: http://debra.dgbt.doc.ca
      

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

From rem-conf-request@es.net Tue Feb 25 12:57:37 1997 
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From: "Jung, Michael" <mikej@finall.com>
To: 'Alessio Casati' <Alessio.Casati@italtel.it>, 
    "'rem-conf@es.net'" <rem-conf@es.net>
Subject: RE: Windows mbone clients
Date: Tue, 25 Feb 1997 12:56:28 -0500
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VIC, VAT, RAT, SDR, SD, Win32-SDR etc can be found at

tis-archive.thepoint.net/Software/Internet/Interactive media/MBONE/

Note the space in "Interactive media"

--mike
Michael Jung
mikej@finall.com

>-----Original Message-----
>From:	Alessio Casati [SMTP:Alessio.Casati@italtel.it]
>Sent:	Tuesday, February 25, 1997 10:44 AM
>To:	rem-conf@es.net
>Subject:	Re: Windows mbone clients
>
>
>> I'd appreciate hearing about any free Windows based mbone clients I
>>might try. 
>> 
>> 
>
>I'm interested in it too. Please make a pubblic reply.
>
>Thank you

From rem-conf-request@es.net Tue Feb 25 13:08:46 1997 
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          Tue, 25 Feb 1997 13:07:12 -0500 (EST)
Sender: liao@ctr.columbia.edu
Message-ID: <33132A47.ABD322C@ctr.columbia.edu>
Date: Tue, 25 Feb 1997 13:07:03 -0500
From: Raymond Liao <liao@ctr.columbia.edu>
Organization: CTR, Columbia University
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MIME-Version: 1.0
To: rem-conf@es.net
CC: mnl@ctr.columbia.edu
Subject: Comet Group Seminar: H. V. Jagadish, AT&T Labs, Data Architectures for 
         Managing Networks
Content-Type: text/plain; charset=us-ascii
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MBone Broadcast Announcement

Title:
    Data Architectures for Managing Networks: From Billing and 
    Account Management to Real Time Performance Management

Speaker:
    H. V. Jagadish
    AT&T Labs
 
Date:
    Thursday, Feb. 27, 1997

Time:
    10:00 AM - 11:00 AM EST

Multicast Address:
    audio:  DVI, RTP, 224.2.243.158/30528 , ttl=127
    video:  H.261, RTP, 224.2.154.154/59442 , ttl=127

Contact:
    Andrew T. Campbell <campbell@ctr.columbia.edu>

URL:
    http://comet.ctr.columbia.edu/activities/seminars/

Place:
    Multimedia Network Laboratory
    8LE1 Schapiro Research Building
    Center for Telecommunications Research
    Columbia University
    New York City, USA

Abstract:

In large networks, huge volumes of data are produced at the element 
level. The challenge is to aggregate this distributed and heteregeneous 
source of information in a manner that greatly decreases the volumes of 
data to be stored or communicated while still retaining the ability to 
answer queries of interest efficiently.

We describe an approach to this problem in the context of a billing
system, and hypothesize extensions of our approach toreal-time 
performance management.  


-- 
Raymond R.-F. Liao
(212) 939-7158
http://www.ctr.columbia.edu/~liao
Center for Telecommunications Research, Columbia University

From rem-conf-request@es.net Tue Feb 25 13:24:55 1997 
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From: Matthew Feldman <mfeldman@VideoConferencing.com>
To: "'rem-conf@es.net'" <rem-conf@es.net>
Subject: FW: Windows mbone clients
Date: Tue, 25 Feb 1997 13:24:00 -0500
X-Mailer: Microsoft Exchange Server Internet Mail Connector Version 4.0.837.3
Encoding: 22 TEXT, 40 UUENCODE
X-MS-Attachment: WINMAIL.DAT 0 00-00-1980 00:00

Me too.
I heard the IPMI group may have some soon.

Matthew Feldman

----------
From: 	Alessio Casati[SMTP:Alessio.Casati@italtel.it]
Sent: 	Tuesday, February 25, 1997 10:43 AM
To: 	rem-conf@es.net
Subject: 	Re: Windows mbone clients


> I'd appreciate hearing about any free Windows based mbone clients I might 
try. 
> 
> 

I'm interested in it too. Please make a pubblic reply.

Thank you



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M'; Z-#,3<$US-!\N+51O-E\P:QJ0;2(M!:!N9D 'D"YNQQ(`.F\U+G5B:BPQ
M/(\I,&M294' 5PN 9&_.=P0@!M$_("!C*; V(7YS(O\+*!0B# $K]D6E/OT@
M0"<@L!=P%@`%D <P+"#;(&,+@&=(D ;@=05 `'#O(> #4 G@0]=B,< )@$1<
MH2!!;6EG: 5 =#D@CBY'R$U>'^8G;2 +@/LL(!J0<RP@(+ +@$^ 3.&Y'[$@
M4#% 2U$AL6L?@,T6H'!!8 )@:6,^40M0"TT@13Q4$<!N:R!Y7PA@1:\>F$==
M&;$`5P`````#`/$_"00```,`)@```````P`V```````"`4<``0```#@```!C
M/553.V$](#MP/4EN=&5L;&5C="!6:61E;R [;#U-04E,0T]-+3DW,#(R-3$X
M,C0P,%HM,3@P``(!^3\!````80````````#<IT#(P$(0&K2Y" `K+^&"`0``
M```````O3SU)3E1%3$Q%0U0@5DE$14\@0T].1D5214Y#24Y'+T]5/4E60RU.
M64,O0TX]4D5#25!)14Y44R]#3CU-05142$571@`````>`/@_`0```! ```!-
M871T:&5W($9E;&1M86X``@'[/P$```!A`````````-RG0,C 0A :M+D(`"LO
MX8(!`````````"]//4E.5$5,3$5#5"!6241%3R!#3TY&15)%3D-)3D<O3U4]
M259#+4Y90R]#3CU214-)4$E%3E13+T-./4U!5%1(15=&`````!X`^C\!````
M$ ```$UA='1H97<@1F5L9&UA;@! ``<P('X5$DDCO % ``@P8+I8$DDCO $#
M``TT_3\```(!%#0!````$ ```%24H< I?Q ;I8<(`"LJ)1<>`#T``0````4`
M``!&5SH@``````L`*0``````"P`C```````"`7\``0```%@````\8SU54R5A
M/5\E<#U);G1E;&QE8W1?5FED96]?)6P]34%)3$-/32TY-S R,C4Q.#(T,#!:
F+3$X,$!M86EL8V]M+G9I9&5O8V]N9F5R96YC:6YG+F-O;3X`/70Q
`
end

From rem-conf-request@es.net Tue Feb 25 16:02:54 1997 
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Date: Tue, 25 Feb 1997 13:01:25 -0800 (PST)
From: Stephane Chatre <chatre@ENGR.ORST.EDU>
cc: rem-conf@es.net
Subject: Multicast simulation
In-Reply-To: <33132A47.ABD322C@ctr.columbia.edu>
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I would like to simulate an MBone session being multicasted using Matlab.
For now, I'm looking for a model of network simulation. Does anyone know
about tools/programs/matlab files available for network simulation?

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


From rem-conf-request@es.net Tue Feb 25 16:50:39 1997 
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Date: Tue, 25 Feb 1997 22:49:07 +0100 (MET)
From: "Lutz Grueneberg, RVS" <gruen@rvs.uni-hannover.de>
X-Sender: gruen@borneo
To: Alessio Casati <Alessio.Casati@italtel.it>
cc: rem-conf@es.net
Subject: Re: Windows mbone clients
In-Reply-To: <9702251540.AA20973@ic8u32.settimo.italtel.it>
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Hi,

On Tue, 25 Feb 1997, Alessio Casati wrote:

>=20
> > I'd appreciate hearing about any free Windows based mbone clients I mig=
ht try.=20
> >=20
> >=20
>=20
> I'm interested in it too. Please make a pubblic reply.
>=20
> Thank you
>=20
A hopefully most current collection of tools for Win95 systems
is available in the RVS Mbone Collection:
ftp://ftp.rvs.uni-hannover.de/pub/products/mbone_collection/mbonew32.zip

Collection of the tools for other OS are available from
the same directory.

German docs on the installation and usage of the tools
is available:
http://www.rvs.uni-hannover.de/products/mbone_collection

Cheers,
 Lutz
--
// Lutz Gr=FCneberg                Lehrgebiet Rechnernetze und Verteilte Sy=
steme
//                               Universit=E4t Hannover
//                               Schlo=DFwender Str. 5
// gruen@rvs.uni-hannover.de     D-30159 Hannover, Germany
// Public PGP-Key available: http://www.rvs.uni-hannover.de/people/gruen.ht=
ml
//                           http://bs.mit.edu:8001/pks-toplev.html


From rem-conf-request@es.net Tue Feb 25 20:14:34 1997 
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          Tue, 25 Feb 1997 17:05:00 -0800
From: jcole@precept.com
Message-Id: <970225170512.ZM3502@little-bear>
Date: Tue, 25 Feb 1997 17:05:09 -0800
Reply-To: jcole@precept.com
X-Mailer: Z-Mail 4.0 (4.0.0 Aug 21 1995)
To: rem-conf@es.net
Subject: Multicast of "ACM 97" Conference Sessions
Mime-Version: 1.0
Content-Type: text/plain; charset=us-ascii

Precept Software will be transmitting 2 1/2 days worth of sessions from the
ACM 97 Conference and Exposition in San Jose, California over the MBONE. The
theme of the event is "The Next 50 Years Of Computing". Video will be H261,
and audio will be PCM, both to be delivered using RTPv2 with Precept's IP/TV
server software.

	Program Name:       ACM 97
	Start Date & Time:  Mar/03/97 - 08:30 PST (16:30 GMT)
	Length:             7 1/2 Hours

	Program Name:       ACM 97
	Start Date & Time:  Mar/04/97 - 08:30 PST (16:30 GMT)
	Length:             7 1/2 Hours

	Program Name:       ACM 97
	Start Date & Time:  Mar/05/97 - 08:30 PST (16:30 GMT)
	Length:             3 1/2 Hours

For more details, see http://www.acm.org/acm97/conference/sched.html.

You can tune in to this MBONE event two ways. The event is being
advertised with Precept's IP/TV Program Guide, which sends out an
sdr-compatible session advertisement to be received by sdr or a local
Program Guide server. Those who want to use Precept's viewer software
for MS Windows but are not running a local Program Guide may access
the one at www.precept.com (this is set by default for the downloaded
viewer). Also located on that web site are details on downloading the
viewer.

ACM has requested this copyright notice be included in the session 
advertisement...

---
Copyright c 1997 ACM,  Association for Computing

The files distributed by this server have been provided by ACM. Copyright 
and all rights therein are maintained by ACM. It is understood that all 
persons copying this information will adhere to the terms and constraints 
invoked by ACM+s  copyright. These works may not be reposted without the 
explicit permission of ACM. Reuse and/or reposting for noncommercial 
classroom use is permitted. Questions regarding usage rights and 
permissions may be addressed to: permissions@acm.org
---

-- 
    John Cole, Systems Engineer         Tel: 415.845.5217      
    Precept Software, Inc.              Fax: 415.845.5235
    1072 Arastradero Road               http://www.precept.com
    Palo Alto, CA 94304                 E-Mail: jcole@precept.com






From rem-conf-request@es.net Wed Feb 26 00:42:18 1997 
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Subject: Subscribe
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subscribe


From rem-conf-request@es.net Wed Feb 26 04:07:12 1997 
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unsubscribe

From rem-conf-request@es.net Wed Feb 26 08:46:15 1997 
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Date: Wed, 26 Feb 1997 05:47:50 -0500
To: hermenh@ix.netcom.com, rem-conf@es.net
From: Christine Perey <cperey@spider.lloyd.com>
Subject: Re: earn $75

How is your study going?

CP


From rem-conf-request@es.net Wed Feb 26 14:30:10 1997 
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To: rem-conf@es.net
Cc: band@std.org
Subject: Severe Tire Damage tonight ~8pm PST
Date: Wed, 26 Feb 97 11:21:46 -0800
From: berc@pa.dec.com
X-Mts: smtp


Who:   Severe Tire Damage
What:  Rock & Roll
Where: The Fabulous SubForum, Palo Alto CA
When:  26-Feb-97 ~8pm PST
Why:   Would you rather smoke two joints or do your homework?
       I knew you were going to say that!

Another session of Severe Tire Damage mcasting their bandwidth-friendly 
message of love to their growing international audience.  This week 
an interview crew from Stern (guten tag bis Deutchland) will be on 
hand to buy decent beer.  Camera control will be available from www.std.org
via the "remote cam" link.  I don't know if we'll have the web controlled 
mixer or fog machine set up this evening.

Many thanks to those who've offered to translate the article in "InternNet 
Guiden" from Swedish.  We now have a reasonable version in English, 
though some phrases like "We know that STD is magnetic, but it is 
magnetic in such new and interesting ways" may have gained new meaning 
in the process.

From rem-conf-request@es.net Wed Feb 26 18:46:01 1997 
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Date: Wed, 26 Feb 1997 15:29:40 -0800 ()
From: Stephen Casner <casner@precept.com>
To: Arlie Davis <arlie@thepoint.net>
cc: rem-conf@es.net
Subject: RE: Progressing IDMR items
In-Reply-To: <c=US%a=_%p=ThePoint_Interne%l=TIS_MAIL-970226201447Z-10@tis-mail.thepoint.net>
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Arlie,

> RTCP puts a much greater strain than is necessary on (S,G)-state
> implementations.  This is rather unfortunate.

Both on the rem-conf list and in the last AVT meeting, we have had
significant discussion of the use of RTCP in large broadcast
scenarios.  I'll attempt to move this discussion back there [bcc to
mbone].

> Imagine 5 million people watching a single multicast of the NFL Super
> Bowl.  Instead of N routers maintaining state for a single (S,G) pair,
> they have to maintain 5,000,001 (S,G) pairs, _no matter how infrequently
> the receivers advertise_.  Aggravating, isn't it?

With a shared-tree routing protocol, there is only (*,G) state for the
receivers because the RTCP data rate is low enough not to trigger the
formation of a source-rooted tree.

> Also, truly, RTCP data isn't always that interesting.  Do I really care
> that much who is listening to the same radio broadcast?  If it's a
> conference, yes.  If it's a broadcast, usually no.

At the AVT meeting, we concluded that there should be additional RTP
profiles specifying alternative RTCP behavior.  It is likely that one
of those would specify that receivers would not send RTCP at all, for
use in scenarios where privacy of reception or other factors outweigh
the value of RTCP's reception feedback.

BTW, at the AVT meeting I invited the submission of proposals for
additional profiles, but I don't think any such proposals have been
submitted yet.  (Not to neglect Bernard Aboba's I-D on the issues and
the useful email on this topic.)
							-- Steve


From rem-conf-request@es.net Thu Feb 27 01:59:00 1997 
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Date: Thu, 27 Feb 1997 08:57:48 +0200 (EET)
From: Seppo Kallio <kallio@cc.jyu.fi>
To: Christine Perey <cperey@spider.lloyd.com>
cc: hermenh@ix.netcom.com, rem-conf@es.net
Subject: Re: earn $75
In-Reply-To: <2.2.32.19970226104750.00ad3284@spider.lloyd.com>
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MIME-Version: 1.0
Content-Type: TEXT/PLAIN; charset=US-ASCII

On Wed, 26 Feb 1997, Christine Perey wrote:

> How is your study going?
> 
> CP

No problems.

Seppo



From rem-conf-request@es.net Thu Feb 27 09:49:32 1997 
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          Thu, 27 Feb 1997 06:48:57 -0800
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Date: Thu, 27 Feb 1997 09:48:54 -0500 (EST)
From: kevin@cc.gatech.edu (Kevin C. Almeroth)
Message-Id: <199702271448.JAA13106@flora.cc.gatech.edu>
To: rem-conf@es.net
Subject: Vic from a file


In my thesis writing-induced haze I cannot remember if I have ever
successfully asked or had the following question answered:

   Has anyone hacked vic, or is there an option to allow someone
   to pipe a file (created through the SunVideo GUI for example)
   
I realize a hack to the source may not be too difficult, but I'd 
prefer not to duplicate effort.

RTPdump and RTPplay are great for analog to digital conversion but
there really isn't any way to get existing digital content (in the
proper format) onto the MBone.

-Kevin Almeroth

From rem-conf-request@es.net Thu Feb 27 13:18:04 1997 
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          Thu, 27 Feb 1997 10:17:14 -0800
Date: Thu, 27 Feb 1997 10:17:14 -0800
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To: 298-list@bmrc.Berkeley.EDU, rem-conf@es.net
From: Jerrlyn Iwata <jerrlyn@postgres.Berkeley.EDU>
Subject: Berkeley Multimedia & Graphics Seminar

                Berkeley Multimedia and Graphics Seminar 
        (Wednesday March 5, 1997 12:30-2:00 PDT 405 Soda Hall) 

                  "The Internet as a Populated Place" 

                             Pavel Curtis 
                            PlaceWare, Inc. 

Advances in technology are radically changing the face of the Internet, from
a huge, comprehensive, but lonely library into a vast web of interconnected
places bustling with live human presences. I'll talk about how these changes
are finally making it possible for people all over the world to use the
Internet as a whole as a vast, powerful tool for interpersonal
communication, collaboration, and sharing. 

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

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

Please direct all technical questions to davesimp@cs.berkeley.edu


From rem-conf-request@es.net Thu Feb 27 13:53:03 1997 
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                   3.2 #7) Thu, 27 Feb 97 10:52:12 PST
PP-warning: Illegal Received field on preceding line
Date: Thu, 27 Feb 97 10:50:00 PST
From: John H Wilson <John_H_Wilson@ccm.jf.intel.com>
Message-ID: <Thu, 27 Feb 97 10:52:10 PST_2@ccm.hf.intel.com>
To: rem-conf@es.net
Subject: RTSP "sessions", "streams", "components", and URIs

IMHO, RTSP could avoid some confusion over the term "session" by being 
self-consistent.

SDP defines the term in one sense: a collection of streams with a common 
time-line.

RTP defines the term in another sense: a single media stream (along with 
its parallel "control" stream (RTP/RTCP).

RTSP has absorbed both terms.

In Section 1.3, it is used in the SDP sense; in section 6.2, GET 
retrieves a session description, and in section 6.11, SESSION allows the 
server to send the client a modified session description with new or 
deleted "components" (it is not clear, but I think "component" is 
synonymous with "stream".)

In section 8.24, session is clearly refering to a single stream 
(consistent with RTP, but RTSP does not insist on its use). It defines 
the session header field which identifies the state of a single stream 
on which SETUP(6.3), PLAY(6.4), PAUSE(6.5), and CLOSE(6.6) operate.

IMHO, "stream" should be used for SETUP, PLAY, PAUSE, CLOSE, and should 
replace the term "component" in the definition of the SESSION request.
The "session" header field would become the "stream" header field.

On a minor, related, point, I wonder why all requests must contain a 
Request-URI. PAUSE and CLOSE, for example, must operate on an existing 
"stream"; if the Request-URI does not match the "stream" ID, I suppose 
it is a protocol error? What would the Request-URI be used for in these 
two requests?

john_h_wilson@ccm.jf.intel.com







From rem-conf-request@es.net Thu Feb 27 14:10:31 1997 
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Date: Thu, 27 Feb 1997 11:09:49 -0800 (PST)
Message-Id: <199702271909.LAA13865@george.arc.nasa.gov>
From: lamaster@george.arc.nasa.gov
To: mbone@isi.edu
Subject: RE: Progressing IDMR items
Cc: rem-conf@es.net



[ "Arlie Davis" <arlie@thepoint.net> wrote:]

|> >> Imagine 5 million people watching a single multicast of the NFL Super
|> >> Bowl.  Instead of N routers maintaining state for a single (S,G) pair,
|> >> they have to maintain 5,000,001 (S,G) pairs, _no matter how infrequently
|> >> the receivers advertise_.  Aggravating, isn't it?
 
["Dino Farinacci" <dino@cisco.com>:]

|>     One would also argue that the RTCP backoff as the group gets larger is
|>     more of a detriment to routers, since the (S,G) timeout will occur
|>     more frequently than the RTCP transmission interval. Therefore, in a
|>     dense-mode router, there is a malloc() for every transmission of the
|>     5,000,001 sources to create (S,G) state. This can't be good in any
|>     router both from a memory fragmentation point of view and a switching
|>     point of view.

I think it might make sense to, by default, automatically drop 
individual RTCP transmissions back to the source at a certain point.
[And, certainly, there should be an option to do so.  See below].  

OTOH, RTCP has proven useful [to me] in cases where there are on 
the order of 1000 listeners.  But, at a certain point, individual 
RTCP listener reports would seem to no longer make sense.  I'm not 
suggesting that no feedback to the source be given.  It would often 
be extremely useful to know how many people are watching.  But, long 
before one got to TV/radio-sized audiences- 10^^6 listeners- you 
would only want to know about how many people are watching, not who.
[Unless "you" are the secret police, or something [e.g. Advertisers].  
See Privacy below.]  

Given the example above, how many listeners does it take before 
the RTCP transmission interval drops to below the timeout threshold?  
Maybe that is ~about a logical place to switch.  Likewise, if the 
average time someone tunes in to an Internet Radio program is short 
relative to the average retransmission time, the statistics would 
no longer be valid anyway.  Maybe there is a better way to estimate 
the number of listeners when large-scale broadcasts are involved.  

|> >> Also, truly, RTCP data isn't always that interesting.  Do I really care
|> >> that much who is listening to the same radio broadcast?  If it's a
|> >> conference, yes.  If it's a broadcast, usually no.  

As above, for small broadcasts, yes, it is useful.  I don't think
one should take the TV analogy too far - for purely TV-like
functions, with on the order of 10-200 possible "channels",
you already have: FM radio, AM radio, broadcast TV, cable TV,
DirecTV [and other satellite TV companies], and so on.  Does
anyone think that "The Mbone" is going to be able to deliver
MPEG1/2/HDTV quality video into the home worldwide in direct 
competition to DirecTV and/etc.????  For little more than the
cost of connecting to an ISP with a 28.8 modem, today you can 
already get MPEG1/2 quality [ie, better than VCR, better than
most cable] satellite TV.  Even when the entire Internet is 
higher-bandwidth, multimedia, multicast enabled, I still don't 
see the Internet in direct competition with existing systems.

I see the "broadcast market" for mbone-like capability to be 
analogous to the Shuttle broadcasts- smaller, "niche" markets 
and communities too small and geographically diverse to get 
air time on a cable or satellite system with a few hundred 
channels.  In other words, the same as the internet today, 
but multicast/multimedia enabled.  The appeal of the internet 
has always been individual and "niche" communication, not 
"mass" communication, for which efficient channels have already 
long existed.  Sure, we need to make sure everything scales up
to a large number of internet users, maybe even 6 billion, but, 
I doubt if they want to be internet users so they can "watch TV".
People can already do that, *more cheaply*, just by watching TV.  

In fact, this diversity, from a routing point of view, 
represents one of the challenges: millions of daytime 
professional internet users now want their own personal 
class C home networks, and they want routes to similar 
people.  Now, add multicast.

Anyway, getting back to RTCP and "niche" broadcasting, assuming 
a larger number of smaller, niche sources:  If we are talking 
about 1000-10000 listeners to one of 10,000 different multicast 
broadcast sources, then, yes, RTCP would be very useful, including 
the information on how well "most people" are receiving the broadcast.

                                                      Do we really want to
|> >> burden precious router resources with (S,G) state that is of dubious
|> >> value?

As Steve Casner above, and Dino Farinacci below, point out, 
this concern is (now) invalid.  A shared tree will handle the
RTCP listener report flow, and the traffic will scale back
to a proportionately small amount.

I *am* still concerned about the total amount of (memory) 
state routers have to carry, but, not for *this* case.  
If 5 million of people are watching the same broadcast, 
so what if there are a few router bytes allocated for multicast?  
We just need to make sure that everything scales up for
this exceptional case, but, frankly, I doubt seriously
whether this will be a constant widespread problem any
time soon.  

[Of much larger concern is the amount of router memory consumed 
by 10^^5 DIS-like ~100-person multiuser games along the lines 
of what Intel described at their interactive multimedia/etc. 
presentation last summer.  If you want to worry about something,
worry about that.  I really don't think RTCP is much of a problem.]

In short, RTCP could be enhanced to scale to millions 
of people who are listening over asymmetric- bandwidth 
media [e.g. cable modems], and also for Privacy.  


----

[Dino Farinacci:]

|>    This is why SPT thresholds were created in PIM. If you set the SPT
|>    threshold in a cisco to say 8kbps, only the active "speakers" get an
|>    (S,G), everyone else uses the (*,G).


And, Steve Casner also replied:
["Stephen Casner" <casner@precept.com>:]

   [Arlie Davis:]
|> > RTCP puts a much greater strain than is necessary on (S,G)-state
|> > implementations.  This is rather unfortunate.
|> 
|> Both on the rem-conf list and in the last AVT meeting, we have had
|> significant discussion of the use of RTCP in large broadcast
|> scenarios.  I'll attempt to move this discussion back there [bcc to
|> mbone].
:

|> With a shared-tree routing protocol, there is only (*,G) state for the
|> receivers because the RTCP data rate is low enough not to trigger the
|> formation of a source-rooted tree.

|> > Also, truly, RTCP data isn't always that interesting.  Do I really care
|> > that much who is listening to the same radio broadcast?  If it's a
|> > conference, yes.  If it's a broadcast, usually no.
|> 
|> At the AVT meeting, we concluded that there should be additional RTP
|> profiles specifying alternative RTCP behavior.  It is likely that one
|> of those would specify that receivers would not send RTCP at all, for
|> use in scenarios where privacy of reception or other factors outweigh
|> the value of RTCP's reception feedback.
|> 
|> BTW, at the AVT meeting I invited the submission of proposals for
|> additional profiles, but I don't think any such proposals have been
|> submitted yet.  (Not to neglect Bernard Aboba's I-D on the issues and
|> the useful email on this topic.)

---

One of the things which several people at IETF requested,
after a similar discussion to this mailing list discussion,
[I guess nobody has responded with anything concrete] 
was some method to [reliably] estimate the number of 
listeners without actually getting individual reports.
Normally, it is a Bad Thing to ask Routers to do
something other than Route, so, this method likely 
should be something driven by the multicast source itself,
to which [some of] the receivers can [randomly] respond, 
if asked.

Also, if we are talking about millions of viewers and
commercially significant advertising dollars possibly based
on viewership, then, there probably needs to be some kind
of difficult-to-spoof authentication built in, too.  After all, 
what is there, today, to prevent a single source from falsifying
non-existent-listener reports?  So, we need a statistically
valid, authenticated, but private, low-upstream-bandwidth,
router-independent method for estimating viewership.  Ideas?


-Hugh LaMaster






From rem-conf-request@es.net Thu Feb 27 15:17:21 1997 
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From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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To: John H Wilson <John_H_Wilson@ccm.jf.intel.com>
CC: robla@prognet.com, rem-conf@es.net, confctrl@isi.edu
Subject: Re: RTSP "sessions", "streams", "components", and URIs
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Thanks for pointing this out - all the authors are aware that the
current terminology is both inconsistently used within and outside the
document. One proposal would be use the terms "track" and
"presentation", since stream has too many conflicting uses already.
(RTSP "lives in" both the SDP name space and the RTP name space, for
better and for worse.) Just using "session" doesn't help, since we do
need to name individual media streams (say, only the audio part of a
movie).

I cc'ed this to confctrl, I think that's where the discussion belongs.

John H Wilson wrote:
> 
> IMHO, RTSP could avoid some confusion over the term "session" by being
> self-consistent.
> 
> SDP defines the term in one sense: a collection of streams with a common
> time-line.
> 
> RTP defines the term in another sense: a single media stream (along with
> its parallel "control" stream (RTP/RTCP).
> 
> RTSP has absorbed both terms.
> 
> In Section 1.3, it is used in the SDP sense; in section 6.2, GET
> retrieves a session description, and in section 6.11, SESSION allows the
> server to send the client a modified session description with new or
> deleted "components" (it is not clear, but I think "component" is
> synonymous with "stream".)
> 
> In section 8.24, session is clearly refering to a single stream
> (consistent with RTP, but RTSP does not insist on its use). It defines
> the session header field which identifies the state of a single stream
> on which SETUP(6.3), PLAY(6.4), PAUSE(6.5), and CLOSE(6.6) operate.
> 
> IMHO, "stream" should be used for SETUP, PLAY, PAUSE, CLOSE, and should
> replace the term "component" in the definition of the SESSION request.
> The "session" header field would become the "stream" header field.
> 
> On a minor, related, point, I wonder why all requests must contain a
> Request-URI. PAUSE and CLOSE, for example, must operate on an existing
> "stream"; if the Request-URI does not match the "stream" ID, I suppose
> it is a protocol error? What would the Request-URI be used for in these
> two requests?

You can PAUSE and CLOSE an individual track (say, the audio track)
without PAUSEing or CLOSEing the whole presentation (audio and video,
say) or you can indeed PAUSE the whole show. The session id is the same
in both cases, the URI is not. "Invalid session id" is, I think, one of
the possible error codes. (Whether Session should be a concept within
RTSP or whether RTSP should borrow the HTTP state maintenance mechanisms
for its purposes is another debate...)


> 
> john_h_wilson@ccm.jf.intel.com


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

From rem-conf-request@es.net Thu Feb 27 15:27:46 1997 
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From: Petri Helenius <pete@sms.fi>
To: lamaster@george.arc.nasa.gov
Cc: mbone@isi.edu, rem-conf@es.net
Subject: RE: Progressing IDMR items
In-Reply-To: <199702271909.LAA13865@george.arc.nasa.gov>
References: <199702271909.LAA13865@george.arc.nasa.gov>

lamaster@george.arc.nasa.gov writes:

 > DirecTV [and other satellite TV companies], and so on.  Does
 > anyone think that "The Mbone" is going to be able to deliver
 > MPEG1/2/HDTV quality video into the home worldwide in direct 
 > competition to DirecTV and/etc.????  For little more than the

I'm in for this one. Many companies in US and abroad have announced
*DSL offerings in the sub-$100/month range. That's in the ballpark
what you pay for your cable, phone and 28.8 ISP connection.
(but does or is going to do you much more than that)

 > cost of connecting to an ISP with a 28.8 modem, today you can 
 > already get MPEG1/2 quality [ie, better than VCR, better than
 > most cable] satellite TV.  Even when the entire Internet is 
 > higher-bandwidth, multimedia, multicast enabled, I still don't 
 > see the Internet in direct competition with existing systems.
 >
I don't want to talk about competing either, I would call it
replacement. After you have the bits flowing you don't need to have a
wire per purpose. You can have it all from the same wire. 

 > I see the "broadcast market" for mbone-like capability to be 
 > analogous to the Shuttle broadcasts- smaller, "niche" markets 
 > and communities too small and geographically diverse to get 
 > air time on a cable or satellite system with a few hundred 
 > channels.  In other words, the same as the internet today, 
 > but multicast/multimedia enabled.  The appeal of the internet
 > has always been individual and "niche" communication, not 
 > "mass" communication, for which efficient channels have already 
 > long existed.  Sure, we need to make sure everything scales up

The issue here that the incremental cost of doing both is extremely
small. (if going from the "small" market content to mass market, the
other direction is more expensive)

 > to a large number of internet users, maybe even 6 billion, but, 
 > I doubt if they want to be internet users so they can "watch TV".
 > People can already do that, *more cheaply*, just by watching TV.  
 >
I disagree with the cost argument here. With the digital television
technology, all is going to be bits anyway, so I don't think consumers
are going to really care what delivers their bits. As long as they get
the bits. Also there is huge forces behind on the integration on
computing devices and larger screens (read: PC and TV) to provide
multimedia content to the couch potatoes.

 > In fact, this diversity, from a routing point of view, 
 > represents one of the challenges: millions of daytime 
 > professional internet users now want their own personal 
 > class C home networks, and they want routes to similar 
 > people.  Now, add multicast.
 >
Remember CIDR? An average home does not need eight bits address space (yet).
I used to have a C-class network at home but swapped it to 4 bit
subnet some while back (just to be a good citizen).

 > Anyway, getting back to RTCP and "niche" broadcasting, assuming 
 > a larger number of smaller, niche sources:  If we are talking 
 > about 1000-10000 listeners to one of 10,000 different multicast 
 > broadcast sources, then, yes, RTCP would be very useful, including 
 > the information on how well "most people" are receiving the broadcast.
 >
You can build your apps to send the RTCP messages to the source, not
to the multicast group. This way the RTCP "problem" goes away.
Regardless of the routing protocol. I for myself wouldn't care of my
neighbor knowing what I watch on my "internet television" nor I would
like you to know.

 > [Of much larger concern is the amount of router memory consumed 
 > by 10^^5 DIS-like ~100-person multiuser games along the lines 
 > of what Intel described at their interactive multimedia/etc. 
 > presentation last summer.  If you want to worry about something,
 > worry about that.  I really don't think RTCP is much of a problem.]

Exactly :-) In a game everybody is a source. That's going to be the
greater challenge.
 > 
 > In short, RTCP could be enhanced to scale to millions 
 > of people who are listening over asymmetric- bandwidth 
 > media [e.g. cable modems], and also for Privacy.  
 > 
Glad that you mentioned privacy as an issue too.
 > 
 > Also, if we are talking about millions of viewers and
 > commercially significant advertising dollars possibly based
 > on viewership, then, there probably needs to be some kind
 > of difficult-to-spoof authentication built in, too.  After all, 
 > what is there, today, to prevent a single source from falsifying
 > non-existent-listener reports?  So, we need a statistically
 > valid, authenticated, but private, low-upstream-bandwidth,
 > router-independent method for estimating viewership.  Ideas?
 > 
I think the keyword here is "estimating". The figures need to be in
the ballpark but they do not need to be exact. However, no clever
technical ideas come into mind right now :-(

Pete

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From: Arlie Davis <arlie@thepoint.net>
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Cc: "'mbone@isi.edu'" <mbone@isi.edu>, "'rem-conf@es.net'" <rem-conf@es.net>
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Simply tossing RTCP packets toward the source is not a good idea at all.
 The source may be a machine with minimal intelligence, whose sole
responsibility is to read digital video and multicast it, for example,
or a stream server who could care less about membership.

Also, there would be a huge flood of RTCP packets toward the source.

So, there are two issues that I have with this: The source may not be
interested, and the flood of packets may be too great.  There are two
relatively easy ways to deal with this.

Let "session watcher" mean a machine which is interested in the RTCP
reports for a given multicast source or session, and "session source"
mean the machine which is actually sourcing the data, such as video or
audio.

The session source could periodically multicast a list of preferred
session watchers.  (How this information is conveyed is irrelevant -- it
could be in-band RTP data with a specific, reserved payload type, or any
number of ways.  The only important thing is that the data is multicast
to the exact same set of machines who are receiving the data multicast.)
 Each RTCP-aware client would receive the list of session watchers and
randomly decide on which one to use.  (If the list of session watchers
changes (watchers arrive, go away), the clients just re-decide.)  Then,
the client unicasts it's RTCP data to the session watcher.

This would allow for very simple and very complex implementations.  The
simplest implementations would announce a fixed list that just points
toward the session source.  The more complex ones would manage a
distributed database of all of the session listeners, their
quality-of-service statistics, etc.

Also, in the same place that specifies the list of session watchers, the
session source could simply indicate that clients are to send their RTCP
the "old fashioned way" -- multicast it to the same group as the session
data.  This would be a good choice for small- and medium-sized group
conferences.  The above model makes sense mostly for traditional
broadcast-style multicasts (few sources, many many listeners).

As we all know, the higher the group membership, the lower the rate at
which the members announce themselves.  As we approach Large numbers of
listeners, the rate essentially approaches zero, which makes RTCP pretty
much useless.  In a large session, usually there are only a few people
truly interested in who is listening.  Why not have the session
watcher(s) collect this information, index it, and (for example) make it
available via a web query.

-- arlie

>-----Original Message-----
>From:	lamaster@george.arc.nasa.gov [SMTP:lamaster@george.arc.nasa.gov]
>Sent:	Thursday, February 27, 1997 2:10 PM
>To:	mbone@isi.edu
>Cc:	rem-conf@es.net
>Subject:	RE: Progressing IDMR items
>
>I think it might make sense to, by default, automatically drop 
>individual RTCP transmissions back to the source at a certain point.
>[And, certainly, there should be an option to do so.  See below].  
>
>OTOH, RTCP has proven useful [to me] in cases where there are on 
>the order of 1000 listeners.  But, at a certain point, individual 
>RTCP listener reports would seem to no longer make sense.  I'm not 
>suggesting that no feedback to the source be given.  It would often 
>be extremely useful to know how many people are watching.  But, long 
>before one got to TV/radio-sized audiences- 10^^6 listeners- you 
>would only want to know about how many people are watching, not who.
>[Unless "you" are the secret police, or something [e.g. Advertisers].  
>See Privacy below.]  
>
>Given the example above, how many listeners does it take before 
>the RTCP transmission interval drops to below the timeout threshold?  
>Maybe that is ~about a logical place to switch.  Likewise, if the 
>average time someone tunes in to an Internet Radio program is short 
>relative to the average retransmission time, the statistics would 
>no longer be valid anyway.  Maybe there is a better way to estimate 
>the number of listeners when large-scale broadcasts are involved.  
>
>|> >> Also, truly, RTCP data isn't always that interesting.  Do I really care
>|> >> that much who is listening to the same radio broadcast?  If it's a
>|> >> conference, yes.  If it's a broadcast, usually no.  
>
>As above, for small broadcasts, yes, it is useful.  I don't think
>one should take the TV analogy too far - for purely TV-like
>functions, with on the order of 10-200 possible "channels",
>you already have: FM radio, AM radio, broadcast TV, cable TV,
>DirecTV [and other satellite TV companies], and so on.  Does
>anyone think that "The Mbone" is going to be able to deliver
>MPEG1/2/HDTV quality video into the home worldwide in direct 
>competition to DirecTV and/etc.????  For little more than the
>cost of connecting to an ISP with a 28.8 modem, today you can 
>already get MPEG1/2 quality [ie, better than VCR, better than
>most cable] satellite TV.  Even when the entire Internet is 
>higher-bandwidth, multimedia, multicast enabled, I still don't 
>see the Internet in direct competition with existing systems.
>
>I see the "broadcast market" for mbone-like capability to be 
>analogous to the Shuttle broadcasts- smaller, "niche" markets 
>and communities too small and geographically diverse to get 
>air time on a cable or satellite system with a few hundred 
>channels.  In other words, the same as the internet today, 
>but multicast/multimedia enabled.  The appeal of the internet 
>has always been individual and "niche" communication, not 
>"mass" communication, for which efficient channels have already 
>long existed.  Sure, we need to make sure everything scales up
>to a large number of internet users, maybe even 6 billion, but, 
>I doubt if they want to be internet users so they can "watch TV".
>People can already do that, *more cheaply*, just by watching TV.  
>
>In fact, this diversity, from a routing point of view, 
>represents one of the challenges: millions of daytime 
>professional internet users now want their own personal 
>class C home networks, and they want routes to similar 
>people.  Now, add multicast.
>
>Anyway, getting back to RTCP and "niche" broadcasting, assuming 
>a larger number of smaller, niche sources:  If we are talking 
>about 1000-10000 listeners to one of 10,000 different multicast 
>broadcast sources, then, yes, RTCP would be very useful, including 
>the information on how well "most people" are receiving the broadcast.
>
>                                                      Do we really want to
>|> >> burden precious router resources with (S,G) state that is of dubious
>|> >> value?
>
>As Steve Casner above, and Dino Farinacci below, point out, 
>this concern is (now) invalid.  A shared tree will handle the
>RTCP listener report flow, and the traffic will scale back
>to a proportionately small amount.
>
>I *am* still concerned about the total amount of (memory) 
>state routers have to carry, but, not for *this* case.  
>If 5 million of people are watching the same broadcast, 
>so what if there are a few router bytes allocated for multicast?  
>We just need to make sure that everything scales up for
>this exceptional case, but, frankly, I doubt seriously
>whether this will be a constant widespread problem any
>time soon.  
>
>[Of much larger concern is the amount of router memory consumed 
>by 10^^5 DIS-like ~100-person multiuser games along the lines 
>of what Intel described at their interactive multimedia/etc. 
>presentation last summer.  If you want to worry about something,
>worry about that.  I really don't think RTCP is much of a problem.]
>
>In short, RTCP could be enhanced to scale to millions 
>of people who are listening over asymmetric- bandwidth 
>media [e.g. cable modems], and also for Privacy.  
>
>|> At the AVT meeting, we concluded that there should be additional RTP
>|> profiles specifying alternative RTCP behavior.  It is likely that one
>|> of those would specify that receivers would not send RTCP at all, for
>|> use in scenarios where privacy of reception or other factors outweigh
>|> the value of RTCP's reception feedback.
>

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From: John H Wilson <John_H_Wilson@ccm.jf.intel.com>
Message-ID: <Thu, 27 Feb 97 17:25:44 PST_4@ccm.jf.intel.com>
To: schulzrinne <schulzrinne%cs.columbia.edu_at_internet_gateway@ccm.jf.intel.com>, 
    rem-conf-request@es.net
cc: robla@prognet.com, rem-conf@es.net, confctrl@isi.edu
Subject: Re[2]: RTSP "sessions", "streams", "components", and URIs


Text item: 

OK, suppose we use "Presentation" instead of "Session"
               and "Track"        instead of "Stream"

I'm still a bit puzzled about the model:

You wrote:

>You can PAUSE and CLOSE an individual track (say, the audio track)
>without PAUSEing or CLOSEing the whole presentation (audio and video,
>say) or you can indeed PAUSE the whole show. The session id is the same
>in both cases, the URI is not. "Invalid session id" is, I think, one of
>the possible error codes. (Whether Session should be a concept within
>RTSP or whether RTSP should borrow the HTTP state maintenance mechanisms
>for its purposes is another debate...)

My questions are:

1. I assume SETUP is always track-specific; whether it is the intial 
   SETUP or not.

2. I assume that a PLAY that does not specify a track id, must 
   specifiy a track URI, and gets a track id in return. (I.e., 
   cannot operate on a presentation basis) True?

3. Is there such a thing as a presentation id? I assume not, since 
   I see no request that generates one.

4. If (as you say) the "session id" is the same to PAUSE/CLOSE either a         
   presentation or a track, and the difference is in the URI, then  to          
   PAUSE/CLOSE a presentation, I must be able to use ANY track id in the        
   presentation + the presentation URI....true? If this is the case, I 
   assume that, once a presentation has been PAUSEd, it can be resumed 
   by sending a PLAY with ANY track id + the presentation URI?

5. I assume that track id's are not client unique. If there are multiple        
   listeners to a track (it is being multicast), these clients can provide some 
   way to pass the track id around, so that any of them can use the same (track 
   id + URI) to PAUSE/PLAY/CLOSE the track. This is the so-called "pass the     
   remote" scenario, except that RTSP provides no mechanism to prevent "multiple
   remotes"?

[So far, I've concluded that SETUP is track-specific, while PLAY, PAUSE & CLOSE 
are not.]

Two presentation description questions:

  - SDP does not seem provide the necessary track-specific URI information; 
    do you intend to propose the necessary extensions?
  - where can I find a definition of SDF?

john_h_wilson@ccm.jf.intel.com






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Date: Fri, 28 Feb 1997 10:07:05 +0200 (EET)
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From: Petri Helenius <pete@sms.fi>
To: Arlie Davis <arlie@thepoint.net>
Cc: "'lamaster@george.arc.nasa.gov'" <lamaster@george.arc.nasa.gov>, 
    "'mbone@isi.edu'" <mbone@isi.edu>, "'rem-conf@es.net'" <rem-conf@es.net>
Subject: RE: Progressing IDMR items
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Arlie Davis writes:
 > Simply tossing RTCP packets toward the source is not a good idea at all.
 >  The source may be a machine with minimal intelligence, whose sole
 > responsibility is to read digital video and multicast it, for example,
 > or a stream server who could care less about membership.
 > 
 > Also, there would be a huge flood of RTCP packets toward the source.

I should have been more clear here. I'm not suggesting that this
should be done without indication from the source to do so. The source
should tell the receivers either send the RTCP packets to the group,
send them to the source or not send them at all.
 > 
 > The session source could periodically multicast a list of preferred
 > session watchers.  (How this information is conveyed is irrelevant -- it
 > could be in-band RTP data with a specific, reserved payload type, or any
 > number of ways.  The only important thing is that the data is multicast
 > to the exact same set of machines who are receiving the data multicast.)
 >  Each RTCP-aware client would receive the list of session watchers and
 > randomly decide on which one to use.  (If the list of session watchers
 > changes (watchers arrive, go away), the clients just re-decide.)  Then,
 > the client unicasts it's RTCP data to the session watcher.

This is a good optimization of what I propose above.
 > 
Pete

From rem-conf-request@es.net Fri Feb 28 03:16:33 1997 
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Date: Fri, 28 Feb 1997 10:15:49 +0200 (EET)
From: Seppo Kallio <kallio@cc.jyu.fi>
To: rem-conf@es.net
Subject: Vic binaries for Linux RedHat 4.1+b/w QuickCam + MatroxMeteor
In-Reply-To: <199702271448.JAA13106@flora.cc.gatech.edu>
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Is there binaries or easy to port sources for vic using b/w qcam and vic
using Matrox Meteor?

I tryed to compile vic-2.8, but have some problems with it.

Need some fast grabber for Meteor olso, is metgrab ported to Linux? I
would like to do some mpegs.

Seppo



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Date: Fri, 28 Feb 1997 09:17:38 -0500 (EST)
From: kevin@cc.gatech.edu (Kevin C. Almeroth)
Message-Id: <199702281417.JAA16163@flora.cc.gatech.edu>
To: rem-conf@es.net
Subject: RE: Progressing IDMR items

[Pruned MBone list, just 'case I hate sending to both]

>>From: lamaster@george.arc.nasa.gov

>>most cable] satellite TV.  Even when the entire Internet is 
>>higher-bandwidth, multimedia, multicast enabled, I still don't 
>>see the Internet in direct competition with existing systems.

I do!  :-)  But let's not meander on predictions.

>>As Steve Casner above, and Dino Farinacci below, point out, 
>>this concern is (now) invalid.  A shared tree will handle the
>>RTCP listener report flow, and the traffic will scale back
>>to a proportionately small amount.

Pulling more statistics from my mlisten stuff, I've noted that
the ratio of RTCP traffic to RTP traffic is typically pretty
well behaved.  For all the audio sessions the RTCP traffic lately
has been about 6-8 Kbps.  While this number is dependent on the
total number of receivers (lately it has been about averaging about 
75 because not much is going on) there is another factor.  And that 
is the number of groups.  As the number of groups grows the RTCP 
traffic will increase (scaling only works within a group).  Something 
to consider.  Oh, and the video traffic has been behaving similarly.

>>["Stephen Casner" <casner@precept.com>:]
>>
>>|> At the AVT meeting, we concluded that there should be additional RTP
>>|> profiles specifying alternative RTCP behavior.  It is likely that one
>>|> of those would specify that receivers would not send RTCP at all, for
>>|> use in scenarios where privacy of reception or other factors outweigh
>>|> the value of RTCP's reception feedback.

Sorry I missed this discussion.  A careful analysis of existing sessions
would yield some good insight into the number and types of profiles needed.

>>Normally, it is a Bad Thing to ask Routers to do
>>something other than Route, so, this method likely 
>>should be something driven by the multicast source itself,
>>to which [some of] the receivers can [randomly] respond, 
>>if asked.

Probabilistic feedback has been shown to be quite accurate.


-Kevin Almeroth

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Date: Fri, 28 Feb 1997 13:26:19 -0800 ()
From: Stephen Casner <casner@precept.com>
To: rem-conf@es.net
Subject: RE: Progressing IDMR items
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I'm replying to several messages here that present a range of ideas.
Most of these ideas were discussed in the last AVT meeting, minutes of
which were sent to this list and are accessible from www.ietf.org.
There you can see a list of the solutions that have been proposed and
some potential problems that can arise.


"Dino Farinacci" <dino@cisco.com>:
> |>     Therefore, in a
> |>     dense-mode router, there is a malloc() for every transmission of the
> |>     5,000,001 sources to create (S,G) state.

Right, so using a shared tree to eliminate this cost is a big win.
Then the (S,G) state timeout is not an issue.


lamaster@george.arc.nasa.gov writes:
> I think it might make sense to, by default, automatically drop 
> individual RTCP transmissions back to the source at a certain point.

I see trouble in making an automatic transision.  Since the RTCP
packets from other participants are the means to determine how many
participants there are, when the threshold is crossed and some stop
sending RTCP, then the count will drop below the threshold again and
this will oscillate.  Instead, I believe the different RTCP modes
should be selected by the choice of a particular RTP profile specified
as a session parameter.

> One of the things which several people at IETF requested,
> after a similar discussion to this mailing list discussion,
> [I guess nobody has responded with anything concrete] 
> was some method to [reliably] estimate the number of 
> listeners without actually getting individual reports.

I think the question was slightly different: how to estimate the
number of receivers without having to store the SSRC id's from all the
RTCP packets you've heard.  This can be done with a sampling
technique, which I think was described in messages to the list.

> Normally, it is a Bad Thing to ask Routers to do
> something other than Route, so, this method likely 
> should be something driven by the multicast source itself,
> to which [some of] the receivers can [randomly] respond, 
> if asked.

The scalability of IP multicast (in number of receivers) derives
directly from the fact that no entity needs to know how many receivers
there are.  So, the routers could not really answer the question
anyway.


On Thu, 27 Feb 1997, Petri Helenius wrote:
> You can build your apps to send the RTCP messages to the source, not
> to the multicast group. This way the RTCP "problem" goes away.

The problem is to avoid an implosion at the sender.  We discussed ways
of letting the sender control the responses by sampling the receivers,
including work by Wakeman, Bolot and Turletti a couple of years ago.


On Thu, 27 Feb 1997, Arlie Davis wrote:
> The session source could periodically multicast a list of preferred
> session watchers.  (How this information is conveyed is irrelevant -- it
> could be in-band RTP data with a specific, reserved payload type, or any
> number of ways.  The only important thing is that the data is multicast
> to the exact same set of machines who are receiving the data multicast.)
>  Each RTCP-aware client would receive the list of session watchers and
> randomly decide on which one to use.  (If the list of session watchers
> changes (watchers arrive, go away), the clients just re-decide.)  Then,
> the client unicasts it's RTCP data to the session watcher.

Indeed, it may be desirable to have the reception reporting go to one
or more third-party monitors ("session watchers") rather than the
sender.  However, you still need to control the rate at which clients
send, because they won't be hearing the multicast RTCP packets to
adapt for themselves.  So, presumably the sender or some other entity
needs to tell them how often to send as well as where to send.  As
Henning Schulzrinne pointed out, this provides an easy way for some
malicious entity to instruct all the receivers to packet bomb some
other innocent party.

> As we all know, the higher the group membership, the lower the rate at
> which the members announce themselves.  As we approach Large numbers of
> listeners, the rate essentially approaches zero, which makes RTCP pretty
> much useless.

The long interval may prevent collecting a complete list of who is
listening, but you still get feedback about loss rate that, in the
aggregate over all receivers, comes in quite frequently.


On Fri, 28 Feb 1997, Petri Helenius wrote:
>  > Also, there would be a huge flood of RTCP packets toward the source.
> 
> I should have been more clear here. I'm not suggesting that this
> should be done without indication from the source to do so. The source
> should tell the receivers either send the RTCP packets to the group,
> send them to the source or not send them at all.

Again, it is essential to control the rate as well as the
destination.  Also, if there is more than one source, having the
source specify the mode can be problematic.  Also, unless the default
is to send nothing at all, the source could get swamped before it had
a chance to tell the receivers what to do.  But this default is not
good for small teleconferences.  For these reasons, I believe this
should be a session-specific parameter.


On Fri, 28 Feb 1997, Kevin C. Almeroth wrote:
> As the number of groups grows the RTCP 
> traffic will increase (scaling only works within a group).  Something 
> to consider.

Sure, but so will the data traffic.  We assume that the network
capacity will increase over time to handle the demand.  The current
situation of having several channels that are up all the time but idle
most of the time is artificial, I think.

> Sorry I missed this discussion.  A careful analysis of existing sessions
> would yield some good insight into the number and types of profiles needed.

That would be welcome.
							-- Steve


From rem-conf-request@es.net Fri Feb 28 16:50:47 1997 
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From: Chris Marshall <clmarshall@anl.gov>
Subject: Mbone Broadcast of Seminar
Cc: mike_petrick@qmgate.anl.gov (Michael Petrick/ANL), 
    bertolac@che.udel.edu (Ralph Bertolacini/Amoco/retired), mmarino@anl.gov
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On Thursday, March 6, 1997 the Argonne Petroleum Seminar Series will
broadcast the following seminar live.  The broadcast will begin at 10:30 am
CST.  Visit the Argonne Petroleum Seminar Series web site at
(http://www.anl.gov/Petroleum) for further information.

		REFINERY ENERGY EFFICIENCY ECONOMICS
				(ABSTRACT BELOW)
                                                    Jerry L. Robertson
                (Retired Engineering Advisor Exxon Research and Engineering
Co.)
                                                      Sr. Consultant
____________________________________________

As a preliminary test to this live broadcast we will be broadcasting two of
the 1996 talks from 10 am to 2 pm on March 4 & 5, 1997.  These talks are:

	"Catalysis in the Petroleum Industry Research Strategies for the Late
Nineties and Beyond"
                                                 Paul B. Venuto
                                           Retired Manager of Catalysis
                                                 Mobil R & D
                                             Central Research Division

		"Trends in Fluid Cracking Catalyst Technology"
                                              Carmo J. Pereira
                                          W. R. Grace & Company
                                             
________________________________________________

REFINERY ENERGY EFFICIENCY ECONOMICS

ABSTRACT

Petroleum Refineries are large complex systems. In the base case typical
scenerio a few crude feeds produce many products, through a set of
processes. The product mix and specifications change with season.
Furthermore, since most of the products are ultimately burned, many of the
constituent product molecules are interchangeable among products-- for
example jet fuel and gasoline. Individual conversion, desulfurization,
separation and other processes are connected either directly or through
intermediate tankage which means that changes in refinery feeds or
intermediate processing conditions are reflected in changes in the product
streams. Design techniques have required the partitioning of the effort so
that, for example, the crude pipe still, cat cracker, catalytic reformer,
cat light ends, cat deethanizer absorber are individually designed with a
specific window of feed and product throughput and specifications. On top
of the process requirements, a set of energy systems--the utility
system--includes: steam at several pressure levels, electric power, cooling
water and a fuel gas system provides additional process interconnectivity.
Most refinery systems have been modified many times by adding new processes
or revamping existing ones for bottleneck removal or new operating
conditions. This typical scenario provides a refinery that consists of many
reactors, heat exchangers, distillation towers, drums, pumps compressors,
etc. connected together with various sizes of pipes and controlled with an
array of control valves. 

Because of the interactions, large scale of the problem and difficulty in
making meaningful cost studies, many energy conservation studies and
projects have resulted in less than acceptable results. Steam has been
saved on the basis of its marginal cost to one process only to have the
steam vent open in another part of the refinery when the project was
installed. Similarly heat pumps have been installed to reboil distillation
towers saving low pressure steam that could have easily been generated by
waste heat in an adjacent process. Systems have been designed with too much
heat exchange in one process area and too little in others. Major studies
have been done (and even books written) to save availability. These
sometimes, but not always, resulting in economic energy efficiency. 

This seminar will outline an economics driven method for providing a
consistent approach to refinery wide energy efficiency. The method is
based on the refinery wide compositing of the heat requirements for all of
the processes (from feed to reactor inlet). Similarly, the heat
availability from all the hot reactor outlets on the way to other processes
or product tankage are also composited. This pinch analysis approach is
expanded by calculation of the marginal value of energy saved (based on its
actual cost) versus the marginal cost of effective heat exchange area (or
additional heat transfer coefficient). The technique is then expanded to
determine the value of marginal energy saved versus temperature. Use of
this value provides a consistent goal for energy saving projects across the
entire refinery and assures consistent approach temperatures for the design
of various energy recovery exchangers. It also allows justification of heat
transfer coefficient enhancers such as turbulence promoters or anti
foulents on an economically consistent basis. 

While the method might appear to be useful only to design, it is also
useful to both daily and longer term operations of refineries. It can be
used to relate the value of marginal internal product (such as vacuum gas
oil or catalytic reformer feed) to the cost of energy for additional
recovery.  Finally it can be used to adjust operational stewardship goals
such as allowable energy consumption for a specific process unit to current
economics and operating parameters. 
===============================================
Christopher L. Marshall, Research Scientist          
Chemistry Division/Coal Chemistry Group
Argonne National Laboratory, CHM/200
9700 South Cass Avenue
Argonne, IL 60439-4831
===============================================
E-mail: CLMARSHALL@ANL.GOV
Phone: (630)252-4310
Fax: (630)252-9288      
=============================================== 
 Current Issues in Petroleum Processing Seminar Series at Argonne
< http://www.anl.gov/Petroleum >
 15th North American Catalysis Society Meeting, Chicago, 1997
< http://www.anl.gov/NAM >
===============================================

From rem-conf-request@es.net Fri Feb 28 22:07:33 1997 
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From: Bernard Aboba <aboba@internaut.com>
To: "rem-conf@es.net" <rem-conf@es.net>
Subject: Profiles
Date: Fri, 28 Feb 1997 19:03:39 -0800
Encoding: 15 TEXT

OK, I'll bite. The profile I would like to see would be:

A profile that brings the bandwidth fraction down below 5 percent. 
To prevent this from being used for abusive purposes, it could be
interpretted as a fraction of 5 percent. That way, you could specify 
a bandwidth fraction of less than 5 percent, but never more. 

Setting the fraction to zero would be equivalent to turning off
RTCP entirely. 

While we talked about a number of other possibilities at the AVT
meeting, in thinking more about them, I'm either convinced that
they don't help (putting RTCP on a separate group), or could
be abused too easily (sending RTCP to a unicast address). 



