From rem-conf-request@es.net Sat Mar 01 04:33:06 1997 
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From: Amitabh <amitabh@fcpl.co.in>
To: rem-conf <rem-conf@es.net>
Subject: Doubts regarding Multicasting
Date: Sat, 1 Mar 1997 15:04:00 +0530
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Message-ID: <19970301090641102.AAA148@Forest_Gump.fcpl.co.in>

Hi,
I am Amitabh Mathrawala and am working on an experimental winsock (1.1 for 
now) compliant tcp/ip stack. Sorry to 
bother you with these doubts.  Since I could'nt get any relevant
information in this regard on the  net, I am writing to you. 
I dont wish to take your time so in case I have missed out any obvious 
documentation, please let me know or any pointers 
to where I can get  the relevant information. If you are aware of any news
groups on this  subject, it would be very helpful. I 
was in look out of any  documentation that is authoritative on the subject
(something like  RFCs) because people have been 
suggesting me some common sense based  solutions.
Let me start with my doubts.

Doubts regarding multi-casting ---------------------------------
1. Assume a single homed host. A datagram kind of socket S is opened. It
joins a multicast group on the only inteface. Now 
is it necessary for the socket to specify INADDR_ANY (and port P) as the
binding addresses while binding in order to 
receive multicast datagrams with target port P ? Or the socket can still
receive multicast datagrams even if it is bound to the 
specific interface address (which is the only one available) and port P ?
2. Assume a multihomed host. A socket S joins a multicast group on
interface I1. What address should it specify while 
binding in order to receive the multicast datagrams (of the joined group)
coming on that interface and not any other 
interfaces?
3. Another similar situation. Lets say a machine has two interfaces I1 and
I2. Open two sockets S1 and S2 on the machine. 
Let them join the same multicast group, say M, on the interfaces I1 and I2
respectively. (Note that such a thing is allowed). 
Let them be bound to addresses <I1, P> and <I2, P>.  Now should the
multicast datagrams of group M coming on 
interface I1 be forwarded/delivered to socket S2 ? 

 Binding of datagram sockets --------------------------------------- 
1. In a multihomed host, a datagram socket has been bound to    INADDR_ANY
address. What IP datagrams are 
delivered to such     socket ? Meaning datagrams targeted to any of the
interfaces or some    specific addr . This is a 
redundant question if a datagram does get     bound immediately after the
bind call.

 With Regards.
~~~~~~~~~~~~~~ 
Amitabh Mathrawala 
Frontier Computers Pvt Ltd Pune, India 
Ph : (91)(212) 328640/321536 
email : amitabh@fcpl.co.in 
~~~~~~~~~~~~~~ 


If u have chosen to be a streetsweeper then sweep
the streets as beethoven composed music or as
michaelangelo painted . Sweep the streets so well
that all the hosts of heaven and earth will stop by to
say " Here lived a great street sweeper that did his job well".


From rem-conf-request@es.net Sun Mar 02 18:28:20 1997 
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To: "Henning Schulzrinne (BL)" <hgs@cs.columbia.edu>
Cc: rem-conf@es.net, confctrl@isi.edu
Subject: Re: New version of RTSP draft
Reply-to: sumisu@nttlabs.com (Jeffrey D. Smith)
In-reply-to: Your message of "Fri, 21 Feb 1997 17:58:47 EST"
References: <330E28A7.1FC6@cs.columbia.edu>
Date: Sun, 02 Mar 1997 15:27:39 -0800
From: Jeff Smith <sumisu@nttlabs.com>

A few meta-level comments and questions before I dive in deeper...

Regarding recording facilities: 
- who decides the name to be used for the stored stream in a record
request?  as written it appears that it is determined by the
requesting party.  Similar to HTTP PUT and POST it would seem
necessary to have the ability for the server or the client to specify
the abs_path or the URL.
- SETUP seams to be for "setup stream" so I'm not sure this would be
the proper place, but it is the request required before RECORD

PAUSE:
- can PAUSE indicate an SMPTE or Absolute Time stamp?

Along the same lines:
- a PLAY range will result in which? the server going to PAUSE at end
of range or CLOSE at the end of range?

js

--
 Jeffrey D. Smith
 Nippon Telegraph and Telephone Corporation   
 Multimedia Communications Laboratories
 250 Cambridge Ave., Suite 205			TEL  +1 415 833 3605
 Palo Alto, CA 94306				FAX  +1 415 326 1878
					 	ISDN +1 415 843 0667

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 old-pgp-fingerprint: C1 EE A9 BD B1 E9 2E 9A  03 CF 6B E1 CF C4 D0 0D
 e-mail: sumisu@nttlabs.com

From rem-conf-request@es.net Sun Mar 02 22:27:36 1997 
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unsubscribe


From rem-conf-request@es.net Mon Mar 03 09:20:55 1997 
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To: rem-conf@es.net
Subject: MBone camera for Windows 95/NT
From: wchang@nist.gov (Wo Chang, x3439)

I'm looking for a low cost, auto-focus, office use, 
monitor-/desk-top mounting with wide-angle video camera 
and a frame grabber (some might use parallel port instead,
such as QuickCam) for running MBone vic video under 
Windows 95/NT.  Any good suggestions or comments from
your hands-on experiences?

Thanks in advanced for any pointers.

--Wo Chang <wchang@nist.gov>

From rem-conf-request@es.net Mon Mar 03 11:15:50 1997 
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Subject: MBone broadcast of the CERN ATLAS meeting
To: rem-conf@es.net, hepnet-l@slacvm.slac.stanford.edu, hrc@frcpn11.in2p3.fr, 
    htasc@listbox.cern.ch, net-teleconferencing@listbox.cern.ch
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	    CERN is pleased to announce the MBONE broadcast of the

	   ATLAS Physics working group meeting and Plenary meetings
	   --------------------------------------------------------

		    on Thursday/Friday 6/7 March 1997
			from the CERN Auditorium

Provisional Agenda:
-------------------

>>> Times are UTC+1 <<<


 Thursday 6 March 1997
 ---------------------

    09h00 to 13h00    Physics working group

    14h00 to 18h00    Plenary meeting I


 Friday 7 March 1997
 -------------------

    09h00 to 13h00    Plenary meeting II


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

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

This message is sent to distribution lists, sorry if you receive multiple
copies of it.

Best regards,
--------------------------------------------------------------------
Martin Fluckiger & Christian Isnard                  CERN - CN/CS/EN
multicast@noc.cern.ch       European Laboratory for Particle Physics
Computers and Networks division      CH-1211 Geneva 23 - Switzerland


From rem-conf-request@es.net Mon Mar 03 12:24:34 1997 
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	MBone Broadcast Announcement
	----------------------------

Title:       
	Cliff Taubes "Four Dimensional Topology I"
Date:        
	Mar 04, 1997

Time:        
	12:00 PST8PDT 1 hours

Contact:     
	m1@msri.org

URL:         
	http://www.msri.org/events/taubes/

Description:        
	An introduction to 4-dimensional topology and the  Seiberg-Witten equations. Part I.









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

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

Title:       
	Cliff Taubes "Four Dimensional Topology II"
Date:        
	Mar 05, 1997

Time:        
	12:00 PST8PDT 1 hours

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	m1@msri.org

URL:         
	http://www.msri.org/events/taubes/

Description:        
	An introduction to 4-dimensional topology and the  Seiberg-Witten equations. Part II.









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

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	MBone Broadcast Announcement
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	Cliff Taubes "Four Dimensional Topology III"
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	Mar 06, 1997

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	12:00 PST8PDT 1 hours

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	m1@msri.org

URL:         
	http://www.msri.org/events/taubes/

Description:        
	An introduction to 4-dimensional topology and the  Seiberg-Witten equations. Part III.









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

From rem-conf-request@es.net Mon Mar 03 12:28:39 1997 
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	MBone Broadcast Announcement
	----------------------------

Title:       
	John Conway "Knotation"
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	Mar 10, 1997

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	16:00 PST8PDT 1 hours

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	joe@msri.org

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

Description:        
	How can we choose names for simple knots so that we can always rebuild a knot from its name, and yet so that  it will usually be easy to find the name from any form of the knot? I shall describe a simple arithmetical notation  that accomplishes this for a large class of knots.   









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

From rem-conf-request@es.net Mon Mar 03 12:45:44 1997 
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X-Mailer: exmh version 1.6.9 8/22/96
To: wchang@nist.gov (Wo Chang, x3439)
cc: rem-conf@es.net
Subject: Re: MBone camera for Windows 95/NT
In-reply-to: Your message of "Mon, 03 Mar 1997 09:20:21 EST." <199703031420.JAA03191@snad.ncsl.nist.gov>
Mime-Version: 1.0
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Date: Mon, 03 Mar 1997 09:43:31 -0800
From: Amancio Hasty <hasty@rah.star-gate.com>

Check out the WinCastTV it usually sells for $149: http://www.hauppauge.com/hcw/html/wc_data.htm

This is a high speed PCI video capture board and it also has a tuner.

	Enjoy,
	Amancio

>From The Desk Of Wo Chang, x3439 :
> I'm looking for a low cost, auto-focus, office use, 
> monitor-/desk-top mounting with wide-angle video camera 
> and a frame grabber (some might use parallel port instead,
> such as QuickCam) for running MBone vic video under 
> Windows 95/NT.  Any good suggestions or comments from
> your hands-on experiences?
> 
> Thanks in advanced for any pointers.
> 
> --Wo Chang <wchang@nist.gov>



From rem-conf-request@es.net Mon Mar 03 14:18:30 1997 
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From: Jerrlyn Iwata <jerrlyn@postgres.Berkeley.EDU>
Subject: [Reminder] 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. 

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


From rem-conf-request@es.net Mon Mar 03 14:43:54 1997 
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From: Scott Burris <scott@CTNS.UCLA.EDU>
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Subject: University of California Teching and Learning Technologies Conference
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This invitational conference will bring together faculty, library, computing 
and instructional development professionals, administrators and students 
 from throughout the University for an intensive consideration of the 
possibilities and challenges we face in incorporating new information and 
communication technologies into the heart of the teaching mission.

This event is scheduled for March 25 and March 26.  Selected programs are 
planned for broadcast over the MBONE. See http://www.ucop.edu/ucophome/auc/ 
and the sdr announcements for details.

We plan to use one video and one audio channel.

Questions about the conference may be directed to Martha Winnacker 
<martha.winnacker@ucop.edu>.

Questions or concerns about the broadcast may be directed to mbone-
admin@ctns.ucla.edu.


----------------
Scott Burris
UCLA Campus Telecommunications and Network Services
scott@ctns.ucla.edu

From rem-conf-request@es.net Mon Mar 03 16:05:43 1997 
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From: "Michael F. Speer" <Michael.Speer@Eng.Sun.COM>
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SUNERGY 25
"NEXT GENERATION NETWORKING"
 Live from Mountain View, California
 March 6, 1997
 
 Broadcast Time:  8:30 - 9:30 am (Pacific Time)
		  16:30-17:30 (GMT)

SUNERGY 25 breaks the network bottleneck, exploring routes 
around perceived bandwidth limitations in networked environments. 
Strategies for extending the capacity of the existing copper 
resources and launching community based fiber initiatives are 
explored. Streaming video, object switching, and wireless 
technologies bring the data to your doorstep. Check the Sunergy
website for guest bios, satellite coordinates and other details.
>From outside of Sun:  http://www.sun.com/sunergy/
Within Sun:  http://sunwww.ebay/sunergy/ 

GUESTS:

  JOHN GAGE (host), Director of the Science Office, Sun 
		Microsystems Computer Company

  GEOFFREY BAEHR (co-host), Chief Networking Officer,
		Sun Microsystems 

  TOM LYON, Founder and CTO, Ipsilon Networks, Inc.

  CHARLES PERKINS, Head, Internet Engineering Task Force,
  		Mobile IP Group

  LAWRENCE ROBERTS, President, ATM Systems


AUDIENCE PROFILE:

	  - Technologists wishing to keep up with the latest 
	    developments in networking
	  - Developers for whom networked or distributed 
	    computing represents a competitive advantage 
	  - Business planners who need to consider local and 
	    global networking for running their enterprises 
          - Anyone interested in keeping up with the business 
	    of distributed computing.

TECHNICAL CONTENT RATING: 4
(On a scale of 1-5, 5 being most technical)

STUDIO AUDIENCE:
Those in the Mountain View, California area are welcome to 
join us in the studio for the live broadcast.  If you are
interested in doing so, please send a request to sunergy@sun.com
for directions to the studio and to be included on the audience 
list.  

^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
This interactive television broadcast will be available
live, via satellite in NORTH AMERICA, SOUTH AMERICA and
EUROPE.  

Participate in the show by asking a question of the guests 
during the live broadcast:
	Telephone: 1-415-988-1171
	Fax:	   1-415-988-7003
	Or via the "Chat Room" on the Sunergy website:
	  http://www.sun.com/sunergy/front.html

SATELLITE INFORMATION:
	http://www.sun.com/sunergy/satellite.html
	[within Sun, go to: 
	http://sunwww.ebay/sunergy/satellite.html]

VIDEO ORDERING INFORMATION:  
	http://www.sun.com/sunergy/videoorder.html
 	[within Sun, go to:  
	http://sunwww.ebay/sunergy/videoorder.html]

Questions?  Or if you want any of the information available
on the website emailed to you, contact:
Sunergy Office
1-415-786-8205
sunergy@sun.com

From rem-conf-request@es.net Mon Mar 03 20:41:01 1997 
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Date: Mon, 03 Mar 1997 17:40:25 -0800
From: Amancio Hasty <hasty@rah.star-gate.com>


The latest Bt848 device driver supports the following cards:

o	WinCast/TV
o       STB PC TV
o       Intel Smart Video Recorder III

A module is included to support vic.

The latest driver also supports direct PCI to PCI data transfer so 
for instance one can display the capture frames directly on your
display. This is a PCI bus master transfer that is the CPU is not
used at all and all the data transfer goes thru the PCI bus.
The overall effect is well very smooth video display. For more
info just download the driver and see the README file.


ftp://rah.star-gate.com/pub/bt848-0.3.tar.gz

For on-going discussions about multimedia on FreeBSD post to
multimedia@freebsd.org

	Enjoy,
	Amancio





From rem-conf-request@es.net Mon Mar 03 20:47:40 1997 
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________________________________________________________
Benoit GAILLARD - B29 Rm138 
Phone : 00-44-(0)1473-647609
________________________________________________________

>----------
>From: 	Sprinter
>Sent: 	Tuesday, March 4, 1997 3:46 am
>To: 	rem-conf@es.net
>Subject: 	who
>
>
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>

From rem-conf-request@es.net Tue Mar 04 05:13:44 1997 
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Date: Tue, 4 Mar 1997 09:42:34 GMT
From: David Holter <D.A.Holter@soton.ac.uk>
Subject: Re: MBone camera for Windows 95/NT
To: "Wo Chang, x3439" <wchang@nist.gov>
cc: rem-conf@es.net
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On Mon, 3 Mar 1997 09:20:21 -0500 (EST) Wo Chang, x3439 
wrote:

> From: Wo Chang, x3439 <wchang@nist.gov>
> Date: Mon, 3 Mar 1997 09:20:21 -0500 (EST)
> Subject: MBone camera for Windows 95/NT
> To: rem-conf@es.net
> 
> I'm looking for a low cost, auto-focus, office use, 
> monitor-/desk-top mounting with wide-angle video camera 
> and a frame grabber (some might use parallel port instead,
> such as QuickCam) for running MBone vic video under 
> Windows 95/NT.  Any good suggestions or comments from
> your hands-on experiences?
> 
> Thanks in advanced for any pointers.
> 
> --Wo Chang <wchang@nist.gov>

Ive been using a Stinger card and the Flexcam camera from 
MacLine 0181 401 1118 which works quite well with some 
applications but I havent been able to get vic to run on the 
PC yet is crashes after a few seconds each time. I think 
there are still a few things to sort out in the PC software.


Regards

David Holter
Networking Group
Computing Services
University of Southampton
UK
+44 1703 593571
+44 1703 593939
D.A.Holter@soton.ac.uk




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

Title:       
	Xiao-Song Lin "Combinatorics and low dimensional topology"
Date:        
	Mar 11, 1997

Time:        
	12:00 PST8PDT 1 hours

Contact:     
	m1@msri.org

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

Description:        
	Lin's lecture from the  Combinatorial Problems Arising in Knots and  3-Manifolds workshop.









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

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

Title:       
	Serge Tabachnikov "Legendrian and transverse knots in tight contact structures"
Date:        
	Mar 12, 1997

Time:        
	12:00 PST8PDT 1 hours

Contact:     
	m1@msri.org

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

Description:        
	Tabachnikov's lecture from the  Combinatorial Problems Arising in Knots and  3-Manifolds workshop.









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

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

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Date:        
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Contact:     
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Description:        
	Bar Natan's lecture from the  Combinatorial Problems Arising in Knots and  3-Manifolds workshop.









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

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________________________________________________________________________

We apologize for multiple copies=20
________________________________________________________________________


			CALL FOR PAPERS

		1997 International Conference

	Intelligent Networks and Intelligence in Networks
=20
				(2IN97)


Paris - France=20
September 2-5, 1997

Organizer : IFIP TC WG 6.7 - Intelligent Networks

Sponsorship requested : Alcatel, Ericsson, France Telecom, Nokia, Nordic
teleoperators, Siemens, Telecom Finland, Lab. PRiSM

Aim of the conference
To identify and study current issues related to the development of
intelligent capabilities in networks. These issues include the
development and distribution of services in broadband and mobile
networks.
This conference belongs to a series of IFIP conference on Intelligent
Network. The first one took place in Lappeeranta August 94, the second
one in Copenhagen, August 95. The proceedings of both events have been
published by Chapman&Hall.
IFIP Working Group 6.7 on IN has concentrated with the research and
development of Intelligent Networks architectures. First the activities
have concentrated in service creation, service management, database
issues, feature interaction, IN performance and advanced signaling for
broadband services. Later on the research activities have turned towards
the distribution of intelligence in networks and IN applications to
multimedia and mobility. The market issues of new services have also
been studied. From the system development point of view, topics from OMG
and TINA-C have been considered.

Main fields of interest to 2IN97:

IN Markets and trends

- Value added services market=20
- Development in telecom area
- Multimedia IN
- IN and the Internet


Standardization/integration and legacies

- Standardization aspects
- TINA
- IN and B-ISDN
- IN and mobile networks
- Intelligence in ATM networks


Service types, service architecture and=20
- Service creation, distribution and  management
- IN service architecture
- Service creation architecture applied in multimedia services (voice,
data, mobile)
- Multimedia services


Tools

- Intelligent control schemes
- Multimedia call models
- Multimedia call control protocols
- Intelligent routing
- Continuous media
- Performance issues
- Intelligent agents
- Distributed artificial intelligence and multi-agent systems for
networks
- JAVA technology for IN services


IN systems and platforms

- IN architecture
- Intelligence and Network Management
- Real time data bases and caches for services



Workshop organization
Lab PRiSM (University of Versailles)

Place of the conference : French Ministry of Telecommunication, Avenue
de Segur, Paris, France

Conference chair:=20
Olli Martikainen FIN=20
Guy Pujolle F

Program committee chair :
Dominique Ga=EFti F

Program committee members

James Aitken UK
Almeida Maria J.B. BR
Andy Bihain USA
Gilles Bregant F
Carla Capellmann, GER
Christian Chabernaud F
Peter Delgado UK
Heinz Dibold GER
Jacques Ferber F
Frank Gallivan USA
Philip Ginzboorg FIN
Shri Goyal USA
Serge Haddad F
Heikki Hammainen FIN
Villy Baek Iversen DK
Bijan Jabbari USA
Jorma Jormakka FIN
Koos Koen RSA
Ulf K=F6rner S
Paul K=FChn GER
Roberto Kung F
Jacques Labetoulle F
Xuejia Lai CH
Martine Lapierre F
Kari Lautanala FIN
Aurel A. Lazar USA
Olli Martikainen FIN=20
Valeri A. Naoumov RF
Karl W. Neunast GER
Jorgen Norgaard DK
Herv=E9 Precioso F
Guy Pujolle F
Kimmo Raatikainen FIN
Konstantin E. Samouylov RF
Manfred Schneps-Schneppe RUS
Lennart S=F6derberg S
Haitao Tang CN


Organization chair :
Nadia Boukhatem F
Thierry Hua F

Important dates:
 March 21, 1997: Deadline for Paper Submission
 May 15, 1997: Notification of Acceptance
 July 1, 1997: Final Version Due
 September 2, 1997: Tutorial
 September 3-5, 1997: Conference

Instructions:
Submit five copies of a full paper by March 21, 1997. Papers should be
no longer than 20 double spaced typewritten pages, including tables and
figures. The proceedings will be published by Chapman&Hall

All correspondence should be addressed:
	- by email (postcript version of the paper) - 2IN97@prism.uvsq.fr
	- or by mail to
2IN97 Conference
Lab PRiSM - University of Versailles
45 av. des Etats-Unis
78035 Versailles Cedex - France

------------------------------------------------------------------------
2IN'97
1st Name: ..............................................................
Last Name:..............................................................
email:..................................................................
Title:..................................................................
Organization:...........................................................
Postal address:.........................................................
........................................................................
City:..............................Postal code:.........................
Country:................................................................
Telephone:..............................................................
Fax:....................................................................


O Topics I am interested in:............................................
O I wish to attend the ICCC'97 conference.
O I wish to present a paper and will submit a paper by March 21, 1997.=20
Title of the paper :
........................................................................
........................................................................
........................................................................
________________________________________________________________________
________________________________________________________________________

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Date: Tue, 4 Mar 1997 18:34:31 GMT
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    urban@egd.igd.fhg.de, xtp-relay@cs.concordia.ca, yevin@sam.imash.ras.ru
Subject: EuropIA

 europia`97
               @EDINBURGH
       ------------------
       DESIGN and the NET
       ------------------

                        http://www.caad.ed.ac.uk/europia/

Details and Provisional Programme 
---------------------------------

Sixth International Conference on the Application/Implications of Computer
Networking in Architecture, Construction, Design, Civil Engineering and
Urban Planning

Sixime Confrence Internationale sur les applications/implications des
nouvelles technologies de communication en Architecture, Construction,
Design, Ingnierie et Planification urbaine

     2-3 April 1997

     Department of Architecture
     University of Edinburgh
     Edinburgh  EH1 1JZ  Scotland
     http://www.caad.ed.ac.uk/europia/


This conference brings together researchers in design, architecture,
engineering, construction management, cognitive science, computer science,
artificial intelligence, sociology, geography and education as well as
industry partners, practitioners and other users of networked
communications.

The focus for this diverse group is design - how we design with and for
networked communications - appropriating recent exciting developments in
medium and broad band local area and wide area networks, new protocols for
interaction and new communications tools such as Web browsers and network
aware multimedia tools.

DATES

   Conference                    2-3 April    1997

REGISTRATION FEE

   Authors           175 pounds stirling
   Non-authors       250 pounds stirling
   Conference Dinner  27 pounds stirling

VENUE

   UnivEd Training and Conference Centre
   11 South College Street
   Edinburgh  EH8 9AA


SEND QUEURIES ABOUT REGISTRATION AND ACCOMMODATION TO

   Ronnie Galloway
   Conference Manager
   Ronnie.Galloway@ed.ac.uk
   UnivEd Technologies Limited
   11 South College Street, Edinburgh, EH8 9AA
   Ph   +44 131 650 9020
   Fax  +44 131 650 9019



EuropIA'97 Design and the Net 2-3 April 1997 
+++++++++++++++++++++++++++++++++++++++++

Provisional Programme
=====================

(4/3/97) 

Wednesday

9.15-9.30      Welcome

               Working with the web: Networked CAD applications 

9.30-10.00   . Common libraries for networked engineering 
               applications
                 Tuttle et al
10.00-10.30  . The VRML model of the city of Bath
                 Bourdakis and Day

10.30-11.00    break

11.00-11.30  . Shared information system for urban and architectural 
               design
                 Caneparo
11.30-12.00  . Communication in distributed statistical frameworks 
               for historic building documentation: The abbey of 
               Valmagne
                 Warden & Vasquez
12.00-12.30  . CAD On-line
                 Coyne & Lee

12.30-2.00      lunch

               Working with the web: Innovation and technology

2.00-2.30    . A method for organising and sharing architectural 
               project information in the WWW
                 Ofluoglu & Kalisperis
2.30-3.00    . Designing learning environments using java
                 Pang & Edmonds
3.00-3.30    . A web -based design support environement for knowledge 
               discovery and machine learning
                 Branki, Bridges, Wallis & Aird

3.30-4.00      break

4.00-4.30    . Ways of aiding navigation in VRML worlds
                 Caritos and Rutherford
4.30-5.00    . Multilevel representations of architectural designs
                 Koutamanis
5.00-5.30    . A network aid for design project management
                 Roussel & Zriek

Thursday 

               The web in practice: Education in cyberspace 

9.30-10.00   . The tex-mex virtual design studio
                 Vasquez & Trigo
10.00-10.30  . Networked based decision support for building design
                 Burry & Smithers
10.30-11.00  . Digital design: students communicating experiential 
               spatial qualities in a networked computing environment
                 Burley, Megza & Bauer

11.00-11.30    break

               The web in practice: The emerging webculture

11.30-12.00  . HQ video conferencing and its application to the 
               teaching of architecture
                 Lindsay & Grant
12.00-12.30  . Sedimented practices of 'reading' design descriptions: 
               from paper to screen
                 Tweed
12.30-1.00   . The architect's role in the information age
                 Rossis

1.00-2.00      lunch

2.00-2.30    . Collaborative design by discovery and the web
                 Peng
2.30-3.00    . Method DTN: Design through the net
                 Reze
3.00-3.30    . A cyberspace in time
                 Szalapaj & Filho

3.30-4.30      Poster Session

               The web in practice: The emerging webculture (cont.)

4.30-5.00    . A virtual monoculture of visual communication
                 Newton & Shi
5.00-5.30    . Design information and the superhighway: 
               roadworks ahead?
                 Ramscar & Lee

5.30-6.00      Plenary Session

Poster Session

             . A web based experience of a collaborative 
               argumentation system design
                 Chen, Branki & Newman

             . Networked collaborative design implementation
                 Burry, Coll & Serrano

             . Up the down staircase...
                 Brady

             . Broadband videoconference workstation
                 Brandt et al

             . DesignNet electronic journal
                 Westphal & Burley

             . Cybernetic theory & architecture
                 Hunt

             . Intranets as tools for strategy
                 Thomas

             . Building an intranet model for CAAD
                 Paranandi


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        I'm posting the following information as a public service:

        To UNSUBSCRIBE from the rem-conf list send your UNSUBSCRIBE
        request (in both Subject: line and body of message) to:

                        rem-conf-request@es.net

        The list maintainers do NOT read the messages sent to the whole
        list.

        Back when I ran/maintained this list, I used to post a periodic
        reminder for change requests and archive(s) location...I also
        studiously ignored drop requests sent to the whole list unless
        I had some time available to service them.

        Please do NOT send *me* any requests for changes...I've moved on...

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



From rem-conf-request@es.net Tue Mar 04 21:31:43 1997 
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Subject: Re: How to UNSUBSCRIBE from rem-conf
To: ari@sprintlabs.com (Ari Ollikainen)
Date: Tue, 4 Mar 1997 18:31:54 -0800 (PST)
Cc: rem-conf@es.net
In-Reply-To: <v02140b0baf42843c292f@[199.2.53.28]> from "Ari Ollikainen" at Mar 4, 97 06:03:26 pm
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You may also unsubscribe from the web at ...
http://www.msri.org/mbone/ml.html

: 
:         I'm posting the following information as a public service:
: 
:         To UNSUBSCRIBE from the rem-conf list send your UNSUBSCRIBE
:         request (in both Subject: line and body of message) to:
: 
:                         rem-conf-request@es.net
: 
:         The list maintainers do NOT read the messages sent to the whole
:         list.
: 
:         Back when I ran/maintained this list, I used to post a periodic
:         reminder for change requests and archive(s) location...I also
:         studiously ignored drop requests sent to the whole list unless
:         I had some time available to service them.
: 
:         Please do NOT send *me* any requests for changes...I've moved on...
: 
:            _/_/   _/_/_/_/    _/  Ari Ollikainen <ari@sprintlabs.com>
:         _/  _/   _/     _/   _/ Sprint Advanced Technology Laboratories
:      _/_/_/_/   _/_/_/_/    _/ Switching and Interworking Development
:    _/     _/   _/     _/   _/ 1 Adrian Court (M/S: CABURS0102)
:  _/      _/   _/       _/ _/ Burlingame, CA 94010
: ~~RECOM Technologies Inc.~~ Voice: 415.375.4265 FAX: 415.375.4490
: 
: 
: 



 ________  o      Dave Wright 
 _______ _/\_>    dave@msri.org
 _______O=>// O   91CBR600F2 AFM# 316
 --------------   Mathematical Sciences Research Institute


From rem-conf-request@es.net Tue Mar 04 23:58:37 1997 
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          Tue, 4 Mar 1997 23:57:35 -0500 (EST)
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Date: Tue, 04 Mar 1997 23:57:35 -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: Comet Group Seminar: B. Badrinath, Rutgers Univ., "On Supporting an 
         Internet Cellular Phone Network"
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MBone Broadcast Announcement

Title:
    ON SUPPORTING AN INTERNET CELLULAR PHONE NETWORK

Speaker:
    Br. Badrinath
    Department of Computer Science,
    Rutgers University

Date:
    Thursday, March. 06, 1997

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

Multicast Address:
    audio:  DVI, RTP, 224.2.194.51/21104 , ttl=127
    video:  H.261, RTP, 224.2.170.239/57392 , 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:

Providing service guarantees in a mobile internetwork requires that
the network account for host mobility. Host mobility imposes spatial
resource demands on the network. In this talk, we will explore new
service models, admission control policies, and spatial reservation
schemes for mobile hosts in ISPN(Integrated Services Packet Network). 
We will also discuss aspects of MRSVP; a Mobile RSVP scheme to support
applications requiring quality of service in a mobile internetwork. One
such example: An internet cellular phone.


-- 
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 Mar 05 08:16:05 1997 
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To: congres@prism.uvsq.fr
From: N.Oloughlin@lcad.ac.uk (Niall O'Loughlin)
Subject: 21N97
Date: Wed, 5 Mar 1997 12:48:45 +0000

I have just received part of a message about a call for papers but the
important part is missing. Would you send the message again, please?

Regards,

Niall O'Loughlin



Dr Niall O'Loughlin
Loughborough College of Art and Design
Epinal Way, Loughborough, Leicestershire
LE11 3GE   U.K.

Telephone       01509-261515 ext 176
International   +44-1509-261515 ext 176
Fax             01509-265515
International   +44-1509-265515
e-mail:         N.Oloughlin@lcad.ac.uk



From rem-conf-request@es.net Wed Mar 05 12:48:17 1997 
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To: rem-conf@es.net
Subject: socks/mbone firewalls
Date: Wed, 05 Mar 1997 17:47:26 +0000
Message-ID: <2466.857584046@cs.ucl.ac.uk>
From: Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>



in order to help some commercial intranet and ISP company managers
feel happier about mbone deployment:

can anyone point me at a discussion paper/description of firewall 
usage, multicast UDP and mbone applications, and any rationale as to
a) attacks (e.g. denial of service esp., but usual service breakins
and disclosure etc obviously useful too)
b) defenses
c) procedures for mbone app<->firewall auth

etc etc

thanks
jon

From rem-conf-request@es.net Wed Mar 05 16:28:51 1997 
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Date: Wed, 5 Mar 1997 16:27:04 -0500 (EST)
From: MADPR@aol.com
Message-ID: <970305162702_2028873154@emout04.mail.aol.com>
To: rem-conf@es.net
Subject: Media Demos of Player-less Audio Broadcasting

Vosaic Audio for Java, announced worldwide last week, will be exclusively
demonstrated to the media at InternetWorld, Mar. 12-15, Los Angeles 
Convention Center.  Highlights of the demonstration will include live radio
broadcasting and individual audio streams on request.  

Vosaic Audio for Java is the first audio streaming system which runs 
without a proprietary player.
     
Call Mitch Koppel or Allyson Stinchfield (312) 541-8787 or e-mail at
mkoppel@msn.com to confirm an appointment for private demonstration in 
Press Suite Room 11.  Copy of announcement release follows:
------------------------------------------------------------------------------
-------------------------------------------
								CONTACTS:	
								Mitch Koppel/ Phil Dunne
								MINKUS & DUNNE
								312/ 541-8787
								mkoppel@msn.com


	VOSAIC ANNOUNCES "VOSAIC AUDIO FOR JAVA"

A revolutionary server-less, plug-in free, firewall-busting, Java-based sound
streaming system for the Web

	FEBRUARY 26, 1997 (Chicago, IL) --- Vosaic L.L.C. today announced the
introduction of what is believed to be the first pure Java-based sound
streaming system using the global ITU standard, GSM audio, VOSAIC Audio for
Java.  It is a Java revolution in the making as it enables Web masters to
easily include audio on their web pages and take advantage of Java's unique
cross-platform capabilities.  All major browsers are totally compliant;
simply point and click on a VOSAIC Audio for Java enabled HTML web page and
immediately listen to audio...no plug-ins to fuss with, no installation
routines to screw up your machine.  It is simply no-brainer, push-button,
turn-on-the-radio simplicity to hear audio over the Net.
	"Vosaic has created an exciting new multimedia enhancement for the Web which
I believe will become an important part of how we communicate globally," says
Lana Coronado, Director of Worldwide Information Technology Services at CNNfn
Interactive, a leading financial news network in New York City.  "Music,
news, teaching and training will become the staple of the Web and Vosaic's
 unique Java-based audio streaming system is a big step forward --- an ideal
way to enhance anyone's Web site."
	The VOSAIC Audio for Java authoring tool is also revolutionary in its
simplicity.  Based on pure Java, it works on all platforms and makes use of
the global ITU standards for sound, the GSM audio codec, into a universal
audio sound source that is used in most of the world's cellular phone
systems.  The VOSAIC Audio for Java authoring tool takes the universally
available WAV and AIFF files as sound sources and simply applies the built-in
auto-conversion tool and GSM sound compressor to immediately generate VOSAIC
Audio for Java compliant sound files for instant audio streaming over the
Net.  Once converted, simply add a simple HTML statement to the Web page and
start streaming off any Web server immediately.
	"What Vosaic has done with this new technology has changed the entire
business model for multimedia and the Internet.  No longer is the installed
base or number of plug-ins downloaded a factor.  This VOSAIC Audio for Java
is by far the easiest system there is to use.  This is what multimedia and
the Web should be and this is what is going to make the Web come alive...a
paradigm shift for both Intra and Internets and the new broadcast ExtraNets
(BxNets)," says Jeff Marshall, Senior Managing Director of Communications
Technology at Bear, Stearns & Co, Inc., a leading investment firm on Wall
Street.
	The VOSAIC Audio for Java sound streaming also provides tremendous value for
money.  For a limited time only, VOSAIC Audio for Java is FREE for
downloading.  Unlike other audio streaming systems, VOSAIC Audio for Java is
a flat-rate, $ 249 each, UNLIMITED audio streaming system. "Web operators and
content providers will no longer will feel constrained by other audio
casters' usurious per-stream-pricing, dollars per hit or percent of revenue
demands. They are now free to let audio be the universally accepted medium it
was meant to be since the days of Marconi and BBC World radio", says Stuart
Johnstone, CEO and co-founder of Vosaic L.L.C. "Vosaic sees the delivery of
audio over the Web as the fundamental basis of global human communications.
 As such, we believe the delivery of audio should be a commodity, available
to all ubiquitously and for the benefit of the planet.  Look to Vosaic in the
near future to deliver a number of exciting musical and special events in
concert with content partners", states Johnstone. 
	VOSAIC Audio for Java has the capability to punch through most known
firewalls so that content creators/providers can be relatively confident that
their message is punching through to the rest of the world.  As GSM can be
easily streamed through 28.8 modems, the world becomes the author's oyster.
	Coming soon---look out for VOSAIC Stereo for Java using Dolby Laboratories
famous AC3 codec, known as DOLBY Net, and for VOSAIC Radio for Java, the
unicasting, multicasting audio streaming system for "live" events.
	Vosaic L.L.C. is in the visual and audio IP-network telecommunications
business and was founded in 1996 to research, develop, and commercialize
technologies arising out of the University of Illinois's Systems Research
Group (comprised of Digital Computer Labs and the NCSA). Vosaic L.L.C. is
located at 1248 North Lake Shore Dr., Chicago, Il 60610, Phone +(312)
943-6764, FAX  +(312) 943-6765 and on the web at <http://www.vosaic.com>.  

						# # #
Edior's Notes: 
- visit the Vosaic homepage to run the instant playing, streaming VOSAIC
Audio for Java 
- call or E-mail <madpr@mcimail.com> or mkoppel@msn.com for evaluation
software.
- trademarks are properties of their respective trademark owners.
			
(end)


From rem-conf-request@es.net Wed Mar 05 17:14:23 1997 
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Date: Wed, 5 Mar 1997 14:13:38 -0800 (PST)
From: Richard Core <rich@nttlabs.com>
Message-Id: <199703052213.OAA04469@panda.nttlabs.com>
To: rem-conf@es.net
Subject: vic option needed (xmit at startup)
X-Sun-Charset: US-ASCII

I would like vic to be able to startup in Transmit mode.  Anyone have or seen
such a vic?  A command line option would work nicely in my usage.

Thanks,
Richard

From rem-conf-request@es.net Wed Mar 05 17:49:57 1997 
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Date: Wed, 5 Mar 1997 14:49:02 -0800 (PST)
From: Marcy Kamps <kamps@Nersc.GOV>
Message-Id: <199703052249.OAA04097@yosemite.lbl.gov>
To: kruse@nordicglobal.com
Subject: Re: HELP
Cc: rem-conf@es.net
X-Sun-Charset: US-ASCII

Holger,

The rem-conf@es.net mailing list does not use listserv commands.
What kind of help do you need from the rem-conf@es.net list
maintainer?

Thanks,


Marcy L. Kamps
************************************************************************
National Energy Research Scientific Computing Center (NERSC)
Energy Sciences Network (ESnet)  Lawrence Berkeley National Laboratory
Phone   : 1-510-486-6448         One Cyclotron Road, Bldg. 50C, Room 103
 (or)   : 1-800-66-NERSC         Berkeley, CA 94720  USA   
Email   : kamps@nersc.gov        Internet: postmaster@es.net
************************************************************************

----- Begin Included Message -----

>From rem-conf-postmaster-request@es.net Wed Mar  5 07:58 PST 1997
To: rem-conf-request <rem-conf-request@es.net>
From: Holger Kruse <kruse@nordicglobal.com>
Date: Wed, 05 Mar 1997 18:55:41 -0400
Reply-To: kruse <kruse@nordicglobal.com>
X-Mailer: GoodNews 1.5 (Dec 28 1996)
Subject: HELP
Organization: Nordic Global Inc.
Content-Type: text
Content-Length: 90

HELP
--
Holger Kruse   kruse@nordicglobal.com
               http://www.nordicglobal.com



----- End Included Message -----



From rem-conf-request@es.net Wed Mar 05 18:09:32 1997 
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Message-ID: <503A2A3C2932CF118D8800805FD44E1802BBD79B@RED-68-MSG>
From: Eric Fleischman <ericfl@MICROSOFT.com>
To: "'confctrl@isi.edu'" <confctrl@isi.edu>, 
    "'rem-conf@es.net'" <rem-conf@es.net>
Subject: In Defense of Interleaved Data Formats
Date: Wed, 5 Mar 1997 15:05:00 -0800
X-Priority: 3
X-Mailer: Internet Mail Service (5.0.1457.3)

The list of reasons given in Section 5.2 of RFC 1889 as to why RTP
sessions must be segregated by media type reminds me of a discussion I
overheard a few years back. The one guy was saying "Trucks should be
banned from the roadways. They are reckless, prone-to-overturn hazards
belching out huge amounts of pollution. Their compression breaks cause
deafness and their weight fosters the premature breakdown of the road
surfaces..." At this point he lost me because I started to think about
the implications of hauling manure, gravel, or plywood with cars. My
brother-in-law, of course, does exactly that by using a specially
constructed trailer made just for such purposes. But then, he gets stuck
in some pretty pitiful places. 

Mind you, I'm not adverse to cars - I own one myself and use it for my
daily commute. Similarly, I would gladly say "amen" in any praise
meeting about the advantages of payload-segregated RTP traffic. I
readily concur that this approach leads to some marvelously creative
solutions for a wide variety of practical problems.  I am actively
interested in Van Jacobson's and Steve McCanne's "Receiver-driven
Layered Multicast" research. Hey! I really do believe! Don't get me
wrong: I'm all for cars! ... I just want to use my truck when it comes
to hauling gravel. 

Separately encoded media RTP sessions are a marvelous solution to
conferencing applications. They can also be used for "On-Demand"
Streaming applications -- just like my brother-in-law can successfully
haul gravel with his trailer. However, there are many occasions when
"On-Demand Streaming" requires the use of interleaved data formats.
Here's a few:

1.	You can save substantial amount of overhead for sessions
accessed by low speed modems. The argument here is that by interleaving,
only one IP, UDP, and RTP header is used, instead of the X number of
duplicate headers for each of the X media types. (Of course everything
is packetized so this saving accumulates at a constant rate.) The
counter-argument against this is that Van Jacobson's header compression
can convert these extra headers down to the size of pinheads, saving
even more than the original savings. Then the counter-argument to that
counter-argument is that if you are spending that type of overhead
(i.e., on-the-fly compression) to process your "on demand" application
then the number of sessions you can simultaneously support on that
server is detrimentally impacted. (Is this a ploy by high-end machine
builders to sell more product or what?) 

2.	While some "on-demand" content is thrown together by
chimpanzees, most of it is carefully crafted by artisans. For this
reason, most on-demand streaming applications go to extraordinary
lengths to ensure that the multimedia experience presented to the client
is the identical multimedia experience which was designed by the content
creator. Interleaving encourages tight synchronization and offers a
higher degree of probability for preserving the designed experience in
the presence of data loss, network variability, and sunspots.

3.	On-demand media may be composed of many media types. Some of
these are "streaming" media types (e.g., voice, video, and MIDI) while
others are not (e.g., still images, URL flippings, script commands and
executables). "On demand uses" also encourage the innovative invention
of rare media types. Regardless, the net effect is that there will be N
media streams for on-demand usages where N may be quite a handful.
Interleaving permits these many media streams to be presented in a
harmonious, orchestrated and optionally synchronized fashion. While this
is an expansion on the previous point, it also seeks to highlight that
conferencing and "on-demand" uses are quite different when it comes to
range of media which must be coordinated and the variability and
precision demanded for controlling the interaction between these media
types.

4.	Disk-head movement efficiency on the server. The argument here
is that interleaving encourages more efficient disk storage approaches
with superior I/O processing capabilities. This directly effects the
construction of more efficient network streaming approaches. This
efficiency directly translates into server performance including the
ability to support a larger number of simultaneous "On-demand" sessions
by that server.

I own both a truck and a car. I similarly want to use media segregated
streams over RTP in environments where it makes sense to do so and to
use interleaved data streams over RTP in environments where it makes
sense to do so.
One exciting aspect of the current RTSP proposal is that it should
theoretically permit both approaches to be carried over RTP. Traditional
RTP traffic can be carried in the traditional way. Interleaved data
traffic can be supported via encapsulating the media definitions (e.g.,
the information contained within an ASF or RMFF header) which can be
obtained by an RTSP "get" command. A dynamic RTP payload type value can
then be associated with that header information for that session. The
ASF or RMFF data can then be "streamed" via RTP and interpreted
correctly.

From rem-conf-request@es.net Wed Mar 05 18:42:41 1997 
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          Wed, 5 Mar 1997 15:41:28 -0800
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          Wed, 5 Mar 1997 15:41:23 -0800 (PST)
Date: Wed, 5 Mar 1997 15:41:23 -0800 (PST)
From: Richard Core <rich@nttlabs.com>
Message-Id: <199703052341.PAA04719@panda.nttlabs.com>
To: DAAgarwal@lbl.gov
Subject: vic transmitOnStartup (NeverMind!)
Cc: vic@ee.lbl.gov, rem-conf@es.net
X-Sun-Charset: US-ASCII

Never mind... just found "yesno transmitOnStartup" and the -X resource=value arg

I'm feeling foolish...
Thanks,
Richard

From rem-conf-request@es.net Wed Mar 05 19:39:27 1997 
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          Wed, 5 Mar 1997 16:37:53 -0800
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From: Mark Handley <mjh@isi.edu>
X-Organisation: Information Sciences Institute, USC
X-Phone: +1 617 253 6011
To: Eric Fleischman <ericfl@MICROSOFT.com>
cc: "'confctrl@isi.edu'" <confctrl@isi.edu>, 
    "'rem-conf@es.net'" <rem-conf@es.net>
Subject: Re: In Defense of Interleaved Data Formats
In-reply-to: Your message of "Wed, 05 Mar 1997 15:05:00 PST." <503A2A3C2932CF118D8800805FD44E1802BBD79B@RED-68-MSG>
Date: Wed, 05 Mar 1997 19:36:18 -0500
Message-ID: <8946.857608578@buttle.lcs.mit.edu>
Sender: mjh@buttle.lcs.mit.edu


>The list of reasons given in Section 5.2 of RFC 1889 as to why RTP
>sessions must be segregated by media type reminds me...
...
>However, there are many occasions when
>"On-Demand Streaming" requires the use of interleaved data formats.
>Here's a few:

>1.	You can save substantial amount of overhead for sessions
>accessed by low speed modems.
...
>(i.e., on-the-fly compression) to process your "on demand" application
>then the number of sessions you can simultaneously support on that
>server is detrimentally impacted.

Note that header compression is not end-to-end, but only between your
end-system and its local terminal server.  It doesn't impact 
multimedia data servers at all.


>2.  ... Interleaving encourages tight synchronization and offers a
>higher degree of probability for preserving the designed experience in
>the presence of data loss, network variability, and sunspots.

Actually this isn't true.  Interleaving gives the network less
oportunity to preserve your audio preferentially from your video if
the network gets loaded (which is usually the right thing to do).

In addition, synchronisation is no easier.  You have all the same
information in separate RTP streams that you have in interleaved
streams.  In either case you need to cope with mismatches in clock
rate between the video stream and a receiving audio device that which
is not clocked at exactly the frequency the manufacturer claims.  Once
you're re-tuning the audio buffering periodically, it makes no
difference whether the streams were transported together or
separately.

>3.	On-demand media may be composed of many media types. Some of
>these are "streaming" media types (e.g., voice, video, and MIDI) while
>others are not (e.g., still images, URL flippings, script commands and
>executables). "On demand uses" also encourage the innovative invention
>of rare media types. Regardless, the net effect is that there will be N
>media streams for on-demand usages where N may be quite a handful.

This seems to be an argument *against* interleaving to me...

>4.	Disk-head movement efficiency on the server. The argument here
>is that interleaving encourages more efficient disk storage approaches
>with superior I/O processing capabilities. This directly effects the
>construction of more efficient network streaming approaches.

Separate streams lets you store your audio and video on different
discs, and hence *improves* performance in some cases :-)

Besides, if your server isn't doing some form of tuning to available
bandwidth, it's going to be a dinosaur in the long run.  Right now you
might get away with it but the internet won't stand large numbers of
unadaptive streams, and so we'll get schemes like RED, CBQ, WFQ, etc
deployed to defend it.  Once the server is doing bandwidth adaption of
such streams, I don't really believe that the "interleaving impacts
the disc" argument holds well because you'll likely be using layered
codecs anyway.  At the very least, you won't be able to stream
straight from disc to net.

Mark

From rem-conf-request@es.net Wed Mar 05 21:51:12 1997 
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From: Eric Fleischman <ericfl@MICROSOFT.com>
To: 'Mark Handley' <mjh@isi.edu>
Cc: "'confctrl@isi.edu'" <confctrl@isi.edu>, 
    "'rem-conf@es.net'" <rem-conf@es.net>
Subject: RE: In Defense of Interleaved Data Formats
Date: Wed, 5 Mar 1997 18:46:43 -0800
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Your comments appear to be written from a conferencing perspective. I
fear that I haven't helped you to understand the fundamental requirement
of a significant percentage (i.e., probably the majority) of "on demand"
usage. This requirement is that the "multimedia experience" is preserved
independently of what is happening on the network. Thus, if the network
becomes saturated, the worst thing to do would be to drop a given media
stream in order to preserve another stream based on network assumptions
as to which stream may be "more important" (i.e., your suggestion to
drop video but to preserve audio in a congested situation). This
incorrectly presumes that one media type is more important than another
in a "multimedia experience". They all are important if the "experience"
is to be maintained in tact. Now mind you, the content creator may
indeed choose for certain streams to be given preferential treatment but
it is simply impossible to know which stream will be given such
preference in the general case - unlike the conferencing experience.

Rather, the appropriate thing to do in the face of severe network
congestion would be to do one of the following:
*	Cancel the experience outright due to underlying network
problems
*	Delay the experience until such a time as the network can handle
the load
*	Continue the experience with "choppy" reception

Thus, as you can see, interleaving is the most appropriate approach for
preserving a multimedia experience since it is more difficult to manage
an appropriate response with segregated media streams and they are more
prone to distort the experience in the face of network congestion.

It is doubtful that you will be able to understand what I'm talking
about as long as you think as an engineer. I am talking "experience" not
"information". This is a left-brain, right-brain type of dichotomy. To
understand what I'm saying, you have to think as an artist. What we are
talking about is the preservation of an artistic creation across the
network. That this creation may educate or convey critical, essential
information does not diminish the fact that we are talking about
art-forms when we are talking about "on demand" streaming.

I therefore encourage you to re-read my original message from this
point-of-view. Should you do so, then I think that you will find points
2-4 to be very intuitive and obvious. I thank you for your input about
point #1 since that was information about which I was previously
unaware.

Also, you seem to be thinking only about audio and video when I say
"multimedia".  While interesting experiences are indeed created using
only those two media types, I would prefer you to think of audio + video
+ MIDI + URL flipping + streaming text + active background applications
like stock tickers + still images + ....   Why limit ourselves to only
two media types? Certainly the people making streaming "on-demand"
content don't.

I implied in my original message that there were many reasons supporting
interleaving of data, even though I only cited four. Here's another
(and, if you keep disagreeing with me, I'll have to keep giving other
reasons ;-) ):

When the artist creates the multimedia experience, the artist creates
that experience independent from any underlying network concerns (in the
general case) with the sole exception of data rates. The experience is
indeed targeted to be streamed at a given data rate and that rate
constrains how much data is available at any point of the timeline. The
artist doesn't care in many cases whether the experience is carried over
RTP or whether it is carried via TCP, UDP, or HTTP, IPX/SPX, SNA, or
whatever just as long as it arrives as designed.  Now you will perhaps
argue that streaming content can't be carried over connection-oriented
transports like the TCP or HTTP such as were mentioned in the previous
sentence. Experience has demonstrated that this observation may very
well be true for segregated media streams, however, it is emphatically
*not* true for interleaved media streams (given a certain preroll
value). 

Here's the deal: the content creator determines the experience but the
local client's environment determines the mechanism by which that
experience is delivered. When we made our product, therefore, we were
compelled to de-couple the experience from the transport. This means
that any given experience will need to be carried via RTP here but HTTP
there. Why HTTP and why native TCP? Well, it turns out that many users
are very concerned about their firewalls with the degree of concern
varying between companies. Some corporations will permit us to "bore
through" their firewalls via a UDP session, others will permit a single
port via TCP streaming, while others refuse to permit any holes in their
firewalls whatsoever. In the latter case we stream via HTTP if they
permit that protocol to enter their environment, otherwise we are
blocked out.

		-----Original Message-----
		From:	Mark Handley [SMTP:mjh@isi.edu]
		Sent:	Wednesday, March 05, 1997 4:36 PM
		To:	Eric Fleischman
		Cc:	'confctrl@isi.edu'; 'rem-conf@es.net'
		Subject:	Re: In Defense of Interleaved Data
Formats 


		Your comments appear to be written from a conferencing
perspective. They don't seem to understand the fundamental requirements
of a significant percentage of "on demand" usage. This requirement is
that the "multimedia experience" is preserved independently of what is
happening on the network. Thus, if the network becomes saturated, the
absolutely wrong thing to do would be to drop a given media stream to
preserve another stream (i.e., your suggestion to drop video but to
preserve audio in a congested situation). Rather, the appropriate thing
to do would be to do one of the following:

			>2.  ... Interleaving encourages tight
synchronization and offers a
			>higher degree of probability for preserving the
designed experience in
			>the presence of data loss, network variability,
and sunspots.

		Actually this isn't true.  Interleaving gives the
network less oportunity to preserve your audio preferentially from your
video if the network gets loaded (which is usually the right thing to
do).
		In addition, synchronisation is no easier.  You have all
the same information in separate RTP streams that you have in
interleaved streams.  In either case you need to cope with mismatches in
clock rate between the video stream and a receiving audio device that
which is not clocked at exactly the frequency the manufacturer claims.
Once you're re-tuning the audio buffering periodically, it makes no
difference whether the streams were transported together or separately.
			>3.	On-demand media may be composed of many
media types. Some of
			>these are "streaming" media types (e.g., voice,
video, and MIDI) while
			>others are not (e.g., still images, URL
flippings, script commands and
			>executables). "On demand uses" also encourage
the innovative invention
			>of rare media types. Regardless, the net effect
is that there will be N
			>media streams for on-demand usages where N may
be quite a handful.

		This seems to be an argument *against* interleaving to
me...
			>4.	Disk-head movement efficiency on the
server. The argument here
			>is that interleaving encourages more efficient
disk storage approaches
			>with superior I/O processing capabilities. This
directly effects the
			>construction of more efficient network
streaming approaches.

		Separate streams lets you store your audio and video on
different discs, and hence *improves* performance in some cases :-)
		Besides, if your server isn't doing some form of tuning
to available bandwidth, it's going to be a dinosaur in the long run.
Right now you might get away with it but the internet won't stand large
numbers of unadaptive streams, and so we'll get schemes like RED, CBQ,
WFQ, etc deployed to defend it.  Once the server is doing bandwidth
adaption of such streams, I don't really believe that the "interleaving
impacts the disc" argument holds well because you'll likely be using
layered codecs anyway.  At the very least, you won't be able to stream
straight from disc to net.
		Mark

From rem-conf-request@es.net Wed Mar 05 22:17:43 1997 
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Date: Wed, 5 Mar 1997 22:16:10 -0500
From: Keith Chow <keith@nal.utoronto.ca>
Organization: University of Toronto
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To: rem-conf@es.net
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Content-Type: text/plain; charset=us-ascii
Content-Transfer-Encoding: 7bit

Hi there,

I have a problem making vic-2.8 on Solaris 2.5.1 as follows:

ld: fatal: symbol `TkPlatformInit' is multiply defined:
        (file main.o and file /usr/local/lib/libtk4.1.a(tkUnixInit.o));
ld: warning: symbol `random' has differing types:
        (file random.o type=NOTY; file /usr/lib/libc.so type=FUNC);
        random.o definition taken
ld: fatal: File processing errors.  No output written to vic
gmake: *** [vic] Error 1

Can anybody give me a help?

Thanks in advance.

Keith.

-- 

-----------------------------------------------
Keith HungKei Chow
Network Architecture Lab, Communication Group,
Department of Electrical & Computer Engineering
University of Toronto
Ontario, Canada.
Tel: (416) 978-1611
Fax: (416) 978-4425
Email: keith@nal.utoronto.ca
       keith@comm.utoronto.ca
-----------------------------------------------

From rem-conf-request@es.net Thu Mar 06 00:58:49 1997 
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Date: Wed, 5 Mar 1997 21:54:48 -0800 (PST)
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To: Eric Fleischman <ericfl@MICROSOFT.com>
From: Ross Finlayson <finlayson@lvn.com>
Subject: Re: In Defense of Interleaved Data Formats
Cc: "'confctrl@isi.edu'" <confctrl@isi.edu>, 
    "'rem-conf@es.net'" <rem-conf@es.net>

At 03:05 PM 3/5/97 -0800, Eric Fleischman wrote:
>4.	Disk-head movement efficiency on the server. The argument here
>is that interleaving encourages more efficient disk storage approaches
>with superior I/O processing capabilities. This directly effects the
>construction of more efficient network streaming approaches. This
>efficiency directly translates into server performance including the
>ability to support a larger number of simultaneous "On-demand" sessions
>by that server.

Let's not forget that we're designing network protocols here, not storage
formats.  There's nothing to preclude the various media that make up a
multimedia session being interleaved on the *disk*, even though the data
within each of the resulting network packets is not interleaved.  (In fact,
one can even imagine a media server storing its data with partially
precomputed network headers, to optimize the process of spitting its data
out onto the network.)

Incidentally, this confusion - between storage formats and network data
formats - is one that I've seen recently occur in other IETF working groups,
e.g., the "calendar scheduling" group.  Microsoft (& other companies) are
free to continue to define and use whatever storage formats they wish.  We
don't care; this is the *Internet* Engineering Task Force.  All we care
about is what goes over the network.

In your follow-up message, you wrote:

>It is doubtful that you will be able to understand what I'm talking
>about as long as you think as an engineer. I am talking "experience" not
>"information". This is a left-brain, right-brain type of dichotomy. To
>understand what I'm saying, you have to think as an artist. What we are
>talking about is the preservation of an artistic creation across the
>network. That this creation may educate or convey critical, essential
>information does not diminish the fact that we are talking about
>art-forms when we are talking about "on demand" streaming.

I'm sorry to sound rude, but this "touchy-feely" stuff makes me want to
vomit.  This is the Internet *Engineering* Task Force, and *engineering* is
what we do here.  Fortunately, there *are* some real engineering questions
that arises from your concerns, namely:

You want it to be possible to implement one of the following end-to-end
semantics when confronted with network loss or delay:
        1/ cancel the presentation
        2/ delay the presentation (as appropriate)
        3/ continue the presentation in spite of missing data
The questions are:
a) Can the particular choice (between 1, 2 and 3) be adequately conveyed -
e.g., in a session description - if different media are *not* interleaved
within network packets.
b) Can each of these choices be implemented (efficiently), if different
media are *not* interleaved within network packets.

We should, I think, be able to answer these questions without having to
stick our heads inside a pyramid.

        Ross.


From rem-conf-request@es.net Thu Mar 06 03:15:40 1997 
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From: Karl Auerbach <karl@precept.com>
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Subject: RE: In Defense of Interleaved Data Formats
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You are arguing that interleaved representations are necessary to
preserve "the multimedia experience".  And further that the ability of
a non-interleaved, packet-per-media type approach to suppress one of
the media types of the presentation isn't useful because it would
destroy "the multimedia experience".

You seem to be saying that it is better to simply stop the
presentation, in total, rather than try to carry any less than perfect
rendition.

I don't agree.

The choice depends on the type of presentation.

When reality strikes, we will notice that a lot of presentations
happen to be audio and video presentations or conferences (often with
less than stellar camera work or microphone mixing -- hardly an
artistic creation.)

And our own experience has tought us that there are techniques, such
as layered encodings and the like, which can preserve the
intelligibility of the interchange at the cost of some audio quality
or even the video itself.

Interleaved streams will prevent us from making use of those
techniques on those sessions where quality degredation is not only
acceptable, but is desiragle to preserve the information content.

And on those presentations in which one wants to stop the data flow or
postpone it, whether the data is interleaved or not really makes no
difference.  It's just as easy to tell a server to stop sending two
separate RTP streams as it is to tell it to stop one.

And as someone mentioned, the argument for interleaving avoids dealing
with the fact that inside a receiving computer, the audio and video
paths are very different and have different time delays.  So the data
streams will be separated and independently timed for rendition no
matter whether they arrive in the same or different packets.

So even if I were to agree with you that total quality must not be
sacrificed, I still don't agree with you that that leads to the conclusion
that one must have interleaved streams.

> It is doubtful that you will be able to understand what I'm talking
> about as long as you think as an engineer.

Speaking as one who spends a lot of my time doing live theatre, I
wouldn't be so quick to deprecate the ability of engineers to
comprehend aesthetic issues.

OK, I'm sitting here pretending that I'm on-stage looking into the
lights and hearing the rustle of the audience while I'm waiting for
the music cue... .... .... Whew, that was one artistic experience!
... And I still don't agree with you.

> ... Now you will perhaps
> argue that streaming content can't be carried over connection-oriented
> transports like the TCP or HTTP such as were mentioned in the previous
> sentence. Experience has demonstrated that this observation may very
> well be true for segregated media streams, however, it is emphatically
> *not* true for interleaved media streams (given a certain preroll
> value). 

Of course "streaming content" can be carried over connection-oriented
transports like TCP.  Just not in real-time.

TCP engines are designed with various algorithms to avoid adding to
network congestion.  As such they exhibit potentially long, possibly
difficult to bound, delays.  Unless you know the characteristics of
the TCP engines involved, you can't come up with a completely safe
"preroll" value.  There is always a chance that you will underrun.

Your comments about firewalls are well taken.  However, I have faith
that the firewall folks can figure out how to deal with UDP based
traffic.

			--karl--






From rem-conf-request@es.net Thu Mar 06 03:25:18 1997 
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To: Mark Handley <mjh@isi.edu>
cc: Eric Fleischman <ericfl@MICROSOFT.com>, 
    "'confctrl@isi.edu'" <confctrl@isi.edu>, 
    "'rem-conf@es.net'" <rem-conf@es.net>
Subject: Re: In Defense of Interleaved Data Formats
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From: Amancio Hasty <hasty@rah.star-gate.com>

>From The Desk Of Mark Handley :
> >4.	Disk-head movement efficiency on the server. The argument here
> >is that interleaving encourages more efficient disk storage approaches
> >with superior I/O processing capabilities. This directly effects the
> >construction of more efficient network streaming approaches.
> 
> Separate streams lets you store your audio and video on different
> discs, and hence *improves* performance in some cases :-)

Well, sort of ...

If you are going to have a server I would imagine that one can do software or
hardware disk stripping , if such is the case then perhaps an interleave
media stream is more efficient than two different media streams.
If one media stream requires more bandwidth than the other then it
may be nice to have separate streams for a better system utilization
by transmitting the prioritized stream. I would imagine that the best
way of handling interleaving vs. separate media streams is to leave 
the choice to the tool and not mandate it as a protocol procedure.

	Cheers,
	Amancio

	



From rem-conf-request@es.net Thu Mar 06 04:16:15 1997 
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To: Ross Finlayson <finlayson@lvn.com>
cc: Eric Fleischman <ericfl@MICROSOFT.com>, 
    "'confctrl@isi.edu'" <confctrl@isi.edu>, 
    "'rem-conf@es.net'" <rem-conf@es.net>
Subject: Re: In Defense of Interleaved Data Formats
In-reply-to: Your message of "Wed, 05 Mar 1997 21:54:48 PST." <1.5.4.16.19970305223954.0a076550@pop.best.com>
Date: Thu, 06 Mar 1997 09:13:44 +0000
Message-ID: <1148.857639624@cs.ucl.ac.uk>
From: Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>



 >You want it to be possible to implement one of the following end-to-end
 >semantics when confronted with network loss or delay:
 >        1/ cancel the presentation
 >        2/ delay the presentation (as appropriate)
 >        3/ continue the presentation in spite of missing data
 >The questions are:
 >a) Can the particular choice (between 1, 2 and 3) be adequately conveyed -
 >e.g., in a session description - if different media are *not* interleaved
 >within network packets.
 >b) Can each of these choices be implemented (efficiently), if different
 >media are *not* interleaved within network packets.
 
yes, well said...

actually, what interleaving says is :

It is a REQUIREMENT of this type of presentation that all media suffer
network effects exactly in proportion to the way they are interleaved....

since this is pretty arbitrary compared with the way the QUALITY of
the presentation will then vary (e.g. consider a variation in rate or loss
for video versus a variance in audio for a given relative sample size
in the packet), i suspect most content providers would be fairly upset
by the way such a presentaiton would degrade......

interleaving: lets see now, what was the end2end principle again?

 >We should, I think, be able to answer these questions without having to
 >stick our heads inside a pyramid.

:-)

 jon


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Scott Petrack
03/06/97 02:52 PM

I would like to understand if there is any issue here which actually
affects the specification of RTP or requires the intervention of AVT in any
way. I claim that there is not.

If the issue is merely "hey guys, interleaved formats are better. Really"
then I
think that the thread should be stopped, because I know of no horse in the
entire AVT/MMUSIC realm which is more dead.

Despite the use of the words "interleaving", there is no real desire to
interleave
currently defined RTP audio and RTP video packets within a single RTP
session.
What is desired is to have a single RTP session which contains packet after
packet
of new special beautifully crafted payloads.

As far as the RTP spec is concerned, all that you need to
do is to write an "RTP profile for the XXYYZZ media type". This will
satisfy the
RTP spec (that is, it will allow you to say "our server confirms to the
Internet
standard") and the only penalty will be that it will be "disallowed" to mix
packets of "audio" or "video" media type with packets of the new XXYYZZ
media types. But you weren't planning to do this anyway.

To put it most simply - You don't need a change in the RTP spec to do what
you
want to do. You need to go and write a new profile and payload format.

No amount of dickey-ing with the words in the spec
will alter the fact that what you do will not be interoperable with
applications which conform to the current profile for basic audio and
video. This
is not due to the lack of artistic sentiment on the part of the application
writers - it is due to the fact that you have yet to invent the new media
types, define payload formats and a profile for the RTP, and share this
information with the rest of us.

Once this is done, who knows? Maybe you'll convince people that your
profile is more suitable for a particular purpose. Maybe.

Until that time, I'll have to remind you that:

"Wer sich der Einsamkeit ergibt,
Ach! der is bald allein."

(sorry, I was carried away by artistic inspiration)

Scott




From rem-conf-request@es.net Thu Mar 06 08:06:05 1997 
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From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
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It should be pointed out that we had this same discussion many moons
ago, on the rem-conf list, at least once. Note that there is one
interleaved MPEG format (Civanlar) already.
-- 
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 Mar 06 11:22:36 1997 
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From: Eric Fleischman <ericfl@MICROSOFT.com>
To: "'Scott_Petrack@vocaltec.com'" <Scott_Petrack@vocaltec.com>, 
    "'rem-conf@es.net'" <rem-conf@es.net>
Subject: RE: In Defense of Interleaved Data Formats
Date: Thu, 6 Mar 1997 08:17:46 -0800
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Thank you, Scott, for your posting.

I am not interested in amending the text of RFC 1889. Rather, I am 
interested in amending the text of the RTSP draft and hence will pursue 
this interest within mmusic and not bother avt further.

My goal in bothering avt to date was to determine the sentiment within avt 
concerning a two prong use of RTP within the auspices of RTSP:
1)	Short term stream interleaved content over RTP via using a dynamic 
payload type established for that session via an RTSP "get" command.
2)	Medium/Long term define a generic RTP profile and payload format for a 
class of streaming interleaved data.

Your response satisfies both of these goals. Therefore, if it represents 
the generic opinion of the avt group then I will be most content to return 
to silence and not trouble you further.

Thanks for the poetry. Let's hope that you "wax artistic" frequently.

BTW: A 6th argument to support interleaved data is that most existing 
multimedia data file formats (solely) support interleaved data if multiple 
media types are present. Thus, if one is to directly stream data from those 
files the most efficient way is to do so via interleaved data streams.

-----Original Message-----
From:	Scott_Petrack@vocaltec.com [SMTP:Scott_Petrack@vocaltec.com]
Sent:	Thursday, March 06, 1997 4:53 AM
To:	rem-conf@es.net
Subject:	RE: In Defense of Interleaved Data Formats






Scott Petrack
03/06/97 02:52 PM

I would like to understand if there is any issue here which actually
affects the specification of RTP or requires the intervention of AVT in 
any
way. I claim that there is not.

If the issue is merely "hey guys, interleaved formats are better. Really"
then I
think that the thread should be stopped, because I know of no horse in the
entire AVT/MMUSIC realm which is more dead.

Despite the use of the words "interleaving", there is no real desire to
interleave
currently defined RTP audio and RTP video packets within a single RTP
session.
What is desired is to have a single RTP session which contains packet 
after
packet
of new special beautifully crafted payloads.

As far as the RTP spec is concerned, all that you need to
do is to write an "RTP profile for the XXYYZZ media type". This will
satisfy the
RTP spec (that is, it will allow you to say "our server confirms to the
Internet
standard") and the only penalty will be that it will be "disallowed" to 
mix
packets of "audio" or "video" media type with packets of the new XXYYZZ
media types. But you weren't planning to do this anyway.

To put it most simply - You don't need a change in the RTP spec to do what
you
want to do. You need to go and write a new profile and payload format.

No amount of dickey-ing with the words in the spec
will alter the fact that what you do will not be interoperable with
applications which conform to the current profile for basic audio and
video. This
is not due to the lack of artistic sentiment on the part of the 
application
writers - it is due to the fact that you have yet to invent the new media
types, define payload formats and a profile for the RTP, and share this
information with the rest of us.

Once this is done, who knows? Maybe you'll convince people that your
profile is more suitable for a particular purpose. Maybe.

Until that time, I'll have to remind you that:

"Wer sich der Einsamkeit ergibt,
Ach! der is bald allein."

(sorry, I was carried away by artistic inspiration)

Scott





From rem-conf-request@es.net Thu Mar 06 11:44:06 1997 
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From: Eric Fleischman <ericfl@MICROSOFT.com>
To: 'Jon Crowcroft' <J.Crowcroft@cs.ucl.ac.uk>, 
    Ross Finlayson <finlayson@lvn.com>
Cc: "'confctrl@isi.edu'" <confctrl@isi.edu>, 
    "'rem-conf@es.net'" <rem-conf@es.net>
Subject: RE: In Defense of Interleaved Data Formats
Date: Thu, 6 Mar 1997 08:36:33 -0800
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Your point
		>It is a REQUIREMENT of this type of presentation that
all media suffer
		>network effects exactly in proportion to the way they
are interleaved....
is right on the mark. 

However, your follow-on comment isn't. Because "on demand" content isn't
constructed in "real time", we have been able to give attention to
implementing error mitigation (error discovery, error hiding, error
correction) approaches within our data streams. We could certainly
benefit from your considerable expertise in improving our current
implementation of forward error correction. Might you be interested in
pursuing this issue "off line" with us?

From rem-conf-request@es.net Thu Mar 06 12:12:22 1997 
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From: Michael.Speer@Eng.Sun.COM (Michael F. Speer)
Message-Id: <199703061711.JAA16436@valathar.eng.sun.com>
To: rem-conf@es.net
Subject: Sunergy #25 Broadcast
Cc: Michael.Speer@Eng.Sun.COM


Folks:

Due some technical problems, I will be arranging to re-broadcast this
Sunergy for late next week.  Sorry for the technical difficulties.

Michael

From rem-conf-request@es.net Thu Mar 06 18:56:14 1997 
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To: Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>
cc: rem-conf@es.net
Subject: Re: socks/mbone firewalls
In-reply-to: Your message of "Wed, 05 Mar 1997 17:47:26 GMT." <2466.857584046@cs.ucl.ac.uk>
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From: George Michaelson <ggm@connect.com.au>


Didn't this get discussed on the list about 3 years ago?  a well archived
copy might find some of this. Maybe it took place live on the mbone and not
via email (another reason audio is at a disadvantage compared to the written
word)

Bellovin & Cheswick has about 10 references in ther index BTW.

-George



From rem-conf-request@es.net Thu Mar 06 19:12:16 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 March 12, 1997 12:30-2:00 PDT 405 Soda Hall) 

  "Interactive Commerce: Technology Solutions for Business on the Web" 

                               Pia Chamberlain 
                                Connect, Inc. 

Conducting business on the web generates challenges not encountered by
static content providers.  OneServer(tm), an internet commerce application
technology platform offered by Connect, Inc., provides a comprehensive,
integrated solution set to address these challenges, including transacted
data access, integrated text search, an object-relational database back-end,
usage tracking, Java integration, and sophisticated access control
capabilities. I will present an architectural overview of OneServer(tm),
briefly discussing how its capabilities support the creation of high-end
internet commerce applications. 

Connect, Inc. has openings for CS/CE graduates for systems, application and
quality assurance engineering positions. Please send resumes to Pia Chamberlain.
--------------------------------------------------------------------------------
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 Fri Mar 07 10:49:50 1997 
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From: Matthew Feldman <mfeldman@VideoConferencing.com>
To: "'rem-conf@es.net'" <rem-conf@es.net>
Subject: IETF e-mails
Date: Fri, 7 Mar 1997 10:48:41 -0500
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Why I am always getting  copies of the same e-mail constantly.  Is there a 
problem with the rem-conf e-mail reflector?  Can someone fix this problem? 
 Is there a problem?

Matthew Feldman
CTO
Intelect Visual Communications
(212) 317-9600




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From rem-conf-request@es.net Fri Mar 07 15:44:14 1997 
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          Fri, 7 Mar 1997 12:43:01 -0800
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Date: Fri, 7 Mar 1997 22:47:52 +0200 (EET)
From: Sampo Syreeni <decoy@nexus.edu.lahti.fi>
To: Mark Handley <mjh@isi.edu>
cc: Eric Fleischman <ericfl@MICROSOFT.com>, 
    "'confctrl@isi.edu'" <confctrl@isi.edu>, 
    "'rem-conf@es.net'" <rem-conf@es.net>
Subject: Re: In Defense of Interleaved Data Formats
In-Reply-To: <8946.857608578@buttle.lcs.mit.edu>
Message-ID: <Pine.LNX.3.95.970307223552.30188C-100000@nexus.edu.lahti.fi>
MIME-Version: 1.0
Content-Type: TEXT/PLAIN; charset=US-ASCII

On Wed, 5 Mar 1997, Mark Handley wrote:

> >However, there are many occasions when
> >"On-Demand Streaming" requires the use of interleaved data formats.
> >Here's a few:
> 
> >1.	You can save substantial amount of overhead for sessions
> >accessed by low speed modems.
> ...
> >(i.e., on-the-fly compression) to process your "on demand" application
> >then the number of sessions you can simultaneously support on that
> >server is detrimentally impacted.
> 
> Note that header compression is not end-to-end, but only between your
> end-system and its local terminal server.  It doesn't impact 
> multimedia data servers at all.

But I assume some form can be implemented here too, if found advantageous?

> >2.  ... Interleaving encourages tight synchronization and offers a
> >higher degree of probability for preserving the designed experience in
> >the presence of data loss, network variability, and sunspots.
> 
> Actually this isn't true.  Interleaving gives the network less
> oportunity to preserve your audio preferentially from your video if
> the network gets loaded (which is usually the right thing to do).

But gives a tighter synch with less buffering. And eases up the
requirements for throttle-back in RTP. It also gives the server a more
predictable data stream to work on.

> In addition, synchronisation is no easier.  You have all the same
> information in separate RTP streams that you have in interleaved
> streams.  In either case you need to cope with mismatches in clock
> rate between the video stream and a receiving audio device that which
> is not clocked at exactly the frequency the manufacturer claims.  Once
> you're re-tuning the audio buffering periodically, it makes no
> difference whether the streams were transported together or
> separately.

How about the case when audio arrives slower than the corresponding 
video? In this case either sizable buffers, fast throttle-back or
frame-discarding abilities are required.

> >3.	On-demand media may be composed of many media types. Some of
> >these are "streaming" media types (e.g., voice, video, and MIDI) while
> >others are not (e.g., still images, URL flippings, script commands and
> >executables). "On demand uses" also encourage the innovative invention
> >of rare media types. Regardless, the net effect is that there will be N
> >media streams for on-demand usages where N may be quite a handful.
> 
> This seems to be an argument *against* interleaving to me...

Hmm. Keep a separate state for each of the streams/connections and you
have quite a bit of data. So interleaving might ease up the requirements
imposed upon lower protocol layers. How about routing considerations, for 
example?

> >4.	Disk-head movement efficiency on the server. The argument here
> >is that interleaving encourages more efficient disk storage approaches
> >with superior I/O processing capabilities. This directly effects the
> >construction of more efficient network streaming approaches.
> 
> Separate streams lets you store your audio and video on different
> discs, and hence *improves* performance in some cases :-)

But how about the basic case where you do  have both the streams
originating from the same server?

> Besides, if your server isn't doing some form of tuning to available
> bandwidth, it's going to be a dinosaur in the long run.

Very true. That is a good argument against interleaving. But I'd say at
the present BW adaptation isn't the norm. What we need now is efficiency
>from simple servers. But in the future multiple independent data streams
will probably be the norm. At least if stream aggregation doesn't take
place in the lower levels...

Sampo Syreeni (Decoy/dAWN), student, <decoy@edu.lahti.fi>


From rem-conf-request@es.net Fri Mar 07 16:00:10 1997 
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Date: Fri, 7 Mar 1997 23:04:23 +0200 (EET)
From: Sampo Syreeni <decoy@nexus.edu.lahti.fi>
To: Eric Fleischman <ericfl@MICROSOFT.com>
cc: 'Mark Handley' <mjh@isi.edu>, "'confctrl@isi.edu'" <confctrl@isi.edu>, 
    "'rem-conf@es.net'" <rem-conf@es.net>
Subject: RE: In Defense of Interleaved Data Formats
In-Reply-To: <503A2A3C2932CF118D8800805FD44E1802BBD79F@RED-68-MSG>
Message-ID: <Pine.LNX.3.95.970307225504.30188E-100000@nexus.edu.lahti.fi>
MIME-Version: 1.0
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On Wed, 5 Mar 1997, Eric Fleischman wrote:

> usage. This requirement is that the "multimedia experience" is preserved
> independently of what is happening on the network. Thus, if the network
> becomes saturated, the worst thing to do would be to drop a given media
> stream in order to preserve another stream based on network assumptions
> as to which stream may be "more important" (i.e., your suggestion to
> drop video but to preserve audio in a congested situation).

Indeed. This brings out the most basic difference between hand-crafted
multimedia 'experiences' and the needs of real-time conferencing. The
first carries artistic value as a whole, the second is primarily a means
of communication. As such, in the first case one needs to preserve the
overall experience whereas in the second case getting the message
delivered is of prime importance.

> "multimedia".  While interesting experiences are indeed created using
> only those two media types, I would prefer you to think of audio + video
> + MIDI + URL flipping + streaming text + active background applications
> like stock tickers + still images + ....   Why limit ourselves to only
> two media types? Certainly the people making streaming "on-demand"
> content don't.

Because they're the hardest to handle due to their high resource
requirements and the relatively high priority they have among our
perceptual channels, probably. When you can do both audio and video, you
can certainly do the other things you refer to.

> sentence. Experience has demonstrated that this observation may very
> well be true for segregated media streams, however, it is emphatically
> *not* true for interleaved media streams (given a certain preroll
> value). 

But it is true that as long as you have some minimum guarantee of
maximum latency between the different data streams, you can buffer
multiple streams as well as you can interleaved ones.

Sampo Syreeni (Decoy/dAWN), student, <decoy@edu.lahti.fi>


From rem-conf-request@es.net Fri Mar 07 16:12:08 1997 
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Date: Fri, 7 Mar 1997 23:16:09 +0200 (EET)
From: Sampo Syreeni <decoy@nexus.edu.lahti.fi>
To: Ross Finlayson <finlayson@lvn.com>
cc: Eric Fleischman <ericfl@MICROSOFT.com>, 
    "'confctrl@isi.edu'" <confctrl@isi.edu>, 
    "'rem-conf@es.net'" <rem-conf@es.net>
Subject: Re: In Defense of Interleaved Data Formats
In-Reply-To: <1.5.4.16.19970305223954.0a076550@pop.best.com>
Message-ID: <Pine.LNX.3.95.970307230659.30188F-100000@nexus.edu.lahti.fi>
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On Wed, 5 Mar 1997, Ross Finlayson wrote:

> >4.	Disk-head movement efficiency on the server. The argument here
[snip]
> 
> Incidentally, this confusion - between storage formats and network data
> formats - is one that I've seen recently occur in other IETF working groups,
> e.g., the "calendar scheduling" group.

This is a direct consequence of trying to make things simple: the most
basic server conceivable is the one that just spits out the data it has.
(Say, HTTP...)

> You want it to be possible to implement one of the following end-to-end
> semantics when confronted with network loss or delay:
>         1/ cancel the presentation
>         2/ delay the presentation (as appropriate)

1-2 seem partially equivalent. Just as in the PAUSE issue.

> a) Can the particular choice (between 1, 2 and 3) be adequately conveyed -
> e.g., in a session description - if different media are *not* interleaved
> within network packets.

The easiest way to go would be to add a single word (such as those
'encrypted' and other pseudo terms the current draft uses) to indicate the
type for each track (session?).

> b) Can each of these choices be implemented (efficiently), if different
> media are *not* interleaved within network packets.

When fighting congestion, one shouldn't assume the client can do anything
extra. So it's all about server throttle-back. So these semantics need to
conveyed to the server when initiating a stream. This approach isn't
dependent on whether the data is interleaved or not.

Sampo Syreeni (Decoy/dAWN), student, <decoy@edu.lahti.fi>


From rem-conf-request@es.net Fri Mar 07 16:14:26 1997 
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Subject: Re: In Defense of Interleaved Data Formats
In-reply-to: Your message of "Fri, 07 Mar 1997 23:04:23 +0200." <Pine.LNX.3.95.970307225504.30188E-100000@nexus.edu.lahti.fi>
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>On Wed, 5 Mar 1997, Eric Fleischman wrote:
>
>> usage. This requirement is that the "multimedia experience" is preserved
>> independently of what is happening on the network. Thus, if the network
>> becomes saturated, the worst thing to do would be to drop a given media
>> stream in order to preserve another stream based on network assumptions
>> as to which stream may be "more important" (i.e., your suggestion to
>> drop video but to preserve audio in a congested situation).
>
>Indeed. This brings out the most basic difference between hand-crafted
>multimedia 'experiences' and the needs of real-time conferencing. The
>first carries artistic value as a whole, the second is primarily a means
>of communication. As such, in the first case one needs to preserve the
>overall experience whereas in the second case getting the message
>delivered is of prime importance.

I believe they're not so different.  If you need *no* degradation,
you'd better be prepared to reserve network resources, and to pay for
them - in this case interleaving gives no advantage over separate
streams. 

If you're not prepared or able to do that, some degree of degradation
is very likely, and separate streams at least allow some degree of
control over that degradation process - remember packet loss rate does
not correspond directly to quality degradation when you compare
different media.

Separate streams cleanly separate policy for dealing with degradation
>from mechanism.  Interleaving does not (especially with multicast).

Summary: in the best case there's no difference (except for the packet
header argument).  In typical or worst case, interleaving is a bad
idea.

Mark

From rem-conf-request@es.net Fri Mar 07 17:27:04 1997 
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From: Sampo Syreeni <decoy@nexus.edu.lahti.fi>
To: Mark Handley <mjh@isi.edu>
cc: Eric Fleischman <ericfl@MICROSOFT.com>, 
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Subject: Re: In Defense of Interleaved Data Formats
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On Fri, 7 Mar 1997, Mark Handley wrote:

> >> usage. This requirement is that the "multimedia experience" is preserved
> >> independently of what is happening on the network. Thus, if the network
> >> becomes saturated, the worst thing to do would be to drop a given media
> >> stream in order to preserve another stream based on network assumptions
> >> as to which stream may be "more important" (i.e., your suggestion to
> >> drop video but to preserve audio in a congested situation).
> >
> >Indeed. This brings out the most basic difference between hand-crafted
> >multimedia 'experiences' and the needs of real-time conferencing. The
> >first carries artistic value as a whole, the second is primarily a means
> >of communication. As such, in the first case one needs to preserve the
> >overall experience whereas in the second case getting the message
> >delivered is of prime importance.
> 
> I believe they're not so different.  If you need *no* degradation,
> you'd better be prepared to reserve network resources, and to pay for
> them - in this case interleaving gives no advantage over separate
> streams. 

How about when no guaranteed QoS is available (e.g. you're on an
Ethernet)? In this case dealing with real-time streams is a pain in the
ass. An easy, ready-made presentation would suffice, and therefore
interleave is useful.

> If you're not prepared or able to do that, some degree of degradation
> is very likely, and separate streams at least allow some degree of
> control over that degradation process - remember packet loss rate does
> not correspond directly to quality degradation when you compare
> different media.

Indeed. But always having to do multiple streams is a true problem for
simple clients. I think I'm taking the position that interlaced data
streams as a unit must be supported as much as simple streams do. They
truly are the most common ones today. And expect nothing less when digital
TV takes on...

> Separate streams cleanly separate policy for dealing with degradation
> from mechanism.  Interleaving does not (especially with multicast).

But when multicast isn't available?

> Summary: in the best case there's no difference (except for the packet
> header argument).  In typical or worst case, interleaving is a bad
> idea.

Explicit interleaving is. But if the original format of the data is
interleaved, it would be nice if it could be used as-is as just another
stream. Especially since there are interleaved protocols out there.

Sampo Syreeni (Decoy/dAWN), student, <decoy@edu.lahti.fi>


From rem-conf-request@es.net Fri Mar 07 17:44:44 1997 
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Subject: Re: In Defense of Interleaved Data Formats
In-reply-to: Your message of "Sat, 08 Mar 1997 00:30:56 +0200." <Pine.LNX.3.95.970308002451.30188T-100000@nexus.edu.lahti.fi>
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>Indeed. But always having to do multiple streams is a true problem for
>simple clients. I think I'm taking the position that interlaced data
>streams as a unit must be supported as much as simple streams do. They
>truly are the most common ones today. And expect nothing less when digital
>TV takes on...

No.  

An interleaved client has to de-interleave the streams and timestamp
them, then decode them (different media have different decode delays)
then buffer to perform playout adjustment for audio-device clock skew,
different decode times, network jitter and audio device buffer
latencies, then actually play the media streams out.

A non-interleaved client does the same thing, except the streams
arrive de-interleaved and pre-timestamped.

In either model, you can move the playout buffering around somewhat,
but it doesn't change the general principle.


Interleaved streams are common because switched circuits are.  We
don't have that constraint.

>But if the original format of the data is
>interleaved, it would be nice if it could be used as-is as just another
>stream. Especially since there are interleaved protocols out there.

Depends if the original format was designed to cope with packet loss.
H.221 for example is not, and takes many seconds to recover after a
packet loss.  It's consituent H.261 and audio codings can however be
packetised to cope quite well with loss.  Just because your data
originated interleaved doesn't mean it's a good idea to transmit it
that way.

Mark

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From: Sampo Syreeni <decoy@nexus.edu.lahti.fi>
To: Mark Handley <mjh@isi.edu>
cc: Eric Fleischman <ericfl@MICROSOFT.com>, 
    "'rem-conf@es.net'" <rem-conf@es.net>
Subject: Re: In Defense of Interleaved Data Formats
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On Fri, 7 Mar 1997, Mark Handley wrote:

> >Indeed. But always having to do multiple streams is a true problem for
> >simple clients. I think I'm taking the position that interlaced data
> >streams as a unit must be supported as much as simple streams do. They
> >truly are the most common ones today. And expect nothing less when digital
> >TV takes on...
> 
> No.  
> 
> An interleaved client has to de-interleave the streams and timestamp
> them, then decode them (different media have different decode delays)
> then buffer to perform playout adjustment for audio-device clock skew,
> different decode times, network jitter and audio device buffer
> latencies, then actually play the media streams out.

But the code exists. As does the experience. As do ready-made data.

> A non-interleaved client does the same thing, except the streams
> arrive de-interleaved and pre-timestamped.

But you'll have to manage more streams. Handling all the transport
protocols isn't very nice.

> Interleaved streams are common because switched circuits are.  We
> don't have that constraint.

Assume you own some links and can use policy routing. Then using
interlaced data with QoS and your own links will make things a lot easier.
With a lot less protocol hassles.

> >But if the original format of the data is
> >interleaved, it would be nice if it could be used as-is as just another
> >stream. Especially since there are interleaved protocols out there.
> 
> Depends if the original format was designed to cope with packet loss.
> H.221 for example is not, and takes many seconds to recover after a
> packet loss.  It's consituent H.261 and audio codings can however be
> packetised to cope quite well with loss.  Just because your data
> originated interleaved doesn't mean it's a good idea to transmit it
> that way.

I assume we have QoS service. Thus not much packet loss. And on-the-fly
conversion isn't going to do good to your servers' performance.

Sampo Syreeni (Decoy/dAWN), student, <decoy@edu.lahti.fi>


From rem-conf-request@es.net Sun Mar 09 04:47:14 1997 
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Date: Sun, 9 Mar 1997 01:44:00 -0800 (PST)
From: Stephen Casner <casner@precept.com>
To: "M. Reha Civanlar" <civanlar@research.att.com>
cc: rem-conf@es.net
Subject: Re: Problems with RTP Payload Types
In-Reply-To: <330E1103.2D28@research.att.com>
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On Fri, 21 Feb 1997, M. Reha Civanlar wrote:

> 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.

It may be sufficient for payload format specifications to be posted
as Internet Drafts and reviewed via email to the rem-conf and/or ietf
mailing lists without requiring a face-to-face AVT meeting.  There
are plenty of other examples, such as MIME type definitions, which
are managed by the IANA and RFC Editor processes subsequent to the
completion of the main MIME working group.  There could well be a
very specific, short-term working group formed to handle some future
payload format specification if it were complex enough to warrant
that.
							-- Steve


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From: Stephen Casner <casner@precept.com>
To: Bernard Aboba <aboba@internaut.com>
cc: "rem-conf@es.net" <rem-conf@es.net>
Subject: Re: Profiles
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Bernard,

> OK, I'll bite. The profile I would like to see would be:

Thanks for responding to my request for proposals.

> 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. 

Where would the bandwidth fraction be specified?  I have suggested an
alternate profile that specified no sending of RTCP, to be indicated
in SDP files by something like "RTP/AVU" (for unidirectional) where
now there is only one choice "RTP/AVP".  There could be more to
indicate other fractions, but this would likely be unwieldy.

Note that it is important for all participants to know what the
fraction is and use the same value as explained in section 6.2.1 or
thereabouts.  If it were a parameter communicated in RTCP, say by
"the" sender, this could be a problem on startup.

> 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). 

I agree that there were at least considerations requiring caution
with all of the ideas we discussed at the AVT meeting.

							-- Steve


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From: Stephen Casner <casner@precept.com>
To: rem-conf@es.net
Subject: AVT Meeting in Memphis
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In case you were curious, the Audio/Video Transport working group
will meet at the Memphis IETF.  AVT is not shown on the draft agenda
because the schedule for working groups in the Transport Area is
established separately as a "track" by the Area Directors.  Those WGs
are added to the agenda whenn the track schedule is completed.

I have requested two slots again since we have never failed to use
all the allotted time and even a few extra minutes.  I hereby solicit
requests for agenda slots.  I will be out this next week, but will
work on organizing the meeting agenda with your input when I return.

							-- Steve


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To: Video <rem-conf@es.net>
Date: Sun, 9 Mar 1997 08:10:55 -0500
Subject: Asking Advice on MPEG/MJPEG
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Would you be able to make any suggestions about the relative
merits of M-JPEG or MPEG, for either production or distribution
of video?

Chris Brown
chris@storm.ca
http://www.storm.ca/~chris

From rem-conf-request@es.net Mon Mar 10 15:20:26 1997 
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Subject: Re: Asking Advice on MPEG/MJPEG
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>Would you be able to make any suggestions about the relative
>merits of M-JPEG or MPEG, for either production or distribution
>of video?

I don't know about production, but for distribution it's a trade-off
(isn't everything!).

Both are DCT based, but M-JPEG compresses every frame completely
independently (intra-frame compression), and sends every block of
pixels in every frame.  This means it's high bandwidth, but resilient
to packet loss as damage is repaired by the next frame.

MPEG can use various levels of inter-frame compression (only coding
differences between frames).  This makes it considerably lower
bandwidth than M-JPEG for the same quality, but damage caused by
packet loss will propogate from frame to frame, and generally cause
much more visible artifacts.

H.261 is pretty similar to MPEG in how it performs its compression.
For typical internet use, what Steve McCanne did in vic was what they
call "robust H.261", where only intra-frame compression is performed,
but only those blocks that change more than some threshold are
actually sent.  This is a good point in the bandwidth/resilience
spectrum.  Whether you can get such a mode out of an off-the-shelf
MPEG encoder I do not know.

Mark

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From rem-conf-request@es.net Mon Mar 10 17:44:08 1997 
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Subject: Re: In Defense of Interleaved Data Formats
Cc: ericfl@microsoft.com, rem-conf@es.net


  *> 
  *> I believe they're not so different.  If you need *no* degradation,
  *> you'd better be prepared to reserve network resources, and to pay for
  *> them

Yes! yes!

Bob Braden


From rem-conf-request@es.net Mon Mar 10 22:10:41 1997 
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Subject: Re: Asking Advice on MPEG/MJPEG
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>>>>> "MH" == Mark Handley <mjh@isi.edu> writes:

 >> Would you be able to make any suggestions about the relative
 >> merits of M-JPEG or MPEG, for either production or distribution
 >> of video?

 MH> I don't know about production, but for distribution it's a trade-off
 MH> (isn't everything!).

 MH> Both are DCT based, but M-JPEG compresses every frame completely
 MH> independently (intra-frame compression), and sends every block of
 MH> pixels in every frame.  This means it's high bandwidth, but resilient
 MH> to packet loss as damage is repaired by the next frame.

JPEG and especially M-JPEG is much more interesting for packet based
transmissions or other environments with relatively high
loss/corruption of data.

The M-JPEG is a specific part of the JPEG standard in which you can
avoid sending some headers and tables (quantization and huffman
tables) for each picture, nothing really revolutionary.  

The current MPEG standards will not survive these environments of hi
loss very well, since the I frames (which holds the most quantity
information) is also most likely to be hit by loss, partly since they
are so large part of a transmission (relative the size of a P or B
frame) but also since it will cause an burst of packets when sent, and
this tends to fill buffers... and retransmissions is usually out of
question...

MPEG is however well suited for lowloss environments where you migth
get better quality out of the same (fixed) bandwidth. If you go for a
good quality of the shelf encoder/decoder box you migth get really
good quality over a considerable lower bandwidth than raw video.

MPEG is now being used in everday commercial TV transmission
links. However, this is normally not recieved as MPEG in the end.

 MH> MPEG can use various levels of inter-frame compression (only coding
 MH> differences between frames).  This makes it considerably lower
 MH> bandwidth than M-JPEG for the same quality, but damage caused by
 MH> packet loss will propogate from frame to frame, and generally cause
 MH> much more visible artifacts.

True. Also, there is many more parameters to tweak in MPEG to balance
the quality in the encoder. Propper setting of these is necessary if
you are at the bandwidth borders...

 MH> H.261 is pretty similar to MPEG in how it performs its compression.
 MH> For typical internet use, what Steve McCanne did in vic was what they
 MH> call "robust H.261", where only intra-frame compression is performed,
 MH> but only those blocks that change more than some threshold are
 MH> actually sent.  This is a good point in the bandwidth/resilience
 MH> spectrum.  Whether you can get such a mode out of an off-the-shelf
 MH> MPEG encoder I do not know.

The H.261 is much more restricted than the original MPEG 1 standard,
but is very usefull in the lower bandwidth ranges (it is originally
specified for up to 1920 kbps lines over primary rate ISDN type of
lines). The H.261 is so similar and the standard is so short, that you
can start of reading that H.261 spec and the walk over to the MPEG 1
standard.

One thing to keep in mind is that these schemes also have various
needs for CPU power and can also cause diffrent propagation
delays. Some of these things have to be kept in mind.

In both MPEG and H.261 you can run motion estimation and this is a
*very* expensive operation, there is smart approximations which gives
a really decent picture for much less CPU, but those have to be
exploited or you have to pay for the extra processing need.

Also, just simple prediction (no motion estimation) will cause an
extra processing need.

In MPEG and H.261 will I and P frames have about the same propagation
delay effect, but as soon as you get the B frames in there you will
need to delay frames so that the necessary reordering can be done
since in B frames is the prediction done both forward and backward in
time... since we can't use data that will come we need to delay the
signal so that the data is there before it is needed. The extra
propagation delay is normally quite OK in CD-ROM and TV link cases but
is certainly something unwanted in interactive situations.

So, there is no system which solves all problems... you have to look
at the specific problem in order to say what to do and what to use.

Regards,
Magnus

From rem-conf-request@es.net Tue Mar 11 06:22:32 1997 
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From: Andrey Bobyshev <bobyshev@x4u2.desy.de>
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       DESY is pleased to announce the MBONE broadcast
      -------------------------------------------------
    of CHEP'97 Plenary meetings from Berlin on April 7-11,1997.
    ----------------------------------------------------------

 Date: 7-11 April, 1997
 
 Time: 9:00 - 18:00 MET
  
  To get more information about CHEP97 conference program, please
 look at URL http://www.desy.de/chep97.
  
  The MBONE broadcast is announced via SDR tool as "CHEP'97 - Plenary
 Sessions from Berlin, Germany". For audio transmission we are going
 to use fphone 3.xx which is free of charge. Release 3.xx
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 fphone.

 Please find more information about fphone at URL
 http://www.inria.fr/rodeo/fphone .

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                   mbone@desy.de

 Best regards,
 			NOC of DESY.
 


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Title:       
	Donald Knuth "Graph Drawing from a User's Perspective"
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	Mar 18, 1997

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	Donald Knuth's 9/18/96 from Graph Drawing 96 at MSRI. No  abstract nor slides available. 









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	MBone Broadcast Announcement
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Title:       
	Donald Knuth "Some Research Problems for Combinatorialists"
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	Donald Knuth's 1/27/97 talk in the MSRI Combinatorics seminar.  No abstract nor slides available. 









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	John Conway's 3/10/97 talk at the MSRI Computational and Algorithmic Methods in 3-Dimensional Topology workshop.    How can we choose names for simple knots so that we can always  rebuild a knot from its name, and yet so that it will usually be easy  to find the name from any form of the knot? I shall describe a  simple arithmetical notation that accomplishes this for a large class  of knots.









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	During the CeBIT 97 there will be several MBone broadcast  with different scopes from the booth "Treffpunkt 22".   This session is used to check  the connectivity.  









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	MBone Broadcast Announcement
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	During the CeBIT 97 there will be several MBone broadcast  with different scopes from the booth "Treffpunkt 22".   This session is used to check  the connectivity.  









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

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From rem-conf-request@es.net Tue Mar 11 13:05:22 1997 
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From: Christian Maciocco <Christian_Maciocco@ccm.jf.intel.com>
Message-ID: <Tue, 11 Mar 97 10:02:58 PST_1@ccm.hf.intel.com>
To: rem-conf@es.net
Subject: Re[2]: Profiles

     >Note that it is important for all participants to know what the 
     >fraction is and use the same value as explained in section 6.2.1 or 
     >thereabouts.  If it were a parameter communicated in RTCP, say by 
     >"the" sender, this could be a problem on startup.
     
     As stated in the RFC, "The control traffic should be limited to a 
     small and known fraction of the session bandwidth,...".
     
     This might be a problem for adaptive applications which could start 
     their sessions in the 64 Kb/s range and reach 3 Mb/s or higher on an 
     Intranet when there is no congestion. I'd like the RTCP bandwidth to 
     get its share of the new session's bandwidth.
     
     Maybe the senders could advertise their new session's bandwidth 
     through an RTCP message when they go over a threshold, e.g. an order 
     of magnitude over what has been advertised for the session, so all 
     receivers use now a new constant. If multiple senders have reached the 
     threshold they'll learn through RTCP that the bandwidth has already 
     been advertized.
     
     Christian

From rem-conf-request@es.net Tue Mar 11 14:01:54 1997 
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Subject: [Reminder] Berkeley Multimedia & Graphics Seminar

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

  "Interactive Commerce: Technology Solutions for Business on the Web" 
                                
                                Pia Chamberlain 
                                 Connect, Inc. 

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


From rem-conf-request@es.net Tue Mar 11 15:06:27 1997 
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To: Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>
cc: rem-conf@es.net, kelly@tis.com, sterne@tis.com
Subject: Re: socks/mbone firewalls
In-reply-to: Your message of "Wed, 05 Mar 1997 17:47:26 GMT." <2466.857584046@cs.ucl.ac.uk>
Date: Tue, 11 Mar 1997 14:58:46 -0500
From: Kelly Djahandari <kelly@tis.com>

>in order to help some commercial intranet and ISP company managers
>feel happier about mbone deployment:
>
>can anyone point me at a discussion paper/description of firewall 
>usage, multicast UDP and mbone applications, and any rationale as to
>a) attacks (e.g. denial of service esp., but usual service breakins
>and disclosure etc obviously useful too)
>b) defenses
>c) procedures for mbone app<->firewall auth
>
>etc etc
>
>thanks
>jon

Jon,

We've written a paper describing work we did developing an MBone
proxy for the TIS Internet Firewall Toolkit.  It has some discussion of
security issues and the approach we took to reducing risk when bringing
multicast UDP datagrams into a firewall protected enclave, without
modifying MBone applications.  The paper is on our web site at 
http://www.tis.com/docs/research/network/mbone/mboneabs.html . 

Hope it helps,
Kelly

From rem-conf-request@es.net Tue Mar 11 15:34:22 1997 
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Since senders get to send more frequently than receivers and thus are
likely to be heard from reasonably often even in large groups, receivers
could simply base their session bandwidth estimate on the sender
reports. If packet loss is ignored, receivers could also monitor the
actual data bandwidth coming in, suitably smoothed. (This tends to cause
problems with on-off sources like voice, however.) SRs have the problem
that the rate during the last interval where a source was sending
doesn't get cleared (since the sender that stopped sending won't be
sending another SR). It also allows buggy or malicious implementations
to lie about the sending rate and cause everyone to send too many RTCP
packets.

Henning

From rem-conf-request@es.net Wed Mar 12 01:48:12 1997 
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Date: Wed, 12 Mar 1997 01:47:09 -0800
From: tsyed <tsyed@pc-comp-sol.ca>
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Subject: RTP
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X-Info: P&C Computer Solutions Inc. http://www.pc-comp-sol.ca

Hi

I am doing my Masters in Multimedia. I am looking an RTP kit on Window
NT environment for my University Lab. Is there any free or cheap RTP Kit
available which I can used for my project? 

Thanks




Tariq A syed

From rem-conf-request@es.net Wed Mar 12 14:18:44 1997 
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I would like to add a few good words for MPEG to MJPEG/MPEG discussions
posted before :-)

Mark Handley wrote:
> MPEG can use various levels of inter-frame compression (only coding
> differences between frames).  This makes it considerably lower
> bandwidth than M-JPEG for the same quality, but damage caused by
> packet loss will propogate from frame to frame, and generally cause
> much more visible artifacts.
> 

Some other advantages of MPEG for transmission include:
1. MQUANT: it is possible to change the quantization factor at the
macroblock level. This way bits can be deployed where they are needed
the most. So, an all intra MPEG (all I frames) can be more efficient
than MJPEG.
2. Slices: Slices have self identifying start codes and can be used to
partition a picture to decodable parts.

Although, some JPEG extensions include similar mechanisms, I have not
seen their implementations yet. All MPEG encoders, however, can
implement the above functionality.

> H.261 is pretty similar to MPEG in how it performs its compression.
> For typical internet use, what Steve McCanne did in vic was what they
> call "robust H.261", where only intra-frame compression is performed,
> but only those blocks that change more than some threshold are
> actually sent.  This is a good point in the bandwidth/resilience
> spectrum.  Whether you can get such a mode out of an off-the-shelf
> MPEG encoder I do not know.
> 

A mode close to this may be achieved by restricting the motion
estimation range to zero and using P frames, both of which are possible
in almost all off-the-shelf MPEG encoders. 

Magnus Danielson wrote:
> The current MPEG standards will not survive these environments of hi
> loss very well, since the I frames (which holds the most quantity
> information) is also most likely to be hit by loss, partly since they
> are so large part of a transmission (relative the size of a P or B
> frame) but also since it will cause an burst of packets when sent, and
> this tends to fill buffers... and retransmissions is usually out of
> question...

To fight with this problem, current MPEG-2 standard do allow using
I-slices (e.g., a bundle of 16 lines that are intra coded) in inter
coded pictures instead of using I frames.


-- 
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 Wed Mar 12 14:58:51 1997 
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To: rem-conf@es.net, 298@dogbert.CS.Berkeley.EDU
Subject: UCB Seminar Announcement
From: David Simpson <davesimp@cs.berkeley.edu>
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Due to problems that that I am still trying to track down, SAP announcements
do not appear to be leaving our local UCB MBone cloud.  The seminar is
still on today at the following addresses:

audio: 224.2.242.81/25378
video: 224.2.178.217/49780
wb: 224.2.167.198/46760

Sorry for the inconvience...
Here is the sdp file:

n=2149589067 2149589067 858195530 224.2.127.254 9875 127 trusted
v=0
o=davesimp 3067184308 3067184330 IN IP4 128.32.32.75
s=UCB Multimedia Seminar
i=UC Berkeley Multimedia Seminar.  This week's speaker is Pia Chamberlain from 
Connect, Inc., on ``Interactive Commerce: Technology Solutions for Business on 
the Web.''
u=http://bmrc.berkeley.edu/298/
e=David Simpson <davesimp@cs.berkeley.edu>
p=David Simpson +1-510-643-7106
c=IN IP4 224.2.242.81/127
t=3067187400 3069613800
r=604800 7200 0
a=tool:sdr v2.3a1
a=type:broadcast
m=audio 25378 RTP/AVP 0
c=IN IP4 224.2.242.81/127
a=ptime:40
m=video 49780 RTP/AVP 31
c=IN IP4 224.2.178.217/127
m=whiteboard 46760 udp wb
c=IN IP4 224.2.167.198/127


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



From rem-conf-request@es.net Wed Mar 12 17:28:13 1997 
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To: Kelly Djahandari <kelly@tis.com>
cc: rem-conf@es.net, sterne@tis.com
Subject: Re: socks/mbone firewalls
In-reply-to: Your message of "Tue, 11 Mar 1997 14:58:46 EST." <199703111958.OAA07014@clipper.hq.tis.com>
Date: Wed, 12 Mar 1997 17:54:11 +0000
Message-ID: <2546.858189251@cs.ucl.ac.uk>
From: Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>


 >We've written a paper describing work we did developing an MBone
 >proxy for the TIS Internet Firewall Toolkit.  It has some discussion of
 >security issues and the approach we took to reducing risk when bringing
 >multicast UDP datagrams into a firewall protected enclave, without
 >modifying MBone applications.  The paper is on our web site at 
 >http://www.tis.com/docs/research/network/mbone/mboneabs.html . 
 
 Kelly

ok - this is nice (at least its a good start!!)

2 problems
1/ you match multicast to unicast - you reasons are good, but many
intranets have better multicast than the internet at large, and it
seems a pitty not to map outside multicast to inside multicast, maybe
on "well known safe ports" and "different addresses"

2/ my program could do that:-)

3/ you could do this with secure sessions

4/ you could use SDR and SIP as tools to initiate the firewall access,
and setup a relay between the outside generalied port, and the inside
one (maybe, as i say on a different mcast addr/port pair for
safety...) - this would be neat, but might not fit quickly with
audited code...as per your toolkit and socks and so on....but its UDP,
so its a lot easier to think through and check than TCP (imho:-)

cheers

 jon


From rem-conf-request@es.net Thu Mar 13 14:51:29 1997 
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From: Jerrlyn Iwata <jerrlyn@postgres.Berkeley.EDU>
Subject: Berkeley Multimedia & Graphics Seminar


>
>                Berkeley Multimedia and Graphics Seminar 
>        (Wednesday March 19, 1997 12:30-2:00 PDT 405 Soda Hall) 
>
>        "The Prince and the Pauper, or, is content really King?" 
>
>                                John Taysom 
>                                Reuters, Ltd. 
>
>
>
>John Taysom, SVP with Reuters based in Palo Alto, will talk about business
models available to content providers on the Internet. Reuters is a supplier
of news to most prominent web sites and an investor in Yahoo, Infoseek and
Sportsline as well as several Internet infrastructure companies. 
>
>Reuters is a 150 year old news agency. Reuters operates one of the largest
private networks in the world providing real-time news, trading data and
transactional systems for the banking sector delivered over approximately
300,000 Pcs. This is in addition to being the premier world news agency for
text, photographs and video news. Reuters has been experimenting with the
public Internet as a distribution system for the last three years. 
>
>John Taysom has worked for Reuters for 13 years, living and working in the
Far East, the Middle East, Europe and now the U.S.A. John runs a corporate
venture fund for Reuters investing in companies whose technology or content
Reuters wishes to use or market internationally. 
>
>------------------------------------------------------------------------------
>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 Fri Mar 14 12:57:05 1997 
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To: Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>
cc: Kelly Djahandari <kelly@tis.com>, rem-conf@es.net, sterne@tis.com
Subject: Re: socks/mbone firewalls
In-reply-to: Your message of "Wed, 12 Mar 1997 17:54:11 GMT." <2546.858189251@cs.ucl.ac.uk>
Date: Fri, 14 Mar 1997 12:52:10 -0500
From: Kelly Djahandari <kelly@tis.com>


Jon,

  >ok - this is nice (at least its a good start!!)
  >
  >2 problems
  >1/ you match multicast to unicast - you reasons are good, but many
  >intranets have better multicast than the internet at large, and it
  >seems a pitty not to map outside multicast to inside multicast, maybe
  >on "well known safe ports" and "different addresses"

Yes - we discuss that as useful extension in the Control vs. 
Performance section.

  >2/ my program could do that:-)

Yes - we cite your monstermash program in the Related Work section.

  >3/ you could do this with secure sessions

Yes, but we needed to support safe participation in *insecure* 
public sessions.

  >4/ you could use SDR and SIP as tools to initiate the firewall access,
  >and setup a relay between the outside generalied port, and the inside
  >one (maybe, as i say on a different mcast addr/port pair for
  >safety...) - this would be neat, but might not fit quickly with
  >audited code...as per your toolkit and socks and so on....but its UDP,
  >so its a lot easier to think through and check than TCP (imho:-)

This would be neat.  However, we wanted the design of the proxy to be
consistent with the rest of the fwtk, which is strongly oriented toward
TCP.

Kelly

From rem-conf-request@es.net Fri Mar 14 15:00:43 1997 
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From: Mark Handley <mjh@isi.edu>
X-Organisation: Information Sciences Institute, USC
X-Phone: +1 617 253 6011
To: Kelly Djahandari <kelly@tis.com>
cc: rem-conf@es.net
Subject: Re: socks/mbone firewalls
In-reply-to: Your message of "Fri, 14 Mar 1997 12:52:10 EST." <199703141752.MAA25187@clipper.hq.tis.com>
Date: Fri, 14 Mar 1997 14:58:20 -0500
Message-ID: <17587.858369500@buttle.lcs.mit.edu>
Sender: mjh@buttle.lcs.mit.edu


>  >4/ you could use SDR and SIP as tools to initiate the firewall access,
>  >and setup a relay between the outside generalied port, and the inside
>  >one (maybe, as i say on a different mcast addr/port pair for
>  >safety...) - this would be neat, but might not fit quickly with
>  >audited code...as per your toolkit and socks and so on....but its UDP,
>  >so its a lot easier to think through and check than TCP (imho:-)
>
>This would be neat.  However, we wanted the design of the proxy to be
>consistent with the rest of the fwtk, which is strongly oriented toward
>TCP.

Note that the current SIP spec allows both UDP and TCP access.

Mark

From rem-conf-request@es.net Fri Mar 14 15:46:09 1997 
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From: Frank Troy <ftroy@explorer.csc.com>
To: rem-conf@es.net
Date: Fri, 14 Mar 1997 15:42:50 +0000
MIME-Version: 1.0
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Subject: sd, vic, vat, wb installation on SGI
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Hello all,

I'm installing sd, vic, vat, and wb on an SGI running IRIX 6.2 and 
I'm having problems.  Any suggestions from experienced installers on 
this platform is welcome.

Thanks
Frank

------------------------------------------------------------------------
Frank Troy
Computer Sciences Corporation
ftroy@csc.com

From rem-conf-request@es.net Fri Mar 14 17:09:32 1997 
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From: Mark Handley <mjh@isi.edu>
X-Organisation: Information Sciences Institute, USC
X-Phone: +1 617 253 6011
To: Kelly Djahandari <kelly@tis.com>
cc: rem-conf@es.net
Subject: Re: socks/mbone firewalls
In-reply-to: Your message of "Fri, 14 Mar 1997 12:52:10 EST." <199703141752.MAA25187@clipper.hq.tis.com>
Date: Fri, 14 Mar 1997 17:07:12 -0500
Message-ID: <18113.858377232@buttle.lcs.mit.edu>
Sender: mjh@buttle.lcs.mit.edu


>  >4/ you could use SDR and SIP as tools to initiate the firewall access,
>  >and setup a relay between the outside generalied port, and the inside
>  >one (maybe, as i say on a different mcast addr/port pair for
>  >safety...) - this would be neat, but might not fit quickly with
>  >audited code...as per your toolkit and socks and so on....but its UDP,
>  >so its a lot easier to think through and check than TCP (imho:-)
>
>This would be neat.  However, we wanted the design of the proxy to be
>consistent with the rest of the fwtk, which is strongly oriented toward
>TCP.

In the current draft, SIP can use both TCP and UDP.

Mark




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From rem-conf-request@es.net Fri Mar 14 19:46:17 1997 
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                   3.2 #7) Fri, 14 Mar 97 16:44:23 PST
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Date: Fri, 14 Mar 97 16:41:00 PST
From: John H Wilson <John_H_Wilson@ccm.jf.intel.com>
Message-ID: <Fri, 14 Mar 97 16:44:23 PST_1@ccm.hf.intel.com>
To: rem-conf@es.net
Subject: Encrypting RTP streams

The following is some text to be included in H.323 that proposes 
how to apply various modes of encryption to its RTP streams. 
I have posted it to achieve wider scrutiny......

There are a number of issues involved in how an encryption algorithm 
may be applied to an RTP stream:
  - to allow for lost (or out-of-sequence) packets
  - to pad packets to an appropriate multiple of octets.
  - to minimize bandwidth demands
  - for endpoint performance

In H.323, encryption is applied just to the RTP payload; the headers 
remaining in the clear. 

Since packets can be lost, deciphering the stream must be stateless; 
each packet being decipherable on its own merits. The cases being 
considered are: Block algorithms in ECB, CBC, OFB or CFB modes, and 
Stream algorithms.

1. Block Algorithms
 
   a) Initialization vectors
      Most block modes involve some "chaining". That is, each encryption 
      cycle depends in some way on the output of the previous cycle. 
      Therefore, at the beginning of a packet, some initial block value 
      (usually called an Initialization Vector - IV) must be provided in 
      order to start the encryption process. Independent of how many 
      stream octets are processed on each encryption cycle, the length 
      of the IV is always equal to the length of a block. All modes 
      except Electronic Code Book (ECB) mode require an IV. In all       
      cases, an IV shall be constructed from the first B octets of: 
      (Seq# + Timestamp + Seq#) from the RTP header.
 
   b) Padding
      ECB and CBC modes always process the input stream a block at a 
      time, and, while CFB and OFB can process the input in any number 
      of octets, N (<=B), it is recommended that N=B. [This              
      recommendation is for security reasons in OFB mode (see Schneier,  
      2nd edition, p205), and for performance].
 
      Two methods are proposed to handle packets whose payload is not 
      a multiple of blocks.

      i) Ciphertext Stealing[1] for ECB & CBC; "Null pad" for CFB & OFB

         There is a description of Ciphertext Stealing in Schneier's 
         2nd edition, p191, p196. With "Null Pad", if the final block 
         is short, it is processed as though it were padded with 0 (in   
         both CFB and OFB, the final operation is an XOR of the final    
         block of plaintext with the some excrypted material). Only the  
         amount of Ciphertext equal to the length of the short block 
         is appended to the encrypted payload.

      ii) Padding in the manner prescribed by RTP 

          RFC 1889 section 5.1 describes a method of padding in which    
          the payload is padded to a multiple of blocks, the last 
          octet set with the number of padding octets (including the     
          last), and the P bit set in the header. The value of the 
          pad should be determined by the normal convention of the       
          cipher algorithm.
 
      It is recommended that all H.Secure implementations must support 
      both schemes. The scheme in use can be deduced as follows: If 
      the P bit is set, then the packet is padded; if the payload is 
      not a multiple of B octets and the P bit is not set, then          
      Ciphertext Stealing or Null Padding applies, else the packet 
      is a multiple of B, and there is no padding.
 
2. Stream Algorithms

   Stream algorithms depend on being able to generate identical key-
   streams at each end, and XORing each plaintext bit with the same 
   key stream bit at each end. Therefore, if packets are lost or re-
   ordered, the algorithm can get out of synchronization. However, 
   if, for each packet, it were possible to determine the number of 
   preceding bits in the stream, the decryption operation would be 
   able to re-sync. (Packets that are out-of-order by the time they 
   reach the decryption algorithm would have to be discarded). 
   In order to determine the number of preceding bits in the stream, 
   the first 8 octets of the payload shall contain the number of 
   preceding octets in the stream. [Note: there seems to be no way 
   to overload the Sequence number field to provide some of this 
   information. Sequence numbers start with a random value, and 
   cycle every 64K packets.]


Reference:
[1]  J. Daemon, "Cipher and Hash function design", Ph.D. Thesis, 
     Katholieke Universiteit Leuven, Mar 95.

From rem-conf-request@es.net Sat Mar 15 11:52:27 1997 
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From: Mark Handley <mjh@isi.edu>
X-Organisation: Information Sciences Institute, USC
X-Phone: +1 617 253 6011
To: John H Wilson <John_H_Wilson@ccm.jf.intel.com>
cc: rem-conf@es.net
Subject: Re: Encrypting RTP streams
In-reply-to: Your message of "Fri, 14 Mar 1997 16:41:00 PST." <Fri, 14 Mar 97 16:44:23 PST_1@ccm.hf.intel.com>
Date: Sat, 15 Mar 1997 11:43:29 -0500
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Sender: mjh@buttle.lcs.mit.edu


>In H.323, encryption is applied just to the RTP payload; the headers 
>remaining in the clear. 

>      of the IV is always equal to the length of a block. All modes 
>      except Electronic Code Book (ECB) mode require an IV. In all       
>      cases, an IV shall be constructed from the first B octets of: 
>      (Seq# + Timestamp + Seq#) from the RTP header.

I don't claim to be an encryption expert, but my understanding was
that the IV was supposed to be secret unless the first block contains
random data.  If this is not the case, chained ciphers are weakened
significantly.

Thus taking the IV from the plain text header doesn't seem to be a
good idea without additional precautions.  A better bet might be to
use a fixed IV and add a random block of dummy data to the front of
the payload before encryption.  This way the second block which
contains predictable data is chained from the random first block
rather than from a known IV.

Mark

From rem-conf-request@es.net Sat Mar 15 12:53:23 1997 
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Date: Sat, 15 Mar 1997 12:52:41 -0500
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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CC: John H Wilson <John_H_Wilson@ccm.jf.intel.com>, rem-conf@es.net
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Mark Handley wrote:
> 
> >In H.323, encryption is applied just to the RTP payload; the headers
> >remaining in the clear.
> 
> >      of the IV is always equal to the length of a block. All modes
> >      except Electronic Code Book (ECB) mode require an IV. In all
> >      cases, an IV shall be constructed from the first B octets of:
> >      (Seq# + Timestamp + Seq#) from the RTP header.
> 
> I don't claim to be an encryption expert, but my understanding was
> that the IV was supposed to be secret unless the first block contains
> random data.  If this is not the case, chained ciphers are weakened
> significantly.
> 
> Thus taking the IV from the plain text header doesn't seem to be a
> good idea without additional precautions.  A better bet might be to
> use a fixed IV and add a random block of dummy data to the front of
> the payload before encryption.  This way the second block which
> contains predictable data is chained from the random first block
> rather than from a known IV.

See Kaufman/Perlman/Speciner, "Network Security", p. 83. The book
recommends a random IV (not necessarily secret), so that the same
plaintext doesn't get encrypted into the same ciphertext. In particular:
"A randomly chosen IV guarantees that even if the same message is sent
repeatedly, the ciphertext will be completelely different each time.
Finally, a randomly chosen IV prevents attackers from supplying chosen
plaintext to the underlying encryption algorithm even if they can supply
chosen plaintext to the CBC." A random initial message block would have
the same effect. Choosing the IV from the *payload* plaintext would
defeat this randomization purpose of the IV and you might as well use,
say, 0. Given that the same payload transmitted twice is very unlikely
to have the same seq. #/ts combination, this seems ok.

Schneier (2nd ed., p. 194) says "... So, if there are n blocks, there
are n-1 exposed 'IVs,', even if the original IV is kept secret. So
there's no reason to keep the IV secret; the IV is just a dummy
ciphertext block...".


> 
> Mark

Henning

From rem-conf-request@es.net Sat Mar 15 15:35:45 1997 
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From: Toerless Eckert <Toerless.Eckert@Informatik.Uni-Erlangen.de>
Message-Id: <199703152035.VAA03479@faui40.informatik.uni-erlangen.de>
Subject: vic/vat recording tool
To: rem-conf@es.net
Date: Sat, 15 Mar 1997 21:35:08 +0100 (MET)
Organisation: CSD IMMD IV, University of Erlangen, Germany
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A stupid question for a change:

What's a good interactive tool to record vic/vat sessions ?
What's expected to be started from sdr for that purpose ?

-- 
Toerless Eckert
Universitaet Erlangen Nuernberg
RRZE / IMMD4
Martensstrasse 1
D-91058 Erlangen, Germany
Tel.: +49 9131 85 7278
FAX: +49 9131 85 8732
Toerless.Eckert@Informatik.Uni-Erlangen.DE
/C=De/A=D400/P=Uni-Erlangen/OU=Informatik/S=Eckert/G=Toerless/

From rem-conf-request@es.net Mon Mar 17 12:19:44 1997 
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To: Toerless Eckert <Toerless.Eckert@informatik.uni-erlangen.de>
cc: rem-conf@es.net
Subject: Re: vic/vat recording tool
In-reply-to: Your message of "Sat, 15 Mar 97 12:35:08 PST." <199703152035.VAA03479@faui40.informatik.uni-erlangen.de>
Date: Mon, 17 Mar 1997 08:26:52 PST
Sender: Bill Fenner <fenner@parc.xerox.com>
From: Bill Fenner <fenner@parc.xerox.com>
Message-Id: <97Mar17.082659pst.177490@crevenia.parc.xerox.com>

I've been using the MBone VCR from http://www.icsi.berkeley.edu/mbone-vcr/.
I haven't done enough recording to be sure that it really works well,
though, it seems to sometimes have trouble with RTPv2 audio.

I use the attached perl script, named record_session, to make the
sdr "record" button work.

  Bill

#!/import/misc/bin/perl
#
# record_session
#
# An interface between sdr and the MBone VCR.
# sdr launches this script with an SDP session description as input.
# This script creates an MBone VCR .vcr file out of the session description,
# and creates a .cmd file that starts recording immediately, and starts
# the MBone VCR with the .cmd file on the command line.
#
# Bill Fenner <fenner@parc.xerox.com> 22 Apr 1996
#
#
# Set the next line to the MBone VCR executable location
$vcr = "mbone_vcr";

# You should not need to change anything below this line.
require 'ctime.pl';

$ver = "0.9";

$fn = $ARGV[0];
$fn =~ s/\.vcr$//;

$vcrfn = $fn . ".vcr";
$cmdfn = $fn . ".cmd";

open(VCR,">$vcrfn") || die "$vcrfn: $!\n";
print VCR "<MBONE_VCR 1.0>\n";
print VCR "# This file was generated by record_session v$ver\n";
print VCR "session_date=", &ctime(time);
$medianum = 1;

# I don't know if MBone VCR requires any particular ordering of its file.
# Given the SDP fields in the order that sdr outputs them, this loop should
# create the .vcr file in the same order as MBone VCR does.

while (<STDIN>) {
	chop;
	($key,$data) = split(/=/,$_,2);
	if ($key eq "s") {
		print VCR "session_name=$data\n";
	} elsif ($key eq "i") {
		print VCR "session_desc=$data\n";
	} elsif ($key eq "c") {
		if ($medianame) {
			($mediaip,$mediattl) = ($data =~ m|([0-9.]+)/(\d+)$|);
		} else {
			($ip,$ttl) = ($data =~ m|([0-9.]+)/(\d+)$|);
			if ($ttl) {
				print VCR "session_max_ttl=$ttl\n";
			} else {
				print VCR "session_max_ttl=0\n";	# XXX
			}
		}
	} elsif ($key eq "m") {
		if ($medianame) {
			print VCR "${medianum}=media_port=$mediaport\n";
			print VCR "${medianum}=media_ip=$mediaip\n";
			print VCR "${medianum}=media_name=$medianame\n";
			print VCR "${medianum}=media_type=$mediatype\n";
			print VCR "${medianum}=media_mute=0\n";
			print VCR "${medianum}=media_desc=$medianame\n";
			print VCR "${medianum}=media_cmd=$mediacmd\n";
			print VCR "${medianum}=media_cid=$mediacid\n";
			print VCR "${medianum}=media_ttl=$mediattl\n";
			$medianum++;
			undef $medianame;
		}
		($medianame, $mediaport, $type, $fmt) = split(/\s+/, $data);
		$mediacid = 0;
		undef $mediatype;
		undef $mediacmd;
		undef $mediattl;
		$mediaip = $ip;
		if ($type eq "vat") {
			$mediatype = 2;
		} elsif ($type =~ /^rtp$/io || $type =~ m|^rtp/avp$|io ) {
			$mediatype = 1;
		} else {
			print STDERR "Warning: don't know about media type $type so ";
			print STDERR "can't record $medianame media...\n";
			undef $medianame;
			next;
		}
		if ($medianame eq "audio") {
			$mediacmd = "vat";
		} elsif ($medianame eq "video") {
			$mediacmd = "vic";
		} else {
			$mediacmd = "unknown";
		}
	} elsif ($key eq "a") {
		if ($medianame && $data =~ /id:(\d+)/) {
			$mediacid = $1;
		}
	}
}
if ($medianame) {
	print VCR "${medianum}=media_port=$mediaport\n";
	print VCR "${medianum}=media_ip=$mediaip\n";
	print VCR "${medianum}=media_name=$medianame\n";
	print VCR "${medianum}=media_type=$mediatype\n";
	print VCR "${medianum}=media_mute=0\n";
	print VCR "${medianum}=media_desc=$medianame\n";
	print VCR "${medianum}=media_cmd=$mediacmd\n";
	print VCR "${medianum}=media_cid=$mediacid\n";
	print VCR "${medianum}=media_ttl=$mediattl\n";
}
close(VCR);

open(CMD,">$cmdfn") || die "$cmdfn: $!\n";
print CMD "# Load the session description file\n";
print CMD "load $vcrfn\n";
print CMD "# Just in case we're being asked to add to an existing file\n";
print CMD "goto end\n";
print CMD "# Allow recording\n";
print CMD "rec_enabled on\n";
print CMD "# and start recording immediately!\n";
print CMD "rec\n";
close(CMD);

exec "$vcr -c $cmdfn &";

From rem-conf-request@es.net Mon Mar 17 14:27:31 1997 
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Date: Mon, 17 Mar 1997 11:26:40 -0800
To: internet-drafts@ietf.org
From: "C. Zhu" <czhu@ibeam.jf.intel.com>
Subject: revision of RTP payload format for H.263 video streams, v04
Cc: rem-conf@es.net, czhu@ibeam.jf.intel.com
X-Attachments: D:\doc\ietf\rtp\r263id04.txt;

--=====================_858655600==_
Content-Type: text/plain; charset="us-ascii"

Hi all,

Attached please find vision 04 of "RTP Payload Format for H.263 
Video Streams", named draft-ietf-avt-rtp-payload-04.txt.

Changes for this revision are mostly editorial to make it more
readable.

If you any comments or questions, please send it to this mailing list or 
to czhu@ibeam.intel.com.

--Chad

--=====================_858655600==_
Content-Type: text/plain; charset="us-ascii"


Internet Engineering Task Force                Audio-Video Transport WG
INTERNET-DRAFT                                                   C. Zhu
                                                            Intel Corp.
                                                         March 15, 1997
                                            Expires: September 15, 1997


               RTP Payload Format for H.263 Video Streams


Status of This Memo

This document is an Internet-Draft.  Internet-Drafts are working
documents of the Internet Engineering Task Force (IETF), its
areas, and its working groups.  Note that other groups may also
distribute working documents as Internet-Drafts.

Internet-Drafts are draft documents valid for a maximum of six
months and may be updated, replaced, or obsoleted by other
documents at any time.  It is inappropriate to use Internet-
Drafts as reference material or to cite them other than as
``work in progress.''

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

Distribution of this document is unlimited.



Abstract 

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



1. Introduction

This document describes a scheme to packetize an H.263 video stream for 
transport using RTP [1]. H.263 video stream is defined by ITU-T
Recommendation H.263 (referred to as H.263 in this document) [4] for 
video coding at very low data rates. RTP is defined by the Internet 
Engineering Task Force (IETF) to provide end-to-end network transport 
functions suitable for applications transmitting real-time data over 
multicast or unicast network services.

Zhu                                                             [Page 1]

Internet Draft       draft-ietf-avt-rtp-payload-04.txt       March, 1997


2. Definitions

The following definitions apply in this document:

CIF: Common Intermediate Format. For H.263, a CIF picture has 352 x 288 
pixels for luminance, and 176 x 144 pixels for chrominance.

QCIF: Quarter CIF source format with 176 x 144 pixels for luminance and
88 x 72 pixels for chrominance.

Sub-QCIF:  picture source format with 128 x 96 pixels for luminance and
64 x 48 pixels for chrominance.

4CIF: Picture source format with 704 x 576 pixels for luminance and
352 x 288 pixels for chrominance.

16CIF: Picture source format with 1408 x 1152 pixels for luminance and 
704 x 576 pixels for chrominance.

GOB: For H.263, a Group of Blocks (GOB) consists of  k*16 lines, where
k depends on the picture format (k=1 for QCIF, CIF and sub-QCIF; k=2 
for 4CIF and k=4 for 16CIF).

MB: A macroblock (MB) contains four blocks of luminance and the 
spatially corresponding two blocks of chrominance. Each block consists 
of 8x8 pixels. For example, there are eleven MBs in a GOB in QCIF
format and twenty two MBs in a GOB in CIF format.


3. Design Issues for Packetizing H.263 Bitstreams

H.263 is based on the ITU-T Recommendation H.261 [2] (referred to as 
H.261 in this document). Compared to H.261, H.263 employs similar 
techniques to reduce both temporal and spatial redundancy, but there 
are several major differences between the two algorithms that 
affect the design of packetization schemes significantly. This 
section summarizes those differences.

3.1 Optional Features of H.263

In addition to the basic source coding algorithms, H.263 supports four 
negotiable coding options to improve performance: Advanced Prediction, 
PB-frames, Syntax-based Arithmetic Coding, and Unrestricted Motion 
Vectors. They can be used in any combination. 

Advanced Prediction(AP): One or four motion vectors can be used 
for some macroblocks in a frame. This feature makes recovery from 
packet loss difficult, because more redundant information has to be 
preserved at the beginning of a packet when fragmenting at a macroblock 
boundary.

Zhu                                                             [Page 1]

Internet Draft       draft-ietf-avt-rtp-payload-04.txt       March, 1997


PB-frames:  Two frames (a P frame and a B frame) are coded into one 
bitstream with macroblocks from the two frames interleaved. From a 
packetization point of view, a MB from the P frame and a MB from the B 
frame must be treated together because each MB for the B frame is coded
based on the corresponding MB for the P frame. A means must be provided 
to ensure proper rendering of two frames in the right order. Also, if 
part of this combined bitstream is lost, it will affect both frames, 
and possibly more.

Syntax-based Arithmetic Coding (SAC): When the SAC option is used, the 
resultant run-value pair after quantization of Discrete Cosine 
Transform (DCT) coefficients will be coded differently from Huffman 
codes, but the macroblock hierarchy will be preserved. Since context 
variables are only synchronized after fixed length codes in the 
bitstream, any fragmentation starting at variable length codes 
will result in difficulty in decoding in the presence of packet 
loss without carrying the values of all the context variables in each 
H.263 payload header.  
 
The Unrestricted motion vectors feature allows large range of motion
vectors to improve performance of motion compensation for inter-coded
pictures. This option also affects packetization because it uses 
larger range of motion vectors than normal. 

To enable proper decoding of packets received, without dependency on 
previous packets, the use of these optional features is signaled in the 
H.263 payload header, as described in Section 5.

3.2 GOB Numbering

In H.263, each picture is divided into groups of blocks (GOB). GOBs are 
numbered according to a vertical scan of a picture, starting with the
top GOB and ending with the bottom GOB. In contrast, a GOB in H.261 
is composed of three rows of 16x16 MB for QCIF, and three half-rows 
of MBs for CIF. A GOB is divided into macroblocks in H.263 
and the definition of the macroblocks are the same as in H.261. 

Each GOB in H.263 can have a fixed GOB header, but the use 
of the header is optional. If the GOB header is present, it may or may 
not start on a byte boundary. Byte alignment can be achieved by proper 
bit stuffing by the encoder, but it is not required by the H.263
bitstream specification [4].

In summary, a GOB in H.263 is defined and coded with finer granularity 
but with the same source format, resulting in more flexibility for 
packetization than with H.261.





Zhu                                                             [Page 2]

Internet Draft       draft-ietf-avt-rtp-payload-04.txt       March, 1997


3.3 Motion Vector Encoding

Differential coding is used to code motion vectors as variable length 
codes. Unlike in H.261, where each motion vector is predicted from the 
previous MB in the GOB, H.263 employs a more flexible prediction scheme,
where one or three candidate predictors could be used depending on
the presence of GOB headers.

If the GOB header is present in a GOB, motion vectors are coded with 
reference to MBs in the current GOB only. If a GOB header is not 
present in the current GOB, three motion vectors must be available to 
decode one macroblock, where two of them might come from the previous
GOB. To correctly decode a whole inter-coded GOB, all the motion 
vectors for MBs in the previous GOB  must be available to compute the 
predictors or the predictors themselves must be present. The optional
use of three motion vector predictors can be a major problem for a 
packetization scheme like the one defined for H.261 when packetizing 
at MB boundaries [5].

Consider the case that a packet starts with a MB but the GOB header
is not present. If the previous packet is lost, then all the motion 
vectors needed to predict the motion vectors for the MBs in the current
GOB are not available. In order to decode the received MBs correctly, 
all the motion vectors for the previous GOB or the motion vector 
predictors would have to be duplicated at the beginning of the packet. 
This kind of duplication would be very expensive and unacceptable in 
terms of bandwidth overhead.

The encoding strategy of each H.263 CODEC (CODer and DECoder) 
implementation is beyond the scope of this document, even though it 
has significant effect on visual quality in the presence of 
packet loss. However, we strongly recommend use of the GOB header 
for every GOB at the beginning of a packet to address this problem.  

3.4 Macroblock Address

As specified by H.261, a macroblock address (MBA) is encoded with a 
variable length code to indicate the position of a macroblock within 
a group of MBs in H.261 bitstreams. H.263 does not code the MBA 
explicitly, but the macroblock address within a GOB is necessary to 
recover from packet loss when fragmenting at MB boundaries. 
Therefore, this information must be included in the H.263 payload 
header for modes (mode B and mode C as described in Section 5) that 
allow packetization at MB boundaries. 







Zhu                                                             [Page 3]

Internet Draft       draft-ietf-avt-rtp-payload-04.txt       March, 1997


4. Usage of RTP

When transmitting H.263 video streams over the Internet, the output
of the encoder can be packetized directly. All the bits resulting 
>from the bitstream including the fixed length codes and  variable 
length codes will be included in the packet. Also multiplexing audio 
and video signals in the same packet is not accommodated, as UDP and 
RTP provide a much more efficient way to achieve multiplexing.

RTP does not guarantee a reliable and orderly data delivery service, 
so a packet might get lost in the network. To achieve a best-effort 
recovery from packet loss, the decoder needs assistance to proceed with 
decoding of other packets that are received. Thus it is desirable to 
be able to process each packet independent of other packets. 
Some frame level information is included in each packet, such as source 
format and flags for optional features to assist the decoder in 
operating correctly and efficiently in presence of packet loss. The 
flags for H.263 optional features also provide information about 
coding options used in H.263 video bitstreams that can be used by 
session management tools.

H.263 video bitstreams will be carried as payload data within RTP 
packets. A new H.263 payload header is defined in section 5 on the H.263
payload header. This section defines the usage of RTP fixed header 
and H.263 video packet structure. 

4.1 RTP Header Usage

Each RTP packet starts with a fixed RTP header [1]. The following 
fields of the RTP fixed header are used for H.263 video streams:

Marker bit (M bit): The Marker bit of the RTP fixed header  is set to 1 
when the current packet carries the end of current frame; Set to 0 
otherwise.
 
Payload Type (PT): The Payload Type shall specify H.263 video payload 
format with decimal value 34.

Timestamp: The RTP timestamp encodes the sampling instance of the first 
video frame contained in the RTP data packet. The RTP timestamp may be 
the same  on successive packets if a video frame occupies more than one 
packet. For H.263 video streams, the RTP timestamp is based on a 90 kHz 
clock, the same as the RTP timestamp for H.261 video streams [5]. 
 
4.2 Video Packet Structure

For each RTP packet, the RTP fixed header is followed by the H.263 
payload header, which is followed by the standard H.263 compressed 
bitstream [4]. 


Zhu                                                             [Page 4]

Internet Draft       draft-ietf-avt-rtp-payload-04.txt       March, 1997


The size of the H.263 payload header is variable depending on modes 
used as detailed in the next section. The layout of an RTP H.263 
video packet is shown as:

 0                   1                   2                   3
 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                 RTP header                                    |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                 H.263 payload header                          |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                 H.263 bitstream                               |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

5. H.263 Payload Header

For H.263 video streams, each RTP packet carries only one H.263 video
packet. The H.263 payload header is always present for each H.263 video 
packet. 

Three formats (mode A, mode B and mode C) are defined for H.263 
payload header. In mode A, a H.263 payload header of four bytes is 
present before actual compressed H.263 video bitstream in a packet. 
It allows fragmentation at GOB boundaries. In mode B, a eight byte H.263
payload header is used and each packet starts at MB boundaries without
the PB-frames option. Finally, a twelve byte H.263 payload header is 
defined in mode C to support fragmentation at MB boundaries for 
frames that are coded with the PB-frames option. 

The mode of each H.263 payload header is indicated by the F and 
P fields in the header. Packets of different modes can be 
intermixed. All client application are required to be able to 
receive packets in any mode, but decoding of mode C packets 
is optional because the PB-frames feature is optional.

In this section, the H.263 payload format is shown as rows of 32-bit 
words. Each word is transmitted in network byte order. Whenever a
field represents a numeric value, the most significant bit is at
the left of the field.

5.1 Mode A

In this mode, an H.263 bitstream will be packetized
on a GOB boundary or a picture boundary. Mode A packets always 
start with the H.263 picture start code [4] or a GOB, but do not 
necessarily contain complete GOBs. Four bytes are used for the 
mode A H.263 payload header. The H.263 payload header definition 
for mode A is shown as follows with F=0.



Zhu                                                             [Page 5]

Internet Draft       draft-ietf-avt-rtp-payload-04.txt       March, 1997


 0                   1                   2                   3
 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F|P|SBIT |EBIT | SRC |I|U|S|A|R      |DBQ| TRB |    TR         |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


F: 1 bit
The flag bit indicates the mode of the payload header. F=0, mode A; 
F=1, mode B or mode C depending on P bit defined below.

P: 1 bit
Optional PB-frames mode as defined by the H.263 [4]. "0" implies normal 
I or P frame, "1" PB-frames. When F=1, P also indicates modes: mode B 
if P=0, mode C if P=1.

SBIT: 3 bits
Start bit position specifies number of most significant bits that 
shall be ignored in the first data byte.

EBIT: 3 bits
End bit position specifies number of least significant bits that 
shall be ignored in the last data byte.

SRC : 3 bits
Source format specifies the resolution of the frame as 
defined by the H.263 [4].

I:  1 bit. 
Picture coding type, defined by H.263[4], "0" is intra-coded, "1" is
inter-coded.

U: 1 bit
Unrestricted Motion Vector option as defined by H.263 [4], "0" off, 
"1" on.

S: 1 bit
Optional Syntax-based Arithmetic Coding option as defined by the 
H.263 [4], 0" off, "1" on.

A: 1 bit
Optional Advanced Prediction option as defined by H.263 [4], "0" off, 
"1" on.

R: 4 bits
Reserved, must be set to zero.





Zhu                                                             [Page 6]

Internet Draft       draft-ietf-avt-rtp-payload-04.txt       March, 1997


DBQ: 2 bits
Differential quantization parameter used to calculate quantizer for 
the B frame based on quantizer for the P frame, when PB-frames option 
is used. The value should be the same as DBQUANT defined by 
H.263 [4].  Set to zero if PB-frames option is not used.
 
TRB: 3 bits
Temporal Reference for the B frame as defined by H.263 [4]. Set to 
zero if PB-frames option is not used.

TR: 8 bits
Temporal Reference for the P frame as defined by H.263 [4]. Set to 
zero if the PB-frames option is not used.

5.2 Mode B

In this mode, an H.263 bitstream can be fragmented at MB boundaries. 
Whenever a packet starts at a MB boundary, this mode shall be used
without PB-frames option. Mode B packets are intended for a GOB 
whose size is larger than the maximum packet size allowed in the 
underlying protocol, thus making it impossible to fit one or more 
complete GOBs in a packet. This mode can only be used without the 
PB-frames option. Mode C as defined in the next section can be used 
to fragment H.263 bitstreams at MB boundaries with the PB-frames option.
The H.263 payload header definition for mode B is shown as follows 
with F=1 and P=0:

 0                   1                   2                   3
 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F|P|SBIT |EBIT | SRC | QUANT   |  GOBN   |   MBA           |R  |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|I|U|S|A| HMV1        | VMV1        | HMV2        | VMV2        |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The following fields are defined the same as in mode A: F, P, SBIT, 
EBIT, SRC, I, U, S and A. Other fields are defined as follows:

QUANT: 5 bits
Quantization value for the first MB coded at the starting of the packet.
Set to 0 if the packet begins with a GOB header. This is the equivalent 
of GQUANT defined by the H.263 [4].

GOBN: 5 bits
GOB number in effect at the start of the packet. GOB number is specified
differently for different resolutions. See H.263 [4] for details.

MBA: 9 bits
The absolute address of the first MB within its GOB, counting from 0.
 

Zhu                                                             [Page 7]

Internet Draft       draft-ietf-avt-rtp-payload-04.txt       March, 1997


HMV1, VMV1: 7 bits each.
Horizontal and vertical motion vector predictors for the first MB  
in this packet [4]. When four motion vectors are used for current
MB with advanced prediction option, these would be the motion 
vector predictors for block number 1 in the MB. Each 7 bits field
encodes a motion vector predictor in half pixel resolution as a 2's 
complement number.

HMV2, VMV2: 7 bits each.
Horizontal and vertical motion vector predictors for block number 3 
in the first MB in this packet when four motion vectors are used with 
the advanced prediction option. This is needed because block number 3 
in the MB needs different motion vector predictors from 
other blocks in the MB. These two fields are not used when the 
MB only has one motion vector. See the H.263 [4] for 
block organization in a macroblock.  Each 7 bits field encodes a motion 
vector predictor in half pixel resolution as a 2's complement number. 

R : 2 bits
Reserved, must be set to zero.

5.3 Mode C

In this mode, an H.263 bitstream is fragmented at MB boundaries of P 
frames with the PB-frames option. It is intended for those GOBs 
whose sizes are larger than the maximum packet size allowed in the 
underlying protocol when PB-frames option is used. The H.263 payload 
header definition for mode C is shown as follows with F=1 and P=1:

 0                   1                   2                   3
 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F|P|SBIT |EBIT | SRC | QUANT   |  GOBN   |   MBA           |R  |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|I|U|S|A| HMV1        | VMV1        | HMV2        | VMV2        |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| RR                                  |DBQ| TRB |    TR         | 
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The following fields are defined the same as in mode B: F, P, SBIT, 
EBIT, SRC, QUANT, GOBN, MBA, R, I, U, S, A, HMV1, VMV1, HMV2, VMV2. The
rest of the fields (TR, DBQ, TRB) are defined the same as in mode A, 
except field RR. RR is reserved and its value must be zero.







Zhu                                                             [Page 8]

Internet Draft       draft-ietf-avt-rtp-payload-04.txt       March, 1997


5.4 Selection of Modes for the H.263 Payload Header

Packets carrying H.263 video streams with different modes can be 
intermixed. The modes shall be selected carefully based on network 
packet size, H.263 coding options and underlying network protocols. 
More specifically, mode A shall be used for packets starting with a GOB
or the H.263 picture start code [4], and mode B or C shall be used 
whenever a packet has to start at a MB boundary. Mode B or C are 
necessary for those GOBs with sizes larger than network packet size. 

We strongly recommend use of mode A whenever possible. 
The major advantage of mode A over mode B and C is its simplicity. 
The H.263 payload header is smaller than mode B and C. Transmission 
overhead is reduced and the savings may be very significant when 
working with very low data rates or relatively small packet sizes. 

Another advantage of mode A is that it simplifies error recovery in the
presence of packet loss. The internal state of a decoder can be 
recovered at GOB boundaries instead of having to synchronize with MBs 
as in mode B and C. The GOB headers and the picture start code are 
easy to identify,  and their presence will normally cause a H.263 
decoder to re-synchronize its internal states. Requiring a decoder to 
synchronize its internal states at MBs introduces extra overhead and 
complexity for the decoder.


Finally, we would like to stress that recovery from packet loss 
depends on a decoder's ability to use the information provided 
in the H.263 payload header within RTP packets.

6. References

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

[2] International Telecommunication Union.
    Video Codec for Audiovisual Services at  p x 64 kbits/s, 
    ITU-T Recommendation H.261, 1993.

[3] H. Schulzrinne.
    RTP Profile for Audio and Video Conference with Minimal 
    Control, RFC 1890.
    IETF, 1996.

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



Zhu                                                             [Page 9]

Internet Draft       draft-ietf-avt-rtp-payload-04.txt       March, 1997


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

7. Author's Address

C. "Chad"  Zhu
Mail Stop: JF2-78
Intel Corporation
2111 N.E. 25th Avenue
Hillsboro, OR 97124
USA

Email: czhu@ibeam.intel.com 
Tel: (503) 264-8849
Fax: (503) 264-6067



































Zhu                                                            [Page 10]

--=====================_858655600==_
Content-Type: text/plain; charset="us-ascii"


--=====================_858655600==_--


From rem-conf-request@es.net Mon Mar 17 14:28:41 1997 
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          Mon, 17 Mar 1997 11:28:14 -0800
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          by nobozo.CS.Berkeley.EDU (8.6.10/8.6.3) with SMTP id LAA01708 
          for <rem-conf@es.net>; Mon, 17 Mar 1997 11:02:53 -0800
Date: Mon, 17 Mar 1997 11:02:53 -0800
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Content-Type: text/plain; charset="us-ascii"
To: rem-conf@es.net
From: Jerrlyn Iwata <jerrlyn@postgres.Berkeley.EDU>
Subject: [Reminder] Berkeley Multimedia & Graphics Seminar

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

        "The Prince and the Pauper, or, is content really King?" 

                                John Taysom 
                                Reuters, Ltd. 

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


From rem-conf-request@es.net Mon Mar 17 20:31:52 1997 
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          by osi-west.es.net with ESnet SMTP (PP);
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          Mon, 17 Mar 1997 17:32:50 -0800
Message-ID: <503A2A3C2932CF118D8800805FD44E1802F46AD3@red-68-msg.dns.microsoft.com>
From: Eric Fleischman <ericfl@MICROSOFT.com>
To: "'mjh@isi.edu'" <mjh@isi.edu>, "'rem-conf@es.net'" <rem-conf@es.net>
Subject: RE: In Defense of Interleaved Data Formats
Date: Mon, 17 Mar 1997 17:29:52 -0800
X-Priority: 3
X-Mailer: Internet Mail Service (5.0.1457.3)

I was tempted to let this message slide by (now that I am finally
reading the messages sent while I was on vacation) but in the interest
of accuracy (Truth) I couldn't do so in this instance.

We have found that "de-interleaving" is very easy to handle at the
client. It can be accomplished quickly and efficiently. 

All our interleaved media objects are pre-time-stamped. 

Our algorithm for interleaved data is much simpler than what you
envision. Thus, these particular comments may be good jousting targets
but don't seem to be born out in "real life" - or at least, in our
experience to date.

As a reminder, my position is that there are times when interleaving
makes the best sense and times when segregated streams make the best
sense. Thus, I am not speaking against segregated streams but rather am
merely saying that these particular arguments are not valid.

Anyway, a seventh reason to use Interleaved Data Streams are for
situations in which you wish to constrain the total streamed bandwidth
to a certain maximum data rate while simultaneously ensuring that that
entire capacity is used (i.e., multimedia content to not dip much below
a pre-specified data rate to ensure that the "most multimedia stuff" can
be jammed into that constrained capacity target).

	-----Original Message-----
	From:	Mark Handley [SMTP:mjh@isi.edu] <mailto:>
	Sent:	Friday, March 07, 1997 2:40 PM
	To:	Sampo Syreeni
	Cc:	Eric Fleischman; 'rem-conf@es.net'
	Subject:	Re: In Defense of Interleaved Data Formats


		>Indeed. But always having to do multiple streams is a
true problem for
		>simple clients. I think I'm taking the position that
interlaced data
		>streams as a unit must be supported as much as simple
streams do. They
		>truly are the most common ones today. And expect
nothing less when digital
		>TV takes on...

	No.  
	An interleaved client has to de-interleave the streams and
timestamp them, then decode them (different media have different decode
delays) then buffer to perform playout adjustment for audio-device clock
skew, different decode times, network jitter and audio device buffer
latencies, then actually play the media streams out.
	A non-interleaved client does the same thing, except the streams
arrive de-interleaved and pre-timestamped.
	In either model, you can move the playout buffering around
somewhat, but it doesn't change the general principle.

	Interleaved streams are common because switched circuits are.
We don't have that constraint.
		>But if the original format of the data is
		>interleaved, it would be nice if it could be used as-is
as just another
		>stream. Especially since there are interleaved
protocols out there.

	Depends if the original format was designed to cope with packet
loss.  H.221 for example is not, and takes many seconds to recover after
a packet loss.  It's consituent H.261 and audio codings can however be
packetised to cope quite well with loss.  Just because your data
originated interleaved doesn't mean it's a good idea to transmit it that
way.
	Mark

From rem-conf-request@es.net Tue Mar 18 04:17:07 1997 
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Sender: campanal@mailer.cefriel.it
Message-ID: <332E54E9.794BDF32@mailer.cefriel.it>
Date: Tue, 18 Mar 1997 09:40:09 +0100
From: Giuseppe Fabio Campanale <campanal@mailer.cefriel.it>
Organization: CEFRIEL - COGEFO
X-Mailer: Mozilla 3.01 (X11; I; SunOS 4.1.4 sun4c)
MIME-Version: 1.0
To: vic@ee.lbl.gov
CC: rem-conf@es.net
Subject: Problems with VIC
Content-Type: text/plain; charset=us-ascii
Content-Transfer-Encoding: 7bit

Hello,

   I retreived the latest version of vic distribution file (v 2.8) and
compiled it successfully on Sun SS20 equipped with SunVideo.
Problems arise when I try to transmit images using (M)Jpeg encoding: all
I can get is a "Bus error" or "Segmentation fault" message. Apart
*this*, everything works fine.

Any hint ?
Has it anything to do with a "buggy" jpeg.cc module or the like ? If so,
is there (where ?) any patch available ?
Or, maybe, is it my fault ?

Thank you very much for your kind attention.
Best Regards.

-- 
 Fabio Campanale

From rem-conf-request@es.net Tue Mar 18 05:13:22 1997 
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Message-ID: <c=GB%a=_%p=BT%l=HAROLD-970318100545Z-6815@mussel.drake.bt.co.uk>
From: Benoit Gaillard <benoit.gaillard@bt-sys.bt.co.uk>
To: "'rem-conf@es.net'" <rem-conf@es.net>
Subject: Trigger join messages
Date: Tue, 18 Mar 1997 10:05:45 -0000
X-Mailer: Microsoft Exchange Server Internet Mail Connector Version 4.0.837.3
Encoding: 18 TEXT

Hi everybody,

Thanks for all this interesting issues discussed on this Email list ...
one personal concern :

I would like to trigger join messages on behalf of external events. 
VIC or VAT applications do not seem easy to modify, do you agree?
Are there other ways of triggering join messages in the multicast
receiver, rather than using multicast applications?

Thanks for your interest,


Benoit.
________________________________________________________
Benoit GAILLARD - B29 Rm138 
Phone : 00-44-(0)1473-647609
________________________________________________________

From rem-conf-request@es.net Tue Mar 18 05:25:11 1997 
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Reply-to: Please.reply.via.fax.or.smail@fax.number.or.smail.address.shown.below.Thank.you
Approved: by tearouts@cloud9.net
X-Priority: 1 (Highest)
Date: Tue, 18 Mar 1997 01:49:11 -1812
To: tearouts@cloud9.net
From: tearouts@cloud9.net
Subject: ===>> FREE 1 yr USA Magazine Sub sent worldwide-270+ Choices! Up to 
         $64.00 value!

FOR MORE INFO:   please "cut out" the below form on the "cut" lines shown,
and fax it, for the fastest reply to:                  718-227-9125   (this
is a fax # in the USA)

or send via smail (first class mail or airmail) to:
                                         Tempting Tear-Outs
                                         Att. Free-catalogue-by-email Dept
                                         3835 Richmond Ave.  Suite #200
                                         Staten Island NY  10312-3828
                                         USA

SORRY, BUT.... our software is not set up to accept forms via return email;
WE CAN ONLY acknowledge forms sent in via fax or smail.

--> IMPORTANT complete directions, to ensure that you get a reply, and more
info follow, below the reply form and the catalogue options.


*------------cut here/begin-------------------------------------------*

Name:
Internet email address:
Smail home address:
City-State-Zip:
Country:
Work Tel. #:
Work Fax #:
Home Tel. #:
Home Fax #:

How did you hear about us (name of person/company who referred you or the
area of
the internet that you saw us mentioned in):  Referred by:  Tempting
Tear-Outs
031497-l

Name of USA mags you currently get on the newsstand or in the store:

Name of USA mags you currently get on a subscription basis, through the mail:

Name of USA mags you would like price quotes on when we call you:

Catalogue version desired (list number of choice below):

*------------cut here/end--------------------------------------------*



CATALOGUE VERSION CHOICES:

1.  This version can be read by everyone, no matter what type of
     computer you use, or what type of software you use.  It is a simple
     format, with just our entire catalogue pasted into the body of a
     single email message, 316K in size.  If you use pine or elm on a unix
     system or an advanced software version such as Eudora Pro 3.0 or
     later, you will most likely receive it as a single email message.
     However, if your software limits incoming email messages to a
     certain size, say 32K or so, then your software will split it into
     multiple email message parts.   Whether you receive it as a single
     email message or multiple part email messages, you can easily
     paste it into one whole text document with your word processor, in
     about 10 minutes or so.
2.  For more advanced computer users:  attached plain ascii text file
     ~316K - you must know how to download an attached text file and
     then be able to locate it on your hard drive or system home
     directory;  it can then be opened with any pc or mac word processing
     software.  If in doubt, don't ask for this version.  This isn't for
     internet *newbies.* Better to order option 1 and spend a few minutes
     pasting them into one whole text document with your word processor,
     than to waste hours trying to figure how to deal with this option.
     This version is great for doing keyword searches and jumping around
     within the catalogue with your word processing software, if your
     normal email reading software doesn't allow this.


VERY IMPORTANT DIRECTIONS TO ENSURE THAT YOU GET A REPLY:

1.   you must call from an "unblocked number," ie. one that is not blocked
>from caller id.  We are very sorry for this requirement, but our fax
software requires this before it allows an incoming fax call to connect.
If you have a blocked number, you must first unblock it.  In most cases
this means dialing *82 from a touch-tone phone (or 1182 from a rotary
phone) before you dial 1-718-227-9125.     NOTE:  If you are not sure if
your number is blocked, just try dialing our fax # normally.  If you don't
get a recording telling you your number is blocked, your number has been
transmitted and you may press the start button on your fax when you hear
the fax tone from our fax.
2.   no reply forms can be accepted by email....only via fax or smail.
3.   your form must be typewritten or printed out on your computer printer
before you fax it;  sorry, but *no* handwritten forms will be acknowledged.
If you can't find someone with a typewriter or a computer printer, we
apologize for not being able to reply to you.
4.   faxes with cover pages will be rejected.  You must send *only* the
reply form.
5.   forms not *completely* filled in will not be acknowledged.
6.   you will receive a reply within 1 business day directly from the
company making the offer via email.  Therefore you must have an email
address.  If you read this message, then you must have an email address, or
access to one, at least.   :-)
7.   your fax must not exceed 1 page in length.   Faxes of 2 or more pages
will be sensed, then auto-terminated and deleted.  Your fax goes directly
onto our 5.0 gigabyte hard drive and we must limit all incoming faxes to 1
page.
8.   all faxes must begin with:
*------------cut here/begin-------------------------------------------*
and must end with:
*------------cut here/end--------------------------------------------*
9. Any fax not conforming to this format will be sensed by our software,
then auto-terminated and deleted from the hard drive, before any human ever
gets to see it.
10. The type on your fax must be dark and legible.   If in doubt, please
print it out darker before faxing it in.  If we can't read it, we can't
reply to you or send you our FREE catalogue.     :-(
11.  If this all seems too complicated for faxing, just do it the old
fashioned way via smail!!!


WHO WE ARE:

Tempting Tear-Outs is an advertising company that brings potential new
customers to the companies they advertise for.

MORE ABOUT THE COMPANY MAKING THE FREE OFFER:

The company making the offer is a magazine subscription agency based in the
USA.  They have over 1,500 popular USA titles available to be shipped to
*any* country, including of course, to anywhere in the USA!    They offer a
FREE 1 yr. subscription to your choice of over 270 of the titles in their
catalogue to any new customer using them for the first time.       The
dollar value of the freebies, based on the subscription prices directly
>from the publishers, ranges from $6.97 all the way up to $50.00!

For new customers in the USA, there is no charge for FPH (foreign postage &
handling), so the freebie is 100% free!   For new customers living
overseas, the only charge on the freebie would be for the FPH (foreign
postage & handling).

Their president has been in the magazine subscription business since 1973
and they are very customer-service oriented.   They will even help you with
address changes on your magazines, even if you move from one country to
another country.   They have thousands of happy customers in over 59
countries.

Their price guarantee is very simple:       they guarantee that their
subscription prices are the lowest available and they will BEAT any
legitimate, verifiable offer before you pay them or match it afterwards, by
refunding you the difference in price PLUS the cost of the postage stamp
you would use sending in the special offer to them, even 6 months after you
pay them, as long as it was current at the time of your offer.    Does that
sound fair?       Wouldn't it be great if everything you bought came with
that price guarantee?

Sometimes they are less than half of the next best deal out there,
sometimes just a little cheaper, but always you get the lowest rates
without having to shop around.     With 1,500 titles on their list, they
would like to think that they have also the best selection around!

Within the USA, for their USA customers, they are cheaper than all their
competitors and even the publishers themselves.  This is their price
guarantee.         The 1 yr. freebie that you get with your first order is
completely free!

Overseas, (even after you factor in the cost of the FPH (foreign postage &
handling) and the conversion from USA Dollars to your currency), on the
average, they are generally around one-fourth to one-half of what the
newsstands overseas charge locally for USA magazines.  On some titles they
are as little as one-tenth of what the newsstands charge.  They are also
the cheapest subscription source for delivery overseas, including directly
>from the publishers themselves!   Some publishers don't even offer
subscriptions overseas.........but overseas subscriptions are this
company's specialty!  They feel that magazines should not be a luxury
overseas.   In the USA, people buy magazines and then toss them after
reading them for just a few minutes or hours.  They are so cheap in the
USA!   Well, this company would like to make it the same way for their
overseas members.  They are also cheaper than all their competitors in the
USA and overseas, including the publishers themselves!   It is also *highly
unlikely* you will find any of their USA competitors calling you overseas,
in order to offer that personal touch, just to sell you a couple of
magazines!  But that is what this company specializes in and loves doing!
Around one-half their business comes from overseas, so they are very
patient with new members who only speak limited English as a 2nd language.
Subscription prices quoted for overseas consist of the subscription price,
plus the FPH.   You add the two together and that is your total cost.   The
exception is the 1 yr. freebie you get with your first order.   On that
title, you pay *only* the FPH for the 1 yr. term.

Their prices are so cheap because when you deal with them, you cut-out all
the middlemen.


HERE IS HOW YOU CAN GET MORE INFO AND GET STARTED WITH THEM:

Simply fax or smail back to us the reply form listed at the top of this
message.   We will then forward your form on to the subscription agency.
They will then email their "big and juicy" catalogue to you, in whichever
of the four formats you chose.   The catalogue is FREE and makes for hours
of fascinating reading, on its own. It includes the complete list of
freebies, a complete list of all the titles they sell, as well as detailed
descriptions on most of the titles, along with lists of titles by category
of interest and their terms of sale.

They will then give you a friendly, no-pressure, no obligation, 5-minute
call to go over how they work and to answer any questions that you might
have, as well as give you up-to-the minute price quotes on any titles you
might be considering.     They will call you in whatever country you live
in, taking the time difference into account.        As they like to
emphasize the personal touch they give to each new customer, all first-time
orders can only be done via phone, so they can answer all your questions
completely and personally.   Once you have placed your first order via
phone, you will be able to place future orders and make inquiries on your
account, get price quotes, etc., all via email, if that is most convenient
for you.

Within the USA, they accept payment via check over the phone, Mastercard,
Visa and American Express.      Overseas, they accept Mastercard, Visa and
American Express, even if your credit card is a local one in local currency!

That's our introduction of our client that we represent.   We hope that we
have piqued your interest and that you will take the next step to get their
free catalogue!   Thank you for your time and interest.

--
Tempting Tear-Outs.
For more info on advertising rates, please write us on your company
letterhead, w/business card, via smail to:   Tempting Tear-Outs, 3835
Richmond Ave. Suite #200, Staten Island NY  10312-3828, USA.



From rem-conf-request@es.net Tue Mar 18 14:50:23 1997 
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To: rem-conf@es.net
From: m1 <m1@msri.org>
Reply-to: m1@msri.org
Subject: Curtis Greene "Representations, Posets, and Moebius Functions"

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

Title:       
	Curtis Greene "Representations, Posets, and Moebius Functions"
Date:        
	Mar 25, 1997

Time:        
	12:00 PST8PDT 1 hours

Contact:     
	m1@msri.org

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

Description:        
	Curtis Greene's talk from MSRI sponsor's day, February 28, 1997.









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

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Message-Id: <199703181951.LAA15162@hille.msri.org>
To: rem-conf@es.net
From: m1 <m1@msri.org>
Reply-to: m1@msri.org
Subject: Andrew Casson "Geometric Topology in Dimension 3"

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

Title:       
	Andrew Casson "Geometric Topology in Dimension 3"
Date:        
	Mar 26, 1997

Time:        
	12:00 PST8PDT 1 hours

Contact:     
	m1@msri.org

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

Description:        
	Andrew Casson's talk from MSRI sponsor's day, February 28, 1997.









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

From rem-conf-request@es.net Tue Mar 18 14:54:32 1997 
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To: rem-conf@es.net
From: m1 <m1@msri.org>
Reply-to: m1@msri.org
Subject: Curt McMullen "Geometrically Infinite Kleinian Groups"

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

Title:       
	Curt McMullen "Geometrically Infinite Kleinian Groups"
Date:        
	Mar 27, 1997

Time:        
	12:00 PST8PDT 1 hours

Contact:     
	m1@msri.org

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

Description:        
	Curt McMullen's talk from  <a href="/sched/comp_top/">    Computational and Algorithmic Methods in 3-Dimensional    Topology Workshop  </a>, March 13, 1997.









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

From rem-conf-request@es.net Wed Mar 19 14:11:01 1997 
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From: deleon@hplabsz.hpl.hp.com (Laura de Leon)
Message-Id: <9703191110.ZM335@hplabsz.hpl.hp.com>
Date: Wed, 19 Mar 1997 11:10:07 -0800
X-Mailer: Z-Mail (3.0.0 15dec93)
To: baylisa@baylisa.org, sage-members@usenix.org, rem-conf@es.net
Subject: (BayLISA: Paul Vixie on BIND Version 8
Cc: deleon@hplabsz.hpl.hp.com
Content-Type: text/plain; charset=us-ascii
Mime-Version: 1.0

The BayLISA group meets monthly to discuss topics of interest to systems
and network administrators.  The meetings are free and open to the public.

BayLISA holds monthly meetings on the third Thursday of each month at
7:30 PM PST.  We meet at Cisco building J in San Jose, on Tasman Drive near
First street. See www.baylisa.org for more information.  The meetings are also
broadcast via MBONE.

Schedule
--------


March 20
Paul Vixie
BIND Version 8 is in final public beta testing.  It has a completely new
configuration file format, many new features including dynamic update and
online change notification, and a better build mechanism than BIND 4.9.5.
Paul will give a brief overview of the new config syntax, new features, and
will tell once again why we didn't call it BIND Version 5.  Paul will also
describe the Internet Software Consortium and its other software projects


[Schedule subject to change]

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

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

	ftp.baylisa.org:/location

For any other information, please send email to:

	info@baylisa.org

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




From rem-conf-request@es.net Thu Mar 20 10:17:20 1997 
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To: IETF-Announce:;
cc: rem-conf@es.net
From: Internet-Drafts@ietf.org
Reply-to: Internet-Drafts@ietf.org
Subject: I-D ACTION:draft-ietf-avt-rtp-payload-03.txt
Date: Thu, 20 Mar 1997 09:56:18 -0500
Sender: cclark@ietf.org
Message-ID: <9703200956.aa13691@ietf.org>

--NextPart

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

       Title     : RTP Payload Format for H.263 Video Streams              
       Author(s) : C. Zhu
       Filename  : draft-ietf-avt-rtp-payload-03.txt
       Pages     : 10
       Date      : 03/19/1997

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

Internet-Drafts are available by anonymous FTP.  Login with the username
"anonymous" and a password of your e-mail address.  After logging in,
type "cd internet-drafts" and then
     "get draft-ietf-avt-rtp-payload-03.txt".
A URL for the Internet-Draft is:
ftp://ds.internic.net/internet-drafts/draft-ietf-avt-rtp-payload-03.txt
 
Internet-Drafts directories are located at:	
	                                                
     o  Africa:  ftp.is.co.za                    
	                                                
     o  Europe:  ftp.nordu.net            	
                 ftp.nis.garr.it                 
	                                                
     o  Pacific Rim: munnari.oz.au               
	                                                
     o  US East Coast: ds.internic.net           
	                                                
     o  US West Coast: ftp.isi.edu               
	                                                
Internet-Drafts are also available by mail.	
	                                                
Send a message to:  mailserv@ds.internic.net. In the body type: 
     "FILE /internet-drafts/draft-ietf-avt-rtp-payload-03.txt".
							
NOTE: The mail server at ds.internic.net can return the document in
      MIME-encoded form by using the "mpack" utility.  To use this
      feature, insert the command "ENCODING mime" before the "FILE"
      command.  To decode the response(s), you will need "munpack" or
      a MIME-compliant mail reader.  Different MIME-compliant mail readers
      exhibit different behavior, especially when dealing with
      "multipart" MIME messages (i.e., documents which have been split
      up into multiple messages), so check your local documentation on
      how to manipulate these messages.
							
							

Below is the data which will enable a MIME compliant mail reader 
implementation to automatically retrieve the ASCII version
of the Internet-Draft.

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

--OtherAccess
Content-Type:  Message/External-body;
        access-type="mail-server";
        server="mailserv@ds.internic.net"

Content-Type: text/plain
Content-ID: <19970319134105.I-D@ietf.org>

ENCODING mime
FILE /internet-drafts/draft-ietf-avt-rtp-payload-03.txt

--OtherAccess
Content-Type:   Message/External-body;
        name="draft-ietf-avt-rtp-payload-03.txt";
        site="ds.internic.net";
        access-type="anon-ftp";
        directory="internet-drafts"

Content-Type: text/plain
Content-ID: <19970319134105.I-D@ietf.org>

--OtherAccess--

--NextPart--


From rem-conf-request@es.net Thu Mar 20 13:11:31 1997 
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X-Mailer: exmh version 2.0gamma 1/24/96
To: rem-conf@es.net
cc: ekumedia@sun3.oulu.fi, tkuusiva@syy.oulu.fi
Subject: Event announcement: Finnish POP
Mime-Version: 1.0
Content-Type: text/plain; charset=us-ascii
Date: Thu, 20 Mar 1997 20:10:46 +0200
From: Janne Himanka <shem@oyt.oulu.fi>

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

Title:       
	A Grand Day Out: A Slice of the Finnish Pop/Rock Scene
Date:        
	Apr 14 1997

Time:        
	14:00 GMT 4 hours

Contact:     
	shem@katiska.oulu.fi

URL:         
	http://katiska.oulu.fi/xmission.html

Description:        
	A few videos of some of the more promising Finnish bands.
	We will also test the Nordic MBone topology. 

Please contact shem@katiska.oulu.fi if something more serious and academic 
collides with this, and we will settle a more suitable time.

Janne Himanka


From rem-conf-request@es.net Thu Mar 20 20:23:06 1997 
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To: rem-conf@es.net
From: m1 <m1@msri.org>
Reply-to: m1@msri.org
Subject: Donald Knuth "Graph Drawing from a User's Perspective"

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

Title:       
	Donald Knuth "Graph  Drawing from a User's Perspective" 
Date:        
	Apr 01, 1997

Time:        
	12:00 PST8PDT 1 hours

Contact:     
	m1@msri.org

URL:         
	http://www.msri.org/lecturenotes/96/knuth/

Description:        
	Donald Knuth's 9/18/96 from Graph Drawing 96 at MSRI.









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

From rem-conf-request@es.net Thu Mar 20 20:25:01 1997 
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To: rem-conf@es.net
From: m1 <m1@msri.org>
Reply-to: m1@msri.org
Subject: Donald Knuth "Some Research Problems for Combinatorialists"

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

Title:       
	Donald Knuth "Some Research Problems for Combinatorialists"
Date:        
	Apr 02, 1997

Time:        
	08:00 PST8PDT 1 hours

Contact:     
	m1@msri.org

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

Description:        
	Donald Knuth's 1/27/97 talk in the MSRI Combinatorics  seminar.









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

From rem-conf-request@es.net Thu Mar 20 20:27:01 1997 
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To: rem-conf@es.net
From: m1 <m1@msri.org>
Reply-to: m1@msri.org
Subject: T. Y. Lam "Kap"

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

Title:       
	T. Y. Lam "Kap"
Date:        
	Apr 03, 1997

Time:        
	12:00 PST8PDT 1 hours

Contact:     
	m1@msri.org

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

Description:        
	T. Y. Lam reviews the works and life of MSRI director  emeritus Irving Kaplansky on the occasion of his 80th  birthday. 









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

From rem-conf-request@es.net Thu Mar 20 23:06:07 1997 
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To: ru-comp-dev-ietf-rem-conf@rutgers.edu
Path: usenet
From: m1@msri.org (m1)
Newsgroups: ru.comp.dev.ietf.rem-conf
Subject: Donald Knuth "Some Research Problems for Combinatorialists"
Date: 20 Mar 1997 23:04:26 -0500
Organization: Rutgers University
Lines: 29
Sender: news@RUTGERS.EDU
Message-ID: <5gt1ca$5p6$1@rutgers.rutgers.edu>

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

Title:       
	Donald Knuth "Some Research Problems for Combinatorialists"
Date:        
	Apr 02, 1997

Time:        
	08:00 PST8PDT 1 hours

Contact:     
	m1@msri.org

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

Description:        
	Donald Knuth's 1/27/97 talk in the MSRI Combinatorics  seminar.









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

From rem-conf-request@es.net Thu Mar 20 23:56:08 1997 
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To: ru-comp-dev-ietf-rem-conf@rutgers.edu
Path: usenet
From: m1@msri.org (m1)
Newsgroups: ru.comp.dev.ietf.rem-conf
Subject: Donald Knuth "Graph Drawing from a User's Perspective"
Date: 20 Mar 1997 23:55:24 -0500
Organization: Rutgers University
Lines: 29
Sender: news@RUTGERS.EDU
Message-ID: <5gt4bs$74m$1@rutgers.rutgers.edu>

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

Title:       
	Donald Knuth "Graph  Drawing from a User's Perspective" 
Date:        
	Apr 01, 1997

Time:        
	12:00 PST8PDT 1 hours

Contact:     
	m1@msri.org

URL:         
	http://www.msri.org/lecturenotes/96/knuth/

Description:        
	Donald Knuth's 9/18/96 from Graph Drawing 96 at MSRI.









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

From rem-conf-request@es.net Fri Mar 21 00:06:51 1997 
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Path: usenet
From: m1@msri.org (m1)
Newsgroups: ru.comp.dev.ietf.rem-conf
Subject: T. Y. Lam "Kap"
Date: 21 Mar 1997 00:05:30 -0500
Organization: Rutgers University
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Sender: news@RUTGERS.EDU
Message-ID: <5gt4uq$7iu$1@rutgers.rutgers.edu>

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

Title:       
	T. Y. Lam "Kap"
Date:        
	Apr 03, 1997

Time:        
	12:00 PST8PDT 1 hours

Contact:     
	m1@msri.org

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

Description:        
	T. Y. Lam reviews the works and life of MSRI director  emeritus Irving Kaplansky on the occasion of his 80th  birthday. 









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

From rem-conf-request@es.net Fri Mar 21 14:30:14 1997 
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          Fri, 21 Mar 1997 11:29:30 -0800
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Mime-Version: 1.0
Content-Type: text/plain; charset="us-ascii"
Date: Fri, 21 Mar 1997 11:29:06 -0800
To: rem-conf@es.net
From: Steve Deering <deering@cisco.com>
Subject: portable MBone terminal?

Has anyone managed to put together a PC laptop system adequate to serve
as a portable MBone terminal and demo machine?  The necessary pieces would
presumably be:

	- a PC laptop, with built-in pointing device suitable for
	  drawing in wb.

	- some flavor of BSD Unix with support for IP multicast and for
	  mrouted 3.8 (in case it has to tunnel to reach the MBone)

	- mrouted, mtrace, sdr, vat, vic, and wb, all working correctly

	- a 10BASE-T ethernet port

	- a (VGA? SVGA?) video output port to drive an LCD projector or
	  overhead projector panel

	- analog audio output port for connecting to, e.g., a hotel
	  meeting room PA system

	- built-in microphone, or audio input port to accept an external
	  microphone.

	- (optional) video input port and tiny camera

If any of you has succeeded in putting together such a system, I would love
to get a "parts list" and any other advice/suggestions/handholding you'd
be willing to share.

Steve



From rem-conf-request@es.net Fri Mar 21 17:31:23 1997 
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          Fri, 21 Mar 1997 14:30:37 -0800
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Date: Fri, 21 Mar 1997 22:30:30 GMT
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          via smap (V1.3mjr) id smaMzsDmb; Fri Mar 21 22:30:05 1997
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X-Mailer: Windows Eudora Light Version 1.5.2
Mime-Version: 1.0
Content-Type: text/plain; charset="us-ascii"
To: videophone@es.net
From: Leora Kudo <lkudo@ibm.net>
Subject: PC-desktop videoconferencing
Cc: rem-conf@es.net

I am searching for any relevant, updated information regarding :
1. number of ISDN lines available worldwide (with country break-down i.e.
US, Europe, Asia-Pacific)
2. ISDN-based PC desktop videoconferencing \ teleconferencing systems : 
2.1. market estimates and forecasts 
	2.1.1. number of units sold
	2.1.2. number of units forecasted for 1997 and 1998
2.3. predicted growth of ISDN based videocommunications equipment revenue
(not network)
2.4. major players in this arena and their respective market share (with
country break-down i.e. US, Europe, Asia-Pacific)
2.5. unit pricing
2.6. main applications 
2.7. trends
3. PC desktop videoconferencing over  INTERNET and POTS
3.1. market estimates and forecasts 
3.2. major players in this arena and their respective market share
3.3. trends
4. Fast Modems
4.1. market estimates and forecasts
4.2. major players


If you can guide me to any web sites, send me information or provide me with
any pointers, I will be more than thankful. 
Until then, thanking you in advance,
Regards,
Leora


From rem-conf-request@es.net Fri Mar 21 18:10:37 1997 
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          Fri, 21 Mar 1997 15:09:18 -0800
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To: ru-comp-dev-ietf-rem-conf@rutgers.edu
Path: usenet
From: deering@cisco.com (Steve Deering)
Newsgroups: ru.comp.dev.ietf.rem-conf
Subject: portable MBone terminal?
Date: 21 Mar 1997 18:09:09 -0500
Organization: Rutgers University
Lines: 32
Sender: news@RUTGERS.EDU
Message-ID: <5gv4el$at8$1@rutgers.rutgers.edu>

Has anyone managed to put together a PC laptop system adequate to serve
as a portable MBone terminal and demo machine?  The necessary pieces would
presumably be:

	- a PC laptop, with built-in pointing device suitable for
	  drawing in wb.

	- some flavor of BSD Unix with support for IP multicast and for
	  mrouted 3.8 (in case it has to tunnel to reach the MBone)

	- mrouted, mtrace, sdr, vat, vic, and wb, all working correctly

	- a 10BASE-T ethernet port

	- a (VGA? SVGA?) video output port to drive an LCD projector or
	  overhead projector panel

	- analog audio output port for connecting to, e.g., a hotel
	  meeting room PA system

	- built-in microphone, or audio input port to accept an external
	  microphone.

	- (optional) video input port and tiny camera

If any of you has succeeded in putting together such a system, I would love
to get a "parts list" and any other advice/suggestions/handholding you'd
be willing to share.

Steve



From rem-conf-request@es.net Fri Mar 21 20:35:35 1997 
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          Fri, 21 Mar 1997 17:34:46 -0800
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          Fri, 21 Mar 1997 20:34:42 -0500
To: ru-comp-dev-ietf-rem-conf@rutgers.edu
Path: usenet
From: lkudo@ibm.net (Leora Kudo)
Newsgroups: ru.comp.dev.ietf.rem-conf
Subject: PC-desktop videoconferencing
Date: 21 Mar 1997 20:34:39 -0500
Organization: Rutgers University
Lines: 29
Sender: news@RUTGERS.EDU
Message-ID: <5gvcvf$f1v$1@rutgers.rutgers.edu>

I am searching for any relevant, updated information regarding :
1. number of ISDN lines available worldwide (with country break-down i.e.
US, Europe, Asia-Pacific)
2. ISDN-based PC desktop videoconferencing \ teleconferencing systems : 
2.1. market estimates and forecasts 
	2.1.1. number of units sold
	2.1.2. number of units forecasted for 1997 and 1998
2.3. predicted growth of ISDN based videocommunications equipment revenue
(not network)
2.4. major players in this arena and their respective market share (with
country break-down i.e. US, Europe, Asia-Pacific)
2.5. unit pricing
2.6. main applications 
2.7. trends
3. PC desktop videoconferencing over  INTERNET and POTS
3.1. market estimates and forecasts 
3.2. major players in this arena and their respective market share
3.3. trends
4. Fast Modems
4.1. market estimates and forecasts
4.2. major players


If you can guide me to any web sites, send me information or provide me with
any pointers, I will be more than thankful. 
Until then, thanking you in advance,
Regards,
Leora


From rem-conf-request@es.net Fri Mar 21 22:36:35 1997 
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Date: Sat, 22 Mar 1997 10:37:23 +0700 (JVT)
From: Albertus Agung Wardana <albertaw@elang.stts.ac.id>
To: Internet-Drafts@ietf.org
cc: rem-conf@es.net
Subject: Re: I-D ACTION:draft-ietf-avt-rtp-payload-03.txt
In-Reply-To: <9703200956.aa13691@ietf.org>
Message-ID: <Pine.LNX.3.93.970322103718.12452L-100000@elang.stts.ac.id>
MIME-Version: 1.0
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On Thu, 20 Mar 1997 Internet-Drafts@ietf.org wrote:

>  A Revised Internet-Draft is available from the on-line Internet-Drafts 
>  directories. This draft is a work item of the Audio/Video Transport 
>  Working Group of the IETF.                                                
> 
>        Title     : RTP Payload Format for H.263 Video Streams              
>        Author(s) : C. Zhu
>        Filename  : draft-ietf-avt-rtp-payload-03.txt
>        Pages     : 10
>        Date      : 03/19/1997
> 
> This document specifies the payload format for encapsulating an H.263 
> bitstream in the Real-Time Transport Protocol (RTP). Three modes are 
> defined for the H.263 payload header. An RTP packet can use one of the 
> three modes for H.263 video streams depending on the desired network packet
> size and H.263 encoding options employed.  The shortest H.263 payload 
> header (mode A) supports fragmentation at Group of Block (GOB) boundaries. 
> The long H.263 payload headers (mode B and C) support fragmentation at 
> Macroblock (MB) boundaries.                                                
> 
> Internet-Drafts are available by anonymous FTP.  Login with the username
> "anonymous" and a password of your e-mail address.  After logging in,
> type "cd internet-drafts" and then
>      "get draft-ietf-avt-rtp-payload-03.txt".
> A URL for the Internet-Draft is:
> ftp://ds.internic.net/internet-drafts/draft-ietf-avt-rtp-payload-03.txt
>  
> Internet-Drafts directories are located at:	
> 	                                                
>      o  Africa:  ftp.is.co.za                    
> 	                                                
>      o  Europe:  ftp.nordu.net            	
>                  ftp.nis.garr.it                 
> 	                                                
>      o  Pacific Rim: munnari.oz.au               
> 	                                                
>      o  US East Coast: ds.internic.net           
> 	                                                
>      o  US West Coast: ftp.isi.edu               
> 	                                                
> Internet-Drafts are also available by mail.	
> 	                                                
> Send a message to:  mailserv@ds.internic.net. In the body type: 
>      "FILE /internet-drafts/draft-ietf-avt-rtp-payload-03.txt".
> 							
> NOTE: The mail server at ds.internic.net can return the document in
>       MIME-encoded form by using the "mpack" utility.  To use this
>       feature, insert the command "ENCODING mime" before the "FILE"
>       command.  To decode the response(s), you will need "munpack" or
>       a MIME-compliant mail reader.  Different MIME-compliant mail readers
>       exhibit different behavior, especially when dealing with
>       "multipart" MIME messages (i.e., documents which have been split
>       up into multiple messages), so check your local documentation on
>       how to manipulate these messages.
> 							
> 							
> 
> Below is the data which will enable a MIME compliant mail reader 
> implementation to automatically retrieve the ASCII version
> of the Internet-Draft.
> 


From rem-conf-request@es.net Sat Mar 22 11:01:06 1997 
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From: ShouJen <sjn@walsin.com.my>
To: videophone@es.net, rem-conf@es.net
Date: Sat, 22 Mar 1997 22:54:16 +0000
MIME-Version: 1.0
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Subject: Video Format
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X-mailer: Pegasus Mail for Win32 (v2.42a)

Hi everybody,

This probably sounds like a newbie question to you guys, pls put in 
your comments.

In videoconferencing, video codec plays a huge role.  I learn that 
there is quite a number of video codec technology around today, such 
as H.261 for standard interoperability, M-JPEG, Indeo, etc.  As the 
camera captures the frames and before the frames are compressed, what 
sort of video format is used?

What's the relationship between H.261 and CIF?  Can I say CIF files 
are compression files of H.261?

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



From rem-conf-request@es.net Sat Mar 22 13:18:57 1997 
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To: ru-comp-dev-ietf-rem-conf@rutgers.edu
Path: usenet
From: sjn@walsin.com.my (ShouJen)
Newsgroups: ru.comp.dev.ietf.rem-conf
Subject: Video Format
Date: 22 Mar 1997 13:18:21 -0500
Organization: Rutgers University
Lines: 21
Sender: news@RUTGERS.EDU
Message-ID: <5h17pd$bt1$1@rutgers.rutgers.edu>

Hi everybody,

This probably sounds like a newbie question to you guys, pls put in 
your comments.

In videoconferencing, video codec plays a huge role.  I learn that 
there is quite a number of video codec technology around today, such 
as H.261 for standard interoperability, M-JPEG, Indeo, etc.  As the 
camera captures the frames and before the frames are compressed, what 
sort of video format is used?

What's the relationship between H.261 and CIF?  Can I say CIF files 
are compression files of H.261?

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



From rem-conf-request@es.net Sat Mar 22 21:13:57 1997 
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          Sat, 22 Mar 1997 21:15:23 -0500 (EST)
Message-ID: <3334AE64.1BFD@BellSouth.Net>
Date: Sat, 22 Mar 1997 20:15:33 -0800
From: Gary O'Neal <goneal@comm.net>
Reply-To: goneal@comm.net
Organization: CommTech
X-Mailer: Mozilla 3.01 (Win95; I; 16bit)
MIME-Version: 1.0
To: videophone@es.net, rem-conf@es.net
Subject: DTVC Co Research
Content-Type: text/plain; charset=us-ascii
Content-Transfer-Encoding: 7bit

To the Group:
Over the past year, I, personally, have had two very disagreeable
employment/representative relationships with start-up videoconferencing
equipment companies:

1. Visual Communications Network, Ltd (VCN) (NASDAQ/OTC-BB: BNET) (or
Intermedia Net, Inc)

2. Visual Telephone International(VTI) (NASDAQ/OTC-BB: VTPI)

Since both of these companies have exhibited at DTVC and COMNET in the
Last year, I am curious as to whether or not any of the group members
had any contact with either or both. To any investors/potential
investors who read these lists, how were you handled by the corporate
management? To the suppliers, carriers and DTVC OEMs of VCN, I'm doing
some research so any info would be appreciated....Anonymous remailers
will be accepted. As both of these organs have publically stated that
they provide the "Most Advanced" DTVC equipment available on the market,
I figure that some group members may have run across them.

Thanks
Gary

From rem-conf-request@es.net Sat Mar 22 23:14:40 1997 
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To: ru-comp-dev-ietf-rem-conf@rutgers.edu
Path: usenet
From: goneal@comm.net ("Gary O'Neal")
Newsgroups: ru.comp.dev.ietf.rem-conf
Subject: DTVC Co Research
Date: 22 Mar 1997 23:13:53 -0500
Organization: CommTech
Lines: 22
Sender: news@RUTGERS.EDU
Message-ID: <3334AE64.1BFD@BellSouth.Net>

To the Group:
Over the past year, I, personally, have had two very disagreeable
employment/representative relationships with start-up videoconferencing
equipment companies:

1. Visual Communications Network, Ltd (VCN) (NASDAQ/OTC-BB: BNET) (or
Intermedia Net, Inc)

2. Visual Telephone International(VTI) (NASDAQ/OTC-BB: VTPI)

Since both of these companies have exhibited at DTVC and COMNET in the
Last year, I am curious as to whether or not any of the group members
had any contact with either or both. To any investors/potential
investors who read these lists, how were you handled by the corporate
management? To the suppliers, carriers and DTVC OEMs of VCN, I'm doing
some research so any info would be appreciated....Anonymous remailers
will be accepted. As both of these organs have publically stated that
they provide the "Most Advanced" DTVC equipment available on the market,
I figure that some group members may have run across them.

Thanks
Gary

From rem-conf-request@es.net Sun Mar 23 12:22:36 1997 
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Date: Sun, 23 Mar 1997 09:21:07 -0800
From: Michael.Speer@Eng.Sun.COM (Michael F. Speer)
Message-Id: <199703231721.JAA20425@valathar.eng.sun.com>
To: rem-conf@es.net
Subject: AVT meetings at April IETF
Cc: Michael.Speer@Eng.Sun.COM


How many times are we meeting in Memphis?

Thanks,
Michael

From rem-conf-request@es.net Sun Mar 23 14:57:38 1997 
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Path: usenet
From: Michael.Speer@Eng.Sun.COM (Michael F. Speer)
Newsgroups: ru.comp.dev.ietf.rem-conf
Subject: AVT meetings at April IETF
Date: 23 Mar 1997 14:56:43 -0500
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Message-ID: <199703231721.JAA20425@valathar.eng.sun.com>


How many times are we meeting in Memphis?

Thanks,
Michael

From rem-conf-request@es.net Sun Mar 23 21:20:27 1997 
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Date: Sun, 23 Mar 1997 20:19:49 -0600 (CST)
From: Marty Hoag <hoag@plains.nodak.edu>
To: rem-conf@es.net
Subject: Test Message
Message-ID: <Pine.SOL.3.91.970323201549.10178K-100000@plains>
MIME-Version: 1.0
Content-Type: TEXT/PLAIN; charset=US-ASCII

   This is only a test. 

   I've been getting duplicates of rem-conf mail this weekend which seems 
to be coming from Rutgers.  This message is being sent at the request of 
the postmaster there.    

   Thank you.  Marty Hoag  hoag@plains.nodak.edu

From rem-conf-request@es.net Mon Mar 24 04:52:54 1997 
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From: Dermot Tynan <dtynan@wfa.digital.ie>
Message-Id: <9703240952.AA28316@karpov.wfa.digital.ie>
Subject: Re: Video Format
To: sjn@walsin.com.my
Date: Mon, 24 Mar 1997 09:52:17 +0000 (GMT)
Cc: videophone@es.net, rem-conf@es.net
In-Reply-To: <33344fbb.walsin@walsin.com.my> from "ShouJen" at Mar 22, 97 10:54:16 pm
Mime-Version: 1.0
Content-Type: text/plain; charset=US-ASCII
Content-Transfer-Encoding: 7bit

ShouJen wrote:
> 
> In videoconferencing, video codec plays a huge role.  I learn that 
> there is quite a number of video codec technology around today, such 
> as H.261 for standard interoperability, M-JPEG, Indeo, etc.  As the 
> camera captures the frames and before the frames are compressed, what 
> sort of video format is used?

The video format prior to compression is by and large irrelevant, except
in how it pertains to the size and quality of the compressed image.  The
most common format is CCIR 601, which is the international broadcast
standard for digital representation of video signals.  It uses an
8-bit luminance and two 8-bit chrominance channels, with a standard
sample rate (13.5MHz) regardless of whether the signal is NTSC-encoded
or PAL-encoded.  In most cases, the chrominance channels are subsampled
at half the luminance rate, known as 4:2:2.  The frame size is usually
quoted as 720x480x29.97 or 720x576x25, representing the number of
horizontal lines, vertical lines, and frames per second.  The first
is NTSC and the second is PAL.  Yes, the NTSC standard does not use
an integer number of frames per second.  On the other hand, if you have
a quickcam or some sort of digital CCD mini camera, the data may never
start out in NTSC/PAL, and can be delivered directly to the codec in
its native format.

> What's the relationship between H.261 and CIF?  Can I say CIF files 
> are compression files of H.261?

H.261 is a DCT-based compression standard specifically for video
conferencing.  CIF on the other hand, is the Common Image Format
as defined by CCIR 601.  In answer to your second question, you
can say that a H.261 file is the compressed version of a CIF file.
						- Der
-- 
Dermot Tynan						+353 91 754608
dtynan@wfa.digital.ie					 DTN: 822-4608

AltaVista Internet Software, Galway, Ireland

From rem-conf-request@es.net Mon Mar 24 10:53:23 1997 
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Reply-To: oka@kobe-u.ac.jp
To: rem-conf@es.net, mbone@ISI.EDU
cc: infocom411@ckp.or.jp
Subject: Broadcasting of Infocom'97
Mime-Version: 1.0
Content-Type: text/plain;charset="ISO-2022-JP"
Date: Tue, 25 Mar 1997 00:56:27 +0900
From: Koji OKAMURA <oka@nanotsu.kobe-u.ac.jp>

Cyber Kansai is pleased to announce the MBONE broadcast of Infocom'97
>from Kobe on April 7-11,1997.  Including Gigabit Network Workshop,
keynotes by Vinton Serf (MCI) and Toshiharu Aoki (NTT), panels and
sessions. Broadcasting of recorded material is also available for US
and Europe.

 Date: 
	7-11 April, 1997
 
 Time: 4/7 (Mon) 9:00 - 17:00 GMT+9 and 14:00 - 22:00 PST (recorded)
	Gigabit Networking Workshop
       4/9 (Wed) 8:00 - 17:00 GMT+9 and 13:00 - 22:00 PST (recorded)
	Keynotes, Sessions and Panel
       4/10 (Thu) 8:30 - 17:00 GMT+9 and 13:30 - 22:00 PST (recorded)
	Sessions and Panel
       4/11 (Fri) 8:30 - 17:00 GMT+9 and 13:30 - 22:00 PST (recorded)
        Sessions and Panel
 
 Title:
      	Infocom' 97 from Kobe, Japan 
 
 URL:
	http://www.ieee.org/comsoc/infocom.
  

 Contact:     
	oka@kobe-u.ac.jp

Best regards,
	  	Cyber Kansai (http://www.ckp.or.jp).

From rem-conf-request@es.net Mon Mar 24 12:07:18 1997 
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Date: Mon, 24 Mar 1997 06:19:01 -0500
From: "H. DO-DUY" <106025.3357@compuserve.com>
Subject: Vat sources for windows
To: vat <rem-conf@es.net>
Message-ID: <199703240619_MC2-132B-481A@compuserve.com>

Hello,

        I'm trying to make an ethernet conferencing tool based upon VAT
4.0b2. I've retrieved the appropriate archives, and tried to undersant the
code, but it seems to be quite difficult, because it is very few commented.
Does anybody know where i could find some documentation on the vat coding ?
I tried to e-mail the people who created vat, but i can't get any reply.

Thanks a lot.

Rodolphe PETIT
SEEE
24, march, 1997, Toulouse, FRANCE
E-mail : 106025.3357@compuserve.com


From rem-conf-request@es.net Mon Mar 24 12:52:43 1997 
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From: Vinay Bannai <vinay@agni.nuko.com>
Message-Id: <199703241751.JAA28798@agni.nuko.com>
Subject: Re: Video Format
To: sjn@walsin.com.my (ShouJen)
Date: Mon, 24 Mar 1997 09:51:21 -0800 (PST)
Cc: ru-comp-dev-ietf-rem-conf@rutgers.edu
In-Reply-To: <5h17pd$bt1$1@rutgers.rutgers.edu> from "ShouJen" at Mar 22, 97 01:18:21 pm
X-Mailer: ELM [version 2.4 PL24]
Content-Type: text

According to ShouJen:
> 
> Hi everybody,
> 
> This probably sounds like a newbie question to you guys, pls put in 
> your comments.
> 
> In videoconferencing, video codec plays a huge role.  I learn that 
> there is quite a number of video codec technology around today, such 
> as H.261 for standard interoperability, M-JPEG, Indeo, etc.  As the 
> camera captures the frames and before the frames are compressed, what 
> sort of video format is used?

It depends on the encoding device used and the country you are in. In US,
it is NTSC, in most parts of Europe it is PAL (France uses SECAM, I
think). NTSC uses 30 frames/second and PAL uses 24 frames/second.

> 
> What's the relationship between H.261 and CIF?  Can I say CIF files 
> are compression files of H.261?

CIF stands for Common Intermediate Format (CIF). This a just a video
source input format. There are others too, like SIF and CCIR601. For
example, if you are using CIF video input, typically it means you are
using 352x288 resolution. It has nothing to do with compression and is not
the compression file.

H.261 uses CIF and QCIF (quarter CIF) video source input formats.
Mpeg-1 uses SIF (source input format) which uses 525x625 resolution.

Hope this helps.
-- 
Vinay Bannai                     E-mail: vinay@agni.nuko.com
(408)-526-0280 x 275 (Work)     http://agni.nuko.com/~vinay


From rem-conf-request@es.net Mon Mar 24 13:40:51 1997 
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Date: Mon, 24 Mar 1997 10:39:38 -0800
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To: rem-conf@es.net
From: Jerrlyn Iwata <jerrlyn@postgres.Berkeley.EDU>
Subject: [Reminder] Berkeley Multimedia & Graphics Seminar

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

                "PC vs. TV : The Battle for Eyeballs" 

                        Stephen C. Beck 
                    Focus Enhancements Inc. 

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


From rem-conf-request@es.net Mon Mar 24 14:28:09 1997 
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X-Priority: 1 (Highest)
Date: Mon, 24 Mar 1997 01:43:21 -1812
To: ts@okcom.net
From: ts@okcom.net
Subject: a very interesting free offer


From rem-conf-request@es.net Mon Mar 24 14:47:42 1997 
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          Mon, 24 Mar 1997 14:43:15 -0500
Date: Mon, 24 Mar 1997 14:43:15 -0500
From: rodgers@nlm.nih.gov (R. P. C. Rodgers)
Message-Id: <199703241943.OAA01443@billings.csb>
To: rem-conf@es.net, mbone@ISI.EDU
Subject: 7-11 April is going to be tight on the MBONE (Was: Broadcasting of 
         Infocom'97)
Cc: infocom411@ckp.or.jp, oka@kobe-u.ac.jp, dem@hep.net, likavec@igd.fhg.de
X-Sun-Charset: US-ASCII


Dear Colleagues,

The announcement from Kobe indicates that there will be at least four
conferences of international scope contending for MBONE bandwidth during
the week of 7-11 April.  WWW6 and CHEP '97 have announced to this group
many weeks ago and are actively advertising at the moment via sd/sdr.
I don't remember seeing a posting to these mail lists from the IETF,
but it is widely known that they plan on multicasting their traditional
two concurrent sessions that week.  We at WWW6 had also hoped to send
out two parallel tracks.  CHEP and Kobe may be using slightly different
time slots, but add all of this together and it is going to be a rather
tight week, requiring lots of coordination and good humor all around.

This highlights several broader issues:

1) Scheduling and priorities.  I know that we started advertising on
   the various MBONE calendars, these mail lists, and with sdr many
   weeks back.  CHEP started doing so about 7-10 days ago.  I may
   have missed email to these lists from IETF, but they are still not
   on sdr.  Kobe just announced today.  I am perfectly willing to work
   as we have in the past, trying best to set up machine and telephone
   communications between these four groups to try to coordinate to
   keep to the ancient and honorable "gentleperson's agreement" of not
   using more total bandwidth that the total of two concurrent A/V
   sessions, and treating the 4 events as "peers," regardless of how
   or when they announced their plans publically, but based on past
   events, just trying to coordinate two meetings can be a tricky task.
   If we have many more weeks like 7-11 April, are we going to have to
   get more serious about enforcing some rules of priority, based on
   advanced reservations, and perhaps some discussion of relative
   priorities?  I'd hate to get into this business, but will it be
   necessary?

2) Discussion of how international conferences might create temporary
   CBONEs (Conference backbones) that might allow concurrent conferences
   to occur with less mutual interference, perhaps getting their
   traffic off of the "public" Internet backbones.  This might even be
   coupled with (shudder) participation fees for remote sites to defray
   conference costs (of course, charges would be hard to require in the
   absence of some reasonable guaranteed level of service).  In
   principle, I would assume that widespread implementation of the
   appropriate multicast routing standards would obviate the need for
   a "CBONE" (or not?).

Some of these ideas might be pure crock (I'm not pushing them, just
trying to engage some of our better minds in discussing the underlying
issues!).  Perhaps some fodder hear from our colleagues at IETF
beer BOFs, if nothing else... 8^)


Best Regards (and drink a beer for me!), Rick Rodgers

> From oka@nanotsu.kobe-u.ac.jp Mon Mar 24 11:26:48 1997
> Reply-To: oka@kobe-u.ac.jp
> To: rem-conf@es.net, mbone@ISI.EDU
> cc: infocom411@ckp.or.jp
> Subject: Broadcasting of Infocom'97
> Mime-Version: 1.0
> Date: Tue, 25 Mar 1997 00:56:27 +0900
> From: Koji OKAMURA <oka@nanotsu.kobe-u.ac.jp>
> 
> Cyber Kansai is pleased to announce the MBONE broadcast of Infocom'97
> from Kobe on April 7-11,1997.  Including Gigabit Network Workshop,
> keynotes by Vinton Serf (MCI) and Toshiharu Aoki (NTT), panels and
> sessions. Broadcasting of recorded material is also available for US
> and Europe.
> 
>  Date: 
> 	7-11 April, 1997
>  
>  Time: 4/7 (Mon) 9:00 - 17:00 GMT+9 and 14:00 - 22:00 PST (recorded)
> 	Gigabit Networking Workshop
>        4/9 (Wed) 8:00 - 17:00 GMT+9 and 13:00 - 22:00 PST (recorded)
> 	Keynotes, Sessions and Panel
>        4/10 (Thu) 8:30 - 17:00 GMT+9 and 13:30 - 22:00 PST (recorded)
> 	Sessions and Panel
>        4/11 (Fri) 8:30 - 17:00 GMT+9 and 13:30 - 22:00 PST (recorded)
>         Sessions and Panel
>  
>  Title:
>       	Infocom' 97 from Kobe, Japan 
>  
>  URL:
> 	http://www.ieee.org/comsoc/infocom.
>   
> 
>  Contact:     
> 	oka@kobe-u.ac.jp
> 
> Best regards,
> 	  	Cyber Kansai (http://www.ckp.or.jp).
> 

From rem-conf-request@es.net Mon Mar 24 17:50:12 1997 
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From: Mark Handley <mjh@isi.edu>
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To: rodgers@nlm.nih.gov (R. P. C. Rodgers)
cc: rem-conf@es.net, oka@kobe-u.ac.jp, dem@hep.net, likavec@igd.fhg.de
Subject: Re: 7-11 April is going to be tight on the MBONE (Was: Broadcasting of 
         Infocom'97)
In-reply-to: Your message of "Mon, 24 Mar 1997 14:43:15 EST." <199703241943.OAA01443@billings.csb>
Date: Mon, 24 Mar 1997 17:45:37 -0500
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Sender: mjh@buttle.lcs.mit.edu


>The announcement from Kobe indicates that there will be at least four
>conferences of international scope contending for MBONE bandwidth during
>the week of 7-11 April.

Fortunately they're all in different timezones, so the overlap is not
as bad as it could be unless you want to listen to all of them.  The
problem is the rebroadcasts that people are discussing.  These could
always be moved to a different week - the potential audience is likely
to be larger if it's not split between sessions.

According to all the ISPs present at the MBoned WG at the last IETF,
no major ISP is running rate-limiting anymore, and we've regularly
seen traffic rates above the old nominal 512Kbps.  The Mbone is slowly
growing up.  It *should* be able to handle four simultaneous
conferences.  But lets not push our luck...

Mark

From rem-conf-request@es.net Mon Mar 24 18:48:23 1997 
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Date: Mon, 24 Mar 1997 18:47:38 -0500 (EST)
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To: rodgers@nlm.nih.gov (R. P. C. Rodgers), rem-conf@es.net, mbone@ISI.EDU
From: sprinter@infobahnos.com
Subject: QUERY: Publicly accessible MBone sites
Cc: infocom411@ckp.or.jp, oka@kobe-u.ac.jp, dem@hep.net, likavec@igd.fhg.de


Dear list members,

Are there any publicly accessible MBone sites?
I know of private networks using and broadcasting on it but was wondering
if there are any universities, research centers or organizations who
generously offer access to broadcast of tune in to the MBone from their
offices.

--
JOSEE M. PROULX
Sprinter Consulting
3601 Ste-Famille St. Ste 1003
Montreal, Quebec H2X 2L6
1-514-844-5304 (telephone)
1-514-844-5133 (facsimile)
sprinter@infobahnos.com (e-mail)



From rem-conf-request@es.net Tue Mar 25 02:47:19 1997 
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Date: Mon, 24 Mar 1997 23:44:11 -0800 (PST)
From: Stephen Casner <casner@precept.com>
Reply-To: Stephen Casner <casner@precept.com>
To: "Michael F. Speer" <Michael.Speer@Eng.Sun.COM>
cc: rem-conf@es.net
Subject: Re: AVT meetings at April IETF
In-Reply-To: <199703231721.JAA20425@valathar.eng.sun.com>
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I received word today that the AVT schedule for the Memphis meeting
has been changed.  Our time is now all on Wednesday, which is the day
chopped into shorter 1-hour slots.  We have three assigned:

    Wednesday Morning I  (9:00-10:00)
    Wednesday Evening I and Evening II (contiguous, 20:00-22:00)

This is one hour less than we have had for recent meetings, but it
seems that we may need less time at this meeting.  I received a few
requests for agenda items (MIB, H.263, MPEG); are there others?

There has not been as much activity since the last meeting as I
anticipated, especially with regard to specification of additional
profiles for other RTCP scenarios.  It may be that some of the more
complicated ideas have fatal problems or simply require more time to
develop -- more research than engineering.  Bernard Aboba responded
with a suggestion that perhaps the only new profile we need is one
that allows the RTCP bandwidth fraction to be specified, and my
response was to wonder where that parameter would be signalled.

Based on a recent conversation with Mark Handley, I think that
instead of defining a new profile, perhaps all we need is to change
the current profile to establish 5% as the default, but to allow the
fraction to be specified as a session parameter.  For example, in SDP
where there is now a "session bandwidth" attribute, we could define
an additional, separate attribute to specify the RTCP bandwidth.  In
fact, because one may want to reduce the bandwidth for reception
reports to zero, but one would usually still want sender reports to
establish the CNAME and provide synchronization timestamps, Mark
suggested an attribute with two paramaters: one giving the RTCP
bandwidth for data senders, and a second giving the RTCP bandwidth
for receivers.  (Currently, senders get at least 1/4 of the 5% BW).

This would cover the applications Bernard wanted (some with a low
feedback BW fraction, and some with none at all).  For sessions that
might adapt the coding over a large BW range, this separate attribute
would also allow specifying an RTCP BW corresponding to a nominal
session BW that might be significantly below the max BW given by the
existing attribute for the overall session BW.

Beyond this, I think the outcome of the discussion at the last
meeting is that "timer reconsideration" should be added to the
interval calculation algorithm to manage simultaneous joins.  I also
believe the current fixed minimum interval value of 5 seconds and
corresponding initial interval value of 2.5 seconds should be scaled
with the RTCP bandwidth so that there is enough time for timer
reconsideration to work on low bandwidth lines.  These things would
just be edits to the RTP spec before Draft Standard.

Do others agree with this summary of where we are?  Are there issues
to be discussed (though let's not simply repeat the discussion from
last time, please)?  Other new ideas?
							-- Steve


From rem-conf-request@es.net Tue Mar 25 02:54:17 1997 
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Date: Tue, 25 Mar 1997 02:53:26 -0500
To: rem-conf@es.net
From: Bill Immerman <immerman@darlington.com>
Subject: subscribe

Please subscribe me


From rem-conf-request@es.net Tue Mar 25 03:21:58 1997 
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    'Stephen Casner' <casner@precept.com>
Cc: "rem-conf@es.net" <rem-conf@es.net>
Subject: RE: AVT meetings at April IETF
Date: Tue, 25 Mar 1997 00:16:56 -0800
Encoding: 7 TEXT

The idea of separate attributes for sender and receiver report bandwidths
is a good one. 

Question: will this bandwidth be specified as an absolute number, or as
a fraction of 5 percent? The fraction might be better security-wise so that
someone could not increase the bandwidth unreasonably.  



From rem-conf-request@es.net Tue Mar 25 03:26:28 1997 
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Subject: Re: Video Format
To: vinay@agni.nuko.com (Vinay Bannai)
Date: Tue, 25 Mar 1997 09:25:30 +0100 (MET)
Cc: sjn@walsin.com.my, ru-comp-dev-ietf-rem-conf@rutgers.edu
In-Reply-To: <199703241751.JAA28798@agni.nuko.com> from "Vinay Bannai" at Mar 24, 97 09:51:21 am
Organisation: CSD IMMD IV, University of Erlangen, Germany
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> CIF stands for Common Intermediate Format (CIF). This a just a video
> source input format. There are others too, like SIF and CCIR601. For
> example, if you are using CIF video input, typically it means you are
> using 352x288 resolution. It has nothing to do with compression and is not
> the compression file.

Well, i think that CIF does too imply to have non-square-pixel resolution,
i.e.: a CIF-picture is supposed to transport a video picture that has
a square-pixel resolution of 375x288. This is often not done correctly
(i.e.: i think vic and a lot of other videoconferencing applications
 actually transport square pixel-video-content in the 352x288 resolution).

From rem-conf-request@es.net Tue Mar 25 07:56:52 1997 
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To: rem-conf@es.net
cc: Claude.Castelluccia@imag.fr
Subject: Question about IPMulticast!
Date: Tue, 25 Mar 1997 13:55:33 +0100
From: Claude Castelluccia <Claude.Castelluccia@imag.fr>


When a host joins a multicast group, it takes some time
before this host starts receiving data from this group?
How long does it really take? Are they studies
 that evaluate or measure this time?

Thanks.

Claude.

From rem-conf-request@es.net Tue Mar 25 09:14:52 1997 
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From: kevin@cc.gatech.edu (Kevin C. Almeroth)
Message-Id: <199703251414.JAA10541@flora.cc.gatech.edu>
To: rem-conf@es.net
Subject: Re: 7-11 April is going to be tight on the MBONE
Cc: rodgers@nlm.nih.gov

>>The announcement from Kobe indicates that there will be at least four
>>conferences of international scope contending for MBONE bandwidth during
>>the week of 7-11 April.  WWW6 and CHEP '97 have announced to this group
>>many weeks ago and are actively advertising at the moment via sd/sdr.

>>1) Scheduling and priorities.

This is a BIG can of worms.  I suggest we consider all four to be equally
important.  Is anything else really an option?

   * Trying to have a discussion of which is more important is
     going to be useless and among other things, how do you enforce it?
     Telling Infocom not to broadcast is a bit laughable.

   * Using when an advertisement was made as a scheduling priority isn't
     going to work.  Ever play the who-gets-the-front-seat-in-the-car game?  
     After a while you start having to make rules about how far in advance 
     you are going to allow reservations, etc.

My suggestion is if you/we REALLY want to do any limiting then start up
a whiteboard session and have a discussion about what times will be needed
to broadcast live.  As Mark suggested, the time zones should help a lot.

I also think re-broadcasts should be delayed until some time later.

-------------------- Cut here to ignore personal plug --------------------

Actually, what about making the sessions available via the IMJ?  :-)
(That's http://imj.gatech.edu for those who haven't seen/heard.)
This would work especially well for the IETF sessions since they could
be chopped into per-agenda-item pieces.  For conferences you could do
per-speaker segments.

We've thought about doing this in the past but some of the IETF sessions
seemed pretty dry especially compared to Bugs!  But there certainly could
an actual need.  I'm pretty sure Evi still tapes the sessions and these
could be encoded and put on the IMJ.

-Kevin Almeroth

From rem-conf-request@es.net Tue Mar 25 09:34:00 1997 
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From: Sneha Kumar Kasera <kasera@cs.umass.edu>
Message-Id: <199703251431.JAA08513@giane.cs.umass.edu>
Subject: Re: Question about IPMulticast!
To: Claude.Castelluccia@imag.fr (Claude Castelluccia)
Date: Tue, 25 Mar 1997 09:31:05 -0500 (EST)
Cc: rem-conf@es.net, Claude.Castelluccia@imag.fr
In-Reply-To: <199703251255.NAA17221@guadalupe.inrialpes.fr> from "Claude Castelluccia" at Mar 25, 97 01:55:33 pm
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You could  look at "Performance and Resource Cost Comparisons
for the CBT and PIM Multicast Routing Protocols in DIS Environments"
by T.Billhartz and others. I think this paper appeared in 
Infocom 1996. 

Sneha

> 
> 
> When a host joins a multicast group, it takes some time
> before this host starts receiving data from this group?
> How long does it really take? Are they studies
>  that evaluate or measure this time?
> 
> Thanks.
> 
> Claude.
> 


From rem-conf-request@es.net Tue Mar 25 10:25:32 1997 
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Date: Tue, 25 Mar 1997 07:23:36 -0800
To: Claude Castelluccia <Claude.Castelluccia@imag.fr>
From: Steve Deering <deering@cisco.com>
Subject: Re: Question about IPMulticast!
Cc: rem-conf@es.net, Claude.Castelluccia@imag.fr

At 4:55 AM -0800 3/25/97, Claude Castelluccia wrote:
> When a host joins a multicast group, it takes some time
> before this host starts receiving data from this group?
> How long does it really take? Are they studies
>  that evaluate or measure this time?

For shortest-path-style multicast routing protocols like DVMRP or MOSPF,
the worst case delay is roughly the round-trip time from the receiver to
the sender.  For shared-tree-style protocols like PIM-SM and CBT, worst
case is the RTT from receiver to RP/core, plus RTT from RP/core to source.
Those are the worst case; it is often better than that, i.e., the RTT
>from the receiver to the nearest point on the existing multicast delivery
tree.  That's the theory; I am not aware of any measurement results that
show that reality matches theory.

Steve



From rem-conf-request@es.net Tue Mar 25 12:07:39 1997 
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To: Steve Deering <deering@cisco.com>
cc: Claude.Castelluccia@imag.fr, rem-conf@es.net
Subject: Re: Question about IPMulticast!
In-reply-to: Your message of "Tue, 25 Mar 1997 07:23:36 PST." <v03020900af5d9c90e89a@[171.69.199.124]>
Date: Tue, 25 Mar 1997 18:06:03 +0100
From: Claude Castelluccia <Claude.Castelluccia@imag.fr>


>For shortest-path-style multicast routing protocols like DVMRP or MOSPF,
>the worst case delay is roughly the round-trip time from the receiver to
>the sender.  For shared-tree-style protocols like PIM-SM and CBT, worst
>case is the RTT from receiver to RP/core, plus RTT from RP/core to source.
>Those are the worst case; it is often better than that, i.e., the RTT
>from the receiver to the nearest point on the existing multicast delivery
>tree.  That's the theory; I am not aware of any measurement results that
>show that reality matches theory.

Thanks for your answer, but I think that 2 additional componants should be added 
to your estimation:

 t1- When a host wants to join a group, it musts wait for the next Query
 sent by the Designated Router. This might take t1.

 t2- Then if this mulicast router is not part of the tree, its must wait until
    the pruned branches grow back. This might take t2.

t1 could be minimized by having hosts sending join queries to their designated 
router instead of waiting for the router's query.
t2 could be minimized by having designated routers sending messages to the mrouters 
toward the source to make the pruned branches grow back faster. 
Do those options exist? Are implemented and used? Am I missing something?

Thanks a lot!

Claude.

From rem-conf-request@es.net Tue Mar 25 12:35:02 1997 
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References: Your message of "Tue,
            25 Mar 1997 07:23:36 PST." <v03020900af5d9c90e89a@[171.69.199.124]>
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Date: Tue, 25 Mar 1997 09:33:13 -0800
To: Claude Castelluccia <Claude.Castelluccia@imag.fr>
From: Steve Deering <deering@cisco.com>
Subject: Re: Question about IPMulticast!
Cc: Claude.Castelluccia@imag.fr, rem-conf@es.net

At 9:06 AM -0800 3/25/97, Claude Castelluccia wrote:
> Thanks for your answer, but I think that 2 additional componants should
>be added
> to your estimation:
>
>  t1- When a host wants to join a group, it musts wait for the next Query
>  sent by the Designated Router. This might take t1.

That is incorrect.  When a host first joins a group, it immediately sends
a couple of IGMP Reports for that group; it does not wait for the next
IGMP Query.

>  t2- Then if this mulicast router is not part of the tree, its must wait
>      until the pruned branches grow back. This might take t2.

That is also incorrect (and a widely-held misconception, it seems).  With
both DVMRP and PIM-DM, when a member of a new group appears on a subnet
that has previously been pruned for that group, the router(s) on that
subnet immediately send "graft" messages to undo the pruning.

So, yes, both of the options that you suggested for reducing join latency
have always been part of the multicast routing machinery.  Thus my original
answer to you: the join latency is at worst the RTT from receiver to source
(ignoring the hop-by-hop processing delays, which are negligible on paths
of any significant distance).

Steve



From rem-conf-request@es.net Tue Mar 25 12:48:12 1997 
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To: rem-conf-request@es.net, deering@cisco.com
cc: Claude.Castelluccia@imag.fr, rem-conf@es.net
Subject: Re[2]: Question about IPMulticast!


Text item: 

     

>For shortest-path-style multicast routing protocols like DVMRP or MOSPF, 
>the worst case delay is roughly the round-trip time from the receiver to 
>the sender.  For shared-tree-style protocols like PIM-SM and CBT, worst 
>case is the RTT from receiver to RP/core, plus RTT from RP/core to source. 
>Those are the worst case; it is often better than that, i.e., the RTT 
>from the receiver to the nearest point on the existing multicast delivery 
>tree.  That's the theory; I am not aware of any measurement results that 
>show that reality matches theory.
     
Thanks for your answer, but I think that 2 additional componants should 
be added
to your estimation:
     
 t1- When a host wants to join a group, it musts wait for the next Query 
 sent by the Designated Router. This might take t1.
     
 t2- Then if this mulicast router is not part of the tree, its must wait until
    the pruned branches grow back. This might take t2.
     
t1 could be minimized by having hosts sending join queries to their designated 
router instead of waiting for the router's query.
t2 could be minimized by having designated routers sending messages 
to the mrouters
toward the source to make the pruned branches grow back faster.
Do those options exist? 
Yes. For DVMRP, if this mulicast router is not part of the tree, it will send a 
GRAFT to its upstream neighbor. So t2 is not necessary.

Are implemented and used? Am I missing something?
     
Thanks a lot!
     
Claude.

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From: Claude Castelluccia <Claude.Castelluccia@imag.fr>
Date: Tue, 25 Mar 1997 18:06:03 +0100
In-reply-to: Your message of "Tue, 25 Mar 1997 07:23:36 PST." <v03020900af5d9c90
e89a@[171.69.199.124]>
Subject: Re: Question about IPMulticast!
cc: Claude.Castelluccia@imag.fr, rem-conf@es.net
To: Steve Deering <deering@cisco.com>
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From rem-conf-request@es.net Tue Mar 25 13:03:06 1997 
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From: Matt Crawford <crawdad@FNAL.GOV>
Subject: Re: Question about IPMulticast!
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> Thanks for your answer, but I think that 2 additional componants
> should be added to your estimation:
> 
>  t1- When a host wants to join a group, it musts wait for the next Query
>  sent by the Designated Router. This might take t1.

No.  Read RFC1112, the middle of page 13:

   When a host joins a new group, it should immediately transmit a
   Report for that group, rather than waiting for a Query, in case it is
   the first member of that group on the network.  To cover the
   possibility of the initial Report being lost or damaged, it is
   recommended that it be repeated once or twice after short delays.

>  t2- Then if this mulicast router is not part of the tree, its must
> wait until the pruned branches grow back. This might take t2.

No.  The mechanism to avoid this delay depends on the multicast
routing protocol, but I think every such protocol includes such a
mechanism.

> Am I missing something?

Umm, access to the protocol specs?

From rem-conf-request@es.net Tue Mar 25 13:07:11 1997 
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To: Steve Deering <deering@cisco.com>
cc: Claude Castelluccia <Claude.Castelluccia@imag.fr>, rem-conf@es.net
Subject: Re: Question about IPMulticast!
In-reply-to: Your message of "Tue, 25 Mar 97 09:33:13 PST." <v03020915af5db997ba8c@[171.69.199.124]>
Date: Tue, 25 Mar 1997 10:06:07 PST
Sender: Bill Fenner <fenner@parc.xerox.com>
From: Bill Fenner <fenner@parc.xerox.com>
Message-Id: <97Mar25.100608pst.177486@crevenia.parc.xerox.com>

Steve Deering <deering@cisco.com> wrote:
>At 9:06 AM -0800 3/25/97, Claude Castelluccia wrote:
>>  t2- Then if this mulicast router is not part of the tree, its must wait
>>      until the pruned branches grow back. This might take t2.
>
>That is also incorrect (and a widely-held misconception, it seems).

Not surprising; it's in Cisco's documentation.  Cisco's web site has lots
of half-truths (or less) about DVMRP.

  Bill

From rem-conf-request@es.net Tue Mar 25 13:13:19 1997 
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FCFS priority also doesn't work since some events schedule/advertise
Mbone transmissions only after most of the registration period is over,
in the (possibly misguided) belief that they don't want the freebie
Mbone to cannibalize the paying attendance base.

From rem-conf-request@es.net Tue Mar 25 13:47:51 1997 
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Date: Tue, 25 Mar 1997 10:44:30 -0800
From: Dino Farinacci <dino@cisco.com>
Message-Id: <199703251844.KAA14481@stilton.cisco.com>
To: fenner@parc.xerox.com
CC: deering@cisco.com, Claude.Castelluccia@imag.fr, rem-conf@es.net
In-reply-to: <97Mar25.100608pst.177486@crevenia.parc.xerox.com> (message from Bill Fenner on Tue, 25 Mar 1997 10:06:07 PST)
Subject: Re: Question about IPMulticast!

>> Not surprising; it's in Cisco's documentation.  Cisco's web site has lots
>> of half-truths (or less) about DVMRP.

    This is not the case on the engineering web page. Bill, tell me where
    this bug is and I'll get it fixed.

Dino

From rem-conf-request@es.net Tue Mar 25 13:51:02 1997 
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From: Sneha Kumar Kasera <kasera@cs.umass.edu>
Message-Id: <199703251845.NAA11341@giane.cs.umass.edu>
Subject: Re: Question about IPMulticast!
To: deering@cisco.com (Steve Deering)
Date: Tue, 25 Mar 1997 13:45:00 -0500 (EST)
Cc: Claude.Castelluccia@imag.fr, rem-conf@es.net
In-Reply-To: <v03020915af5db997ba8c@[171.69.199.124]> from "Steve Deering" at Mar 25, 97 09:33:13 am
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> That is incorrect.  When a host first joins a group, it immediately sends
> a couple of IGMP Reports for that group; it does not wait for the next
> IGMP Query.
> 
> >  t2- Then if this mulicast router is not part of the tree, its must wait
> >      until the pruned branches grow back. This might take t2.
> 
> That is also incorrect (and a widely-held misconception, it seems).  With
> both DVMRP and PIM-DM, when a member of a new group appears on a subnet
> that has previously been pruned for that group, the router(s) on that
> subnet immediately send "graft" messages to undo the pruning.
> 
> So, yes, both of the options that you suggested for reducing join latency
> have always been part of the multicast routing machinery.  Thus my original
> answer to you: the join latency is at worst the RTT from receiver to source
> (ignoring the hop-by-hop processing delays, which are negligible on paths
> of any significant distance).
> 
> Steve
> 
I think that join latency could be more than RTT depending upon the processing
load at an intermediate hop unless join messages are treated with high
priority. As it needs to be received at each  hop, a join message has to go 
through additional "CPU" queueing (different from the fast H/W queues) at each
hop. One could ignore the processing delay which is of the order of a few
milliseconds but not ignore the additional queueing delays. 
The difference between "join latency" and RTT is likely to increase with 
increasing number of hops.

I had once conducted RTT experiments where packets were sent with IP options
and without IP options. The packets with IP options usually had a much
higher RTT(with more fluctuation) in comparison to packets without 
IP options. 

I understand that the processing of join messages is different from the 
processing of packets with IP options. Don't you think we could still get an 
estimate of the delay due to the "additional" queueing that takes place at 
the hops ?

Sneha

> 
> 


From rem-conf-request@es.net Tue Mar 25 14:01:10 1997 
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From: Bill Fenner <fenner@parc.xerox.com>
To: dino@cisco.com, fenner@parc.xerox.com
Subject: Re: Question about IPMulticast!
Cc: Claude.Castelluccia@imag.fr, deering@cisco.com, rem-conf@es.net
Message-Id: <97Mar25.105638pst.177486@crevenia.parc.xerox.com>
Date: Tue, 25 Mar 1997 10:56:27 PST

Well, typing "DVMRP" in the search engine on cio.cisco.com brings up at
least 2 documents with errors, and searching at
http://cio.cisco.com/univercd/home/search.htm brings up more.  A quick
survey shows

http://cio.cisco.com/warp/public/732/IP/cgmp_wp.htm
http://cio.cisco.com/warp/public/543/21.html
http://cio.cisco.com/warp/public/614/17.html
http://www.cisco.com/univ-src/ccden/data/doc/cintrnet/idg3/idgmulti.htm

but I don't really have time to go through all the documents on Cisco's
web site.  Given that I found the above in 5 minutes I bet a wider
search could find more.

  Bill

From rem-conf-request@es.net Tue Mar 25 14:34:13 1997 
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References: <v03020915af5db997ba8c@[171.69.199.124]> from "Steve Deering" at Mar 25,
            97 09:33:13 am
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To: Sneha Kumar Kasera <kasera@cs.umass.edu>
From: Steve Deering <deering@cisco.com>
Subject: Re: Question about IPMulticast!
Cc: rem-conf@es.net

> I think that join latency could be more than RTT depending upon the
>processing
> load at an intermediate hop unless join messages are treated with high
> priority.

That's why I said in my first reply that that join latency is "roughly"
the RTT, which based on the nature of the original question is close enough
to the right answer.  The questioner was apparently concerned that the
latencies would be on the order of many seconds, like the IGMP query
interval or DVMRP prune lifetime. In fact the latencies should be on the
order of the RTT (perhaps some smallish multiple), not on the order of
10s of seconds or minutes.  (It's also why I said "in theory", because
you are right that there might be some pathologically bad behavior
observable in the field.)

Steve



From rem-conf-request@es.net Tue Mar 25 14:54:31 1997 
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Subject: Re: Question about IPMulticast!
To: dino@cisco.com (Dino Farinacci)
Date: Tue, 25 Mar 1997 20:45:52 +0100 (MET)
Cc: fenner@parc.xerox.com, deering@cisco.com, Claude.Castelluccia@imag.fr, 
    rem-conf@es.net
In-Reply-To: <199703251844.KAA14481@stilton.cisco.com> from "Dino Farinacci" at Mar 25, 97 10:44:30 am
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>> Not surprising; it's in Cisco's documentation.  Cisco's web site has lots
>> of half-truths (or less) about DVMRP.

What's the shortest half-truth ?

      "Cisco  Does DVMRP"

(do not confuse with:
      "Debbie Does Dallas")

Sorry, couldn't resist ;-)))))

From rem-conf-request@es.net Tue Mar 25 14:55:56 1997 
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Date: Tue, 25 Mar 1997 11:28:02 -0800 ()
From: Stephen Casner <casner@precept.com>
To: Bernard Aboba <aboba@internaut.com>
cc: "rem-conf@es.net" <rem-conf@es.net>
Subject: Specifying RTCP BW (was: AVT meetings at April IETF)
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> The idea of separate attributes for sender and receiver report bandwidths
> is a good one.
> 
> Question: will this bandwidth be specified as an absolute number, or as
> a fraction of 5 percent? The fraction might be better security-wise so that
> someone could not increase the bandwidth unreasonably.

There are many different ways we could do this, and I think any of
them could be abused.  What is to keep someone from specifying a
fraction greater than 1?  Also, there may be applications where more
than 5% is appropriate.

I think be best guideline is a choice that minimizes the likelihood of
confusion.  In separate email, Mark Handley proposed:

  We have two choices for how to implement this - a bandwidth field or
  an attribute.

  Given that this is RTP-specific, and that this is relative to a
  bandwidth field, I'd probably favour an attribute.  Perhaps:
  a=rtcpbw:5 1.25

  Where the first number is the total proportion of session bandwidth to
  be used by RTCP (RR+SRs) and the second is the minimum proportion of
  the session b/w to be used by RTCP SRs.  Both numbers are percentages
  (seems more compact than fractions of the b/w).  If the two numbers
  are the same, no b/w is used by RRs.

This keeps the numbers the same as two numbers that currently appear
in the interval calculation algorithm.  It seems like a reasonable
choice to me.
							-- Steve


From rem-conf-request@es.net Tue Mar 25 14:56:53 1997 
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From: "Balanesi, Terry H., Cntr." <BALANESITH@afmcfafb.fafb.af.mil>
To: 'AVT Working Group' <rem-conf@es.net>
Subject: FW: IP Multicasting
Date: Tue, 25 Mar 97 12:43:00 MST
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 ----------
From: Steve Coya
To: Balanesi, Terry H., Cntr.
Subject: Re: IP Multicasting
Date: Monday, March 24, 1997 4:23PM


Terry,

Sorry for the previous message - wrong envelope!

>> I am hoping you can direct me to someone who understands IP multicating
>> and the relationship to this type of route addressing and TCP.  I would
>> appreciate any help you can give me.


You might start off by sending a message to the AVT Working Group. They may 
be
able to vector you to the right resource.

The email address is rem-conf@es.net

From rem-conf-request@es.net Tue Mar 25 15:01:02 1997 
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From: "Bernard D. Aboba" <aboba@internaut.com>
To: Stephen Casner <casner@precept.com>
Cc: "rem-conf@es.net" <rem-conf@es.net>
Subject: Re: Specifying RTCP BW (was: AVT meetings at April IETF)
In-Reply-To: <Pine.WNT.3.95.970325112653.-134487G-100000@oak.precept.com>
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Including the percentages is ok, but isn't it desirable to put some
upper limit on this? Is there a compelling reason for why RTCP 
bandwidth might go above 30 percent of the total?

From rem-conf-request@es.net Tue Mar 25 15:10:30 1997 
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unsubscribe

From rem-conf-request@es.net Tue Mar 25 15:16:34 1997 
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To: rodgers@nlm.nih.gov (R. P. C. Rodgers), rem-conf@es.net, mbone@ISI.EDU
From: Petri Helenius <pete@sms.fi>
Subject: Re: 7-11 April is going to be tight on the MBONE (Was: Broadcasting of 
         Infocom'97)
Cc: infocom411@ckp.or.jp, oka@kobe-u.ac.jp, dem@hep.net, likavec@igd.fhg.de

At 14:43 24.3.1997 -0500, R. P. C. Rodgers wrote:
>   If we have many more weeks like 7-11 April, are we going to have to
>   get more serious about enforcing some rules of priority, based on
>   advanced reservations, and perhaps some discussion of relative
>   priorities?  I'd hate to get into this business, but will it be
>   necessary?

No, we just need to push the available bandwidth higher since in my opinion
MBone is much more effective use of it than most of the unicast replication
based apps. I also value the conferences highly. I would like the
transmitters to monitor the incoming RTCP packets and spot bad links while
the transmission is active. It might also be a good idea to try something at
these levels of bandwidth a week or two advance to the actual sessions to
see how the network would behave. 
>
Pete


From rem-conf-request@es.net Tue Mar 25 15:57:14 1997 
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From: Karl Auerbach <karl@precept.com>
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To: Claude Castelluccia <Claude.Castelluccia@imag.fr>
cc: rem-conf@es.net
Subject: Re: Question about IPMulticast!
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> When a host joins a multicast group, it takes some time
> before this host starts receiving data from this group?
> How long does it really take? Are they studies
>  that evaluate or measure this time?

I've been doing some very unscientific measurements of this...

On an extremely well endowed network (underloaded ATM underpinnings, mongo
IP routers, not much IP traffic, few (<4) IP hops end-to-end, the whole
thing in one big room, etc) I've been observing join latencies (the time
between when I send an IGMP Group Membership packet from a host until the
time the data on the multicast group arrives) of from nearly nil to
perhaps 10 seconds.  (I'm guessing about the 10 seconds because it was
more of a "where is that traffic?" kind of thing -- like sitting at a
traffic light -- in which time streeeeeeetches.)

I'm not sure why it sometimes took so long.  Since this network was using
ATM Forum LAN Emulation (only on the router-to-host paths, not on the
inter-router paths), I have this suspicion that some of the time may have
been consumed by IGMP sniffing edge devices.  But I have no specific
evidence to support that. 

		--karl--




From rem-conf-request@es.net Tue Mar 25 16:12:41 1997 
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From: Orion Hodson <O.Hodson@cs.ucl.ac.uk>
Reply-To: rat-trap@cs.ucl.ac.uk
To: rem-conf@es.net
cc: ucl-audio@cs.ucl.ac.uk, merci@cs.ucl.ac.uk, mice-nsc-uk@cs.ucl.ac.uk
Subject: New RAT release (v3.0.18)
Date: Tue, 25 Mar 1997 20:51:34 +0000
Message-ID: <1554.859323094@cs.ucl.ac.uk>
Sender: O.Hodson@cs.ucl.ac.uk

The Robust-Audio Tool (RAT) is designed to allow multiple users to
talk to each other over the internet.  RAT is adaptive to network and
host conditions.  It is RTP compliant and can be operated in unicast
and multicast modes.  It is compatible with other Mbone audio tools
(VAT, FreePhone, IP/TV) and offers redundant audio encoding and
receiver based repair to provide  improved sound quality.  Further
details can be found at http://www-mice.cs.ucl.ac.uk/mice/rat/. 

Binaries (and man page) are available at the UCL ftp server:
	ftp://cs.ucl.ac.uk/mice/rat/
and currently exist for Solaris, SunOS, Irix, Linux, HP-UX and Windows95.

FreeBSD, and possibly OSF/1, versions will be available
shortly.  Users of earlier releases should upgrade now to enjoy
improved audio quality and stability.

This release has the following enhancements, compared to release 2.6:
	- Improved redundancy support
	- Improved packet repetition
	- Transcoder/mixer operation 
	- Flakeway operation 
	- Lecture mode
	- LBL conference bus compatibility
	- Audio device trading between multiple instances of rat/vat
	- Voice switching of video when combined with vic
	- Better RTCP compliance
	- L16 (16bit linear, uncompressed) codec
	- Improved performance of analysis-by-synthesis codecs (GSM/LPC)
	- New user-interface
	- Active speaker indication now works
	- Automatic audio DC bias removal
	- Selective muting of participants
	- Assorted bug-fixes

A large amount of the code has been re-written to make RAT more
flexible and reliable.  The major enhancement in this release is the
ability to operate as a transcoder.  RAT will transcode between two
multicast groups, unicast and multicast, and unicast-to-unicast.  RAT
can transcode between any combination of PCMU, L16, DVI, GSM, LPC and
redundant audio, and can add/remove encryption as appropriate.

Comments, suggestions, and bug-reports should be sent to
rat-trap@cs.ucl.ac.uk.

regards

Colin, Orion, and Isidor [despite being on a plane to Greece :-) ] 





From rem-conf-request@es.net Tue Mar 25 17:16:15 1997 
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To: Steve Deering <deering@cisco.com>
Cc: rem-conf@es.net
Subject: Re: Question about IPMulticast!
In-Reply-To: Your message of "Tue, 25 Mar 1997 09:33:13 PST." <v03020915af5db997ba8c@[171.69.199.124]>
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Date: Tue, 25 Mar 97 15:10:07 PST
From: Daniel Zappala <daniel@ISI.EDU>


deering@cisco.com said:

> So, yes, both of the options that you suggested for reducing join latency
> have always been part of the multicast routing machinery.  Thus my original
> answer to you: the join latency is at worst the RTT from receiver to source
> (ignoring the hop-by-hop processing delays, which are negligible on paths
> of any significant distance).
> 

Assuming, of course, none of the routing protocol's packets are lost...

Daniel Zappala
daniel@isi.edu




From rem-conf-request@es.net Tue Mar 25 17:44:56 1997 
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References: Your message of "Tue,
            25 Mar 1997 09:33:13 PST." <v03020915af5db997ba8c@[171.69.199.124]>
Mime-Version: 1.0
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Date: Tue, 25 Mar 1997 14:25:31 -0800
To: Daniel Zappala <daniel@ISI.EDU>
From: Steve Deering <deering@cisco.com>
Subject: Re: Question about IPMulticast!
Cc: rem-conf@es.net

At 3:10 PM -0800 3/25/97, Daniel Zappala wrote:
> deering@cisco.com said:
>
> > So, yes, both of the options that you suggested for reducing join latency
> > have always been part of the multicast routing machinery.  Thus my original
> > answer to you: the join latency is at worst the RTT from receiver to source
> > (ignoring the hop-by-hop processing delays, which are negligible on paths
> > of any significant distance).
> >
>
> Assuming, of course, none of the routing protocol's packets are lost...

and that the routing doesn't change during the join, and that the network
doesn't partition, and that there are no Byzantine failures in the routers,
and that ...

What a bunch of picky people we have on this list!  Yes, in truth, the
worst-case join latency is infinite.  Is that a more useful answer?

Steve



From rem-conf-request@es.net Tue Mar 25 20:29:19 1997 
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From: Korey Kirschenmann <kkirsche@cogs.gps.com>
To: rem-conf <rem-conf@es.net>
Subject: MBONE training sites?
Date: Tue, 25 Mar 1997 18:14:31 -0600
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MBONE list,

Do any of you know of training centers that have been setup with MBONE
access? I am looking for training sites for rent, so I can deliver training
to multiple sites at one time utilizing IP multicast and the MBONE
technology. 

Korey Kirschenmann
kkirsche@cogs.gps.com
701-281-3183

From rem-conf-request@es.net Tue Mar 25 20:32:05 1997 
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Date: Tue, 25 Mar 1997 20:28:44 -0500
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From: Ethan Rafferty <eer2@cornell.edu>
Subject: NFS server research project
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Hello,

I am an MBA student at Cornell's Johnson School of Management conducting
research on NFS servers.  My research team would like to send out a
conjoint analysis tool (which would take 15 minutes to complete) to
potential and actual consumers of NFS servers.  The purpose of the study is
to analyze consumers' tradeoffs between attributes of the product (such as
performance, storage capacity, availability, support of multiple client
types, and method of purchase) in order to determine optimal configurations.

I would like to discuss with you your suggestions on contacting appropriate
individuals, such as systems administrators, to participate in this study.
To be more specific, I am seeking lists of contact information for such
individuals.

Please let me know your suggestions regarding directions in which to proceed.

Thanks,

Ethan Rafferty
(607) 256-1201
eer2@cornell.edu


From rem-conf-request@es.net Tue Mar 25 21:20:08 1997 
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From: Sneha Kumar Kasera <kasera@cs.umass.edu>
Message-Id: <199703260218.VAA15535@giane.cs.umass.edu>
Subject: Re: Question about IPMulticast!
To: rem-conf@es.net
Date: Tue, 25 Mar 1997 21:18:01 -0500 (EST)
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> So, yes, both of the options that you suggested for reducing join latency
> have always been part of the multicast routing machinery.  Thus my original
> answer to you: the join latency is at worst the RTT from receiver to source
> (ignoring the hop-by-hop processing delays, which are negligible on paths
> of any significant distance).
>
> Steve
>
I think that join latency could be more than RTT depending upon the processing
load at an intermediate hop unless join messages are treated with high
priority. As it needs to be received at each  hop, a join message has to go
through additional "CPU" queueing (different from the fast H/W queues) at each
hop. One could ignore the processing delay which is of the order of a few
milliseconds but not ignore the additional queueing delays.
The difference between "join latency" and RTT is likely to increase with
increasing number of hops.

I had once conducted RTT experiments where packets were sent with IP options
and without IP options. The packets with IP options usually had a much
higher RTT(with more fluctuation) in comparison to packets without
IP options.

I understand that the processing of join messages is different from the
processing of packets with IP options. Don't you think we could still get an
estimate of the delay due to the "additional" queueing that takes place at
the hops ?

Sneha


From rem-conf-request@es.net Tue Mar 25 21:44:09 1997 
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Date: Tue, 25 Mar 1997 18:38:03 -0800
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Subject: un
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Can someone tell me how to unsubscribe from this list?

From rem-conf-request@es.net Wed Mar 26 04:57:06 1997 
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          Wed, 26 Mar 1997 13:53:33 +0100 (MET)
From: Toerless Eckert <Toerless.Eckert@Informatik.Uni-Erlangen.de>
Message-Id: <199703261253.NAA16720@faui40.informatik.uni-erlangen.de>
Subject: Wanted: progressive scan video cameras
To: rem-conf@es.net
Date: Wed, 26 Mar 1997 13:53:31 +0100 (MET)
Organisation: CSD IMMD IV, University of Erlangen, Germany
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Hi

I was wondering If someone could point me to inexpensive progressive-scan
color video cameras. I would prefer a camera that delivers NTSC or PAL
compatible video signal on output so that it can be transferred into the
computer via standard frame-grabbers. I guess something like a SuperNTSC
camera would do this trick nicely but i've got no idea where to get those and
how expensive they are. I'm also much more interested in a PAL compatible
camera.

If you know progressive scan cameras that are not PAL/NTSC compatible i'd like
to learn which framegrabbers you use with them.

The background of the question is that we'd like to go for SCIF (CCIR601)
resolution with our research video compression scheme, but that we would
like to avoid handling of interlace motion artefact.

Best regards
	Toerless

From rem-conf-request@es.net Wed Mar 26 05:47:11 1997 
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 ----------
From: BALANESITH
To: Balanesi  Terry H.  Cntr.
Subject: Returned mail
Date: Tuesday, March 25, 1997 7:49PM

 ---- message ----

Error transferring to NOTESMAIL1/PICTEL mail.box; Server error:
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Error transferring to NOTESMAIL1/PICTEL mail.box; Server error:
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To: 'AVT Working Group' <rem-conf@es.net>
bcc: rem-conf <rem-conf@smtpnotes.pictel.com>
From: "Balanesi  Terry H.  Cntr." <BALANESITH@afmcfafb.fafb.af.mil>
Date: 25 Mar 97 12:43:00 EDT
Subject: FW: IP Multicasting
MIME-Version: 1.0
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 ----------
From: Steve Coya
To: Balanesi, Terry H., Cntr.
Subject: Re: IP Multicasting
Date: Monday, March 24, 1997 4:23PM


Terry,

Sorry for the previous message - wrong envelope!

>> I am hoping you can direct me to someone who understands IP multicating
>> and the relationship to this type of route addressing and TCP.  I would
>> appreciate any help you can give me.


You might start off by sending a message to the AVT Working Group. They may
be
able to vector you to the right resource.

The email address is rem-conf@es.net

 ---- message ------

From rem-conf-request@es.net Wed Mar 26 09:03:27 1997 
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Date: Tue, 25 Mar 1997 23:22:28 -0800 (PST)
From: Ahmed A-G Helmy <ahelmy@catarina.usc.edu>
To: Daniel Zappala <daniel@ISI.EDU>
cc: Steve Deering <deering@cisco.com>, rem-conf@es.net
Subject: Re: Question about IPMulticast!
In-Reply-To: <199703252208.AA01720@zephyr.isi.edu>
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> deering@cisco.com said:
> 
> > So, yes, both of the options that you suggested for reducing join latency
> > have always been part of the multicast routing machinery.  Thus my original
> > answer to you: the join latency is at worst the RTT from receiver to source
> > (ignoring the hop-by-hop processing delays, which are negligible on paths
> > of any significant distance).
> > 
> 
> Assuming, of course, none of the routing protocol's packets are lost...

Daniel,

I believe that grafts trigger graft acks hbh, and so are more robust to 
loss of pkts than pure soft state [note that the acks only ameliorate the 
loss/failure problems but are still susceptible to other node/state 
failures, and there soft state probably performs better by periodic 
refreshes]... 

There is a probabilistic analysis of the join latency for PIM-SM taking 
packet losses into consideration, as a function of the per link av 
probability of pkt loss and the domain/network size. This analysis was 
conducted as part of the bootstrap convergence time estimates found in:

http://catarina.usc.edu/pim/pimsm/bootstrap-TechReport.ps.gz

or

http://www.usc.edu/dept/cs/technical_reports.html
TR#: 97-644

Regards,
-A

> 
> Daniel Zappala
> daniel@isi.edu
> 
> 
> 
> 

From rem-conf-request@es.net Wed Mar 26 12:33:26 1997 
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To: karl@precept.com
cc: Claude Castelluccia <Claude.Castelluccia@imag.fr>, rem-conf@es.net
Subject: How to test join latency on the MBone (was Re: Question about 
         IPMulticast!)
In-reply-to: Your message of "Tue, 25 Mar 97 12:41:52 PST." <Pine.SOL.3.93.970325123351.10641B-100000@big-bear>
Date: Wed, 26 Mar 1997 12:04:51 PST
Sender: Bill Fenner <fenner@parc.xerox.com>
From: Bill Fenner <fenner@parc.xerox.com>
Message-Id: <97Mar26.120503pst.177486@crevenia.parc.xerox.com>

Here's how I would test join latency on the MBone:

1. Find a source and group that is transmitting data.  IMJ is a good
source if there's nothing else that's transmitting constantly, since
you can make it transmit on demand.
2. Mtrace to that source and group and find the closest router that's
on the tree.  e.g.

Mtrace from 130.207.8.30 to 13.2.116.11 via group 224.2.1.2
Querying full reverse path... 
  0  crevenia (13.2.116.11)
 -1  tibia-235 (13.2.116.239)  DVMRP  thresh^ 1  -9 ms  Prune sent upstream
 -2  disco-dart.parc.xerox.com (204.162.228.251)  DVMRP  thresh^ 0  0 ms  Prune sent upstream
 -3  dodgedart.parc.xerox.com (204.162.228.7)  DVMRP  thresh^ 1  -85 s   Prune sent upstream
 -4  ames.dart.net (140.173.144.1)  DVMRP  thresh^ 1  338 ms  Prune sent upstream
 -5  mbone.nsi.nasa.gov (192.203.230.241)  DVMRP  thresh^ 32  48 ms  Output pruned
 -6  sanjose1-mbone1.bbnplanet.net (4.0.0.34)  PIM/Special  thresh^ 0  356 ms
 -7  paloalto-mbone1.bbnplanet.net (131.119.0.197)  PIM/Special  thresh^ 32  411 ms
 -8  collegepk-mbone1.bbnplanet.net (128.167.252.196)  PIM/Special  thresh^ 64  399 ms
 -9  atlanta3-mbone1.bbnplanet.net (192.221.48.234)  PIM/Special  thresh^ 64  -98 s
-10  feta-fddi.gatech.edu (130.207.244.30)  DVMRP  thresh^ 32  -175 s
-11  siwenna.cc.gatech.edu (130.207.9.11)  DVMRP  thresh^ 8  400 ms
-12  ophelia-video.cc.gatech.edu (130.207.10.26)  DVMRP  thresh^ 1  405 ms
-13  muerto.cc.gatech.edu (130.207.8.30)
Round trip time 522 ms; total ttl of 69 required.

This means that mbone.nsi.nasa.gov is the closest router on the tree,
since it says "Output pruned" instead of "Prune sent upstream".  The
statistics section also shows this:

192.203.230.241 mbone.nsi.nasa.gov Output pruned
     |     ^      ttl   69       120 pps      358/358  =100%  25 pps
     v     |      hop -301 ms
192.52.195.128 
140.173.144.1   ames.dart.net      Prune sent upstream

This 100% loss is caused by the prune.

3. Run tcpdump on the group in question
4. Join the group and watch the time between the IGMP join and the first
packet.

If the join latency is longer than seconds, you can use mtrace as above
to find the hop that failed to properly graft.

  Bill

From rem-conf-request@es.net Thu Mar 27 07:02:23 1997 
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Subject: I-D ACTION:draft-ietf-avt-rtp-mib-00.txt
Date: Thu, 27 Mar 1997 09:54:15 -0500
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--NextPart

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

       Title     : Real-Time Transport Protocol Management Information Base
       Author(s) : M. Baugher, J. Du, S. Naudus
       Filename  : draft-ietf-avt-rtp-mib-00.txt
       Pages     : 25
       Date      : 03/26/1997

This memo defines an experimental  Management Information Base (MIB) for 
use with network management protocols in TCP/IP-based internets.  In 
particular, it defines objects for managing Real-Time Transport Protocol 
Systems [1].  Comments should be made to the IETF Audio/Video Transport 
Working Group.                               
                              
This memo does not specify a standard for the Internet community.          

Internet-Drafts are available by anonymous FTP.  Login with the username
"anonymous" and a password of your e-mail address.  After logging in,
type "cd internet-drafts" and then
     "get draft-ietf-avt-rtp-mib-00.txt".
A URL for the Internet-Draft is:
ftp://ds.internic.net/internet-drafts/draft-ietf-avt-rtp-mib-00.txt
 
Internet-Drafts directories are located at:	
	                                                
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                 ftp.nis.garr.it                 
	                                                
     o  Pacific Rim: munnari.oz.au               
	                                                
     o  US East Coast: ds.internic.net           
	                                                
     o  US West Coast: ftp.isi.edu               
	                                                
Internet-Drafts are also available by mail.	
	                                                
Send a message to:  mailserv@ds.internic.net. In the body type: 
     "FILE /internet-drafts/draft-ietf-avt-rtp-mib-00.txt".
							
NOTE: The mail server at ds.internic.net can return the document in
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      feature, insert the command "ENCODING mime" before the "FILE"
      command.  To decode the response(s), you will need "munpack" or
      a MIME-compliant mail reader.  Different MIME-compliant mail readers
      exhibit different behavior, especially when dealing with
      "multipart" MIME messages (i.e., documents which have been split
      up into multiple messages), so check your local documentation on
      how to manipulate these messages.
							
							

Below is the data which will enable a MIME compliant mail reader 
implementation to automatically retrieve the ASCII version
of the Internet-Draft.

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

--OtherAccess
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        access-type="mail-server";
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Content-Type: text/plain
Content-ID: <19970326103143.I-D@ietf.org>

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

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

Content-Type: text/plain
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--OtherAccess--

--NextPart--


From rem-conf-request@es.net Thu Mar 27 09:08:42 1997 
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Date: Thu, 27 Mar 1997 12:03:50 -0500
From: "H. DO-DUY" <106025.3357@compuserve.com>
Subject: Building Vat under Windows NT4.0
To: Vat mailing list <rem-conf@es.net>
Message-ID: <199703271203_MC2-135E-F182@compuserve.com>

Hi, 

I'm trying to build Vat 4.0b2 from sources found on the Net, on my Pentium
200 Pro, using NT 4.0 and Visual C++ 4.2.
I've got a problem of variable definition that i can't find anywhere. they
are declared as extern in two modules, put are never defined, so the link
can't be done. Those variables seem to be used to convert mulaw format to
linear format : 
        mulawtolin
and     lintomulaw
and     lintomulawX


Please, tell me, if you can, where i could find the definition of those
variables.

Thanks a lot


Rodolphe PETIT
SEEE, Toulouse 28th, march, 97
e-mail : 106025.3357@compuserve.com

From rem-conf-request@es.net Thu Mar 27 11:53:36 1997 
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Date: Thu, 27 Mar 1997 11:32:52 -0800
To: rem-conf@es.net
From: Mark Baugher <mbaugher@mailbox.jf.intel.com>
Subject: I-D ACTION:draft-ietf-avt-rtp-mib-00.txt
Mime-Version: 1.0
Content-Type: text/plain; charset="us-ascii"

hi,
  this I-D is a result of some ongoing work we are doing
on management of RTP on our network.  The MIB definition
can be viewed in hypertext format at www.rdrop.com\rtp.mi2.

Mark
>To: IETF-Announce:;
>cc: rem-conf@es.net
>From: Internet-Drafts@ietf.org
>Reply-to: Internet-Drafts@ietf.org
>Subject: I-D ACTION:draft-ietf-avt-rtp-mib-00.txt
>Date: Thu, 27 Mar 1997 09:54:15 -0500
>Sender: cclark@ietf.org
>Content-Length: 3484
>
> A New Internet-Draft is available from the on-line Internet-Drafts 
> directories. This draft is a work item of the Audio/Video Transport 
> Working Group of the IETF.                                                
>
>       Title     : Real-Time Transport Protocol Management Information Base
>       Author(s) : M. Baugher, J. Du, S. Naudus
>       Filename  : draft-ietf-avt-rtp-mib-00.txt
>       Pages     : 25
>       Date      : 03/26/1997
>
>This memo defines an experimental  Management Information Base (MIB) for 
>use with network management protocols in TCP/IP-based internets.  In 
>particular, it defines objects for managing Real-Time Transport Protocol 
>Systems [1].  Comments should be made to the IETF Audio/Video Transport 
>Working Group.                               
>                              
>This memo does not specify a standard for the Internet community.          
>
>Internet-Drafts are available by anonymous FTP.  Login with the username
>"anonymous" and a password of your e-mail address.  After logging in,
>type "cd internet-drafts" and then
>     "get draft-ietf-avt-rtp-mib-00.txt".
>A URL for the Internet-Draft is:
>ftp://ds.internic.net/internet-drafts/draft-ietf-avt-rtp-mib-00.txt
> 
>Internet-Drafts directories are located at:	
>	                                                
>     o  Africa:  ftp.is.co.za                    
>	                                                
>     o  Europe:  ftp.nordu.net            	
>                 ftp.nis.garr.it                 
>	                                                
>     o  Pacific Rim: munnari.oz.au               
>	                                                
>     o  US East Coast: ds.internic.net           
>	                                                
>     o  US West Coast: ftp.isi.edu               
>	                                                
>Internet-Drafts are also available by mail.	
>	                                                
>Send a message to:  mailserv@ds.internic.net. In the body type: 
>     "FILE /internet-drafts/draft-ietf-avt-rtp-mib-00.txt".
>							
>NOTE: The mail server at ds.internic.net can return the document in
>      MIME-encoded form by using the "mpack" utility.  To use this
>      feature, insert the command "ENCODING mime" before the "FILE"
>      command.  To decode the response(s), you will need "munpack" or
>      a MIME-compliant mail reader.  Different MIME-compliant mail readers
>      exhibit different behavior, especially when dealing with
>      "multipart" MIME messages (i.e., documents which have been split
>      up into multiple messages), so check your local documentation on
>      how to manipulate these messages.
>							
>							
>
>Below is the data which will enable a MIME compliant mail reader 
>implementation to automatically retrieve the ASCII version
>of the Internet-Draft.
>
><ftp://ds.internic.net/internet-drafts/draft-ietf-avt-rtp-mib-00.txt>
>

From rem-conf-request@es.net Thu Mar 27 17:40:36 1997 
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Date: Thu, 27 Mar 1997 17:32:50 -0800
To: rem-conf@es.net
From: Alagu Periyannan <alagu@mailman.apple.com>
Subject: RTP over TCP


Hi all,

Sometime back there was talk about RTP over TCP and
maybe even an internet draft.

Can anyone provide a pointer to this work?

Thanks.




Alagu Periyannan                   alagu@mailman.apple.com

Communications Products and Technology
Apple Computer, Inc.



From rem-conf-request@es.net Fri Mar 28 08:08:03 1997 
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Date: Fri, 28 Mar 1997 11:04:26 -0500 (EST)
From: kevin@cc.gatech.edu (Kevin C. Almeroth)
Message-Id: <199703281604.LAA17213@flora.cc.gatech.edu>
To: rem-conf@es.net
Subject: Las Vegas Interop Broadcasts


   This is to announce our intentions to broadcast parts of the Las Vegas
Networld+Interop Show.  The broadcast dates are during the day (LV time)
>from Monday, May 5th through Thursday, May 8th.

   I'm not going to say exactly what we are planning to broadcast but
it will be content from parts of the General Conference, Intranet Conference,
Engineers Conference, and the keynotes.

   Actually, if you have any interest in making suggestions, you can go
to the N+I LV WWW page, take a look at what'll be playing, and send me
any preferences or suggestions.  The URL is:

   http://www.interop.com/Interop_Online/htdocs/events/index.html

-Kevin Almeroth

From rem-conf-request@es.net Fri Mar 28 10:49:46 1997 
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From: Internet-Drafts@ietf.org
Reply-to: Internet-Drafts@ietf.org
Subject: I-D ACTION:draft-ietf-avt-profile-new-00.txt
Date: Fri, 28 Mar 1997 10:13:52 -0500
Sender: cclark@ietf.org
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--NextPart

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

       Title     : RTP Profile for Audio and Video Conferences 
                   with Minimal Control                                                 
       Author(s) : H. Schulzrinne
       Filename  : draft-ietf-avt-profile-new-00.txt
       Pages     : 26
       Date      : 03/27/1997

This memo describes a profile called "RTP/AVP" for the use of the real-time
transport protocol (RTP), version 2, and the associated control protocol, 
RTCP, within audio and video multiparticipant conferences with minimal 
control. It provides interpretations of generic fields within the RTP 
specification suitable for audio and video conferences. In particular, this
document defines a set of default mappings from payload type numbers to 
encodings.                                   
                           
The document also describes how audio and video data may be carried 
within RTP. It defines a set of standard encodings and their names when 
used within RTP. However, the encoding definitions are independent of 
the particular transport mechanism used. The descriptions provide pointers 
to reference implementations and the detailed standards. This document 
is meant as an aid for implementors of audio, video and other real-time 
multimedia applications.                                                              

Internet-Drafts are available by anonymous FTP.  Login with the username
"anonymous" and a password of your e-mail address.  After logging in,
type "cd internet-drafts" and then
     "get draft-ietf-avt-profile-new-00.txt".
A URL for the Internet-Draft is:
ftp://ds.internic.net/internet-drafts/draft-ietf-avt-profile-new-00.txt
 
Internet-Drafts directories are located at:	
	                                                
     o  Africa:  ftp.is.co.za                    
	                                                
     o  Europe:  ftp.nordu.net            	
                 ftp.nis.garr.it                 
	                                                
     o  Pacific Rim: munnari.oz.au               
	                                                
     o  US East Coast: ds.internic.net           
	                                                
     o  US West Coast: ftp.isi.edu               
	                                                
Internet-Drafts are also available by mail.	
	                                                
Send a message to:  mailserv@ds.internic.net. In the body type: 
     "FILE /internet-drafts/draft-ietf-avt-profile-new-00.txt".
							
NOTE: The mail server at ds.internic.net can return the document in
      MIME-encoded form by using the "mpack" utility.  To use this
      feature, insert the command "ENCODING mime" before the "FILE"
      command.  To decode the response(s), you will need "munpack" or
      a MIME-compliant mail reader.  Different MIME-compliant mail readers
      exhibit different behavior, especially when dealing with
      "multipart" MIME messages (i.e., documents which have been split
      up into multiple messages), so check your local documentation on
      how to manipulate these messages.
							
							

Below is the data which will enable a MIME compliant mail reader 
implementation to automatically retrieve the ASCII version
of the Internet-Draft.

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

--OtherAccess
Content-Type:  Message/External-body;
        access-type="mail-server";
        server="mailserv@ds.internic.net"

Content-Type: text/plain
Content-ID: <19970327150436.I-D@ietf.org>

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

--OtherAccess
Content-Type:   Message/External-body;
        name="draft-ietf-avt-profile-new-00.txt";
        site="ds.internic.net";
        access-type="anon-ftp";
        directory="internet-drafts"

Content-Type: text/plain
Content-ID: <19970327150436.I-D@ietf.org>

--OtherAccess--

--NextPart--

From rem-conf-request@es.net Fri Mar 28 14:31:43 1997 
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Cc: algirdas.pakstas@hia.no
Reply-to: a.pakstas@ieee.org
Subject: Workshop On Quality and Productivity for Communications
Mime-Version: 1.0
Content-Type: text/plain; charset=us-ascii
Date: Fri, 28 Mar 1997 23:16:11 +0100
From: Algirdas Pakstas <Algirdas.Pakstas@hia.no>
Message-ID: <"torsk.hia..039:28.02.97.22.24.19"@hia.no>

Sorry if you receive mutual copies of this Call for Participation...

------- Forwarded Message

The Communications Software committee is co-sponsoring the following
workshop. So if you are not interested in attending, then please forward
this notice to others you think would be interested. More information can
be found on the WWW at
http://gump.bellcore.com:8000/ieee/commsoft/workshop.html


Thanks,

Stan Moyer


<center>International Workshop On

Quality and Productivity for Communications in the

</center>                               next Millennium



<center>May 13-15, 1997

Ojai Valley Inn

Ojai, California, USA

</center>

<center>Co-sponsored by the IEEE Communications Society Communications
Software and Quality Assurance and Management Technical Committees and
the IEICE

Communications Society Communications Quality Committee

</center>

DESCRIPTION


The current telecommunications environment is one of

change and corporate downsizing. Telecommunications

deregulation is creating more telecom firms. Businesses

are having to perform the same (or more) functions with

fewer people. The time to market for new services and

products is decreasing (e.g. the Internet world). At the

same time, even more stringent quality standards must be

maintained in this era of competition. 


The goal of this workshop is to bring together end users,

service providers and developers from the

communications industries to share the latest products,

services, processes and methodologies for improving

productivity while enhancing quality. Throughout each

session, one or more of the following issues should be

addressed: 


     How can service providers do more with less? 

     How to rapidly bring new services to market? 

     How to maintain (or improve) quality in rapid

     development environments. 

     How to maintain quality while outsourcing work. 



------- End of Forwarded Message






From rem-conf-request@es.net Fri Mar 28 21:48:28 1997 
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          id <06637-0@osi-east2.es.net>; Fri, 28 Mar 1997 21:43:50 -0800
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          Fri, 28 Mar 1997 21:43:37 -0800
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          id OAA23509; Sat, 29 Mar 1997 14:47:31 +0900
Message-Id: <199703290547.OAA23509@nanotsu.kobe-u.ac.jp>
To: rem-conf@es.net
cc: Mark Handley <mjh@isi.edu>, rodgers@nlm.nih.gov (R. P. C. Rodgers), 
    dem@hep.net, likavec@igd.fhg.de, shimojo@center.osaka-u.ac.jp
Reply-To: oka@nanotsu.kobe-u.ac.jp
Subject: Broadcasting of Infocom'97
In-reply-to: Your message of Mon, 24 Mar 1997 17:45:37 EST. <4883.859243537@buttle.lcs.mit.edu>
Mime-Version: 1.0
Content-Type: text/plain;charset="ISO-2022-JP"
Date: Sat, 29 Mar 1997 14:47:30 +0900
From: Koji OKAMURA <oka@nanotsu.kobe-u.ac.jp>


We, Cyber Kansai Project, has decided to change the broadcasting
schedule of Infocom'97.

At the week of 7-11 April, we will broadcast Infocom on only daytime of
Japan and will re-broadcast on US daytime at the next or later week.

These are our broadcasting plan.

			Channel 1		Channel 2
 === 4/7 ===
 8:30-17:00(GMT+9)	Gigabit Workshop

 === 4/9 ===
 8:00-10:00(GMT+9)	Keynotes
10:30-12:00(GMT+9)	ATM Switching I		Wireless Networks I
13:30-15:00(GMT+9)	ATM Traffic Control	Optical Networks I
15:30-17:00(GMT+9)	ATM ABR Services I	Panel I:  
						Broadband Service to the Home

 === 4/10 ===
 8:30-10:00(GMT+9)	ATM ABR Services II	Optical Networks II
10:30-12:00(GMT+9) 	ATM Switching II	Wireless Networks II
12:00-13:30(GMT+9)  				Panel II: 
						Routing vs Switching
13:30-15:00(GMT+9)	ATM Traffic Cntrl II	Optical Network III
15:30-17:00(GMT+9)	ATM ABR Services III	Wireless Networks III

 === 4/11 ===
 8:30-10:00(GMT+9)	ATM Switching III	Wireless Networks IV
10:30-12:00(GMT+9)	ATM Traffic Cntrl III	Panel III:
						Micro cellular Communication 
						Systems: PHS, DECT and PACS
13:30-15:00(GMT+9)	LANs and MANs		Optical Networks IV
15:30-17:00(GMT+9)	ATM ABR Services IV	Wireless Networks V

Best Regards.

--
Koji

From rem-conf-request@es.net Sun Mar 30 06:12:23 1997 
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          for <rem-conf@es.net>; Sun, 30 Mar 1997 17:06:28 +0200
From: Scott_Petrack@vocaltec.com
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To: rem-conf@es.net
Message-ID: <4225646A.004D394C.00@maskit1.vocaltec.co.il>
Date: Sun, 30 Mar 1997 16:07:36 +0200
Subject: Anyone not using a hotel reservation or have a spare room on April 7?
Mime-Version: 1.0
Content-type: text/plain; charset=US-ASCII






Scott Petrack
03/30/97 04:07 PM

I've really screwed up now and didn't properly book a room for the IETF.
I have been able to find a room for all days EXCEPT April 7.

If anyone has reserved a room that s/he no longer needs, or knows of a
spare room for that
day, I would be most grateful to hear about it.

I of course would prefer a reservation for the full week, rather than
switching hotels in the middle for one day,
but beggars can't be choosers.

Sincerely,

Scott Petrack



From rem-conf-request@es.net Mon Mar 31 06:50:11 1997 
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Content-Type: Multipart/Mixed; Boundary="NextPart"
To: IETF-Announce:;
cc: rem-conf@es.net
From: Internet-Drafts@ietf.org
Reply-to: Internet-Drafts@ietf.org
Subject: I-D ACTION:draft-ietf-avt-crtp-02.txt
Date: Mon, 31 Mar 1997 09:35:26 -0500
Sender: cclark@ietf.org
Message-ID: <9703310935.aa04843@ietf.org>

--NextPart

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

       Title     : Compressing IP/UDP/RTP Headers for Low-Speed Serial 
                   Links                                                   
       Author(s) : S. Casner, V. Jacobson
       Filename  : draft-ietf-avt-crtp-02.txt
       Pages     : 20
       Date      : 03/27/1997

This document describes a method for compressing the headers of IP/UDP/RTP 
datagrams to reduce overhead on low-speed serial links. In many cases, all 
three headers can be compressed to 2-4 bytes.        
                      
Comments are solicited and should be addressed to the working group 
mailing list rem-conf@es.net and/or the author(s).                                 

Internet-Drafts are available by anonymous FTP.  Login with the username
"anonymous" and a password of your e-mail address.  After logging in,
type "cd internet-drafts" and then
     "get draft-ietf-avt-crtp-02.txt".
A URL for the Internet-Draft is:
ftp://ds.internic.net/internet-drafts/draft-ietf-avt-crtp-02.txt
 
Internet-Drafts directories are located at:	
	                                                
     o  Africa:  ftp.is.co.za                    
	                                                
     o  Europe:  ftp.nordu.net            	
                 ftp.nis.garr.it                 
	                                                
     o  Pacific Rim: munnari.oz.au               
	                                                
     o  US East Coast: ds.internic.net           
	                                                
     o  US West Coast: ftp.isi.edu               
	                                                
Internet-Drafts are also available by mail.	
	                                                
Send a message to:  mailserv@ds.internic.net. In the body type: 
     "FILE /internet-drafts/draft-ietf-avt-crtp-02.txt".
							
NOTE: The mail server at ds.internic.net can return the document in
      MIME-encoded form by using the "mpack" utility.  To use this
      feature, insert the command "ENCODING mime" before the "FILE"
      command.  To decode the response(s), you will need "munpack" or
      a MIME-compliant mail reader.  Different MIME-compliant mail readers
      exhibit different behavior, especially when dealing with
      "multipart" MIME messages (i.e., documents which have been split
      up into multiple messages), so check your local documentation on
      how to manipulate these messages.
							
							

Below is the data which will enable a MIME compliant mail reader 
implementation to automatically retrieve the ASCII version
of the Internet-Draft.

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

--OtherAccess
Content-Type:  Message/External-body;
        access-type="mail-server";
        server="mailserv@ds.internic.net"

Content-Type: text/plain
Content-ID: <19970328112342.I-D@ietf.org>

ENCODING mime
FILE /internet-drafts/draft-ietf-avt-crtp-02.txt

--OtherAccess
Content-Type:   Message/External-body;
        name="draft-ietf-avt-crtp-02.txt";
        site="ds.internic.net";
        access-type="anon-ftp";
        directory="internet-drafts"

Content-Type: text/plain
Content-ID: <19970328112342.I-D@ietf.org>

--OtherAccess--

--NextPart--


From rem-conf-request@es.net Mon Mar 31 12:07:35 1997 
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          id <09847-0@osi-east2.es.net>; Mon, 31 Mar 1997 11:53:34 -0800
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Date: Mon, 31 Mar 1997 12:52:22 -0700 (MST)
From: Evi Nemeth <evi@rupertsberg.cs.colorado.edu>
Message-Id: <199703311952.MAA01465@rupertsberg.cs.colorado.edu>
To: rem-conf@es.net
Subject: IETF Broadcast, April 7-10
Cc: evi@rupertsberg.cs.colorado.edu

The IETF meeting held April 7-11 in Memphis, Tennessee, USA will
broadcast 2 channels of audio/video/wb of the sessions Monday-Thurs.
There will be no broadcasts on Friday.  The traditional delayed 
broadcasts will be done at a later time, during a less busy week
for the MBone.

We suggest that the other 3 major events during this same week also
delay any rebroadcasts so that we can accomodate the bandwidth needs
for each of the live events.  Maybe we can get the various "radio"
stations to shutdown during that week too.

We will request that all presenters provide us with postscript that
will fit in the whiteboard (<32kbytes/slide) and will stop transmitting
video when it is just the speaker roaming around and talking.  Others
with whiteboard materials might do the same.  We usually show the speaker
for a minute or two and then stop transmitting, which keeps his/her
picture in vic, but does not use further bandwidth.

Thanks to everyone for cooperating to make this busy week work.

-evi

Evi Nemeth
IETF Multicast Staff

