From rem-conf-request@es.net Wed Jan 01 11:10:53 1997 
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To: rem-conf@es.net
From: Ross Finlayson <finlayson@lvn.com>
Subject: New multicast session browser available for testing
Cc: multimedia@freebsd.org, karl@best.com

Happy New Year.

I'm pleased to announce the availability (for 'alpha testing') of
"multikit": a new multicast session browser.

As a special case, multikit implements a SDP browser; this can be used as an
alternative to "sdr".  (This is analogous to the way that Web browsers can
also be used as FTP clients.)  However, multikit also extends the SDP/SAP
model in several ways - most notably:
        - directories are 'first-class' objects, and can be 'announced'
just like any other session
        - sessions can have arbitrary attributes (not just the fixed set of
attributes that were defined for SDP)
        - SDP's session bundling mechanism is generalized: Any
arbitrary collection of sessions can be combined and announced as a single
'bundle'
        - a variety of 'announcement' protocols are possible.  Different
directories can use different protocols to share information.  (The
protocol for each directory is specified as an attribute of the directory.)

multikit provides an extensible framework that can be used for
experimenting with new multicast-based protocols (e.g., with differing
degrees of reliability).  It will, I hope, help to motivate new uses for the
MBone, beyond the traditional audio/video applications.

multikit is currently available for two platforms: Windows 95 (and NT), and
FreeBSD Unix.  (I can port the system to other versions of Unix if anyone
wants to give me a guest account.)

Note: This is free software, but please be aware that the current version
is 'alpha' software.  (However, I anticipate that future versions of
multikit will also remain free for experimental and non-commercial use.)

For more information about multikit - including downloading instructions -
please see:
    http://www.lvn.com/multikit

        Ross.


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subscribe

From rem-conf-request@es.net Mon Jan 06 05:58:21 1997 
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To: rem-conf@es.net
Subject: New RTP audio redundancy draft available
Date: Mon, 06 Jan 1997 10:57:06 +0000
Message-ID: <847.852548226@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>

A revised draft of the RTP payload type for redundant audio is now
available as draft-perkins-rtp-redundancy-02.txt and is enclosed 
below. There are no protocol changes in this revision, but a number 
of clarifications have been made based on comments received,.

The major outstanding issue is the choice of RTP payload type. Currently,
RAT and FreePhone use a dynamic payload type for redundant audio, yet as 
the number of tools implementing redundancy increases, it may make sense
to assign a static payload type, to ease interoperability. I'm interested
in hearing opinions on this matter...

Colin

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

INTERNET-DRAFT                              3 January 1997


                                              Colin Perkins
                                            Isidor Kouvelas
                                              Vicky Hardman
                                  University College London

                                               Mark Handley
                                                        ISI

                                     Jean-Chrysostome Bolot
                                         Andres Vega-Garcia
                                        Sacha Fosse-Parisis
                                     INRIA Sophia Antipolis



             RTP Payload for Redundant Audio Data
              draft-perkins-rtp-redundancy-02.txt


Status of this Memo


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

Internet-Drafts are draft documents valid for a maximum of six months and
may be updated, replaced, or obsoleted by other documents at any time.  It
is inappropriate to use Internet-Drafts as reference material or to cite
them other than as ``work in progress''. To learn the current status of any
Internet-Draft, please check the ``1id-abstracts.txt'' listing contained in
the Internet-Drafts Shadow Directories on ftp.is.co.za (Africa), nic.nordu.net 
(Europe), munnari.oz.au (Pacific Rim), ds.internic.net (US East Coast), or
ftp.isi.edu (US West Coast).

Distribution of this document is unlimited.

Comments are solicited and should be addressed to the authors and/or
the AVT working group's mailing list at rem-conf@es.net.

                         Abstract

    This document describes a payload type for use with the real-time
    transport protocol (RTP), version 2, for encoding redundant data.  The
    primary motivation for the scheme described herein is the development
    of audio conferencing tools for use with lossy packet networks such as
    the Internet Mbone, although this scheme is not limited to such
    applications.



Perkins et al                                      Page 1


INTERNET-DRAFT                              3 January 1997


1  Introduction

If multimedia conferencing is to become widely used by the Internet Mbone
community, users must perceive the quality to be sufficiently good for most
applications. We have identified a number of problems which impair the
quality of conferences, the most significant of which is packet loss over
the Internet Mbone.  Packet loss is a persistent problem, particularly
given the increasing popularity, and therefore increasing load, of the
Internet.  The disruption of speech intelligibility even at low loss rates
which is currently experienced may convince a whole generation of users
that multimedia conferencing over the Internet is not viable. The addition
of redundancy to the data stream is offered as a solution [1]. If a packet
is lost then the missing information may be reconstructed at the receiver
>from the redundant data that arrives in the following packet(s), provided
that the average number of consequutively lost packet is small. Recent work
[4,5] shows that packet loss patterns in the Internet are such that this
scheme typically functions well.

This document proposes an RTP payload format for the transmission of data
encoded in such a redundant fashion.


2  Requirements

The requirements for a redundant encoding scheme under RTP are as follows:

  o Packets have to carry a primary encoding and one or more redundant
    encodings.

  o As a multitude of encodings may be used for redundant information,
    each block of redundant encoding has to have an encoding type
    identifier.

  o As the use of variable size encodings is desirable, each encoded
    block in the packet has to have a length indicator.

  o The RTP header provides a timestamp field that corresponds to
    the time of creation of the encoded data.  When redundant encodings
    are used this timestamp field can refer to the time of creation
    of the primary encoding data. Redundant blocks of data will correspond
    to different time intervals than the primary data, and hence each
    block of redundant encoding will require its own timestamp.  To
    reduce the number of bytes needed to carry the timestamp, it can
    be encoded as the difference of the timestamp for the redundant
    encoding and the timestamp ofthe primary.

There are two essential means by which redundant audio may be added
to the standard RTP specification:  a header extension may hold the


Perkins et al                                      Page 2


INTERNET-DRAFT                              3 January 1997


redundancy, or one, or more, additional payload types may be defined.
These are now discussed in turn.


3  Use of RTP Header Extension

The RTP specification [2] states that applications should be prepared
to ignore a header extension.  Including all the redundancy information
for a packet in a header extension would make it easy for applications
that do not implement redundancy to discard it and just process the
primary encoding data. There are, however, a number of disadvantages
with this scheme:

  o There is a large overhead from the number of bytes needed for
    the extension header (4) and the possible padding that is needed
    at the end of the extension to round up to a four byte boundary
    (up to 3 bytes). For many applications this overhead is unacceptable.

  o Use of the header extension limits applications to a single redundant
    encoding, unless further structure is introduced into the extension.
    This would result in further overhead.

For these reasons, the use of RTP header extension to hold redundant audio
encodings is disregarded.


4  Use Of Additional RTP Payload Types


The RTP profile for audio and video conferences [3] lists a set of
payload types and provides for a dynamic range of 32 encodings that
may be defined through a conference control protocol. This leads to
two possible schemes for assigning additional RTP payload types for
redundant audio applications:

  1.A dynamic encoding scheme maybe defined, for each combination
    of primary/redundant payload types, using the RTP dynamic payload
    type range.

  2.A single fixed payload type may be defined to represent a packet
    with redundancy. This may then be assigned to either a static
    RTP payload type, or the payload type for this may be assigned
    dynamically.


4.1 Dynamic Encoding Schemes

It is possible to define a set of payload types that signify a particular
combination of primary and secondary encodings for each of the 32 dynamic
payload types provided. This would be a slightly restrictive yet feasible

Perkins et al                                      Page 3


INTERNET-DRAFT                              3 January 1997


solution for packets with a single block of redundancy as the number
of possible combinations is not too large.  However the need for multiple
blocks of redundancy greatly increases the number of encoding combinations
and makes this solution not viable.

A modified version of the above solution could be to decide prior to the
beginning of a conference on a set a 32 encoding combinations that will be
used for the duration of the conference.  All tools in the conference can
be initialized with this working set of encoding combinations. Communication 
of the working set can be made through the use of an external, out of band,
mechanism.  Setup is complicated as great care needs to be taken in
starting tools with identical parameters. This scheme is more efficient as
only one byte is used to identify combinations of encodings.

It is felt that the complication inherent in distributing the mapping
of payload types onto combinations of redundant data preclude the use
of this mechanism.


4.2 Payload Type to Mean ``Packet-With-Redundancy''

A more flexible solution is to have a single payload type which signifies a
packet with redundancy and have each of the encoding blocks in the packet
contain it's own payload type field:  such a packet acts as a container,
encapsulating multiple packets into one.

Such a scheme is flexible, since any number of redundant encodings
may be enclosed within a single packet.  There is, however, a small
overhead since each encapsulated packet must be preceded by a header
indicating the type of data enclosed. This is the preferred solution,
since it is both flexible, extensible, and has a relatively low overhead.
The remainder of this document describes this solution.


5  RTP payload type for redundant data

The assignment of an RTP payload type for this new packet format is
outside the scope of this document, and will not be specified here.
It is expected that the RTP profile for a particular class of applications
will assign a payload type for this encoding, or if that is not done
then a payload type in the dynamic range shall be chosen.

An RTP packet containing redundant data shall have a standard RTP header,
with payload type indicating redundancy.  The other fields of the RTP
header relate to the primary data block of the redundant data.

Following the RTP header are a number of additional headers, defined
in the figure below, which specify the contents of each of the encodings
carried by the packet. Following these additional headers are a number
of data blocks, which contain the standard RTP payload data for these

Perkins et al                                      Page 4


INTERNET-DRAFT                              3 January 1997


encodings.  It is noted that all the headers are aligned to a 32 bit
boundary, but that the payload data will typically not be aligned.

 0                   1                   2                   3
 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F|   block PT  |  timestamp offset         |   block length    |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The bits in the header are specified as follows:

F: 1 bit First bit in header indicates whether another header block
    follows.  If 1 further header blocks follow, if 0 this is the
    last header block.

block PT: 7 bits RTP payload type for this block.

timestamp offset:  14 bits Unsigned offset of timestamp of this block
    relative to timestamp given in RTP header.  This uses the same
    clock as the primary encoding. The use of an unsigned offset
    implies that redundant data must be sent after the primary data,
    and is hence a time to be subtracted from the current timestamp
    to determine the timestamp of the data for which this block is
    the redundancy.

block length:  10 bits Length in bytes of the corresponding data block
    excluding header.

It is noted that the use of an unsigned timestamp offest limits the
use of redundant data slightly: it is not possible to send redundancy
before the primary encoding. This may affect schemes where a low bandwidth
coding suitable for redundancy is produced early in the encoding process,
and hence could feasibly be transmitted early.  However, the addition
of a sign bit would unacceptably reduce the range of the timestamp
offset, and increasing the size of the field above 14 bits limits the
block length field. It seems that limiting redundancy to be transmitted
after the primary will cause fewer problems than limiting the size
of the other fields.

It is further noted that the block length and timestamp offset are
10 bits, and 14 bits respectively; rather than the more obvious 8 and
16 bits.  Whilst such an encoding complicates parsing the header information
slightly, and adds some additional processing overhead, there are a
number of problems involved with the more obvious choice:  An 8 bit
block length field is sufficient for most, but not all, possible encodings:
for example 80ms PCM and DVI audio packets comprise more than 256 bytes,
and cannot be encoded with a single byte length field. It is possible
to impose additional structure on the block length field (for example
the high bit set could imply the lower 7 bits code a length in words,
rather than bytes), however such schemes are complex.  The use of a

Perkins et al                                      Page 5


INTERNET-DRAFT                              3 January 1997


10 bit block length field retains simplicity and provides an enlarged
range, at the expense of a reduced range of timestamp values. A 14
bit timestamp value does, however, allow for 4.5 complete packets delay
with 48KHz audio, more at lower sampling rates, and it is felt that
this is sufficient.

The primary encoding block header is placed last in the packet. It
is therefore possible to omit the timestamp and block-length fields
>from the header of this block, since they may be determined from the
RTP header and overall packet length.  The header for the primary (final)
block comprises only a zero marker bit, and the block payload type
information, a total of 8 bits.  This is illustrated in the figure
below:






































Perkins et al                                      Page 6


INTERNET-DRAFT                              3 January 1997


 0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+
|0|   BlockPT   |
+-+-+-+-+-+-+-+-+

The final header is followed, immediately, by the data blocks, stored
in the same order as the headers.  There is no padding or other delimiter
between the data blocks, and they are typically not 32 bit aligned.
Again, this choice was made to reduce bandwidth overheads, at the expense
of additional decoding time.


6  Limitations


The RTP marker bit is not preserved for redundant data blocks. Hence
if the primary (containing this marker) is lost, the marker is lost.
It is believed that this will not cause undue problems: even if the
marker bit was transmitted with the redundant information, there would
still be the possibility of its loss, so applications would still have
to be written with this in mind.

In addition, CSRC information is not preserved for redundant data.
The CSRC data in the RTP header of a redundant audio packet relates
to the primary only. Since CSRC data in an audio stream is expected
to change relatively infrequently, it is recommended that applications
which require this information assume that the CSRC data in the RTP
header may be applied to the reconstructed redundant data.


7  Security Considerations


RTP packets containing redundant information are subject to the security
considerations discussed in the RTP specification [2], and any appropriate
RTP profile (for example[3]).  This implies that confidentiality of
the media streams is achieved by encryption.

Encryption of a redundant data-stream may occur in two ways:


  1.The entire stream is to be secured, and all participants are expected
    to have keys to decode the entire stream.  In this case, nothing
    special need be done, and encryption is performed in the usual
    manner.

  2.A portion of the stream is to be encrypted with a different key
    to the remainder.  In this case a redundant copy of the last packet
    of that portion cannot be sent, since there is no following packet
    which is encrypted with the correct key in which to send it.  Similar
    limitations may occur when enabling/disabling encryption.



Perkins et al                                      Page 7


INTERNET-DRAFT                              3 January 1997


The choice between these two is a matter for the encoder only.  Decoders
can decrypt either form without modification.


8  Example Packet

An RTP audio data packet containing a DVI4 (8KHz) primary, and a single
block of redundancy encoded using 8KHz LPC is illustrated:


 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 
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|X| CC=0  |M|      PT     |  sequence number of primary   |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|              timestamp of primary encoding                    |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|           synchronization source (SSRC) identifier            |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1| block PT=7  |  timestamp offset         |   block length    |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0| block PT=5  |                                               |
+-+-+-+-+-+-+-+-+                                               +
|                                                               |
+               LPC encoded redundant data (PT=7)               +
|                                                               |
+                               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                               |                               |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+                               +
|                                                               |
+                                                               +
|               DVI4 encoded primary data (PT=5)                |
+                                                     +---------+
|                                                     |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
















Perkins et al                                      Page 8


INTERNET-DRAFT                              3 January 1997


9  Author's Addresses

Colin Perkins/Isidor Kouvelas/Vicky Hardman
Department of Computer Science
University College London
London WC1E 6BT
United Kingdom
Email:  {c.perkins|i.kouvelas|v.hardman}@cs.ucl.ac.uk

Mark Handley
USC Information Sciences Institute
c/o MIT Laboratory for Computer Science
545 Technology Square
Cambridge, MA 02139, USA
Email:  mjh@isi.edu

Jean-Chrysostome Bolot/Andres Vega-Garcia/Sacha Fosse-Parisis
INRIA Sophia Antipolis
2004 Route des Lucioles, BP 93
06902 Sophia Antipolis
France
Email:  {bolot|avega}@sophia.inria.fr


10 References

[1] V.J. Hardman, M.A.Sasse, M. Handley and A. Watson; Reliable Audio
for Use over the Internet; Proceedings INET'95, Honalulu, Oahu, Hawaii,
September 1995.  http://www.isoc.org/in95prc/

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

[3] H. Schulzrinne; RTP Profile for Audio and Video Conferences with
Minimal Control; RFC 1890, January 1996

[4] M. Yajnik, J.Kurose and D. Towsley; Packet loss correlation in
the MBone multicast network; IEEE Globecom Internet workshop, London,
November 1996

[5] J.-C. Bolot and A. Vega-Garcia; The case for FEC-based error control
for packet audio in the Internet; Multimedia Systems, 1997










Perkins et al                                      Page 9
-------------------------------------------------------------------------------

From rem-conf-request@es.net Mon Jan 06 10:20:40 1997 
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          by max.fys.ruu.nl (8.7.5/8.7.3/hjm) with SMTP id QAA27694 
          for <rem-conf@es.net>; Mon, 6 Jan 1997 16:15:42 +0100 (MET)
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Date: Mon, 06 Jan 1997 16:15:42 +0100
From: Christiaan Balke <A.C.Balke@fys.ruu.nl>
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To: rem-conf@es.net
Subject: What RFC-number for Session Description Protocol
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Hallo people,

And the best wishes for 1997. My wish at the moment is to 
know the RFC number in which SDPv2 is described, if such a 
rfc is not obsoleted.
And if you also have an URL, please mail me so.

-- 

Hartelijke groeten 

Christiaan

-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
   A.C. Balke	|			E-mail : a.c.balke@fys.ruu.nl
=-=-=-=-=--=-=-=-			Tel :    +31-030-2534504  	
   			 		FAX    : +31-302537555
   WWW : http://www.fys.ruu.nl/~balke/chris_home_pers.html    
   Department of Physics & Astronomy, Utrecht University         
   Post box 80.000, 3508 TA Utrecht, The Netherlands.            
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-= ><> =-

From rem-conf-request@es.net Mon Jan 06 10:45:41 1997 
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From: Mark Handley <mjh@isi.edu>
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X-Phone: +1 617 253 6011
To: Christiaan Balke <A.C.Balke@fys.ruu.nl>
cc: rem-conf@es.net
Subject: Re: What RFC-number for Session Description Protocol
In-reply-to: Your message of "Mon, 06 Jan 1997 16:15:42 +0100." <32D1171E.5902@fys.ruu.nl>
Date: Mon, 06 Jan 1997 10:44:22 -0500
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>And the best wishes for 1997. My wish at the moment is to 
>know the RFC number in which SDPv2 is described, if such a 
>rfc is not obsoleted.

It's still an internet-draft right now.
draft-ietf-mmusic-sdp-02.{txt,ps}

Mark

From rem-conf-request@es.net Mon Jan 06 22:16:28 1997 
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          for <rem-conf@es.net>; Mon, 6 Jan 1997 19:09:06 -0800 (PST)
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Date: Mon, 6 Jan 1997 19:15:35 -0800
To: rem-conf@es.net
From: Joe Pallas <Pallas@Apple.COM>
Subject: Event: MacWorld Expo keynote - Jan. 7

Technology willing, the MacWorld Expo keynote speech by Apple CEO Dr. Gil
Amelio will be transmitted via multicast RTP tomorrow (Tuesday), Jan. 7,
1200 PST.  Dr. Amelio is expected to talk about how Apple's future
Macintosh operating systems will take advantage of NeXT's software.

The session is advertised with sdr.

joe

--
Joe Pallas
Apple Research Labs
When your only problem is a nail, every tool looks like a hammer.



From rem-conf-request@es.net Tue Jan 07 02:36:54 1997 
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From: anand@ece.iisc.ernet.in (SVR Anand)
Message-Id: <9701070734.AA06964@ece.iisc.ernet.in>
Subject: RTP timestamps and devices
To: rem-conf@es.net
Date: Tue, 7 Jan 97 13:04:22 GMT+5:30
Cc: anand@ece.iisc.ernet.in (SVR Anand)
X-Mailer: ELM [version 2.3 PL11]

Hi,

	I would like to know the mechanism(s) for getting the sampling 
instants from the audio device for RTP timestamping. How can I implement
the monotonically increasing clock using the audio device ?
	My development platform is MSWindows, and I am using the multimedia
library for recording, and playback audio samples. It would be of great help 
to me, if you can let me know how to go about realising the clock. 
	Since RTP is not tied to any particular operating environment, I am 
sure there will be a way of getting around my problem. 

Thanks.

Regards
Anand. 


From rem-conf-request@es.net Tue Jan 07 05:22:25 1997 
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X-Mailer: Windows Eudora Light Version 1.5.4 (16)
Mime-Version: 1.0
Content-Type: text/plain; charset="us-ascii"
To: rem-conf@es.net, mbone@isi.edu
From: Ross Finlayson <finlayson@lvn.com>
Subject: multikit update: corrected URL; new mailing list; guest account(s) 
         desired
Cc: multimedia@freebsd.org

Thanks to those of you so far who have tested the "multikit" enhanced
multicast session browser.  More testers are always welcome; for more
information, please see "http://www.lvn.com/multikit/".
                                ^
                                |
NOTE: For now you apparently need to include the trailing "/" (my web
provider's site is misconfigured; I'll see if I can get this fixed).  My
apologies to those of you who got tripped up over this.

I hope to port the software to more Unix variants - especially SunOS,
Solaris and Linux.  Would anyone like to set up a guest account for me on
one or more of these platforms?

In other news, I've set up a new mailing list - "multikit@lists.best.com" -
as an open forum for discussion of multikit - e.g.,
        - Comments and discussions on various features of multikit,
including bug reports and suggestions for improvements
        - Announcements of new versions of multikit (or related software)
        - Announcements of new "invoked command" implementations, or of new
public services accessible from a multikit directory

To subscribe to the mailing list, send a message to
"multikit-request@lists.best.com", with the word
        subscribe
in the BODY of the message.

        Ross.


From rem-conf-request@es.net Tue Jan 07 15:03:23 1997 
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          Tue, 7 Jan 1997 12:02:42 -0800
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          id OAA10950 for rem-conf@es.net; Tue, 7 Jan 1997 14:02:56 -0600
Date: Tue, 7 Jan 1997 14:02:56 -0600
From: "Milind M. Buddhikot" <milind@dworkin.wustl.edu>
Message-Id: <199701072002.OAA10950@dworkin.wustl.edu>
To: rem-conf@es.net
Subject: CFP for NOSSDAV97 (Submission Deadline: Jan 15)

Hi:

Enclosed please find a Call-for-Papers (CFP) for the 7th International
Workshop on Network and Operating Systems Support for Digital Audio
and Video to be held in St. Louis, Missouri (USA) from May 19-21,
1997.

Please feel free to circulate the CFP by any means you deem
appropriate.  Also, please excuse any multiple copies of this CFP you
may receive due to your memberships in multiple mailing lists.

Please note that the deadline for submissions is Jan 15, 1997 (fast
approaching)

Thanks and regards.

Milind


*****************************************************************************
*                                                                            *
*                                NOSSDAV'97                                  *
*                                                                            *
*                              Call-for-Papers                               *
*                                                                            *
*                                                                            *
*      The 7th International Workshop on Network and Operating System        *
*             Support for Digital Audio and Video (NOSSDAV 97)               *
*                                                                            *
*                URL: http://www.arl.wustl.edu/NOSSDAV97/nossdav.html        *
*                                                                            *
*                            May 19 - 21 1997                                *
*                                                                            *
*                               Hosted by:                                   *
*                   The Applied Research Laboratory                          *
*                    Department of Computer Science                          *
*               Washington University in St. Louis, Missouri                 *
*                                                                            *
*                Sponsored by IEEE Communications Society                    *
*                In cooperation with ACM SIGCOMM and SIGMM                   *
******************************************************************************

Objectives

The 7th International Workshop on Network and Operating Systems
Support for Digital Audio and Video (NOSSDAV 97) is the international
workshop concerned with state of the art technology in networking and
operating system support for multimedia systems.  For seven years,
NOSSDAV has proven to be an outstanding forum for researchers involved
in building innovative multimedia systems, networks and applications
on both the research and industrial front. Other topics that will be
examined include "middleware" for multimedia, media toolkits, mobile
communications, Virtual Reality (VR), real-time systems, software
agents, digital libraries, and other digital media besides audio and
video.


A key aspect of the workshop is that it provides extensive discussion
periods during which attendees can informally discuss their current
work and future research directions.  Traditionally, NOSSDAV has
emphasized on high quality experimental research that prototypes
systems to explore innovative solutions to the problems in the diverse
areas of multimedia computing. NOSSDAV97 will continue this tradition.


Relevant topics for the workshop include:

   * APIs and Continuous Media (CM) programming abstractions for
     multimedia
   * Cell-based system architectures
   * Communication protocols for multimedia
   * Distributed multimedia systems
   * End-to-end admission control
   * High-speed/ATM networks
   * Micro-kernel and OS support for real-time communications
   * Mobile multimedia systems
   * Multicast protocols and media scaling
   * Multimedia network interfaces
   * Multimedia-oriented desk, local and wide area networks
   * Multimedia and the Internet
   * Multimedia storage, server, and I/O architectures 
   * Quality of service and synchronization frameworks
   * Resource management and reservation in the OS and network
   * Software agents for multimedia systems
   * TV set-top device communication
   * VOD system architecture
   * VR systems
   * Workstation and PDA architectures for multimedia


Submissions

Two types of submissions are solicited: position papers and research
papers.  For the purpose of paper review, position papers are
restricted to three single-spaced pages. Research papers are
restricted to an extended abstract no longer than five formatted
postscript pages. To complete your submission, please send the
following items by electronic mail to NOSSDAV97@arl.wustl.edu

(1) The research or position paper in POSTSCRIPT form.

(2) The title of the paper, a list of authors with complete contact
information in the form of email address and phone number, and an
abstract summarizing the paper in PLAIN TEXT.

Only if electronic submission is impossible, papers may be sent to the
following mailing address.

Dr. Gurudatta M. Parulkar
ATTN: NOSSDAV 97
Washington University
Department of Computer Science
Applied Research Laboratory
Campus Box 1045
St. Louis, Missouri 63130
USA


The paper abstracts will be circulated among the NOSSDAV97 program
committee members to solicit reviewers.

Please note that the proceedings of the workshop will be published as
a book by Springer-Verlag and the best papers will be forwarded to
selected journals for publication.

Important Dates

Submission Deadline:            15 January 1997 (A FIRM DEADLINE)
Acceptance Notification:        15 March 1997
Final Paper Due:                15 April 1997 (A FIRM DEADLINE)
Workshop:                       19 - 21 May 1997

Program Chair

Dr. Gurudatta M. Parulkar
Director, Applied Research Lab
Department of Computer Science
Washington University, St. Louis MO. USA
guru@arl.wustl.edu, TEL: (314) 935-7534


Local Arrangements Chair

John D. DeHart
Senior Research Associate, Applied Research Lab
Department of Computer Science
Washington University, St. Louis MO. USA
jdd@arl.wustl.edu, TEL: (314) 935-7534

Other Correspondance:

NOSSDAV97@arl.wustl.edu TEL: (314) 935-7534

Program Committee

Andrew T. Campbell, Columbia University
Domenico Ferrari, Universita` Cattolica, Italy
Kevin Jeffay, Univ of North Carolina
Mike Jones, Microsoft, Inc.
Chuck Kalmanek, AT&T Research
Dilip Kandlur, IBM, TJ Watson Research
S. Keshav, Cornell University
Jim Kurose, University of Masachusetts
Monica Lam, Stanford University
Ian Leslie, Cambridge University, UK
Tom Little, Boston University
Derek McAuley, University of Glasgow, UK
Steve McCanne, Univ of California, Berkeley
A. Desai Narasimhalu, National University of Singapore
Gerald Neufeld, Univ of British Columbia, Canada
Duane Northcutt, Sun Microsystems 
Joe Pasquale, Univ of California, San Diego
Bernhard Plattner, ETH, Zurich
Steve Pink, SICS, Sweden
P Venkat Rangan, Univ of California, San Diego
KK Ramakrishnan, AT&T Research 
Henning Schulzrinne, Columbia University 
Brian Smith, Cornell University 
Doug Shepherd, University of Lancaster, UK
Ralf Steinmetz, University of Darmstadt, Germany
James Sterbenz, GTE 
Dan Swinehart, Xerox PARC
Hideyuki Tokuda, Keio University, Japan
Harrick Vin, Univ of Texas, Austin
Raj Yavatkar, Intel
Radu Popescue-Zeletin, GMD FOKUS, Germany
Hui Zhang, CMU


Publicity Chair

Milind M. Buddhikot
Graduate Research Assistant 
Department of Computer Science
Washington University, St. Louis MO, USA
milind@dworkin.wustl.edu TELE (314) 935 4302

Publications Chair

Vykky A. Klingenberg
Technical Assistant, Applied Research Laboratory
Department of Computer Science
Washington University, St. Louis MO, USA
vykky@arl.wustl.edu TELE (314) 935 7534


LOCATION
 
As is traditional, the workshop will take place at an elite resort,
away from the hustle and bustle of daily life.  Innsbrook Estates
Executive Conference Center is located on 3,200 acres of the most
gorgeous rolling Missouri woodland, dotted by crystal clear lakes.
For accommodations, there are 1, 2, and 3-bedroom condominiums which
are fully equipped with living and dining areas, fireplaces, cable
television and kitchens to offer conferees all the comforts of home in
a lakeside or wooded hideaway. You want to relax after a day of
lectures?  There is an eighteen hole golf-course, a junior-olympic
swimming pool, lighted tennis courts, mini-fitness center with a sauna
and hot tub, softball, volleyball, horseback riding, miniature golf
course, fishing, lake swimming, canoeing, and sailing, just to name a
few of the amenities.  A relaxing excursion to a scenic location away
>from the conference site is also being planned.



From rem-conf-request@es.net Wed Jan 08 04:28:54 1997 
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          with ESnet SMTP (PP); Wed, 8 Jan 1997 01:28:16 -0800
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          id CAA15785 for rem-conf@es.net; Wed, 8 Jan 1997 02:28:12 -0700 (MST)
Date: Wed, 8 Jan 1997 02:28:12 -0700 (MST)
From: Evi Nemeth <evi@rupertsberg.cs.colorado.edu>
Message-Id: <199701080928.CAA15785@rupertsberg.cs.colorado.edu>
To: rem-conf@es.net
Subject: USENIX Technical Conference

Wednesday - Friday, January 8 -10, 1997, vat/vic/wb

The keynote address (James Gosling on Developing in "Internet Time")
and selected refereed sessions (performance, caching, client tricks,
user tools) and invited talks (matt blaze on cyrptography, rob pike
on inferno, bill cheswick on stupid net tricks, louis monier on 
altavista and steve deering on ipv6 will be broadcast.  See 
http://www.usenix.org/ana97/index.html#TECHNICAL for details.

-evi

evi nemeth
computer science dept
univ of colorado
boulder, colorado

From rem-conf-request@es.net Wed Jan 08 08:55:35 1997 
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Sender: hgs@cs.columbia.edu
Message-ID: <32D3A71B.7930@cs.columbia.edu>
Date: Wed, 08 Jan 1997 08:54:35 -0500
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
X-Mailer: Mozilla 3.0 (X11; I; SunOS 5.5.1 sun4u)
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To: rem-conf@es.net
Subject: Headphone/microphone combinations?
Content-Type: text/plain; charset=us-ascii
Content-Transfer-Encoding: 7bit

I'd like to update my A/V hardware page at
http://www.cs.columbia.edu/~hgs/rtp/hardware.html. Any suggestions for,
say, good headphone/microphone combinations are appreciated.
-- 
Henning Schulzrinne         email: schulzrinne@cs.columbia.edu
Dept. of Computer Science   phone: +1 212 939-7042
Columbia University         fax:   +1 212 666-0140
New York, NY 10027          URL:   http://www.cs.columbia.edu/~hgs

From rem-conf-request@es.net Wed Jan 08 09:18:18 1997 
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          Wed, 8 Jan 1997 06:17:13 -0800
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          Wed, 8 Jan 1997 09:15:39 -0500
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Message-Id: <9701081413.AA01417@boggs.wpine.com.wpine.com>
To: rem-conf@es.net
Subject: RTCP Scalability problem.
Date: Wed, 08 Jan 1997 09:13:52 -0500
From: Brian O'Shea <boshea@wpine.com>


This is not a solution, only some thoughts about another possible approach to
solving the problem.  There may be significant holes in the idea, and I invite
those more knowledgable then I to point them out, since I can't think of them
at the moment.

During a sleepless night it occured to me that nobody had proposed using the
IP address of the sender of a multicast RTCP message to determine whether they
should send one.  

We could have a scheme whereby the RTCP message is sent only after a random
timer expires.  If, sometime before that timer expires you receive an RTCP
message with a recvfrom address that matches your subnet, then you reset the
timer.  That way RTCP messages would only be sent from a subnet, not from
everyone in the subnet.

So, what's wrong with this, and why won't it work?

-bos

+***********************************************************************+
+ Brian O'Shea					White Pine Software	+
+ Reflector Group Leader			542 Amherst St		+
+ bos@wpine.com					Nashua NH, 03063	+
+ Phone: 603-886-9050				Fax: 603-886-9051	+
+		All it takes is all you've got.				+
+***********************************************************************+



From rem-conf-request@es.net Wed Jan 08 09:53:28 1997 
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Message-ID: <c=US%a=_%p=ThePoint_Interne%l=TIS_MAIL-970108145212Z-418@tis-mail.thepoint.net>
From: Arlie Davis <arlie@thepoint.net>
To: "'rem-conf@es.net'" <rem-conf@es.net>, 'Brian O'Shea' <boshea@wpine.com>
Subject: RE: RTCP Scalability problem.
Date: Wed, 8 Jan 1997 09:52:12 -0500
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In other words, you wan to make RTCP participants (can't really call
them clients) sensitive to nearby neighbors, so that they reduce the
frequency of their updates.  A good, scalable idea.  The problem is, how
do you define who is a close neighbor?  Your first shot at it, "is peer
on the same subnet?" works for one specific scenario -- machines on
LANs, watching for peers on the exact same LAN.  This will work, and
it's a good first step toward _localized_ reduction of RTCP traffic, but
it doesn't seem general enough.

Here's a scenario that _should_ benefit from localized reduction of RTCP
traffic (I'll just call it Localized Back-off or LB to avoid typing all
morning): Several machines on different LAN segments very "near" each
other.  For example, a college campus might have 40 different segments,
15 of which have people watching a popular multicast session.  If there
are only 2-3 people per segment watching, then each participant backs
off a little bit, but not nearly as much as they would if they all
realized they were "near" each other.  Also, how do you define "subnet"
in a meaningful way when you have, say, 256 people dialed into modem- or
ISDN-based access servers?  The idea of subnet is a rather squishy one
when you are dealing with point-to-point links.

Here's an idea that might improve the situation a little bit: Include in
the RTCP packet the original MTTL.  When the packet is received, find
the difference between the original TTL (in the RTCP payload) and the
actual TTL (in the IP header).  The difference is the number of hops
that the packet had to traverse.  That number is a much more interesting
metric of how close a participant is than my-subnet-or-not.  You can use
that to scale how frequently you multicast your own presence.

Has anyone ever considered having an RTP statistics monitor?  You could
have all of the RTCP participants _unicast_ their reports to the
statistics monitor, which would save on bandwidth and on processor time
on zillions of machines.  The stats monitor could then summarize and
periodically multicast the results.  How do you decide who runs the
statistics monitor?  I dunno.  Just an idea.  You could have the
statistics monitor periodically multicast an RTCP datagram that says
"I'm the statistics monitor -- send your information to me, instead of
the group".  RTCP participants could run in "compatability" mode (old
way) until they saw a statistics monitor announcing its presence.  The
RTCP monitor could also support a TCP protocol (to be used infrequently)
that could be used to browse the members (past and present) of the
multicast session, session statistics, multicast traffic analysis, and
so on.

It just seems goofy for every single client to talk to every other
client all the time.  I know there are some good reasons for it, but it
seems there are some bad reasons, too.

Again, just some ideas.  I welcome (intelligent and constructive)
criticism.

-- arlie

>----------
>From: 	Brian O'Shea[SMTP:boshea@wpine.com]
>Sent: 	Wednesday, January 08, 1997 9:13 AM
>To: 	rem-conf@es.net
>Subject: 	RTCP Scalability problem.
>
>
>This is not a solution, only some thoughts about another possible approach to
>solving the problem.  There may be significant holes in the idea, and I
>invite
>those more knowledgable then I to point them out, since I can't think of them
>at the moment.
>
>During a sleepless night it occured to me that nobody had proposed using the
>IP address of the sender of a multicast RTCP message to determine whether
>they
>should send one.  
>
>We could have a scheme whereby the RTCP message is sent only after a random
>timer expires.  If, sometime before that timer expires you receive an RTCP
>message with a recvfrom address that matches your subnet, then you reset the
>timer.  That way RTCP messages would only be sent from a subnet, not from
>everyone in the subnet.
>
>So, what's wrong with this, and why won't it work?
>
>-bos
>
>+***********************************************************************+
>+ Brian O'Shea					White Pine Software	+
>+ Reflector Group Leader			542 Amherst St		+
>+ bos@wpine.com					Nashua NH, 03063	+
>+ Phone: 603-886-9050				Fax: 603-886-9051	+
>+		All it takes is all you've got.				+
>+***********************************************************************+
>
>
>

From rem-conf-request@es.net Wed Jan 08 10:24:04 1997 
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Date: Wed, 08 Jan 1997 10:22:49 -0500
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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To: Brian O'Shea <boshea@wpine.com>
CC: rem-conf@es.net
Subject: Re: RTCP Scalability problem.
References: <9701081413.AA01417@boggs.wpine.com.wpine.com>
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Brian O'Shea wrote:
> 

> 
> We could have a scheme whereby the RTCP message is sent only after a random
> timer expires.  If, sometime before that timer expires you receive an RTCP
> message with a recvfrom address that matches your subnet, then you reset the
> timer.  That way RTCP messages would only be sent from a subnet, not from
> everyone in the subnet.
> 
> So, what's wrong with this, and why won't it work?

This was discussed earlier, I believe. However:

- Gives inaccurate receiver count.
- Fails for small groups where you want to know *all* the people
participating, not just the subnet.
- Doesn't help much with groups where every subnet has, on average,
close to one listener.
- The application has to know the definition of "subnet" if the subnet
mask is not "classical". Subnet mask information may or may not be
readily available to the *application* (obviously, the kernel or
equivalent knows it).
- A remote monitor can't easily know (because it certainly does not know
the subnet mask) as to whether two reports from different SSRCs reflect
basically the same QOS conditions (even assuming that all receivers on
the subnet indeed share the same QOS, which may be true for yellow-cable
Ethernet, but may not be true for more switched environments)
- Since the interval between transmissions for a single member will be
larger than given by the nominal RTCP interval, stations will time out
at the other group members if there are more than just a few in a
subnet.

Note that there are other mechanisms that alleviate (but don't
completely solve) the problem of excessive RTCP traffic during start-up.
In steady-state, the rate is not the problem for any group size; you may
just never hear from a large part of your audience... The
one-report-per-subnet proposal makes somewhat more economical use of the
RTCP bandwidth to distribute "interesting" feedback.
> 
> -bos
> 
> +***********************************************************************+
> + Brian O'Shea                                  White Pine Software     +
> + Reflector Group Leader                        542 Amherst St          +
> + bos@wpine.com                                 Nashua NH, 03063        +
> + Phone: 603-886-9050                           Fax: 603-886-9051       +
> +               All it takes is all you've got.                         +
> +***********************************************************************+

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

From rem-conf-request@es.net Wed Jan 08 12:29:28 1997 
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          Wed, 8 Jan 1997 09:28:49 -0800
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          Wed, 8 Jan 97 12:28:41 EST
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Mime-Version: 1.0
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Date: Wed, 8 Jan 1997 12:29:26 -0400
To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>, rem-conf@es.net
From: macker@itd.nrl.navy.mil (Joseph P. Macker)
Subject: Re: Headphone/microphone combinations?

Henning:

FYI, for 3-4 years we have had workable results with low rate LPC-10 based
network phone applications using, speaker & headset mic combination.  Most
of our low rate 2400 bps LPC-10 code and vector quantized 1200,800 bps code
is quite sensitive to any background noise and microphone reproduction
quality. The headset mics we have used on a number of workstations are
Audio-Technica ATV-75, ATM-73a and others in that series (profession music,
audio outlets carry these in typical ranges from $100-250).  Some models
have proximity effect compensation.  We used these headset mics with most
apps (e.g., vat,nevot,rat,ivox) with excellent results.  A reasonable
compromise for those desiring a speaker/mic combo rather than headphones.

These model types are relatively inexpensive for the quality audio produced
and IMHO are non-intrusive and comfortable.  I assume there are also good
wireless versions of these mic types, but I have had the opportunity to try
these (i'd love to try these out).  Early on we did try quality aviation
headsets with a number of workstations, but found them ackward for many
casual AVT situations (for extreme noise environments
..helicopters,etc...this is the only sane way to go).

-joe

    _/      _/ _/_/_/_/  _/       Joseph P. Macker, Code 5544
   _/_/    _/ _/     _/ _/        Center for High Assurance Computer Systems
  _/  _/  _/ _/_/_/_/  _/         (202)-767-2001
 _/    _/_/ _/   _/   _/          macker@itd.nrl.navy.mil
_/      _/ _/     _/ _/_/_/_/     http://tonnant.itd.nrl.navy.mil/macker.html



From rem-conf-request@es.net Wed Jan 08 13:53:36 1997 
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From: Dermot Tynan <dtynan@fws.ilo.dec.com>
Message-Id: <9701081840.AA11003@karpov.fws.ilo.dec.com>
Subject: Re: RTCP Scalability problem.
To: arlie@thepoint.net (Arlie Davis)
Date: Wed, 8 Jan 1997 18:40:27 +0000 (GMT)
Cc: rem-conf@es.net, boshea@wpine.com
In-Reply-To: <c=US%a=_%p=ThePoint_Interne%l=TIS_MAIL-970108145212Z-418@tis-mail.thepoint.net> from "Arlie Davis" at Jan 8, 97 09:52:12 am
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Arlie Davis wrote:
> 
> Has anyone ever considered having an RTP statistics monitor?  You could
> have all of the RTCP participants _unicast_ their reports to the
> statistics monitor, which would save on bandwidth and on processor time
> on zillions of machines.

I believe this is mentioned in the Aboba paper as a workable method,
although I may be a bit confused.  He certainly covered a lot of the
ins and outs of various schemes.  One thing that occurs to me is that
in the case of a large multicast, say, for example, some sort of
broadcast like a Presidents Address or a lecture or something, what
is of interest isn't the names and addresses of the receivers, but
just simply the count.  The Aboba paper talks in terms of monitoring
counts for administrative and business reasons.  In this particular
case, reception reports are of little interest to the main server,
nor are the names of the listeners.  Intermediate translators and
mixers will want the reception reports, but I can't see the main
server doing anything with say, a million RRs.  In terms of a large
audio conference with several mixers and translators, then any given
translator could take the various RRs and NAME types and save them,
only releasing basic stats such as the number of listeners.  It would
retain the NAME information in case one of the listeners transmits
data.  In which case, it would make sure that the appropriate RTCP
packet for that listener was broadcast back up the stream, using the
standard algorithm for transmission.  RRs would be sent only on behalf
of the translator/mixer.  Does this seem reasonable?
						- Der
-- 
Dermot Tynan						+353 91 754608
dtynan@ilo.dec.com					 DTN: 822-4608

AltaVista Internet Software, Galway, Ireland

From rem-conf-request@es.net Wed Jan 08 15:58:34 1997 
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To: macker@itd.nrl.navy.mil (Joseph P. Macker)
Cc: Henning Schulzrinne <schulzrinne@cs.columbia.edu>, rem-conf@es.net
Subject: Re: Headphone/microphone combinations?
In-Reply-To: Message of Wed, 8 Jan 1997 12:29:26 -0400 from macker@itd.nrl.navy.mil (Joseph P. Macker) <9701081728.AA19823@itd.nrl.navy.mil>
Date: Wed, 08 Jan 97 12:51:21 -0800
From: berc@pa.dec.com
X-Mts: smtp


Besides Plantronics Mirage headsets (you already have a pointer to 
Plantronics), I sometimes use a Sony WRT-27A/WRR-27 (957MHz) wireless 
setup.  It works fine in my office and around our floor, but anyone 
going wireless should try before they buy since local physical can 
have a large affect on performance and noise.

When going wireless I use a Shure ST4300 echo canceller, which helps 
but doesn't get rid of everything.

lance

From rem-conf-request@es.net Thu Jan 09 05:21:32 1997 
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Subject: MBONE announcement for CERN LHCC meeting
To: rem-conf@es.net, hepnet-l@slacvm.slac.stanford.edu, hrc@frcpn11.in2p3.fr, 
    htasc@listbox.cern.ch, net-teleconferencing@listbox.cern.ch
Date: Thu, 9 Jan 1997 11:20:13 +0100 (MET)
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	 CERN is pleased to announce the MBONE broadcast of the

		  Large Hadron Collider Committee meeting
		  ---------------------------------------

		       on Thursday 23 January 1997
			 from the CERN Auditorium

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

>>> Times are UTC+2 <<<

 9.00 -  9.10  Status of the LHC project (C. Llewellyn Smith)

 9.15 -  9.40  ATLAS status report (P. Jenni)

 9.45 - 10.35  ATLAS calorimeter Technical Design Report

10.45 - 11.15  COFFEE BREAK

11.15 - 12.05  ATLAS calorimeter Technical Design Report continued

12.15 - 12.30  RD42 status report (P. Weilhammer)


14.00 - 14.50  ATLAS Computing Technical Proposal

15.00 - 15.50  CMS Computing Technical Proposal


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

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

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

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


From rem-conf-request@es.net Fri Jan 10 13:41:51 1997 
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To: rem-conf@es.net
From: lee <lee@math.washington.edu>
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Subject: Pacific Northwest Geometry Seminar

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

Title:       
	Pacific Northwest Geometry Seminar
Date:        
	Jan 10, 1996

Time:        
	08:00 GMT 2 hours

Contact:     
	lee@math.washington.edu

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

Description:        
	Quarterly regional meeting for geometers.









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

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To: rem-conf@es.net
From: lee <lee@math.washington.edu>
Reply-to: lee@math.washington.edu
Subject: Pacific Northwest Geometry Seminar

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

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

Time:        
	09:00 PST8PDT 8 hours

Contact:     
	lee@math.washington.edu

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

Description:        
	Quarterly regional meeting for geometers.









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

From rem-conf-request@es.net Fri Jan 10 18:21:40 1997 
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From: Ross Finlayson <finlayson@lvn.com>
Subject: New multikit ports (SunOS, Solaris) now available

SunOS and Solaris/SPARC versions of the "multikit" enhanced multicast
session browser are now available.  (Thanks to Kalyan Kidambi and Mark
Handley for providing guest accounts.)

For downloading instructions, please see http://www.lvn.com/multikit/

        Ross.


From rem-conf-request@es.net Fri Jan 10 18:48:35 1997 
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To: rem-conf@es.net
Subject: last minute reminder -- USENIX closing session

mbone event, USENIX closing session:

severe tire damage - music and tricks.  16:15 PST (GMT -9 i think).

-evi

From rem-conf-request@es.net Fri Jan 10 19:01:59 1997 
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To: rem-conf@es.net
From: mallard@mail.arc.nasa.gov (Mark Allard)
Subject: NASA STS-81 Shuttle MBone Session

NASA-Ames Research Center is pleased to provide MBone coverage of the
STS-81 Shuttle Mission.

This will be the fifth docking between the Space Shuttle and the Russian
Space Station MIR, the second mission involving an exchange of US
Astronauts.

Liftoff is currently anticipated for January 12, 1997 at 4:27AM(EST) with
an anticipated mission duration of just over 10 days.

There is an updated Shuttle Web Page available at <http://shuttle.nasa.gov>
that will carry mission information and links of related interest; the
MBone session will carry the link.

Contact Mark Allard <mallard@mail.arc.nasa.gov> with any issues pertaining
to this broadcast being supplied as a public service to the MBone community
by NASA-Ames Research Center.

Enjoy,
M



From rem-conf-request@es.net Mon Jan 13 11:31:24 1997 
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Date: Mon, 13 Jan 1997 10:26:35 -0600
From: Eric Hauch <ehauch@mail.state.tn.us>
To: rem-conf@es.net
Cc: ehauch@mail.state.tn.us
Subject: ANSI Standards WG T1A1.5 - Multimedia Communications Coding & 
         Performance
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Greetings to the members of the IETF-AVT Working Group,

As chairman of WG T1A1.5, I have had recent discussions
with Stephen Casner to explore areas of mutual interest
between our groups.  Dan Klenke (a recent T1A1.5
vice-chair) attended part of your December meeting in San
Jose to gain further insight.

Based on Dan's observations at your meeting, it looks very
much like there would be benefit to our groups in aligning our
activities and supporting each other - some of Dan's thoughts
on this follow.

I will be continuing a dialogue with Stephen - and will foster a
discussion on the activities of your group at our meeting next
week in Orlando.  I may also be able to attend part of your
April meeting in Memphis.

In the meantime, both Stephen and I encourage anyone
interested to look through the T1 Web Page (www.t1.org).  I
am also attaching the meeting notice for our upcoming
meeting - anyone who is able to attend on such short notice
is more than welcome.

Please feel free to contact me if you need any further
information.

Eric Hauch
WG T1A1.5 Chair
ehauch@mail.state.tn.us
615-532-2365


Dan Klenke's Thoughts As Follows:

This message is an introduction to T1A1, an ANSI-accredited
telecommunication standards body.  It has become clear
that the work of T1A1.5 and the IETF (especially the AVT
WG) have much in common.  We propose the following
questions:

	. How should we work together?
	. How should we communicate our work results to
each other?
	. Are there opportunities for joint activity?
	. How can the IETF take advantage of this US channel
into the ITU?

Following is a high-level description of T1A1, and an invitation
to IETF members to attend the upcoming meeting of T1A1 in
Orlando, Florida during the week of January 20, 1997.


		  Technical Sub-Committee T1A1
		Performance and Signal Processing
		==========================

   Working Groups:

	T1A1.2 - Network Survivability

	T1A1.3 - Performance of Digital Networks and Services

    **	T1A1.5 - Multimedia Communications Coding and
Performance

	T1A1.7 - Signal Processing and Network Performance
for Voiceband Services

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


	T1A1.5 is (roughly) the equivalent of what was ITU-T
Study Group 15, Working Party 1.  It has responsibility for
the North American versions of the H.3xx set of multimedia
terminal definitions.  It serves as a "clearing house" for US
contributions to SG 15 (now SG 16).

	T1A1 typically meets four times a year.  Attendance
and participation is open to all interested parties without
charge.  Membership in T1A1 is required for voting privleges.

	For more information about T1 and T1A1, try the T1
Web Site at www.t1.org

	For more information about the activities of T1A1.5,
contact the Working Group Chairman at Eric Hauch
(ehauch@mail.state.tn.us)

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

Orlando Meeting Notice Follows:

T1A1 ADVANCED MEETING LOCATION ANNOUNCEMENT
Orlando, Florida
January 20 through 24, 1997

HOSTED BY:	ECI Telecom, Inc. and AT&T

MEETING PLACE:	Sheraton Orlando North Hotel
	600 North Lake Destiny Drive
	Maitland, FL 32751
	Phone:	(407) 660-9000 or 1~-800/ 628-6660
	FAX:		(407) 660-9008

CONTACT:	Dan Etz-Hadar/Elayna Monts
	ECI Telecom, Inc. (407) 830~-2544

ACCOMMODATIONS:	For reservations, call the Sheraton
Orlando North Hotel at the number listed above.  A block of
rooms has been set aside in the name of T1A1/ECI
TELECOM at the rate of $109.00 per night (plus
tax).  This rate applies for single or double occupancy. The
block of rooms will be released by January 5, 1997.  After
that date, reservations will be accepted on a space 
available basis only.  PLEASE MENTION THE GROUP
NAME WHEN MAKING RESERVATIONS 
TO QUALIFY FOR THE RATE.

LOCAL TRANSPORTATION:	Public transportation is not
convenient in Central Florida. We recommend a rental car.
Rates are very competitive. If you prefer, a shuttle service is
available from the airport to the hotel and Avis Rental 
Car has a desk located in the hotel lobby for short or
long-term rentals.  Transtar provides regularly scheduled
shuttle service to the hotel at a cost of $19 one way or $35
round-trip.  Cab fare from the airport to the Sheraton 
North Hotel is approximately $50.

DIRECTIONS:	Departing the airport, take Semoran Blvd
(State Road 436 North).  Stay on 436 for six miles, to the
East/West Expressway.  Go west on the 
Expressway five miles to I-4, then seven miles to Maitland
Blvd, West (Exit 47B).  The hotel is located on Lake Destiny
Drive (First right off of Maitland Blvd, West).

MAILING INSTRUCTIONS: 	Documents should be shipped
to hotel no earlier than 3 days in advance.  No charge for
receipt of boxes.

	The mailing address should read as follows:

	Sheraton Orlando North Hotel & Towers
	600 North Lake Destiny Drive
	Maitland, FL 32751
ATTEN: (RECIPIENT'S NAME) GUEST
T1A1 /ECI Telecom Inc.
c/o Convention Services

WEATHER: 	Florida in January cool, jacket weather. 
Temperatures range from 40 to 70 degrees.

##################################################

NEXT T1A1 MEETING
Jan 20 - 24, 1997

SCHEDULE FOR THE WEEK

		TIME KEY (except Plenary)
			AM = 8:30 a.m. to noon
			PM = 1:30 to 5:00 p.m.
	

ACTIVITY			Meeting Days & Times
----------------------		----------------------

T1A1.2 Working Group:		Mon AM thru Tue PM
(Network Survivability)
  Editing Group:			Wed AM thru Thurs PM

T1A1.3 Working Group:		Mon PM thru Thurs PM		
(Perf. of Dig. Serv. & Netw.)

T1A1.5 Working Group: 		Tue AM thru Thurs PM 
(Multimedia Comm. & Coding)

T1A1.7 Working Group:		Weds AM thru Thurs PM 
(Sig. Proc.+Netw. Perf. VB Serv.)
ATM Editing Mtg:		Tues AM
Access Model			Tues PM

T1A1 Plenary			Friday 8:00 a.m.



From rem-conf-request@es.net Mon Jan 13 14:40:10 1997 
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Sender: hgs@cs.columbia.edu
Message-ID: <32DA8F5C.934@cs.columbia.edu>
Date: Mon, 13 Jan 1997 14:39:08 -0500
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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To: rem-conf@es.net
Subject: Packet audio patent
Content-Type: text/plain; charset=us-ascii
Content-Transfer-Encoding: 7bit

There was some discussion a while back on a recent packet audio patent.
This patent is now available on-line for inspection.

http://patent.womplex.ibm.com/details?patent_number=5526353
-- 
Henning Schulzrinne         email: schulzrinne@cs.columbia.edu
Dept. of Computer Science   phone: +1 212 939-7042
Columbia University         fax:   +1 212 666-0140
New York, NY 10027          URL:   http://www.cs.columbia.edu/~hgs

From rem-conf-request@es.net Mon Jan 13 15:33:24 1997 
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          Mon, 13 Jan 1997 15:32:48 -0500 (EST)
From: Kathy Lynn Hewitt <klhewitt@eos.ncsu.edu>
Message-Id: <9701131532.ZM27493@eos.ncsu.edu>
Date: Mon, 13 Jan 1997 15:32:44 -0500
X-Mailer: Z-Mail (3.2.1 10oct95)
To: rem-conf@es.net
Subject: Importing Powerpoint 7.0 slides into wb
Mime-Version: 1.0
Content-Type: text/plain; charset=us-ascii

Hi!  I'm trying to import Powerpoint 7.0 slides into wb by converting them to
ps.  But I can't seem to convert it into proper format.  I've tried using
distill from Adobe.  I've tried different ps print drivers like Digital LN0R3
and then setting the postscript option to optimize for portabiltiy.  But
nothing seems to work.

I'm trying to get this by 7pm tonight so any help would be greatly appreciated.

Thank you!!

--Kathy Hewitt

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


From rem-conf-request@es.net Mon Jan 13 18:32:13 1997 
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Date: Mon, 13 Jan 1997 18:32:30 -0400
To: Kathy Lynn Hewitt <klhewitt@eos.ncsu.edu>, rem-conf@es.net
From: macker@itd.nrl.navy.mil (Joseph P. Macker)
Subject: Re: Importing Powerpoint 7.0 slides into wb

At  3:32 PM 1/13/97 -0500, Kathy Lynn Hewitt wrote:
>Hi!  I'm trying to import Powerpoint 7.0 slides into wb by converting them to
>ps.  But I can't seem to convert it into proper format.  I've tried using
>distill from Adobe.  I've tried different ps print drivers like Digital LN0R3
>and then setting the postscript option to optimize for portabiltiy.  But
>nothing seems to work.

Kathy:
There used to be a limit on the size of imported ps in wb (~32K per page).
This might make things a little tight with "busy", converted Powerpoint slides.

Just thought this might be what you are running into at present. 
-joe



From rem-conf-request@es.net Mon Jan 13 20:01:47 1997 
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From: Mark Handley <mjh@isi.edu>
X-Organisation: Information Sciences Institute, USC
X-Phone: +1 617 253 6011
To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
cc: rem-conf@es.net
Subject: Re: Headphone/microphone combinations?
In-reply-to: Your message of "Wed, 08 Jan 1997 08:54:35 EST." <32D3A71B.7930@cs.columbia.edu>
Date: Mon, 13 Jan 1997 20:00:20 -0500
Message-ID: <19631.853203620@buttle.lcs.mit.edu>
Sender: mjh@buttle.lcs.mit.edu


>I'd like to update my A/V hardware page at
>http://www.cs.columbia.edu/~hgs/rtp/hardware.html. Any suggestions for,
>say, good headphone/microphone combinations are appreciated.

Having never been very satisfied with any of the headsets I've tried
(and we had a pretty good selection at UCL), I finally resorted to
building my own.  

My requirements were to have headphones that go over my ears rather
than press on them because this puts the drivers a little further from
my ears meaning load sounds hurt less (and MBone sessions often have
loud sounds) and because I find them more comfortable for wearing for
long periods of time.
	
I spent some time wandering around audio shops till I found a set of
headphones that were suitably comfortable without really caring too
much about the quality because, lets face it, good high frequency
response is actually a liability with workstations - the high
frequency hiss from SparcStations on walkman type headsets always
annoys me.

What I got were a pair of $30 Aiwa headphones - I was prepared to
spend a lot more, but these were more comfortable.

I also got a tiny ( < 10mm) $2 condenser microphone from Radio Shack,
assorted cables and plugs, a 10uF capacitor and a wire coathanger.
Assorted soldering, bending of coathanger and insulating tape later I
now have the headset I wanted.  The SparcStation supplies the 5V bias
voltage for the mike just perfectly.  This is on an SS20 - don't know 
how universal it is, but I think SS10s and SS5s are the same.

     +------+                                               +---+
     |      |-------------- bias (+5V) ---------------------|   |
     |      |                                                \ /
     | mike |      ||                                       +===+
     |      |------||--------- output ----------------------|   | 3.5mm plug
     |      |      || 10 uF                                 |===|
     |      |                                               |   |
     |      |---------------- ground -----------------------|   |
     +------+                                               |   |
                                                          +=======+
                                                          |       |

I have to admit I was surprised that the bias and the output were the
opposite way round from what I expected.

You need a good breathshield over the mike when it's only 2-3cm from
your mouth, but Radio Shack sell those too.  Anyway, this setup works
pretty well.  Total cost: about $40.  

See http://buttle.lcs.mit.edu/~mjh/headphones.gif for a not terribly
good GIF.

Mark

From rem-conf-request@es.net Mon Jan 13 23:54:29 1997 
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Date: Mon, 13 Jan 1997 20:33:51 -0800 ()
From: Stephen Casner <casner@precept.com>
To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
cc: rem-conf@es.net
Subject: Re: Packet audio patent
In-Reply-To: <32DA8F5C.934@cs.columbia.edu>
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On Mon, 13 Jan 1997, Henning Schulzrinne wrote:
> There was some discussion a while back on a recent packet audio patent.
> This patent is now available on-line for inspection.
> 
> http://patent.womplex.ibm.com/details?patent_number=5526353

See also patent 4,100,377, "Packet transmission of speech", from
1978, and 4,748,620, "Time stamp and packet virtual sequence
numbering for reconstructing information signals from packets", from
1988.  And probably others...
							-- Steve


From rem-conf-request@es.net Tue Jan 14 13:02:48 1997 
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Message-ID: <32DB983D.4BF9@mbone.com>
Date: Tue, 14 Jan 1997 10:29:17 -0400
From: Vinay Kumar <vinay@mbone.com>
Organization: ICAST Communications, Inc.
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CC: rem-conf@es.net
Subject: mbone website update
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the mbone website at http://www.mbone.com/ now maintains html hyperlinked 
archives of the 'rem-conf' and 'mbone' mailing lists. The archives are 
also indexed and keyword searchable.

please feel free to send suggestions, feedback, bug reports. also, parts 
of the site are either slow or under construction/update, so please bear 
with me on that one. this will be fixed real soon now.
happiness.
---
 vinay kumar
vinay@mbone.com
http://www.mbone.com/

From rem-conf-request@es.net Tue Jan 14 13:32:58 1997 
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Date: Tue, 14 Jan 1997 19:31:16 +0100
From: Roland Mech <mech@tnt.uni-hannover.de>
Organization: Universitaet Hannover, Theoretische Nachrichtentechnik
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Subject: running wb on Windows95
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Hello,

I am trying to run the whiteboard-tool wb on a PC using Windows 95,
and I have the problem that Windows gives the error message
"No 32 Bit application!"

I downloaded the wb-tool from
 ftp.ee.lbl.gov/conferencing/wb/i386-wb-1.59.

Many thanks in advance for helpful answers.

  Roland

________________________________________________________________________________

Dipl.-Inform. Roland Mech          Universitaet Hannover
                                   Institut fuer Theoretische
Nachrichtentechnik
                                   und Informationsverarbeitung
phone:  +49-511-762-5308           Appelstrasse 9A
fax:    +49-511-762-5333           30167 Hannover, Germany
email:  mech@tnt.uni-hannover.de
www:	http://www.tnt.uni-hannover.de/wiss/mech.html    
________________________________________________________________________________

From rem-conf-request@es.net Tue Jan 14 14:31:24 1997 
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Date: Tue, 14 Jan 1997 11:30:38 -0800
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To: sage-members@usenix.org, baylisa@baylisa.org, rem-conf@es.net
Subject: BayLISA: Brent Chapman on Containment Zones
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The BayLISA group meets monthly to discuss topics of interest to systems
and network administrators.  The meetings are free and open to the public.

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


Schedule
--------

January 16
Containment Fields: An Intranet Application of Internet Firewalls Technology

Brent Chapman
Great Circle Associates, Inc.
Brent@GreatCircle.COM
+1 415 962 0841 or 800/270-2562

Many organizations require different security models for different parts of
their networks; for instance, they may have classrooms and demonstration
facilities which customers are allowed access to, as well as internal
development networks where outsiders are strictly prohibited.  Containment
fields are a way of using standard Internet firewall technologies (packet
filters and IP tunnels) to establish and link pieces of a network that have
special security requirements without compromising the security of the
larger network they are a part of.  The containment field concept was
developed as part of the SGI/Cray merger, in order to reconcile the
significantly different security models of those two organizations as they
merged their existing wide area networks.  This talk will describe
containment fields in the context of the SGI/Cray merger, including why and
how we developed them, what we expect to do with them, and how we plan to
build them.

February 20
Nick Stoughton
Standards efforts and how they will effect you

March 20
Paul Vixie

[Schedule subject to change]

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

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

	ftp.baylisa.org:/location

For any other information, please send email to:

	info@baylisa.org

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




From rem-conf-request@es.net Tue Jan 14 23:39:57 1997 
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Date: Wed, 15 Jan 1997 05:39:22 +0100
From: Hans Mayer <Hans.Mayer@gmd.de>
Message-Id: <199701150439.AA20841@mail.gmd.de>
To: mech@tnt.uni-hannover.de, rem-conf@es.net
Subject: Re: running wb on Windows95

>I downloaded the wb-tool from
> ftp.ee.lbl.gov/conferencing/wb/i386-wb-1.59.

I'm not shure this executable is a W'95 one. As far as I know wb is Interviews
based and this is so X'centric I would guess it's impossible to port to any-
thing else.

Hans

From rem-conf-request@es.net Wed Jan 15 03:58:21 1997 
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To: Hans Mayer <Hans.Mayer@gmd.de>
cc: mech@tnt.uni-hannover.de, rem-conf@es.net
Subject: Re: running wb on Windows95
In-reply-to: Your message of "Wed, 15 Jan 1997 05:39:22 +0100." <199701150439.AA20841@mail.gmd.de>
Date: Wed, 15 Jan 1997 08:26:20 +0000
Message-ID: <660.853316780@cs.ucl.ac.uk>
From: Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>



 >>I downloaded the wb-tool from
 >> ftp.ee.lbl.gov/conferencing/wb/i386-wb-1.59.
 
 >I'm not shure this executable is a W'95 one. As far as I know wb is Interviews
 >based and this is so X'centric I would guess it's impossible to port to any-
 >thing else.

right

the best freebie shared media tool for windows right now is probasbly
mark handley's nte (network text editor) from a MICE FTP site near
you....


 jon


From rem-conf-request@es.net Wed Jan 15 17:02:30 1997 
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To: rem-conf@es.net
Cc: band@srd.com
Subject: The Triumph of Severe Tire Damage
Date: Wed, 15 Jan 97 13:55:41 -0800
From: berc@pa.dec.com
X-Mts: smtp


If you weren't at Usenix, this is your chance to experience the 8KHz 
monaural audio splendor of Severe Tire Damage accompanied by a blazing 
128Kb of H.261 video.

What: Severe Tire Damage
When: January 15th, 8pm PST
From: The Fabulous SubForum, in Palo Alto California
Why:  Because rock & roll will keep South Korea free

Remote camera control will be present, and maybe the remote mixer.
We're still working on the software for the remote fog machine.

lance

From rem-conf-request@es.net Thu Jan 16 09:21:52 1997 
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From: Carlos Becker Westphall <westphal@orion.lrg.ufsc.br>
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To: sarikaya@u-aizu.ac.jp, cellular@dfv.rwth-aachen.de, perform@tay1.dec.com, 
    tccc@cs.umass.edu, ghafoor@ecn.purdue.edu
Subject: Please distribute widely
Cc: announcements.chi@xerox.com, arl@arl.wustl.edu, atm@bbn.com, 
    ccrc@dworkin.wustl.edu, cip@bbn.com, cnom@maestro.bellcore.com, 
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>From sarikaya@u-aizu.ac.jp Wed Jan  8 05:36 EDT 1997
Date: Wed, 08 Jan 1997 16:24:37 +0900
From: Behcet Sarikaya <sarikaya@u-aizu.ac.jp>
Subject: Please distribute widely




                    PRELIMINARY CALL FOR PAPERS - PROMSMmNet'97
         International Conference on Protocols for Multimedia Systems -    
                              Multimedia Networking
                       Santiago, Chile, November 24-26, 1997



IMPORTANT DATES: 
   Deadline for Receipt of Papers                   June 2, 1997
   Notification of Acceptance Mailed                July 14, 1997
   Final Camera Ready Papers Due to IEEE CS Press   September 4, 1997



Please visit our Web page for submission information:
http://cc-lab.u-aizu.ac.jp/promsmmnet97/

From rem-conf-request@es.net Thu Jan 16 19:00:14 1997 
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Subject: INFOCOM'98 - Preliminary Call for Papers

   ------------------------------------------------------------------
  |         SEVENTEENTH IEEE INTERNATIONAL CONFERENCE              |
  |                         ON                                     |
  |               COMPUTER COMMUNICATIONS                          |
  |                                                                |
  |        INFOCOM'98 - Gateway to the 21st Century                |
  |              March 29 - April 2, 1998                          |
  |           Hotel Nikko, San Francisco, USA                      |
  |                                                                |
  |        Sponsored by:                                           |
  |                                                                |
  |          Computer Communications Technical Committees of the   |
  |          IEEE Computer and Communications Societies            |
  |                                                                |
  ------------------------------------------------------------------
 
  CALL FOR PAPERS:

  Authors are invited to submit full papers reporting results, strongly
  substantiated by mathematical analysis, experiment, implementation,
  or simulation, in all areas of computer communications and
  networks; however, more qualitative exploration of important 
  architectural issues is also encouraged. A more detailed list of
  specific areas of interest can be obtained from the INFOCOM'98 WWW site.

  SCHEDULE AND IMPORTANT DATES:
 
  Full Paper Due                             -    July 15, 1997
  Notification of Acceptance                 -    October 31, 1997
  Camera Ready Version of Accepted Paper Due -    January 5, 1998
  Conference                                 -    March 31-April 2, 1998
  Tutorials                                  -    March 29-30, 1998

  SUBMISSION INSTRUCTIONS:

  The first page of your paper should contain paper title, author
  name(s), complete address (telephone, fax, email) for all authors,
  indication of who is the corresponding author, abstract, followed by
  key words and optionally one or more areas of interest from the list
  on the WWW site. The paper should be limited to 20 pages excluding 
  illustrations and 30 pages overall.

  We strongly encourage electronic submission of manuscripts in
  Postscript format.  Please consult the INFOCOM '98
  WWW site for electronic submission instructions.  Hard copy
  submissions will still be handled on an exception basis, but require
  explicit permission from the Infocom'98 Technical Program Chair who 
  can be reached at:

  Professor Ian Akyildiz
  Technical Program Chair, IEEE INFOCOM '98
  School of ECE
  Georgia Tech, Atlanta, Georgia 30332.
  (email) infocom98@ee.gatech.edu 
  (phone) +1 404-894-5141
  (fax)   +1 404-894-0035 

  STUDENT GRANTS:

  You may qualify for a Student Travel Grant to defray some of your
  costs for presenting a paper at this conference.  IEEE Communications
  Society will award a limited number of Student Travel Grants to
  full-time students who represent their accepted papers at INFOCOM '98. 
  Please refer to the INFOCOM'98 WWW site for further details.

  WORLD-WIDE WEB: 

  Information on Infocom'98 is available through the World Wide Web at
  USA:           http://www.comsoc.org/~infocom98  
  Europe:        http://www.cs.ucl.ac.uk/research/infocom98
  Asia/Pacific:  http://www.iss.nus.sg/INFOCOM

  +------------------------------------------------------------------+

  INFOCOM'98 EXECUTIVE COMMITTEE:

  GENERAL CHAIR                             VICE GENERAL CHAIR

  Roch Guerin                               Henning Schulzrinne
  IBM T.J. Watson Research Center           Columbia Univ.
  Yorktown Heights, NY, USA.                New York, NY, USA.

  TECHNICAL CHAIR                           

  Ian Akyildiz
  Georgia Institute of Technology                         
  Atlanta, Georgia, USA.

  PUBLICITY CHAIR                       PUBLICATIONS CHAIR
  Ramesh Nagarajan, Bell labs, USA.     Abhijit Choudhury, Bell labs, USA.

  TUTORIAL CO-CHAIRS                    PANELS CO-CHAIRS
  Arvind Krishna, IBM, USA.             David Tipper, Univ. of Pittsburgh, USA.
  Catherine Rosenberg, Nortel, UK.      Vittorio Trecordi, CEFRIEL, Italy.

  LOCAL ARRANGEMENTS CHAIR              FINANCIAL CO-CHAIRS
  Fred Bauer, SRI, USA.                 Ariel Orda, Technion, Israel.
                                        Ioannis Korilis, Bell labs, USA.

  INFOCOM STANDING COMMITTEE CHAIR:

  Harvey Freeman, LANWORKS, Inc., U.S.A.
 
  +----------------------------------------------------------------+






From rem-conf-request@es.net Fri Jan 17 13:19:31 1997 
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Date: Fri, 17 Jan 1997 10:18:30 -0800
From: hermenh@ix.netcom.com
To: rem-conf@es.net
To: videophone@es.net
To: CU-SeeMe-L@cornell.edu
To: mbone@isi.edu
Message-Id: <1997117101858641@ix.netcom.com>
Subject: earn $75
X-Mailer: NETCOMplete v3.20, from NETCOM On-Line Communications, Inc.
MIME-Version: 1.0
Content-Type: text/plain; charset=us-ascii

Hello,

As an independent researcher I will be conducting one-on-one, face-to-face interviews with people that use desktop videoconferencing systems for personal purposes.  The interviews will be conducted either in San Jose, CA or in Sacramento, CA.

If  you have a videoconferenicing system in your home, and if you are willing to be interviewed for approximately 20 minutes (I can pay you a $75 cash incentive for your time) please contact me and tell me:
1) what system you use
2) which of the above mentioned cities is more convenient for you to come to.

Hermen


From rem-conf-request@es.net Fri Jan 17 13:55:44 1997 
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Date: Fri, 17 Jan 97 10:54:03 PST
From: deby@lan.nsc.com (Deby Worsley)
Message-Id: <9701171854.AA14871@lan.nsc.com>
To: rem-conf@es.net, videophone@es.net, CU-SeeMe-L@cornell.edu, mbone@isi.edu, 
    hermenh@ix.netcom.com
Subject: Re: earn $75

Please provide your contact details on this.


> From videophone-request@es.net Fri Jan 17 10:43:01 1997
> To: rem-conf@es.net
> To: videophone@es.net
> To: CU-SeeMe-L@cornell.edu
> To: mbone@isi.edu
> Subject: earn $75
> X-Mailer: NETCOMplete v3.20, from NETCOM On-Line Communications, Inc.
> Mime-Version: 1.0
> Content-Type> : > text/plain> ; > charset=us-ascii> 
> Content-Length: 569
> X-Lines: 10
> 
> Hello,
> 
> As an independent researcher I will be conducting one-on-one, face-to-face interviews with people that use desktop videoconferencing systems for personal purposes.  The interviews will be conducted either in San Jose, CA or in Sacramento, CA.
> 
> If  you have a videoconferenicing system in your home, and if you are willing to be interviewed for approximately 20 minutes (I can pay you a $75 cash incentive for your time) please contact me and tell me:
> 1) what system you use
> 2) which of the above mentioned cities is more convenient for you to come to.
> 
> Hermen
> 
> 

From rem-conf-request@es.net Sun Jan 19 13:44:04 1997 
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Subject: Some comments on current SDP usage
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Date: Sun, 19 Jan 1997 19:41:57 +0000
From: Christian Zahl <zahl@fokus.gmd.de>
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Hello,

I'm currently working the interpretation of SDP announcements currently
send on the mbone. It seems that currently many tools generate SDP
packets, which does NOT conform to the (still ID) specifiaction!
It seems that also sdr v2.3a1 violates this spec.

The major problem is the order of the various attribute lines. E.g.
the order of the email (e=) and phone (p=) attributes is often
wrong. Is there any chance that this will be changed in the near
future? One aim of the SDP spec (to my understanding) was to allow
to write a simple parser and to give some hints on corrupt packets
(from the ID: "... but all must appear in exactly the order given
here (the fixed order greately enhances error detection and allows
for a simple parser).")

I'm currently trying to compress the SDP packets, and for doing so
I rely on the strict order. So for the moment it seems to be necessary
to reoder the fields prior to compression... :-(

Any comments are welcome, 
Chris


From rem-conf-request@es.net Sun Jan 19 13:53:50 1997 
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To: Christian Zahl <zahl@fokus.gmd.de>
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Subject: Re: Some comments on current SDP usage
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Date: Sun, 19 Jan 1997 19:52:45 +0100
From: Peter Parnes <peppar@cdt.luth.se>

Where does the specs say that fields must be in any strict order?

/P




From rem-conf-request@es.net Mon Jan 20 02:52:36 1997 
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To: Christian Zahl <zahl@fokus.gmd.de>
From: Ross Finlayson <finlayson@lvn.com>
Subject: Re: Some comments on current SDP usage
Cc: rem-conf@es.net

As I have commented to Mark (Handley) on more than one occasion, I just
don't believe the claim that requiring the various "=" tags in SDP to appear
in a fixed order "allows for a simpler parser".  If you allow the tags to
appear in any order, then your parser can simply be a "switch" statement
inside of a "while" loop.  (Then, having completed the parsing, you then
check that all of the manditory tags are present.)  I defy anyone to come up
with a simpler parser that assumes that the tags appear in a fixed order,
while still allowing for the fact that some of them are optional.

In any case, it's generally a good idea for SDP receivers to follow the "be
liberal in what you accept" philosophy, and accept SDP descriptions that
make sense, even if their "=" tags don't appear in the exact prescribed
order.  multikit's SDP parser does this, and I assume that many SDP
receivers do as well.

With this in mind, the SDP spec should not be so restrictive.  While there
are good reasons for requiring the "v=" (version) tag to be first, and
perhaps for requiring that all of the 'session-global' tags appear before
the 'media-specific' tags, there seems no reason for the spec to be so
anal-retentive about the order of these tags.

        Ross.


From rem-conf-request@es.net Mon Jan 20 03:03:47 1997 
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From: Achim Steinacker <stein@pi4.informatik.uni-mannheim.de>
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On Sun, 19 Jan 1997, Peter Parnes wrote:

> Where does the specs say that fields must be in any strict order?
> 
> /P

SDP draft 03.2 Page 5: ".. Some lines in each description are required and
some are optional but all must appear in exactly the order given here...."

Sdr violates not only the order of the 'e'- and 'p'-fields. It also
doesn't include a 'c'-field in each session-description if there is one in
each media-description, although it is not marked optional in the spec.


From rem-conf-request@es.net Mon Jan 20 14:15:07 1997 
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Date: Mon, 20 Jan 1997 11:14:26 -0800
To: rem-conf@es.net
From: Chunrong Chad Zhu <czhu@ibeam.jf.intel.com>
Subject: revison of RTP payload format for H.263
Cc: czhu@ibeam.jf.intel.com
X-Attachments: D:\doc\ietf\rtp\rtp263id.txt;

--=====================_853816466==_
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Hi all,

Attached please find the revision of "RTP Payload Format for H.263 
Video Streams".

This revision reflected feedback from last meeting in San Jose. I removed
the version field in RTP H.263 header added in the previous version.


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

--Chad

--=====================_853816466==_
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Internet Engineering Task Force                Audio-Video Transport WG
INTERNET-DRAFT                                                   C. Zhu
                                                            Intel Corp.
                                                       January 10, 1997
                                                 Expires: July 10, 1997


               RTP Payload Format for H.263 Video Streams


Status of This Memo

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

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

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


Distribution of this document is unlimited.




Abstract 

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









Zhu                                                             [Page 1]


Internet Draft      draft-ietf-avt-payload-03.txt       January 10, 1997


1. Introduction

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

The complete specification of RTP for a particular application will 
require a profile specification document [3], a payload format 
specification, and an RTP protocol document [1]. This document is 
intended to serve as the payload format specification for H.263 video 
streams.

2. Definitions

For the purpose of this document, the following definitions apply:

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

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

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

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

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

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

MB: a macroblock (MB) relates to four blocks of luminance and the 
spatially corresponding two blocks of chrominance. Each block consists 
of 8x8 pixels.

3. Design Issues for Packetizing H.263 Bitstream

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

Zhu                                                             [Page 2]


Internet Draft      draft-ietf-avt-payload-03.txt       January 10, 1997

3.1 Optional Features of H.263
 
In addition to the basic source coding algorithms, H.263 supports four 
negotiable features to improve performance: Advanced Prediction, PB 
frames, Syntax-based Arithmetic Coding, and Unrestricted Motion Vectors.
They can be used in any combination. 
 
Advanced Prediction(AP): four motion vectors instead of one can be used 
for some macroblocks in the frame. This feature makes recovery from 
packet loss harder, because more redundant information has to be 
preserved at the beginning of the packet when fragmenting at macroblock 
boundaries.
 
PB frames:  two frames ( P frame and B frame) are coded into one 
bitstream with macroblocks from the two frames interleaved. From a 
packetization point of view, a MB from the P frame and a MB from the B 
frame must be treated together because each MB for the B frame are coded
based on the corresponding MB for the P frame. A means must be provided 
to ensure proper rendering of two frames in the right order. Also if 
part of this combined bitstream is lost, it will affect the two frames, 
and possibly more.
 
Syntax-based Arithmetic Coding (SAC): Huffman codes are not the only 
choice for variable length coding. When the SAC option is on, the 
resultant run value pair after quantization of Discrete Cosine 
Transform (DCT) coefficients will be coded differently from Huffman 
codes, but the macroblock hierarchy will be preserved. Since context 
variables are only synchronized before fixed length codes in the 
bitstream, any fragmentation at variable length codes will result in 
difficulty in decoding in the presence of packet loss without carrying 
the values of all the context variables in each packet header.  
 
Unrestricted motion vector feature also has impact on packetization 
because of larger range of motion vector than normal. 

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

3.2 GOB Numbering

In H.263, each picture is divided into groups of blocks (GOB). GOBs are 
numbered by vertical scan of the GOBs, starting with the upper GOB and 
ending with the lower GOB. In contrast, a GOB in  H.261 is composed of 
three rows of 16x16 MB for QCIF, and three half rows of MB for CIF 
format. Like H.261, a GOB is divided into macroblocks in H.263. The 
definition of a macroblock in H.263 is the same as in H.261. 

Each GOB in H.263 can have a fixed GOB header, but unlike H.261 the use 
of the header is optional. If the GOB header is present, it may or may 
not start on a byte boundary. Byte alignment can be achieved by proper 
bit stuffing by the encoder, but it is not required.

Zhu                                                             [Page 3]


Internet Draft      draft-ietf-avt-payload-03.txt       January 10, 1997


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



3.3 Motion Vectors Encoding

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

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

Let's assume a packet starts with one MB but the GOB header is not 
coded. If the previous packet is lost, then all the motion vectors 
to predict the motion vector for the MBs in this GOB are not available. 
In order to decode the received MBs correctly, all the motion vectors 
for the previous GOB or the motion vector predictors would have to be 
saved at the beginning of the packet. This would be very expensive 
and unacceptable in terms of bandwidth overhead.

The encoding strategy of each H.263 codec implementation is beyond 
the scope of this document, even though it has very significant impact 
on visual quality in the presence of packet loss. However, we strongly
recommend use of the GOB header for every GOB at the beginning of 
a packet to address these problems.  


3.4 Macroblock Address

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


Zhu                                                             [Page 4]


Internet Draft      draft-ietf-avt-payload-03.txt       January 10, 1997

4. Usage of RTP

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

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

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

4.1 RTP Header usage

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

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. 0 otherwise.
 
Payload Type (PT): The Payload Type shall specify H.263 video payload 
format.
 
Timestamp: The RTP Timestamp encodes the sampling instance of the first 
video frame contained in the RTP data packet. The RTP timestamp may be 
the same  on successive packets if a video frame occupies more than one 
packet. For H.263 video stream, the RTP timestamp is based on a 90 kHz 
clock, the same as that of RTP payload for H.261 stream [5]. 
 
4.2 Video Packet Structure

H.263 compressed bitstream is carried as payload within each RTP packet.
For each RTP packet, the RTP header is followed by the H.263 payload 
header, which is followed by the standard H.263 compressed bitstream. 
The size of the H.263 payload header is variable depending on modes 
used as detailed in the next section. The layout of the RTP H.263 
video packet is shown as:

Zhu                                                             [Page 5]


Internet Draft      draft-ietf-avt-payload-03.txt       January 10, 1997


 0                   1                   2                   3
 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                 RTP Header                                    |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                 H.263 Payload Header                          |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                 H.263 stream                                  |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+



5. H.263 Payload Header

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

Three formats (Mode A, Mode B and Mode C) are defined for RTP H.263 
payload header. In Mode A, a H.263 payload header of four bytes is 
present before actual compressed H.263 video bitstream in the packet. 
It allows fragmentation at GOB boundaries. In Mode B, a eight byte H.263
payload header is used and each packet starts at MB boundaries with the 
PB frame option off. Finally, Mode C with a 12 byte header is provided 
to support fragmentation at MB boundaries for frames that are coded with
the PB frame option. The mode is indicated by the F field and P field
in the header. The three modes can be intermixed for one compressed
frame. All client application are required to be able to receive 
packets in all three modes, but decoding of Mode C packets is optional 
because PB-frame is an optional feature of H.263 decoder.


In this section, the H.263 payload format is shown as rows of 32-bit 
word. Each word is transmitted in network byte order with the most 
significant byte shown at the right in the following diagrams.

5.1 Mode A

In this mode, H.263 bitstream will be packetized at GOB boundaries. 
In other words, each packet will start at the beginning of a GOB, and it 
can carry one or more MBs or GOBs. Only four bytes are used for the 
header. Mode A can be used with or without PB frame option. For those
GOBs that are smaller than network packet size, this mode is 
recommended. The H.263 payload header definition for Mode A is shown 
as follows with F=0.

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

Zhu                                                             [Page 6]


Internet Draft      draft-ietf-avt-payload-03.txt       January 10, 1997

F: 1 bit
The flag bit indicates the format of the header. F=0, Mode A, F=1, 
Mode B or Mode C.

P: 1 bit
Optional PB-frame mode as defined by the H.263 [4]. "0" implies normal 
I or P frame, "1" PB-frame. When F=1, P also indicates modes. Mode B 
if P=0, Mode C if P=1.
 
SBIT: 3 bits
Start bits specifies number of bits that should be ignored in 
the first data byte.
 
EBIT: 3 bits
End bits specifies number of bits that should be ignored in the 
last data byte.

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

I:  1 bit. 
Picture coding type, defined by H.263[4], "0" is intra-coded, "1" is
inter-coded.
 
U: 1 bit
Unrestricted Motion Vector mode as defined by H.263 [4], "0" off, 
"1" on.
 
S: 1 bit
Optional Syntax-based Arithmetic Code mode as defined by the H.263 [4], 
0" off, "1" on.

A: 1 bit
Optional Advanced Prediction mode as defined by H.263 [4], "0" off, 
"1" on.
 
R: 4 bits
Reserved, must be set to zero.
 
DBQ: 2 bits
Differential quantization parameter used to calculate quantizer for 
the B frame based on quantizer for the P frame, when PB frame option 
is on. The value should be the same as DBQUANT defined by the H.263 [4].
Set to zero if PB frame option is off.
 







Zhu                                                             [Page 7]


Internet Draft      draft-ietf-avt-payload-03.txt       January 10, 1997

TRB: 3 bits
Temporal Reference for the B frame as defined by the H.263 [4]. Set to 
zero if PB frame option is off.

TR: 8 bits
Temporal Reference for the P frame as defined by the H.263 [4]. Set to 
zero if the PB frame option is off.
 
5.2 Mode B

In this mode, the H.263 stream can be fragmented at MB boundaries. Thus 
necessary information is needed at the start of a packet to recover 
the decoder internal state in the presence of packet loss. It is 
intended for those GOBs whose sizes are larger than the maximum packet 
size allowed in the underlying protocol. This mode can only 
be used with PB frame option off. Mode C as defined in the next section 
can be used to fragment at MB boundaries with PB frame option on. The 
H.263 payload header definition for Mode B is shown as follows with 
F=1 and P=0:

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



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

QUANT: 5 bits
Quantization value for the first MB coded at the starting of the packet.
Set to 0 if the packet begins with a GOB header. This is the equivalent 
of GQUANT defined by the H.263 [4].
 
GOBN: 5 bits
GOB number in effect at the start of the packet. GOB number is specified
differently for different resolutions. See H.263 [4] for details.
 
MBA: 9 bits
The absolute address of the first MB within its GOB, counting from 0.
 
HMV1, VMV1: 7 bits each.
Horizontal and vertical motion vector predictors for the first MB  
in this packet[4]. When four motion vectors are used for current MB with
advanced prediction option, these would be the motion vector predictors 
for block number 1 in this MB.


Zhu                                                             [Page 8]


Internet Draft      draft-ietf-avt-payload-03.txt       January 10, 1997

HMV2, VMV2, 7 bits each.
Horizontal and vertical motion vector predictors for block number 3 
in the current MB when four motion vectors are used for this MB in 
advanced prediction mode. This is needed because block number 3 in the 
first MB needs different motion vector predictors from other blocks in 
the MB. These two fields are not used when the current MB only has one
motion vector. See the H.263 [4] for block organization in a frame.

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

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


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



The following fields are defined the same as in Mode B: F, P, SBIT, 
EBIT, SRC, QUANT, GOBN, MBA, R, I, U, S, A, HMV1, VMV1, HMV2, VMV2. The
rest of the fields (TR, DBQ, TRB) are defined the same as in Mode A, 
except field RR of 19 bits is reserved and its value must be zero.

5.4 Selection of Modes for the H.263 Payload Header

Packets with different modes can be intermixed. The modes shall be 
selected carefully based on performance characteristics, H.263 coding 
modes and underlying network protocols.

We strongly recommend use of Mode A whenever possible. The major 
advantage of Mode A over Mode B and C is its simplicity. The header is 
one half and one third of the size of Mode B and C respectively. 
Transmission overhead is reduced and the savings may be very 
significant when working with very low bit rates, especially when low 
latency is desired.



Zhu                                                             [Page 9]


Internet Draft      draft-ietf-avt-payload-03.txt       January 10, 1997


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

Mode A shall be used for packets starting with a GOB of size smaller 
than the network packet size. The major disadvantage of Mode A is lack 
of flexibility in packetization when a GOB can not fit in a network 
packet. 

Mode B has the advantage of flexibility with fragmentation at MB 
boundaries with PB frame option off. This mode is necessary when a 
GOB is larger than the network packet size. It has the disadvantage of 
higher overhead with a long header of 8 bytes. For small packets, this 
may not be desirable. Mode C is the same as B, except it allows 
fragmentation with PB option on at the price of 4 additional bytes.

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

6. 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, ITU-T Recommendation 
    H.263, 1996
 
[5] T. Turletti, C. Huitema, RTP Payload Format for H.261 Video Stream. 
    RFC 2032. 

7. Author's Address

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

Zhu                                                            [Page 10]


Internet Draft      draft-ietf-avt-payload-03.txt       January 10, 1997


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
















































Zhu                                                            [Page 11]

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From rem-conf-request@es.net Mon Jan 20 14:55:13 1997 
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Date: Mon, 20 Jan 1997 11:54:53 -0800
From: Anders Klemets <klemets@vxtreme.com>
Organization: VXtreme, Inc.
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To: Peter Parnes <peppar@cdt.luth.se>
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------------633011824D9219
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 Peter Parnes wrote:
> Where does the specs say that fields must be in any strict order?

The BNF syntax at the end of the document, (appendix A, I think) makes
it
quite clear that there is a strict order in which the fields must
appear.

Anders


------------633011824D9219
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<HTML><BODY>

<DT>&nbsp;Peter Parnes wrote:<BR>
&gt; Where does the specs say that fields must be in any strict order?</DT>

<DT>&nbsp;</DT>

<DT>The BNF syntax at the end of the document, (appendix A, I think) makes
it</DT>

<DT>quite clear that there is a strict order in which the fields must appear.</DT>

<DT>&nbsp;</DT>

<DT>Anders</DT>

<DT>&nbsp;</DT>

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ADR:;;701 Welch Rd., Building C;Palo Alto;CA;94086
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From rem-conf-request@es.net Mon Jan 20 16:57:35 1997 
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Date: Mon, 20 Jan 1997 16:56:53 -0500 (EST)
From: Videoduck1@aol.com
Message-ID: <970120164835_504954690@emout05.mail.aol.com>
To: rem-conf@es.net, videophone@es.net
Subject: UK ISDN Videoconferencing

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

Three demo numbers are available to H.320 ISDN videoconferencing users in the
UK
0897 596001
0891 517488
0891 517488

Hope you like them.


From rem-conf-request@es.net Mon Jan 20 20:47:08 1997 
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From rem-conf-request@es.net Tue Jan 21 04:16:07 1997 
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Date: Tue, 21 Jan 1997 10:17:39 +0100
From: Peter Hoermann <peha@darmstadt.gmd.de>
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Hi everybody,

I'm looking for a solution to bring Internet and ISDN videoconferences
together. Has anyone already experiences in this area ? And furthermore:
has one of you already experienced RADVision's H.320/H.323 Gateway(s) ?
Are there maybe products of other companies that are solving the same
problem ?
Hope to hear from you. Thanks in advance.

Peter Hoermann

From rem-conf-request@es.net Tue Jan 21 04:57:10 1997 
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To: Peter Hoermann <peha@darmstadt.gmd.de>
cc: merci@cs.ucl.ac.uk, rem-conf@es.net
Subject: Re: Gateways
In-reply-to: Your message of "Tue, 21 Jan 1997 10:17:39 +0100." <32E489B3.59EB@darmstadt.gmd.de>
Mime-Version: 1.0
Content-Type: text/plain; charset=us-ascii
Date: Tue, 21 Jan 1997 09:56:01 +0000
Message-ID: <10045.853840561@cs.ucl.ac.uk>
From: Roy Bennett <R.Bennett@cs.ucl.ac.uk>

Peter
 >Hi everybody,
 >
 >I'm looking for a solution to bring Internet and ISDN videoconferences
 >together. Has anyone already experiences in this area ? And furthermore:

This solution is being addressed in our project MERCI (Multimedia European 
Research Conferencing Integration) - a successor to the MICE project and 
also funded by the European Union. 

You can find information on our plans at our web site
   http://www-mice.cs.ucl.ac.uk/merci/
Most of the work on this aspect of the project is being done by our partner
in Berlin, TELES AG, which specialises in ISDN videoconferencing.

We expect to have an initial H323/MBONE G/W to allow a single MBONE terminal
to join an H320 conference by the end of next month. After that we will
develop the gateway further to allow multiple participants on each side.

 >has one of you already experienced RADVision's H.320/H.323 Gateway(s) ?
 >Are there maybe products of other companies that are solving the same
 >problem ?
I, personally, have not seen this G/W.
 >Hope to hear from you. Thanks in advance.
 >
 >Peter Hoermann

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



From rem-conf-request@es.net Tue Jan 21 06:05:30 1997 
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From: Ross Finlayson <finlayson@lvn.com>
Subject: New mailing list of interest to many

Many of the subscribers to "rem-conf" may also be interested in a mailing
list - described below - that has recently been created.

To subscribe: Send email to "mafp-request@lists.best.com", with the word
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in the BODY of the message.

Description of the mailing list:
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The "mafp@lists.best.com" mailing list is an open forum for the discussion
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applications.  (MAFP stands for "multicast attribute framing protocol".)

A large class of distributed applications involve the dynamic sharing,
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The primary goal of this mailing list is to help achieve consensus on one or
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format could conceivably be standardized by the IETF.

A document describing one such format has been published as an Internet
Draft: "draft-finlayson-mafp-00.txt".  (This document can also be found
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are encouraged to comment on this draft, and/or submit additional documents
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        Ross Finlayson
        finlayson@lvn.com


From rem-conf-request@es.net Tue Jan 21 08:26:52 1997 
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From: Amotz Shemi <amotz@radvision.rad.co.il>
To: 'Peter Hoermann' <peha@darmstadt.gmd.de>
Cc: "'rem-conf@es.net'" <rem-conf@es.net>
Subject: RE: Gateways
Date: Tue, 21 Jan 1997 15:26:35 +0200
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Peter,=20

RADVision's LAN to WAN gateway is in the market for about two years. It =
connects four LAN segments with four WAN ports (V.35, ISDN or leased =
lines).  The next H.323/H.320 GW version has passed interop tests and =
now in a beta phase (more info in www.radvision.com - am not one of the =
marketing folks)

Amotz Shemi
RADVision
amotz@radvision.rad.co.il


----------
From: 	Peter Hoermann[SMTP:peha@darmstadt.gmd.de]
Sent: 	Tuesday, January 21, 1997 11:17 AM
To: 	rem-conf@es.net
Subject: 	Gateways

Hi everybody,

I'm looking for a solution to bring Internet and ISDN videoconferences
together. Has anyone already experiences in this area ? And furthermore:
has one of you already experienced RADVision's H.320/H.323 Gateway(s) ?
Are there maybe products of other companies that are solving the same
problem ?
Hope to hear from you. Thanks in advance.

Peter Hoermann




From rem-conf-request@es.net Tue Jan 21 09:47:00 1997 
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Subject: Re: Gateways
To: peha@darmstadt.gmd.de
Date: Tue, 21 Jan 1997 15:45:59 +0100 (MET)
Cc: cabo@informatik.uni-bremen.de (Carsten Bormann), rem-conf@es.net
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Hi Peter,

we doing a great deal of work in the area of conferencing
interoperability for ITU-T H.320/H.324, H.323, and systems based on IETF
MMUSIC/AVC documents (part of which is done in the context of the MERCI
project).

TELES is offering products in this area, at the university we are doing
research and protocol development (together with the University of Bremen).
Send me an e-mail, if you like further information.

Gruesse,
Joerg

-------------------------------------------------------------------------------
Joerg Ott                                                    jo@cs.tu-berlin.de
Technische Universitaet Berlin, Germany                  fax + 49 30 314-25 156
TELES AG                                               voice + 49 30 314-73 389


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From rem-conf-request@es.net Tue Jan 21 10:54:23 1997 
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From rem-conf-request@es.net Tue Jan 21 12:42:36 1997 
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To: rem-conf@es.net, Peter Hoermann <peha@darmstadt.gmd.de>
Subject: Re: Gateways

     Peter,
     
     Try VideoServer.  You can find their web page at www.videosever.com
     
     Regards,
     
     Jim Hirni


______________________________ Reply Separator _________________________________
Subject: Gateways
Author:  Peter Hoermann <peha@darmstadt.gmd.de> at InterNet
Date:    1/21/97 8:12 AM


Hi everybody,
     
I'm looking for a solution to bring Internet and ISDN videoconferences 
together. Has anyone already experiences in this area ? And furthermore: 
has one of you already experienced RADVision's H.320/H.323 Gateway(s) ? 
Are there maybe products of other companies that are solving the same 
problem ?
Hope to hear from you. Thanks in advance.
     
Peter Hoermann


From rem-conf-request@es.net Tue Jan 21 13:01:07 1997 
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To: rem-conf@es.net
From: ari@sprintlabs.com (Ari Ollikainen)
Subject: How to UNSUBSCRIBE from rem-conf


        I humbly suggest that the recent unsubscribers try to do it the
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        The list maintainers do NOT read the messages sent to the
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        Back when I ran/maintained this list, I used to post a periodic
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        Please do NOT send *me* any requests for changes...I've moved on...

           _/_/   _/_/_/_/    _/  Ari Ollikainen <ari@sprintlabs.com>
        _/  _/   _/     _/   _/ Sprint Advanced Technology Laboratories
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~~RECOM Technologies Inc.~~ Voice: 415.375.4265 FAX: 415.375.4490



From rem-conf-request@es.net Tue Jan 21 13:36:40 1997 
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From: Jerrlyn Iwata <jerrlyn@postgres.berkeley.edu>
Subject: Berkeley Multimedia & Graphics Seminar

                Berkeley Multimedia and Graphics Seminar 
        (Wednesday January 22, 1997 12:30-2:00 PDT 405 Soda Hall) 

             "Music Delivery and Promotion on the Internet" 

                            Philip R. Wiser 
                             Liquid Audio 

The promise of commercial music distribution via the Internet is ready to be
fulfilled. This talk will describe the technical and legal complexities of
online music distribution. It will also present solutions to current online
distribution barriers. 

Music is a natural product for online delivery. Digital representations are
of desirable quality and only marginal value is added by physical packaging.
In fact, delivery of musical information via the Internet has been available for
many years. None of this content, however, has been distributed
commercially. It is only recently that large amounts of easily accessible,
high quality content has been exposed. The natural progression is to
commercially distribute this information electronically. 

Several technologies are required to make digital music distribution a
reality. Audio compression, embedded digital signalling, and cryptography
provide the framework for these systems. In addition, commerce and rights
reporting
components must be implemented. This discussion will address all of these
features. 

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

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


From rem-conf-request@es.net Tue Jan 21 13:56:14 1997 
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Date: Tue, 21 Jan 1997 10:53:45 -0800
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unsubscribe

From rem-conf-request@es.net Tue Jan 21 15:17:18 1997 
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Date: Tue, 21 Jan 1997 15:14:42 -0500 (EST)
From: Bob Dixon <rdixon@stargate.acs.ohio-state.edu>
To: rem-conf@es.net
Subject: Re: Gateways
Message-ID: <Pine.SOL.3.91.970121151104.13065R-100000@stargate.acs.ohio-state.edu>
MIME-Version: 1.0
Content-Type: TEXT/PLAIN; charset=US-ASCII

 >Hi everybody,
 >
 >I'm looking for a solution to bring Internet and ISDN videoconferences
 >together. Has anyone already experiences in this area ? And furthermore:


In addition to the RADVision research project I mentioned earlier,
we are also looking into doing this using a Mac as a gateway.
Feed the Mac baseband video and audio, and have the Mac running any desired
IP videoconferencing software. We use this now to send video
>from our campus TV station and satellite dishes to people on the
Internet, using CU-SeeMe.


                                     Bob Dixon
                                     Ohio State University

From rem-conf-request@es.net Tue Jan 21 16:40:07 1997 
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Date: Tue, 21 Jan 1997 13:42:28 -0800
To: Jim_Hirni@multilink.com
From: ari@sprintlabs.com (Ari Ollikainen)
Subject: Re: Gateways
Cc: rem-conf@es.net


>     Try VideoServer.  You can find their web page at www.videosever.com
>

        Could you cite specific products? The only references to H.323
        and "Internet" are the press releases dealing with Videoserver's
        relationships with Compaq and cisco:

        12/11/96

        "...the combined Compaq and VideoServer conferencing solution will
        integrate the Compaq ProLiant line of servers and VideoServer hardware
        modules and software. The VideoServer conferencing modules will
        consist of audio, video and data processing units, H.323 gateway
        solutions, and ISDN and Ethernet connectivity.
        Customers will be able to add an optional module for Continuous
        Presence, which enables up to four conference sites to be viewed
        simultaneously. Initially, two base configurations will be available:
        a four-user or eight-user conferencing system. In addition, either
        configuration can be upgraded to support up to 48 users.
        The joint VideoServer/Compaq collaboration and conferencing systems
        will augment Compaq's existing videoconferencing terminal offerings...
        "

        and

        10/28/96

        "...The collaboration mobilizes the emerging videoconferencing
        standard H.323, making conferencing available to a broader range of
        customers. Joining the strengths of Cisco IOS software and
        VideoServer's conferencing infrastructure capabilities, improves
        H.323-based multimedia connectivity . The two companies will integrate
        VideoServer's gateway software and hardware modules into a wide range
        of Cisco platforms.

        "We are committed to delivering a full array of network services
        through Cisco IOS software and this partnership is an important step
        in the evolution of real-time, multimedia conferencing solutions,"
        said Stephen DeWitt, vice president of Cisco IOS software marketing at
        Cisco Systems. "The results of this joint effort will contribute to
        making videoconferencing a mainstream application for corporations."


        "This is a business of standards," said Robert Castle, president of
        VideoServer, Inc. "Cisco and VideoServer share this belief and have
        put standards support and interoperability at the top of our product
        requirements. Working with the major endpoint manufacturers, the two
        companies have already tested key elements of standards support and we
        are working to help make the combinations of endpoints, routers, and
        conferencing gateways a seamless solution that fits the corporate
        network. Together we will build software capabilities which
        incorporate key elements of the ITU standard H.323 for LAN
        conferencing."

        Under the understanding, Cisco will OEM gateway modules from
        VideoServer to enable conferencing between LAN and WAN endpoints.
        VideoServer is demonstrating the gateway this week at Telecon in
        Anaheim, CA the leading annual exposition for conferencing products.
        ..."

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



From rem-conf-request@es.net Tue Jan 21 18:07:58 1997 
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Message-ID: <32E550C3.BAC@ctr.columbia.edu>
Date: Tue, 21 Jan 1997 18:26:59 -0500
From: Raymond Liao <liao@ctr.columbia.edu>
Organization: CTR, Columbia University
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MIME-Version: 1.0
To: rem-conf@es.net
CC: mnl@ctr.columbia.edu
Subject: Comet Group Seminar, CTR, Columbia Univ.
References: <199701200341.MAA17811@tsbgw.wide.toshiba.co.jp> <32E2F762.7D52@ctr.columbia.edu> <32E36728.677C@cs.columbia.edu> <32E3A781.1556@ctr.columbia.edu> <32E3DD1B.41C67EA6@ctr.columbia.edu> <32E4245C.3A30@cs.columbia.edu>
Content-Type: text/plain; charset=us-ascii
Content-Transfer-Encoding: 7bit

Hi there:

There is going to be a live broadcast on Mbone for the seminar 
of the Comet Group at the Center for Telecommunications Research, 
Columbia University, New York City.

The seminar is held weekly on Thursday from 10 AM to 11 AM EST, 
with the exception of this week which will be on Friday, Jan. 24th.

The audio part will be PCM encoded, through RTP, IP address is
224.2.131.73/25792, ttl=127
The video part will be H.261 encoded, through RTP, IP address is
224.2.189.122/57242, ttl=127.

The seminar home page is: 
http://comet.ctr.columbia.edu/activities/seminars/

Best,
Raymond. 
                               ?????
                              \ @ @ /      
---------------------------.oOO--0--OOo.---------------------------
Raymond R.-F. Liao                  Email: liao@ctr.columbia.edu
Center for Telecomm. Research         Web: www.ctr.columbia.edu/~liao
Columbia University                 Phone: (212) 939-7158
New York City, NY10027-6699           Fax: (212) 316-9068 
___________________________.oooO___Oooo.___________________________

From rem-conf-request@es.net Tue Jan 21 18:11:34 1997 
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Date: Tue, 21 Jan 97 18:13:57 PST
Encoding: 84 Text
Message-Id: <9700218538.AA853899780@multimail.multilink.com>
To: ari@sprintlabs.com (Ari Ollikainen)
Cc: rem-conf@es.net
Subject: Re[2]: Gateways

     Ari,
     
     I'm not a Videoserver employee, but I bet if you call their Burlington Ma 
     number (617-229-2000) and ask for sales, someone there will be happy to 
     talk with you about specific producs.
     
     Regards,
     
     Jim Hirni


______________________________ Reply Separator _________________________________
Subject: Re: Gateways
Author:  ari@sprintlabs.com (Ari Ollikainen) at InterNet
Date:    1/21/97 6:12 PM


>     Try VideoServer.  You can find their web page at www.videosever.com 
>
     
        Could you cite specific products? The only references to H.323 
        and "Internet" are the press releases dealing with Videoserver's 
        relationships with Compaq and cisco:
     
        12/11/96
     
        "...the combined Compaq and VideoServer conferencing solution will 
        integrate the Compaq ProLiant line of servers and VideoServer hardware 
        modules and software. The VideoServer conferencing modules will 
        consist of audio, video and data processing units, H.323 gateway 
        solutions, and ISDN and Ethernet connectivity.
        Customers will be able to add an optional module for Continuous 
        Presence, which enables up to four conference sites to be viewed 
        simultaneously. Initially, two base configurations will be available: 
        a four-user or eight-user conferencing system. In addition, either 
        configuration can be upgraded to support up to 48 users.
        The joint VideoServer/Compaq collaboration and conferencing systems 
        will augment Compaq's existing videoconferencing terminal offerings... 
        "
     
        and
     
        10/28/96
     
        "...The collaboration mobilizes the emerging videoconferencing 
        standard H.323, making conferencing available to a broader range of 
        customers. Joining the strengths of Cisco IOS software and 
        VideoServer's conferencing infrastructure capabilities, improves 
        H.323-based multimedia connectivity . The two companies will integrate 
        VideoServer's gateway software and hardware modules into a wide range 
        of Cisco platforms.
     
        "We are committed to delivering a full array of network services 
        through Cisco IOS software and this partnership is an important step 
        in the evolution of real-time, multimedia conferencing solutions," 
        said Stephen DeWitt, vice president of Cisco IOS software marketing at 
        Cisco Systems. "The results of this joint effort will contribute to 
        making videoconferencing a mainstream application for corporations."
     
     
        "This is a business of standards," said Robert Castle, president of 
        VideoServer, Inc. "Cisco and VideoServer share this belief and have put 
        standards support and interoperability at the top of our product 
        requirements. Working with the major endpoint manufacturers, the two 
        companies have already tested key elements of standards support and we 
        are working to help make the combinations of endpoints, routers, and 
        conferencing gateways a seamless solution that fits the corporate 
        network. Together we will build software capabilities which incorporate 
        key elements of the ITU standard H.323 for LAN conferencing."
     
        Under the understanding, Cisco will OEM gateway modules from 
        VideoServer to enable conferencing between LAN and WAN endpoints. 
        VideoServer is demonstrating the gateway this week at Telecon in 
        Anaheim, CA the leading annual exposition for conferencing products. 
        ..."
     
           _/_/   _/_/_/_/    _/  Ari Ollikainen <ari@sprintlabs.com>
        _/  _/   _/     _/   _/ Sprint Advanced Technology Laboratories
     _/_/_/_/   _/_/_/_/    _/ Switching and Interworking Development
   _/     _/   _/     _/   _/ 1 Adrian Court (M/S: CABURS0102)
 _/      _/   _/       _/ _/ Burlingame, CA 94010
~~RECOM Technologies Inc.~~ Voice: 415.375.4265 FAX: 415.375.4490
     
     


From rem-conf-request@es.net Wed Jan 22 05:36:53 1997 
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          Wed, 22 Jan 97 11:36:53 +0100
From: richter@ro2.informatik.uni-hamburg.de (Jan-Peter Richter)
Message-Id: <9701221036.AA01657@ro2.informatik.uni-hamburg.de>
Subject: Re: How to UNSUBSCRIBE from rem-conf
To: rem-conf@es.net
Date: Wed, 22 Jan 1997 11:36:15 +0100 (MET)
X-Mailer: ELM [version 2.4 PL25]
Content-Type: text


Hi,

being bugged by an ever increasing number of 'unsubscribe' messages to
the list, Juergen Schoenwaelder - the maintainer of another mailing list -
has decided to append automatically the following message to _every_ 
posting:
(quote)
--
!! This message is brought to you via the `tkined & scotty' mailing list.
!! Please do not reply to this message to unsubscribe. To subscribe or
!! unsubscribe, send a mail message to <tkined-request@ibr.cs.tu-bs.de>.
!! See http://wwwsnmp.cs.utwente.nl/~schoenw/scotty/ for more information.
(unqoute)

Something similar should be done with the rem-conf distributions also!

JPR

------------------------------------------------------------------------------
Jan-Peter Richter				Vogt-Koelln-Str.30
Universitaet Hamburg				D-22527 Hamburg
Fachbereich Informatik		       		Tel: +49 40 5494-2350
Arbeitsgruppe Telekommunikation        		Fax: +49 40 5494-2328 
und Rechnernetze

e-mail:	richter@informatik.uni-hamburg.de
------------------------------------------------------------------------------

From rem-conf-request@es.net Wed Jan 22 14:18:31 1997 
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From: Joe Metzger <metzger@arioch.ams.ameslab.gov>
Message-Id: <9701221317.ZM8885@arioch.ams.ameslab.gov>
Date: Wed, 22 Jan 1997 13:17:50 -0600
In-Reply-To: richter@ro2.informatik.uni-hamburg.de (Jan-Peter Richter) "Re: How to UNSUBSCRIBE from rem-conf" (Jan 22, 11:36am)
References: <9701221036.AA01657@ro2.informatik.uni-hamburg.de>
Organization: ESnet
X-Phone: (515)294-1918
X-Fax: (515)294-4491
X-Mailer: Z-Mail (3.2.0 26oct94 MediaMail)
To: rem-conf@es.net
Subject: Upcoming changes to rem-conf
Mime-Version: 1.0
Content-Type: text/plain; charset=us-ascii

Hello,
Currently rem-conf@es.net is a manually maintained list.  Mail sent to
rem-conf-request@es.net is forwarded to a real person who updates the
distribution list.  Mail sent to rem-conf@es.net is not sent to the
maintainer.  This list currently contains over a thousand subscribers.
 Around 100 of those subscribers appears to be re-distribution lists at
other sites.

As part of a larger task to update the ESnet email infrastructure, I am
evaluating mailing list software to take over this task.  The goal is to
increase the level of service provided and to reduce the human resources
required to maintain lists.

Several of the mailing list systems I am looking at contain mechanisms to
detect and prevent mailing loops.  Some of these systems will have
problems if a subscriber is actually another list managed by the same
software. If you are currently running a rem-conf re-distribution list
with automated software, please let me know.  Send these notes to
METZGER@ES.NET.  NOT TO THE LIST.

I will be providing additional updates before we actually switch this
list over to a new system.  If you have any questions or comments, please
send mail to metzger@es.net.

Thanks.


*************************************************************************
* Joe Metzger                          Ames Office: (515)294-1939       *
* Computer Systems Engineer            LBNL Office: (510)486-5740       *
* Network Information Services Group   metzger@es.net                   *
* ESnet, Lawrence Berkeley Laboratory  314 Wilhelm, ISU, Ames IA, 50011 *
*************************************************************************

From rem-conf-request@es.net Thu Jan 23 12:31:19 1997 
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Original-From: ramesh@hoserve.ho.lucent.com
Message-Id: <9701231502.AA10193@hopaw33>
To: rem-conf@es.net
Subject: IEEE INFOCOM'98 - Advance Call for Papers

  Colleagues,
 
  Please consider submitting your work to INFOCOM'98. Apologies
  for multiple copies and please feel free to distribute this call
  for papers by any means appropriate. Thanks.

  Ramesh.


  ------------------------------------------------------------------
  |     INFOCOM'98 - SEVENTEENTH IEEE INTERNATIONAL CONFERENCE     |
  |                         ON                                     |
  |               COMPUTER COMMUNICATIONS                          |
  |                                                                |
  |             Gateway to the 21st Century                        |
  |                                                                |
  |              March 29 - April 2, 1998                          |
  |           Hotel Nikko, San Francisco, USA                      |
  |                                                                |
  |        Sponsored by:                                           |
  |                                                                |
  |          Computer Communications Technical Committees of the   |
  |          IEEE Computer and Communications Societies            |
  |                                                                |
  ------------------------------------------------------------------
 
  CALL FOR PAPERS:

  Authors are invited to submit full papers reporting results, strongly
  substantiated by mathematical analysis, experiment, implementation,
  or simulation, in all areas of computer communications and
  networks; however, more qualitative exploration of important 
  architectural issues is also encouraged. A more detailed list of
  specific areas of interest can be obtained from the INFOCOM'98 WWW site.

  SCHEDULE AND IMPORTANT DATES:
 
  Full Paper Due                             -    July 15, 1997
  Notification of Acceptance                 -    October 31, 1997
  Camera Ready Version of Accepted Paper Due -    January 5, 1998
  Conference                                 -    March 31-April 2, 1998
  Tutorials                                  -    March 29-30, 1998

  SUBMISSION INSTRUCTIONS:

  The first page of your paper should contain paper title, author
  name(s), complete address (telephone, fax, email) for all authors,
  indication of who is the corresponding author, abstract, followed by
  key words and optionally one or more areas of interest from the list
  on the WWW site. The paper should be limited to 20 pages excluding 
  illustrations and 30 pages overall.

  We strongly encourage electronic submission of manuscripts in
  Postscript format.  Please consult the INFOCOM '98
  WWW site for electronic submission instructions.  Hard copy
  submissions will still be handled on an exception basis, but require
  explicit permission from the Infocom'98 Technical Program Chair who 
  can be reached at:

  Professor Ian Akyildiz
  Technical Program Chair, IEEE INFOCOM '98
  School of ECE
  Georgia Tech, Atlanta, Georgia 30332.
  (email) infocom98@ee.gatech.edu 
  (phone) +1 404-894-5141
  (fax)   +1 404-894-0035 

  STUDENT GRANTS:

  You may qualify for a Student Travel Grant to defray some of your
  costs for presenting a paper at this conference.  IEEE Communications
  Society will award a limited number of Student Travel Grants to
  full-time students who represent their accepted papers at INFOCOM '98. 
  Please refer to the INFOCOM'98 WWW site for further details.

  WORLD-WIDE WEB: 

  Information on Infocom'98 is available through the World Wide Web at
  USA:           http://www.comsoc.org/~infocom98  
  Europe:        http://www.cs.ucl.ac.uk/research/infocom98
  Asia/Pacific:  http://www.iss.nus.sg/INFOCOM

  +------------------------------------------------------------------+

  INFOCOM'98 EXECUTIVE COMMITTEE:

  GENERAL CHAIR                             VICE GENERAL CHAIR

  Roch Guerin                               Henning Schulzrinne
  IBM T.J. Watson Research Center           Columbia Univ.
  Yorktown Heights, NY, USA.                New York, NY, USA.

  TECHNICAL CHAIR                           

  Ian Akyildiz
  Georgia Institute of Technology                         
  Atlanta, Georgia, USA.

  PUBLICITY CHAIR                       PUBLICATIONS CHAIR
  Ramesh Nagarajan, Bell labs, USA.     Abhijit Choudhury, Bell labs, USA.

  TUTORIAL CO-CHAIRS                    PANELS CO-CHAIRS
  Arvind Krishna, IBM, USA.             David Tipper, Univ. of Pittsburgh, USA.
  Catherine Rosenberg, Nortel, UK.      Vittorio Trecordi, CEFRIEL, Italy.

  LOCAL ARRANGEMENTS CHAIR              FINANCIAL CO-CHAIRS
  Fred Bauer, SRI, USA.                 Ariel Orda, Technion, Israel.
                                        Yannis Korilis, Bell labs, USA.

  INFOCOM STANDING COMMITTEE CHAIR:

  Harvey Freeman, LANWORKS, Inc., U.S.A.
 
  +----------------------------------------------------------------+






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To: rem-conf@es.net
From: joe <joe@msri.org>
Reply-to: joe@msri.org
Subject: Bruce Sagan on Group Representations and Symmetric Functions, Part I

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

Title:       
	Bruce Sagan on Group Representations and Symmetric Functions, Part I
Date:        
	Jan 28, 1997

Time:        
	12:00 PST8PDT 1 hours

Contact:     
	joe@msri.org

URL:         
	http://www.msri.org/events/introLN.html

Description:        
	Bruce Sagan's first lecture from  the MSRI Introductory Lecture Series in Combinatorics,  August 12 - 23, 1996









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

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From: neal <neal@msri.org>
Reply-to: neal@msri.org
Subject: Bruce Sagan on Group Representations and Symmetric Functions, Part II

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

Title:       
	Bruce Sagan on Group Representations and Symmetric Functions, Part II
Date:        
	Jan 29, 1997

Time:        
	12:00 PST8PDT 1 hours

Contact:     
	neal@msri.org

URL:         
	http://www.msri.org/events/introLN.html 

Description:        
	Bruce Sagan's second lecture lecture from  the MSRI Introductory Lecture Series in Combinatorics,  August 12 - 23, 1996









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

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To: rem-conf@es.net
From: neal <neal@msri.org>
Reply-to: neal@msri.org
Subject: Bruce Sagan on GroupRepresentations and Symmetric Functions, Part III

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

Title:       
	Bruce Sagan on GroupRepresentations and Symmetric Functions, Part III
Date:        
	Jan 30, 1997

Time:        
	12:00 PST8PDT 1 hours

Contact:     
	neal@msri.org

URL:         
	http://www.msri.org/events/introLN.html

Description:        
	Bruce Sagan's third lecture from  the MSRI Introductory Lecture Series in Combinatorics,  August 12 - 23, 1996









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

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MBone Broadcast Announcement

Title:  
    CSR (Cell Switch Router):  High Performance Router
    for Realtime Internet and Intranet

Speaker:  
    Hiroshi Esaki
    Chief Architect, CSR Project
    Computer & Network Division
    Toshiba Corporation, Japan

Date:  
    Friday, January 24, 1997

Time:  
    10:00 A.M. - 12:00 noon EST

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

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

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

Abstract:

Architecture overview of Cell Switch Router (CSR) is presented.  CSR
contains cell switch fabric and IP packet switch fabric to achieve high
throughput IP forwarding.  CSR can operate both with PVC (Permanent
Virtual
Connection) and with SVC as the VC for cut-thru packet forwarding.  With
cut-thru packet forwarding, CSR does not need to analyze the IP header
to
forward packet but uses VPI/VCI information for packet forwarding. 
Since
the CSR is router, CSR has full functionality of existing router.  For
example, the packet routing path can be dynamically changed according to
the network status, even regarding cut-thru packet forwarding.  Similar
architecture models are proposed by many researchers.  The architectural
difference and the technical features will also be discussed in the
presentation.

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Date: Fri, 24 Jan 1997 14:34:42 +1100 (EST)
From: Anna Cicognani <anna@arch.usyd.EDU.AU>
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To: rem-conf@es.net
Subject: VC97 - Mbone Broadcast
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Title:  

	VC97 International Conference on Creative Collaboration in Virtual
	Communities

Start: 13 february 1997, 21.00 GMT
End:   15 february 1997, 08.00 GMT

Test session from 11.2.97, 23.00 GMT to 12.2.97, 06.00 GMT

Contact:  

	Mrs. Anna Cicognani <anna@arch.usyd.edu.au> - Chair

URL:     

	http://www.arch.usyd.edu.au/kcdc/conferences/VC97

Place:  
	Sydney University
	Faculty of Architecture
	Key Centre of Design Computing


Abstract:

VC97 is a leading Conference about VCs and virtual enviroments. Keynote
speakers include: Howard Rheingold and Stewart Brand. Topics from politics
in cyberspace, to design an architecture of virtual environments.
Scenario planning at the end of the Conference.


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From: Bernard Aboba <aboba@internaut.com>
To: "'rem-conf@es.net'" <rem-conf@es.net>
Subject: New mailing list on multicasting of Web content
Date: Fri, 24 Jan 1997 06:51:48 -0800
Encoding: 45 TEXT

New mailing list on multicasting Web content
 
A new mailing list has been set up for discussing the distribution of
Web content via "unreliable multicast." To
subscribe, send a message to "www-multicast-request@w3.org", and put
"subscribe" in the subject line. The archive of the mailing list is
available as http://lists.w3.org/Archives/Public/www-multicast/, and
the FTP server (used for archiving of related documents) is available at
ftp://ftp.zilker.net/pub/mailcom/WWW-MULTICAST/. 
 
The primary initial goals of this list will be to produce an Internet
Draft reviewing existing implementations of unreliable multicasting of
Web content, as well as a draft reviewing potential applications and
their requirements. Should an IETF working group be
formed in this area, it is envisioned that this list would
subsequently be used for discussions relating to working group
business.
 
Many Web sites that distribute popular, frequently updated content
experience problems with "flash crowds": thousands of access per
minute to the same Web page overload the Web server or the network
link leading to the server. It is possible that integrating
IP multicast with the transport protocol used by the Web (HTTP) may
solve this problem.
 
This mailing list focusses on unreliable multicasting of Web content
due to the current immaturity of reliable multicast protocols (see
"IETF Criteria For Evaluating Reliable Multicast Transport and
Application Protocols", A. Mankin, A. Romanow, 11/27/1996
(ftp://ietf.org/Internet-drafts/draft-mankin-reliable-multicast-00.txt).).
Reliable multicast protocols will be dealt with by a research group
within the IRTF (Internet Research Task Force).
 
Unreliable multicast is better understood. Moreover, there is
considerable interest in the Web community to use this technology,
both to solve serious existing operational problems like "flash
crowds", and to avoid creating new problems due to "server push"
applications. Several applications (i.e. cache stuffing) appear to be
naturally amenable to use of unreliable multicast. Thus, moving ahead
on unreliable webcasting will serve to advance these applications as
well as to potentially provide short-term solutions for other
applications. The newly created mailing list will focus on  
these solutions.




From rem-conf-request@es.net Fri Jan 24 11:06:44 1997 
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Date: Fri, 24 Jan 1997 11:00:40 -0500 (EST)
From: Baldine Paul <baldine@research.att.com>
Message-Id: <199701241600.LAA25685@learnx.research.att.com>
To: rem-conf@es.net
Subject: RTP/H.263 Motion Vector Predictors
Cc: basso@research.att.com, mrc@research.att.com


In sections 5.2 and 5.3 of RTP Payload Format
for H.2363 Video Streams, references are made
to HMV1, VMV1, HMV2, VMV2, motion vector predictors
included in the packet header for type B and type C
packets. The document does not specify the representation
of these 7-bit integer quantities. Does anyone know
of any other document to clarify this point?
The ITU-T H.263 Document will not help since motion vector
predictors are not coded in the bit stream but inferred
>from the previously decoded macroblock and/or BOG or initialized
to zero, depending on the decoding context.

Please, send reply to
 Baldine-Brunel Paul
 AT&T Labs-Research
 baldine@research.att.com

From rem-conf-request@es.net Fri Jan 24 11:42:08 1997 
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To: rem-conf@es.net
cc: kling@indiana.edu, robelr@netlab.indiana.edu
Subject: Event: Cyberspace and the Future of Community
Mime-Version: 1.0
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Date: Fri, 24 Jan 1997 11:41:16 -0500
From: Allen Robel <robelr@stone51.netlab.indiana.edu>

Indiana University is pleased to provide Mbone coverage of the following event.

Cyberspace and the Future of Community
February 7, 1997
14:00 - 21:30 UTC (09:00 - 16:30 EST)
 
http://www/cs/indiana.edu/horizon/970207

Cyberspace and the Future of Community is jointly sponsored at Indiana 
University by the Computer Science department, as part of their "Horizon Days" 
series, by the Philosophy department, and by the Center for Social
Informatics (of the School of Library and Information Science), with support 
>from Sigma Chi for interdisciplinary campus meetings.

At a time when the vitality of community is in serious question in the U.S., 
some hope that Cyberspace can open up new ways for people to join together 
around common interests. Electronic mail, newsgroups, civic nets
(such as HoosierNet), chat rooms, cybercafes, moderated discussion groups, 
muds and moos -- these are just some of the ways people are starting to 
interact on-line.

How well does the new technology underwrite age-old human tendencies to 
congregate and communicate? Does it reflect something genuinely novel? Are 
they good, bad, or transformative? Can electronic forums strengthen local
community, or will the new connections be national or international? How much 
privacy must we sacrifice when we communicate on the net? What will happen to 
ethics, responsibility, and personal identity?

Come participate in a public forum with five nationally-known commentators on 
the future of community in the age of the internet.

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

9:00 a.m.	Reception

9:15 a.m.	William Mitchell
		Dean of the MIT School of Architecture and Planning;
		author of City of Bits: Space, Place and the Infobahn 
		(published on the Net)

10:45 a.m.	Geoffrey Nunberg
		Xerox Palo Alto Research Center and Stanford University;
		editor, The Future of the Book; commentator, NPR's "Fresh
		Air," Usage Editor, The American Heritage Dictionary

1:30 p.m.	Langdon Winner
		Professor of Political Science and Technology Studies, 
		Rennselaer Polytechnic Institute; author of Democracy
		in a Technological Society


3:00 p.m.	Comments and discussion with
 
		Rob Kling
		Professor, School of Library and Information Science, 
		Director, Center for Social Informatics, Indiana University
		editor of Computerization and Controversy
		
		Gregory Rawlins
		Professor of Computer Science, Indiana University;
		author of Moths to the Flame

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

Available Abstracts and Biographies:


William J. Mitchell, MIT School of Architecture and Planning

City of Bits

In the past, we always had to go places to do things; we went to work, we went 
to school, and sometimes we just went out. Now, the unfolding Digital 
Revolution has changed things; telepresence is an increasingly viable and 
attractive alternative to physical presence, and virtual spaces compete with 
physical venues as sites for many transactions. This does not mean that 
architecture becomes irrelevant, or that cities will just disappear, but it 
does suggest that we must fundamentally reconsider our concepts of place and 
community. The cities of the 21st century will be very different from those we 
know today.

Biography:

William J. Mitchell is Professor of Architecture and Media Arts and Sciences 
and Dean of the School of Architecture and Planning at the Massachusetts 
Institute of Technology.  He teaches courses and conducts research in design
theory, computer applications in architecture and urban design, and imaging 
and image synthesis.  He consults extensively in the field of computer-aided 
design and was the co-founder of a California software company.

Mitchell's most recent book, City of Bits: Space, Place and the Infobahn, 
examines architecture and urbanism in the context of the digital 
tele-communications revolution and the growing domination of software over 
materialized form.  In The Reconfigured Eye: Visual Truth in the 
Post-Photographic Era (MIT Press, 1992), Mitchell examined the social and 
cultural impact of digitally altered photographs and synthesized 
photorealistic scenes.

In addition to numerous articles, Mitchell is also the author of Digital 
Design Media, with Malcolm McCullough (Van Nostrand Reinhold, second ed., 
1995; orig. publ. 1991); The Logic of Architecture: Design, Computation, and 
Cognition (MIT Press, 1990); The Poetics of Gardens, with Charles Moore and 
William Turnbull (MIT Press, 1988); The Art of Computer Graphics Programming, 
with Robin S. Liggett and Thomas Kvan (Van Nostrand Reinhold, 1987); and 
Computer-Aided Architectural Design (Van Notrand Reinhold, 1977).  With 
Patrick Purcell and Malcolm McCullough he edited, and contributed essays to, 
The Electronic Design Studio (MIT Press, 1990).


Geoffrey Nunberg, Xerox PARC

Biography:

Geoffrey Nunberg (PhD, CUNY 1977) is a Principal Scientist at the Xerox Palo 
Alto Research Center and a professor of linguistics at Stanford University. He 
has done research on a number of aspects of natural language and on the
history and use of media; he is editor of the recent collection The Future of 
the Book (UC Press). He also does a regular language feature on the NPR 
program "Fresh Air," and is Usage Editor and Chair of the Usage Panel of the 
American Heritage Dictitionary.


Langdon Winner, Science and Technology Studies, Rennselaer Polytechnic 
Institute

Cyberlibertarian Dreams and the Future of Civil Society

The introduction of digital technology provides an occasion for transforming 
countless social practices and institutions.  At present there is a strong 
tendency to argue that events are moving in a particular direction, that the 
future will unfold in a particular way.  This view is expressed in an emerging 
ideology, cyberlibertarianism, a collection of ideas that links ecstatic 
enthusiasm for electronically mediated forms of living with right wing 
libertarian ideas about the proper definition of freedom, social life, 
economics, and politics in the years to come. But is their vision of a wired 
world truly the utopia it claims to be?  Is it a reliable guide to the choices 
before us?

Biography:

Langdon Winner is a political theorist who focuses upon social and political 
issues that surround modern technological change.  He is the author of 
Autonomous Technology, a study of the idea of "technology-out-of-control" in 
modern social thought, The Whale and the Reactor: A Search for Limits in an 
Age of High Technology, and editor of Democracy in a Technological Society.

Mr. Winner was born and raised in San Luis Obispo, California.  He received 
his B.A., M.A., and Ph.D. in political science from the University of 
California at Berkeley.  He is Professor of Political Science in the 
Department of Science and Technology Studies at Rensselaer Polytechnic 
Institute and Director of Graduate Studies in his department.  He has also 
taught at the New School for Social Research, M.I.T., the University of 
California at Santa Cruz, and the University of Leiden in the Netherlands, and 
has lectured widely throughout the United States and Europe.  In 1991-1992 he 
was visiting research fellow at the Center for Technology and Culture at the 
University of Oslo, Norway.


Rob Kling, Center for Social Informatics, School of Library and Information 
Science, Indiana University

Biography:

Rob Kling's current research focuses on the ways that computerization is a 
social process with technical elements, how intensive computerization 
transforms work, and how computerization entails many social choices.  He has 
also
studied the ways that complex information systems and expert systems are 
integrated into the social life of organizations.  He has conducted studies in 
numerous kinds of organizations, including local governments, insurance
companies, pharmaceutical firms, and hi-tech manufacturing firms.  He has 
written about the value conflicts implicit in and social consequences 
ofcomputerization which directly affect the public.  He is currently studying
the effective use of digital libraries to support research and teaching, and 
the conditions that foster effective public use of the emerging National 
Information Infrastructure ("data superhighways").

Dr. Kling is co-author of Computers and Politics: High Technology in American 
Local Governments (Columbia University Press, 1982).  He is co-editor of 
PostSuburban California: The Transformation of Postwar Orange County
(University of California Press, 1990).  Computerization and Controversy: 
Value Conflicts & Social Choices (Academic Press, 1991) examines the social 
controversies about computerization in organizations and social life,
regarding productivity, worklife, personal privacy, risks of computer systems, 
and computer ethics.  Dr. Kling is the sole editor of a substantially 
rewritten 2nd edition of Computerization and Controversy that Academic Press 
published in  April 1996.


Gregory Rawlins, Computer Science Department, Indiana University

Gregory Rawlins is associate professor of computer science. He received a BSc 
in mathematics from the University of the West Indies (1980), and an MSc in 
mathematics and a PhD in computer science from the University of Waterloo 
(1983, 1987).  He chaired the first workshop on the foundations of genetic 
algorithms in 1990, is on the editorial board of the Journal of Evolutionary 
Computation, and has published several books on computer science both for 
general audiences and for specialists in genetic algorithms and the analysis 
of algorithms.

Rawlins's research centers on computational complexity and machine learning.  
His current main passion is the investigation of self-adaptive software.  He 
is author of Compared to What?: An Introduction to the Analysis of Algorithms 
(Computer Science Press), Moths to the Flame (MIT Press), and the forthcoming 
Slaves of the Machine: The Quickening of Computer Technology (MIT Press), 
appearing in Spring 1997.


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



From rem-conf-request@es.net Fri Jan 24 17:09:29 1997 
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To: Bill Fenner <fenner@parc.xerox.com>
Cc: mbone@isi.edu, rem-conf@es.net, vvolfson@lobster.wellfleet.com
Subject: draft-ietf-idmr-traceroute-ipm-01 question
In-Reply-To: Your message of "Thu, 19 Dec 1996 14:26:33 PST." <96Dec19.142648pst.177713@crevenia.parc.xerox.com>
Date: Fri, 24 Jan 1997 17:02:50 -0500
From: Vera Volfson <vvolfson@BayNetworks.COM>


Hi,
I have a question about draft-ietf-idmr-traceroute-ipm-01 (5.4.3).

"If the Responce Address is multicast, the router MUST use a globally routable
address, if it has one "

What is the definition of 'globally routable address' ?

Thanks,
Vera

From rem-conf-request@es.net Fri Jan 24 18:39:04 1997 
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Date: Fri, 24 Jan 1997 15:38:01 -0800
To: Baldine Paul <baldine@research.att.com>
From: Chunrong Chad Zhu <czhu@ibeam.jf.intel.com>
Subject: Re: RTP/H.263 Motion Vector Predictors
Cc: rem-conf@es.net, czhu@ibeam.jf.intel.com


Each 7 bits field represents a motion vector predictor in half pixel resolution
as 2's complement. For example, if HMV1=1000000 (bits 4 through 10
repectively), it represents motion vector predictor 0.5 as speficied by
H.263, since
half-pel motion vector is often used in inter-coded frame.

I will add this clearification into the next revision of the document.
Thanks very much for pointing this out.

--Chad
 
At 11:00 AM 1/24/97 -0500, you wrote:
>
>In sections 5.2 and 5.3 of RTP Payload Format
>for H.2363 Video Streams, references are made
>to HMV1, VMV1, HMV2, VMV2, motion vector predictors
>included in the packet header for type B and type C
>packets. The document does not specify the representation
>of these 7-bit integer quantities. Does anyone know
>of any other document to clarify this point?
>The ITU-T H.263 Document will not help since motion vector
>predictors are not coded in the bit stream but inferred
>from the previously decoded macroblock and/or BOG or initialized
>to zero, depending on the decoding context.
>
>Please, send reply to
> Baldine-Brunel Paul
> AT&T Labs-Research
> baldine@research.att.com
>
>


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Message-Id: <199701251911.LAA24888@hille.msri.org>
To: rem-conf@es.net
From: donajski <donajski@sunsite.icm.edu.pl>
Reply-to: donajski@sunsite.icm.edu.pl
Subject: Donajski's Digital Gallery at ICM Warsaw University

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

Title:       
	Donajski's Digital Gallery at ICM Warsaw University
Date:        
	Jan 28, 1997

Time:        
	18:00 GMT 2 hours

Contact:     
	donajski@sunsite.icm.edu.pl

URL:         
	http://sunsite.icm.edu.pl/ddg/opening_a-v.html

Description:        
	Opening day for 'Garden of forklike paths'   exhibition with music concert by Camerata Quartet.  Event will start 28 of Jan. 97 at 6.00 pm.     From 25 of Jan we will give a daily 'like-test'   transmission.









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

From rem-conf-request@es.net Sat Jan 25 23:21:28 1997 
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Subject: CFP: Switching and Traffic Management
To: rem-conf@es.net
Date: Sun, 26 Jan 1997 15:20:44 +1100 (EST)
Reply-to: atiq@eng.monash.edu.au (Mohammed Atiquzzaman)
From: atiq@eng.monash.edu.au (Mohammed Atiquzzaman)
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                              CALL FOR PAPERS

                        SPIE International Symposium
                    Voice, Video, & Data Communications
            Conference on Performance & Control of Network System
                   3-7 November 1997, Dallas, Texas, USA

                            Special Session on
           Switching and Traffic Management in High Speed Networks

   Session Organizers: Mohammed Atiquzzaman, Monash University, Australia.
                       Bhumip Khasnabish, GTE Labs, USA.

Telecommunications in today's competitive marketplace typically involve 
multiple carriers, multiple services, varied equipment from multiple vendors, 
and multiple interconnected networks and  network systems.  The increasing 
demand in bandwidth for computing and communications has given rise to high 
speed networks.  These networks require special types of switching and 
traffic management mechanisms. This special session is intended to cover 
different aspects of high speed switching, congestion control and service 
interworking mechanisms.  Topics include (but are not limited to):

   o  Switching architectures
   o  Buffer management
   o  Congestion control mechanisms
   o  Traffic characterization, control, measurement, models and tools
   o  Prototyping experience and trial results
   o  Scheduling algorithms
   o  Service interworking (Frame Relay and ATM, TCP/IP and ATM, etc.)
   o  Performance evaluation, scalability, reliability and availability

Authors should send the following items by email to one of the session 
organizers (Hard copy of papers may be sent only if electronic submission 
is impossible).

   o  Postscript copy of full paper.
   o  Plain text file containing the title of the paper, a 250-word 
      abstract, and the author listing (principal author first) with 
      full name, affiliation, mailing address, e-mail address, phone 
      number and fax number for each author.

                        Session Organizers:

 Mohammed Atiquzzaman                     Bhumip Khasnabish
 Dept. of Elect. & Comp. Syst. Engg       GTE Labs. Inc., MS-48, Rm.4-292
 Monash University, Clayton               40 Sylvan Road,
 Melbourne 3168, Australia.               Waltham MA 02254, USA
 Tel: +61 3 9905 5383                     Tel: +1-617-466-2080
 Fax: +61 3 9905 3454                     Fax: +1-617-890-9320
 email: atiq@eng.monash.edu.au            email: bhumip@gte.com     
 http://www.monash.edu.au/~atiq           http://www1.acm.org:82/~bhumip/

                         Important Dates:

 Deadline for receipt of manuscripts:                      30 April 1997
 Notification of acceptance/rejection:                     late June 1997

Proceedings of this conference will be published and available at the 
symposium.  Length of camera-ready manuscript is limited to 12 pages  
with single-spaced text.  ASCII and Postscript versions of this 
announcement and information about the conference are available 
>from http://www.monash.edu.au/~atiq


From rem-conf-request@es.net Sun Jan 26 01:26:20 1997 
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From: anand@ece.iisc.ernet.in (SVR Anand)
Message-Id: <9701260624.AA17601@ece.iisc.ernet.in>
Subject: RTCP scalability
To: rem-conf@es.net
Date: Sun, 26 Jan 97 11:54:44 GMT+5:30
X-Mailer: ELM [version 2.3 PL11]

Hi,

	There has been a major thread of discussion on the RTCP scalability
issue. Hoping that this is still an issue and open for discussion...

To reduce the possible initial chaos in the network, as well as the receivers,
I am wondering about the possibility of reduced/truncated RTCP information 
carried in the SR and RR reports at the beginning of the session just enough
to allow for faster convergence in terms of listener participation (may be
an email), and incrementally increase the SR/RR content with time. Once the 
network sees a periodic, and predictable behaviour of the session,eventually, 
we can then think of header compression. 
	I realise that I am not in a position to point out the exact nature of
the information that can cause a minimal RTCP packet to begin with, to 
eventually have complete information.

	Please let me know if the basic idea is acceptable, and worth exploring
further. I appreciate your comments and criticism.

Regards
Anand.

From rem-conf-request@es.net Mon Jan 27 11:16:34 1997 
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Message-Id: <199701271604.IAA18537@bugs-bunny.CS.Berkeley.EDU>
X-Mailer: exmh version 1.6.9 8/22/96
From: Larry Rowe <Rowe@plateau.CS.Berkeley.EDU>
To: smoliar@pal.xerox.com, blattner@ocfkms.llnl.gov, brauer@faslab.com, 
    kelly_parker@maillink.berkeley.edu, kurt@ampex.com, 
    mm-group@bugs-bunny.CS.Berkeley.EDU, mslade@ix.netcom.com, 
    nic@gigabitnet.net, pmcc@hpl.hp.com, rem-conf@es.net, 
    sfchang@ctr.columbia.edu, sullivan@path0.its.berkeley.edu, 
    tierney@george.lbl.gov, jkeyser@linfield.edu
Subject: [Rebroadcast] Berkeley Multimedia Seminar (P. Wiser "Music Delivery 
         and Promotion on the Internet"
Reply-To: Rowe@plateau.CS.Berkeley.EDU
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Date: Mon, 27 Jan 1997 08:03:07 -0800
Sender: larry@plateau.CS.Berkeley.EDU

Hi -

Last week the broadcast of the Berkeley Multimedia and Graphics Seminar did 
not get beyond Berkeley due to a network failure about 5 minutes before the 
broadcast.  Because Phil Wiser's talk was so interesting and several remote 
viewers were already connected to watch the broadcast, we will rebroadcast his 
talk this wednesday during the seminar.  If you want to ask questions after 
the seminar, you can contact Phil at Liquid Audio (see 
http://bmrc.berkeley.edu/298/w1.html for addresses).
	Larry Rowe
-- 
Professor Lawrence A. Rowe               Internet:  Rowe@BMRC.Berkeley.EDU
Computer Science Division - EECS         Phone: 510-642-5117
University of California, Berkeley       Fax: 510-642-5615
Berkeley, CA 94720-1776
URL: http://www.bmrc.berkeley.edu/~larry

---
Berkeley Multimedia and Graphics Seminar 
(Wednesday January 29, 1997 12:30-2:00 PDT 405 Soda Hall) 

"Music Delivery and Promotion on the Internet" 
Philip R. Wiser (Liquid Audio )

The promise of commercial music distribution via the Internet is ready to be 
fulfilled. This talk will describe the technical and legal complexities of 
online music distribution. It will also present solutions to current online 
distribution barriers.

Music is a natural product for online delivery. Digital representations are of 
desirable quality and only marginal value is added by physical packaging. In 
fact, delivery of musical information via the Internet has been available for 
many years. None of this content, however, has been distributed commercially. 
It is only recently that large amounts of easily accessible, high quality 
content has been exposed. The natural progression is to commercially 
distribute this information electronically.

Several technologies are required to make digital music distribution a 
reality. Audio compression, embedded digital signalling, and cryptography 
provide the framework for these systems. In addition, commerce and rights 
reporting components must be implemented. This discussion will address all of 
these features.


From rem-conf-request@es.net Mon Jan 27 13:33:09 1997 
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Date: Mon, 27 Jan 1997 19:33:41 +0100
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Organization: Janot Entertainment
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subscribe

From rem-conf-request@es.net Mon Jan 27 15:55:38 1997 
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Date: Mon, 27 Jan 1997 15:54:01 -0500 (EST)
From: "Todd L. Montgomery" <tmont@cs.wvu.edu>
X-Sender: tmont@backus
To: rem-conf@es.net, rmp-discuss@aurora.jhuapl.edu, rm@irtf.cs.berkeley.edu, 
    floyd@ee.lbl.gov, mccanne@cs.berkeley.edu, gcast_all@gcast.com
Subject: Announcing an SRM implementation in C++
Message-ID: <Pine.SOL.3.91.970127154443.10389E@backus>
MIME-Version: 1.0
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I apologize for any duplicates of this message received, but I 
want to try and reach all the interested parties....

This announcement is also available through:
	http://research.ivv.nasa.gov/projects/RMP/GSRM.html

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
	Generic Scaleable Reliable Multicast (GSRM) Framework

GlobalCast Communications (http://www.gcast.com) announces the
beta test availability of a Scaleable Reliable Multicast protocol. 
The Generic Scaleable Reliable Multicast (GSRM) framework is a C++
library that allows a C++ application to reliably multicast data
to large groups of computers on a LAN, WAN or Internet.  It is
an adaptation and implementation of the SRM work that Steve McCanne,
Sally Floyd, Lixia Zhang, Van Jacobson, and others pioneered as 
part of the wb mbone tool[1].  

GSRM allows a developer to either use the standard stream-based
framework (SSRM), which organizes packets in a stream sent from 
each sender, or to build its own model by defining its own ALF frames 
and application data units (ADUs). In this way, GSRM provides a general
framework within which applications may define their own formats and 
protocol processing.  The flexibility of this ALF approach [2] allows 
GSRM to be applicable for a wide range of potential applications.

GSRM provides a stream-based, best effort reliable multicast 
system called Stream-based SRM (SSRM). SSRM is built on top of the
GSRM framework and provides a partial ALF model that the application
can customize to fits its needs. Most applications will not need to
use anything more than SSRM. However, the framework fully allows
new ALF models to be constructed from scratch.

Some of GSRMs features:
- Supports a connection-less, best-effort, reliable multicast group
	communication model over a WAN.
- Supports a generic notion of application level framing (ALF) of 
	data into application data units (ADUs) as well as support 
	for integrated layer processing (ILP) of ADUs.
- Decouples the SRM request/repair policy on ADUs from the
	ALF model.
- Provides a rate-based flow control mechanism.

Some of SSRMs features:
- Supports a stream-based, best-effort, push model reliable multicast
	delivery system with the following assumptions of ordering, 
	message stability, and group communication models:
	- Ordering requirements of a stream are application defined
	- Message stability is also application defined
	- one-to-many or few-to-many group communication model
	- Finite length streams

Supported Platforms:
	Windows 95/NT with Microsoft Visual C++ 4.x
	Sun Solaris 2.5 with SunPro C++ 4.x

Availability:
	Through the GlobalCast Communications web site at
	http://www.gcast.com through the BETA Program.

Contact for Additional Information
	Thomas Yeh (tyeh@gcast.com)
	Albert Chen (achen@gcast.com)
	Brian Whetten (whetten@gcast.com)
	Todd Montgomery (tmont@cs.wvu.edu)

About GlobalCast:
GlobalCast Communications is a venture capital backed company committed to
providing world leading reliable multicast protocols, tools, and
applications.  It is actively involved in helping drive forward the state of
the art in reliable multicast research and protocols, and is committed to
doing all it can to assist the research community and standards bodies in
developing open standards for reliable multicast.  To this end, it has a
history of open disclosure of its protocols and of allowing free use of its
software for academic or other non-commercial use.  


[1] S. Floyd, V. Jacobson, S. McCanne.  "A Reliable Multicast Framework for
Light-weight Sessions and Application Level Framing".  Proceedings of the
1995 ACM SIGCOMM Conference.  August, 1995, Cambridge, MA.  pp. 342-356.

[2] D. Clark and D. Tennenhouse.  "Architectural Considerations for a New
Generation of Protocols".  Proceedings of ACM SIGCOMM '90, Sept. 1990, pp.
201-208.

------------------------------------------------------------------------------
Todd L. Montgomery                      GlobalCast Communications, Inc.
Senior Scientist and Co-Founder         3777 Depot Road
E-mail: tmont@gcast.com	                Suite 415, 416A
        tmont@research.ivv.nasa.gov     Hayward, CA 94545
        tmont@io.com                    Telephone: (510) 266-0400
        tmont@cs.wvu.edu		FAX: (510) 266-0401
        tmont@access.mountain.net
	tmont@wvu.edu
http://www.cs.wvu.edu/~tmont/
------------------------------------------------------------------------------

From rem-conf-request@es.net Mon Jan 27 16:46:29 1997 
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          Mon, 27 Jan 1997 13:45:47 -0800
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To: "Todd L. Montgomery" <tmont@cs.wvu.edu>
cc: rem-conf@es.net, rmp-discuss@aurora.jhuapl.edu, rm@mash.cs.berkeley.edu, 
    floyd@ee.lbl.gov, mccanne@cs.berkeley.edu, gcast_all@gcast.com
Subject: Re: Announcing an SRM implementation in C++
In-reply-to: Your message of "Mon, 27 Jan 1997 15:54:01 EST." <Pine.SOL.3.91.970127154443.10389E@backus>
Mime-Version: 1.0
Content-Type: text/plain; charset=us-ascii
Date: Mon, 27 Jan 1997 22:39:38 +0100
From: Peter Parnes <peppar@cdt.luth.se>

tmont@cs.wvu.edu said:
> Supported Platforms:
> 	Sun Solaris 2.5 with SunPro C++ 4.x

How about other compilers such as GNU g++ ? 

Any plans on providing a Java-api? 

/P



From rem-conf-request@es.net Mon Jan 27 18:24:23 1997 
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          Mon, 27 Jan 1997 15:23:26 -0800
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To: "Todd L. Montgomery" <tmont@cs.wvu.edu>
cc: rem-conf@es.net, rmp-discuss@aurora.jhuapl.edu, rm@irtf.cs.berkeley.edu, 
    floyd@ee.lbl.gov, mccanne@cs.berkeley.edu, gcast_all@gcast.com
Subject: Re: Announcing an SRM implementation in C++
In-reply-to: Your message of "Mon, 27 Jan 1997 15:54:01 EST." <Pine.SOL.3.91.970127154443.10389E@backus>
Date: Mon, 27 Jan 1997 23:23:12 +0000
Message-ID: <2727.854407392@cs.ucl.ac.uk>
From: Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>



hmmmm.....

but srm never addressed 3 key quesions of (reli)able multicast

0 selective unreliability
1/ congestion
2/ per source ordering

i think this should continue on the rm list only. excet that the only
active discussion to date there has been rmfp........

how long is it since the ietf announcement of the rm list/group?

.....hmmm.... i spose everyone is too busy writing sigcomm papers:-)

jon

From rem-conf-request@es.net Mon Jan 27 19:38:19 1997 
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To: rem-conf@es.net
From: Jerrlyn Iwata <jerrlyn@postgres.Berkeley.EDU>
Subject: [reminder] Berkeley Multimedia & Graphics Seminar

                Berkeley Multimedia and Graphics Seminar 
        (Wednesday January 29, 1997 12:30-2:00 PDT 405 Soda Hall) 

      "[Rebroadcast] Music Delivery and Promotion on the Internet" 

                           Philip R. Wiser 
                            Liquid Audio 

The promise of commercial music distribution via the Internet is ready to be
fulfilled. This talk will describe the technical and legal complexities of
online music distribution. It will also present solutions to current online
distribution barriers. 

Music is a natural product for online delivery. Digital representations are
of desirable quality and only marginal value is added by physical packaging.
In fact, delivery of musical information via the Internet has been available for
many years. None of this content, however, has been distributed
commercially. It is only recently that large amounts of easily accessible,
high quality content has been exposed. The natural progression is to
commercially distribute this information electronically. 

Several technologies are required to make digital music distribution a
reality. Audio compression, embedded digital signalling, and cryptography
provide the framework for these systems. In addition, commerce and rights
reporting
components must be implemented. This discussion will address all of these
features. 



From rem-conf-request@es.net Mon Jan 27 20:08:22 1997 
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Date: Mon, 27 Jan 1997 17:07:26 -0800
To: "Wilson C. Chung" <wchung@ormail.intel.com>
From: Chunrong Chad Zhu <czhu@ibeam.jf.intel.com>
Subject: Re: RTP/H.263 Motion Vector Predictor
Cc: rem-conf@es.net

In case, someone else also missed my earlier replay to this question, here it
is:

Each 7 bits field represents a motion vector predictor in half pixel resolution
as 2's complement. For example, if HMV1=1000000 (bits 4 through 10
repectively), it represents motion vector predictor 0.5 as speficied by
H.263, since
half-pel motion vector is often used in inter-coded frame.

I will add this clearification into the next revision of the document.
Thanks very much for pointing this out.

--Chad


At 04:06 PM 1/27/97 -0500, you wrote:
>Dear BB:
>
>According to Turletti/Huitema's RTP payload format for H.261 video
>streams
>date July 7, 1996 document Section 5.1, HMVD and VMVD is a 5 bits number 
>encoded as a 2's complement number.  I think Chad Zhu from Intel will
>need
>to be more specific about what the 7 bits number represent.
>BTW, Chad's earlier RTP payload format for H.263 Video stream document
>dated
>Feb 26, 1996 and June 10, 1996 have 8 bits instead of 7 bits.
>I did not join the e-mail reflector until recently so I do not know is
>there
>any discussion regarding that.
>
>best regards, Wilson
>
>Baldine Paul wrote:
>> 
>> My understanding is that the RTP packetization will provide some
>> level of error resilience in case of packet losses. In order to recover
>> the motion vector of the current macroblock (in case of loss of the
previous packet)
>> the motion vector associated with the previous macroblock in the GOB is to be
>> encoded as part of the RTP/H.263 packet header. This, in addition to the
MVD that
>> is encoded in the H.263 bitstream, will allow to decoder to reconstruct
the current
>> macroblock, even if the previous is lost. As I was saying to Wilson, what
is not specified
>> in the document is whether the 7 bit represents a motion vector component in
>> sign/magnitude, 2's complement, 1's complement or any other format(like,
e.g. indexing
>> into a table of motion vector values).
>> What fo you think of this interpretation?
>> Baldine
>
>-- 
>Wilson C. Chung <wchung@pictel.com> (T)1-508-623-4667 (F)1-508-749-2804
>PictureTel Corp, 100 Minuteman Road, M/S 635, Andover MA  01810 USA
>
>


From rem-conf-request@es.net Mon Jan 27 20:21:25 1997 
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Date: Mon, 27 Jan 1997 17:20:47 -0800 (PST)
From: Stephen Casner <casner@precept.com>
To: rem-conf@es.net
Subject: Minutes for AVT WG at San Jose IETF
Message-ID: <Pine.SOL.3.95.970127171934.8858A-100000@little-bear.precept.com>
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Content-Type: TEXT/PLAIN; charset=US-ASCII

Minutes of the Audio/Video Transport Working Group

Reported by Steve Casner

1.  Introduction and status

The AVT working group met for two sessions at the San Jose '96 IETF.
AVT has produced the Real-time Transport Protocol, which was published
in January 1996 as a Proposed Standard RFC 1889 along with the
companion RTP profile for audio/video conferencing RFC 1890.  The
minimum interval before advancement to Draft Standard has now passed;
during that time, RTP use has grown with additional independent
implementations and a broader range of applications.  Accordingly, at
this meeting we discussed what changes to the spec may be required for
advancement to Draft Standard.  In particular, the biggest topic was
RTCP scalability for very large sessions, especially with asymmetric
or low data rate links.  Also of significance for low-speed links was
the second major topic of this meeting, an update on the IP/UDP/RTP
header compression protocol.

Since the last meeting, RFCs 2029, 2032, 2035 and 2038 defining the
RTP payload formats for CellB, H.261, JPEG and MPEG video encodings
were also published as Proposed Standards, in October 1996.  In the
second AVT session, a problem with RFC 2038 regarding packet loss
resilience for MPEG2 was presented, as were RTP Payload Formats for
Redundant Audio, H.728 audio and H.263 video.

A status report on development of an RTP MIB was presented; this is a
potential future work item for the group, along with completion of the
changes for transition of the RTP spec to Draft Standard and
standardization of additional profiles and payload formats.


2.  RTCP scaling for large "broadcast" sessions

Before this IETF meeting, there were several messages to the AVT
mailing list regarding use of RTCP in very large "broadcast"
scenarios, especially those where endpoints are connected by
low-speed modems or asymmetric cable systems with low-speed uplinks.
Bernard Aboba and Henning Schulzrinne gave presentations on the
problems that arise along with a variety of possible solutions.  

The interval between RTCP packets increases with the group size so
that RTCP consumes a constant fraction of the session bandwidth.
This keeps RTP well behaved as it scales to large groups.  However,
there are scenarios for which the current spec is inadequate:

  - Because the initial RTCP report is sent within a fixed interval
    after joining the session independent of the group size, there
    can be a large bandwidth spike if many participants join at the
    same time, e.g., for a scheduled event.  Convergence to the
    proper report interval may take too long, in particular for
    low-bandwidth sessions.

  - For sessions larger than a few thousand, the long RTCP interval
    may preclude collection of the desired information.

  - The memory required by the suggested RTCP interval calculation
    algorithm to track all the participant SSRC identifiers may be
    excessive for very large sessions.

  - For some types of sessions, privacy of reception may be more
    important than the feedback RTCP provides.

  - On low-speed modems, giving even 5% of the bandwidth to RTCP
    draws cries from the people who have to squeeze the data to fit.

  - Network operators are concerned about the multicast routing state
    required to track all the participants as multicast sources.

Several potential solutions for these problems were presented and
evolved in the discussion that followed:

  - The initial bandwidth spike can be avoided if the report interval
    is "reconsidered" before sending a report.  That is, sending is
    delayed until the updated group size estimate at the end of the
    interval is not much larger than it was at the start.  Group size
    estimation can also be accelerated by having each participant send
    the estimate it has counted, but use the maximum of all estimates
    received to calculate the interval.

  - Memory requirements may be reduced by storing only a sampling of
    the SSRC identifiers heard, based on a probability function.  The
    probability can be scaled as needed to fit the group size.

  - The report interval can be reduced by having only some of the
    participants report, perhaps controlled by a sampling function
    from the sender, if partial feedback is appropriate.

  - Receiver RTCP reports could be sent via unicast to the sender or
    some monitor, or sent on a separate multicast address, but the
    report interval must be controlled by some external means since
    the receiver can't calculate the interval itself.  There is a
    danger that if receivers allow some other entity to control their
    report rate and destination, this could be subverted to packet
    bomb some innocent party.

  - It is always safe (from a network load point of view) if
    receive-only participants do not send RTCP.  This avoids several
    of the problems listed, but precludes feedback.  The development
    of diagnostic tools such as mtrace makes RTCP less critical for
    network diagnosis than before.

  - Report bandwidth can be reduced by summarization of reports by
    selected receivers through self-organization using TTL scoping, or
    by RTP translators explicitly interposed in the distribution tree.
    This adds significant complication.

Many good points were brought up in the discussion of these possible
solutions:

  - Using a distributed consensus algorithm driven by multicast is
    robust because a single defector can't inflate the rate.

  - Encryption can be used for privacy, though this can add legal
    complications in some countries.

  - The current RTCP algorithm represents a design point that we
    believe works well for a wide range of applications and which has
    been tested in experiments on the MBone.  But there may be changes
    needed to extend operation to highly asymmetric networks or other
    scenarios outside that range.

Each of the potential solutions has some drawbacks, and there is not
one solution that is the clear answer in all cases.  Further
experimentation is needed to test the proposed ideas.  However, it was
agreed that the wording in the main RTP spec should be modified to
allow more flexibility in how RTCP may be used, and that the means by
which different modes should be defined and selected is through the
specification of (a small number of) additional RTP Profiles.  These
profiles might be specified in the description of a session, e.g., in
SDP using "RTP/AVB" ("B" for broadcast) rather than "RTP/AVP".

Proposals are solicited in the form of Internet Drafts defining
additional profiles to be the subject of experimentation and then
consideration for standardization.


3.  IP/UDP/RTP header compression

Steve Casner presented an update on the proposal developed with Van
Jacobson for hop-by-hop compression of IP/UDP/RTP headers to allow
use over low-speed lines.  This proposal was introduced at the
previous AVT meeting in Montreal.  The update incorporates changes
agreed in Montreal and subsequent improvements:

  - The context ID is always sent, and in the first byte to allow
    overlap with a link-level segmentation scheme when feasible.

  - The sequence number is increased to 4 bits and is changed to be
    context-specific rather than global.

  - A new delta encoding scheme was designed to fit the patterns
    encountered in typical RTP applications, replacing the temporary
    use of the scheme from RFC 1144 TCP/IP header compression.

  - RTP header compression is now integrated into the IPv6 header
    compression scheme (which also supports IPv4 and encapsulation in
    multiple headers).

  - It is now specified that RTCP packets will not be compressed,
    since the traffic fraction is small and the required increase in
    shared context would be impractical.

Manoj Leelanivas presented a report on his implementation of the
header compression algorithm and its performance.  He emphasized that
since the identification of which UDP streams carry RTP is only though
heuristics, it is essential to have a negative cache for UDP streams
that don't compress; otherwise, the context cache will thrash as each
new packet may fail to match any existing context.  The compression
scheme performed as expected: most audio packet headers compressed to
2 bytes without UDP checksum or 4 with; most video packet headers
compressed to 4 or 6 bytes.

During discussion of the proposal, it was agreed that the delta
encoding should be specified to be table driven, with the default
table being as presented but with holes in the encoding space used to
encode negative deltas that may occur with MPEG or out-of-order
packets.  Also, for use of header compression over higher-speed lines
where there may be a larger number of contexts, either an 8- or 16-bit
context ID should be allowed via negotiation.  The working group
agreed that this proposal should go to Last Call with these revisions
in place.

Steve Casner also made a short presentation in the PPPEXT working
group meeting to coordinate the allocation of PPP packet types that
will be required for this compression scheme.


4.  Transport aspects of RTSP Proposal

The Real-Time Streaming Protocol is under discussion primarily in the
MMUSIC working group, but also has some transport aspects that were
discussed briefly in AVT.  The RTSP proposal has included a
reduced-sized variant of RTP that was termed "compressed RTP".  Since
the AVT working group agreed in Montreal that the combined IP/UDP/RTP
compression scheme described in the previous section should be
standardized and that an end-to-end RTP-only compression scheme should
not, the RTSP authors will extract the definition of RTP variant into
a separate non-standards-track interim proposal named CUSH, for
Compressed UDP Stream Header.  In the main RTSP specification, the
means of specifying the underlying stream transport protocol will be
made more flexible to provide better separation between control and
transport.


5.  RTP payload formats
 
The second AVT session covered topics beyond the main RTP spec,
primarily additional payload formats plus some potential new
application areas for RTP.

5.1  Redundant audio payload format

Colin Perkins gave an update on draft-perkins-rtp-redundancy-01.txt
which defines a mechanism for carrying redundant audio formats in RTP
to compensate for packet loss.  The redundant copy is usually more
heavily compressed to reduce overhead.  This updated draft includes
modifications suggested at the Montreal meeting.  Two interoperating
implementations are in regular use.  The working group agreed that
this proposal should go to Last Call after a few minor edits.

5.2  Payload format for H.263 video

Chad Zhu presented revisions to the payload format for H.263 video in
draft-ietf-avt-rtp-payload-02.txt.  These revisions reflect changes in
the ITU H.263 spec in addition to suggested clarifications and other
minor changes from the AVT group.  One change, prompted by the
development of H.263+, was to add a version number in the payload
format.  This change generated some discussion because it introduces
another multiplexing point in the processing of each data packet.  The
group concluded that it would be preferable to design the payload
format to accommodate the expected changes as well as possible, and
then assign a new payload type if an incompatibility arises.  The
H.263 payload format spec should also be ready for Last Call after the
version number is removed.

5.3  Problems with MPEG2 payload format

The payload format spec for MPEG1 and MPEG2 was recently published as
Proposed Standard RFC 2038.  Reha Civanlar has identified a problem
with this spec in that the packet loss resilience information is
insufficient for MPEG2.  He presented two recommendations for
supplying the necessary information:

  - An optional second payload-specific header word would be added to
    carry the additional picture layer information required, and
    redundant sequence and GOP headers would be transmitted
    periodically as needed to achieve the desired loss probability.
    The overhead would be less than 1% for a 4 Mbps data rate assuming
    2 sequence/GOP header retransmissions per second.

  - To reduce the redundancy overhead, the "high priority" header
    information could be sent using a "reliable" protocol prior to the
    main transmission of the video data using RTP.  Since inclusion of
    the additional redundancy information in the payload header is
    optional, use of this method does not require changes to RTP.

The group accepted the proposed extensions, and the chair asked Reha
to write up the proposed change for inclusion into the MPEG payload
format specification before its transition to Draft Standard.

Reha also proposed an alternate format for bundled audio + video MPEG
payload format described in draft-civanlar-bmpeg-00.txt.  This format
would give reduced overhead and better error resilience compared to
the encapsulation defined in RFC 2038 for MPEG1 Systems or MPEG2
Transport streams.  However, such improvements were considered in an
early draft of RFC 2038 but were discarded because applications that
have data in the Systems or Transport streams formats and can't afford
the data handling required to take advantage of the benefits of
separate audio and video Elementary streams also could not afford to
use the more efficient bundled format.  Reha's proposal is put forth
for comparison testing and will be considered as an alternate format.

5.4  Payload format for G.728 audio

Ofer Shapiro has submitted text to be added to the RTP Audio/Video
Profile to describe the payload format for G.728 audio.  No separate
payload format spec is needed since only a few paragraphs are needed
to describe the format.  The four 10-bit vectors per audio frame are
simply packed into 5 bytes, MSB first.  Multiple frames may be packed
into each packet.  This format has been accepted by the ITU study
group covering the use of RTP in H.323.


6.  New applications of RTP

Several proposals for new applications of RTP have been recently
submitted to the working group.  Two of them, a payload format for
carrying HTTP over RTP in draft-aboba-rtp-http-01.txt, and a proposal
for adding Scalable Reliable Multicast mechanisms to RTP, described in
draft-parnes-rtp-ext-srm-01.txt, fall into the general area of
reliable multicast which the Transport Area Directors want to organize
as a separate IRTF Research Group and later IETF Working Groups.  The
group discussed this plan and the fit of new work into the AVT
charter, but discussion of these proposals was deferred until
organization of the reliable multicast research area is sorted out.

6.1  Using RTP with caching

Two presentations on new applications were made at this meeting.  The
first was by Roger Kermode on the idea of layering audio and video
streams in time as well as quality, then combining this layering with
"proactive" caching to reduce latency and bandwidth requirements for
on-demand playback.  Multiple RTP streams (layers) from a particular
video subject would be used for the different access phases implied by
(near) on-demand access as well as for multiple quality levels as
desired by different receivers.  Receivers would combine multiple
streams from caches and original sources with local storage to produce
the desired quality of playback at the desired time.

This work is at the research stage, but is likely to draw on RTP
(possibly with SRM extensions), RTCP, RTSP and SDP as building
blocks.

6.2  Aggregation Service within RTP

The second presentation was by Jonathan Rosenberg on multiplexing
several RTP audio sessions into one packet stream.  This could be
used, for example, to increase packet efficiency substantially between
Internet Telephony gateways.  Such gateways have recently been
developed and deployed to allow long-distance telephone callers to
dial a local gateway which then establishes an RTP stream to another
gateway near the desired destination and instructs the remote gateway
to dial the callee's telephone.  There may be many parallel streams
of small packets between these gateways; aggregating these streams
into larger packets conserves header overhead without increasing delay
as would be the case for large packets in a single stream.

Jonathan presented various methods for assigning individual streams to
logical channels in the aggregate and for carrying the information
that must remain specific to each stream.  The variations trade off
efficiency and the extent to which interpretation of fields in the RTP
header must change.  Full details of the options and efficiency
results are given in draft-rosenberg-itg-00.txt and .ps.  Further work
is needed before deciding whether some standardization should be
undertaken.


7.  Status report on RTP MIB

Stan Naudus gave a status report on the development of an RTP MIB.
While RTCP provides an efficient and scalable means to obtain feedback
>from end systems in a multicast session, it does not provide third
party management of unicast sessions such as in Internet Telephony.
There may also be a need for additional remote control of RTP mixers
and translators beyond what would be practical to implement with RTCP.

The challenge in designing a MIB for RTP is to make it scalable over
the wide variety of applications in which RTP might be used.  The
design currently includes 8 tables and 62 objects.  Greg Minshall
questioned whether 62 was too many objects for a practical MIB.  The
goal of the authors is to continue work on the MIB, including
implementation testing to assess practicality, and submit it for AVT
consideration at the next meeting in Memphis.


8.  Advancing RTP to Draft Standard

As noted in the introduction, it is time to advance the RTP spec, RFC
1889, and the RTP Profile, RFC 1890 to Draft Standard.  Steve Casner
presented a list of (potential) changes to be addressed for
advancement:

  - changes for RTCP scaling as described earlier
  - rule changes proposed for layered encodings as described in
    draft-speer-avt-layered-video-01.txt 
  - revision of the loop/collision algorithm for separate RTP and RTCP
    source port numbers
  - some clarifications of the wording and small changes such as
    allowing separate unicast ports that have already been made in the
    spec source files
  - allowing separate multicast addresses for RTP/RTCP

These items are all straightforward or were previously discussed
except for the last.  Should the specification of an RTP session be
generalized to allow different addresses to be used for RTP and RTCP?
This would allow RTCP delivery to be constrained for improved
scalability in some scenarios.  After some discussion, the consensus
of the group was that we should take the conservative approach and not
make this change at this time.

In addition to any required revisions of the RTP specification,
advancing to Draft Standard requires evidence that interoperability
requirements have been met.  This should be easy to provide since
there are several genetically distinct implementations of RTP, both
research and commercial, in use.  The exact form required for the
evidence has not been determined by the Area Directors yet, but it is
likely that an applicability document will be needed to list the areas
in which RTP has been successfully used and the areas to which we
believe it can be extended.  We may also be required to document the
scalability of RTP in a manner similar to that proposed by the Area
Directors as a requirement for reliable multicast protocol proposals.


9.  Advancing RTP Profile to Draft Standard

Since the RTP Profile is a simpler document than the main RTP spec,
there are fewer issues for its advancement.  One question of
significance is whether the Profile should restrict the format of the
RTCP SDES CNAME item beyond the suggestions of the RTP spec so that
separate implementations of audio and video tools will be more likely
to use a common format.  This commonality is important to allow audio
and video streams to be associated for synchronization and other
presentation management.  The group agreed that the profile should
recommend use of only the numeric address form of the CNAME.

In addition to this change, there are several of the simple audio
payload formats included in the Profile that need some additional
details specified, such as bit and byte order for packing.  Some
additional formats, such as G.723 and G.728, will also be added.


10.  Future work

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


From rem-conf-request@es.net Mon Jan 27 22:39:48 1997 
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Date: Mon, 27 Jan 1997 19:43:00 -0800
To: Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>, 
    "Todd L. Montgomery" <tmont@cs.wvu.edu>
From: Brian Whetten <whetten@gcast.com>
Subject: Re: Announcing an SRM implementation in C++
Cc: rem-conf@es.net, rmp-discuss@aurora.jhuapl.edu, rm@mash.CS.Berkeley.EDU, 
    floyd@ee.lbl.gov, mccanne@cs.berkeley.edu, gcast_all@gcast.com

These are things we are working to help fix, through whatever contributions
we can.

Our SRM implementation (which we would again like to reiterate does not
necessarily have fidelity to the original intentions of the SRM authors) has
provided a solution for per-source ordering.  The framework we have provided
can be adapted for selective unreliability, and an instance of this, which
we call source/time ordering (per-source ordering, with a time bound) will
be released shortly.  Congestion control for _all_ reliable multicast
protocols is still a very active research area, and while we have proposed
two solutions (one in RMP, and one in SRM), we are working hard in this area
as well, and will be releasing new WAN congestion control algorithms soon as
well.

You are right...let's switch this discussion over to the rm mailing list,
and see if we can get some traffic going there. 

Brian

At 11:23 PM 1/27/97 +0000, Jon Crowcroft wrote:
>
>
>hmmmm.....
>
>but srm never addressed 3 key quesions of (reli)able multicast
>
>0 selective unreliability
>1/ congestion
>2/ per source ordering
>
>i think this should continue on the rm list only. excet that the only
>active discussion to date there has been rmfp........
>
>how long is it since the ietf announcement of the rm list/group?
>
>.....hmmm.... i spose everyone is too busy writing sigcomm papers:-)
>
>jon
>
>
Brian Whetten					     3777 Depot, Suite 415
President, GlobalCast Communications	 	        Hayward, CA  94545
whetten@gcast.com			      (510) 266-0400  FAX 266-0401


From rem-conf-request@es.net Tue Jan 28 00:39:08 1997 
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Date: Mon, 27 Jan 1997 21:38:56 -0800
From: Anders Klemets <klemets@vxtreme.com>
Organization: VXtreme, Inc.
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------------54FC1E5610BA2
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> The group agreed that the profile should
> recommend use of only the numeric address form of the CNAME.

I don't recall that we actually agreed on that.  But in any case, only
having the
numeric address form in the CNAME, and nothing else, is not such a good
idea.
If you have two applications running on the same machine that try to
talk to each
other, the RTP loop detection function may think that an RTP loop has
occurred.

Some of the RTP implementations that are available today might avoid
false
RTP loop detection by checking the source IP address and port number.
But that is not a very nice hack, since RTP is supposed to be
independent of the
underlying protocol.

Anders

P.S.  Sorry for sending HTML'fied stuff to the list, that's a feature of
my
mail program, and it cannot be turned off.


------------54FC1E5610BA2
Content-Transfer-Encoding: 7bit
Content-Type: text/html; charset=us-ascii

<HTML><BODY>

<DT>&gt; The group agreed that the profile should<BR>
&gt; recommend use of only the numeric address form of the CNAME.<BR>
<BR></DT>

<DT>I don't recall that we actually agreed on that.&nbsp; But in any case,
only having the</DT>

<DT>numeric address form in the CNAME, and nothing else, is not such a
good idea.</DT>

<DT>If you have two applications running on the same machine that try to
talk to each</DT>

<DT>other, the RTP loop detection function may think that an RTP loop has
occurred.</DT>

<DT>&nbsp;</DT>

<DT>Some of the RTP implementations that are available today might avoid
false</DT>

<DT>RTP loop detection by checking the source IP address and port number.</DT>

<DT>But that is not a very nice hack, since RTP is supposed to be independent
of the</DT>

<DT>underlying protocol.</DT>

<DT>&nbsp;</DT>

<DT>Anders</DT>

<DT>&nbsp;</DT>

<DT>P.S.&nbsp; Sorry for sending HTML'fied stuff to the list, that's a
feature of my&nbsp;</DT>

<DT>mail program, and it cannot be turned off.</DT>

<DT>&nbsp;</DT>

</BODY>
</HTML>
------------54FC1E5610BA2--



From rem-conf-request@es.net Tue Jan 28 02:02:51 1997 
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unsubscribe


From rem-conf-request@es.net Tue Jan 28 07:58:31 1997 
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From: k claffy <kc@nlanr.net>
Message-Id: <199701281257.EAA21895@nlanr.net>
Subject: announce broadcast of apricot conference 28-29th jan 97
To: rem-conf@es.net
Date: Tue, 28 Jan 1997 04:57:47 -0800 (PST)
Cc: evi@xor.com (evi n)
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what:  APRICOT [asian pacific rim
	        Internet conference on
		operational techniques]

where: from hong kong, excelsior hotel
who: apricot and APNIC and commercial sponsors
when:  28-29th jan 97  [29-30th in .hk]

program details: http://202.77.253.68/program.html
source host: 166.222.30.4
check sdr for advertisement

note there are three tracks and only one mbone channel.  
so from the program at the URL above,
we will try to broadcast track 2 in real time
and then try to rebroadcast tapes of 
track 3 and then track 1 during the .hk off hours

  track 	hk time		US PST time   	GMT
   2	  	0830-1700       1730-0200	0130-1000
   3	  	1700-2300       0200-0800	1000-1600
   1	  	2300-0500       0800-1400	1600-2400

[not that we have any legitimate reason to belive
that any of this will work anyway, 
so advise to add grains of salt to the hour glass]

k
send complaints to  kc@nlanr.net
		 or evi@cs.colorado.edu


From rem-conf-request@es.net Tue Jan 28 08:04:42 1997 
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          Tue, 28 Jan 1997 14:04:07 +0100 (MET)
Date: Tue, 28 Jan 1997 14:04:07 +0100 (MET)
From: Biersack Ernst <Ernst.Biersack@eurecom.fr>
Message-Id: <199701281304.OAA12837@alpes.eurecom.fr>
To: rem-conf@es.net
Subject: SIGCOMM 97 submission deadline is coming up
Cc: erbi@alpes.eurecom.fr


Dear colleagues,
Its time to hurry up!
The deadline for submission to SIGCOMM 97 is Jan 31st, i.e. 
the  end of this week.
Yours,
\Ernst Biersack
(Sigcomm Publicity Chair)


The next ACM SIGCOMM will be in September 1997 
at the Palais des Festivals in Cannes, France.

Tutorials, September 14-15, 1997
Conference, September 16-18, 1997


The deadlines are:
	   Paper submissions: 31 January 1997
	   Tutorial proposals: 31 January 1997 
	   Notification of acceptance: 28 April 1997
	   Camera-ready paper due: 30 May 1997 


For more details, please  see:

http://www.inria.fr/rodeo/sigcomm97/



From rem-conf-request@es.net Tue Jan 28 15:49:34 1997 
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CC: rem-conf@es.net, videophone@es.net
Subject: book availability announcement: Duran/Sauer, Mainstream 
         Videoconferencing
Content-Type: text/plain; charset=us-ascii
Content-Transfer-Encoding: 7bit

Mainstream Videoconferencing: A Developer's Guide to Distance
Multimedia, by Joe Duran and myself, is now available.

For more information, see http://www.aw.com/cp/duran-sauer.html. That
location includes table of contents, preface, first chapter and an
appendix of web resources.

Charlie

From rem-conf-request@es.net Tue Jan 28 20:45:22 1997 
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Date: Tue, 28 Jan 1997 17:42:36 -0800 (PST)
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To: rem-conf@es.net
From: mallard@mail.arc.nasa.gov (Mark Allard)
Subject: NASA-Remote University Course

The second in a series of Remote University Courses in Telerobotics will be
broadcast on the MBone Wednesday, February 29, 1997 from 2PM-7PM.

Contact Mark Allard <mallard@mail.arc.nasa.gov> for more information or go
to <http://quest.arc.nasa.gov/courses/telerobotics/590/> for additional
details on the course.

This course material is being provided as a service to the MBone community
by NASA-Ames Research Center.



From rem-conf-request@es.net Wed Jan 29 03:54:31 1997 
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Date: Wed, 29 Jan 1997 00:53:11 -0800 (PST)
From: Stephen Casner <casner@precept.com>
To: Anders Klemets <klemets@vxtreme.com>, 
    Henning Schulzrinne <schulzrinne@cs.columbia.edu>
cc: rem-conf@es.net
Subject: Re: CNAME
In-Reply-To: <32EDF7B0.6445@cs.columbia.edu>
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Anders and Henning,

Anders> > The group agreed that the profile should
Anders> > recommend use of only the numeric address form of the CNAME.
Anders> 
Anders> I don't recall that we actually agreed on that.

We can certainly discuss this topic further here on this list.  My
recollection of the discussion was that Colin Perkins agreed that RAT
should change to use the numeric form, and that there there were a
couple of other speakers in favor but no strong objections voiced.
This topic was near the end of the meeting, and there was not a lot of
discussion on any side.

Henning> I figured there must have been a straw poll of some sort with
Henning> unanimous agreement.

Well, consensus process is a rough one.  On this topic, I don't recall
asking for raising of hands on both sides, but I think I asked if
there was general agreement and got some hums but no shouts.  (A WG
chair also often looks out on a relatively large number of blank
stares as well, which is one of the harder aspects of the job.)
As I say, I would be happy to hear more input.

Anders> But in any case, only having the numeric address form in the
Anders> CNAME, and nothing else, is not such a good idea.  If you have
Anders> two applications running on the same machine that try to talk
Anders> to each other, the RTP loop detection function may think that
Anders> an RTP loop has occurred.

Sorry, I did not mean to imply that the "user@address" form would be
replaced with the "address" only form, only that the two address forms
would be recommended over the FQDN forms.

Anders> Some of the RTP implementations that are available today might
Anders> avoid false RTP loop detection by checking the source IP
Anders> address and port number.  But that is not a very nice hack,
Anders> since RTP is supposed to be independent of the underlying
Anders> protocol.

I believe one must use the IP address and port in the collision/loop
detection algorithm, and that the CNAME turns out to provide only
marginal information.  Since I wrote the collision/loop detection
algorithm as it currently appears in the spec, that's what it says.  I
Henning and I have had some debate about this, but I firmly believe
it.  I can go back and dig up that debate if needed.  I invite folks
to take another look at the algorithm, but note that the following
section:

       IF the source transport address is found in the list of
         conflicting addresses:
       THEN IF the source identifier is not from an RTCP SDES chunk
               containing a CNAME item OR if that CNAME is the
               participant's own:
            THEN (optionally) count occurrence of own traffic looped.
                 mark current time in conflicting address list entry.
                 ABORT processing of data packet or control element.

the "ABORT" should be move out one indentation level, i.e., it is
always done in the first THEN clause independent of the second IF.

There are two points I'd like to make on the use of source IP
addresses and ports:

1.  Doing this check does not strictly make RTP dependent upon the
    lower layer protocol.  The IP addresses and ports are opaque to
    the algorithm; it only checks for equality.  Well, we do depend
    upon there being a source address.

2.  The original source address is lost for packets that go through a
    translator, but that translator is also responsible for performing
    the collision/loop detection with the original source info.


Henning> I was also surprised by "agreed", given all the unresolved
Henning> issues (Net10 replication, IPv4/IPv6, debuggability, which
Henning> interface to pick and how to get all interfaces, etc.)

Net10 is indeed a cause for concern.  Unfortunately, the places where
Net10 is used are often the ones that don't provide sufficient
information to obtain an FQDN either, since those hosts don't need to
be known outside the firewall.  To the extent that UUID's depend upon
addresses, they don't help, either (and temporal uniqueness is a
problem for matching tools that don't start at the same exact
instant).  I think IPv4/IPv6 and interface choice are much less
significant issues (less likely to cause trouble).  I'm not sure what
the debuggability issue is.

I've been exchanging messages with Dermot Tynan who works on the
AltaVista Firewall.  He suggests that firewalls may be required to
rewrite CNAMEs to hide internal information.  If this will be the
case, and if we can devise a common translation rule (that keeps
uniqueness of the machine identities), that may solve the Net10
problem.

Henning> It might also be worth looking into what if any effect the 8/8 IPv6
Henning> addressing proposal might have, where the address prefix might change
Henning> during a session.

That may not be a problem if we only need to coordinate tools that
start at the same time.  The CNAME does not need to change when the
prefix changes.  I guess I'll burn this bridge when I come to it.

							-- Steve


From rem-conf-request@es.net Wed Jan 29 04:14:08 1997 
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To: Charlie Sauer <sauer@technologists.com>
cc: rem-conf@es.net, videophone@es.net
Subject: Re: book availability announcement: Duran/Sauer, Mainstream 
         Videoconferencing
In-reply-to: Your message of "Tue, 28 Jan 1997 14:47:47 CST." <32EE65F2.7934@technologists.com>
Date: Wed, 29 Jan 1997 09:13:05 +0000
Message-ID: <1046.854529185@cs.ucl.ac.uk>
From: Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>



 >Mainstream Videoconferencing: A Developer's Guide to Distance
 >Multimedia, by Joe Duran and myself, is now available.
 
 Charlie

looks good

meanwhile, our internet multimedia  book is at
http://www.cs.ucl.ac.uk/staff/jon/mmbook/book/book.html

feel free to browse/download/comment....
its mostly mbone/rsvp/rtp technology type book......

to be published soonsh....

 jon


From rem-conf-request@es.net Wed Jan 29 06:47:02 1997 
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To: Stephen Casner <casner@precept.com>
cc: Anders Klemets <klemets@vxtreme.com>, 
    Henning Schulzrinne <schulzrinne@cs.columbia.edu>, rem-conf@es.net
Subject: Re: CNAME
In-reply-to: Your message of "Wed, 29 Jan 1997 00:53:11 PST." <Pine.SOL.3.95.970129000331.20716A-100000@little-bear.precept.com>
Date: Wed, 29 Jan 1997 11:44:21 +0000
Message-ID: <970.854538261@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>

--> Stephen Casner writes:
>Anders> > The group agreed that the profile should
>Anders> > recommend use of only the numeric address form of the CNAME.
>Anders> 
>Anders> I don't recall that we actually agreed on that.
>
>We can certainly discuss this topic further here on this list.  My
>recollection of the discussion was that Colin Perkins agreed that RAT
>should change to use the numeric form, and that there there were a
>couple of other speakers in favor but no strong objections voiced.
>This topic was near the end of the meeting, and there was not a lot of
>discussion on any side.

I made the change to RAT, starting with RAT-3.0.8 (we hope to have a
general release of RAT-3.0.x relatively soon).

It seems that allowing two different, but equivalent, forms for a CNAME is
unnecessary, and can only serve to cause confusion. Given that, in some
cases, it's not possible to determine the FQDN it seems sensible to mandate
the user@dotted-quad form of CNAME in the a/v profile.

Henning> I was also surprised by "agreed", given all the unresolved
Henning> issues (Net10 replication, IPv4/IPv6, debuggability, which
Henning> interface to pick and how to get all interfaces, etc.)

These other issues seem much more serious: my particular concern being the
choice of interface for multi-homed hosts (since I'm currently working on
an audio transcoder/mixer).

Colin

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From: richter@ro2.informatik.uni-hamburg.de (Jan-Peter Richter)
Message-Id: <9701291508.AA09758@ro2.informatik.uni-hamburg.de>
Subject: ext. deadline Feb 10. ANMT/QoS Modeling Conference
To: rem-conf@es.net
Date: Wed, 29 Jan 1997 16:07:25 +0100 (MET)
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Content-Type: text



Please notice that the deadline for the ANMT/ QoS Modelling Conference
to be held in Singapore, September 1-4, has been extended to >>> Feb. 10 <<<

	Jan-Peter Richter

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



                            Call for Papers

                    
                     4th International Conference on

	       Analytical and Numerical Modelling Techniques 

		          with the application to 

                    QUALITY OF SERVICE (QoS) MODELLING


                      Singapore, September 1-4, 1997


The submission of papers is solicited for the "4th Analytical and Numerical
Modelling Techniques Conference"  on QUALITY OF SERVICE (QoS) MODELLING
which is organized by SCS as part of the 1st  World Congress on Systems 
Simulation.

Technical Background:
--------------------
Recently, the concept of Quality of Service (QoS) has attracted a lot of 
attention in research and development of high speed computer and 
communication networks with multimedia applications. Real-time and other 
qualities such as guaranteed performance and dependability  are vital 
properties of offered services to be provided by future communication networks.
Contractual relationships between service user and service provider 
regarding prospective services and their required qualities have to
be realized.  Guarantees should be  given even in the presence of stochastic 
uncertainties. Network components may fail, transmission errors cannot 
be totally avoided, or temporary overload conditions are likely to occur 
due to incorporated multiplexing techniques in the usage of system resources.

Quantitative modeling represents a promising approach to appropriately
support system design and dynamic management. Accurate projections of 
computer and communication systems' behavior are needed. The first 
challenge for QoS modeling comes from the fact that timeliness, 
performance, and dependability properties are equally vital issues for 
the applications. The second challenge is due to the peculiar novelty 
that application and customer specific metrics rather than system related 
ones are of major interest. As a third challenge it becomes increasingly
important that QoS support is provided in heterogeneous environments.

Original papers which have not been published and which deal with 
QoS related quantitative modeling topics are solicited for submission. 
The conference focus is  on modeling techniques and tools as well as on
evaluation studies relevant to the theme. QoS modeling combined with
practical implementation experience is of predominant interest.
The following is an incomplete list of topics relevant to the conference:

- QoS modeling techniques and tools for evaluation of
  multimedia services in high speed networks
- QoS management
- QoS specification
- QoS mapping
- QoS negotiation and admission control
- QoS modeling and evaluation case studies
- QoS modeling and practical implementation experience
- QoS support in heterogeneous environments
- User oriented QoS modeling
- Neurocomputing and cognitive modeling for QoS support 
- Real-time services
- Adaptive and reconfigurable systems
- Responsive systems modeling
- Dependability evaluation
- Performability modeling
- Stochastic Petri net models
- Queueing systems
- Markov models
- Simulation studies
- Measurement techniques and studies


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

 Conference Chair:
 ----------------
 
 Gunter Bolch
 Department of Computer Science - OS, University Erlangen-Nuremberg
 Martensstrasse 1, D-91058 Erlangen, Germany
 bolch@informatik.uni-erlangen.de
 Tel.: +49 9131 85-7903, Fax: +49 9131 39388
 
 Program  Chair:
 --------------
 
 Hermann de Meer
 Department of Computer Science  -  TKRN, University of Hamburg 
 Vogt-Koelln-Strasse 30, D-22527, Hamburg, Germany
 demeer@informatik.uni-hamburg.de
 Tel: +49 40 5494-2347, Fax: +49 40 5494-2328
 
 
 Program Committee:
 -----------------
 
 Andres Albanese, International Computer Science Institute at Berkeley, USA
 Gordon Blair, Lancaster Univ., UK
 Andrew Campbell, Columbia Univ., USA
 Mario Dal Cin, Erlangen Univ., Germany
 Petre Dini, Univ. de Montreal, Canada
 Winfried Dulz, Erlangen Univ., Germany
 Pankaj  Garg, Hewlett Packard Lab. at Palo Alto, USA
 Stefan Greiner, Erlangen Univ., Germany
 Roch Guerin, IBM T.J. Watson Research Center, USA 
 Abdelhakim Hafid, Univ. de Montreal, Canada
 Guenter Haring, Univ. of Vienna, Austria
 Boudewijn Haverkort, Aachen Univ., Germany
 Ulrich Herzog, Erlangen Univ., Germany
 Gardeep Hura, Nanyang Technological University, Singapore
 Klaus Irmscher, Bergakademie Freiberg, Germany
 Werner Jarschel, Siemens AG Erlangen, Germany
 Yoshiaki Kakuda, Osaka Univ., Japan
 David Kirsh, Univ. of California at San Diego, USA
 Edward Knightly, Rice Univ., USA
 Udo Krieger, Deutsche Telekom AG Darmstadt, Germany
 Jorg Liebeherr, University of Virginia, USA
 Dimitris Logothetis, Lucent Technologies, USA
 Miroslaw Malek,  Humboldt Univ. Berlin, Germany
 Jan de Meer, GMD-Fokus Berlin, Germany
 Bruno Mueller-Clostermann, Essen Univ., Germany
 Jogesh Muppala, Hong Kong University of Science and Technology, Hong Kong
 Klara Nahrstedt, University of Illinois at Urbana-Champaign, USA 
 Victor Nicola, Univ. of Twente, The Netherlands
 Giovanni Pacifici, IBM T.J. Watson Research Center, USA
 Antonio Puliafito, Universita di Catania, Italy
 Jan-Peter Richter, Univ. of Hamburg, Germany
 Miklos Telek, Budapest Univ., Hungary
 Don Towsley, Univ. of Massachusetts at Amherst, USA
 Satish Tripathi, Univ. of Maryland, USA
 Kishor Trivedi, Duke University, USA
 Andreas Vogel, Univ. of Queensland, Australia
 Sandy Wang, Stratacom at Palo Alto, USA
 Weiguo Wang, ISS, National University of Singapore, Singapore
 Bernd Wolfinger, Univ. of Hamburg, Germany
 Martina Zitterbart, Braunschweig Univ., Germany
 

 Deadlines:
 ---------

 Paper Submission (extended deadline): Feb. 10, 1997

 Notification of Acceptance: April 1, 1997

 Camera Ready Copy: May 15, 1997


 
 Submission Policy:
 -----------------
 
 The submission of SIX COPIES of double-spaced full manuscripts is 
 solicited.  The length of double-spaced papers should not exceed 15 PAGES, 
 including figures and references. The camera ready copy will be 
 required to be in double column format, not exceeding five pages of length. 
 
 The papers should be sent to:

 Office of the
 1st WORLD CONGRESS ON SYSTEMS SIMULATION
 ANMT: QoS Modeling

 Attention: Mr. Jonnovan Hong
 c/o IEEE Singapore Section
 59D Science Park Drive, The Fleming
 Singapore 118243
 Republic of Singapore

 Tel: +65 773 1141 Fax: +65 773 1142
 ieeesgp@pacific.net.sg
 

 ATTENTION!

 Please help to speed up the reviewing process and submit also
 a POSTSCRIPT version of your paper by EMAIL to the program chair:
     
 demeer@informatik.uni-hamburg.de



 Publication Policy:
 ------------------
 
 All submitted papers will be reviewed by at least four members of the
 international program committee.  Accepted papers will be published in 
 the IEEE Computer Society and SCS-Europe Publishing House proceedings.
  
 

 Exhibitions and Tutorials:
 -------------------------
 
 Exhibitions, tutorials, and vendor sessions are also solicited for 
 presentation. If interested, please contact the appropriate 
 conference or program chair person.
 


 Organization:
 ------------

 Organized and sponsored by:        SCS The Society for Computer Simulation
					  International
				    National University of Singapore 
				    Nanyang Technical University, Singapore 
				    University Erlangen-Nuremberg, Germany 
				    University of Ottawa, Canada 


					       
 Participating Societies:           ASIM, CASS, CSSS, FRANKOSIM, HSS,
				    IEEE Singapore, ISCS, JSST, KSS, 
				    SCS Europe, SCS International,
				    SiE, SIMS, UKSS, NUS, NTU 


From rem-conf-request@es.net Wed Jan 29 11:15:54 1997 
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Date: Wed, 29 Jan 1997 11:14:55 -0500 (EST)
From: Wendy Fleming <fleming@lis.pitt.edu>
Subject: Re: book availability announcement: Duran/Sauer, Mainstream 
         Videoconferencing
To: Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>
Cc: Charlie Sauer <sauer@technologists.com>, rem-conf@es.net, videophone@es.net
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unsubscribe


From rem-conf-request@es.net Wed Jan 29 17:31:26 1997 
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Date: Wed, 29 Jan 1997 14:08:04 -0800 ()
From: Stephen Casner <casner@precept.com>
To: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
cc: Anders Klemets <klemets@vxtreme.com>, 
    Henning Schulzrinne <schulzrinne@cs.columbia.edu>, rem-conf@es.net
Subject: Re: CNAME
In-Reply-To: <970.854538261@cs.ucl.ac.uk>
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> These other issues seem much more serious: my particular concern being the
> choice of interface for multi-homed hosts (since I'm currently working on
> an audio transcoder/mixer).

If there is a conventional way to get one of the addresses, we don't
care which one it is for the CNAME.  We just want all the tools to
get the same one.
							-- Steve


From rem-conf-request@es.net Wed Jan 29 18:11:29 1997 
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Date: Wed, 29 Jan 1997 15:10:28 -0800
To: rem-conf@es.net
From: Steve Deering <deering@cisco.com>
Subject: MBone teleconf tools for Macs?

Has anyone ported any of the MBone teleconferencing tools to the
Macintosh yet?

Apple has their MBone TV app which is a partial implementation of some of
the capabilities in the Unix MBone suite, all nicely integrated with
Apple's QuickTime architecture. I'm sure it will be fabulous when (if) they
finish it, but for now it is only good for passive reception of MBone audio
and video using some subset of the common encodings -- it doesn't support
transmission of audio, so it's not yet useable for teleconferencing with
vat users.

Some folks at UCLA began an implementation (USeeLA), but according to
their web page, they decided to stop development since Apple was already
doing it.  Too bad.

CUSeeMe, of course, supports teleconferencing on Macs, but as far as I
know it does not interoperate with the MBone RTP-based tools.

I am particularly interested in Mac versions of sdr and vat, or
interoperable equivalents.  wb would also be great, as an adjunct to
the first two.  vic is much lower priority for me, though it seems that
implementors always want to do the flashy video stuff first.

Freeware or payware, either will do.  I'm willing to be a tester for
pre-release implementations.  Just tell me I don't have to buy a PeeCee!

Steve



From rem-conf-request@es.net Wed Jan 29 20:51:30 1997 
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Date: Wed, 29 Jan 97 17:49 PST
To: Steve Deering <deering@CISCO.COM>
From: Denis DeLaRoca <CSP1DWD@MVS.OAC.UCLA.EDU>
Subject: Re: MBone teleconf tools for Macs?
CC: rem-conf@ES.NET

On Wed, 29 Jan 1997 15:10:28 -0800,
   Steve Deering <deering@CISCO.COM> said:
> Has anyone ported any of the MBone teleconferencing tools to the
> Macintosh yet?

I am afraid the answer is no! Apple's Mbone TV stuff is way too limited
and in practice not very useful... UCLA's UseeLA is long dead and didn't
get too far. CUSeeMe's latest version, at least their Windows version,
is multicast aware, but I don't know to whet extent it interoperates
with the MBONE tools -- the White Pine people came to the IP Multicast
summit. The UCL people also looked at doing MBONE tools for the MAC
but gave up as well... Apple sort of said they would solve the problem
but so far we have no decent tools yet. So maybe the only reasonable
option is to get yourself an Intel notebook and run Windows or some-
thing like FreeBSD or Linux.

There's now TCL/TK for the Macintosh, and Apple's Open Transport does
support multicasting. So I think the framework to port vat, vic and sdr
to the Macintosh is there, pretty much like it was done for Windows '95
-- the audio and video programming is all done thru Quicktime.

Let's hope someone else can give us better news. :-(

-- Denis


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I noticed that an announcment i made a while ago, has disappeared from the
sdr calendar (when I open my suite). Is that possible? 
Do they have a TTL too....?


From rem-conf-request@es.net Wed Jan 29 22:16:59 1997 
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Date: Wed, 29 Jan 1997 22:22:00 -0500
To: Denis DeLaRoca <CSP1DWD@MVS.OAC.UCLA.EDU>
From: daveb@wpine.com (David Bundy)
Subject: Re: MBone teleconf tools for Macs?
Cc: Steve Deering <deering@CISCO.COM>, rem-conf@ES.NET

Hello folks,
        We are working on such interopability in our upcoming 3.0, 3.1 
releases for Mac and Windows of CU-SeeMe client and reflector
(eg. MBONE TOOLS, RTP, H.323).
          I do
not have a date on this I can legally give you but we have stated publicly
we will have it this year.  Our current 2.1.1 Windows release follows the
SDR and Mutlicast standards but didn't get the RTP and Mbone Codecs.

Dave B.

At 05:49 PM 1/29/97 PST, Denis DeLaRoca wrote:
>On Wed, 29 Jan 1997 15:10:28 -0800,
>   Steve Deering <deering@CISCO.COM> said:
>> Has anyone ported any of the MBone teleconferencing tools to the
>> Macintosh yet?
>
>I am afraid the answer is no! Apple's Mbone TV stuff is way too limited
>and in practice not very useful... UCLA's UseeLA is long dead and didn't
>get too far. CUSeeMe's latest version, at least their Windows version,
>is multicast aware, but I don't know to whet extent it interoperates
>with the MBONE tools -- the White Pine people came to the IP Multicast
>summit. The UCL people also looked at doing MBONE tools for the MAC
>but gave up as well... Apple sort of said they would solve the problem
>but so far we have no decent tools yet. So maybe the only reasonable
>option is to get yourself an Intel notebook and run Windows or some-
>thing like FreeBSD or Linux.
>
>There's now TCL/TK for the Macintosh, and Apple's Open Transport does
>support multicasting. So I think the framework to port vat, vic and sdr
>to the Macintosh is there, pretty much like it was done for Windows '95
>-- the audio and video programming is all done thru Quicktime.
>
>Let's hope someone else can give us better news. :-(
>
>-- Denis
>
>
David O. Bundy
VP Engineering
White Pine Software

LOOK WE MOVED!

542 Amherst St
Nashua NH, 03063
===============
dbundy@wpine.com
http://www.wpine.com/~dob
Phone: 603-886-9050
Fax:   603-886-9051


From rem-conf-request@es.net Thu Jan 30 04:52:45 1997 
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unsubscribe

pocsai@rpk.uni-karlsruhe.de 


From rem-conf-request@es.net Thu Jan 30 05:25:18 1997 
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Date: Thu, 30 Jan 1997 12:25:40 +0200
To: Steve Deering <deering@cisco.com>, rem-conf@es.net
From: Jan Lucenius <jan.lucenius@vtt.fi>
Subject: Re: MBone teleconf tools for Macs?
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At 15:10 29.1.1997 -0800, Steve Deering wrote:
>
>CUSeeMe, of course, supports teleconferencing on Macs, but as far as I
>know it does not interoperate with the MBone RTP-based tools.
>
According to answers that I got last spring about CuSeeMe, it is able to
communicate with NV through a reflector. (Do not remeber whether it
supports Mbone and not either about vat/vic/sdr compatibility.) CuSeeMe
does not itself support RTP itself. I also remind that I have seen some
notice that some of the Mac products should cooperate with Microsoft's Net
Meeting in the future - not sure if it was CuSeeMe ? It also seems like
NetPhone would be NV compatible, but that's only audio.

I think one of the best summaries , where to start finding out about
existing products is http://www3.ncsu.edu/dox/video/products.html .

 > Just tell me I don't have to buy a PeeCee!

My feeling is also that Macintosh would be at least as good if not better
platform for audio and video I also think that you should use it, if its
possible. Write what you find out.

	Jan

From rem-conf-request@es.net Thu Jan 30 08:39:52 1997 
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Date: Thu, 30 Jan 1997 08:38:04 -0500 (EST)
From: Yves Lepage <yves@CC.McGill.CA>
Message-Id: <199701301338.IAA03310@maelstrom.CC.McGill.CA>
To: deering@cisco.com, rem-conf@es.net
Subject: Re: MBone teleconf tools for Macs?

Hi Steve,

With the merge of Apple and NeXT, I'm not too sure that either Apple
or third parties `will be willing to be put too much energy in an OS
that's destined to die (die of old age however). 

I was able to get some inter-operability between Mbone and Cu-SeeMe
by hacking around difficulties. The CuSeeMe reflector is supposed
the perfectly sned and receive mcast traffic but:

1- for me, it didn't
2- It can only support the CuSeeMe video format and DVI audio, which 
   makes it useless

I know I'm not being very useful here but I felt I had to say
if I was you, I'd not hold my breath. Do an altavista search
of the words MBone and CuSeeMe (ok, they're not really words)
and you'll several web pages of people who tried to make these
two worlds inter-operate. Some succeded.

Regards,
Yves Lepage

From rem-conf-request@es.net Thu Jan 30 12:02:26 1997 
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          Thu, 30 Jan 1997 09:01:32 -0800
Date: Thu, 30 Jan 1997 09:01:22 -0800
From: touch@ISI.EDU
Posted-Date: Thu, 30 Jan 1997 09:01:22 -0800
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To: rem-conf@es.net, aboba@internaut.com
Subject: Re: New mailing list on multicasting of Web content
X-Auto-Sig-Adder-By: faber@isi.edu

> From rem-conf-request@es.net Fri Jan 24 10:45:51 1997
> X-UIDL: 854242089.010
> From: Bernard Aboba <aboba@internaut.com>
> To: "'rem-conf@es.net'" <rem-conf@es.net>
> Subject: New mailing list on multicasting of Web content
> Date: Fri, 24 Jan 1997 06:51:48 -0800
> Encoding: 45 TEXT
> 
> New mailing list on multicasting Web content
>  
> A new mailing list has been set up for discussing the distribution of
> Web content via "unreliable multicast." To
> subscribe, send a message to "www-multicast-request@w3.org", and put
> "subscribe" in the subject line. The archive of the mailing list is
> available as http://lists.w3.org/Archives/Public/www-multicast/, and
> the FTP server (used for archiving of related documents) is available at
> ftp://ftp.zilker.net/pub/mailcom/WWW-MULTICAST/. 
>  

Why do we need another list??


> From owner-w-bone@net.lut.ac.uk Fri Jun 14 06:04:35 1996
> From: Mark Handley <M.Handley@cs.ucl.ac.uk>
> X-Organisation: University College London, CS Dept.
> X-Phone: +44 171 419 3666
> To: dauphin@sig.enst.fr (Gilles Dauphin)
> Cc: mbone@ISI.EDU, w-bone@mrrl.lut.ac.uk
> Subject: Re: ANNOUNCE: webcast mailing-list
> Date: Fri, 14 Jun 96 13:49:10 +0100
> 
> 
> >   I'd like to announce a new mailing-list called webcast.
> >
> >   The goal of this list is to provide an informal forum for people interested
> >   in multicasting hypertext.
> >
> > Subscription:
> >
> >  Anyone can subscribe by sending a special message to the address 
> >  "listserv@sig.enst.fr" without subject indication but with, on a line by
> >  itself, the message "subscribe webcast" followed by identification (person 
> >  or group); e.g.:
> >
> >  To: listserv@sig.enst.fr
> >  Subject:
> >  subscribe webcast Gilles Dauphin
> >
> >  Subscription confirmation is normally sent back a few minutes later.
> 
> Perhaps I'm confused, but I though this was also (partly) the purpose of
> w-bone@mrrl.lut.ac.uk
> 
> Do these lists serve the same purpose?  If so, can we agree to use only
> one.  If not, could you enlighten me as to the difference between them?
> 
> Mark
> 

----------------------------------------------------------------------
Joe Touch - touch@isi.edu		    http://www.isi.edu/~touch/
ISI / Project Leader, ATOMIC-2, LSAM       http://www.isi.edu/atomic2/
USC / Research Assistant Prof.                http://www.isi.edu/lsam/

From rem-conf-request@es.net Thu Jan 30 14:36:42 1997 
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Date: Thu, 30 Jan 1997 11:33:43 -0800
From: touch@ISI.EDU
Posted-Date: Thu, 30 Jan 1997 11:33:43 -0800
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To: rem-conf@es.net, aboba@internaut.com, touch@ISI.EDU
Subject: Re: New mailing list on multicasting of Web content
X-Auto-Sig-Adder-By: faber@isi.edu

I'm resending this because I believe the brain-dead
rem-conf remailer punted it due to "too much included
text, not enough new".

Please excuse the defeater attached at the bottom...

PS - there appear to be three lists, as a result:

	w-bone@mrrl.lut.ac.uk
	webcast (Mark H's proposal)
	www-multicast@w3.org

I propose that w-bone is the correct place for this
list, because it already exists.

Joe

> From: Bernard Aboba <aboba@internaut.com>
> To: "'rem-conf@es.net'" <rem-conf@es.net>
> Subject: New mailing list on multicasting of Web content
> Date: Fri, 24 Jan 1997 06:51:48 -0800
> 
> New mailing list on multicasting Web content
>  
> A new mailing list has been set up for discussing the distribution of
> Web content via "unreliable multicast." To
> subscribe, send a message to "www-multicast-request@w3.org", and put

Why do we need another list??


> From: Mark Handley <M.Handley@cs.ucl.ac.uk>
> To: dauphin@sig.enst.fr (Gilles Dauphin)
> Cc: mbone@ISI.EDU, w-bone@mrrl.lut.ac.uk
> Subject: Re: ANNOUNCE: webcast mailing-list
> Date: Fri, 14 Jun 96 13:49:10 +0100
> 
> >   I'd like to announce a new mailing-list called webcast.
> >
> >   The goal of this list is to provide an informal forum for people interested
> >   in multicasting hypertext.
> 
> Perhaps I'm confused, but I though this was also (partly) the purpose of
> w-bone@mrrl.lut.ac.uk
> 
> Do these lists serve the same purpose?  If so, can we agree to use only
> one.  If not, could you enlighten me as to the difference between them?
> 
> Mark

(defeater text follows, to ensure that
including two old messages plus a small
amount of information isn't mistaken as
'me-too' mail

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----------------------------------------------------------------------
Joe Touch - touch@isi.edu		    http://www.isi.edu/~touch/
ISI / Project Leader, ATOMIC-2, LSAM       http://www.isi.edu/atomic2/
USC / Research Assistant Prof.                http://www.isi.edu/lsam/

From rem-conf-request@es.net Thu Jan 30 15:19:29 1997 
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Date: Thu, 30 Jan 1997 11:51:28 -0800
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To: 298-list@bugs-bunny.CS.Berkeley.EDU
From: Jerrlyn Iwata <jerrlyn@postgres.berkeley.edu>
Subject: Berkeley Multimedia & Graphics Seminar

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

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

                                Dave Mentley 
                            Stanford Resources, Inc. 
                                San Jose, CA 

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

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

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


From rem-conf-request@es.net Thu Jan 30 15:58:06 1997 
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To: rem-conf@es.net
Subject: SunFastEthernet if_bqe driver
Mime-Version: 1.0
Content-Type: text/plain; charset=us-ascii
Date: Thu, 30 Jan 1997 15:57:09 -0500
From: Mike Minnich <minnich@aladdin.es.dupont.com>


Does anyone have a recent (working) if_bqe driver for SunOS 4.1.3_U1
with the requisite IP Multicast patches applied?

The version supplied with the release3.5 kernel mods does not work
with our boards (rev -08).  Sun patch 103345-01 supplies a more
recent driver which works fine, but as you might guess it is not
multicast capable.

Mike



From rem-conf-request@es.net Thu Jan 30 16:34:15 1997 
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From: Bernard Aboba <aboba@internaut.com>
To: "rem-conf@es.net" <rem-conf@es.net>, "'touch@ISI.EDU'" <touch@ISI.EDU>
Subject: RE: New mailing list on multicasting of Web content
Date: Thu, 30 Jan 1997 13:29:51 -0800
Encoding: 9 TEXT

The WWW-MULTICAST list was created for the explicit purpose of preparing
for a future IETF BOF on the topic of unreliable multicasting of Web content. 
Should an IETF working group be formed, it is expected that the list will
transition over to use for working group discussions. 

Since the list has explicit goals (i.e. preparation of Internet-Drafts
reviewing existing work, and defining the problem) relating to IETF
activity, its purpose is distinct from the other lists you mention.  



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To: rem-conf@es.net
From: osserman <osserman@msri.org>
Reply-to: osserman@msri.org
Subject: Gordon Shaw: Computation by Symmetry Operations in a Structured Neural 
         Model of the Brain: Music and Abstract Reasoning

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

Title:       
	Gordon Shaw: Computation by Symmetry Operations in a Structured Neural Model of the Brain:  Music and Abstract Reasoning
Date:        
	Feb 03, 1997

Time:        
	16:15 PST8PDT 1 hours

Contact:     
	osserman@msri.org

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

Description:        
	 Predictions from our structured neural model of the cortex led us to the hypothesis that music could  causally enhance spatial-temporal reasoning. We have shown a) College students scored significantly  higher on a spatial-temporal reasoning task after listening to a Mozart Sonata (K.448), but not after  listening to silence or to minimalist music. b) Preschool children who received private keyboard lessons for 6  months improved dramatically on a spatial-temporal reasoning task while appropriate control groups did  not improve significantly. Enhancement a) lasted roughly 10 minutes and established the causal effect,  while enhancement b) lasts long enough to have major educational implications. Here we review the model,  in particular, the "built-in" ability of the cortex to recognize symmetry relations among the inherent  spatial-temporal firing patterns, which we suggest is a crucial feature of the cortical relationship between  music and spatial-temporal reasoning. Then,!
!
 we summarize the behavioral results, and make further  predictions relevant to education.









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

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To: rem-conf@es.net
From: osserman <osserman@msri.org>
Reply-to: osserman@msri.org
Subject: Gordon Shaw: Computation by Symmetry Operations in a Structured Neural 
         Model of the Brain: Music and Abstract Reasoning

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

Title:       
	Gordon Shaw: Computation by Symmetry Operations in a Structured Neural Model of the Brain:  Music and Abstract Reasoning
Date:        
	Feb 04, 1997

Time:        
	12:00 PST8PDT 1 hours

Contact:     
	osserman@msri.org

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

Description:        
	Predictions from our structured neural model of the cortex led us to the hypothesis that music could  causally enhance spatial-temporal reasoning. We have shown a) College students scored significantly  higher on a spatial-temporal reasoning task after listening to a Mozart Sonata (K.448), but not after  listening to silence or to minimalist music. b) Preschool children who received private keyboard lessons for 6  months improved dramatically on a spatial-temporal reasoning task while appropriate control groups did  not improve significantly. Enhancement a) lasted roughly 10 minutes and established the causal effect,  while enhancement b) lasts long enough to have major educational implications. Here we review the model,  in particular, the "built-in" ability of the cortex to recognize symmetry relations among the inherent  spatial-temporal firing patterns, which we suggest is a crucial feature of the cortical relationship between  music and spatial-temporal reasoning. Then, !
!
we summarize the behavioral results, and make further  predictions relevant to education.









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

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To: rem-conf@es.net
From: osserman <osserman@msri.org>
Reply-to: osserman@msri.org
Subject: Gordon Shaw: The Mozart Effect

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

Title:       
	Gordon Shaw: The Mozart Effect
Date:        
	Feb 04, 1997

Time:        
	16:15 PST8PDT 1 hours

Contact:     
	osserman@msri.org

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

Description:        
	    Music Training Causes Long-Term Enhancement of Spatial-Temporal  Reasoning and Mathematical Conceptualization in Young School Children     Professor Gordon Shaw of UC Irvine has been a key figure in the music-and-mathematics educational  experiments that have been in the news during the past few years. In some of the experiments, students who  listened to Mozart for ten minutes before taking a standard exam did considerably better than those who did  not. (Various controls were used, including ten minutes of silence, ten minutes of a "relaxation" tape, and  ten minutes of Philip Glass.) More recently, extensive experiments have been carried out with pre-schoolers  to see if exposure to music training has a long-lasting effect on brain development and the ability to master  certain kinds of spatial-temporal tasks, such as mental rotation of the different parts of a fragmented  picture to recognize the original.     Gordon Shaw was featured recently on Canadian television i!
!
n connection with a school district's decision to  drop its music program in order to purchase large numbers of computers. He describes his current  experiment with a public school kindergarten class in which half the class is given extra music training and  the other half is given extra computer training. Toward the end of the year they will be tested to see if one  or the other half does significantly better on certain standard tests, such as those described above.     The work of Gordon Shaw and his collaborators has broad implications, both for basic theoretical questions  about the structure and function of the brain, and for practical matters involving educational programs. 









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

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Subject: Gordon Shaw: The Mozart Effect

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

Title:       
	Gordon Shaw: The Mozart Effect
Date:        
	Feb 05, 1997

Time:        
	12:00 PST8PDT 1 hours

Contact:     
	osserman@msri.org

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

Description:        
	Music Training Causes Long-Term Enhancement of Spatial-Temporal  Reasoning and Mathematical Conceptualization in Young School Children     Professor Gordon Shaw of UC Irvine has been a key figure in the music-and-mathematics educational  experiments that have been in the news during the past few years. In some of the experiments, students who  listened to Mozart for ten minutes before taking a standard exam did considerably better than those who did  not. (Various controls were used, including ten minutes of silence, ten minutes of a "relaxation" tape, and  ten minutes of Philip Glass.) More recently, extensive experiments have been carried out with pre-schoolers  to see if exposure to music training has a long-lasting effect on brain development and the ability to master  certain kinds of spatial-temporal tasks, such as mental rotation of the different parts of a fragmented  picture to recognize the original.     Gordon Shaw was featured recently on Canadian television in co!
!
nnection with a school district's decision to  drop its music program in order to purchase large numbers of computers. He describes his current  experiment with a public school kindergarten class in which half the class is given extra music training and  the other half is given extra computer training. Toward the end of the year they will be tested to see if one  or the other half does significantly better on certain standard tests, such as those described above.     The work of Gordon Shaw and his collaborators has broad implications, both for basic theoretical questions  about the structure and function of the brain, and for practical matters involving educational programs









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

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To: rem-conf@es.net
From: joe <joe@msri.org>
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Subject: George Andrews: The Death of Proof

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

Title:       
	George Andrews: The Death of Proof
Date:        
	Feb 06, 1997

Time:        
	12:00 PST8PDT 1 hours

Contact:     
	joe@msri.org

URL:         
	http://www.msri.org/lecturenotes/96/andrews/1/

Description:        
	Recently John Horgan in the Scientific American and Doron  Zeilberger in the A.M.S. Notices provided lengthy articles  suggesting that proof as we know it is on the skids. The  object of this talk will be to examine the mathematical  evidence for these assertions. We hope to show that proof  still has some life left in it. We shall especially examine the  role of computer algebra system in this controversy









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

From rem-conf-request@es.net Thu Jan 30 20:43:15 1997 
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Date: Thu, 30 Jan 1997 17:40:41 -0800
To: rem-conf@es.net
From: alagu@mailman.apple.com (Alagu Periyannan)
Subject: Re: MBone teleconf tools for Macs?

At 3:10 PM 1/29/97, Steve Deering wrote:
>Has anyone ported any of the MBone teleconferencing tools to the
>Macintosh yet?
>

Here is what Apple already has or is looking into having
as far as MBone/RTP tools ...

Reception Tools:

Apple has an MBone client called "MBone TV" that handles,
Video: H.261, JPEG, NV
Audio: PCM, GSM

There is an implementation of SDR called "MBone Directory"
which handles receiving announcements. We are looking into
adding announcement generation capabilities.

All the MBone/RTP client software can be downloaded at,
http://qttv.quicktime.apple.com/qttv/qttv.RTP.html

Broadcast Tools:

Apple has a broadcast tool called "Broadcaster Lite" that let's
you broadcast audio/video. We are looking into adding MBone/RTP
capabilities into the broadcaster application soon.

(It currently uses the "MovieTalk" protocol, i.e. does not
use multicast or RTP. It works with Apple's reflector to make
distribution easier in the absence of multicast. Broadcasts
can be viewed with the "Quicktime TV" application.)

The Broadcaster can be downloaded at,
http://qttv.quicktime.apple.com/qttv/qttv.blite.sysreq.html

We're also looking into getting support for other RTP
audio/video payload types such as DVI, etc. for both MBone TV
as well as the Broadcaster.

There has been some change in personel on these projects
which has caused temporary delays.

We will keep users updated through our website at,
http://qttv.quicktime.apple.com/

Thanks for your patience. And stay tuned (to our website).

>
>I am particularly interested in Mac versions of sdr and vat, or
>interoperable equivalents.  wb would also be great, as an adjunct to
>the first two.  vic is much lower priority for me, though it seems that
>implementors always want to do the flashy video stuff first.
>

There is a tool called "Maven" available on the internet
that will interoperate with VAT.




Alagu Periyannan                   alagu@mailman.apple.com

Communications Products and Technology
Apple Computer, Inc.



From rem-conf-request@es.net Thu Jan 30 21:33:08 1997 
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Date: Thu, 30 Jan 1997 18:31:00 -0800
To: alagu@mailman.apple.com (Alagu Periyannan)
From: Steve Deering <deering@cisco.com>
Subject: Re: MBone teleconf tools for Macs?
Cc: rem-conf@es.net

Thanks a lot for the status report, Alagu.  I'm delighted to hear you
say that Apple plans to continue development of RTP- and multicast-
compatible tools.  From the private responses to my message I have
learned that there are a fair number of us Mac lovers anxious to use
the MBone from our favorite platform.

The partitioning into "reception tools" and "broadcast tools" makes some
sense for broadcast-style (one-to-many) applications, but doesn't seem
a very comfortable fit with interactive use of multicast, such as
teleconferencing (many-to-many).  A tool that integrates the sending
and receiving capabilities, like vat, would be much more natural,
in my opinion.  Perhaps extending the Apple VideoPhone app (which I
have not yet tried out) to support multiple participants via multicast
and RTP would be something powerful and easy to do.

> There is a tool called "Maven" available on the internet
> that will interoperate with VAT.

Last I knew, Maven supported point-to-point (i.e., unicast) operation only.

Steve



From rem-conf-request@es.net Fri Jan 31 02:12:23 1997 
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From: Bernard Aboba <aboba@internaut.com>
To: "rem-conf@es.net" <rem-conf@es.net>, 'Stephen Casner' <casner@precept.com>
Subject: RE: Minutes for AVT WG at San Jose IETF
Date: Thu, 30 Jan 1997 23:08:34 -0800
Encoding: 26 TEXT

Steve Casner said:

Should the specification of an RTP session be
generalized to allow different addresses to be used for RTP and RTCP?
This would allow RTCP delivery to be constrained for improved
scalability in some scenarios.  After some discussion, the consensus
of the group was that we should take the conservative approach and not
make this change at this time.

I think this point is worth some debate. While placement of RTP and RTCP
on separate groups is certainly not a panacea, allowing this would have
a pretty big impact for networks running SM-PIM. For a session with
1 sender and 20K receivers, we're talking about the difference between
20K potential (S,G) forwarding table entries, and one (S,G) entry,
with the rest on the shared tree. So this modification has the potential
to dramatically enhance scalability in one to many scenarios. 

What I would suggest is to keep RTP/RTCP on the same group
by default, but define a profile that will cause them to use
separate groups. Similarly, I would argue for a profile with
no RTCP, and one which defines the bandwidth fraction to be
some number less than 5 percent, for asymmetric networks. 

If we do all this, plus fix the RTCP "spike" problem via reconsideration,
we'll have made a substantial step toward enabling major multicast
deployments. 


From rem-conf-request@es.net Fri Jan 31 03:10:13 1997 
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From: Stephen Casner <casner@precept.com>
To: Bernard Aboba <aboba@internaut.com>
cc: "rem-conf@es.net" <rem-conf@es.net>
Subject: RE: Minutes for AVT WG at San Jose IETF
In-Reply-To: <01BC0F02.85024140@multi>
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> I think this point is worth some debate. While placement of RTP and RTCP
> on separate groups is certainly not a panacea, allowing this would have
> a pretty big impact for networks running SM-PIM. For a session with
> 1 sender and 20K receivers, we're talking about the difference between
> 20K potential (S,G) forwarding table entries, and one (S,G) entry,
> with the rest on the shared tree. So this modification has the potential
> to dramatically enhance scalability in one to many scenarios. 

I agree that SM could be a significant savings for this scenario.
However, I thought that the decision to create (S,G) state was made
independently for each source, based on the data rate from that
source.  If that is accurate, wouldn't the savings be the same even
with one address?

> Similarly, I would argue for a profile with
> no RTCP, and one which defines the bandwidth fraction to be
> some number less than 5 percent, for asymmetric networks. 

Doesn't the 5-second minimum interval keep the data rate low enough
for the small-group case on asymmetric networks?  You cited an example
of a net with 1Mb/s downstream and 50kb/s upstream (or something like
that); 1 RTCP packet every 5 seconds is only about 200b/s.  As the
upstreams of multiple users come together, I would hope that the
bandwidth available increases.

> If we do all this, plus fix the RTCP "spike" problem via reconsideration,
> we'll have made a substantial step toward enabling major multicast
> deployments. 

That, and specify the sampling technique for tracking SSRC id's to
reduce memory requirements.
							-- Steve


From rem-conf-request@es.net Fri Jan 31 03:35:21 1997 
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From: Manoj Leelanivas <mleelani@cisco.com>
Message-Id: <199701310832.AAA22467@chow.cisco.com>
Subject: Re: Minutes for AVT WG at San Jose IETF
To: aboba@internaut.com (Bernard Aboba)
Date: Fri, 31 Jan 1997 00:32:07 -0800 (PST)
Cc: rem-conf@es.net, casner@precept.com
In-Reply-To: <01BC0F02.85024140@multi> from "Bernard Aboba" at Jan 30, 97 11:08:34 pm
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=>    
=>    Steve Casner said:
=>    
=>    Should the specification of an RTP session be
=>    generalized to allow different addresses to be used for RTP and RTCP?
=>    This would allow RTCP delivery to be constrained for improved
=>    scalability in some scenarios.  After some discussion, the consensus
=>    of the group was that we should take the conservative approach and not
=>    make this change at this time.
=>    
=>    I think this point is worth some debate. While placement of RTP and RTCP
=>    on separate groups is certainly not a panacea, allowing this would have
=>    a pretty big impact for networks running SM-PIM. For a session with
=>    1 sender and 20K receivers, we're talking about the difference between
=>    20K potential (S,G) forwarding table entries, and one (S,G) entry,
=>    with the rest on the shared tree. So this modification has the potential
=>    to dramatically enhance scalability in one to many scenarios. 

Just to clarify, the protocol specification for PIM-SM specifies:

- Creation of (S, G) at the RP and subsequent joining of source tree,
only if the register rate from S warrants it. 
- Creation of (S, G) in the leaf routers is also triggered only if the
packets from that particular S has crossed the threshold for switching
to source tree.

So, even with one group for both RTP/RTCP, sparse mode as specified by
the draft should be OK. It wont create tons of (S, G)s.
Of course, some implementations switch to
shortest path tree if the cumulative threshold on the (*, G) is
exceeded (rather than per S) and in that case having two groups is
beneficial to avoid lot of (S, G) states.

There is another motivation for assigning RTP and RTCP different
groups. If you have 20K receivers sending RTCP reports, the RP is
going to get the load of large number of registers(intermittent from
each source). If we have 2 RPs, one serving the RTP group-range and
another serving an RTCP group range, then the RTCP register load (This
may be helpful only if you have 20K+ receivers as you mentioned),
could be diverted to the 2nd RP.

-Manoj

From rem-conf-request@es.net Fri Jan 31 08:47:46 1997 
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Date: Fri, 31 Jan 1997 14:47:41 +0100
From: Sven Claussen <claussen@ira.uka.de>
Organization: University of Karlsruhe
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To: rem-conf@es.net
Subject: Preferences for SDR-Tools???
Content-Type: text/plain; charset=us-ascii
Content-Transfer-Encoding: 7bit

I am using the sdr-tools on a sun-machine for broadcasting
lessons out of the university. After starting the programs
(nv, wb, vic, vat) I have to configure them. And this again
and again after each new session.
Can I create a preference file somewhere to store my favorite
configuration? Or can I put all the parameter I needed into
the function call, e.g. vic -f h261 -t 47 etc.??
Who has further information about the parameters of the call?
The manpages I got are not sufficient:-( 
I like to call the tools as a cron-job. Does this work?

Sven
-- 
***************************
* Sven Claussen           *
* Fakultaet f. Informatik *
* Universitaet Karlsruhe  *
* claussen@ira.uka.de     *
***************************

From rem-conf-request@es.net Fri Jan 31 20:26:38 1997 
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Date: Fri, 31 Jan 1997 17:21:01 -0800 ()
From: Stephen Casner <casner@precept.com>
To: Manoj Leelanivas <mleelani@cisco.com>
cc: rem-conf@es.net
Subject: Re: Minutes for AVT WG at San Jose IETF
In-Reply-To: <199701310832.AAA22467@chow.cisco.com>
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Manoj,

> Just to clarify, the protocol specification for PIM-SM specifies:
> 
> - Creation of (S, G) at the RP and subsequent joining of source tree,
> only if the register rate from S warrants it. 
> - Creation of (S, G) in the leaf routers is also triggered only if the
> packets from that particular S has crossed the threshold for switching
> to source tree.
> 
> So, even with one group for both RTP/RTCP, sparse mode as specified by
> the draft should be OK. It wont create tons of (S, G)s.

Thanks for clarifying; that's what I thought.

> Of course, some implementations switch to
> shortest path tree if the cumulative threshold on the (*, G) is
> exceeded (rather than per S) and in that case having two groups is
> beneficial to avoid lot of (S, G) states.

Is that because the leaf router essentially has to keep some (S,G)
state in order to measure the rate for that S?  Does this negate the
points you clarified in most (not just some) implementations?  That
is, in practice would we need two groups in order to avoid lots of
(S,G) state?  Or do we just need to make sure implementers know that
there is a good reason to do what the spec says.

> There is another motivation for assigning RTP and RTCP different
> groups. If you have 20K receivers sending RTCP reports, the RP is
> going to get the load of large number of registers(intermittent from
> each source). If we have 2 RPs, one serving the RTP group-range and
> another serving an RTCP group range, then the RTCP register load (This
> may be helpful only if you have 20K+ receivers as you mentioned),
> could be diverted to the 2nd RP.

Good thought, but this doesn't divide the work evenly.  For the RTP RP
there is only one source in this scenario, and all 20K of the other
sources are on the RTCP RP.  To handle many sessions, you could have a
large range assigned to the RTP RP, and a bunch of RTCP RP's each with
a small range, but the effect would be essentially the same with an
equal number of small ranges for RPs serving combined RTP and RTCP.

Also, if we did have separate addresses for RTP and RTCP, we'd
probably want a simple arithmetic relationship between the addresses.
That may make it difficult to achieve the desired mapping to separate
RPs, depending on the mapping function.
							-- Steve


From rem-conf-request@es.net Fri Jan 31 20:54:18 1997 
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Date: Fri, 31 Jan 1997 20:53:29 -0500 (EST)
From: kevin@cc.gatech.edu (Kevin C. Almeroth)
Message-Id: <199702010153.UAA11260@flora.cc.gatech.edu>
To: claussen@ira.uka.de, rem-conf@es.net
Subject: Re: Preferences for SDR-Tools???

>>I am using the sdr-tools on a sun-machine for broadcasting
>>lessons out of the university. After starting the programs
>>(nv, wb, vic, vat) I have to configure them. And this again
>>and again after each new session.
>>Can I create a preference file somewhere to store my favorite
>>configuration? Or can I put all the parameter I needed into
>>the function call, e.g. vic -f h261 -t 47 etc.??
>>Who has further information about the parameters of the call?
>>The manpages I got are not sufficient:-( 
>>I like to call the tools as a cron-job. Does this work?

Vic and vat are extremely flexible this way...  Here is
one set of parameters I use to configure everything and even
start transmitting.

Check out the man pages for all the options.

vat -r -I2 -N"Session Source" -fdvi2 -t16 \
    -XlectureMode=ture \
    -XmikeMute=false \
    -XlineinGain=40  \
    -XinputPort=Line \
    -XspeakerMode=FullDuplex \
    -XlineoutMode=FullDuplex \
    -Xlineout2Mode=FullDuplex \
    224.2.129.147/17982 &
 
vic -Dsunvideo -I2 -B256 -N"Session Source" \
    -Xframerate=30 \
    224.2.129.147/63984//16 &


-Kevin Almeroth

From rem-conf-request@es.net Fri Jan 31 22:28:20 1997 
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From: Manoj Leelanivas <mleelani@cisco.com>
Message-Id: <199702010327.TAA03048@chow.cisco.com>
Subject: Re: Minutes for AVT WG at San Jose IETF
To: casner@precept.com (Stephen Casner)
Date: Fri, 31 Jan 1997 19:27:30 +73600 (PST)
Cc: mleelani@cisco.com, rem-conf@es.net
In-Reply-To: <Pine.WNT.3.95.970131170529.-230513E-100000@oak.precept.com> from "Stephen Casner" at Jan 31, 97 05:21:01 pm
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=>    
=>    Manoj,
=>    
=>    > Just to clarify, the protocol specification for PIM-SM specifies:
=>    > 
=>    > - Creation of (S, G) at the RP and subsequent joining of source tree,
=>    > only if the register rate from S warrants it. 
=>    > - Creation of (S, G) in the leaf routers is also triggered only if the
=>    > packets from that particular S has crossed the threshold for switching
=>    > to source tree.
=>    > 
=>    > So, even with one group for both RTP/RTCP, sparse mode as specified by
=>    > the draft should be OK. It wont create tons of (S, G)s.
=>    
=>    Thanks for clarifying; that's what I thought.
=>    
=>    > Of course, some implementations switch to
=>    > shortest path tree if the cumulative threshold on the (*, G) is
=>    > exceeded (rather than per S) and in that case having two groups is
=>    > beneficial to avoid lot of (S, G) states.
=>    
=>    Is that because the leaf router essentially has to keep some (S,G)
=>    state in order to measure the rate for that S?

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

=>    Does this negate the
=>    points you clarified in most (not just some) implementations?  That
=>    is, in practice would we need two groups in order to avoid lots of
=>    (S,G) state?  Or do we just need to make sure implementers know that
=>    there is a good reason to do what the spec says.

I am not sure what to say here. There is some justification from the
implementors point of view to keep the source state switching simple
(ie not caching each source and its packets) by keeping a cumulative
threshold for the (*, G). But then you need 2 groups, if you want to
keep the state down on the routers. But if all implementations adhere
to the spec(keeping a source/rate cache around), then we dont need 2
groups. So,IMHO we cannot justify 2 groups for this problem.


=>    
=>    > There is another motivation for assigning RTP and RTCP different
=>    > groups. If you have 20K receivers sending RTCP reports, the RP is
=>    > going to get the load of large number of registers(intermittent from
=>    > each source). If we have 2 RPs, one serving the RTP group-range and
=>    > another serving an RTCP group range, then the RTCP register load (This
=>    > may be helpful only if you have 20K+ receivers as you mentioned),
=>    > could be diverted to the 2nd RP.
=>    
=>    Good thought, but this doesn't divide the work evenly.  For the RTP RP
=>    there is only one source in this scenario, and all 20K of the other
=>    sources are on the RTCP RP.  To handle many sessions, you could have a
=>    large range assigned to the RTP RP, and a bunch of RTCP RP's each with
=>    a small range, but the effect would be essentially the same with an
=>    equal number of small ranges for RPs serving combined RTP and RTCP.
=>  

The RP which is switching RTP traffic also may have sources which are
sending RTP streams interrmittently. Even otherwise, it has to switch
the traffic for the active source(ie, it is indeed loaded to an extent though
switching path may be much faster). What we could do is essentially
moving the RTCP load to another RP, so that the primary RP can still
have enough CPU to handle data streams.

-Manoj

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Date: Fri, 31 Jan 1997 19:48:16 -0800 ()
From: Stephen Casner <casner@precept.com>
To: Manoj Leelanivas <mleelani@cisco.com>
cc: rem-conf@es.net
Subject: Re: Minutes for AVT WG at San Jose IETF
In-Reply-To: <199702010327.TAA03048@chow.cisco.com>
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> Yes, it has to keep a small source cache(S, G and the rate) to
> implement the way the protocol specifies it. Still the baggage is much
> lesser than the (S, G) entry which needs to keep track of the outgoing
> interfaces and such.

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

> I am not sure what to say here. There is some justification from the
> implementors point of view to keep the source state switching simple
> (ie not caching each source and its packets) by keeping a cumulative
> threshold for the (*, G). But then you need 2 groups, if you want to
> keep the state down on the routers. But if all implementations adhere
> to the spec(keeping a source/rate cache around), then we dont need 2
> groups. So,IMHO we cannot justify 2 groups for this problem.

Yes, even if we changed the spec for RTP to use two groups, other
protocols may have groups involving senders at both high and low
rates.  It does not seem reasonable to assume that all sources sending
to a group will have comparable rates (so that switching them all to
source-rooted trees when the data rate from one exceeds a threshold
will make sense).

> The RP which is switching RTP traffic also may have sources which are
> sending RTP streams interrmittently. Even otherwise, it has to switch
> the traffic for the active source(ie, it is indeed loaded to an extent though
> switching path may be much faster).

But doesn't that load get moved off to a source-rooted tree?

							-- Steve


