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From rem-conf Thu Jan 01 17:29:40 1998 
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Date: Thu, 01 Jan 1998 17:18:26 -0800
From: Cameron Elliott <cam@elliott.net>
Organization: KEGO - Software & Datacommunications
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-- 
I'm curious is there a RTP payload format specification
for G.723.1 inside RTP?

Or does G.723.1 not really need a payload format, and just
follows the RTP header?
(If so, do you just leave the marker bit clear?)

Thanks
Cameron Elliott






--
Cameron Elliott

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

888-FOR-KEGO
Cam@Kego.com



From rem-conf Fri Jan 02 09:01:25 1998 
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Date: Fri, 02 Jan 1998 11:39:51 -0500
From: "Jonathan D. Rosenberg" <jdrosen@dnrc.bell-labs.com>
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The next version of RFC 1890
(http://www.cs.columbia.edu/~hgs/rtp/drafts/draft-ietf-avt-profile-new-02.txt)
specifies how to encapsulate g.723.1. You are correct in that there are
no additional header fields required for it.

-Jonathan R.

Cameron Elliott wrote:
> 
> --
> I'm curious is there a RTP payload format specification
> for G.723.1 inside RTP?
> 
> Or does G.723.1 not really need a payload format, and just
> follows the RTP header?
> (If so, do you just leave the marker bit clear?)
> 
> Thanks
> Cameron Elliott
> 
> --
> Cameron Elliott
> 
> Kego - Software & Datacommuncations
>   Unix, Embedded Software, TCP/IP, H.323
> 
> 888-FOR-KEGO
> Cam@Kego.com

-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
PHONE: (908) 949-6418                       Rm. 4C-526
FAX:   (908) 834-5379
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen



From rem-conf Mon Jan 05 06:06:25 1998 
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From: goebel@unik.no (Vera Goebel)
Subject: CfP - IDMS'98 (reminder)
Cc: idms98@unik.no
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Please accept our apologies if you receive multiple copies of
this announcement.

You will be doing us a great favor if you disseminate the
this Call for papers among your interested colleagues. Thanks.


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

                      REMINDER:

              Call for Papers IDMS'98
 5th International Workshop on Interactive Distributed
   Multimedia Systems and Telecommunication Services
         8. - 11. September 1998, Oslo, Norway


The Fifth International Workshop on Interactive Distributed Multimedia
Systems and Telecommunication Services follows the successful IDMS
workshops held 1997 in Darmstadt and 1996 in Berlin. The purpose of this

workshop is to bring together researchers, developers, and practitioners

>from academia and industry. The workshop serves as a forum for
discussion, presentation, and exploration of technologies and their
advances in the broad field of interactive distributed multimedia
systems and telecommunication services -- ranging from basic system
technologies
such as networking and operating system support to all kinds of
teleservices and distributed multimedia applications. Case studies and
papers describing experimental work are especially welcome. Relevant
topics include, but are not limited to

=B7 High-speed/ATM networks
=B7 Mobile multimedia systems
=B7 Multimedia over sattelite
=B7 Multimedia middelware
=B7 Quality of service issues
=B7 Media scaling
=B7 Resource management
=B7 Protocol design and implementation
=B7 Distributed multimedia database systems
=B7 Development tools for distributed multimedia applications
=B7 Multimedia-specific intelligent agents
=B7 Computer supported collaborative work
=B7 Distributed virtual reality systems
=B7 Distance education
=B7 Conferencing
=B7 Digital libraries
=B7 Interactive television
=B7 Video-on-demand systems
=B7 Compression algorithms

IDMS'98 will consist of a three day technical program, a full day of
tutorials, and demonstrations during the workshop. In order to keep the
flavour of a workshop, the number of participants will be restricted.
=46urthermore, we encurage contributions in form of full papers and
position papers. Full papers are expected to describe innovative and
significant work. The purpose of position papers is to provide a seed
for debate and discussion. Position papers enable researchers to present

exciting ongoing work in early stages, suggestions for future
directions,
and concerns about current developments. Both types of papers will be
reviewed by the program committee and printed in the workshop
proceedings. The proceedings will be published in the Springer LNCS
series and will be available during the workshop. It is intended to
forward selected papers to a special issue of the "Computer
Communications" Journal.

Information for authors: Authors are invited to submit full papers and
position papers for review. Submitted manuscripts must describe original

work (not submitted or published elsewhere). Full papers must not be
longer than 20 double spaced pages and position papers must not be
longer than 8 double spaced pages. Both types of papers should contain
an
abstract of approximately 300 words, and include title, authors and
affiliations. The submission process of papers will be handled
electronically. Detailed description of the electronic submission
procedures is available on the IDMS'98 web page:

http://www.unik.no/~idms98.

Authors without web access may send mail to
idms98@unik.no requesting electronic submission information. Authors
unable to submit electronically are invited to send 5 copies of their
contribution to one of the workshops chairs ATTN: IDMS'98.

Important dates:
 Submission due: February 1, 1998
 Notification of acceptance: April 15, 1998
 Camera ready version: May 15, 1998
 Workshop: September 9 - 11, 1998

Program co-chairs: Vera Goebel and Thomas Plagemann
UniK - Center for Technology at Kjeller, University of Oslo, P.O. Box
70, N-2007 Kjeller, Norway
Email: {goebel; plageman}@unik.no, Phone: +47/63.81.45.70, Fax:
+47/63.81.81.46

Program Committee:

=46. A. Aagesen, NTNU, Norway
H. Affifi, ENST Bretagne, France
E. Biersack, Institut Eur=E9com, France
G. Bochmann, U. Montreal, Canada
B. Butscher, DeTeBerkom, Germany
A. T. Campbell, Columbia U., USA
S. Chanson , Hongkong U., HK
L. Delgrossi, U. Piacenza, Italy
M. Diaz, LAAS-CNRS, France
=46. Eliassen, U. Troms=F8, Norway
W. Effelsberg, U. Mannheim, Germany
D. Ferrari, U. Cattolica Piacenza, Italy
J.-P. Hubaux, EPFL, Switzerland
D. Hutchison, Lancaster U., UK
W. Kalfa, TU Chemnitz, Germany
T. D. C. Little, Boston U., USA
E. Moeller, GMD FOKUS, Germany
K. Nahrstedt, U. Illinois, USA
G. Parulkar, Washington U., USA
B. Pehrson, KTH Stockholm, Sweden
S. Pink, SICS, Sweden
B. Plattner, ETH Zurich, Switzerland
H. Scholten, U. Twente, Netherlands
R. Steinmetz, GMD, Germany
H. Tokuda, Keio U., Japan
L. Wolf, TH Darmstadt, Germany
M. Zitterbart, TU Braunschweig, Germany

ACM Multimedia'98 takes place in Bristol (UK) in the week following
IDMS'98: http://www.acm.org/sigmm/MM98.

 -------------------------------------------------------------------------
 Dr. Vera Goebel                         Tel. +47/63.81.45.71.219 (direct)
 Associate Professor                  or Tel. +47/63.81.45.70 (swichboard)
 UNIK
 Center for Technology at Kjeller
 University of Oslo                      Fax. +47/63.81.81.46

 PO BOX 70
 N-2007 Kjeller                          email: goebel@unik.no
 Norway                                  http://www.unik.no/~goebel/
 -------------------------------------------------------------------------





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--NextPart

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

	Title		: RTP: A Transport Protocol for Real-Time Applications
	Author(s)	: V. Jacobson, S. Casner, R. Frederick, H. Schulzrinne
	Filename	: draft-ietf-avt-rtp-new-00.txt,.ps
	Pages		: 94
	Date		: 02-Jan-98
	
         This memorandum is a revision of RFC 1889 in preparation
         for advancement from Proposed Standard to Draft Standard
         status. Readers are encouraged to use the PostScript form
         of this draft to see where changes from RFC 1889 are
         marked by change bars. The revision process is not yet
         complete; some changes which have been discussed and
         tentatively accepted in meetings of the Audio/Video
         Transport working group have not yet been incorporated
         into this draft.

         This memorandum describes RTP, the real-time transport
         protocol. RTP provides end-to-end network transport
         functions suitable for applications transmitting real-
         time data, such as audio, video or simulation data, over
         multicast or unicast network services. RTP does not
         address resource reservation and does not guarantee
         quality-of-service for real-time services. The data
         transport is augmented by a control protocol (RTCP) to
         allow monitoring of the data delivery in a manner
         scalable to large multicast networks, and to provide
         minimal control and identification functionality. RTP and
         RTCP are designed to be independent of the underlying
         transport and network layers. The protocol supports the
         use of RTP-level translators and mixers.
 
 
   This specification is a product of the Audio/Video Transport working
   group within the Internet Engineering Task Force. Comments are
   solicited and should be addressed to the working group's mailing list
   at rem-conf@es.net and/or the authors.


 


Internet-Drafts are available by anonymous FTP.  Login with the username
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type "cd internet-drafts" and then
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ENCODING mime
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From rem-conf Mon Jan 05 07:22:09 1998 
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From: Dilip D Kandlur <kandlur@us.ibm.com>
To: <rem-conf@es.net>
Subject: MMCN'98: Final Program and Registration Information
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       Advance  Program
                                    &
                  Call for Participation


           SPIE/ACM MULTIMEDIA COMPUTING AND NETWORKING 1998

                   San Jose, California
      January 26-28 1998



  Conference   Kevin Jeffay, University of North Carolina at Chapel Hill
  Chairs       Dilip Kandlur, IBMJT.J. Watson Research Center
               Timothy Roscoe, Persimmon I.T., Inc.

  Program      Peter Beadle, University of Wollongong
  Committee    Ming-Syan Chen, National Taiwan University
               Wu-Chi Feng, Ohio State University
               Martin Freeman, Philips Research
               J.J. Garcia-Luna-Aceves, U.C. Santa Cruz
               Anoop Gupta, Stanford University
               Mark Hayter, DEC Systems Research Center
               Sugih Jamin, University of Michigan at Ann Arbor
               Paul Jardetzky, Sun Microsystems
               Chuck Kalmanek, AT&T Research
               Ian Leslie, University of Cambridge
               Sape Mullender, University of Twente
               Klara Nahrstedt U.I. Urbana-Champaign
               Guru Parulkar, Washington University
               Lawrence A. Rowe, U.C. Berkeley
               Debanjan Saha, IBM T.J. Watson
               Henning Schulzrinne, Columbia University
               Doug Shepherd, Lancaster University
               Brian Smith, Cornell University
               Cormac Sreenan, AT&T Research
               Ralf Steinmetz, T.U. Darmstadt
               Harrick Vin, University of Texas at Austin
               Jonathan Walpole, Oregon Graduate Institute
               Raj Yavatkar, Intel Corporation
               Hui Zhang, Carnegie Mellon University


Registration & hotel info can be found at URL:

       http://www.spie.org/web/meetings/programs/pw98/ei98_home.html

  +------------------------------------+
  |                                    |
  |   Register by January 7, 1998 for  |
  |  for early registration discount!! |
  |                                    |
  +------------------------------------+


MMCN '98 ADVANCE PROGRAM
------------------------

Monday 26 January

8.45 am: Welcome and Opening Remarks

9.00 to 10.30 am: Session 1: Multimedia System Development Tools

        Middleware support for distributed multimedia and
        collaborative computing, K. Birman, R. Friedman, M. Hayden,
        Cornell Univ.; I. Rhee, Emory Univ.

        Multiplatform simulation of video playout
        performance, L. Gharai, R. Gerber, Univ. of
        Maryland/College Park

        A Software-only video production switcher for the Internet
        MBone, T.H. Wong, K. D. Mayer-Patel, D. Simpson, L. A. Rowe,
        Univ. of California/Berkeley

11.00 am to 12.30 pm: Session 2: Operating Systems I

        Applying statistical process controls to the adaptive
        rate control problem: a framework for the streaming of
        hetergeneous streams, N. R. Manobar,
        M. H. Willebeek-LeMair, IBM Thomas J. Watson Research Ctr.;
        A. Prakash, Univ. of Michigan/Ann Arbor

        An integrated input/output system for kernel data
        streaming, F. W. Miller, S. K. Tripathi, Univ. of
        Maryland/College Park

        Measurement-based admission control and resource
        allocation for multimedia applications, N. Stratford,
        P. Barham,S. Crosby, F. Toomey, M. Huggard, Univ. of Cambridge
        (UK)

Lunch Break

2.00 to 3.30 pm: Keynote Address: Enhanced Display Environments for
                Telecollaboration and Personal Computing in the Office of the
                Future
                Henry Fuchs, Federico Gil Professor of Computer
                Science, Univ. of North Carolina/Chapel Hill


4.00 to 5.30 pm: Session 3: Video-on-Demand

        Supporting interactive scanning operations in VoD
        systems, G. Apostolopoulos, Univ. of Maryland/College Park;
        M. Krunz, Univ of Arizona; S. K. Tripathi, Univ. of
        Maryland/College Park

        Modelling prerecorded compressed video streams for fast
        bandwidth smoothing implementations, W. C. Feng, M. Liu,
        C. C. Lam, The Ohio State Univ.

        A system for demonstrating dynamic service aggregation
        in VoD scenarios, P. Basu, A. Narayanan, R. Krishnan,
        T. D. C. Little, Boston Univ.


Tuesday 27 January

8.30 to 9.15 am: Plenary Speaker

        Multimedia Communications: What's Next?
        Leonardo Chiariglione, CSELT/Telecom Italia (Italy)

9.30 to 11.00 am: Session 4: Operating Systems II

        Symphony: an integrated multimedia file system,
        P. J. Shenoy, P. Goyal, S. S. Rao, H. M. Vin, Univ. of
        Texas/Austin

        Adaptive prefetching for device independent file
        I/O, D. Revel, D. McNamee, D. Steere, J. Walpole, Oregon Graduate
        Institute of Science and Technology

        Resource kernels: a resoure-centric approach to
        real-time and multimedia systems, R. Rajkumar, K. Juvva,
        A. Molano, S Oikawa, Carnegie Mellon Univ.

11.15 am to 12.45 pm: Session 5: The World Wide Web

        Characterizing videos on the World Wide Web,
        S. Acharya, B. C. Smith, Cornell Univ.

        Static caching of Web servers, Z. Liu, P. Nain,
        N. Niclausse, INRIA Ctr. Sophia Antipolis (France); D. Towsley,
        Univ. of Massachusetts/Amherst

        Resource-based caching for Web servers, R. Tewari,
        H. M. Vin, Univ. of Texas/Austin; A. Dan, D. Sitaram, IBM Thomas
        J. Watson Research Ctr.

Lunch/Exhibit Break

2.00 to 3.30 pm: Keynote Address: Low Latency Media Delivery in a
                Consumer Internet Service

        Michael Schwartz, Director of Server Engineering and
        Senior Scientist, @Home Network

4.00  to 5.30 pm: Session 6: Multimedia Applications

        Integrated audio-visual processing for object
        serialization and tracking, G. S. Pingali, Lucent
        Technologies/Bell Labs.

        Accelerating M-JPEG compression with temporal
        information, H. Boenisch, K. Froitzheim, P. Schulthess,
        Univ. Ulm (FRG)

        Cross-model retrieval of scripted speech audio,
        C. B. Owen, F. Makedon, Dartmouth College

Wednesday 28 January

8:30 to 9:15am: Plenary Speaker

        The Computer Revolution Hasn't Happened Yet
        Alan Kay, Disney Fellow and Vice President of Research and
        Development, The Walt Disney Company

9.30 to 11.30 am: Session 7: Flow and congestion control

        Adaptive source rate control for wireless video
        conferencing, H. Liu, M. El Zarki, Univ. of Pennsylvania

        Flow and congestion control for internet streaming
        applications, S. Cen, C. Pu, J. Walpole, Oregon Graduate
        Institute of Science and Technology

        Invited Paper

11.30am to 1.00 pm: Panel Discussion:

        The Future of Multimedia Research, Or, What Am I Doing And Why?
        Moderator: Lawrence A. Rowe, University of California/Berkeley








From rem-conf Mon Jan 05 09:06:44 1998 
From rem-conf-request@es.net Mon Jan 05 09:06:43 1998
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Subject: Re: I-D ACTION:draft-ietf-avt-rtp-new-00.txt,.ps 
In-reply-to: Your message of "Mon, 05 Jan 1998 10:16:23 EST."
             <199801051516.KAA15955@ns.ietf.org> 
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Date: Mon, 05 Jan 1998 18:00:54 +0100
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Hi

On page 7 a photo URL SDES is proposed (well not really as it hasn't been 
discussed yet :-).  How about adding an optional personal home-page URL SDES 
type? I have done this earlier by inventing my own SDES type (and by that 
breaking the specs, yes :-).

The RTP specs should for both these new SDES types (if accepted :-) include 
text about a random delay for when the page can be retrieved from the time the 
URL was first known through the RTCP SDES packet. This could hinder the sudden 
overload of the webserver in question.

/P




From rem-conf Mon Jan 05 09:46:46 1998 
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To: Peter Parnes <peppar@cdt.luth.se>
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Subject: Re: I-D ACTION:draft-ietf-avt-rtp-new-00.txt,.ps
References: <199801051700.SAA20544@basil.cdt.luth.se>
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Peter Parnes wrote:
> 
> Hi
> 
> On page 7 a photo URL SDES is proposed (well not really as it hasn't been
> discussed yet :-).  How about adding an optional personal home-page URL SDES
> type? I have done this earlier by inventing my own SDES type (and by that
> breaking the specs, yes :-).
> 
> The RTP specs should for both these new SDES types (if accepted :-) include
> text about a random delay for when the page can be retrieved from the time the
> URL was first known through the RTCP SDES packet. This could hinder the sudden
> overload of the webserver in question.

One simple and conservative approach would be to delay automatic
retrieval until it is one's turn to send an SDES RR. For a low-bandwidth
audio channel, this would limit total web retrieval rates to about two
hits/second. For web pages instead of photos, retrievals will likely be
triggered by human action, imposing some amount of randomization.

Having the ability to periodically transmit a fixed JPEG or GIF image
via RTP might be an alternative for the photo transmission, but would
tend to use more bandwidth.


> 
> /P



From rem-conf Tue Jan 06 09:07:30 1998 
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Subject: H.263+ without change marks
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Draft 20 of the revised H.263 ("H.263+") is available in PDF format without
change marks at:
http://standard.pictel.com/h263+rtp/draft20ncm.pdf




From rem-conf Fri Jan 09 07:16:23 1998 
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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		: Options for Repair of Streaming Media
	Author(s)	: C. Perkins, O. Hodson
	Filename	: draft-ietf-avt-info-repair-02.txt
	Pages		: 9
	Date		: 08-Jan-98
	
    This document summarizes a range of possible techniques
    for the repair of continuous media streams subject to packet
    loss.  The techniques discussed include redundant transmission,
    retransmission, interleaving and forward error correction.
    The range of applicability of these techniques is noted,
    together with the protocol requirements and dependencies.

Internet-Drafts are available by anonymous FTP.  Login with the username
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type "cd internet-drafts" and then
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From rem-conf Fri Jan 09 09:39:05 1998 
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Date: Fri, 09 Jan 1998 18:26:25 +0100
From: Jorg Nonnenmacher <nonnen@eurecom.fr>
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Hi,

The current RTCP spec requires every receiver to send a
RTCP packet periodically and randomized in an interval corresponding
to the number of receivers.

For very popular multicasts/broadcasts with a very large number
of receivers two things may be of concern:

* The period for sendings of a single receiver becomes very large,
  such that the intended reception information of a single receiver
  is old (and meaningless?)

* Further, for a very large number of receivers the identity of each single
receiver might not be of interest. 

One can give a threshold for the number of receivers
that allows to switch RTCP from the current mechanism
to a sampling method. Another (lower) threshold can be given to switch back
>from the sampling method to the everybody-sends RR RTCP.

One solution to outdated reception status is to define (announce by the 
sender) classes of reception quality.
Receivers group themselves in the defined/announced classes.
The receivers are sampled by each class separately.
For each class an estimate of the number of receivers can be given due
to the number of responses.


Joerg
 





From rem-conf Fri Jan 09 18:56:58 1998 
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Date: Fri, 09 Jan 1998 21:25:27 -0500
From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
Organization: Lucent Technologies
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Jorg Nonnenmacher wrote:
> 

> One can give a threshold for the number of receivers
> that allows to switch RTCP from the current mechanism
> to a sampling method. Another (lower) threshold can be given to switch back
> from the sampling method to the everybody-sends RR RTCP.

There are some problems if some people stop sending RTCP packets, and
others continue, based on group size. The decision about whether to send
or not will depend on the group size, but the group size is determined
by people sending packets. This interrelationship can cause oscillations
and other bad things. For example, lets say (in the extreme case) when
the group size reaches 100, half of these stop sending RR. These members
will eventually time out, and cause the group to drop back to 50. Then,
since the group size is smaller, everyone starts sending again, causing
a sharp increase in the group size back up to 100, and then back down
again, etc. Other similar algorithms all result in these kinds of
behaviors because of the ambiguity.

> 
> One solution to outdated reception status is to define (announce by the
> sender) classes of reception quality.
> Receivers group themselves in the defined/announced classes.
> The receivers are sampled by each class separately.
> For each class an estimate of the number of receivers can be given due
> to the number of responses.

I think this may actually cause more oscillatory problems than the above
approach. Since users will move from class to class as their reception
varies, group sizes will be very dynamic to begin with, which will
probably amplify the oscillations. 

If you want to get updates from receivers more frequently, some kind of
polling from a central monitor seems more appropriate. 

-Jonathan R.

-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
PHONE: (908) 949-6418                       Rm. 4D-534B
FAX:   (908) 834-5379
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen



From rem-conf Sat Jan 10 19:38:06 1998 
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Date: Sat, 10 Jan 1998 22:20:40 -0500
From: Henning Schulzrinne <hgs@cs.columbia.edu>
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Organization: Columbia University (home)
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As Jorg Nonnenmacher points out, large groups raise a particular problem
of stale QOS reports, as it is likely that each receiver will get to
send at most one report covering the whole time since it joined the
group. Finding out about loss averaged over an hour is only modestly
useful for diagnosing problems and doesn't really tell you a whole lot
about the user experience (it could have been 10% constant loss or six
minutes of no reception; the first might have been hidden by local
recovery, the latter wouldn't).

Instead of sampling and further complicating the transmission mechanism,
we might consider two alternatives that require only modest changes to
the current scheme:

1) Make the indication of the RR loss fraction reflect only recent
experience.

2) Add an additional RR field for recent loss experience, where recent
may be the shorter of last RR transmission and, say, two minutes.

A variation on the proposal by Jorg had been made earlier, namely that
only those that experience problems would participate in the receiver
reports, with reconsideration limiting the avalanche effect of sudden
widespread QOS degradation. Getting the names of those might even be
useful: "Dear Mr. Subscriber; we gather that your reception of the world
boxing championship was not as good as you have a right to expect.
Please accept our apologies and a coupon for Wrestlemania pay-per-view."

Henning



From rem-conf Sat Jan 10 20:35:22 1998 
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To: hgs@cs.columbia.edu
From: Tim Dorcey <tdorcey@boxtop.com>
Subject: Re: RTP QOS reports for large groups
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At 10:20 PM -0500 1/10/98, Henning Schulzrinne wrote:
>As Jorg Nonnenmacher points out, large groups raise a particular problem
>of stale QOS reports, as it is likely that each receiver will get to
>send at most one report covering the whole time since it joined the
>group. Finding out about loss averaged over an hour is only modestly

When it is a receiver's turn to send, it should send a batch of reports at
fixed intervals that are small enough to be useful.  So, a typical
receiver's reporting times would look like:

----x-x-x-x-------------------x-x-x-x---------------x-x-x-x-------------

instead of:

---x------x------x------x------x-----x-----x-----x-----x-----x-----x----


I'm not sure how much can be done with aggregate QOS reports in the first
place, but at least this solves the problem of keeping them from getting
too widely spaced in time.

Tim






From rem-conf Sat Jan 10 23:26:26 1998 
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Date: Sat, 10 Jan 1998 23:24:53 -0800 (PST)
From: Stephen Casner <casner@precept.com>
To: rem-conf@es.net
cc: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Subject: Re: RTP QOS reports for large groups
In-Reply-To: <34B83A88.FD9941F2@cs.columbia.edu>
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I support the idea of refining the definition of the "loss fraction"
field in RTCP reception reports.  When that field was added (at the
1995 San Jose IETF, if memory serves), the purpose was to allow single
reception reports to be useful because the interval between reports
could become quite long in large groups.  Because it is difficult to
choose just how long a shorter measurement interval for the loss
fraction should be, the consensus of the working group at that time
was that the interval should still be the whole inter-report
interval.  That's what is currently in the spec.  Some folks expressed
the sentiment then that we should begin with that definition and see
what we might learn during the Proposed Standard stage.

I have always felt that this topic was worthy of further
consideration, and I think it is appropriate to ask now as we prepare
for advancement to Draft Standard status whether this aspect of the
protocol should be changed.

The hard problem is still there: what should be shorter interval be?

  - Henning Schulzrinne suggested (actually for an additional field,
    but I think it could also apply to the existing field): "the
    shorter of last RR transmission and, say, two minutes".

  - Perhaps the interval should be scaled by the session bandwidth, so
    the measurement interval is approximately a constant number of
    bits.

  - Or perhaps there is some network time constant that would reflect
    the period over which changes in congestion were likely to occur,
    suggesting a contant interval rather than constant number of bits.

Does anyone have any analytical or experimental results that would
suggest the right choice?
							-- Steve




From rem-conf Sun Jan 11 03:52:03 1998 
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To: hgs@cs.columbia.edu
cc: rem-conf@es.net
Subject: Re: RTP QOS reports for large groups
In-reply-to: Your message of "Sat, 10 Jan 1998 22:20:40 EST." <34B83A88.FD9941F2@cs.columbia.edu>
Date: Sun, 11 Jan 1998 11:46:22 +0000
Message-ID: <1033.884519182@cs.ucl.ac.uk>
From: Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>
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In message <34B83A88.FD9941F2@cs.columbia.edu>, Henning Schulzrinne typed:


 >>As Jorg Nonnenmacher points out, large groups raise a particular problem
 >>of stale QOS reports, as it is likely that each receiver will get to
 >>send at most one report covering the whole time since it joined the
 >>group. Finding out about loss averaged over an hour is only modestly
 >>useful for diagnosing problems and doesn't really tell you a whole lot
 >>about the user experience (it could have been 10% constant loss or six
 >>minutes of no reception; the first might have been hidden by local
 >>recovery, the latter wouldn't).

henning,

your message points out part of the problem with RTP reports...you
start with talking about both dianosing problems and userexperience

one of these needs aggregate (e.g. per router/leaf/branch) loss info,
while the other needs per recevier/sender info...

i thinone might want to have
i) per session selection of strategies
ii) dynamic selection of strategies
for reporting per receive/send,er and for employing localization and
aggregation of reports (e.g. scoped reports and summarizers)

default would be as per now, but with your reconsideation of course
but one could imagine playing with a system that used a hierarchical
set of summarizers (receives would report on behalf of nearby regions
of receivers, who would report with some scope limit of some kind) and
then maybe switch to global reports in the event of problems...

of coure, just as with reliable multicast work, localizaion and robust
representative summarizer election are somewhat difficualt, ESPECIALLY
with the current IP multicast architecture....a slight change is
called for i think by this too!

(see the rm list for idea on turning points and so on...)

cheers

 >>
 >>Instead of sampling and further complicating the transmission mechanism,
 >>we might consider two alternatives that require only modest changes to
 >>the current scheme:
 >>
 >>1) Make the indication of the RR loss fraction reflect only recent
 >>experience.
 >>
 >>2) Add an additional RR field for recent loss experience, where recent
 >>may be the shorter of last RR transmission and, say, two minutes.
 >>
 >>A variation on the proposal by Jorg had been made earlier, namely that
 >>only those that experience problems would participate in the receiver
 >>reports, with reconsideration limiting the avalanche effect of sudden
 >>widespread QOS degradation. Getting the names of those might even be
 >>useful: "Dear Mr. Subscriber; we gather that your reception of the world
 >>boxing championship was not as good as you have a right to expect.
 >>Please accept our apologies and a coupon for Wrestlemania pay-per-view."
 >>
 >>Henning
 >>

 cheers

   jon




From rem-conf Sun Jan 11 03:54:04 1998 
From rem-conf-request@es.net Sun Jan 11 03:54:04 1998
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	id 0xrLvx-00048f-00; Sun, 11 Jan 1998 03:52:05 -0800
From: Toerless Eckert <Toerless.Eckert@Informatik.Uni-Erlangen.de>
Message-Id: <199801111148.MAA17697@faui45r.informatik.uni-erlangen.de>
Subject: Optimized mpeg encoders ?
To: mbone@isi.edu, rem-conf@es.net
Date: Sun, 11 Jan 1998 12:48:10 +0100 (MET)
Cc: tnzoerne@immd4.informatik.uni-erlangen.de (Thorsten Zoerner (CIP Admin)),
        Carsten.Wiethoff@blns.siemens.de (Carsten Wiethoff)
Organisation: CSD IMMD IV, University of Erlangen, Germany
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Hi

I was wondering if someone on this group knows if there are 
optimized versions of mpeg layer2 audio encoders out there in source for 
free. I currently know only of the ISO reference implementation which is
especially pessimized in terms of performace so that the patent holders
can better sell their commercial stuff (just guessing).

There seem to be a couple of other mpeg layer-2 audio encoders out there, 
but what i've found so far is only binaries for DOS/Windooz/Mac, so
i cannot evaluate their performance anyway, apart from not being able to
guess if they would be available in source for free.

If anybody got hints here for me, please reply. With the layer 2 reference
encoder our best outcome is something slightly more than 100% CPU for
128 Kbps stereo encoding of 44.1 Kzh on an UltraSparc/250Mhz, so this definitely
isn't going to work with a videoconferencing application.

Best regards
	Toerless

P.S.:
I am asking for layer-2 encoder software, because the layer-2
reference encoder does contain a useful psychoaccoustic model and
thus optimization could have been done purely on the algorithmic
implementation to deliver higher throughput. As for layer 3, there
doesn't even seem to be a useful psychoaccoustic model published (i.e.:
the one in the ISO reference encoder delivers only slighter better
quality per bit than layer 2, but far from sufficient to get a better
bitrate/cpu-cycles ratio). Everybody who want's to use a good model seems
to have to buy licenses from FHG, and this doesn't work for software designated
to become freeware. Also, i don't want to support such a monopoly, even though
they're next door to us and are funded with my tax-money too (i 
especially like that part).
If i am mistaken in these point's i would be glad to go for layer 3.



From rem-conf Mon Jan 12 00:09:04 1998 
From rem-conf-request@es.net Mon Jan 12 00:09:04 1998
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	id 0xrelc-0003Gc-00; Sun, 11 Jan 1998 23:58:40 -0800
Date: Sun, 11 Jan 1998 23:58:37 -0800 (PST)
Message-Id: <199801120758.XAA04994@shanti.eecs.berkeley.edu>
From: Matt Podolsky <podolsky@shanti.eecs.berkeley.edu>
To: rem-conf@es.net
CC: Matt Podolsky <podolsky@shanti.eecs.berkeley.edu>,
        Steve McCanne <mccanne@eecs.berkeley.edu>,
        Cindy Romer <cromer@eecs.berkeley.edu>
Subject: Paper available: Simulation of FEC for Internet Audio 
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A copy of our Infocom '98 paper describing simulation
work we have done on FEC for Internet audio is now
available on-line at the link below.  The paper is
entitled "Simulation of FEC-Based Error Control for
Packet Audio on the Internet" and was written by
myself, Cynthia Romer, and Steven McCanne.

http://www-wavelet.eecs.berkeley.edu/~podolsky/publications/infocom98.ps

The paper focuses on FEC techniques jointly developed
by groups at UCL and INRIA, and implemented in such
tools as RAT and FreePhone.  We refine these novel
coding techniques into a formal framework that we call
``signal processing-based FEC'' (SFEC) and use our
framework to more rigorously evaluate the relative
merits of this approach.  Through extensive simulation,
we evaluate the ``scalability'' of SFEC for packet
audio --- \ie, the ability for a coding algorithm to
improve aggregate performance when used by all sources
in the network --- and find that optimal signal quality
is achieved when sources react to network congestion
not by blindly adding FEC, but rather by adding FEC in
a controlled fashion that simultaneously constrains the
source-coding rate.

Matt Podolsky



From rem-conf Mon Jan 12 02:15:45 1998 
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	id 0xrgkd-0004Ol-00; Mon, 12 Jan 1998 02:05:47 -0800
From: Toerless Eckert <Toerless.Eckert@Informatik.Uni-Erlangen.de>
Message-Id: <199801121004.LAA21225@faui45r.informatik.uni-erlangen.de>
Subject: sdr announcement questions
To: rem-conf@es.net
Date: Mon, 12 Jan 1998 11:04:56 +0100 (MET)
Cc: mbone@isi.edu
Organisation: CSD IMMD IV, University of Erlangen, Germany
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Hi

Question a): How do i solve the following problem:
	  
	     I want to make a sdr-announcement that should have a living-time
	     "forever", i.e.: something like a continuous channel, or
	     continually regular channel. Whatever. I currently solved
	     this by guessing which field holds the end-time and then
	     modifying it. Now the problem with this approach to me seems
	     that i cannot allow for any mistakes in this announcement,
	     because the end-time seem also to convey something like the
	     "cache-timeout" for such an announcement. I.e.: My sdr sends
	     out such an announcement and there's a typo in it. I want to
	     correct it, but that doesn't help, because people who've already
	     reveieved the erroneous announcement will still use that from
	     their local cache. There doesn't seem to be something like
	     a sequence number for announcements, or am i mistaken there ?

Replacement question: where's the documentation for the exact format and
semantics of the .sdr announcement files ? ;-))


Best regards
	Toerless



From rem-conf Mon Jan 12 02:26:56 1998 
From rem-conf-request@es.net Mon Jan 12 02:26:55 1998
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	id 0xrh0o-0004eR-00; Mon, 12 Jan 1998 02:22:30 -0800
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To: Toerless Eckert <Toerless.Eckert@Informatik.Uni-Erlangen.de>
cc: rem-conf@es.net, mbone@isi.edu
Subject: Re: sdr announcement questions 
In-reply-to: Your message of "Mon, 12 Jan 1998 11:04:56 +0100."
             <199801121004.LAA21225@faui45r.informatik.uni-erlangen.de> 
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Date: Mon, 12 Jan 1998 11:22:03 +0100
From: Peter Parnes <peppar@cdt.luth.se>
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If you want to change a SDR SDP cache file you have to first quit sdr. The 
edit the correct cache file and change the second field on the line that 
starts with 'o=' to a higher number. This field is the 'version number' of 
this announcement.

If you want a session that never time out then you have to change the t-field 
to
t=0 0 

The format is documented in the SDP specification. 

NOTE that this is illegal according to the SDP specifications ;-) 

/P




From rem-conf Mon Jan 12 06:24:43 1998 
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	id 0xrkcH-0006pP-00; Mon, 12 Jan 1998 06:13:25 -0800
From: Edgar Hellfritsch <edgar.hellfritsch@rrze.uni-erlangen.de>
Subject: Re: sdr announcement questions
To: rem-conf@es.net
Date: Mon, 12 Jan 1998 15:12:36 +0100 (MET)
In-Reply-To: <199801121022.LAA00690@basil.cdt.luth.se> from "Peter Parnes" at Jan 12, 98 11:22:03 am
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> If you want to change a SDR SDP cache file you have to first quit sdr. The 
> edit the correct cache file and change the second field on the line that 
> starts with 'o=' to a higher number. This field is the 'version number' of 
> this announcement.
> 
> If you want a session that never time out then you have to change the t-field 
> to
> t=0 0 
> 
> The format is documented in the SDP specification. 
> 
> NOTE that this is illegal according to the SDP specifications ;-) 
> 
> /P
> 
what about private session announcements? they don't show up in the cache dir.

sdr doesn't seem to receive private session announcements when starting it
AFTER the session initialisation has already taken place. if a private session
is announced while sdr is listening, the session shows up immediately.
looks to me like private sessions are announced once and never again.
ist that a known effect?

edgar



From rem-conf Tue Jan 13 07:21:09 1998 
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	id 0xs7iU-00004H-00; Tue, 13 Jan 1998 06:53:22 -0800
From: "Stavros A. Papadakis" <spapad@csi.forth.gr>
Posted-Date: Tue, 13 Jan 1998 16:40:50 +0200 (EET)
Date: Tue, 13 Jan 1998 16:39:30 +0200
Message-Id: <9801131439.AA30574@panoramix.csi.forth.gr>
Organization: Institute of Computer Science,
	Foundation for Research and Technology-Hellas (FORTH)
	Science and Technology Park of Crete
	Vassilika Vouton, P.O.Box 1385
	GR 711 10 Heraklion, Crete, Greece
	tel.: +30 (81) 39 16 00,  fax: +30 (81) 39 16 01
To: spapad@csi.forth.gr
Subject: Announcing the Second European Conference on Research and Advanced Technology for Digital Libraries
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                     Second European Conference on
                  Research and Advanced Technology for
                          Digital Libraries

     European       European              ICS-FORTH      University of
     Union          Research                             Crete
                    Consortium for
                    Informatics and
                    Mathematics


                        19 - 23 September, 1998
                        Heraklion, Crete, Greece


     Web Page
         http://www.csi.forth.gr/2EuroDL
     E-mail
         ecdl@cc.uch.gr

________________________________________________________________________________

- Objectives 

This conference is the second of a series of European conferences on research
and technology for digital libraries funded by the European Commission's TMR
Programme. Its objectives are: to bring together researchers from multiple
disciplines whose science relates to the development of digital libraries;
to provide an opportunity for these scientists to form a research community
in Europe specific to digital library development and to enable them to discuss
issues and strategies specific to the European context; to assist young
researchers in establishing relationships with senior scientists in their areas
of interest; to enable review and discussion of research under way in Europe,
the US, Japan and other countries on digital libraries; to stimulate
researchers, especially young scientists, to explore new areas of interest in
digital library development; to establish a forum for discussion of issues
specific to Europe such as interoperability, multilinguality, intellectual
property policy, and information commerce; to provide an opportunity for
researchers in the relevant enabling technologies and information sciences,
to discuss issues related to interoperability between world wide distributed
digital libraries. 

>From a technical point of view, the European Conferences series aims to
contribute to the definition of those digital library parameters which
especially influence issues of access, retrieval, and interaction with
information; to identify key problems which must be solved to make digital
library services an effective reality; to identify a general structure or
framework for integrating research and solutions; and to propose and encourage
specific, high priority research directions within such a framework.

________________________________________________________________________________

- Topics 

The conference organisers solicit papers on topics related to digital
libraries, including but not limited to the following list: 

 o Digital Library Models, Frameworks, and System Requirements 
 o Metadata 
 o System Integration and Architecture Issues 
 o Interoperability, Scalability 
 o Networked Information Discovery, Agent Technologies 
 o Information Retrieval, Organisation, Navigation - Tools and Paradigms 
 o Multilinguality 
 o Role of Knowledge Representation Systems in Digital Library Interactions 
 o Collecting, Capturing, Filtering, Cataloging, Indexing, 
 o Preserving 
 o Intellectual Property Rights, Terms and Conditions, Rights Management 
 o Authoring, Electronic Publishing, Electronic Commerce and
   Information Economies 
 o Economic and Social Implications and Issues 
 o User Interfaces 
 o Handling of Graphics, GIS, Medical Data, Multimedia Information,
   Experimental Data and
 o Scientific Models 

________________________________________________________________________________

- Conference Programme 

The conference will be held in Heraklion, Crete, Greece. Tutorials will be
organised on the 19th and 20th of September 1998. The opening session will
take place at 9.00a.m. on Monday the 21th of September 1998 and the final
session will take place on Wednesday afternoon, the 23th of September 1998.
Full details on the scientific programme of the Conference will be published
on our Web site by the 1st of July 1998. 

________________________________________________________________________________

- Important Dates 

       15 March 1998    Proposals for tutorials, panels and demos
                        due to the Programme Chair 
       15 April 1998    Notification of tutorial, panel and demo acceptance 
        15 May 1998     Papers and proposals for posters
                        due to the Programme Chair 
       25 June 1998     Notification of paper and poster acceptance 
        1 July 1998     Scientific Programme on the Web 
       25 July 1998     Final papers due 
   19,20 September 1998 Tutorials 
   21-23 September 1998 Conference 

________________________________________________________________________________

- Panels 

Suggestions for the organisation of panel sessions on one of the proposed
topics or on related topics are welcomed. Proposals should include a short
CV and position paper for each panelist. 

________________________________________________________________________________

- Posters 

During the conference a space will be reserved for poster sessions. Research
projects of any scale are invited to illustrate innovative concepts and
prototype systems. 

Poster proposals should include title, names of presenters and outline
(max. 500 words). 

________________________________________________________________________________

- Tutorials 

Tutorial days will be held before the Conference, on Saturday the 19th and
Sunday the 20th of September 1998. Proposals for tutorials are solicited.
Tutorials would be either half day (3 hours) or full day (6 hours).
Each proposal should include a title, a summary (intentions, objectives, etc.),
duration and a short CV of the instructor(s). 

________________________________________________________________________________

- Demos 

Result demonstrations of on-going projects are strongly encouraged. Those
interested should submit a description of the intended Demo to the
Programme Chair. 

________________________________________________________________________________

- Papers 

Papers (max 20 pages, double spaced) should be submitted electronically in HTML
format, either by e-mail to the Conference Secretariat, ecdl@cc.uch.gr,
or to our ftp site, ftp://ftp.ics.forth.gr/2EuroDL. 

________________________________________________________________________________

- Proceedings 

The Proceedings will be published by Springer as a volume in their Lecture
Notes in Computer Science series and will be distributed at the Conference. 

________________________________________________________________________________

- Fellowship for Young Researchers 

A limited number of fellowships for the Conference and also for Tutorials are
available for young researchers who are citizens of European Union countries or
Liechtenstein, Norway and Iceland. The fellowship offers free registration for
the participants and, in special cases where necessary and appropriately
justified, may pay for or reimburse travel and lodging expenses. 

________________________________________________________________________________

- Programme Chair 

    Christos Nikolaou, University of Crete & ICS-FORTH 
    Leoforos Knossou, GR-71110 Heraklion, Crete, Greece 
    Tel: +30 81 393199, 
    Fax: +30 81 210106 
    E-mail: nikolau@cc.uch.gr 

________________________________________________________________________________

- Programme Committee 

    Serge Abiteboul          INRIA, France 
    Robert B. Allen          Bellcore, USA 
    Thomas Baker             Asian Institute of Technology, Thailand 
    William Birmingham       University of Michigan, USA 
    Panos Constantopoulos    University of Crete & ICS-FORTH, Greece 
    Bruce Croft              University of Massachusetts, USA 
    Costis Dallas            Hellenic Ministry of Foreign Affairs, Greece 
    Edward A. Fox            Virginia Technical University, USA 
    Norbert Fuhr             University of Dortmund, Germany 
    Hector Garcia-Molina     Stanford University, USA 
    Keith Jeffery            RAL-CLRC, UK 
    Martin Kersten           CWI, Netherlands 
    Judith Klavans           Columbia University, USA 
    Carl Lagoze              Cornell University, USA 
    Clifford A. Lynch        Coalition for Networked Information, USA 
    Jeff MacKie-Mason        University of Michigan, USA 
    A. Desai Narasimhalu     National University of Singapore, Singapore 
    Ann Okerson              Yale University, USA 
    Olle Olsson              SICS, Sweden 
    Andreas Paepcke          Stanford University, USA 
    Nicholas Patrikalakis    MIT, USA 
    Carol Peters             IEI-CNR, Italy 
    Jakka Sairamesh          IBM-T.J. Watson Research Center, USA 
    Peter Schauble           ETH Zurich, Switzerland 
    Hans Joerg Schek         ETH Zurich, Switzerland 
    Eric Simon               INRIA, France 
    Ingeborg T. Solvberg     University of Science and Technology, Norway 
    Constantine Stephanidis  ICS-FORTH, Greece 
    Shigeo Sugimoto          University of Library and Information Science,Japan
    Costantino Thanos        IEI-CNR, Italy 
    Ulrich Thiel             GMD-IPSI, Germany 
    Stuart Weibel            OCLC, USA 

________________________________________________________________________________

- Local Organising Committee 

    Sarantos Kapidakis       ICS-FORTH, Greece 
    Penelope Constanta       ICS-FORTH, Greece 
    Spiros Lalis             University of Crete, Greece 
    Gioylh Koraoy            University of Crete, Greece
    Stella Vourou            University of Crete & ICS-FORTH, Greece
    Mixalhs Tzekakhs         University of Crete, Greece
    Maria Stavrakaki         University of Crete, Greece
    Rena Kalaitzaki          University of Crete, Greece
    Maria Prevelianaki       ICS-FORTH, Greece
    Liana Kefalaki           ICS-FORTH, Greece
    Dimitris Papadakis       University of Crete, Greece
    Manolis Marazakis        University of Crete, Greece
    Anastasia Anastasiadi    ICS-FORTH, Greece
    Stavros Papadakis        University of Crete, Greece

________________________________________________________________________________

- Contact Info 

For more information regarding this conference contact the conference
secretariat, 

    Rena Kalaitzaki and Maria Stavrakaki
    University of Crete, 
    Computer Science Department, 
    Tel: +30 81 393504 
    Fax: +30 81 393501 
    E-mail: ecdl@cc.uch.gr 


________________________________________________________________________________

You can subscribe to the announcement list of the "Second European Conference 
on Research and Advanced Technology for Digital Libraries"
by sending electronic mail to 'majordomo@csi.forth.gr' with body
'subscribe ecdl2-announce <your email address>'




From rem-conf Tue Jan 13 09:36:47 1998 
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	id 0xsA90-0001T1-00; Tue, 13 Jan 1998 09:28:54 -0800
Date: Tue, 13 Jan 1998 18:27:14 +0100 (MET)
From: Christian Isnard - CERN/IT/CS/EN <isnard@dxcoms.cern.ch>
To: rem-conf MBONE List <rem-conf@es.net>, hepnet-l@slacvm.slac.stanford.edu,
        hrc@frcpn11.in2p3.fr, htasc@listbox.cern.ch,
        net-teleconferencing LIST <net-teleconferencing@listbox.cern.ch>
Cc: Christian Isnard <isnard@dxcoms.cern.ch>
Subject: MBONE announcement for CERN LHCC meeting
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          CERN is pleased to announce the MBONE broadcast of the

                 LARGE HADRON COLLIDER COMMITTEE Session
		 ---------------------------------------

                       on Thursday 15 January 1998
                         from the CERN Auditorium


*** Times are UTC+1 ***

Open Session (please note change from preliminary announcement)

        at 09.00 hrs - Auditorium

09.00-09.10     Status of the LHC project (C.Llewellyn Smith)

09.15-10.00     Status of the LHC machine (L.Evans)

10.15-10.35     COFFEE BREAK

10.35-11.35     CMS Muon Project Technical Design Report
                (F.Gasparini, G.Mitselmakher, G.Iaselli)

11.45-12.45     CMS Electromagnetic Calorimeter Project Technical Design
                Report
                (H.Hofer, J.-L.Faure, P.Denes, C.Seez)

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

The sessions will also be recorded using the wrtp application. They can be
downloaded from our vod server:
http://sunmed2.cern.ch/cgi-bin/nph-MBone-sessions/

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






From rem-conf Tue Jan 13 17:47:45 1998 
From rem-conf-request@es.net Tue Jan 13 17:47:44 1998
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	id 0xsHoR-0006iB-00; Tue, 13 Jan 1998 17:40:11 -0800
Date: Wed, 14 Jan 1998 02:44:26 +0100
From: csmr98@dsi.UNIFI.IT
Subject: Conf. Prog.: CSMR98+REF98: Maintenance & Reengineering in Florence
To: rem-conf@es.net
Message-id: <9801140144.AA28483@aguirre.dsi.UNIFI.IT>
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Dear colleague:

Here is the preliminary program of the combined conference
that will be help in Florence, Italy, 8-11 March 1998.
Please accept our apologies if you receive multiple copies of this ca=
ll.
Post forward this message to all your interested colleagues.

Please visit our www page: http://www.dsi.unifi.it/~nesi/csmr98.html
thank you=20

---------------------------------------------------------------------=
-----------

----> 2nd  EUROMICRO CONFERENCE on SOFTWARE MAINTENANCE AND REENGINEE=
RING

----> 6th  REENGINEERING FORUM

      Palazzo degli Affari, FIRENZE, Italy, March 8-11, 1998

      csmr98@aguirre.ing.unifi.it, csmr98@dsi.unifi.it
      http://www.dsi.unifi.it/~nesi/csmr98.html, http://www.reenginee=
r.org/ref98/
      Fax: +39-(0)55-4796363

----> Sponsored by: EUROMICRO, REENGINEERING FORUM, IEEE Computer Soc=
iety
                    IEEE Task Force on Information Technology,=20
                    IEEE Technical Council on  Software Engineering=
=20

      In Cooperation with CESVIT, Universita' di Firenze.

      Patroned by: AICA, AIIA, DeQualitate, Dipartimento di Sistemi e=
 Informatica,
                   Jackson, TABOO, UNINFO.

      Local Conference Organization:
            Paolo Nesi, Alessandro Fantechi, Fabrizio Fioravanti
            Dip. Sistemi e Informatica, Universit=E0 degli Studi di F=
irenze=20
            Via S. Marta, 3, 50139 Firenze, Italy=20

--------------------------> Conference Program <---------------------=
---------

Keynote Speakers:=20
- Continuous Engineering for Industrial Scale Software Systems.=20
  H. Weber, ISST, D
- Evolving Software Practice for Year 2000 and Beyond.=20
  S. Bohner, META Group - Software Engineering Strategies, USA

STEVENS LECTURE ON SOFT. DEVELOPMENT METHODS
- Presentation of the Wayne Stevens Award and the Stevens Lecture on=
=20
  Software Development Methods.=20

Session: Tool Architecture
- Architecture and Functions of a Commercial Software Reengineering W=
orkbench=20
  (H. M. Sneed - Software Engineering Service - D)
- Control Flow Normalization for COBOL/CICS Legacy System=20
  (M. van Brand, A. Sellink, C. Verhoef, - University of Amsterdam - =
NL)

Session: Data Reengineering
- A Generic Approach for Data Reverse Engineering Taking into Account=
 Application=20
  Domain Knoledge (S. A. Ghannouchi, , H. H. B. Ghezola, F. Kamoun, E=
NSI, Tun)
- A strategy for reducing the effort for database schema maintenance=
=20
  (D. Castelli - Istituto di Elaborazione dell'Informazione - I)
- Data Reverse Engineering Methods in Practice=20
  (P. Aiken - Virginia Commonwealth University - USA)
- Database Reengineering for Quality (E. Locuratolo - IEI - CNR; M. L=
offredo,=20
  O. Signore, CNUCE - CNR - I)

Session: Business Information Technology
- An Organizational Framework for Mass-Customized Business Applicatio=
n=20
  (P. Ludwig, T. Kaufmann, H. Liessmann - D)
- Driving IT Decisions from Architectural Principles=20
  (E. Chikofsky - DMR Consulting Group - USA)
- Architectural Reflection: Bridging the Gap Between a Running System=
 and its=20
  Architectural Specification (W. Cazzola, A. Savigni, A. Sosio, F. T=
isato,=20
  University of Milano - I)

Session: Year 2000 Problem
- Variable Classification Technique for Software Maintenance and Appl=
ication to The=20
  Year 2000 Problem (K. Kawabe, A. Matsuo, S. Uehara - Fujitsu Labora=
tories Ltd. JP=20
  A. Ogawa - Fujitsu Ltd. - JP)
- From "Y2K" to Data Warehouses [and other beneficial impacts of the =
millennium=20
  problem] (D. Aebi, C. Schucan, ETH Zurich - SW)

Session: Program Understanding
- On Constructing a Tool to Verify Programs for Processors Built in M=
achines=20
  (T. Ohta, N. Matsumara,  Y. Itoh, Shizuoka Univ. - JP)
- Improving Comprehensibility of Software with Diagramming, Folding, =
and=20
  Coloring (J. H. Cross II - Auburn University - USA)

Session: Reuse and Object Oriented Techniques
- A Dependence-Based Representation for Concurrent Object-Oriented So=
ftware Maintenance
  (J. Zhao - Fukouka Institute of Tech. - J. Cheng, K. Ushijima - Kyu=
shu Univ.-JP)
- OOA Metrics for the Unified Modeling Language=20
  (M. Marchesi-Univ. di Cagliari - I)
- Protection Reconfiguration for Reusable Software (C. D. Jensen, D. =
Hagimont -
  Universite Joseph Fourier - F)

Session: System Assessment=20
- Towards Mature Measurement Program (F. Niessink, H. van Vliet-Univ.=
 Amsterdam NL)
- A Tool for Process and Product Assessment of C++ Applications=20
  (F. Fioravanti, P. Nesi, S. Perlini - University of Florence - I)
- Software Testability Measurement derived from Data Flow Analysis (P=
.-L. Yeh,=20
  J.-C. Lin, Tatung Institute of Technology - Taiwan)

Session: System Documentation
- Documentu: A Flexible Architecture for Documentation Production Bas=
ed on a Reverse
  Engineering Strategy (C. de Oliveira Braga, Arndt von Staa, Julio C=
.S.P. Leite,=20
  PUC-Rio - Brazil)
- Documentation Multi-Targeting Using ASML (C. Owen, F. Makedon, J. F=
ord, S. Rebelsky,=20
  Grinnell College - USA)

Session: Software Architecture
- Assessing Architectural Complexity (R. Kazman - Carnege Mellon Univ=
ersity - USA -=20
  M. Burth - University of Mannheim - D)
- Architecture recovery for Software Evolution (J. C. Duenas, W. L. d=
e Oliveira,=20
  J. A. de la Puente - Universidad Politecnica de Madrid - E)
- The Hot-Spots Technique to Scavenge for Architectural Elements=20
  (H. Gall, B. Bellay, Technical University of Vienna - Austria)

Session: Requirements and Specification Evolution
- Requirements Evolution in the Midst of Environmental Change A Manag=
ed Approach (W.=20
  Lam, M. Loomes - Univ. of Hertfordshire - UK)
- A Method for Assessing Legacy Systems for Evolution (J. Ransom, I. =
Sommerville,=20
  I. Warren - Lancaster University - UK)
- System Specification Reengineering Using the SpecView Tool=20
  (T. G. Kirner, R. C. Gratao - Federal University of Sao Carlos - BR=
)
- A Tool Supporting the Re-Design of Legacy Applications=20
  (K. Cremer - RWTH Aachen- D)

Session: Maintenance Effort
- Modeling Maintenance Effort by means of Dynamic System=20
  (F. Calzolari, G. Antoniol,  P. Tonella - IRST - I)
- Improving Defect Removal Effectiveness for Software Development=
=20
  (Hareton K. N. Leung The Honk Kong Poly. Univ.)
- The Extract-Transform-Rewrite Cycle A Step Towards metaCARE=20
  (B. Kullbach, A. Panse - University of Koblenz - D)

Session: Logic Programming, Telecommunications
- A Metric Suite for Concurrent Logic Programs (J. Zhao - Fukouka Ins=
titute of=20
  Technology - J. Cheng, K. Ushijima - Kyushu Univ. - JP)
- Identifying Fault Prone Modules An Empirical Study in Telecom-munic=
ation System=20
  (S.-B. Hong, K. Kim - Call Proc.Section, ETRI - KR)

Session: Reverse Engineering in Practice
- A Process for Reverse Engineering of AXE 10 Software=20
  (S. Haglund, Ericsson Software Technology AB - Sweden)
- Regenerative Verification and Validation: Moving Forward From Rever=
se Engineering=20
  (T. Roberts - TASC - B. Farley - Motorola - USA)
- A Proposed Reverse Engineering Tech to Redocument High-Level Contro=
l Structures of
  Embedded Systems Software (D. Wilkening -USA)
- Current Parsing Techniques in Software Renovation Considered Harmfu=
l=20
  (M. van den Brand, A. Sellink, C. Verhoef,  NL)

Papers in CSMR98 OPEN FORUM Sections
- DBFW A Simple DataBase FrameWork for the Evaluation and Maintenance=
 of Automated
  Theorem Prover Data (P. Jakobi, A. Wolf, D)
- Reengineering of Distributed Systems Using Formal Methods=20
  (S. Kleuker - University of Oldenburg D)
- Metrics-based Evaluation of Object-Oriented Software development Me=
thods=20
  (R. R. Dumke, E. Foltin - University of Magdeburg - D)
- RENAISSANCE, A Method To Migrate From Legacy To Immortal Software S=
ystem=20
  (M. Battaglia, G. Savoia, J. Favaro - Intecs Sistemi - I)
- Visualization of Differences between Versions of Object-Oriented So=
fware=20
  (J. Seemann, J. W. von Gudenberg - Wurzburg University - D)
- Amber Metrics for the Testing & Maintenance of Object-Oriented Desi=
gns=20
  (J. Doake, I. Duncan - Anglia Polytechnic University - UK)
- Tailoring the Process Model for Maintenance and Reengineering=20
  (S. Stoecklin, D. Wiliams, P. Stoecklin, USA)
- Toward a systematic object-oriented transformation of a Merise anal=
ysis=20
  (I. Borne, A. Romanczuk, F. Stefani, Ecole de Nantes - F)
- Object Evolution by Model Evolution (R. T. Mittermeir, H. Pirker,  =
Dominik=20
  Rauner-Reithmayer - Universit=E4t Klagenfurt - A)
- A sound and Pratical Approach to the Re-Engineering of Time Critica=
l System=20
  (H. Zedan, H. Yang, De Montfort University - UK)
- Reengineering a Computerized Numerical Control Towards Object-Orien=
ted=20
  (F. Butera, B. Fontanella, P.Nesi, M.Perfetti - ELEXA S.r.l. - I)
- Software Artifacts Reuse and Maintenance An organizational Framewor=
k=20
  (C. Toffolon, S. Dakhli - Paris-Dauphine University - F)
- The Task Artifact Cycle: Some Experiences from Reengineering Practi=
ce=20
  (S. Kutscha, sd&m GmbH und Co KG, D)

Session: CSMR98 Industrial Track
- ESSI Project CARERRAS (David Fox,  England)
- ESSI Project PROMASYS (Miguel Fernandez Diaz, Spain)

Session: Results of the Reverse Engineering Demonstration Project=
=20
  Companies, research centers, and universities participating in the =
Reverse=20
  Engineering   Demonstration Project will present their results on t=
he analysis=20
  of a single software system using different tools and techniques.=
=20

General Time Frame
The conference will run from Sunday 8th to Wednesday 11th. Registrati=
on will start on
Sunday 8th at the 16:00 and continues till 18:00. Conference will sta=
rt at the 18:00 of
Sunday 8th with the official opening. The registration desk will be a=
lso open in the
morning from 8:00 to 9:00 of Monday and Tuesday. The conference will =
close its
activities Wednesday 11th at the 18:00.

Social Program
On Tuesday 10th the conference dinner will be organized.=20
Tours could be organized with the Agency by BUS or by feet to visit: =
Boboli Garden,=20
the Dome, Giotto Tower, The Old Bridge, Michelangelo Vista Point, Bel=
vedere,=20
Cappelle Medicee, S. Croce Church, S. Miniato Church, Fiesole Village=
, S. Gimignano=20
Village, The Chianti Vine Area, the Old Market Area, Siena Village, M=
useum with the=20
David of Michelangelo,  (Accademia Gallery), Uffizi Museum, Medieval =
Arms Museum,=20
Archeological Museum, and more than 100 churches along the town. etc.

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

For travel information, registration form, accommodation form,=20
and all other details please consult the www pages of the=20
conference:

http://www.dsi.unifi.it/~nesi/csmr98.html
http://www.reengineer.org/ref98/

---------------------------------------------------------------------=
---------------




From rem-conf Wed Jan 14 11:51:15 1998 
From rem-conf-request@es.net Wed Jan 14 11:51:14 1998
Received: from list by mail1.es.net with local (Exim 1.81 #2)
	id 0xsYa6-0006NM-00; Wed, 14 Jan 1998 11:34:30 -0800
Date: Wed, 14 Jan 98 11:32:00 PST
From: Gim L Deisher <Gim_L_Deisher@ccm.jf.intel.com>
Message-ID: <Wed, 14 Jan 98 11:34:12 PST_1@ccm.jf.intel.com>
To: itu-adv-video@ferrari.iterated.com, rem-conf@es.net,
        internet-drafts@ietf.org
cc: Chad_Zhu@mail.intel.com, Donald_Newell@mail.intel.com,
        Christian_Maciocco@mail.intel.com, Thomas_R_Gardos@ccm.jf.intel.com,
        stewe@cs.tu-berlin.de, jo@cs.tu-berlin.de, garys@python.pictel.com,
        cabo@tzi.org, lscline@ideal.jf.intel.com
Subject: New RTP Payload Format for H.263+
X-Mailing-List: <rem-conf@es.net> 
X-Loop: rem-conf@es.net
Precedence: list


Text item: 

Enclosed is a new revision of the H.263+ RTP payload 
specification which is to replace the 11/97 IETF draft, 
draft-ietf-avt-rtp-h263-video-00.txt.  Since the version 
presented in December 97 IETF meeting, only minor 
corrections and clarifications have been made.
     
Comments welcome as usual.  Thank you.

Text item: h263pplh.txt 1/14/98 8:41A

Internet Engineering Task Force                 Audio-Video Transport WG
INTERNET-DRAFT                                 C. Bormann / Univ. Bremen
                                                        L. Cline / Intel
                                                      G. Deisher / Intel
                                                       T. Gardos / Intel
                                                     C. Maciocco / Intel
                                                       D. Newell / Intel
                                                   J. Ott / Univ. Bremen
                                                G. Sullivan / PictureTel
                                                   S. Wenger / TU Berlin
                                                          C. Zhu / Intel

                                            Date Generated: 14 Jan. 1998

               RTP Payload Format for the 1998 Version of
                    ITU-T Rec. H.263 Video (H.263+)


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 made obsolete 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.


1. Introduction

This document specifies an RTP payload header format applicable to the 
transmission of video streams generated based on the 1998 version of
ITU-T Recommendation H.263 [4].  Because the 1998 version of H.263 is a 
superset of the 1996 syntax, this format can also be used with the 1996 
version of H.263.

The 1998 version of ITU-T Recommendation H.263 added numerous coding 
options to improve codec performance over the 1996 version.  The 1998 
version is referred to as H.263+ in this document.  Among the new 
options, the ones with the biggest impact on the RTP payload 
specification and the error resilience of the video content are the 
slice structured mode, the independent segment decoding mode (ISD), the 
reference picture selection mode, and the scalability mode.  This 
section summarizes the impact of these new coding options on 
packetization.  Refer to [4] for more information on coding options.

The slice structured mode was added to H.263+ for three purposes: to 
provide enhanced error resilience capability, to make the bitstream more 
amenable to use with an underlying packet transport such as RTP, and to 
minimize video delay.  The slice structured mode supports fragmentation 
at macroblock boundaries.

With the independent segment decoding option, a video picture frame is 
broken into segments and encoded in such a way that each segment is 
independently decodable.  Utilizing ISD in a lossy network environment 
helps to prevent the propagation of errors from one segment of the 
picture to others.

The reference picture selection mode allows the use of an older 
reference picture rather than the one immediately preceding the current 
picture.  Usually, the last transmitted frame is implicitly used as the 
reference picture for inter-frame prediction.  If the reference picture 
selection mode is used, the data stream carries information on what 
reference frame should be used, indicated by the temporal reference as 
an ID for that reference frame.  The reference picture selection mode 
can be used with or without a back channel, which provides information 
to the encoder about the internal status of the decoder.  However, no 
special provision is made herein for carrying back channel information.

H.263+ also includes bitstream scalability as an optional coding mode.
Three kinds of scalability are defined: temporal, signal-to-noise ratio
(SNR), and spatial scalability.  Temporal scalability is achieved via 
the disposable nature of bi-directionally predicted frames, or B-frames.  
SNR scalability permits refinement of encoded video frames, thereby 
improving the quality (or SNR).  Spatial scalability is similar to SNR 
scalability except the refinement layer is twice the size of the base 
layer in the horizontal dimension, vertical dimension, or both.


2. 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, with the only exception being that when 
the payload of a packet begins with a Picture, GOB, Slice, EOS, or EOSBS 
start code, the first two (all-zero) bytes of the start code are removed 
and replaced by setting an indicator bit in the payload header.

For H.263+ bitstreams coded with temporal, spatial, or SNR scalability, 
each layer may be transported to a different network address.  More 
specifically, each layer may use a unique IP address and port number 
combination.  The temporal relations between layers shall be expressed 
using the RTP timestamp so that they can be synchronized at the 
receiving ends in multicast or unicast applications.

The H.263+ video stream will be carried as payload data within RTP 
packets.  A new H.263+ payload header is defined in section 4.  This 
section defines the usage of the RTP fixed header and H.263+ video 
packet structure.


2.1 RTP Header Usage

Each RTP packet starts with a fixed RTP header.  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 header is set to 1 when 
the current packet carries the end of current frame, and is 0 otherwise.

Payload Type (PT): The Payload Type shall specify the H.263+ video 
payload format.

Timestamp: The RTP Timestamp encodes the sampling instance of the first 
video frame data contained in the RTP data packet.  The RTP timestamp 
shall be the same on successive packets if a video frame occupies more 
than one packet.  In a multilayer scenario, all pictures corresponding 
to the same temporal reference should use the same timestamp.  If 
temporal scalability is used (if B-frames are present), the timestamp 
may not be monotonically increasing in the RTP stream.  If B-frames are 
transmitted on a separate layer and address, they must be synchronized 
properly with the reference frames.  Refer to the 1998 ITU-T 
Recommendation H.263 [4] for information on required transmission order 
to a decoder.  For an H.263+ video stream, the RTP timestamp is based on 
a 90 kHz clock, the same as that of the RTP payload for H.261 stream 
[5].  Since both the H.263+ data and the RTP header contain time 
information, it is required that those timing information run 
synchronously.  That is, both the RTP timestamp and the temporal 
reference (TR in the picture header of H.263) should carry the same 
relative timing information.  If necessary, mathematical rounding should 
be applied to the information of the H.263+ data stream to generate the 
RTP timestamp (this is especially true for the standard picture clock 
frequency of 30000/1001 Hz, and may also be true if custom picture clock 
frequencies are to be used; see [4] for details).


2.2 Video Packet Structure

A section of an H.263+ compressed bitstream is carried as a payload 
within each RTP packet.  For each RTP packet, the RTP header is followed 
by an H.263+ payload header, which is followed by a number of bytes of a 
standard H.263+ compressed bitstream.  The size of the H.263+ payload 
header is variable depending on the payload involved as detailed in the 
section 4.  The layout of the 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+ Compressed Data Stream                            ...
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Any H.263+ start codes can be byte aligned by an encoder by using the 
stuffing mechanisms of H.263+.  As specified in H.263+, picture, slice, 
and EOSBS start codes shall always be byte aligned, and GOB and EOS 
start codes may be byte aligned.  For packetization purposes, GOB start 
codes should be byte aligned, although this is not absolutely required 
herein since it is not required in H.263+.

All H.263+ start codes (Picture, GOB, Slice, EOS, and EOSBS) begin with 
16 zero-valued bits.  If a start code is byte aligned and it occurs at 
the beginning of a packet, these two bytes shall be removed from the 
H.263+ compressed data stream in the packetization process and shall 
instead be represented by setting a bit (the P bit) in the payload 
header.


3. Design Considerations

The goals of this payload format are to specify an efficient way of 
encapsulating an H.263+ standard compliant bitstream and to enhance the 
resiliency towards packet losses.  Due to the large number of different 
possible coding schemes in H.263+, a copy of the picture header with 
configuration information is inserted into the payload header when 
appropriate.  The use of that copy of the picture header along with the 
payload data can allow decoding of a received packet even in such cases 
in which another packet containing the original picture header becomes 
lost.

There are a few assumptions and constraints associated with this H.263+ 
payload header design.  The purpose of this section is to point out 
various design issues and also to discuss several coding options 
provided by H.263+ that may impact the performance of network-based 
H.263+ video.


o The optional slice structured mode described in annex K of H.263+ [4]
  enables more flexibility for packetization.  Similar to a picture
  segment that begins with a GOB header, the motion vector predictors in
  a slice are restricted to reside within its boundaries.  However,
  slices provide much greater freedom in the selection of the size and
  shape of the area which is represented as a distinct decodable region.  
  In particular, slices can have a size which is dynamically selected to 
  allow the data for each slice to fit into a chosen packet size.  
  Slices can also be chosen to have a rectangular shape which is
  conducive for minimizing the impact of errors and packet losses on 
  motion compensated prediction.  For these reasons, the use of the 
  slice structured mode is strongly recommended for any applications 
  used in environments where significant packet loss occurs.

o In non-rectangular slice structured mode, only complete slices should
  be included in a packet.  In other words, slices should not be
  fragmented across packet boundaries.  The only reasonable need for a 
  slice to be fragmented across packet boundaries is when the encoder 
  which generated the H.263+ data stream could not be influenced by an 
  awareness of the packetization process (such as when sending H.263+ 
  data through a network other than the one to which the encoder is 
  attached, as in network gateway implementations).  Optimally, each 
  packet will contain only one slice.

o The independent segment decoding (ISD) described in annex R of [4]
  prevents any data dependency across slice or GOB boundaries in the
  reference picture.  It can be utilized to further improve resiliency
  in high loss conditions.

o If ISD is used in conjunction with the slice structure, the 
  rectangular slice submode shall be enabled and the dimensions and 
  quantity of the slices present in a frame shall remain the same 
  between each two intra-coded frames (I-frames), as required in H.263+.  
  The individual ISD segments may also be entirely intra coded from time 
  to time to realize quick error recovery without adding the latency 
  time associated with sending complete INTRA-pictures.

o When the slice structure is not applied, the insertion of a 
  (preferably byte-aligned) GOB header can be used to provide resync 
  boundaries in the bitstream, as the presence of a GOB header 
  eliminates the dependency of motion vector prediction across GOB 
  boundaries.  These resync boundaries provide natural locations for 
  packet payload boundaries.

o H.263+ allows picture headers to be sent in an abbreviated form in 
  order to prevent repetition of overhead information that does not 
  change from picture to picture.  For resiliency, sending a complete 
  picture header for every frame is often advisable.  This means, that 
  especially in cases with high packet loss probability in which picture 
  header contents are not expected to be highly predictable, the sender 
  may always set the subfield UFEP in PLUSPTYPE to '001' in the H.263+ 
  video bitstream.

o In a multi-layer scenario, each layer may be transmitted to a 
  different network address.  The configuration of each layer such as 
  the enhancement layer number (ELNUM), reference layer number (RLNUM), 
  and scalability type should be determined at the start of the session 
  and should not change during the course of the session.

o All start codes can be byte aligned, and picture, slice, and EOSBS 
  start codes are always byte aligned.  The boundaries of these
  syntactical elements provide ideal locations for placing packet 
  boundaries.

o We assume that a maximum Picture Header size of 504 bits is
  sufficient.  The syntax of H.263+ does not explicitly prohibit larger 
  picture header sizes, but the use of such extremely large picture 
  headers is not expected.


4. 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.  The payload header is of variable length.  A 16 bit field 
of the basic payload header may be followed by an 8 bit field for Video 
Redundancy Coding information, and/or by a variable length picture 
header as indicated by PLEN. These optional fields appear in the order 
given above when present.

If a picture header is included in the payload header, the length of the 
picture header in number of bytes is specified by PLEN.  The minimum 
length of the payload header is 16 bits, corresponding to PLEN equal to 
0 and no VRC information present.

The remainder of this section defines the various components of the RTP 
payload header.  Section five defines the various packet types that are 
used to carry different types of H.263+ coded data, and section six 
summarizes how to distinguish between the various packet types.


4.1 General H.263+ payload header

The H.263+ payload header is structured as follows:

 0                   1
 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|  RR     |P|V|  PLEN     |PEBIT|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

RR: 5 bits
  Reserved bits.  Shall be zero.

P: 1 bit
  Indicates the picture start or a picture segment (GOB/Slice) start or 
  a video sequence end (EOS or EOSBS).  Two bytes of zero bits then have 
  to be prefixed to the payload of such a packet to compose a complete
  picture/GOB/slice/EOS/EOSBS start code.  This bit allows the omission 
  of the two first bytes of the start codes, thus improving the 
  compression ratio.

V: 1 bit
  Indicates the presence of an 8 bit field containing information for 
  Video Redundancy Coding (VRC), which follows immediately after the 
  initial 16 bits of the payload header if present.  For syntax and 
  semantics of that 8 bit VRC field see section 4.2.

PLEN: 6 bits
  Picture header length in number of bytes.  If no additional picture 
  header is attached, PLEN is 0.  If PLEN>0, the additional picture 
  header is attached immediately following the rest of the payload 
  header.

PEBIT: 3 bits
  Indicates the number of bits that shall be ignored in the last byte of 
  the picture header.  If PLEN is zero, then PEBIT shall also be zero.


4.2 Video Redundancy Coding Header Extension

Video Redundancy Coding (VRC) is an optional mechanism intended to 
improve error resilience over packet networks.  Implementing VRC in 
H.263+ will require the Reference Picture Selection option described in 
Annex N.  By having multiple "threads" of independently inter-frame 
predicted pictures, damage of individual frame will cause distortions 
only within its own thread but leave the other threads unaffected.  From 
time to time, all threads converge to a so-called sync frame (an INTRA 
picture or a non-INTRA picture which is redundantly represented within 
multiple threads); from this sync frame, the independent threads are 
started again.  For a more complete description of VRC see [7].

While a VRC data stream is - like all H.263+ data - totally self-
contained, it may be useful for the transport hierarchy implementation 
to have knowledge about the current damage status of each thread.  On 
the Internet, this status can easily be determined by observing the 
marker bit, the sequence number of the RTP header, and the thread-id and 
a circling "packet per thread" number.  The latter two numbers are coded 
in the VRC header extension.

The format of the VRC header extension is as follows:

 0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+
| TID | Trun  |S|
+-+-+-+-+-+-+-+-+

TID: 3 bits
  Thread ID.  Up to 7 threads are allowed. Each frame of H.263+ VRC data 
  will use as reference information only sync frames or frames within 
  the same thread.  By convention, thread 0 is expected to be the 
  "canonical" thread, which is the thread from which the sync frame 
  should ideally be used.  In the case of corruption or loss of the 
  thread 0 representation, a representation of the sync frame with a 
  higher thread number can be used by the decoder.  Lower thread numbers 
  are expected to contain equal or better representations of the sync 
  frames than higher thread numbers in the absence of data corruption or 
  loss.  See [7] for details.

Trun: 4 bits
  Monotonically increasing (modulo 16) 4 bit number counting the packet
  number within each thread.

S: 1 bit
  A bit that indicates that the packet content is for a sync frame.  An
  encoder using VRC may send several representations of the same "sync"
  picture, in order to ensure that regardless of which thread of 
  pictures is corrupted by errors or packet losses, the reception of at 
  least one representation of a particular picture is ensured (within at 
  least one thread).  The sync picture can then be used for the 
  prediction of any thread.  If packet losses have not occurred, then 
  the sync frame contents of thread 0 can be used and those of other 
  threads can be discarded (and similarly for other threads).  Thread 0 
  is considered the "canonical" thread, the use of which is preferable 
  to all others.  The contents of packets having lower thread numbers 
  shall be considered as generally preferred over those with higher 
  thread numbers.


5. Packetization schemes

5.1 Picture Segment Packets and Sequence Ending Packets (P=1)

A picture segment packet is defined as a packet that starts at the 
location of a Picture, GOB, or slice start code in the H.263+ data 
stream.  This corresponds to the definition of the start of a video 
picture segment as defined in H.263+.  For such packets, P=1 always.

An extra picture header can sometimes be attached in the payload header 
of such packets.  Whenever an extra picture header is attached as 
signified by PLEN>0, only the last six bits of its picture start code, 
'100000', are included in the payload header.  A complete H.263+ picture 
header with byte aligned picture start code can be conveniently 
assembled on the receiving end by prepending the sixteen leading '0' 
bits.

When PLEN>0, the end bit position corresponding to the last byte of the 
picture header data is indicated by PEBIT.  The actual bitstream data 
shall begin on an 8-bit byte boundary following the payload header.

A sequence ending packet is defined as a packet that starts at the 
location of an EOS or EOSBS code in the H.263+ data stream.  This 
delineates the end of a sequence of H.263+ video data (more H.263+ video 
data may still follow later, however, as specified in ITU-T 
Recommendation H.263).  For such packets, P=1 and PLEN=0 always.

The optional header extension for VRC may or may not be present as 
indicated by the V bit flag.


5.1.1 Packets that begin with a Picture Start Code

Any packet that contains the whole or the start of a coded picture shall 
start at the location of the picture start code (PSC), and should 
normally be encapsulated with no extra copy of the picture header. In 
other words, normally PLEN=0 in such a case.   However, if the coded 
picture contains an incomplete picture header (UFEP = "000"), then a 
representation of the complete (UFEP = "001") picture header may be 
attached during packetization in order to provide greater error 
resilience.  Thus, for packets that start at the location of a picture 
start code, PLEN shall be zero unless both of the following conditions 
apply:
1) The picture header in the H.263+ bitstream payload is incomplete
   (PLUSPTYPE present and UFEP="000"), and
2) The additional picture header which is attached is not incomplete
   (UFEP="001").

A packet which begins at the location of a Picture, GOB, slice, EOS, or 
EOSBS start code shall omit the first two (all zero) bytes from the 
H.263+ bitstream, and signify their presence by setting P=1 in the 
payload header.

Here is an example of encapsulating the first packet in a frame (without 
an attached redundant complete picture header):

 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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|  RR     |1|V|0|0|0|0|0|0|0|0|0|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-------------------------------+
| bitstream data without the first two 0 bytes of the PSC       |
+---------------------------------------------------------------+


5.1.2 Packets that begin with GBSC or SSC

For a packet that begins at the location of a GOB or slice start code, 
PLEN may be zero or may be nonzero, depending on whether a redundant 
picture header is attached to the packet.  In environments with very low 
packet loss rates, or when picture header contents are very seldom 
likely to change (except as can be detected from the GFID syntax of 
H.263+), a redundant copy of the picture header is not required.  
However, in less ideal circumstances a redundant picture header should 
be attached for enhanced error resilience, and its presence is indicated 
by PLEN>0.

Assuming a PLEN of 9, below is an example of a packet that begins with a
GBSC or a SSC:

 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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|  RR     |1|V|0 0 1 0 0 1|PEBIT|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1 0 0 0 0 0| picture header starting with TR, PTYPE, ...       |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ...                                                           |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ...           | bitstream data begins with GBSC/SCC ...       .
+-+-+-+-+-+-+-+-+-----------------------------------------------+

Notice that only the last six bits of the picture start code, '100000', 
are included in the payload header.  A complete H.263+ picture header 
with byte aligned picture start code can be conveniently assembled if 
needed on the receiving end by prepending the sixteen leading '0' bits.


5.1.3 Packets that Begin with an EOS or EOSBS Code

For a packet that begins with an EOS or EOSBS code, PLEN shall be zero, 
and no Picture, GOB, or Slice start codes shall be included within the 
same packet.  As with other packets beginning with start codes, the two 
all-zero bytes that begin the EOS or EOSBS code at the beginning of the 
packet shall be omitted, and their presence shall be indicated by 
setting the P bit to 1 in the payload header.

System designers should be aware that some decoders may interpret the 
loss of a packet containing only EOS or EOSBS information as the loss of 
essential video data and may thus respond by not displaying some 
subsequent video information.  Since EOS and EOSBS codes do not actually 
affect the decoding of video pictures, they are somewhat unnecessary to 
send at all.  Because of the danger of misinterpretation of the loss of 
such a packet, encoders are generally to be discouraged from sending EOS 
and EOSBS.

Below is an example of a packet containing an EOS code:

 0                   1
 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|  RR     |1|V|0|0|0|0|0|0|0|0|0|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1|1|1|1|1|1|0|0|
+-+-+-+-+-+-+-+-+


5.2 Encapsulating Follow-On Packet (P=0)

A Follow-on packet contains a number of bytes of coded H.263+ data which 
does not start at a synchronization point.  That is, a Follow-On packet 
does not start with a Picture, GOB, Slice, EOS, or EOSBS header, and it 
may or may not start at a macroblock boundary.  Since Follow-on packets 
do not start at synchronization points, the data at the beginning of a 
follow-on packet is not independently decodable.  For such packets, P=0 
always.  If the preceding packet of a Follow-on packet got lost, the 
receiver may discard that Follow-on packet as well as all other 
following Follow-on packets.  Better behavior, of course, would be for 
the receiver to scan the interior of the packet payload content to 
determine whether any start codes are found in the interior of the 
packet which can be used as resync points.  The use of an attached copy 
of a picture header for a follow-on packet is useful only if the 
interior of the packet or some subsequent follow-on packet contains a 
resync code such as a GOB or slice start code.  PLEN>0 is allowed, since 
it may allow resync in the interior of the packet.  The decoder may also 
be resynchronized at the next segment or picture packet.

Here is an example of a follow-on packet (with PLEN=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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|  RR     |0|V|0|0|0|0|0|0|0|0|0|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-------------------------------+
| bitstream data                                                |
+---------------------------------------------------------------+


6. Use of this payload specification

There is no syntactical difference between a picture segment packet and 
a Follow-on packet, other than the indication P=1 for picture segment or 
sequence ending packets and P=0 for Follow-on packets.  See the 
following for a summary of the entire packet types and ways to 
distinguish between them.

For a more detailed discussion on how to use the payload specification, 
the reader should refer to [8].

It is possible to distinguish between the different packet types by 
checking the P bit and the first 6 bits of the payload along with the 
header information.  The following table shows the packet type for 
permutations of this information (see also the picture/GOB/Slice header 
descriptions in H.263+ for details):

--------------+--------------+----------------------+-------------------
 First 6 bits | P-Bit | PLEN |  Packet              |  Remarks
 of Payload   |(payload hdr.)|                      |
--------------+--------------+----------------------+-------------------
 100000       |   1   |  0   |  Picture             |  Typical Picture
 100000       |   1   | > 0  |  Picture             |  Note UFEP
 1xxxxx       |   1   |  0   |  GOB/Slice/EOS/EOSBS |  See possible GNs
 1xxxxx       |   1   | > 0  |  GOB/Slice           |  See possible GNs
 Xxxxxx       |   0   |  0   |  Follow-on           |
 Xxxxxx       |   0   | > 0  |  Follow-on           |  Interior Resync
--------------+--------------+----------------------+-------------------

See [4] for details regarding the possible values of the six bits (a "1" 
bit followed by a five bit GN field explicit or emulated) of GOB, Slice, 
EOS, and EOSBS codes.

As defined in this specification, every start of a coded frame (as 
indicated by the presence of a PSC) has to be encapsulated as a picture 
segment packet.  If the whole coded picture fits into one packet of 
reasonable size (which is dependent on the connection characteristics), 
this is the only type of packet that needs to be observed.  Due to the 
high compression ratio achieved by H.263+ it is often possible to use 
this mechanism, especially for small spatial picture formats such as 
QCIF and typical Internet packet sizes around 1500 bytes.

If the complete coded frame does not fit into a single packet, two 
different ways for the packetization may be chosen.  In case of very low 
or zero packet loss probability, one or more Follow-on packets may be 
used for coding the rest of the picture.  Doing so leads to minimal 
coding and packetization overhead as well as to an optimal use of the 
maximal packet size, but does not provide any added error resilience.

The alternative is to break the picture into reasonably small partitions 
- called Segments - (by using the Slice or GOB mechanism), that do offer 
synchronization points.  By doing so and using the Picture Segment 
payload with PLEN>0, decoding of the transmitted packets is possible 
even in such cases in which the Picture packet containing the picture 
header was lost (provided any necessary reference picture is available). 
Picture Segment packets can also be used in conjunction with Follow-on 
packets for large segment sizes.


7. Security Considerations

RTP packets using the payload format defined in this specification are 
subject to the security considerations discussed in the RTP 
specification [1], and any appropriate RTP profile (for example [3]).
This implies that confidentiality of the media streams is achieved by 
encryption.  Because the data compression used with this payload format 
is applied end-to-end, encryption may be performed after compression so 
there is no conflict between the two operations.

A potential denial-of-service threat exists for data encodings using 
compression techniques that have non-uniform receiver-end computational 
load.  The attacker can inject pathological datagrams into the stream 
which are complex to decode and cause the receiver to be overloaded.
However, this encoding does not exhibit any significant non-uniformity.

As with any IP-based protocol, in some circumstances a receiver may be 
overloaded simply by the receipt of too many packets, either desired or 
undesired.  Network-layer authentication may be used to discard packets 
>from undesired sources, but the processing cost of the authentication 
itself may be too high.  In a multicast environment, pruning of specific 
sources may be implemented in future versions of IGMP [5] and in 
multicast routing protocols to allow a receiver to select which sources 
are allowed to reach it.

A security review of this payload format found no additional 
considerations beyond those in the RTP specification.


8. References

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

[2] "Video Codec for Audiovisual Services at px64 kbits/s", ITU-T
    Recommendation H.261, 1993.

[3] "RTP Profile for Audio and Video Conference with Minimal Control",
    RFC 1890.

[4] "Video Coding for Low Bitrate Communication", Draft ITU-T
    Recommendation H.263, Draft 20, September 1997.

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

[6] C. Zhu, "RTP Payload Format for H.263 Video Streams", RFC 2190.

[7] S. Wenger, "Video Redundancy Coding in H.263+", Proc. AVSPN97, 
    Aberdeen, U.K..

[8] S. Wenger, G. Knorr, J. Ott: "Error resilience support in H.263 
    V.2", submitted for publication to IEEE T-CSVT, 1997.




From rem-conf Wed Jan 14 12:29:15 1998 
From rem-conf-request@es.net Wed Jan 14 12:29:14 1998
Received: from list by mail1.es.net with local (Exim 1.81 #2)
	id 0xsZK1-00076Y-00; Wed, 14 Jan 1998 12:21:57 -0800
To: Toerless Eckert <Toerless.Eckert@informatik.uni-erlangen.de>
cc: rem-conf@es.net, mbone@isi.edu
Subject: Re: sdr announcement questions 
In-reply-to: Your message of "Mon, 12 Jan 98 02:04:56 PST."
             <199801121004.LAA21225@faui45r.informatik.uni-erlangen.de> 
Date: Wed, 14 Jan 1998 12:21:34 PST
Sender: Bill Fenner <fenner@parc.xerox.com>
From: Bill Fenner <fenner@parc.xerox.com>
Message-Id: <98Jan14.122142pst.177480@crevenia.parc.xerox.com>
X-Mailing-List: <rem-conf@es.net> 
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Toerless Eckert <Toerless.Eckert@Informatik.Uni-Erlangen.de> wrote:
>	     There doesn't seem to be something like
>	     a sequence number for announcements, or am i mistaken there ?

The third field in the "o=" line is the version number; if you keep
the other fields the same then the new announcement is supposed to
replace the old one.

>Replacement question: where's the documentation for the exact format and
>semantics of the .sdr announcement files ? ;-))

sdr stores announcements with one or two header lines (n= and/or k= )
followed by an SDP (draft-ietf-mmusic-sdp-05.{txt,ps}) packet.

  Bill



From rem-conf Wed Jan 14 18:38:51 1998 
From rem-conf-request@es.net Wed Jan 14 18:38:51 1998
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	id 0xsf7e-0002UO-00; Wed, 14 Jan 1998 18:33:34 -0800
Sender: "M. Strata Rose" <strata@remarque.org>
Date: Wed, 14 Jan 98 12:33:28 PST
From: strata@virtual.net
Reply-To: strata@virtual.net
To: rem-conf@es.net
Subject: January BayLISA: SSH : Thurs, 1/15/1998
Message-ID: <CMM.0.90.4.884810008.strata@remarque.org>
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-----------------------------------------------------------------------------

January BayLISA:  SSH -- Introduction through Implementation
Mbone broadcast, 01/15/1998 19:30 PST - approx 21:30 PST

BayLISA will broadcast the following talk and take live questions from
the mbone audience during the q&a period.  If you are within the Bay
Area, please feel free to drop by in person!

We are working on generating session information further in advance of
our meetings, so that more people may tune in.  We apologize for the
last-minute notice this month.

Strata Rose,
BayLISA Board Member

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

BayLISA General Meeting
Thursday, 15 January, 1998

Cisco Building J1, LONDON Room   <*** Note temporary room change, 
				      building remains the same.
				      London Room is right next to usual room.
				      Traditional ASCII map at end of posting.
-----------------------------------------------------------------------------
Thursday, 15 January, 1998

SSH -- Introduction through Implementation,
Steve Acheson

Starting at 19:30, and continuing until about 21:30

SSH, the Secure Shell program, has matured into a popular and powerful 
tool for secure system access and securely performing remote functions, 
such as rdist and rsync. 

The talk will focus on: 

     SSH features and authentication methods 
     Overview of the different versions (both public and commercial) 
     How to secure X11 connections using SSH 
     How to do secure port forwarding with SSH 
     Softare available for use with SSH (eg, rdist, rsync) 


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

General Meetings:

Unless otherwise noted, monthly meetings are held on the third Thursday 
of each month, at Cisco Systems in San Jose. Directions are available at 
this time, and please let us know if you have any questions.  Light
refreshments are generally served.

BayLISA meetings are also broadcast on the MBONE using volunteer assistance 
and loaned hardware. MSRI has an MBONE broadcast schedule page as well as 
information about MBONE tools. 

Please see the BayLISA web site for more information about future meetings,
BayLISA membership, and other neat stuff.  http://www.baylisa.org/

Upcoming meetings:

February 19th:  All the News That Fits...
		Large Scale Netnews (Panel)
			Nick Christensen, Earthlink
			Danheil Baker, Supernews
			Landon Knoll, SGI

-----------------------------------------------------------------------------
[http://www.baylisa.org/events/cisco.html]

Getting To The Meeting at Cisco Systems

     Cisco Systems -- Building J Gateway Conference Facility
     236 W. Tasman Drive
     San Jose, CA 95134
     (At Highway 237 and N. First)

Take any freeway to highway 237. If your on the Milpitas (east) side of 
the bay, head west towards Mountain View, if your on the Mountain View 
(west) side of the bay, head east towards Milpitas. 

Take highway 237 to N. First Street. Head south on N. First until you get 
to Tasman Drive. Turn right, following the light rail tracks. Go through 
the second signal, and Building J will be the middle of three buildings on 
your right before Vista Montana. You can enter the conference facility 
through the doors on the north side of the building. 

Look! An ongoing tradition, a terrible ASCII map! 

                Cisco Main Campus

=========================================================
                          \                   Highway 237
                           \
                            \
               Vista Montana/\
                           /  \ 
  ___________________     /    \
    Tasman Drive     \   /      \
                      \ /        \
                    H  /          \
                        \ I        \
                  G      \   J      \
                     F    \          \  N. First Street
                           \ K        \
                     E      \---       \
                             \  L       \
                     D        \          \
                         C     \          \
                                \          \
                            B    \----------\--------- Tasman Drive
                                 /           \
                         A      /             \
                               Rio Robles
                                Drive


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

===========================================================================
Strata Rose [KF6NBZ]				 strata@virtual.net
VirtualNet Consulting				 http://www.virtual.net/
Internet: Installations, Security, Large Site Scaling Issues, E-Commerce
===========================================================================



From rem-conf Thu Jan 15 02:46:25 1998 
From rem-conf-request@es.net Thu Jan 15 02:46:24 1998
Received: from list by mail1.es.net with local (Exim 1.81 #2)
	id 0xsmdv-00066E-00; Thu, 15 Jan 1998 02:35:23 -0800
From: Edgar Hellfritsch <edgar.hellfritsch@rrze.uni-erlangen.de>
Subject: SDR private announcements
To: rem-conf@es.net
Date: Thu, 15 Jan 1998 11:35:17 +0100 (MET)
X-Mailer: ELM [version 2.4 PL23]
MIME-Version: 1.0
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Hi

In order to keep a public sdp announcement posted forever, you just set the
date of expiry to zero in the corresponding cache file. fine. (ok, You're
supposed not to do this, but in our case it's a necessity.)

How can I do this with an encrypted cache file? Decode/edit/re-encode the
cache file? Or do I have to keep re-announcing our regular private sessions
every month?

Any suggestions?

Edgar



From rem-conf Thu Jan 15 03:57:55 1998 
From rem-conf-request@es.net Thu Jan 15 03:57:54 1998
Received: from list by mail1.es.net with local (Exim 1.81 #2)
	id 0xsnrZ-0006vf-00; Thu, 15 Jan 1998 03:53:33 -0800
To: Edgar Hellfritsch <edgar.hellfritsch@rrze.uni-erlangen.de>
cc: rem-conf@es.net
Subject: Re: SDR private announcements
In-reply-to: Your message of "Thu, 15 Jan 1998 11:35:17 +0100." <34bde6661085002@max4.rrze.uni-erlangen.de>
Date: Thu, 15 Jan 1998 11:53:04 +0000
Message-ID: <530.884865184@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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--> Edgar Hellfritsch writes:
>In order to keep a public sdp announcement posted forever, you just set the
>date of expiry to zero in the corresponding cache file. fine. (ok, You're
>supposed not to do this, but in our case it's a necessity.)
>
>How can I do this with an encrypted cache file? Decode/edit/re-encode the
>cache file? Or do I have to keep re-announcing our regular private sessions
>every month?

...or change the SDR UI to allow for non-expiry when creating the session,
and SDR will encrypt it for you.

Colin



From rem-conf Thu Jan 15 08:52:09 1998 
From rem-conf-request@es.net Thu Jan 15 08:52:09 1998
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	id 0xssOD-0002DV-00; Thu, 15 Jan 1998 08:43:33 -0800
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Mime-Version: 1.0
Content-Type: Multipart/Mixed; Boundary="NextPart"
To: IETF-Announce: ;
Cc: rem-conf@es.net
From: Internet-Drafts@ns.ietf.org
Reply-to: Internet-Drafts@ns.ietf.org
Subject: I-D ACTION:draft-ietf-avt-rtp-h263-video-01.txt
Date: Thu, 15 Jan 1998 11:43:16 -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		: RTP Payload Format for the 1998 Version of ITU-T Rec. H.263 Video (H.263+)
	Author(s)	: C. Maciocco, J. Ott, C. Bormann, C. Zhu, L. Cline, D. Newell, G. Sullivan, G. Deisher, S. Wenger
	Filename	: draft-ietf-avt-rtp-h263-video-01.txt
	Pages		: 10
	Date		: 14-Jan-98
	
This document specifies an RTP payload header format applicable to the
transmission of video streams generated based on the 1998 version of
ITU-T Recommendation H.263 [4].  Because the 1998 version of H.263 is a
superset of the 1996 syntax, this format can also be used with the 1996
version of H.263.
 
The 1998 version of ITU-T Recommendation H.263 added numerous coding
options to improve codec performance over the 1996 version.  The 1998
version is referred to as H.263+ in this document.  Among the new
options, the ones with the biggest impact on the RTP payload
specification and the error resilience of the video content are the
slice structured mode, the independent segment decoding mode (ISD), the
reference picture selection mode, and the scalability mode.  This
section summarizes the impact of these new coding options on
packetization.  Refer to [4] for more information on coding options.
 
The slice structured mode was added to H.263+ for three purposes: to
provide enhanced error resilience capability, to make the bitstream more
amenable to use with an underlying packet transport such as RTP, and to
minimize video delay.  The slice structured mode supports fragmentation
at macroblock boundaries.

With the independent segment decoding option, a video picture frame is
broken into segments and encoded in such a way that each segment is
independently decodable.  Utilizing ISD in a lossy network environment
helps to prevent the propagation of errors from one segment of the
picture to others.
 
The reference picture selection mode allows the use of an older
reference picture rather than the one immediately preceding the current
picture.  Usually, the last transmitted frame is implicitly used as the
reference picture for inter-frame prediction.  If the reference picture
selection mode is used, the data stream carries information on what
reference frame should be used, indicated by the temporal reference as
an ID for that reference fra

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-h263-video-01.txt".
A URL for the Internet-Draft is:
ftp://ds.internic.net/internet-drafts/draft-ietf-avt-rtp-h263-video-01.txt

Internet-Drafts directories are located at:

	Africa:	ftp.is.co.za
	
	Europe: ftp.nordu.net
		ftp.nis.garr.it
			
	Pacific Rim: munnari.oz.au
	
	US East Coast: ds.internic.net
	
	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-h263-video-01.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:	<19980115112135.I-D@ietf.org>

ENCODING mime
FILE /internet-drafts/draft-ietf-avt-rtp-h263-video-01.txt

--OtherAccess
Content-Type: Message/External-body;
	name="draft-ietf-avt-rtp-h263-video-01.txt";
	site="ds.internic.net";
	access-type="anon-ftp";
	directory="internet-drafts"

Content-Type: text/plain
Content-ID:	<19980115112135.I-D@ietf.org>

--OtherAccess--

--NextPart--





From rem-conf Thu Jan 15 10:15:03 1998 
From rem-conf-request@es.net Thu Jan 15 10:15:03 1998
Received: from list by mail1.es.net with local (Exim 1.81 #2)
	id 0xstih-0003JM-00; Thu, 15 Jan 1998 10:08:47 -0800
Message-Id: <3.0.5.32.19980115100825.00910bf0@ideal.jf.intel.com>
X-Sender: bstrahm@ideal.jf.intel.com
X-Mailer: QUALCOMM Windows Eudora Pro Version 3.0.5 (32)
Date: Thu, 15 Jan 1998 10:08:25 -0800
To: Edgar Hellfritsch <edgar.hellfritsch@rrze.uni-erlangen.de>,
        rem-conf@es.net
From: Bill Strahm <bstrahm@ideal.jf.intel.com>
Subject: Re: SDR private announcements
In-Reply-To: <34bde6661085002@max4.rrze.uni-erlangen.de>
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At 11:35 AM 1/15/98 +0100, you wrote:
>Hi
>
>In order to keep a public sdp announcement posted forever, you just set the
>date of expiry to zero in the corresponding cache file. fine. (ok, You're
>supposed not to do this, but in our case it's a necessity.)
>
Is there any chance you could define an admin scope to announce your
private sessions on.  Along with the encryption, you get privacy through
obscurity, and non-interested people will not have to see your private
session and even know it is going on.  I am getting more and more concerned
about seeing SessionTitle (private) all over the SDP interface, since these
announcements have only a small audience (all though it might be global) do
they really belong on the public announcement group ?

Bill
Bill Strahm    |Programming today is a race between
bstrahm@       |software engineers striving to build
ibeam.intel.com|bigger and better idiot-proof programs,
(503) 264-4632 |and the Universe trying to produce
               |bigger and better idiots.  So far, the
               |Universe is winning.--Rich Cook
I am not speaking for Intel.  And Intel rarely speaks for me



From rem-conf Fri Jan 16 03:12:08 1998 
From rem-conf-request@es.net Fri Jan 16 03:12:07 1998
Received: from list by mail1.es.net with local (Exim 1.81 #2)
	id 0xt9Tm-00033M-00; Fri, 16 Jan 1998 02:58:26 -0800
X-Authentication-Warning: archimedes.informatik.uni-mannheim.de: kuehne owned process doing -bs
Date: Fri, 16 Jan 1998 11:58:20 +0100 (MET)
From: Gerald Kuehne <kuehne@pi4.informatik.uni-mannheim.de>
X-Sender: kuehne@archimedes
To: rem-conf@es.net
Subject: RTP payload for MPEG4
Message-ID: <Pine.GSO.3.96.980116115646.20163D-100000@archimedes>
MIME-Version: 1.0
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Hi,

are the any information available concerning a RTP payload for
(parts of) MPEG4?

bye 
Gerald





From rem-conf Fri Jan 16 05:51:50 1998 
From rem-conf-request@es.net Fri Jan 16 05:51:48 1998
Received: from list by mail1.es.net with local (Exim 1.81 #2)
	id 0xtC4v-0004MP-00; Fri, 16 Jan 1998 05:44:57 -0800
Date: 16 Jan 1998 08:42 EST
Sender: "Vahe Balabanian" <balabani@nortel.ca>
To: kuehne@pi4.informatik.uni-mannheim.de
Cc: rem-conf@es.net, mpeg4-rtp@valathar.Eng.Sun.COM
From: "Vahe Balabanian" <balabani@nortel.ca>
Subject: re:RTP payload for MPEG4
Message-Id: <E0xtC4t-0004MF-00@mail1.es.net>
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Gerard, a group from IETF AVT and ISO MPEG is presently drafting 
an RFC. It has not been uploaded yet as a draft RFC.

Vahe Balabanian
Nortel
Advanced Tehnology

In message "RTP payload for MPEG4", kuehne@pi4.informatik.uni-mannheim.de writes:

>
>Hi,
>
>are the any information available concerning a RTP payload for
>(parts of) MPEG4?
>
>bye 
>Gerald
>
>
>
>                                                        
>



From rem-conf Fri Jan 16 16:15:53 1998 
From rem-conf-request@es.net Fri Jan 16 16:15:52 1998
Received: from list by mail1.es.net with local (Exim 1.81 #2)
	id 0xtLpb-00022L-00; Fri, 16 Jan 1998 16:09:47 -0800
From: Ken Wong <kenwong@nih.gov>
Reply-To: kenwong@nih.gov
To: rem-conf@es.net
Subject: NIH Mbone Broadcast
Message-ID: <SIMEON.9801161953.B@tsavo.nih.gov>
Date: Fri, 16 Jan 1998 19:02:53 -0500 (EST)
Priority: NORMAL
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The NIH will be multicasting live to the Global Mbone the 
Xenotransplantation Conference.  The conference will be 
held on January 21 and 22 at the NIH in Bethesda, Md from 
8am to 6pm EST.  This multicast is intended to facilitate 
the real-time delivery of health information.

All comments welcomed.

Ken

----------------------
Ken Wong
kenwong@nih.gov




From rem-conf Fri Jan 16 20:57:24 1998 
From rem-conf-request@es.net Fri Jan 16 20:57:24 1998
Received: from list by mail1.es.net with local (Exim 1.81 #2)
	id 0xtQ9A-0003U8-00; Fri, 16 Jan 1998 20:46:16 -0800
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X-Mailer: exmh version 2.0gamma 1/27/96
To: kenwong@nih.gov
cc: rem-conf@es.net
Subject: Re: NIH Mbone Broadcast 
In-reply-to: Your message of "Fri, 16 Jan 1998 19:02:53 EST."
             <SIMEON.9801161953.B@tsavo.nih.gov> 
Mime-Version: 1.0
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Date: Fri, 16 Jan 1998 20:46:03 -0800
From: Amancio Hasty <hasty@rah.star-gate.com>
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Curious what is Xenotransplantation?

	Tnks,
	Amancio





From rem-conf Mon Jan 19 09:48:42 1998 
From rem-conf-request@es.net Mon Jan 19 09:48:42 1998
Received: from list by mail1.es.net with local (Exim 1.81 #2)
	id 0xuKl7-0001aj-00; Mon, 19 Jan 1998 09:13:13 -0800
Date: Mon, 19 Jan 1998 12:11:27 -0500 (EST)
From: Christian Huitema <huitema@bellcore.com>
Message-Id: <980119121126.ZM2984@seawind.bellcore.com>
In-Reply-To: Stephen Casner <casner@precept.com>
        "Re: RTP QOS reports for large groups" (Jan 10, 11:24pm)
References: <Pine.PCW.3.95.980110230248.17286C-100000@revelstoke.precept.com>
X-Mailer: Z-Mail (5.0.0 30July97)
To: Stephen Casner <casner@precept.com>, rem-conf@es.net
Subject: Re: RTP QOS reports for large groups
Cc: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
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> When that field was added (at the
> 1995 San Jose IETF, if memory serves), the purpose was to allow single
> reception reports to be useful because the interval between reports
> could become quite long in large groups.

There was actually a debate between two groups based on, essentially the
VAT experience and the IVS/MICE experience.  Van Jacobson had work hard to
develop diagnostic tools that would take the reports from VAT and use them
in order to detect network problems, or MBONe problems.  To do that, you
need indications that are very well defined, so you can easily compare
reports issued by different implementations, and relate them to physical
variables.

In the MICE project, we had started experimenting with rate adaptation as
a way to cope with congestion (see the communication by Wakeman, Turletti
and Bolot in the 1994 Sigcomm conference).  To adapt, you need a signal
that reflects the short term experience of users -- the average loss rate
over the last 5 minutes is not terribly useful.  If you have short term
averages coming from a random set of recipient, you can assume that the
observers get a statistical sampling of the short term conditions, and you
can then adapt the behavior of sources. The loss rate indication was
initially added to RTP to enable this kind of adaptation; but the whole
purpose was kind of lost in the final definition.

I would like to define the loss rate as a running average over some
period. Clearly, there is a debate on what the period should be.  However,
there are only 8 bits allocated to the loss rate, which sort of constrains
the design space.  What about an exponential average with a half life of
128 packets ?

-- 
Christian Huitema
----------
See you at INET'98, Geneva 21-24,July 98 http://www.isoc.org/inet98/



From rem-conf Mon Jan 19 11:30:39 1998 
From rem-conf-request@es.net Mon Jan 19 11:30:38 1998
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	id 0xuMlO-0002MZ-00; Mon, 19 Jan 1998 11:21:38 -0800
From: Mark Handley <mjh@east.isi.edu>
X-Organisation: Information Sciences Institute, USC
X-Phone: +1 617 253 6011
To: Christian Huitema <huitema@bellcore.com>
cc: Stephen Casner <casner@precept.com>, rem-conf@es.net,
        Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Subject: Re: RTP QOS reports for large groups 
In-reply-to: Your message of "Mon, 19 Jan 1998 12:11:27 EST."
             <980119121126.ZM2984@seawind.bellcore.com> 
Date: Mon, 19 Jan 1998 14:21:30 -0500
Message-ID: <6540.885237690@north.lcs.mit.edu>
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>I would like to define the loss rate as a running average over some
>period. Clearly, there is a debate on what the period should be.  However,
>there are only 8 bits allocated to the loss rate, which sort of constrains
>the design space.  What about an exponential average with a half life of
>128 packets ?

Unless you're actually using this is a feedback loop, it's likely that
the current definition is more useful for general monitoring purposes.
Whilst I agree that the current definition is not terribly useful for
any congestion control scheme, I'm unconvinced that we can fix an
interval now and get it right.

The alternative would seem to be to change the spec to allow any form
of filter to be used to process the loss reports prior to sending, and
to specify the filter characteristics out of band (such as in SDP).
The default could be the current spec, but apps should expect to be
able to be told to use an exponential average or a time interval
filter with a specified parameter.  This would let us experiment with
particular values without having to have access to the receivers
source code.  Different sessions could use different filters if this
turns out to be useful.

To implement this, we'd probably need a bit from somewhere to indicate
that a particular RR is using the specified report filter rather than
the default one to prevent old receivers confusing senders.

Cheers,
	Mark




From rem-conf Mon Jan 19 22:21:59 1998 
From rem-conf-request@es.net Mon Jan 19 22:21:59 1998
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Message-Id: <199801200615.WAA22300@mlk.cs.berkeley.edu>
To: rem-conf@es.net
cc: Brian Smith <bsmith@cs.cornell.edu>
Subject: ACM Multimedia 98
From: mccanne@eecs.berkeley.edu (Steven McCanne)
Date: Mon, 19 Jan 1998 22:15:16 -0800
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I would like to point out to people on this list that
the ACM Multimedia paper deadline is approaching.
This conference is unique in that it brings together
researchers working on content and applications with
those (like many on this list) working on protocols
and network infrastructure.  A number of papers related
to RTP and the MBone have been published in past
Multimedia proceedings.  More info, including the call,
is available at http://www.acm.org/sigmm/MM98/

Steve




From rem-conf Wed Jan 21 01:36:43 1998 
From rem-conf-request@es.net Wed Jan 21 01:36:42 1998
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From: anand@ece.iisc.ernet.in (SVR Anand)
Message-Id: <9801210915.AA03041@ece.iisc.ernet.in>
Subject: Re: RTP QOS reports for large groups
To: hgs@cs.columbia.edu
Date: Wed, 21 Jan 98 14:45:17 GMT+5:30
Cc: rem-conf@es.net
In-Reply-To: <34B83A88.FD9941F2@cs.columbia.edu>; from "Henning Schulzrinne" at Jan 10, 98 10:20 pm
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Hi

Can't we adopt ideas given in the paper "A reliable multicast framework for
light-weight sessions and application level framing" by Sally Floyd et al. 
and bring in some sort of scoping for RR reports in the context of quality
reporting ? 
The same paper mentions "packet loss in multicast traffic is most likely
to occur not in the backbone but in the edges of the multicast network"
and suggests the use of loss neighborhood to limit the number of loss 
requests, repairs. 
Poor audio/video reception can be due to congested links, and the locality
of the congestion from the above paragraph is most likely at the edge of 
the network. So why can't we limit RR reports to "loss neighborhood" for
action to be taken instead of to the entire group ? 

I am not addressing the issue of participants identity conveyed in the
reports here.

Regards
Anand.



From rem-conf Wed Jan 21 02:31:37 1998 
From rem-conf-request@es.net Wed Jan 21 02:31:37 1998
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To: anand@ece.iisc.ernet.in (SVR Anand)
cc: hgs@cs.columbia.edu, rem-conf@es.net
Subject: Re: RTP QOS reports for large groups
In-reply-to: Your message of "Wed, 21 Jan 1998 14:45:17 GMT." <9801210915.AA03041@ece.iisc.ernet.in>
Date: Wed, 21 Jan 1998 10:22:20 +0000
Message-ID: <2125.885378140@cs.ucl.ac.uk>
From: Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>
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In message <9801210915.AA03041@ece.iisc.ernet.in>, SVR Anand typed:

 >>Can't we adopt ideas given in the paper "A reliable multicast framework for
 >>light-weight sessions and application level framing" by Sally Floyd et al. 
 >>and bring in some sort of scoping for RR reports in the context of quality
 >>reporting ? 

yes  - dynamic, continuous
autoconfiguring areas and "representatives" (possibly nested)
who then scope their reports appropriately and dynamically set their
timers as per SRM is a good idea - i think deborah estrin has folks
working on this...

 >>The same paper mentions "packet loss in multicast traffic is most likely
 >>to occur not in the backbone but in the edges of the multicast network"
 >>and suggests the use of loss neighborhood to limit the number of loss 
 >>requests, repairs. 
 >>Poor audio/video reception can be due to congested links, and the locality
 >>of the congestion from the above paragraph is most likely at the edge of 
 >>the network. So why can't we limit RR reports to "loss neighborhood" for
 >>action to be taken instead of to the entire group ? 
 >>
 >>I am not addressing the issue of participants identity conveyed in the
 >>reports here.
 >>
 >>Regards
 >>Anand.
 >>

 cheers

   jon




From rem-conf Wed Jan 21 02:31:38 1998 
From rem-conf-request@es.net Wed Jan 21 02:31:37 1998
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	id 0xuxK0-000778-00; Wed, 21 Jan 1998 02:23:48 -0800
To: anand@ece.iisc.ernet.in (SVR Anand)
cc: hgs@cs.columbia.edu, rem-conf@es.net
Subject: Re: RTP QOS reports for large groups
In-reply-to: Your message of "Wed, 21 Jan 1998 14:45:17 GMT." <9801210915.AA03041@ece.iisc.ernet.in>
Date: Wed, 21 Jan 1998 10:23:09 +0000
Message-ID: <779.885378189@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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--> SVR Anand writes:
>Can't we adopt ideas given in the paper "A reliable multicast framework for
>light-weight sessions and application level framing" by Sally Floyd et al. 
>and bring in some sort of scoping for RR reports in the context of quality
>reporting ? 
>The same paper mentions "packet loss in multicast traffic is most likely
>to occur not in the backbone but in the edges of the multicast network"
>and suggests the use of loss neighborhood to limit the number of loss 
>requests, repairs. 

Well, my experience is that there is a lot of congestion in the backbone
links too, both within SuperJANET (the UK research network), and on the
international links. Maybe the "packet loss in multicast traffic is most
likely to occur not in the backbone but in the edges of the multicast
network" comment is true in some places, but certainly not all.

>Poor audio/video reception can be due to congested links, and the locality
>of the congestion from the above paragraph is most likely at the edge of 
>the network. So why can't we limit RR reports to "loss neighborhood" for
>action to be taken instead of to the entire group ? 

This limits the scope for session monitoring tools somewhat, which are very
useful as the first stage for finding network problems. I'm also not sure
it's the right approach for loss repair: in many cases you want the RR to
go back to the original sender of a media stream, so that sender can be
informed to lower its rate, and add some form of FEC to the stream. 

-- 
Colin Perkins                   Email: c.perkins@cs.ucl.ac.uk
Department of Computer Science  Phone: (+44) 171 419 3666
University College London       WWW  : http://www.cs.ucl.ac.uk/staff/c.perkins/



From rem-conf Wed Jan 21 04:00:11 1998 
From rem-conf-request@es.net Wed Jan 21 04:00:10 1998
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	id 0xuymu-0000UM-00; Wed, 21 Jan 1998 03:57:44 -0800
Date: Wed, 21 Jan 1998 11:47:55 +0000 (GMT)
From: "Alan J. Flavell" <flavell@a5.ph.gla.ac.uk>
Reply-To: A.Flavell@physics.gla.ac.uk
To: rem-conf@es.net
Subject: vat on Win95 - voice controlled mode?
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I don't seem to be able to operate vat in a voice controlled mode in
Win95.  When I'm speaking, the respective line of the vat display "lights
up", and when I stop speaking, it goes out, but I don't seem to hear from
the other participant(s) until I turn off the "talk" button. 

First investigations suggested that it might be an issue whether the sound
card supports full duplex.  Well, according to the documentation, this
sound card _does_ support full duplex.

Should I expect this to work or not?  If so, any suggestions how to go
about debugging it further, please? 






From rem-conf Wed Jan 21 04:00:11 1998 
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Date: Wed, 21 Jan 1998 12:53:37 +0100
From: Markus Hofmann <hofmann@telematik.informatik.uni-karlsruhe.de>
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CC: SVR Anand <anand@ece.iisc.ernet.in>, rem-conf@es.net
Subject: Re: RTP QOS reports for large groups
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Jon Crowcroft wrote:

> yes  - dynamic, continuous
> autoconfiguring areas and "representatives" (possibly nested)
> who then scope their reports appropriately and dynamically set their
> timers as per SRM is a good idea - i think deborah estrin has folks
> working on this...

Right, but scoping based on TTL is not very precise. Why not to build
separate subgroups, each of them having a separate multicast address?
There has also been some interesting discussion during the RM meeting in
Cannes about a modified multicast service model (see
http://www.east.isi.edu/rm/ for presentation material an minutes).)

Cheers
  Markus



From rem-conf Wed Jan 21 04:23:33 1998 
From rem-conf-request@es.net Wed Jan 21 04:23:32 1998
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To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
cc: rem-conf@es.net
Subject: Re: I-D ACTION:draft-ietf-avt-rtp-new-00.txt,.ps 
In-reply-to: Your message of "Mon, 05 Jan 1998 12:42:53 EST."
             <34B11B9D.1A7E1330@cs.columbia.edu> 
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Date: Wed, 21 Jan 1998 13:18:22 +0100
From: Peter Parnes <peppar@cdt.luth.se>
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Hi 

This is probably some very old thing but I was wondering why it says on page 
45 that
"The NAME value is expected to remain constant at least for the duration of a 
session." regarding the SDES NAME field?

If that should remain in there shouldn't all MBone applications enforce this 
then? Today, as you all know, it is very easy to change the name during a 
session.

/P










From rem-conf Wed Jan 21 06:02:10 1998 
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Date: Wed, 21 Jan 1998 08:55:31 -0500
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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Peter Parnes wrote:
> 
> Hi
> 
> This is probably some very old thing but I was wondering why it says on page
> 45 that
> "The NAME value is expected to remain constant at least for the duration of a
> session." regarding the SDES NAME field?
> 
> If that should remain in there shouldn't all MBone applications enforce this
> then? Today, as you all know, it is very easy to change the name during a
> session.

This may be partly due to the fact that not all applications support
NOTE as the appropriate alternative to putting messages in one's name
field. It may be appropriate to add advice to discourage the change of
these fields and use NOTE for temporary anouncements.

> 
> /P

-- 
Henning Schulzrinne        email: schulzrinne@cs.columbia.edu
Dept. of Computer Science  phone: +1 212 939-7042 (@Bell Labs: 732 949
8344)
Columbia University        fax:   +1 212 666-0140
New York, NY 10027         URL:   http://www.cs.columbia.edu/~hgs



From rem-conf Wed Jan 21 11:25:49 1998 
From rem-conf-request@es.net Wed Jan 21 11:25:48 1998
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	id 0xv5f0-0004oB-00; Wed, 21 Jan 1998 11:18:02 -0800
From: Tom Zoerner <Tom.Zoerner@informatik.uni-erlangen.de>
Message-Id: <199801211917.UAA05180@faui01.informatik.uni-erlangen.de>
Subject: Timestamp definition in FEC draft
To: rem-conf@es.net
Date: Wed, 21 Jan 1998 20:17:56 +0100 (MET)
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Hi.

I'm currently implementing a multicast application with FEC according to
draft-ietf-avt-fec-01. However I have some trouble with the way timestamps
are supposed to be set. In the draft is says to use the timestamp of the
oldest xor'ed packet. I see two problems with this:

1) I cannot add LPC redundancy for the latest packet, since the offsets
in redundancy encoded payloads can only be negative. If I do FEC with
one or several later packets however, the offset would be zero or positive.

I'm not sure yet if it's a good idea to combine LPC redundancy with FEC,
but my idea is to recover loss-less from single losses (I'm working with
layered codecs so I need that property) and use the usual repair and error
concealment techniques for multi-packet losses.

2) jitter/delay calculation: well, I cannot actually generate the FEC
packet before I have the data for the latest to-be-coded packet. That
means however, that the timestamp on the packet being sent differs
wildly from the actual time the packet was generated. Hence I can not
easily use FEC packets for jitter calculation.

The simple solution seems to be to exclude FEC packets completely from
jitter calculation, however the sender may use a scheme where he sends
*only* FEC encoded packets, e.g. scheme 2 from Budge's draft.

The only reason the draft gives for chosing the oldest timestamp for
the whole packet is because it "is a somewhat natural definition".
Well, I believe using the latest timestamp would be even more natural,
since in addition to the 2 points above the timestamps in the packet
stream would still increase monotonically.

So I'd like to propose to reconsider the draft in that point.
(Is there actually already software in use based on that draft?)

-tom



From rem-conf Wed Jan 21 15:03:27 1998 
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	id 0xv970-0006N4-00; Wed, 21 Jan 1998 14:59:10 -0800
From: "Edward W. Knightly" <knightly@ece.rice.edu>
Message-Id: <980121165906.ZM25471@qos.ece.rice.edu>
Date: Wed, 21 Jan 1998 16:59:06 -0600
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Subject: IWQoS Call for Papers
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Below is the final CFP for IWQoS '98. The workshop's URL is 
     http://www-ece.rice.edu/conf/iwqos98/

Best regards,
-Ed Knightly (IWQoS '98 co-chair)



             ________________________________________________
                               CALL FOR PAPERS

                  Sixth IEEE/IFIP International Workshop on
                       Quality of Service (IWQoS '98)

                           -Napa, California USA -
                               May 18-20, 1998

                 Sponsored by IEEE Communications Society
              Technical Committees on Computer Communications, 
             Internet, Multimedia Communications, and Software

		  Technical co-sponsorship by IFIP WG6.1* 

                     In co-operation with ACM SIGCOMM 

              Patrons: Hewlett Packard and Nokia Corporation
			      *to be approved
             ________________________________________________

                                  

INVITED PROGRAM
---------------
Keynote Address - Roch Guerin, IBM Research
Keynote Paper - Domenico Ferrari, Universita' Cattolica Piacenza, Italy 
Panel Chair - Hui Zhang, Carnegie Mellon University
   "The Future of Differential and Integrated Services in the Internet" 
Panel Chair - Vaduvur Bharghavan, University of Illinois at Urbana Champaign
   "QoS in Wireless Networks: the Transition from Myth to Reality"

                                  
WORKSHOP SCOPE
--------------
This is the sixth in a series of workshops aimed at providing an
international forum for the exchange of information on quality of
service (QoS) research in distributed and Internet systems. The
objective of the Sixth International Workshop on Quality of Service is
to bring together researchers, developers, and practitioners working
in all facets of QoS research addressing distributed systems, Internet
services, multimedia, operating systems, networking, and middleware to
discuss recent results and future directions.

Contributions are solicited in all areas of QoS research in distributed
systems and networking, including, but not limited to:
- QoS Specification, Resource Allocation, (Re)negotiation, and Monitoring 
- QoS and the Internet 
- QoS Performance Modeling, Analysis and Evaluation 
- Adaptive Applications and Services 
- QoS Measurement, Management and Control 
- QoS Support for Mobility and Wireless Networks 
- Middleware Solutions for End-to-End QoS 
- Media Scaling and QoS Filtering Techniques 
- QoS Support for Information Appliances 
- Storage Systems/Database Support for QoS 
- Experimental Studies on QoS, including Human Perception 
- QoS Programmability, Languages and Formal Method Techniques 
- QoS Economics and Implications

WORKSHOP URL
------------
http://www-ece.rice.edu/conf/iwqos98
 

IMPORTANT DATES
---------------
   * Submission of contribution: February 23, 1998
   * Notification of acceptance: April 6, 1998
   * Camera-ready paper due: April 13, 1998


PAPER SUBMISSION 
---------------- 
Authors are invited to submit full papers and position statements for
review. Full papers should not exceed 12 single-spaced pages including
figures, tables, and references, using a typeface no smaller than 11
points. Position statements should not exceed 3 single-spaced
pages. To expedite the reviewing process, please submit the paper
electronically (in postscript format only, through e-mail) to
iwqos98@ece.rice.edu. All submissions must include name and
affiliation of the authors, a contact address of the main author, an
abstract, and keywords.
                                                                           

PROGRAM CO-CHAIRS:
------------------
Ed Knightly, Rice University
Rich Friedrich, Hewlett Packard Laboratories

PROGRAM COMMITTEE:
------------------
Vaduvur Bharghavan, University of Illinois at Urbana Champaign, USA 
Gordon Blair, University of Lancaster, UK 
Andrew Campbell, Columbia University, USA 
Simon Crosby, Cambridge University, UK 
Jon Crowcroft, UCL, UK 
Hermann de Meer, University of Hamburg, Germany 
Jan de Meer, GMD Fokus, Germany 
Sudhir Dixit, Nokia Research, USA
Alexandros Eleftheriadis, Columbia University, USA 
Domenico Ferrari, Universita' Cattolica Piacenza, Italy 
Leonard Franken, KPN Research, The Netherlands 
Michael Fry, University of Technology, Australia 
Jean-Peirre Hubaux, EPFL, Switzerland 
Brigitte Kerherve, University of Quebec at Montreal, Canada 
Glenford Mapp, Olivetti Research Ltd, UK 
Mahmoud Nagshineh, IBM Thomas J. Watson Research Center, USA 
Klara Nahrstedt, University of Illinois at Urbana Champaign, USA 
Max Ott, C&C Research, NEC USA 
Guru Parulkar, Washington University, USA 
Steve Pink, Lulea University of Technology, Sweden 
Jerry Rolia, Carleton University, Canada 
Douglas Schmidt, Washington University, USA 
Chris Sluman, OpenIT Ltd, UK 
Cormac Sreenan, AT&T Research, USA 
Hideyuki Tokuda, Keio University, Japan 
James VanLoo, Sun Microsystems, USA 
Andreas Vogel, DSTC, Australia 
Lixia Zhang, University of California, Los Angeles, USA 
Hui Zhang, Carnegie Mellon University, USA 






From rem-conf Thu Jan 22 01:55:03 1998 
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Date: Thu, 22 Jan 1998 01:24:19 -0800
From: Chase Bailey <chase@cisco.com>
Organization: Cisco Systems
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        apc_members@hornbill.ee.nus.sg, atmpost@hplabs.hpl.hp.com,
        ccrc@dworkin.wustl.edu, cellular@comnets.rwth-aachen.de,
        citmalum@haas.berkeley.edu, citmlst@haas.berkeley.edu,
        cnom@maestro.bellcore.com, commsoft@cc.bellcore.com,
        comsoc-chapters@ieee.org,
        "comsoc-gicb@ieee.org" <comsoc-gicb@ieee.org>, comsoc.tac@tab.ieee.org,
        cost237-gen@montefiore.ulg.ac.be,
        cost237-teleservices@comp.lancs.ac.uk,
        cost237-transport@comp.lancs.ac.uk, Cost237-WG3@masi.ibp.fr,
        cswg%sunoco@relay.nswc.navy.mil, dbworld@cs.wisc.edu,
        end2end-interest@isi.edu, enternet@bbn.com, epr@infotest.com,
        f-troup@codex.cis.upenn.edu, FORTE94_CFP@admin.unibe.ch,
        g-troup@ccrc.wustl.edu, globecom@signet.com.sg, golshani@asu.edu,
        hipparch@sophia.inria.fr, hpcs95-prog@VM1.NODAK.EDU, ICAD@santafe.edu,
        ieee_rtc_list@cs.tamu.edu, ieeetcpc@ccvm.sunysb.edu, itc@fokus.gmd.de,
        kia@koko.CS.UNLV.EDU, manet@itd.nrl.navy.mil, mobile-ip@tadpole.com,
        mospf@gated.cornell.edu, mpls@external.cisco.com,
        multicomm@cc.bellcore.com, ni@cps.msu.edu, nlanr@nlanr.net,
        ns-users@mash.cs.berkeley.edu, ns-annouce@mash.cs.berkeley.edu,
        OSIMCAST@bbn.com, park@cstp.umkc.edu, perform@tay1.dec.com,
        rdhm@masi.ibp.fr, rem-conf@es.net, reres@laas.fr,
        rm@mash.cs.berkeley.edu, rsvp@isi.edu, SC6WG4@ntd.comsat.com,
        sigmedia@bellcore.com, singhal@cis.ohio-state.edu, tccc@cs.umass.edu,
        tcp-over-satellite@achtung.sp.trw.com, udlr@sophia.inria.fr,
        wiggles@ca.ibm.com, xtp-relay@cs.concordia.ca
Subject: HOT INTERCONNECTS VI - CALL FOR PAPERS
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CALL FOR PAPERS

HOT INTERCONNECTS VI

A Symposium on High Performance Interconnects,
>from system buses and interfaces to networks

August 13-15, 1998
Stanford University
Stanford, California, U.S.A.

Sponsored by the
IEEE Computer Society Technical Committee on
Microprocessors and Microcomputers.

Hot Interconnects is an international symposium focusing
on the hardware and software architecture and implementation
of high-performance interconnects of all scales. The
conference is directed particularly at new and exciting
product and technology innovations in interconnects from
system buses and interfaces to small and large scale networks
and network software. Contributions should focus on real
products, prototypes, or experimental systems and their
performance evaluation. Contributions are solicited in,
but not limited to, the following topics:

     - High speed serial links
     - High performance buses and interfaces
     - Multi-Mbps to the home
     - Processor-memory interconnects
     - Gigabit per second networking technologies
     - Cluster interconnects
     - Terabit per second networking technologies
     - Network-attached storage devices and interfaces
     - Performance evaluation of interconnects
     - Optical Interconnects
     - Web casting technologies
     - Video and audio on the Internet
     - High performance network switches
     - Wireless and mobile networks
     - Bus support chip sets
     - Network security
     - Networking chip sets
     - Innovations in distributed computing
     - Network protocols on a chip
     - Emerging network appliance technologies
     - Innovations in distributed computing
     - Low power devices

The deadline for submissions is March 1, 1998.

Contributions can be in the form of full papers or in
abstracts or position statements for review. Full papers
should not exceed 15 single-spaced pages including
figures, tables, and references. Abstracts or Position
Statements should not exceed 3 pages. To expedite the
reviewing process please submit the paper electronically
in ASCII, postscript, word or pdf format to
hoti@stanford.edu.

Hardcopy submissions should be mailed in 10 copies to
Nick McKeown, Program Co-chair, Hot Interconnects VI,
Computer Systems Lab., Stanford University, Stanford, CA
94305-9030, USA.

All submissions must include name and affiliation of the
authors, a contact address (phone, fax and email address)
of the main author.

Web Site:
  http://www.hoti.org

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

Important Dates:

    -  March 1, 1998: Paper submission deadline
    -  May  15, 1998: Notifications of acceptance mailed to authors
    -  June 15, 1998: Final camera-ready manuscripts due.

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

General Chair:

    - Hasan Alkhatib, Santa Clara University
      halkhatib@scu.edu, 408-554-4485

Program Commitee Co-Chairs:

    - Nick McKeown, Stanford University
      nickm@stanford.edu, 650-725-3641
    - Chase Bailey, Cisco Systems
      chase@cisco.com, 408-527-3765

Program Commitee:

    - Bill Dally, Stanford
    - Chuck Thacker, Microsoft
    - Craig Partridge, BBN/GTE
    - Dan Pitt, Bay Networks
    - Dave Oran, Cisco Systems
    - Greg Chesson, Silicon Graphics
    - James Luciani, Bay Networks
    - Kathleen Nichols, Bay Networks
    - Mark Horowitz, Stanford
    - Martin Izzard, Texas Instruments
    - Qiang Li, Santa Clara University
    - Randy Rettberg, Sun Microsystems
    - Steve Deering, Cisco Systems

Publicity Committee:

    - Kristina Scott, Cisco Systems
      krscott@cisco.com, 408-527-1504







From rem-conf Thu Jan 22 03:28:31 1998 
From rem-conf-request@es.net Thu Jan 22 03:28:31 1998
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	id 0xvKhv-0002a9-00; Thu, 22 Jan 1998 03:22:03 -0800
To: Tom Zoerner <Tom.Zoerner@informatik.uni-erlangen.de>
cc: rem-conf@es.net
Subject: Re: Timestamp definition in FEC draft
In-reply-to: Your message of "Wed, 21 Jan 1998 20:17:56 +0100." <199801211917.UAA05180@faui01.informatik.uni-erlangen.de>
Date: Thu, 22 Jan 1998 11:21:12 +0000
Message-ID: <721.885468072@cs.ucl.ac.uk>
From: Orion Hodson <O.Hodson@cs.ucl.ac.uk>
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<199801211917.UAA05180@faui01.informatik.uni-erlangen.de>Tom Zoerner writes:
> I'm currently implementing a multicast application with FEC according to
> draft-ietf-avt-fec-01. However I have some trouble with the way timestamps
> are supposed to be set. In the draft is says to use the timestamp of the
> oldest xor'ed packet. I see two problems with this:
> 
> 1) I cannot add LPC redundancy for the latest packet, since the offsets
> in redundancy encoded payloads can only be negative. If I do FEC with
> one or several later packets however, the offset would be zero or positive.
> 
> I'm not sure yet if it's a good idea to combine LPC redundancy with FEC,

It looks like a lot of work for an uncertain reward.  I am also
implementing the parity fec proposed in draft-ietf-avt-fec-01, and
have some simulator code for the budge schemes for a prelimary
investigation.  I ran random and burst loss models against parity fec,
interleaving, and redundancy.  Two of the parity fec schemes proposed
by Budge et al performed considerably better than redundancy upto 20%
loss rates before their performance fell sharply.  

The simulation did not account for redundancy generating larger
packets and parity fec more packets, both of which adversely affect
router performance.  Neither did the simulation allow for bigger time
offsets between the media units xor'ed together which makes both
redundancy and parity fec deal better with the bursty losses observed.
We probably want tools to be smart about the ranges and effects of fec
techniques, or perhaps the fec on separate multicast groups.

> 2) jitter/delay calculation: well, I cannot actually generate the FEC
> packet before I have the data for the latest to-be-coded packet. That
> means however, that the timestamp on the packet being sent differs
> wildly from the actual time the packet was generated. Hence I can not
> easily use FEC packets for jitter calculation.

The problem is when you release the packets, if you hold onto all of
the packets with the same timestamp and release them consecutively
then you reduce some of the wild variation.  

> The simple solution seems to be to exclude FEC packets completely from
> jitter calculation, however the sender may use a scheme where he sends
> *only* FEC encoded packets, e.g. scheme 2 from Budge's draft.
> 
> The only reason the draft gives for chosing the oldest timestamp for
> the whole packet is because it "is a somewhat natural definition".
> Well, I believe using the latest timestamp would be even more natural,
> since in addition to the 2 points above the timestamps in the packet
> stream would still increase monotonically.

But this breaks if the offset between xor'ed units is not 1. i.e 

xor  f(x) f(x,x+10) f(x+1) f(x+1,x+11)
ts    x      x+10     x+1      x+11 

-- Orion

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




From rem-conf Thu Jan 22 06:31:45 1998 
From rem-conf-request@es.net Thu Jan 22 06:31:43 1998
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	id 0xvNa4-0003nE-00; Thu, 22 Jan 1998 06:26:08 -0800
Message-ID: <34C7489C.E3B9B484@dnrc.bell-labs.com>
Date: Thu, 22 Jan 1998 08:24:44 -0500
From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
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To: Tom Zoerner <Tom.Zoerner@informatik.uni-erlangen.de>
CC: rem-conf@es.net
Subject: Re: Timestamp definition in FEC draft
References: <199801211917.UAA05180@faui01.informatik.uni-erlangen.de>
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Tom Zoerner wrote:
> 
> 2) jitter/delay calculation: well, I cannot actually generate the FEC
> packet before I have the data for the latest to-be-coded packet. That
> means however, that the timestamp on the packet being sent differs
> wildly from the actual time the packet was generated. Hence I can not
> easily use FEC packets for jitter calculation.
> 
> The simple solution seems to be to exclude FEC packets completely from
> jitter calculation, however the sender may use a scheme where he sends
> *only* FEC encoded packets, e.g. scheme 2 from Budge's draft.

An alternative is to use the redundancy payload format to send the FEC
packets. Lets say you were using a (5,7) scheme (that is, five data
packets, plus two more FEC). The two FEC packets would then be
piggybacked as "redundant codecs" with the first two data packets in the
next block. This actually accomplishes a number of things:

-hides the jitter in FEC packet generation, since the media packet
timestamps, not the FEC, are used for jitter computation
-reduces header overheads
-works well with RTP header compression

I should point out to you that I don't think the current FEC scheme
supports sending *only* FEC encoded packets. This is because the FEC
packets use a bitmask to indicate which media packets the FEC is applied
to. The bitmask indicates offsets in sequence number space. Now, if the
media packets are never actually sent by themselves, they don't have a
sequence number of their own, and therefore cannot be referenced via the
mask. Thus, the FEC payload format would only work for what are called
"systematic codes" where the FEC always follows the original data (with
the current mask definition, you could also send them before). I believe
that there are results from coding theory that have shown that this
restriction does not affect the performance of the code. I'm not sure
whether  these results apply here, though.

The earlier draft didn't have this restriction. The media packets had
their own sequence numbering space. FEC packets had their sequence
number computed in the same way as the media - the xor of the sequence
numbers of the associated packets. The problem with this was that having
screwy, non-contiguous sequence numbering broke RTP compression and made
it harder to actually detect lost packets. 

> 
> The only reason the draft gives for chosing the oldest timestamp for
> the whole packet is because it "is a somewhat natural definition".
> Well, I believe using the latest timestamp would be even more natural,
> since in addition to the 2 points above the timestamps in the packet
> stream would still increase monotonically.

The definition of the timestamp, from RFC1889, is "The timestamp
reflects the sampling instant of the first octet in the RTP data
packet". For FEC, this would be the sampling instant of the oldest
packet covered by the FEC. This is what I meant by "natural" - it is
based on the current definition.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
PHONE: (732) 949-6418                       Rm. 4C-526
FAX:   (732) 834-5379
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen



From rem-conf Thu Jan 22 06:54:26 1998 
From rem-conf-request@es.net Thu Jan 22 06:54:25 1998
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	id 0xvNxO-00048I-00; Thu, 22 Jan 1998 06:50:14 -0800
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From: "Ron Kalian"<Ron_Kalian@vocaltec.com>
To: rem-conf@es.net
cc: "Scott Petrack"<Scott_Petrack@vocaltec.com>
Message-ID: <42256594.004F8E70.00@il4.vocaltec.co.il>
Date: Thu, 22 Jan 1998 16:49:07 +0200
Subject: Proposal for change to the current RTP/RTCP spec
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Hi,

The current RTP/RTCP spec allows for sending RTCP Receiver Report (RR)
extensions, should the existing RR parameters not suffice.  The spec
specifies very little about the RR extensions - allowing anything to be an
RR extension so long as it follows the RRs in the RR packet..

A problem with this is that there is no way to uniquely identify
extensions.  Although it is possible to start the extension with a some
"unique" identifier, one party's RR extensions may get misinterpreted by
another party as their own!  I think there must be a way to uniquely
identify one's RR extensions.

For this reason I would like to propose the following addition to the RTCP
spec:
"The RR extension part of the RR packet must begin with a unique 32-bit
identifier, that must be registered with IANA."

RR extensions are not the only way of passing application specific RTCP
data.  It is also possible to use an APP packet, in which case a new RTCP
packet type may be registered with IANA.  But there are cases in which it
is preferable to use RR extensions, because:

1. Transmitting the same extension data, RR extensions save 64-bits due to
lack of APP packet header, even if a 32-bit identifier is included (in the
case of an APP packet - in the 'name' field).

2. Some extension data are closely related to the RR data, in which case it
must be transmitted together with the RR as RR extensions.

In any case, the readers of RR packets should be able to determine whether
or not they can read a given extension.  This can only be achieved using
unique identifiers for RR extensions.


Ron Kalian.


_________________________________

Ron Kalian
Program Manager
VocalTec Communications Ltd.

Phone: ++972-9-9525852
Email: ron@vocaltec.com

_________________________________

Visit us at CeBIT 98'
Hannover Messe
March 19 - 25, 1998





From rem-conf Fri Jan 23 15:32:06 1998 
From rem-conf-request@es.net Fri Jan 23 15:32:04 1998
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	id 0xvsN1-0000Gq-00; Fri, 23 Jan 1998 15:18:43 -0800
Date: Fri, 23 Jan 1998 15:18:30 -0800
From: almeroth@cs.ucsb.edu (Kevin C. Almeroth)
Message-Id: <199801232318.PAA24126@jackson.cs.ucsb.edu>
To: almeroth@cs.ucsb.edu
Subject: DC IETF MBone content available
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This e-mail is going out to a few mailing lists, and I cringe to do
it, but it is relevant to all...  (lists bcc'ed to prevent accidents).

***

       Announcing Availability of the DC IETF Sessions on the IMJ

I'm ready to announce that all the sessions from the 40th IETF in DC
that were transmitted on the MBone are now available on the 
IMJ (http://imj.gatech.edu) for on-demand playback.  You can schedule 
programs via the WWW page and start the MBone sessions via SDR (or 
the WWW page with a simple plug-in).  For additional information about
the MBone see http://mbone.com.

Now that I've graduated and settled down somewhat in my new position at
the University of California, Santa Barbara, I hope to continue research
on the IMJ and to make the archival of IETF meetings a regular service.  
Part of this work includes encoding the sessions in real-time as they are 
being broadcast.  This will significantly cut down on the turn around 
time between when the content is first broadcast and when it is available 
for replay.

Finally, and as usual, I've included the original IMJ e-mail announcement 
for those who want additional information about the IMJ.

Any suggestions for improvement are welcome.

-Kevin

****
              Announcing the Interactive Multimedia Jukebox


   At one point on the MBone list there was a discussion of what on-demand
servers exist or are being developed.  Well, I'd like to announce our 
version called the Interactive Multimedia Jukebox (IMJ).

  The IMJ web page is a request and scheduling interface for the playout 
of content over the MBone.  CONTENT IS ENCODED SO THAT IT CAN BE RECEIVED
WITH THE LATEST VERSIONS OF VIC AND VAT (ANY PLATFORM).

   The IMJ page is located at http://imj.gatech.edu.  Information about 
the IMJ including general info, a postscript version of a paper about the 
IMJ, how-to-use information, and how it was implemented can be found at 
http://www.cc.gatech.edu/computing/Telecomm/IMJ/.  (Recent note:  the
paper will be appearing in the WWW7 conference in Brisbane, Australia)

   Some quick additional information about the IMJ is included below:


IMJ Audio/Video
-----------------
   The IMJ uses the WWW to submit requests which are then scheduled
for playout on the MBone.  Right now we are offering three channels:
Channels 1 and 2 are being broadcast at a TTL of 127.  Channel 3 is
for internal testing and has a a TTL of 15.  Each channel provides 
DVI-2 audio and 128 Kbps H.261 video.  The encoding was done using 
Henning Schulzrinne's rtpplay and rtpdump.

Scheduling
----------
   Program scheduling uses a set of criteria to decide on which channel
to schedule a program.  The current set of criteria are listed just
below the interactive schedule.  We are exploring lots of different
options.

Content
-------
   We have been working on the jukebox paradigm with Turner Broadcasting,
and as a result of their interest they have agreed to let us broadcast
content on the MBone.  I am particularly excited about this aspect because
of their help and interest in investigating new applications on the Internet.

   The plan is to add a new set of content about every two weeks.  When this 
happens, the old content will be moved to the secondary request menu for two 
weeks.  The old-old stuff will be removed.  Hopefully I'll be able to 
advertise on the MBone list when the new stuff is available.

Development Platform
--------------------
   As is mentioned in the paper on the IMJ information page we are using
the IMJ as a platform for a variety of research issues including
providing interactivity, supporting multiple heterogeneous streams,
video server organization, tracking usage, program scheduling, and
pricing.

Acknowledgments
----------------
   In addition to Turner Broadcasting, several other groups have made the
IMJ possible.  The GT Broadband Telecommunications Center sponsored much 
of the research and some of the equipment, and GT Office of Information 
Technology offered technical assistance and additional equipment.


Please send any feedback or suggestions to almeroth@cs.ucsb.edu or
kevin@cc.gatech.edu.

-Kevin Almeroth




From rem-conf Fri Jan 23 22:43:15 1998 
From rem-conf-request@es.net Fri Jan 23 22:43:14 1998
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	id 0xvzG3-00022P-00; Fri, 23 Jan 1998 22:39:59 -0800
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Date: Fri, 23 Jan 1998 22:35:07
To: "Kevin C. Almeroth" <kevin@CC.GATECH.EDU>
From: Ross Finlayson <rsf@cs.stanford.edu>
Subject: Re: DC IETF MBone content available
Cc: IPMULTICAST@STARDUST.COM, rem-conf@es.net
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At 06:22 PM 1/23/98 -0500, Kevin C. Almeroth wrote:
>[...] I hope to continue research
>on the IMJ and to make the archival of IETF meetings a regular service.
>Part of this work includes encoding the sessions in real-time as they are
>being broadcast.  This will significantly cut down on the turn around
>time between when the content is first broadcast and when it is available
>for replay.
[...]
>Any suggestions for improvement are welcome.

Just one:
Make it possible to people with 28.8 kbps MBone connections to access these
archives.

I'm serious.  If RealNetworks can do this sort of thing, than so can we.
If we want to have IP multicast taken seriously, we have to make it more
usable by 'normal people'.

	Ross Finlayson
	Live Networks
	http://www.lvn.com/




From rem-conf Mon Jan 26 08:23:37 1998 
From rem-conf-request@es.net Mon Jan 26 08:23:37 1998
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	id 0xwqyw-0004nH-00; Mon, 26 Jan 1998 08:01:54 -0800
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From: "Scott Petrack"<Scott_Petrack@vocaltec.com>
To: rem-conf@es.net
Message-ID: <42256598.00554AA1.00@il4.vocaltec.co.il>
Date: Mon, 26 Jan 1998 17:57:31 +0200
Subject: Is there a tool for exact recording of received RTP streams?
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Is there a tool (public domain or otherwise) which stores a received RTP
stream on disk, including the precise time at which each packet was
received, in such a way that the receipt of the RTP can be exactly
"re-enacted" at a later point in time (inlcuding all the jitter, etc.).

I'd like to know  what tools are out there.
Thanks
Scott Petrack
VocalTec Communications Ltd.
petrack@vocaltec.com






From rem-conf Mon Jan 26 09:18:19 1998 
From rem-conf-request@es.net Mon Jan 26 09:18:19 1998
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Message-ID: <34CCC3DB.AF7CAFBF@cs.columbia.edu>
Date: Mon, 26 Jan 1998 12:11:55 -0500
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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To: Scott Petrack <Scott_Petrack@vocaltec.com>
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Scott Petrack wrote:
> 
> Is there a tool (public domain or otherwise) which stores a received RTP
> stream on disk, including the precise time at which each packet was
> received, in such a way that the receipt of the RTP can be exactly
> "re-enacted" at a later point in time (inlcuding all the jitter, etc.).

The RTP tools (http://www.cs.columbia.edu/~hgs/rtptools/) do exactly
that.


> 
> I'd like to know  what tools are out there.
> Thanks
> Scott Petrack
> VocalTec Communications Ltd.
> petrack@vocaltec.com

-- 
Henning Schulzrinne        email: schulzrinne@cs.columbia.edu
Dept. of Computer Science  phone: +1 212 939-7042 (@Bell Labs: 732 949
8344)
Columbia University        fax:   +1 212 666-0140
New York, NY 10027         URL:   http://www.cs.columbia.edu/~hgs



From rem-conf Mon Jan 26 09:23:59 1998 
From rem-conf-request@es.net Mon Jan 26 09:23:59 1998
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To: "Scott Petrack" <Scott_Petrack@vocaltec.com>
cc: rem-conf@es.net
Subject: Re: Is there a tool for exact recording of received RTP streams? 
In-reply-to: Your message of "Mon, 26 Jan 1998 17:57:31 +0200."
             <42256598.00554AA1.00@il4.vocaltec.co.il> 
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Date: Mon, 26 Jan 1998 18:17:52 +0100
From: Peter Parnes <peppar@cdt.luth.se>
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You can use my mMOD system for making an exact copy of the original traffic at 
a lter point in time.

http://www.cdt.luth.se/~peppar/progs/mMOD/

/P




From rem-conf Mon Jan 26 09:30:26 1998 
From rem-conf-request@es.net Mon Jan 26 09:30:25 1998
Received: from list by mail1.es.net with local (Exim 1.81 #2)
	id 0xwsIP-0006Kz-00; Mon, 26 Jan 1998 09:26:05 -0800
Date: Mon, 26 Jan 1998 18:25:15 +0100 (MET)
Message-Id: <199801261725.SAA06670@baby.link.no>
To: rem-conf@es.net
Subject: Re: Is there a tool for exact recording of received RTP streams?
From: Petter Reinholdtsen <pere@td.org.uit.no>
Reply-to: pere@td.org.uit.no
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[Scott Petrack]
> Is there a tool (public domain or otherwise) which stores a received
> RTP stream on disk, including the precise time at which each packet
> was received, in such a way that the receipt of the RTP can be
> exactly "re-enacted" at a later point in time (inlcuding all the
> jitter, etc.).

Might MBone VCR or MBone VCR on Demand be the program you are looking
for?  Have a look at <URL:http://www.icsi.berkeley.edu/mbone-vcr/> and
<URL:http://www.informatik.uni-mannheim.de/informatik/pi4/projects/MVoD/>.
-- 
##>  Petter Reinholdtsen <##    | pere@td.org.uit.no
 O-  <SCRIPT Language="Javascript">window.close()</SCRIPT>
http://www.hungry.com/~pere/    | Go Mozilla, go! Go!



From rem-conf Mon Jan 26 15:52:09 1998 
From rem-conf-request@es.net Mon Jan 26 15:52:09 1998
Received: from list by mail1.es.net with local (Exim 1.81 #2)
	id 0xwyBV-0002JD-00; Mon, 26 Jan 1998 15:43:21 -0800
Date: Mon, 26 Jan 1998 15:41:58 -0800 (Pacific Standard Time)
From: Stephen Casner <casner@precept.com>
To: rem-conf@es.net
Subject: Minutes of AVT meeting in Washington DC
Message-ID: <Pine.WNT.3.96.980126153809.-183571B-200000@oak.precept.com>
X-X-Sender: casner@big-bear.precept.com
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  This message is in MIME format.  The first part should be readable text,
  while the remaining parts are likely unreadable without MIME-aware tools.
  Send mail to mime@docserver.cac.washington.edu for more info.

--11353260-15213-885858118=:-183571
Content-Type: TEXT/PLAIN; charset=US-ASCII

Here are the somewhat lengthy minutes of the AVT meeting at the 40th
IETF in Washington, DC in December.  Any comments or corrections are
welcome.
							-- Steve

--11353260-15213-885858118=:-183571
Content-Type: TEXT/PLAIN; charset=US-ASCII; name="avt/minutes.97dec"
Content-Transfer-Encoding: BASE64
Content-ID: <Pine.WNT.3.96.980126154158.-183571C@oak.precept.com>
Content-Description: Minutes of AVT meeting

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--11353260-15213-885858118=:-183571--



From rem-conf Mon Jan 26 21:21:04 1998 
From rem-conf-request@es.net Mon Jan 26 21:21:04 1998
Received: from list by mail1.es.net with local (Exim 1.81 #2)
	id 0xx3O7-00049W-00; Mon, 26 Jan 1998 21:16:43 -0800
Date: Mon, 26 Jan 1998 21:16:12 -0800 (PST)
From: "T. Todd Elvins" <todd@SDSC.EDU>
To: scwpf@sdsc.edu, web-watchers -- K Claffy <kc@sdsc.edu>, rem-conf@es.net,
        webteam@qualcomm.com, webheads@sio.ucsd.edu, java-ucsd@mib.org,
        flesner@nosc.mil, talks@ucsd.edu, cse-talks@cs.UCSD.EDU
Cc: Greg Johnson <johnson@sdsc.edu>
Subject: Wednesday night Forum - Reminder - MBONE Broadcast
Message-Id: <Pine.SGI.3.96.980126210323.3696B-100000@jessicarabbit.sdsc.edu>
Mime-Version: 1.0
Content-Type: TEXT/PLAIN; charset=US-ASCII
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Southern California Web Programmers Forum
Meeting Agenda
Date: January 28, 1998
Time: 7:00PM
San Diego Supercomputer Center

Invited Speakers

Kelly Looney (VP of Marketing) and Mark Gaither (Senior Technical Staff)
>from Activerse, Inc.

	"The Most Interesting Thing on the Internet: People"

Realtime network presence is changing the Internet. 

	"The Web's primary focus has been one of people accessing 
	information. Over the next 12 months, look for the first signs of a
	fundamental shift. Instead of people accessing information, the Web's 
	focus will shift toward people accessing other people in an 
	information-rich environment". 
		-- Paul Saffo, Director of the Institute for the Future

Kelly and Mark will present two
new software products, Ding! and Ding! Switchboard. Ding! was chosen
as one of the "125 best software titles available in the market today" in
the Winter '98 issue of Your New PC, a special publication from the editors of
PC Magazine. These two products which constitute the first
all-Java Internet/intranet instant collaboration system. 


Speaker #2

T. Todd Elvins (Staff Scientist)
>from the San Diego Supercomputer Center SD Bay Project

	"Using Java to Control VRML"

Todd will demonstrate a 3D GIS application constructed
using a Java applet to trigger changes in VRML content.
Java and VRML were integrated using the Extended Authoring Interface (EAI)
to create BAYVIEW, a 3D virtual environmental.


==========================================================
For parking and directions see   http://www.sdsc.edu/scwpf


-----------------------------------------------------------------
T. Todd Elvins,  Ph.D.				    todd@sdsc.edu
Science Department, San Diego Supercomputer Center			   
University of California, San Diego		   (619) 534-5128	   
P.O. Box 85608				  FAX  	   (619) 534-5117
San Diego, CA 92186-9784               http://www.sdsc.edu/~todd/
-----------------------------------------------------------------




From rem-conf Tue Jan 27 00:48:29 1998 
From rem-conf-request@es.net Tue Jan 27 00:48:29 1998
Received: from list by mail1.es.net with local (Exim 1.81 #2)
	id 0xx6dt-0005Kz-00; Tue, 27 Jan 1998 00:45:13 -0800
X-Authentication-Warning: listbox2.cern.ch: Host pcitcs04.cern.ch [137.138.33.126] claimed to be cern.ch
Message-ID: <34CD9E3C.8D486F5D@cern.ch>
Date: Tue, 27 Jan 1998 09:43:40 +0100
From: Miguel CHAMOCHIN GOMEZ <Miguel.Chamochin@cern.ch>
Organization: CERN
X-Mailer: Mozilla 4.04 [en] (Win95; I)
MIME-Version: 1.0
To: Scott Petrack <Scott_Petrack@vocaltec.com>,
        "rem-conf@es.net" <rem-conf@es.net>
Subject: Re: Is there a tool for exact recording of received RTP streams?
References: <42256598.00554AA1.00@il4.vocaltec.co.il>
Content-Type: text/plain; charset=us-ascii
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Scott Petrack wrote:
> 
> Is there a tool (public domain or otherwise) which stores a received RTP
> stream on disk, including the precise time at which each packet was
> received, in such a way that the receipt of the RTP can be exactly
> "re-enacted" at a later point in time (inlcuding all the jitter, etc.).

   See http://csvod1.cern.ch/ "wrtp project". Wrtp application maintain
a index file with that information.

   Best regards.

-- 
         Miguel Angel Chamochin Gomez
         Information Technology Division
         CERN   CH-1211 Geneve 23  SWITZERLAND
         Phone: +41-22-767-9703
         Fax: +41-22-767-7155
         e-mail: mangel@mail.cern.ch
                 Miguel.Chamochin@cern.ch

                                 ---
"Would you tell me, please, which way I ought to go from here?"
        "That depends a good deal on where you want to get to."
                            - Lewis Carrol, Alice in Wonderland
                                -----



From rem-conf Tue Jan 27 10:23:49 1998 
From rem-conf-request@es.net Tue Jan 27 10:23:49 1998
Received: from list by mail1.es.net with local (Exim 1.81 #2)
	id 0xxFQY-0000k8-00; Tue, 27 Jan 1998 10:08:02 -0800
Message-ID: <19980127100754.45360@caida.org>
Date: Tue, 27 Jan 1998 10:07:54 -0800
From: k claffy <kc@caida.org>
To: rem-conf@es.net
Cc: jay@sdsc.edu, hutton@sdsc.edu, gjohnson@sdsc.edu,
        Tracie Monk <tmonk@nlanr.net>, don <dmitchel@nsf.gov>
Subject: Stephen Spielberg's Survivors of the Shoah Visual History Foundation" - 1/27/98 Presentation
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(will say 'shoah foundation' in sdr)



DIGITAL ASSET MANAGEMENT AT STEPHEN
SPIELBERG'S SURVIVORS OF THE SHOAH
VISUAL HISTORY FOUNDATION:



University of California, San Diego:  On Tuesday, January 27, from
4:00 to 6:00 p.m. in the Auditorium of the San Diego
Supercomputer Center on the UCSD campus, the National
Laboratory for Advanced Network Research (NLANR) will sponsor
a presentation on the development of advanced technology to obtain
and preserve the testimonies of tens of thousands of survivors of the
Holocaust all over the world.  The presentation is titled

Digital Asset Management
at the
Survivors of the Shoah Visual History Foundation

The speaker will be Sam Gustman, Director of Technology at the
Shoah Foundation.  He will demonstrate the advanced technology
being used for the Foundation's undertaking.

"Shoah" is the Hebrew word for the Holocaust.  The Survivors of
the Shoah Visual History Foundation was founded by Hollywood
director Steven Spielberg in 1994.  It is a nonprofit organization
dedicated to videotaping and archiving interviews of Holocaust
survivors all over the world.

The purpose of collecting the testimonies is to foster the teaching of
racial, ethnic, and cultural tolerance in schools and public
organizations around the world.  More than 40,000 testimonies have
already been collected on videotape in 29 different languages,
comprising the collective experience of Jews and non-Jews alike
who emerged from Hitler's concentration camps and from hiding
places all over Europe at the end of World War II.

Gustman and the staff of the Shoah Foundation are putting together
the complex system for scheduling, digitizing, cataloguing, and
distributing the testimonies.  "At present," he says, "we expect the
archive to grow to more than 150 terabytes of MPEG-encoded
video.  We will deliver these testimonies over proprietary and
government ATM networks to five Holocaust memorial institutions
around the world, which will serve as primary repositories.  We are
eager to share our plans with the best developers and leaders in
advanced digital technology and data archiving, and we are seeking
collaborations with them to help make our repository permanently
and easily available to people all over the world."

Gustman notes that the Foundation works with the world's leading
Holocaust museums, educators, archivists, and documentary film
makers.  The compilation will be the most comprehensive library of
survivor testimony ever assembled.  "By preserving the eyewitness
testimonies of tens of thousands of Holocaust survivors, the
Foundation will enable future generations to learn the lessons of this
devastating period in human history from those who survived,"
Gustman says.

To access the Shoah Foundation on the web, see http://www.vhf.org.



From rem-conf Tue Jan 27 11:18:08 1998 
From rem-conf-request@es.net Tue Jan 27 11:18:07 1998
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Date: Tue, 27 Jan 1998 11:14:13 -0800 (Pacific Standard Time)
From: Stephen Casner <casner@precept.com>
To: rem-conf@es.net
Subject: Re: Minutes of AVT meeting in Washington DC
In-Reply-To: <Pine.WNT.3.96.980126153809.-183571B-200000@oak.precept.com>
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I am not sure why my text/plain attachment of the AVT meeting minutes
was encoded BASE64.  Since that has caused trouble for some people,
here's a URL instead:

    ftp://hydra.precept.com/pub/rtp/minutes.97dec

> Here are the somewhat lengthy minutes of the AVT meeting at the 40th
> IETF in Washington, DC in December.  Any comments or corrections are
> welcome.
> 							-- Steve




From rem-conf Wed Jan 28 00:30:43 1998 
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From rem-conf Wed Jan 28 08:10:35 1998 
From rem-conf-request@es.net Wed Jan 28 08:10:34 1998
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	id 0xxZuO-0001Hr-00; Wed, 28 Jan 1998 08:00:12 -0800
From: Tom Zoerner <Tom.Zoerner@informatik.uni-erlangen.de>
Message-Id: <199801281600.RAA25466@faui01.informatik.uni-erlangen.de>
Subject: FEC draft and timestamp recovery
To: rem-conf@es.net
Date: Wed, 28 Jan 1998 17:00:06 +0100 (MET)
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Hi.

I'm still having trouble with the way the FEC draft handles timestamps.
Specifically with the fact that FEC masks operate on sequence numbers;
when a packet of a certain payload type is missing, I don't know it's
sequence number though - in general I only know the timestamp. That however
is not directly recoverable by FEC packets. The draft says in chapter 5.3
Recovery Procedures: "The timestamp for xi is computed by any reasonable
approximation, at the discretion of the implementor."  Is this meant to
suggest that this is trivial?  Because I really don't think so:

Let's assume the last packet I received of my main payload type had
sequence number n and timestamp ts. Let's also assume I haven't received
any later packets from that payload type, so there's no way to guess the
sequence numer of the missing packet. It could be n+1, but also n+2 if
there was a FEC or other payload type inbetween (that also got lost).

Now let's assume I had received a FEC packet n+3 with mask 0x1010 and
timestamp ts. So I know that packet encodes packets n and n+2 - but how
do I know if n+2 contained ts+td or ts+2*td?

In case of audio it would be quite important to know the timestamp,
since it directly determines the position in the playout queue. In the
above scenario: at ts+td, do I have to do error concealment for a
missing packet (n+1), or do I immediately play out the recovered n+2.

Maybe we do need that XOR of all timestamps afterall? Or simply disallow
all masks with gaps, i.e. zeroes inbetween ones in the mask (that'd be a
serious restriction)

-tom



From rem-conf Wed Jan 28 10:46:40 1998 
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	id 0xxcGe-0002s3-00; Wed, 28 Jan 1998 10:31:20 -0800
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Date: Wed, 28 Jan 1998 13:35:48 -0800
From: "Andrew T.Campbell" <campbell@ctr.columbia.edu>
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Organization: Center for Telecommunications Research, Columbia University
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        commsoft@cc.bellcore.com, multicomm@cc.bellcore.com
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First IEEE Conference on Open Architectures and Network Programming

San Francisco, CA, USA, April 3-4, 1998 

http://comet.columbia.edu/openarch/ 


PROGRAM AT A GLANCE:

KEYNOTES:

Mike Nelson, Federal Communications Commission
Bill Edwards, MCI Worldcom  

INVITED SESSIONS:

Research Frontiers in OPEN Architectures 
-- Speaker: Louis-Francois Pau, Ericsson Utvecklings
 
Research Frontiers in Network Programming
-- Speaker: David L. Tennenhouse, DARPA and MIT 

TECHNICAL SESSIONS:

Architectures and Performance 
-- Chair: Chuck Kalmanek, AT&T Labs - Research

Open Signalling 
-- Chair: Giovanni Pacifici, IBM T.J. Watson Research Center

Open Service Management 
-- Chair: Amit Gupta, Sun Microsystems

Protocols 
-- Chair: Andrew Thomas, HP Labs 

Work in Progress 
-- Chair: Tom Laporta, Bell Labs - Lucent Technologies

Active Networks  
-- Chair: Raj Yavatkar, Intel

CONFERENCE PANEL:

Active Networks and/or Broadband Kernels? 
-- Chair: Aurel A. Lazar, Columbia University

FOR FULL DETAILS OF THE OPENARCH'98 PROGRAM AND REGISTRATION: 

http://comet.columbia.edu/openarch/



From rem-conf Wed Jan 28 11:06:27 1998 
From rem-conf-request@es.net Wed Jan 28 11:06:26 1998
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Date: Wed, 28 Jan 1998 13:56:46 -0500
From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
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Tom Zoerner wrote:

> I'm still having trouble with the way the FEC draft handles timestamps.
> Specifically with the fact that FEC masks operate on sequence numbers;
> when a packet of a certain payload type is missing, I don't know it's
> sequence number though - in general I only know the timestamp. That however
> is not directly recoverable by FEC packets. The draft says in chapter 5.3
> Recovery Procedures: "The timestamp for xi is computed by any reasonable
> approximation, at the discretion of the implementor."  Is this meant to
> suggest that this is trivial?  Because I really don't think so:
> 
> Let's assume the last packet I received of my main payload type had
> sequence number n and timestamp ts. Let's also assume I haven't received
> any later packets from that payload type, so there's no way to guess the
> sequence numer of the missing packet. It could be n+1, but also n+2 if
> there was a FEC or other payload type inbetween (that also got lost).
> 
> Now let's assume I had received a FEC packet n+3 with mask 0x1010 and
> timestamp ts. So I know that packet encodes packets n and n+2 - but how
> do I know if n+2 contained ts+td or ts+2*td?

Excellent point. In the original draft, the sequence number space for
the media packets was distinct. The FEC packets had a SN which was the
minimum of the SN in the media packets it protected. Because of this,
you could compute the timestamp for a recovered media packet by
interpolating from adjacent packets. The change to using the same
sequence space has clearly broken this.

> Maybe we do need that XOR of all timestamps afterall? Or simply disallow
> all masks with gaps, i.e. zeroes inbetween ones in the mask (that'd be a
> serious restriction)

It seems there are a number of deficiencies as currently defined with
the current SN/TS method:

1. You cannot recover the TS of the packet, as you point out, which is
needed for playout.
2. You must send the original media packets (systematic codes only),
eliminating some of the schemes in Budge's draft.
3. In multicast, a non-FEC receiver will discard the FEC packets since
the PT are unknown. However, it will probably be confused, since it will
see gaps in the sequence number space, and think there is a loss of real
data. However, if it uses the timestamps to fill-in the required amount
of missing data, it will find that there are no gaps. This is a
consequence of the media sequence number space being "interrupted" by
FEC.
4. There is no clean way to send the FEC on a separate multicast group.
This is because its SN space is intermingled with the media. If you send
the FEC on a separate group, its probably a good idea for it to have its
own SN space. 

I see several possible fixes:

1. Include an additional 32 bit field in the FEC header for timestamp
reconstruction, as you suggest. This fixes (1), but none of the others.
2. Go back to the original proposal. This proposal specified that the SN
in the FEC was the minimum of the media packets is was protecting. The
TS was the xor of the TS of the packets it was protecting. This fixes
(1) and (2), but having packets show up with non-sequential,duplicated
SN will cause confusion. It also has problems with header compression.
3. Require the FEC to always be sent as a separate stream. It uses its
own sequence number space. The FEC header contains an additional 16 bit
SN base field, which indicates the base that the mask is applied to. The
SN that the base/mask are referring to are those for the media packets.
The media packets are sent completely normally. The TS in the FEC
packets is set to the minimum TS of all packets it protects. We can also
add a 32 bit TS reconstruction field into the FEC header, which is the
xor of the TS in the media packets. This seems to solve all four
problems above. It should also work well for RTP compression, since this
way the FEC appear as a separate context. You can even still include the
FEC data piggybacked onto the media using the redundant codec payload
format, if you like.

I am tending to lean towards solution (3). 

-Jonathan R.

-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
PHONE: (732) 949-6418                       Rm. 4C-526
FAX:   (732) 834-5379
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen



From rem-conf Wed Jan 28 11:07:02 1998 
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Date: Wed, 28 Jan 1998 11:02:36 -0800 (Pacific Standard Time)
From: Stephen Casner <casner@precept.com>
To: rem-conf@es.net
Subject: Working group last call on BT.656 and JPEG
Message-ID: <Pine.WNT.3.96.980128110200.-4125059D-100000@oak.precept.com>
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To the AVT working group:

Two RTP payload format Internet-Drafts were presented at the
Washington IETF meeting and were said to be ready to go to working
group last call.  Therefore, this is a request for last call for
comments on:

    draft-tynan-rtp-bt656-01.txt for BT.656-3 uncompressed video
    draft-ietf-avt-jpeg-new-00.txt and .ps for motion-JPEG video

The first is a new payload format; the second is a revision of
RFC2035 which makes functional changes (improvements!), so it will
"cycle in grade".  Each of these drafts needs a small addition before
publication:  the BT656 draft needs to have a security considerations
section added (can be copied from anoter recent payload format draft),
and the JPEG draft needs to have a "changes" section or appendix to
summarize the changes (since it is already on the standards track).
Since these are simple additions, I think that the working group last
call can proceed without them and then any comments can be folded into
a revision to go to the IESG.

Barring any problems with these drafts, in two weeks I will request
IESG Last Call for publication of these drafts as Proposed Standards.

							-- Steve




From rem-conf Wed Jan 28 15:02:00 1998 
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From:	keith@comm.toronto.edu (Keith Chow)
Message-Id: <199801282256.RAA18042@tesla.comm.toronto.edu>
Subject: Question on RTP availability.
To:	rem-conf@es.net
Date:	Wed, 28 Jan 1998 17:56:18 -0500
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Hi,

Is there any work going on providing RTP as a standard protocol with the
OS? Linux? Solaris? Irix? or even Windows?

I believe it would be very helpful in developing application.
But by doing so, we might need to modify the existing Socket API.

Any comment?

Keith.

-----------------------------------------------
Keith HungKei Chow
Network Architecture Lab, Communication Group,
Department of Electrical & Computer Engineering
University of Toronto
Ontario, Canada.
-----------------------------------------------




From rem-conf Thu Jan 29 01:23:27 1998 
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To: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
cc: Tom Zoerner <Tom.Zoerner@informatik.uni-erlangen.de>, rem-conf@es.net
Subject: Re: FEC draft and timestamp recovery
In-reply-to: Your message of "Wed, 28 Jan 1998 13:56:46 EST." <34CF7F6E.FE25517D@dnrc.bell-labs.com>
Date: Thu, 29 Jan 1998 09:07:18 +0000
Message-ID: <706.886064838@cs.ucl.ac.uk>
From: Orion Hodson <O.Hodson@cs.ucl.ac.uk>
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<34CF7F6E.FE25517D@dnrc.bell-labs.com>Jonathan Rosenberg writes:
> Tom Zoerner wrote:
> 
> > I'm still having trouble with the way the FEC draft handles timestamps.
> > Specifically with the fact that FEC masks operate on sequence numbers;
> > when a packet of a certain payload type is missing, I don't know it's
> > sequence number though - in general I only know the timestamp. That however
> > is not directly recoverable by FEC packets. The draft says in chapter 5.3
> > Recovery Procedures: "The timestamp for xi is computed by any reasonable
> > approximation, at the discretion of the implementor."  Is this meant to
> > suggest that this is trivial?  Because I really don't think so:
> > 
> > Let's assume the last packet I received of my main payload type had
> > sequence number n and timestamp ts. Let's also assume I haven't received
> > any later packets from that payload type, so there's no way to guess the
> > sequence numer of the missing packet. It could be n+1, but also n+2 if
> > there was a FEC or other payload type inbetween (that also got lost).
> > 
> > Now let's assume I had received a FEC packet n+3 with mask 0x1010 and
> > timestamp ts. So I know that packet encodes packets n and n+2 - but how
> > do I know if n+2 contained ts+td or ts+2*td?
> 
> Excellent point. In the original draft, the sequence number space for
> the media packets was distinct. The FEC packets had a SN which was the
> minimum of the SN in the media packets it protected. Because of this,
> you could compute the timestamp for a recovered media packet by
> interpolating from adjacent packets. The change to using the same
> sequence space has clearly broken this.

For simple cases you can get around this problem by observation, if
you take an fec coded stream you can reasonably assume that the media
data and fec data occurs at predictable intervals.  As long as you are
not still observing the first cycle of media/fec data then you can
predict what n+1 and n+2 should have been.  

Where this fails is where your fec arises from a joint/source coding,
i.e where your fec arises after some classification of your media
stream and is no longer predictably placed.  

> It seems there are a number of deficiencies as currently defined with
> the current SN/TS method:
> 
> 1. You cannot recover the TS of the packet, as you point out, which is
> needed for playout.
> 2. You must send the original media packets (systematic codes only),
> eliminating some of the schemes in Budge's draft.
> 3. In multicast, a non-FEC receiver will discard the FEC packets since
> the PT are unknown. However, it will probably be confused, since it will
> see gaps in the sequence number space, and think there is a loss of real
> data. However, if it uses the timestamps to fill-in the required amount
> of missing data, it will find that there are no gaps. This is a
> consequence of the media sequence number space being "interrupted" by
> FEC.
> 4. There is no clean way to send the FEC on a separate multicast group.
> This is because its SN space is intermingled with the media. If you send
> the FEC on a separate group, its probably a good idea for it to have its
> own SN space. 
> 
> I see several possible fixes:
> 
> 1. Include an additional 32 bit field in the FEC header for timestamp
> reconstruction, as you suggest. This fixes (1), but none of the others.
> 2. Go back to the original proposal. This proposal specified that the SN
> in the FEC was the minimum of the media packets is was protecting. The
> TS was the xor of the TS of the packets it was protecting. This fixes
> (1) and (2), but having packets show up with non-sequential,duplicated
> SN will cause confusion. It also has problems with header compression.
> 3. Require the FEC to always be sent as a separate stream. It uses its
> own sequence number space. The FEC header contains an additional 16 bit
> SN base field, which indicates the base that the mask is applied to. The
> SN that the base/mask are referring to are those for the media packets.
> The media packets are sent completely normally. The TS in the FEC
> packets is set to the minimum TS of all packets it protects. We can also
> add a 32 bit TS reconstruction field into the FEC header, which is the
> xor of the TS in the media packets. This seems to solve all four
> problems above. It should also work well for RTP compression, since this
> way the FEC appear as a separate context. You can even still include the
> FEC data piggybacked onto the media using the redundant codec payload
> format, if you like.
> 
> I am tending to lean towards solution (3). 

I favour (3).

- Orion

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



From rem-conf Thu Jan 29 05:26:49 1998 
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	id 0xxtq3-0003ML-00; Thu, 29 Jan 1998 05:17:03 -0800
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Date: Thu, 29 Jan 1998 08:16:54 -0500 (EST)
From: Michael Kounavis <mk@comet.columbia.edu>
To: rem-conf@es.net
cc: mnl@comet.columbia.edu
Subject: MBone Broadcast announcement
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MBone Broadcast Announcement

Title:
  Scalable bandwidth allocation in the Internet   

Speaker:
   Zheng Wang, Bell Labs Lucent Technologies


Date:
    Thursday, Jan. 29, 1997

Time:
    11:00 AM - 12:00 AM EST

Multicast Address:
    audio:  DVI, RTP, 224.2.177.25/31716 , ttl=127
    
Contact:
    Michael Kounavis <mk@comet.columbia.edu>

URL:
    http://www.comet.columbia.edu/activities/seminars/spring98

Place:
    Interschool Laboratory
    Schapiro Research Building
    Center for Telecommunications Research
    Columbia University
    New York City, USA

Abstract:

In this talk, I present a new differentiated service scheme called
"User-Share Differentiation (USD)". The USD scheme is based on the
principle of link sharing. The scheme allows ISPs to differentiate
traffic flows on a per-user basis, providing virtual leased lines
and proportional fair sharing. I first look at the lessons we
learnt from the best effor and int-serv models, and the problems 
with the current proposals, then present the details of the USD scheme.
Finally I examine the implementation and standardization issues


-----------------------------------------------------------------------
Michael E. Kounavis       e-mail   : mk@comet.columbia.edu
Room 801, CEPSR           www      : http://www.comet.columbia.edu/~mk/
Columbia University       office   : (212) 854-6900      
New York, NY 10025        lab      : (212) 854-5599               
-----------------------------------------------------------------------




From rem-conf Thu Jan 29 05:32:35 1998 
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Date: Thu, 29 Jan 1998 08:26:54 -0500
From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
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Orion Hodson wrote:
> 
> 
> > you could compute the timestamp for a recovered media packet by
> > interpolating from adjacent packets. The change to using the same
> > sequence space has clearly broken this.
> 
> For simple cases you can get around this problem by observation, if
> you take an fec coded stream you can reasonably assume that the media
> data and fec data occurs at predictable intervals.  As long as you are
> not still observing the first cycle of media/fec data then you can
> predict what n+1 and n+2 should have been.
> 
> Where this fails is where your fec arises from a joint/source coding,
> i.e where your fec arises after some classification of your media
> stream and is no longer predictably placed.

Packet losses may also make this kind of classification hard. The
reconstruction mechanisms should be "stateless" I think; stateless here
meaning that a single packet contains sufficient information to tell you
what to do with it, without requiring other packets to direct behavior.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
PHONE: (732) 949-6418                       Rm. 4C-526
FAX:   (732) 834-5379
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen



From rem-conf Thu Jan 29 06:32:17 1998 
From rem-conf-request@es.net Thu Jan 29 06:32:16 1998
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	id 0xxuvT-0004SP-00; Thu, 29 Jan 1998 06:26:43 -0800
Message-Id: <199801291425.GAA26217@iceland.it.earthlink.net>
Subject: Fwd: Re: [MC] dog the wag
Date: Thu, 29 Jan 98 06:35:48 -0000
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From: ALICE  <itsalice@earthlink.net>
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---------------- Begin Forwarded Message ----------------
Date:        01/29  11:30 AM
Received:    01/29  5:38 AM
From:        Kite-a-holic, revcoal@connix.com
To:          Francis L. Regan, billywompus@juno.com
CC:          mindcontrol-l@sonic.net

On Wed, 28 Jan 1998, Francis L. Regan wrote:
>In the new movie, Wag the Dog, spin doctors create a war to distract
>attention from a sex scandal involving their president.  People have said
>it is interesting how filmmakers could have anticipated a sex scandal and
>an impending war and finished  a movie before any of it broke in the
>news.  However, something else might be going on.  These guys  (I hate
>not citing my sources, but i'm in kind of a hurry and can't remember)

You CAN'T REMEMBER people who claim to know the President and made
substantial campaign contributions?????


>know the president and have donated decent amounts of money to  his
>campaign.  What if the president is planning a war with Iraq as the spark
>for world war III, and needs a distraction until it can get off the
>ground?  He could have asked his friends to make a movie suggesting the
>opposite.  This would also help explain the first lady's lack of flipping
>out about this.

I was thinking this yesterday, first off in the morning when NBC News was
reporting that the U.S. is seriously considering using nuclear weapons
against Iraq, and again last night when it was reported that Congress
'just happens' to be voting today (Thursday) on whether to back Clinton
in his 'Iraq initiative', and it was stated it WOULD vote to support
Clinton by an overwhelming majority...

At that point, I became sure that this Lewinsky thing is the smoke and
mirrors to divert the public's attention AWAY from the obvious steps
being taken to initiate a war...


June  :-7
 
          /\
          \/
         /          
    Q   /
    |-+/
   / \          
        --ahjt

 Go fly a kite ! It's fun!


X-NO-ARCHIVE: YES

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From rem-conf Thu Jan 29 06:34:19 1998 
From rem-conf-request@es.net Thu Jan 29 06:34:18 1998
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	id 0xxuzZ-0004Y2-00; Thu, 29 Jan 1998 06:30:57 -0800
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To: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
cc: Orion Hodson <O.Hodson@cs.ucl.ac.uk>,
        Tom Zoerner <Tom.Zoerner@informatik.uni-erlangen.de>, rem-conf@es.net
Subject: Re: FEC draft and timestamp recovery 
In-reply-to: Your message of "Thu, 29 Jan 1998 08:26:54 EST."
             <34D0839E.8B170A9D@dnrc.bell-labs.com> 
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Date: Thu, 29 Jan 1998 09:29:28 -0500
From: Ken Carlberg <carlberg@time.saic.com>
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> 3. Require the FEC to always be sent as a separate stream. 

perhaps i missed something in the thread, but the above statement
makes me think that one produces a domino effect in that a new problem
arises in having to ensure that separate streams receive the same 
service -- e.g., end-to-end delay bounds. 

feel free to correct me, but an important attribute of FEC packets
is that they arrive when the original packets are received.  but
this problem gets hard(er) when the streams are traversing transit
networks that operate under best effort service and yet use a variety
of queueing and congestion avoidance strategies (CBQ, W-RED, WFQ, 
F-RED, etc.).  

one could provide a counter arguement that integrated, and perhaps
differentiated (depending on how it is realized), services can deal
with this problem.  but that assumes that these services will be 
populated throughout the internet.  from a more extreme perspective,
leaf subnets that have (TDMA/FDMA) links would probably still have
a hard time 'synchronizing' streams whose packets vary widely in size.

please understand, i think that there are some appealing aspects 
about placing the fec packets in another stream. but i guess what 
i'm wondering aloud is...are we setting ourselves up for another 
problem if we use what amounts to an out-of-band means of sending 
fec packets?

just my $0.02,

-ken





From rem-conf Thu Jan 29 08:02:20 1998 
From rem-conf-request@es.net Thu Jan 29 08:02:19 1998
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Date: Thu, 29 Jan 1998 10:51:20 -0500
From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
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CC: Orion Hodson <O.Hodson@cs.ucl.ac.uk>,
        Tom Zoerner <Tom.Zoerner@informatik.uni-erlangen.de>, rem-conf@es.net
Subject: Re: FEC draft and timestamp recovery
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Ken Carlberg wrote:
> 
> > 3. Require the FEC to always be sent as a separate stream.
> 
> perhaps i missed something in the thread, but the above statement
> makes me think that one produces a domino effect in that a new problem
> arises in having to ensure that separate streams receive the same
> service -- e.g., end-to-end delay bounds.
> 
> feel free to correct me, but an important attribute of FEC packets
> is that they arrive when the original packets are received.  but
> this problem gets hard(er) when the streams are traversing transit
> networks that operate under best effort service and yet use a variety
> of queueing and congestion avoidance strategies (CBQ, W-RED, WFQ,
> F-RED, etc.).
> 

With the current best effort service model, packets may arrive late,
early, out of order, or never. Playout buffers and such things are used
to deal with these problems. Thus, the current working assumption is
that there are no "bounds" on the delay jitter between FEC and data
packets anyway. 

I don't think the service you would get from W-RED, RED, or F-RED would
be substantially different. All of those are still FCFS, so the delay
jitter and reordering should be the same as FIFO. Loss rates may be
different, but for things like F-RED they should be better, not worse.
The only problem I see is if you had a WFQ system that isolated traffic
based on destination ports. In that case, the delay jitter could be
worse. However, I'm not really sure how much worse it would be compared
to a congested network with best-effort FIFO service. 

In any case, the problem you point out will happen when you send audio
and video separately anyway. Hosts will have to implement sufficient
playout buffering to allow for inter-media synchronization. This buffer
should also then be sufficient for the FEC mechanisms. 

Proposal (3) still allowed you to send the FEC and data in the same
packets, however, using the redundant audio codecs. So, if you were
really worried about the problem, you could do that.

-Jonathan R. 
-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
PHONE: (732) 949-6418                       Rm. 4C-526
FAX:   (732) 834-5379
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen



From rem-conf Thu Jan 29 09:49:30 1998 
From rem-conf-request@es.net Thu Jan 29 09:49:29 1998
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	id 0xxxxf-0007ZL-00; Thu, 29 Jan 1998 09:41:11 -0800
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Date: Thu, 29 Jan 1998 10:36:40 -0700
To: "...." <rem-conf@es.net>
From: James <joh@mail.arc.nasa.gov>
Subject: SUBCRIBE
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Please add me to your list..i wish to subscribe to MBone annoucements.

james. Oh





From rem-conf Thu Jan 29 23:30:27 1998 
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Date: Fri, 30 Jan 1998 17:19:50 +1000
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From: Liz Armstrong <liz@dstc.edu.au>
Subject: WWW7 newsletter
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Hello

Please find following the latest WWW7 newsletter.

The WWW7 Consortium appreciates your on-going support of the Conference and
thanks you
for taking the time to distribute our updates to your friends and colleagues.

Regards
Liz Armstrong for
WWW7 Consortium

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~


WWW7 Electronic Newsletter - January 1998

A regular email newsletter sent out to over 5,000 people around the
world who are interested in WWW7.  Feel free to forward this newsletter
to your associates.

To unsubscribe from this mailing, simply forward this entire newsletter to
info@www7.conf.au and type Unsubscribe in the subject line of the email.


************************************************************************
A. Developer's Day

B. What will be of particular interest to the cultural sector?

C. Selection of WWW7 papers, posters, tutorial and workshops now finalised

D. Workshops at WWW7

E. Conference Contact Details
************************************************************************

A.      Developers' Day

 Developers' Day will be on Saturday 18 April.  The keynote speaker will
 be James Gosling, Vice President, Sun Microsystems and the father
 of Java - the most revolutionary programming language in years and the
 most empowering language for the Web.  James will talk on the future of
 Web development and the role that Java will play.

Developers' day will include the following confirmed technology tracks:

	Style Sheets
	Markup - XML, HTML, SGML and more
	Java
	Internationalisation
	Resource Description Framework and Metadata 
	Distributed Objects (CORBA) on the Web

More tracks are under development including HTTP, Push Technologies
and Ecommerce.

For more info, visit the WWW7 home page or email <developer@www7.conf.au>


B.  What will be of particular interest to the cultural sector?

 This stream will be working alongside the main conference, with a strong
 focus on the interaction between the emerging technologies and the needs
 and aspirations of the cultural community.  In the special section,
 a physical collection will be converted to a digital collection using
 the latest techniques and equipment.  Those who make the equipment
 and use it will be available to talk to you about the good and bad
 aspects of this process.  You will be able to consider photography,
 VR, textual descriptions, cataloging issues, archival issues and more.
 All these practical problems for the community will be brought to air
 in a context where the developers and users can interact.

 Each day of the main conference there will be three streams from which
 you can choose:

 the usual WWW7 technical papers, working sessions for cultural
 institutional policy and decision-makers, and working sessions for
 those who develop content for cultural institutions, both derivative
 and original digital material.

 On tutorials day (Tuesday, the day before the conference starts), there
 will be tutorials for you.  The first day of the conference, Wednesday,
 will introduce the issues and links to be made with the technical part
 of the conference.  The second day, Thursday, will be particularly
 of value to those who work in museums, with the emphasis on visiting
 cultural institutions - esp. in cyberspace.  The third day, Friday,
 will be of particularly concerned with work in the area of collection
 description - it will be International Metadata Day, a sequel Metadata
 Day held at the National Library of Australia following DC4 in Canberra
 in 1997. Developers' day will extend all this into the development level
 for those with technical expertise.

 For more information about the cultural sector activities, please email:
 www7culture@srl.rmit.edu.au

or contact:

	WWW7 Conference Cultural Sector Activities
		c/- Sunrise at RMIT
		GPO Box 2476V
		Melbourne 3001  VIC  Australia

	Phone +613 9660 3024
	Fax +613 9660 2761


C. Selection of WWW7 papers, posters, tutorial and workshops now finalised.

 The Program Committee were impressed by the number and quality of papers
 and posters submitted. We received 236 submissions from which we selected
 54 papers, 12 posters and 48 short papers.

 The geographic distribution of papers was also remarkable - we were
 pleased to receive submissions from 24 countries. Australasia will be
 well-represented with papers and short papers from Australia, Japan,
 Singapore, Taiwan and Korea. However we have many good papers and short
 papers from the rest of the world, with the USA, the UK, Germany, Sweden,
 Italy, Denmark, the Netherlands, Switzerland, Israel, Greece and Austria
 all represented in the program.

 The topic range was also particularly impressive. We received submissions
 in every one of the topic areas listed in the Call for Papers, with
 a large number papers on topical areas such as Information retrieval
 and modeling, search and indexing techniques, browsers and tools,
 human-computer interaction, applications in education and training,
 metadata systems and many more.

 In addition to the refereed papers and short papers, the conference will
 also devote a track to the activities of the World Wide Web consortium
 (W3C) motivate much of the research on Web technologies and social
 issues. Of course, one of the conference keynote speakers is Tim
 Berners-Lee, co-founder of the Web and Director of the W3C.
 

D.  Workshops at WWW7 - Opportunities for Staff Development

 WWW7 begins with a Workshop and Tutorial Day on April 14, 1998.
 A slate of 11 workshops afford conference delegates the opportunity
 to work together to move Web technology and applications forward.
 The spectrum of areas covered by the conference workshops demonstrates
 that the WWW7 is truly the gathering place for anyone interested in the
 World Wide Web. Some workshops focus on advancing the base technology
 with such topics as HTTP, XSL and XML, and, the emerging area of Web
 engineering. Other workshops focus on important application areas,
 such as libraries and on-line training and education.  A third set of
 workshops focus on pressing social issues such as disability access and
 communication within agricultural communities.

 Workshops provide a stimulating opportunity for participants to pursue the
 issues and ideas associated with their particular interest at a gathering
 of people with similar interests. This is an invaluable opportunity
 for networking and for establishing on-going collaboration. Indeed,
 two of the workshops --  The International Flexible Hypertext Workshop
 and The Hypertext Functionality Workshop -- are part of ongoing series
 of workshops on these topics.

 The Web community benefits from the ongoing work accomplished and
 the new initiatives started during the workshops and the workshops
 certainly reward those who get involved.  The starting point for
 such involvement is the workshop page of the conference Web site,
 http://www7.conf.au/workshop.html


E.  Conference Contact Details

	WWW7 Conference Secretariat
		PO Box WWW7
		St Lucia South
		Queensland Australia 4067

	Phone +61 7 3870 8831
	Fax +61 7 3371 9514
	Email :info@www7.conf.au
	Web:  http://www7.conf.au



7th International World Wide Web Conference
14-18 April 1998
Brisbane Convention & Exhibition Centre
Brisbane, Australia







