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--NextPart

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

	Title		: User Multiplexing in RTP payload between 
                          IP Telephony Gateways
	Author(s)	: B. Subbiah, S. Sengodan 
	Filename	: draft-ietf-avt-mux-rtp-00.txt
	Pages		: 17
	Date		: 31-Aug-98
	
   This draft proposes a method to multiplex a number of low bit rate
   audio streams (upto 256) into a single RTP/UDP/IP connection between
   IP telephony gateways. Audio samples from different users are assem-
   bled into an RTP payload thus reducing the overhead of RTP/UDP/IP
   headers. To identify users sharing a single RTP/UDP/IP connection,
   a 2 byte MINI-Header is proposed. A channel negotiation procedure to
   assign and release channels on a single UDP connection between
   gateways is described.


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From rem-conf Tue Sep 08 05:26:26 1998 
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Date: Tue, 08 Sep 1998 12:36:38 +0100
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I enclose a draft of the minutes for the Chicago AVT meeting. In order to
meet the submission deadline, I'd be grateful if you could send comments
or corrections to me by Thursday evening.

Colin


===============================================================================
Minutes of the Audio/Video Transport Working Group

Reported by Colin Perkins

The Audio/Video Transport working group met twice at the 42nd IETF in
Chicago. The major items of discussion were the revision of RTP and the
Audio/Video profile for advancement to draft standard, multiplexing of RTP
streams and transport of MPEG4 streams using RTP.

The meeting opened with a review of the status of the working group
documents. The RTP header compression scheme and the payload format
documents for JPEG, BT656 and H.263+ video are now either awaiting IESG
approval or are in last call. The payload format for bundled MPEG streams
has been published as RFC2343 (experimental). The Options for Repair of
Streaming Media document has been published as RFC2354 (informational). 

The draft giving guidelines for writers of RTP payload format documents
(draft-ietf-avt-rtp-format-guidelines-00.txt) is still awaiting revision.
Comments and suggestions are encouraged in order that this document can
effectively capture the knowledge of the group as an aid to the designers
of future payload formats. 

The RTP payload format for PureVoice audio (draft-mckay-qcelp-01.txt) was 
discussed briefly with a view to conducting last call for proposed standard 
soon. Some concerns were raised regarding the means for signalling encryption 
of the media data only (not the RTP headers) by means of a bit in the payload 
sub-header, since out-of-band signalling, for example using an SDP attribute, 
may be more appropriate. Clarification of this issue is needed before the 
draft can go forward, but the document is otherwise complete.

The generic RTP payload format was not discussed this time. The authors of
the proposals presented at the last meeting were to produce a combined
proposal, but this is not yet complete. It is expected that further
discussion of this subject will take place during the next meeting.

The RTP MIB (draft-ietf-avt-rtp-mib-02.txt) was presented by Mark Baugher
(Intel). The changes since the previous draft are the elimination of
separate tables for hosts and monitors, with an rtpSessionMonitor variable
being added to distinguish the two modes. The rtpFracLost variable was
removed since it was found not to be useful. The definition of a session
was clarified.

The RTP MIB will be referenced by the H-series MIBs which are being
defined, but the RTP MIB itself will remain an AVT work item (since RTP
experience is in this group). The H-series MIB developers will review the
RTP MIB to ensure that it meets their needs.

The main area of concern with this current version of the RTP MIB is the
complexity of the compliances definitions resulting from the merger of the
tables. It is expected that the MIB will be ready for last call after a
further round of revisions, probably by the time of the next meeting.
A reference implementation of the MIB is now available.

The major topic of discussion on the first day was multiplexing multiple 
streams into a single RTP stream. This discussion started with an introduction 
by Stephen Casner (Cisco) who raised a number of issues for consideration in 
the following discussion:
- Why are we multiplexing: because common handling is needed or to reduce 
  overhead?
- When should we multiplex: what should be separate and what should be 
  bundled?
- Where should we multiplex? At which protocol level? Can we keep all 
  multiplexing at one protocol level?
- How should we multiplex: an application specific solution or a general 
  purpose one?
A pointer was also given to Jonathan Rosenberg's internet-draft on this
subject from December 1996, which discusses these issues. Copies of this
may be obtained from http://www.cs.columbia.edu/~jdrosen/aggregate/rtpdoc.htm 
and a revised and updated version of this will be re-posted.

Following the introduction, a number of proposals were presented: Jonathan
Rosenberg (Lucent) and Barani Subbiah (Nokia) presented somewhat similar
proposals applicable for the efficient connection of PSTN gateways using
RTP.   Tohru Hoshi (Hitachi) and Mark Handley (ISI) presented more general
proposals.

The first presentation was of draft-ietf-avt-aggregation-00.txt by Jonathan 
Rosenberg. This proposal deals with the interconnection of telephony
gateways only - it is not a general purpose multiplexing protocol. Some of
the main features of this proposal are:
- Users are identified by a single 7 bit identifier. The mapping from SSRC 
  to this identifier is done by non-RTP means (eg: SIP/H.323 signalling). If 
  more than 127 streams are being transported between two gateways multiple 
  multiplexed RTP sessions must be used.
- All multiplexed streams must share a common clock and generate packets at 
  integer multiples of a common frame duration. They do not need to use a 
  common codec.
- Payload type information is transported for each multiplexed user. In many 
  cases the payload type specifies the length of the packet so there is no 
  need for a separate length indication field (although one can optionally be 
  provided if necessary).
- User payloads are not word aligned. Aligning them would reduce the bandwidth 
  efficiency significantly.
It was noted that statistical multiplexing of multiple streams using
silence suppression can cause problems: there is a limit to how many
streams can be packed into a single multiplexed packet before exceeding 
the network MTU. To avoid this two multiplexed packets must be generated,
but this will cause the receiver to see non-contiguous sequence numbers 
per user giving the appearance of loss. The solution is to limit the
statistical multiplexing of streams so that this does not occur, this
reduces the efficiency somewhat.

It was also noted that the loss of a single packet from the multiplexed
stream will affect all users multiplexed into the stream. It is not
expected that this will be a problem (in fact the use of multiplexing may
reduce loss rates, since it reduces the both the data and packet rates
compared to non-multiplexed streams).

Barani Subbiah (Nokia) presented draft-ietf-avt-mux-rtp-00.txt. This
proposal is designed to solve a similar problem to that of Rosenberg and is
unsurprisingly somewhat similar to that proposal. The main difference seems
to be that this protocol uses an explicit length indication for each
multiplexed packet (6 bits rather than 16 bits in Rosenberg's proposal) and
that the payload type for each user is signalled out-of-band rather than
carried in the payload (disallowing changing payload types on the fly).

Neither of these two proposals carry the timestamp of the multiplexed
packets explicitly: the timestamp of the enclosing RTP packet must be used.
This may introduce some small additional amount of jitter, although it is
not thought that this will be a problem.

A more generic proposal (draft-tanigawa-rtp-multiplex-00.txt) was presented
by Tohru Hoshi (Hitachi). In this proposal RTP streams are multiplexed,
rather than voice streams, by concatenating multiple RTP packets into one.
This allows for the multiplexing of any sort of data, rather than just
voice data, at the expense of additional overheads. Once again, out of band
signalling is required to indicate that this is a multiplexed stream. It
was noted that it may be possible to generalise this as a generic IP
multiplexing protocol, rather than an RTP multiplexer.

These proposals discuss out-of-band signalling which is required for
correct operation of these protocols. It is noted that, whilst signalling
is required in these cases and should be simple to implement using either
SIP or H.323, it is outside the scope of the AVT group's charter.

It was noted that the presence of multiple multiplexing solutions is not
necessarily desirable, since this hinders interoperability. It would be
desirable to combine these proposals into one if at all possible. However,
it was further noted that the drafts from Rosenberg and Subbiah are
essentially solving the same problem, whilst the proposal presented by
Hoshi is doing something different. The tradeoffs are different and we may
need two protocols: the issues are sufficiently different for the two
scenarios.

None of the three proposals presented so far have solved the generic
multiplexing problem: the first two are clearly very application specific,
the third requires out of band signalling to operate.

The second session started with a brief presentation by Mark Handley (ISI)
describing an idea which resulted from the earlier discussion of
multiplexing. This proposal, MuRGE, uses the techniques of RTP header
compression within a single packet as a generic multiplexing method (all
state is reinitialised within each packet). That is, each packet contains a
standard RTP header followed by a number of payloads each with their own
payload header. The payload headers are coded as differences from the
previous header. Clearly the bandwidth efficiency of this proposal depends
on the similarity between the headers of the multiplexed payloads. If used
between cooperating gateways where SSRC values can be allocated
consecutively and the codecs, timestamps and sequence numbers are
synchronised, this proposal can produce a single byte header for each
multiplexed packet. If there is no cooperation between multiplexing points
the full RTP header has to be sent for each multiplexed stream. If
signalling is employed between multiplexing points (eg: for SSRC mapping)
then some gain can be made even in the most generic case. This proposal is
at a very early stage of development, but introduces some interesting
ideas. Further work is clearly needed.

The discussion of multiplexing concluded with agreement that the development 
of a multiplexing proposal is of interest and should become a work item of
the group.

The next item for discussion was transport of MPEG4 streams using RTP 
draft-ietf-avt-rtp-mpeg4-00.txt and the role of DMIF signalling 
draft-ietf-avt-mpeg4-dmif-01.txt which was presented by Vahe Balabanian 
(Nortel). After outlining the proposals discussion focused on a number 
of open issues:
- Should MPEG4 elementary streams be transported directly over RTP or 
  should they be encapsulated using FlexMux first? There is some concern 
  that the use of FlexMux does not cleanly fit into the RTP model in 
  particular the interaction with RTP mixers is unclear.
- The mapping of MPEG4 scene and object descriptor streams to RTP is 
  unclear. It may be that these need special transport and protocols
  other than RTP may be better suited to their needs: in particular
  the initial session description should not be carried in RTP. The
  transport of dynamic updates to this is an area which needs further
  study: an RTP stream may be appropriate or alternatively a separate
  signalling stream (eg: RTSP using ANNOUNCE) may work better.
- The mapping of MPEG4 decoder timestamps to RTP is unclear, since RTP
  includes only a send timestamp and applications are expected to derive
  their own decode time based on the observed network timing jitter.
The next IETF meeting coincides with the MPEG4 decision meeting, so it is
therefore urgent that these issues are resolved. The last chance to make
changes to MPEG4 version 1 will be at the MPEG meeting on 12-16 October.
Since it is clear that only limited progress can be made in the short time
period available it was decided to continue with the development of the
payload format for MPEG4 elementary streams and to resolve the issues
discussed. Once this is done and further experience has been gained with
actual implementations the group will revisit these issues.

The major discussion item for the second day was the advancement of RTP and 
the Audio/Video profile from proposed- to draft-standard status. A summary 
of the changes in draft-ietf-avt-rtp-new-01.txt was made by Stephen Casner
(Cisco). These include
- Added fudge factor in timer reconsideration
- Added fix for underestimate when using SSRC sampling if the group size
  decreases rapidly
- RTCP sender and receiver bandwidth may be specified as a parameter
  (rather than the default 5%)
- RTCP minimum interval may scale smaller for high bandwidth sessions and
  zero initial delay for unicast sessions
- Specified padding for RTCP only on the last packet
- Specified relative NTP uses the "best" platform clock
- Formal reference to IPsec for security (this concerns some people since
  RTP may be used in scenarios where the presence of IPsec cannot be
  guaranteed...)
- Partial conversion to SHOULD, MUST, MAY, etc

In addition it has been decided not to make a number of the changes which
have previously been suggested: 
- Ignore group size dropping to zero with reverse reconsideration. 
- No scaling of the RTCP interval larger since this could cause time-outs
- No changes to the jitter algorithm for multi-packet video frames
- No additional SDES items were defined (these can be registered with 
  the IANA separately)
- No change to the definition of the RTCP RR loss fraction
- Nothing was added about translators adding random timestamp offsets

The issue of conditional vs unconditional reconsideration was discussed and
it was noted that there is little to choose between these two algorithms in
practise, and that unconditional reconsideration is simplest to implement.
The next revision will therefore only include unconditional reconsideration.

A number of problems which have been discovered in the SSRC sampling
algorithm were presented by Jonathan Rosenberg (Lucent). The problems with
reconsideration and over-weighting of senders have been corrected in the
current RTP draft. A problem remains when the group size decreases rapidly
which results in members using SSRC sampling producing very inaccurate
estimates of the group size. A solution using a "binning" algorithm is
proposed in draft-ietf-avt-rtpsample-00.txt but this algorithm may be
patented by Lucent (who are willing to license on "fair, reasonable and
non-discriminatory terms"). 

Much discussion occurred on this topic since it is considered undesirable to
have patented technology as part of the main specification. The cleanest
solution appears to be to move the SSRC sampling algorithms out of the main
specification into a separate document. The main RTP specification would
note that SSRC sampling may be desirable in certain cases and point to this
new document for implementation advice and sample algorithms.

The changes to the Audio/Video profile (draft-ietf-avt-profile-new-03.txt)
are less extensive: the PureVoice codec has been added and assigned payload
type 12. As discussed at the previous meeting this is the last static
assignment which will be made - this policy is now stated explicitly in
the draft. The static assignment of payload type 77 to redundant audio
has been removed since all known implementations use a dynamic payload
type. References to MPEG1 system streams and MPEG2 program streams
(RFC2250) using dynamic payload types have been added and a number of other
RFC references have been updated.

The new policy regarding static payloads needs to be better described in
the next revision of the draft. The SDP modifiers to explicitly denote the
RTCP fraction (if the default of 5% is not being used) have yet to be
written, but it is felt that these should not be added to the A/V profile
(since that should not be tied to SDP) rather a new document should define
them.

The MIME registration of payload formats has yet to be done. There are many
open issues here regarding how this should occur, what information is bound
to the names, who MIME is extended for types which cannot be represented in 
email, etc.

An update on the RTP payload for redundant audio was presented by Colin
Perkins (UCL). This document (draft-ietf-avt-rtp-redundancy-revised-00.txt)
updates RFC2198 in the light of additional usage experience. The change is
to specify that all packets in a redundant stream should be sent using the
redundancy format, rather than sending the first packet(s) in a talkspurt
using the payload format of the primary codec. This allows for explicit
advertisement of the buffering requirements of a stream which simplifies
implementations and removes the need for an out-of-band parameter to convey
this information.

An update on the RTP payload for generic forward error correction
(draft-ietf-avt-fec-03.txt) was presented by Jonathan Rosenberg (Lucent).
Changes since the previous draft include the addition of code fragments
illustrating the decoding stage, support for FEC using Reed-Solomon codes,
extension of the timestamp recovery to 56 bits and removal of the reference
to the expired draft by Budge. A number of issues with the current draft
were discussed including the required mask size: 24 bits is believed
sufficient so the optional extension to 56 bits will be removed from the
draft. 

It was also decided that parity FEC and Reed-Solomon codes are sufficiently
different that this draft should be split into two. The resulting parity
FEC payload format document is expected to be ready for last call after one
further revision; the Reed-Solomon payload format document will need
further work over the coming months.

The meeting concluded with a reminder that a revised working group charter
has been posted. Comments and discussion of the proposed new charter and
milestones should be directed to the mailing list.  
===============================================================================



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To: Web@Placement
Bcc: liz@ervose.mv.com, heather_fox@es.adp.com, james_kinder@es.adp.com, thomasgf@es.dupont.com, dcewg@es.net, rem-conf@es.net
From: <Huvar@msn.com>
Subject: DRAMATICALLY increase your website placement in as little as 72 hours!!!!
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  Increase Your Website Traffic Within 72 Hours!

    We'll put YOUR website at the top of the search engines - FAST!
 

If your site isn't getting the traffic it should, chances are it's because
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--Search Engine Success Group--
      410-783-8269                 




-----------------------
If you're not interested in any potential future offers, 
just click reply. Thank you.







From rem-conf Wed Sep 09 06:39:35 1998 
From rem-conf-request@es.net Wed Sep 09 06:39:34 1998
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From: Parveen Kumar <parveenk@comp.nus.edu.sg>
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Hi,

Are there applications(reasonably stable) which uses RTP
multiplexing/demultiplexing?

Or most of the applications just use use one rtp session 
for each session?

Sorry for my ignorance but I have a limited knowledge in this area.
Could anyone of you give me pointers to such working applications.


Thanx...

Parveen Kumar
MS, NUS singapore





From rem-conf Wed Sep 09 07:13:39 1998 
From rem-conf-request@es.net Wed Sep 09 07:13:38 1998
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To: Parveen Kumar <parveenk@comp.nus.edu.sg>
cc: rem-conf@es.net
Subject: Re: Hi
In-reply-to: Your message of "Wed, 09 Sep 1998 18:10:59 +0800." <Pine.SOL.3.96.980909180517.2379D-100000@sun450.iscs.nus.edu.sg>
Date: Wed, 09 Sep 1998 15:00:20 +0100
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From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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--> Parveen Kumar writes:
>Are there applications(reasonably stable) which uses RTP
>multiplexing/demultiplexing?

If you mean applications which act as an RTP mixer, then rat-3.0 will do
that. See http://www-mice.cs.ucl.ac.uk/multimedia/software/ for details.

Colin



From rem-conf Wed Sep 09 07:56:37 1998 
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From: Mark Handley <mjh@east.isi.edu>
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To: Parveen Kumar <parveenk@comp.nus.edu.sg>
cc: rem-conf@es.net
Subject: Re: Hi 
In-reply-to: Your message of "Wed, 09 Sep 1998 18:10:59 +0800."
             <Pine.SOL.3.96.980909180517.2379D-100000@sun450.iscs.nus.edu.sg> 
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>Hi,
>
>Are there applications(reasonably stable) which uses RTP
>multiplexing/demultiplexing?
>
>Or most of the applications just use use one rtp session 
>for each session?
>
>Sorry for my ignorance but I have a limited knowledge in this area.
>Could anyone of you give me pointers to such working applications.

To my knowledge, there aren't any publically available.  

In general RTP-level multiplexing of different media or sessions is a
bad idea.  It introduces an additional unnecessary level of
multiplexing above the UDP port level, and makes it impossible to
tailor QoS to the different streams.

However I can also see that there are some cases (typically IP
telephony between large PSTN gateways) where the bandwidth savings
will be significant and different QoS is never required, so this is
going to happen anyway.  It's not clear to me though that the makers
of such gateways are likely to make their code available.

Also we now have four different proposals for how to do this, so its
not clear that having code would help you right now.

Cheers,
	Mark





From rem-conf Wed Sep 09 08:15:54 1998 
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Subject:  rem-conf Let Us Promote Your Site -gnwpvtwk
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</B></I></U>
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<P>------------------------------CUT HERE-------------------------------------</B></I>
<P>SECTION 1: Personal Information </B></I>
<P>Your Full Name:__________________________</B></I>
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<P>NOTE: Each submission site has different submission forms. We will complete all available fields, however, all of the following information may not be listed at all sites. </B></I>
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<P>NOTE: If you wish to have us submit more than one web page please fill out and submit this form for each individual page (simply fill out the form, submit it, make any necessary changes and submit it again). We can submit any page that has a unique URL.</B></I>
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<P>SECTION 3: Payment Information</B></I>
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<P>NOTE:  you can call us in person at our voice or Fax number, OR you may fill out this form, print it out and Fax it to us. </B></I>
<P>Lucid Technologies Inc</B></I>
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<P>Boca Raton, FL 33433</B></I>
<P>Our voice number is: 561-347-7304, our Fax number is: 561-347-7304</B></I>
<P>Also if you would like your site or product promoted by mass email or wish to buy mass e-mail products please call 561-347-7304</B></I>
<P>Method of Payment: We Accept Check by Fax  or Credit cards </B></I>
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<P>To Pay By Check</B></I>
<P>Please Print Out this Form Take a blank check write void on it  and tape it to this form and fax it back to us and enter your drivers licence number or state id here</B></I>
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<P>Fax this form to 561-347-7304</B></I>
<P>------------------------------------------CUT HERE-------------------------------------</B></I>
<P>The Sender Of this E-mail is a Member of the Internet responsible marketers (I.R.M.A) Association #12995</B></I>
<P>The Mailing List that you are being mailed from was filtered against</B></I>
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<P>Also if you wish to have your product or service promoted by Bulk mail please call 561-347-7304</B></I>
<P>DED):</B></I>
<P>________________



From rem-conf Wed Sep 09 17:33:29 1998 
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Our unique Turnkey Solution for ISO-9000 gives you everything you
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For more information, Call 561-367-9880 ext. 222



From rem-conf Wed Sep 09 19:13:03 1998 
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Date: Wed, 09 Sep 1998 22:10:10 -0400
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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Is anybody aware of implementations that stream MPEG over RTP or (better
yet) can receive such? If so, which mode (elementary, system or
transport stream)? I seem to remember Precept having such a thing, but I
can't find it at the moment.

Thanks.
-- 
Henning Schulzrinne   schulzrinne@cs.columbia.edu
Dept. of Comp. Sci.   ph  +1 212 939-7042
Columbia University   fax +1 212 666-0140
New York, NY 10027    http://www.cs.columbia.edu/~hgs



From rem-conf Wed Sep 09 23:46:42 1998 
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From rem-conf Thu Sep 10 01:58:41 1998 
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Date: Thu, 10 Sep 1998 10:53:28 +0200
From: Nicolai Leymann <leymann@berkom.de>
Organization: Deutsche Telekom Berkom GmbH
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Hi,

Precept (now Cisco) has a tool called IP/TV which supports MPEG (as well as
other formats) and RTP. More information can be found under the following
URL:

                   http://www.precept.com/datasheets/html/ipdatasheet.htm


           Nic

Henning Schulzrinne wrote:

> Is anybody aware of implementations that stream MPEG over RTP or (better
> yet) can receive such? If so, which mode (elementary, system or
> transport stream)? I seem to remember Precept having such a thing, but I
> can't find it at the moment.
>
> Thanks.
> --
> Henning Schulzrinne   schulzrinne@cs.columbia.edu
> Dept. of Comp. Sci.   ph  +1 212 939-7042
> Columbia University   fax +1 212 666-0140
> New York, NY 10027    http://www.cs.columbia.edu/~hgs
>





--
Nicolai Leymann                        email: leymann@berkom.de
Deutsch Telekom Berkom GmbH            phone: ++49-30-34973570
Goslarer Ufer 35, 10589 Berlin           fax: ++49-30-34973571





From rem-conf Thu Sep 10 06:01:42 1998 
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To: rem-conf@es.net
Subject: Slides from Chicago meeting available
Date: Thu, 10 Sep 1998 13:53:44 +0100
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From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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Copies of slides presented at the AVT meeting in Chicago are now available
>from http://www-mice.cs.ucl.ac.uk/multimedia/misc/avt/ along with draft
minutes of the meeting.

Colin



From rem-conf Thu Sep 10 06:02:00 1998 
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From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
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All RTP implementations, including libraries, are listed at
http://www.cs.columbia.edu/~hgs/rtp

If you know of anything else, let me know. A very basic RTP
implementation, without codecs, is part of the rtptools.



From rem-conf Thu Sep 10 06:22:36 1998 
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Subject: Re: MPEG/RTP
In-Reply-To: <35F73502.32DFB54B@cs.columbia.edu> from Henning Schulzrinne at "Sep 9, 98 10:10:10 pm"
To: schulzrinne@cs.columbia.edu (Henning Schulzrinne)
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> Is anybody aware of implementations that stream MPEG over RTP or (better
> yet) can receive such? If so, which mode (elementary, system or
> transport stream)? I seem to remember Precept having such a thing, but I
> can't find it at the moment.

	I don't know if you are asking for a commercial implementation
or just any one. If it's the second case, we have developed such a
solution within the SICMA ACTS project.

	The University of Paderborn has developed a multimedia
server on a parallel computer which streams MPEG-1 System streams
using RTP (RTSP for control), and a client that plays them
(written in C++ and using Active Movie for the decoding, implemented
as a Netscape Plug-in).
	Apart from them, we are aware of implementation from other
companies that work with us in other projects. The german company
"Axcent" has also developed an RTSP/RTP server for PC platforms
and the corresponding client. I don't know about the commercial
status of the product though. 
	People from Siemens ZT in Munich are also working on the
adaptation from a media server and client (Java applet, JMF based)
to work with RTP and RTSP, and as far as I know have already a working
version, but I doubt that it is an open product.
	If you have interest I could put you in touch with people in
each group.


	As far as RTP is concerned, all the implementations use
only MPEG-1 System streams, and a straightforward packetizing.



-------------------------------------------------------------------
Francisco Cortes		office: F2.315
University of Paderborn		Tel.:   +49 5251 60 6693
Dept. of Math. & Comp. Sci.	Fax.:   +49 5251 60 6697
Fuerstenallee 11		email:  cortes@uni-paderborn.de
D-33102 Paderborn, Germany
http://www.uni-paderborn.de/fachbereich/AG/monien/PERSONAL/CORTES/
===
PGP fingerprint  69 B9 77 98 D7 6A 8F CF  32 78 43 85 4E 7C E0 96
-------------------------------------------------------------------



From rem-conf Thu Sep 10 06:37:09 1998 
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Cisco has an IP/TV server. Anyway, you can try ICAST viewer
(http://www.icast.com), or NetShow from Microsoft.

RAMIRO





From rem-conf Thu Sep 10 08:38:34 1998 
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Date: Thu, 10 Sep 1998 08:33:28 -0700
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From: Chris Williams <cwilliams@bmrc.berkeley.edu>
Subject: Sept. 16 Berkeley, Multimedia, Interfaces, and Graphics
  Seminar 
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<center>

</center>

<center>ICEBERG: From POTS to PANS


Wednesday September 16, 1998 12:30-2:00 PDT 

405 Soda Hall

</center>

<center>Anthony Joseph

Computer Science Division - EECS

University of California, Berkeley 

</center>

Technological progress is yielding a convergence of network access and
backbone infrastructure: wide-area wireless networking, digital cellular
telephony, wired and wireless IP networks, and public switched telephone
networks. However, the process is far from over and many significant
design challenges remain.


The UC Berkeley's Iceberg project is exploring these challenges,
including how to merge the different design philosophies and requirements
that are associated with each of the access and backbone technologies. We
are in the process of building a large-scale testbed, based upon an IP
core network, that will incorporate current and prototype networking
technology.


Once it is deployed, our testbed will provide significant additional
functionality beyond that which is available from the individual
components, including the ability to handoff services from one network to
another (e.g., transferring a call from a cell phone to wireless IP
connected laptop). In addition, the infrastructure will offer ubiquitous
access to information, anywhere, anyplace, anytime, and using any I/O
device.


This talk will discuss the design of the Iceberg architecture and our
progress is deploying the testbed.



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



____________________________________________


Chris Williams

Berkeley Multimedia Research Center (BMRC)

http://bmrc.berkeley.edu/

626 Soda Hall #1776

University of California

Berkeley, CA 94720-1776

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

Email: cwilliams@bmrc.berkeley.edu



From rem-conf Thu Sep 10 09:15:03 1998 
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Date: Thu, 10 Sep 1998 12:12:22 -0400 (EDT)
From: Sukesh Garg <sukeshgg@caip.rutgers.edu>
To: rem-conf@es.net
Subject: Video and Java
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 Hi,
 I am working on a project that involves sending real-time data over the
n/w using Java. 
 I have built an application and found the my application to be very slow.

Is there any implementation available that is fast....
or any pointers as to increase the speed...(using java)

Thanks

Sukesh





From rem-conf Thu Sep 10 10:35:22 1998 
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References: <Pine.GSO.3.96.980910121034.1845B-100000@biko.rutgers.edu>
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Sukesh Garg wrote:
> 
>  Hi,
>  I am working on a project that involves sending real-time data over the
> n/w using Java.
>  I have built an application and found the my application to be very slow.
> 
> Is there any implementation available that is fast....
> or any pointers as to increase the speed...(using java)
> 
> Thanks
> 
> Sukesh

Make sure you are using a JIT. A lot depends on your threading model.
Checkout http://developer.java.sun.com - it usually has some good
performance tips.

-Anup.



From rem-conf Thu Sep 10 11:41:07 1998 
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Hi,

The High performance native compiler from IBM could also help you.

http://www.alphaWorks.ibm.com/

/Serge Lachapelle




From rem-conf Thu Sep 10 12:08:55 1998 
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Subject: Re: Video and Java 
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>http://www.alphaWorks.ibm.com/
jikes only speeds up compilation of java files .

	amancio



From rem-conf Thu Sep 10 12:19:10 1998 
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Amancio Hasty wrote:

> >http://www.alphaWorks.ibm.com/
> jikes only speeds up compilation of java files .
>
>         amancio
>

Not Jikes, but the High Performance Compiler.

<snip from http://www.alphaWorks.ibm.com/formula/hpc >

Compiles Java bytecode (or sourcecode) into optimized, platform-specific
native code

Platform(s):
  AIX, OS/2, Windows 95, Windows NT

High Performance Compiler for Java compiles Java bytecode into optimized
platform-specific native (object) code. The resulting code is
(generally) significantly faster than the bytecode executed in a JVM
(Java Virtual Machine)/JIT (Just In Time) environment. The degree of
performance improvement depends upon the application.

<end of snip>

/Serge Lachapelle




From rem-conf Thu Sep 10 14:53:19 1998 
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Date: Thu, 10 Sep 1998 23:47:56 +0200 (MET DST)
From: Christoph Kuhmuench <cjk@pi4.informatik.uni-mannheim.de>
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To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
cc: rem-conf@es.net
Subject: Re: MPEG/RTP
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On Wed, 9 Sep 1998, Henning Schulzrinne wrote:

> Is anybody aware of implementations that stream MPEG over RTP or (better
> yet) can receive such? If so, which mode (elementary, system or
> transport stream)? I seem to remember Precept having such a thing, but I
> can't find it at the moment.

I can offer an experimental implementation of the RTP payload for MPEG.
We implemented the payload (video only) and integrated it into vic-2.8.
Together with the MPEG-2 reference decoder version 1.2 of the MPEG
Software Simulation Group which we also integrated into vic it is possible
to stream MPEG-1 or MPEG-2 videos and display them. The name of the
MPEG file can be entered in vic's Menu window.

More info can be found at
  http://www.informatik.uni-mannheim.de/~cjk/ihl/rtp/

Regards,
   Chris

------------------------------------------------------------------------------
Christoph Kuhmuench             email:cjk@pi4.informatik.uni-mannheim.de
Praktische Informatik IV        http://www.informatik.uni-mannheim.de/~cjk
Universitaet Mannheim           Tel: ++49 621 292 1554
L15,16                          Fax: ++49 621 292 5745
68131 Mannheim, Germany
------------------------------------------------------------------------------




From rem-conf Thu Sep 10 14:56:32 1998 
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Subject: RTP payload type names
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Is it legal to use a dynamic payload type for payloads which have a static
payload ID? If it is, is there a table which gives the official RTPmap
names for the existing static types?

On a related topic, what is the RTPmap name for H.263+?

Thank you!

David Singer
Apple Computer/QuickTime 408-974-3162





From rem-conf Thu Sep 10 15:27:29 1998 
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>Is it legal to use a dynamic payload type for payloads which have a static
>payload ID? If it is, is there a table which gives the official RTPmap
>names for the existing static types?

I seem to remember that this was the consensus when we talked about it
a couple of IETFs ago.

>On a related topic, what is the RTPmap name for H.263+?

On the assumption that the H.263 payload format goes to historic,
probably "H263".  Whatever it is, this either needs to be defined in
the AV profile document itself, or in the H.263+ payload format
document, or separately registered in the MIME-types registry.  

We still are missing the formal rules for how the MIME-types registry
is used for RTP names.  Any progress on these Steve, Colin?

Cheers,
	Mark



From rem-conf Thu Sep 10 17:22:47 1998 
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To: Chris Williams <cwilliams@bmrc.berkeley.edu>, rem-conf@es.net
From: Markus Buchhorn <markus@acsys.anu.edu.au>
Subject: Re: Sept. 16 Berkeley, Multimedia, Interfaces, and Graphics 
  Seminar 
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At 08:33 10/09/98 -0700, Chris Williams wrote:

<excerpt>

<center>ICEBERG: From POTS to PANS


Wednesday September 16, 1998 12:30-2:00 PDT 

405 Soda Hall

</center></excerpt><<<<<<<<

[...]

>>>>

<excerpt><center>

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

</excerpt>

<<<<<<<<

I would be very keen to see/hear this talk, but there is currently no
organised MBONE infrastructure in Australia (although I'm working on
that) and no quality international feed (and I'm working on that too
;-)).


Would somebody (hopefully close to the source) be able to record
(rtpdump) the audio and video and make it available for me to ftp?


Thanks for any help ! Much appreciated !


Cheers,

	Markus



Markus Buchhorn,  Advanced Computational Systems CRC     | Ph: +61 2 62798810

email: markus@acsys.anu.edu.au, snail: ACSys, RSISE Bldg,|Fax: +61 2 62798602

Australian National University, Canberra 0200, Australia |Mobile: 0417 281429  



From rem-conf Fri Sep 11 05:23:27 1998 
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of possible interest, given recent examination of RTP-MPEG4 interoperablity.

regards, Don
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Subject: ANNOUNCE:  Advanced Interactive Content Initiative 
Date: Thu, 10 Sep 1998 00:08:47 -0700
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Please note the formation of the 'Advanced Interactive Content Initiative',
or 'AIC Initiative'.

Experts from several different communities feel the time and technology are
ripe for TV (set top) platforms to offer a range of basic through advanced
interactive applications using 3D as well as 2D content, in stored and
streamed fashion.

Key technologies to be used are Broadcast HTML (BHTML), VRML and MPEG-4.

The goal of the initiative is to develop syntax and techniques for
integration of BHTML and VRML/BIFS content, and for the use of MPEG-4 for
streaming of object-based and BHTML content. Content may be delivered using
MPEG-2 or IP transport protocols.  The objective is to produce a common
specification which undergoes technical review and formal approval by
multiple organizations.

A charter, requirements document, FAQ, and details are available at:

http://toocan.philabs.research.philips.com/misc/aici/

Specifications should be frozen by 20 December 1998. A validation of the
specifications in a working software implementation should be completed by
20 March 1999.

A mailing list has been established.  To join, email to:

aici-request@toocan.philabs.research.philips.com

In the message body:

subscribe aici

We look for to your participation in what we hope will be an exciting and
important endeavor.


Rob



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From: Ken Wong <kenwong@nih.gov>
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The National Institutes of Health (NIH) will be multicasting live to 
the Internet Mbone the NIH Wednesday Afternoon Lecture Series (WALS).  
The WALS features weekly lectures of top medical reseachers and 
scientists covering a variety of medical related topics.

Each lecture is scheduled for Wednesday at 3pm EST.  The first 
scheduled lecture will be on September 16th.

Please consult http://www1.od.nih.gov/wals/index.html for the list of 
scheduled speakers.  Please note schedule deviations and bye weeks.

Please tune into the session "NIH WALS Lecture."

__________________________________________________

"I believe I am the most fortunate sentient in
 this sector of the galaxy."

                                      - Data

Ken Wong
National Institutes of Health
kenwong@nih.gov




From rem-conf Fri Sep 11 12:11:52 1998 
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From: Dave Singer <singer@apple.com>
Subject: Re: RTP payload type names
Cc: rem-conf@es.net
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At 6:26 PM -0400 9/10/98, Mark Handley wrote:
>>Is it legal to use a dynamic payload type for payloads which have a static
>>payload ID? If it is, is there a table which gives the official RTPmap
>>names for the existing static types?
>
>I seem to remember that this was the consensus when we talked about it
>a couple of IETFs ago.
>
>>On a related topic, what is the RTPmap name for H.263+?
>
>On the assumption that the H.263 payload format goes to historic,
>probably "H263".  Whatever it is, this either needs to be defined in
>the AV profile document itself, or in the H.263+ payload format
>document, or separately registered in the MIME-types registry.
>

The new I-D for H.263+ is not a superset of the old RFC (2190?) and thus we
need distinct payload types and/or names for the two.

Currently the RFC has the type 34 and the name H263.  Seems to me that the
new ID had better have no static type and the name H263+. Or am I missing
something?

David Singer
Apple Computer/QuickTime 408-974-3162





From rem-conf Fri Sep 11 12:44:56 1998 
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Date: Fri, 11 Sep 1998 12:40:12 -0700 (Pacific Daylight Time)
From: Stephen Casner <casner@cisco.com>
To: Dave Singer <singer@apple.com>
cc: Mark Handley <mjh@east.isi.edu>, rem-conf@es.net
Subject: Re: RTP payload type names
In-Reply-To: <v04003a08b21f237be879@[17.255.20.102]>
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On Fri, 11 Sep 1998, Dave Singer wrote:

> The new I-D for H.263+ is not a superset of the old RFC (2190?) and thus we
> need distinct payload types and/or names for the two.

Correct.

> Currently the RFC has the type 34 and the name H263.  Seems to me that the
> new ID had better have no static type and the name H263+. Or am I missing
> something?

Yes, the new ID (and I hope soon it will get through the process to
become an RFC) is intended to have no static type assigned.  For the
name, I suspect something like H263-1998 would be appropriate.

Or more specifically, a name that references this particular payload
format specification in some way.  I say that because trouble arose
for the earlier H263 payload format when some implementations of early
I-D revisions using the static PT 34 were shipped as products and then
later the I-D was updated before publication as an RFC.  At the draft
stage, if a dynamic payload type is used and is bound to the
particular revision of the draft, this problem can be avoided.

							-- Steve




From rem-conf Fri Sep 11 13:53:09 1998 
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Date: Fri, 11 Sep 1998 14:00:39 -0700 (MST)
From: gamze <seckin@enuxsa.eas.asu.edu>
To: rem-conf@es.net
Subject: Close Caption
In-Reply-To: <35F80C21.CE44A298@cisco.com>
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Hi,

Sorry if this topic is already discussed in this list, but I need 
information on how an additional media can be added to MPEG-2 transport 
stream (if possible at all).
More specifically, how is close captioning handled with MPEG-2 streams ?

Thank you,
Gamze

----------------------------------------------------------
"Change is inevitable; Progress is optional." K. McMilan
 
Gamze Seckin

Department of Computer Science and Engineering
College of Engineering and Applied Sciences
Arizona State University Box 875406
Phone : (602) 965-1207
Fax : (602) 965-2751
E-mail : gamze@asu.edu




From rem-conf Fri Sep 11 17:13:45 1998 
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From: almeroth@cs.ucsb.edu (Kevin C. Almeroth)
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Subject: Exodus doing MBone?
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Does anyone know if exodus.net is doing MBone?  I sent them an email
about multicast support (since UCnet's is now going through them first)
but any inside info would probably get me further.

On a related note...  this change in service provider makes our MBone
tunnel pretty inefficient which is why anyone out there looking at
the 42nd IETF IMJ content is seeing reasonably poor performance.

A new tunnel endpoint will hopefully solve this problem.

-Kevin



From rem-conf Sat Sep 12 09:42:33 1998 
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To: Stephen Casner <casner@cisco.com>
cc: Dave Singer <singer@apple.com>, Mark Handley <mjh@east.isi.edu>, 
    rem-conf@es.net
Subject: Re: RTP payload type names
In-reply-to: Your message of "Fri, 11 Sep 1998 12:40:12 PDT." <Pine.WNT.3.96.980911123235.-3942131C-100000@oak.precept.com>
Date: Sat, 12 Sep 1998 17:34:59 +0100
Message-ID: <3027.905618099@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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--> Stephen Casner writes:
>On Fri, 11 Sep 1998, Dave Singer wrote:
>> Currently the RFC has the type 34 and the name H263.  Seems to me that the
>> new ID had better have no static type and the name H263+. Or am I missing
>> something?
>
>Yes, the new ID (and I hope soon it will get through the process to
>become an RFC) is intended to have no static type assigned.  For the
>name, I suspect something like H263-1998 would be appropriate.
>
>Or more specifically, a name that references this particular payload
>format specification in some way.  I say that because trouble arose
>for the earlier H263 payload format when some implementations of early
>I-D revisions using the static PT 34 were shipped as products and then
>later the I-D was updated before publication as an RFC.  At the draft
>stage, if a dynamic payload type is used and is bound to the
>particular revision of the draft, this problem can be avoided.

H263-RFCxxxx maybe?

Or could we use a version parameter when the names are registered as MIME
types? "video/h263; version=RFCxxxx"?

Colin



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From rem-conf Sun Sep 13 08:29:09 1998 
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From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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> Yung Yi wrote:
> 
> Hi.
> As you know, receiver buffering is needed to reduce jitter and
> continue natural playback at audio and video data transmission.
> Could you please give me the pointer(web site or papers) on that if
> you know?
> 
> I'll be very grateful if you do.

Search http://www.cs.columbia.edu/~hgs/netbib for "playout delay" or
"playout".



From rem-conf Mon Sep 14 04:59:10 1998 
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From: The IESG <iesg-secretary@ietf.org>
SUBJECT: Last Call: RTP Payload Format for the 1998 Version of ITU-T
	 Rec. H.263 Video (H.263+) to Proposed Standard
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The IESG has received a request from the Audio/Video Transport Working
Group to consider RTP Payload Format for the 1998 Version of ITU-T Rec.
H.263 Video (H.263+) <draft-ietf-avt-rtp-h263-video-02.txt> as a
Proposed Standard.

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send any comments to the
iesg@ietf.org or ietf@ietf.org mailing lists by September 28, 1998.

Files can be obtained via
ftp://ftp.ietf.org/internet-drafts/draft-ietf-avt-rtp-h263-video-02.txt




From rem-conf Mon Sep 14 11:55:04 1998 
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> From: "M. Reha Civanlar" <civanlar@research.att.com>
> To: <rem-conf@es.net>
> Subject: A new I-D: AT&T's Error Resilient Video Transmission Technique
> Date: Wed, 29 Jul 1998 15:14:14 -0400
>
> A new I-D: "AT&T's Error Resilient Video Transmission Technique"
> is now available at:
> http://www.ietf.org/internet-drafts/draft-civanlar-hplp-00.txt
>
> Comments and questions are encouraged.
>
I was wondering if there was any advantage to sending high priority
information (mentioned in the I-D) using the data channel. In our product,
we have been sending such high priority information using the control
channel when clients open multimedia assets. Thus, before data is
streamed through the data channel (using RTP), the client has the
necessary data to handle lost packets.

Given that DESCRIBE messages in RTSP are designed to provide media
specific information, servers could return such high priority
information in a response to a DESCRIBE message. Does SDP need to be
extended to do this? It appears that one place to return this information
in the SDP header is to use the "a=fmtp:<format><format specific parameters>
attribute.

Does this proposal fit the intended usage? Is there another way in SDP to
specify this?

Thanks.
Ram

-- 
Ram Kordale



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From rem-conf Mon Sep 14 23:14:46 1998 
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From: Stephen Casner <casner@cisco.com>
To: rem-conf@es.net
cc: "Eric C. Rosen" <ecr@qualcomm.com>
Subject: AVT WG last call and question on draft-mckay-qcelp-01.txt
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To the AVT working group:

At the AVT meeting in Chicago, I said that I would issue a last call
for comments within the working group on the RTP payload format for
PureVoice audio (draft-mckay-qcelp-01.txt), which I would like to do
by this message.  The -00 and -01 versions of that document have been
out for a while, but haven't received any comments.

In the discussion of this draft during the meeting, an issue was
raised that I would like to identify in particular.  The draft
specifies a method for encrypting the voice payload and a bit in the
payload header to indicate when the payload is encrypted.

In the meeting, we discussed why one might want sometimes to encrypt
only the RTP payload and not the headers, so I think the motivation
for providing a means to do that was established.  However, there was
a question raised whether it is appropriate to have a payload-format
specific way of indicating encryption when the RTP spec already
describes a payload-format-independent mechanism, namely to define a
(dynamic) payload type to indicate an encrypted payload format.  We
did not really resolve that question in the meeting, so I pose it now.

Enclosed below is an edited sequence of messages that were exchanged
in a side conversation that I repeat here to start the discussion.

							-- Steve

Date: Fri, 11 Sep 1998 16:44:16 +0100
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>

My opinion on the technical issue is that signalling this out-of-band (as
an SDP parameter) is more appropriate, since this is a generic solution
rather than being specific to this payload format. Of course, this requires 
us to define an appropriate SDP parameter but that should be a simple
addition (and can possibly be done in the A/V profile?).

Colin


Date: Fri, 11 Sep 1998 09:37:26 -0700
From: "Eric C. Rosen" <ecr@qualcomm.com>

Colin and Steve,

Although I agree that, in general, an out-of-band indication of
the encryption algorithm is appropriate, there are some good arguments
to consider for allowing a bit to label the media itself as encrypted.

There are some applications where distributing a formal SDP session
announcement is not practical, but RTP is still preferred for streaming
the media.  In such cases, it is useful for the endpoints to be able
to quickly determine whether the media is encrypted and take application
specific action accordingly.  Key exchange and encryption algorithm
definition would still take place a priori and out-of-band.

Certain telephonic switching services may need to treat (i.e. route)
encrypted streams differently than clear streams.  In this case, listening
for and caching session announcements to support high-speed switching of
RTP formatted media is impractical, or at least, inconvenient.

--Eric


Date: Fri, 11 Sep 1998 17:21:18 -0700 (Pacific Daylight Time)
From: Stephen Casner <casner@cisco.com>

Eric,

> There are some applications where distributing a formal SDP session
> announcement is not practical, but RTP is still preferred for streaming
> the media. ... In this case, listening
> for and caching session announcements to support high-speed switching of
> RTP formatted media is impractical, or at least, inconvenient.

When Colin mentioned signaling the use of encryption out-of-band as an
SDP parameter, SDP was only stated as an example of out-of-band
signaling.  Use of RTP does not imply use of SDP, and use of SDP does
not imply use of SAP (the announcement protocol).

> In such cases, it is useful for the endpoints to be able
> to quickly determine whether the media is encrypted and take application
> specific action accordingly.  Key exchange and encryption algorithm
> definition would still take place a priori and out-of-band.

Exactly.  Some out-of-band signaling mechanism is needed.  That
signaling also needs to establish the selection of encodings and
payload formats to be used, and in the case of dynamic payload types,
to establish the binding between payload types and payload formats.
This signaling can establish one or more dynamic payload types to
indicate the encrypted payload formats.  As I mentioned during the AVT
meeting, the RTP spec already includes a description of this
alternative method of encryption:

   As an alternative to encryption at the RTP level as described above,
   profiles may define additional payload types for encrypted encodings.
   Those encodings must specify how padding and other aspects of the
   encryption should be handled. This method allows encrypting only the
   data while leaving the headers in the clear for applications where
   that is desired. It may be particularly useful for hardware devices
   that will handle both decryption and decoding.

> Certain telephonic switching services may need to treat (i.e. route)
> encrypted streams differently than clear streams.

It certainly doesn't seem like you'd want those routing decisions to
be based on the setting of a bit in the payload-type-specific header.
If some other encoding were used, the routing system would need to
look for some other indication.  The only "advantage" I see in having
a bit in the payload header to indicate encryption is that you could
easily turn encryption on and off on a packet-by-packet basis.
However, this is probably a bad idea anyway, and is contrary to your
statement above about special routing.  Besides, if you want to have
the flexibility to turn the encryption on and off per packet, it was
our intention that you'd simply define one (dynamic) payload type to
indicate the unencrypted format and another to indicate the encrypted
format.  RTP allows the payload type to change on the fly (though
H.323 may not in all circumstances, and if not, it should be fixed).

							-- Steve


Date: Fri, 11 Sep 1998 17:40:20 -0700
From: "Eric C. Rosen" <ecr@qualcomm.com>

Steve,

Let me bring up one additional point.

The RTP payload definition for PureVoice defines extra fields
when encryption is active.  The bit tells us that the vocoder
data is encrypted and that the payload structure has changed
to allow for the possible inclusion of cryptosync and crytocheck
fields, which are not needed otherwise.

Without the 'E' bit accompanying the media, an application
receiving RTP encapsulated PureVoice must rely on out-of-band
information to determine how to parse the payload.  We see this
as a potential source for problems.

--Eric


Date: Sat, 12 Sep 1998 17:30:02 +0100
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>

I'm not sure I understand... Any system which wishes to parse the encrypted
data must know the encryption key. When it is told the key for a stream, it
will know that the stream has a different format, surely? 

Colin


Date: Sat, 12 Sep 1998 21:37:43 -0700 (PDT)
From: Eric Rosen <ecr@qualcomm.com>

But it may not know a priori that the session is encrypted.  Without the
bit, the receiver *must* rely on session announcements or something similar.
Otherwise, when packets arrive, the receiver won't know how to decode them.

Similarly, if the receiver does not know the key or is not expecting
encrypted traffic, it should still be able to detect the fact that 
encrypted traffic is arriving in order to not attempt to treat the
packets as clear traffic.

--Eric




From rem-conf Tue Sep 15 00:28:09 1998 
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Date: Tue, 15 Sep 1998 00:30:04 -0700 (PDT)
From: Stephen Casner <casner@cisco.com>
To: "Eric C. Rosen" <ecr@qualcomm.com>
cc: rem-conf@es.net
Subject:  draft-mckay-qcelp-01.txt question
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Eric,

> The RTP payload definition for PureVoice defines extra fields
> when encryption is active.  The bit tells us that the vocoder
> data is encrypted and that the payload structure has changed
> to allow for the possible inclusion of cryptosync and crytocheck
> fields, which are not needed otherwise.

If the payload structure is different, then using different payload
type values for the unencrypted and encrypted cases seems all the more
appropriate.

> Without the 'E' bit accompanying the media, an application
> receiving RTP encapsulated PureVoice must rely on out-of-band
> information to determine how to parse the payload.  We see this
> as a potential source for problems.

Indeed it would be a problem, except that the different payload type
values would provide the same indication.

> But it may not know a priori that the session is encrypted.  Without the
> bit, the receiver *must* rely on session announcements or something similar.
> Otherwise, when packets arrive, the receiver won't know how to decode them.

For the encrypted case, you need some a priori communication to supply
the key.  At the same time, the application is informed what dynamic
payload type value will be used to indicate that the data is in the
encrypted format.  As agreed at the AVT meeting in Los Angeles, a
static payload type has been assigned for the unencrypted format, but
a dynamic payload type would have worked for that case as well.

> Similarly, if the receiver does not know the key or is not expecting
> encrypted traffic, it should still be able to detect the fact that 
> encrypted traffic is arriving in order to not attempt to treat the
> packets as clear traffic.

Yes.  Of course, if the sender uses the wrong payload type value, the
receiver is likely to get mighty confused, but the same is true if you
send ADPCM data and claim that it is PCM.  For the primary RTP
encryption scheme in which the entire packet is encrypted, there's
enough information tracked by the RTP packet validation procedures to
determine whether the (decrypted) packets are valid RTP or not.
However, the information in a payload header or in the payload data
may not have enough structure to validate it by itself.

							-- Steve




From rem-conf Tue Sep 15 00:58:39 1998 
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From: Martin Fluckiger <Martin.Fluckiger@cern.ch>
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Subject: MBone broadcast of the CERN ATLAS plenaries (16-18.9.98)
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	  CERN is pleased to announce the MBONE broadcast of the

			 ATLAS Plenary Meetings
			 ----------------------

			from the CERN Auditorium


*** Times are UTC+2 ***


Wednesday 16 September 1998
---------------------------

 9:00    Introduction and Status
	 - general CERN and LHC matters
	 - MoU
	 - LHCC
	 - goals of this overview week
 9:30    The LHC Machine
	 - Short general status
	 - Critical technical issues
10:15    Status of the experimental area

10:45    --- coffee break ---

11:15    Main Issues Entering the Construction Phase
	 - master schedule
	 - milestones, and their follow-up
	 - quality controls
	 - project follow-up
12:00    Electronics Co-ordination

12:45    --- lunch break  ---

14:00    Technical Integration of the Overall Detector
15:00    Computing Status I

15:45    --- tea break    ---

16:15    Computing Status II
16:45    Trigger, DAQ and DCS Global Overview
17:15    Trigger, DAQ and DCS Status Reports

18:45    End


Thursday 17 September 1998
--------------------------

 9:00    Toroid Magnet System
	 - status of B0
	 - construction schedules
	 - test station and common cryogenics
 9:45    Muon Spectrometer Global Overview
10:15    Muon Spectrometer Status Reports

10:45    --- coffee break ---

11:15    Muon spectrometer Status Reports

12:15    --- lunch break  ---

14:00    Tile Calorimeter Global Overview
14:30    Tile Calorimeter Status Reports

15:30    --- tea break    ---

16:00    LAr cryostats and cryogenics
16:30    LAr System Global Overview
17:00    LAr System Status Reports

18:30    End


Friday 18 September 1998
------------------------

 9:00    Inner Detector Global Overview
 9:30    Inner Detector Status Reports

10:30    --- coffee break ---

11:00    Inner Detector Status Reports
11:30    Solenoid
12:00    Conclusions

12:30    End


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From rem-conf Tue Sep 15 01:46:47 1998 
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Subject: Re: Error resilience thru control (RTSP/SDP) not data channel
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>Given that DESCRIBE messages in RTSP are designed to provide media
>specific information, servers could return such high priority
>information in a response to a DESCRIBE message. Does SDP need to be
>extended to do this? It appears that one place to return this information
>in the SDP header is to use the 
>"a=fmtp:<format><format specific parameters> attribute.
>
>Does this proposal fit the intended usage? Is there another way in SDP to
>specify this?

At least it is possible to request some important media-information,
e.g. GOP update in H.263. (See "SDP syntax for H.263" draft)

Example:
a=fmtp 34 GOP-UPDATE=1,3


--
Petri



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From: "Stepping, Michael" <michael.stepping@FernUni-Hagen.de>
To: 'MPEG-4 Reflector Dmif' <dmif@nortel.ca>, 
    "'MPEG4 Franceschini, Guido'" <Guido.Franceschini@CSELT.IT>, 
    'MPEG4 Zvi Lifshitz' <zvil@csi.com>
Cc: "rem-conf@es.net" <rem-conf@es.net>
Subject: DAI Callbacks
Date: Tue, 15 Sep 1998 10:41:42 +0200
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Dear DMIFers,

is it correct, that DMIF is a symmetric delivery mechanism and the
DMIF application interface - so called DAI - is also symmetric?

If it is, have I correct detected, that there are local DAI-calls
and remote DAI-calls (so called CALLBACK) which have different
parameters?
This indicates a non symmetric DAI - interface. Is this correct?

Is it feasable to remove the DAI-CALLBACKs?

What will happen, if a remote Application Server tries to add a channel?
If I have detected correct, this direction is not implemented in
IM1-core code?


Michael Stepping

-------------------------------------------------------------------------------
Dipl.-Ing. Michael Stepping an der FernUniversitaet-Hagen
Lehrgebiet fuer Kommunikationssysteme, Herr Prof. Dr. Kaderali
Feithstrasse 142 im TGZ in 58084 Hagen
Tel. (0 23 31) 9 87 - 41 04, Fax: - 3 97
EMail: Michael.Stepping@Fernuni-hagen.de
Homepage: http://www-kommsys.fernuni-hagen.de/~stepping
-------------------------------------------------------------------------------
FTK Research Institute for Telecommunications
Image Processing & Multimedia Technology
Martin-Schmeisser-Weg 4, D-44227 Dortmund
Phone: (02 31) 97 50 56 - 36, Fax: - 10




From rem-conf Tue Sep 15 06:08:14 1998 
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From: Franceschini Guido <Guido.Franceschini@CSELT.IT>
Subject: RE: DAI Callbacks
To: 'MPEG-4 Reflector Dmif' <dmif@nortel.ca>,
 "'MPEG4 Franceschini, Guido'" <guido.franceschini@cselt.it>,
 'MPEG4 Zvi Lifshitz' <zvil@csi.com>,
 "'Stepping, Michael'" <michael.stepping@FernUni-Hagen.de>
Cc: rem-conf@es.net
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Michael,

> ----------
> From: 	Stepping,
> Michael[SMTP:michael.stepping@FernUni-Hagen.de]
> Sent: 	Tuesday, September 15, 1998 10:41 AM
> To: 	'MPEG-4 Reflector Dmif'; 'MPEG4 Franceschini, Guido'; 'MPEG4 Zvi
> Lifshitz'
> Cc: 	rem-conf@es.net
> Subject: 	DAI Callbacks
> 
> Dear DMIFers,
> 
> is it correct, that DMIF is a symmetric delivery mechanism and the
> DMIF application interface - so called DAI - is also symmetric?
quite correct

> If it is, have I correct detected, that there are local DAI-calls
> and remote DAI-calls (so called CALLBACK) which have different
> parameters?
> This indicates a non symmetric DAI - interface. Is this correct?
No.
Symmetry applies to sender/receiver functionalities, not to Delivery
Layer/Application Layer functionalities.
The terminology adopted indicates with CALLBACK the calls directed from
Delivery Layer to Application Layer

> Is it feasable to remove the DAI-CALLBACKs?
No

> What will happen, if a remote Application Server tries to add a
> channel?
> If I have detected correct, this direction is not implemented in
> IM1-core code?
Yes. IM1 is not yet a complete implementation of DMIF.

> Michael Stepping
> 
> ----------------------------------------------------------------------
> ---------
> Dipl.-Ing. Michael Stepping an der FernUniversitaet-Hagen
> Lehrgebiet fuer Kommunikationssysteme, Herr Prof. Dr. Kaderali
> Feithstrasse 142 im TGZ in 58084 Hagen
> Tel. (0 23 31) 9 87 - 41 04, Fax: - 3 97
> EMail: Michael.Stepping@Fernuni-hagen.de
> Homepage: http://www-kommsys.fernuni-hagen.de/~stepping
> ----------------------------------------------------------------------
> ---------
> FTK Research Institute for Telecommunications
> Image Processing & Multimedia Technology
> Martin-Schmeisser-Weg 4, D-44227 Dortmund
> Phone: (02 31) 97 50 56 - 36, Fax: - 10
> 
> Guido Franceschini
> 
> -------------------------------
> CSELT, Via Reiss Romoli 274 , I-10148 Torino - Italy
> Tel + 39 011 228 6137
> Fax + 39 011 228 6190
> Email guido.franceschini@cselt.it



From rem-conf Tue Sep 15 10:42:40 1998 
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From: Christoph Kuhmuench <cjk@pi4.informatik.uni-mannheim.de>
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        Martin Mauve <mauve@pi4.informatik.uni-mannheim.de>
Subject: Guidelines for RTP Payload Format specifications?
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Hi,

We are currently looking into the development of an RTP profile for
"interactive media".  This profile specifies a foundation for payloads
like animations in VRML or Java simulations.  In preparation we read the
guidelines for writers of RTP payload format specifications
(draft-ietf-avt-rtp-format-guidelines-00).
The Guideline identifies the following two requirements:

 o A payload format should be devised so that in the presence of
   loss, the payload can still be decoded.

 o Ideally all the contents of every packet should be possible to
   be decoded and played out irrespective of whether preceding
   packets have been lost or arrive late.

We believe that these two requirements restrict the usage of RTP to audio
and video streams.  However for "interactive media" we need the
functionalities of RTP (e.g. time stamps, sender/ receiver reports,
SSRC...).  Unfortunatly the nature of interactive media does not permit
the complete fulfilment of both requirements.

Consider an example application were a VRML world is shared between a
number of users.  In this scenario the initial state of the world must be
transmitted to all participants and following user interaction (events)
must be broadcasted.  Since events cannot be decoded without having
received the world and/or previous events this scenario violates both
requirements.  It would be very convenient to transmit the inital world
and user interaction (events) via reliable multicast beneath RTP.

This leads to the following two quetstions:

Is it acceptable to specify an RTP payload that does not completely fulfil
both requirements?

Is it acceptable to specify an RTP payload that requires reliable
multicast beneath RTP?

Christoph Kuhmuench, Volker Hilt, Martin Mauve




From rem-conf Tue Sep 15 11:11:42 1998 
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        Martin Mauve <mauve@pi4.informatik.uni-mannheim.de>
Subject: Re: Guidelines for RTP Payload Format specifications? 
In-reply-to: Your message of "Tue, 15 Sep 1998 19:37:05 +0200."
             <Pine.OSF.3.95.980915183901.3275A-100000@pi4> 
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>The Guideline identifies the following two requirements:
>
> o A payload format should be devised so that in the presence of
>   loss, the payload can still be decoded.
>
> o Ideally all the contents of every packet should be possible to
>   be decoded and played out irrespective of whether preceding
>   packets have been lost or arrive late.
...
>This leads to the following two quetstions:
>
>Is it acceptable to specify an RTP payload that does not completely fulfil
>both requirements?
>
>Is it acceptable to specify an RTP payload that requires reliable
>multicast beneath RTP?

Firstly I should say that the draft in question attempts to provide
guidelines but they're not absolute rules.


In the case of VRML, it seems you have one protocol for disseminating
the world model, and that needs to be reliable.

Then you have another protocol for disseminating moving objects
(events).  This doesn't strictly need to be reliable - if a packet is
lost and a subsequent packet contains information that supercedes the
first packet, then you don't need reliability as part of the transport
protocol.  An example is a moving bullet in a wargame, where it's
position is sent repeatedly.  You don't need every position because if
you miss one packet you'll get the new position soon enough, and any
retransmitted old positions will arrive too late for playout anyway.

The first (world dissemination) protocol doesn't seem to have any
benefit from using RTP, whereas the second (real-time update) protocol
seems like a good use for RTP.

Now the complication comes if you have a reliable-multicast-like
protocol for ensuring resynchronization after packet losses when the
final packet(s) got lost (like where that bullet finally ended up).
Such a protocol has more in common with the world-dissemination
protocol than the real-time update protocol.  It's also not strictly
reliable, because again later events may render earlier information
obsolete before reliable transmission to everyone has succeeded.

So back to your second question:

>Is it acceptable to specify an RTP payload that requires reliable
>multicast beneath RTP?

It's not clear to me what you mean by "beneath" RTP.  It seems what
you might have is an RM protocol that runs _over_ RTP.  Peter Parnes did
some work on SRM over RTP a few years back - you might look at his
work.

Another possibility is to define a new RTP profile for semi-reliable
delivery that defines RTCP packet types for NACK, etc.  It seems this
might also be usable by a number of people for streaming audio and
video from servers, where there is sufficient playout delay for
retransmission to take place.

Hope this helps,
	
Mark



From rem-conf Tue Sep 15 13:24:51 1998 
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To: rem-conf@es.net
Subject: ORWH Women's Health Seminar on Eating Disorders
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This program will be multicasted live onto the Internet Mbone on Wed, 
Sept 16, from noon-2pm EST.  Please tune into the session "NIH ORWH 
Seminar."

Apologies for the late program addition and notice.

>Approved-By: nih-staff-moderators@LIST.NIH.GOV
>Date:         Tue, 8 Sep 1998 10:21:19 -0400
>Reply-To: "ORWH HOME PAGE (OD)" <ORWHPUBS@OD1TM1.OD.NIH.GOV>
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>From: "ORWH HOME PAGE (OD)" <ORWHPUBS@OD1TM1.OD.NIH.GOV>
>Subject:      ORWH Women's Health Seminar on Eating Disorders
>To: NIH-STAFF@LIST.NIH.GOV
>
>The NIH Office of Research on Women's Health
>Women's Health Seminar Series
>
>EATING DISORDERS:  FADS AND FACTS
>
>September 16, 1998
>
>Noon - 2 p.m.
>
>Masur Auditorium, Building 10
>
>
>Overview
>Harold Goldstein, Ph.D., NIMH
>
>The Neurobiology of Eating Disorders
>Sarah Leibowitz, Ph.D., Rockefeller University
>
>The Psychology of Risk Factors:
>A Developmental Approach to What Causes Eating Disorders
>Ruth H. Striegel-Moore, Ph.D., Wesleyan University
>
>Overview of Effective Treatments for Eating Disorders
>W. Stewart Agras, M.D., Stanford University
>
>Discussion and Questions
>
>Presentations are open to the general public.  Reservations are not required.
>Sign language interpretation will be provided.
>
>
>


__________________________________________________

"I believe I am the most fortunate sentient in
 this sector of the galaxy."

                                      - Data

Ken Wong
National Institutes of Health
kenwong@nih.gov




From rem-conf Tue Sep 15 16:37:18 1998 
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From: "Ram Kordale" <kordale@relic.engr.sgi.com>
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Date: Tue, 15 Sep 1998 16:32:17 -0700
In-Reply-To: Koskelainen Petri <petkos@cs.tut.fi>
        "Re: Error resilience thru control (RTSP/SDP) not data channel" (Sep 15, 11:43am)
References: <199809150843.LAA13514@harakka.cs.tut.fi>
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On Sep 15, 11:43am, Koskelainen Petri wrote:
> Subject: Re: Error resilience thru control (RTSP/SDP) not data channel
>
> >Given that DESCRIBE messages in RTSP are designed to provide media
> >specific information, servers could return such high priority
> >information in a response to a DESCRIBE message. Does SDP need to be
> >extended to do this? It appears that one place to return this information
> >in the SDP header is to use the
> >"a=fmtp:<format><format specific parameters> attribute.
> >
> >Does this proposal fit the intended usage? Is there another way in SDP to
> >specify this?
>
> At least it is possible to request some important media-information,
> e.g. GOP update in H.263. (See "SDP syntax for H.263" draft)
>
                                ^^^^^^^^^^^^^^^^^^^^^^^^^ Where is such
description available. Is this a separate draft? Is there one such draft for
MPEG-1 and MPEG-2 format data?

Thanks.
Ram


-- 
Ram Kordale



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From: Koskelainen Petri <petkos@cs.tut.fi>
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Subject: Re: RTP payload type names
To: rem-conf@es.net
Date: Wed, 16 Sep 1998 10:04:50 +0300 (EET DST)
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>Is it legal to use a dynamic payload type for payloads which have a static
>payload ID? If it is, is there a table which gives the official RTPmap
>names for the existing static types?
>
>On a related topic, what is the RTPmap name for H.263+?
>

Another related point, it makes no sense to announce only H.263(+)
without any knowledge about the codec options or parameters.
In draft 
http://search.ietf.org/internet-drafts/draft-koskelainen-sdp263-02.txt
we have suggested such format for H.263+ SDP syntax using a=fmtp attribute.

It seems that static payload type 34 must be changed to dynamic one.
So, is the correct form now as follows ?

m=video 8888 RTP/AVP 96
a=rtpmap:96 H263-1998
a=fmtp 96 CIF=4 QCIF=2/MaxBR=1000/E F

(assuming H263-1998 is the RTPmap name)


Thanks,
Petri



From rem-conf Wed Sep 16 00:11:53 1998 
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Date: Wed, 16 Sep 1998 00:04:13 -0700 (PDT)
From: Stephen Casner <casner@cisco.com>
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To: Mark Handley <mjh@east.isi.edu>
cc: Christoph Kuhmuench <cjk@pi4.informatik.uni-mannheim.de>, rem-conf@es.net,
        Volker Hilt <hilt@pi4.informatik.uni-mannheim.de>,
        Martin Mauve <mauve@pi4.informatik.uni-mannheim.de>
Subject: Re: Guidelines for RTP Payload Format specifications? 
In-Reply-To: <6061.905882880@north.lcs.mit.edu>
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On Tue, 15 Sep 1998, Mark Handley wrote:
> It's not clear to me what you mean by "beneath" RTP.  It seems what
> you might have is an RM protocol that runs _over_ RTP.  Peter Parnes did
> some work on SRM over RTP a few years back - you might look at his
> work.
> 
> Another possibility is to define a new RTP profile for semi-reliable
> delivery that defines RTCP packet types for NACK, etc.  It seems this
> might also be usable by a number of people for streaming audio and
> video from servers, where there is sufficient playout delay for
> retransmission to take place.

For some time, I've thought that adding "variable reliability" to RTP
would be a useful enhancement for several applications.  We have not
undertaken that in AVT because of concerns that reliable multicast
protocols should not be undertaken in IETF until the IRTF RM Research
Group has made some progress on this topic.
							-- Steve




From rem-conf Wed Sep 16 00:11:53 1998 
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To: Ram Kordale <kordale@relic.engr.sgi.com>
cc: rem-conf@es.net
Subject: Re: Error resilience thru control (RTSP/SDP) not data channel?
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On Mon, 14 Sep 1998, Ram Kordale wrote:

> Given that DESCRIBE messages in RTSP are designed to provide media
> specific information, servers could return such high priority
> information in a response to a DESCRIBE message. Does SDP need to be
> extended to do this? It appears that one place to return this information
> in the SDP header is to use the "a=fmtp:<format><format specific parameters>
> attribute.

I believe this fits within the intended usage of fmtp.  In the
discussions of generic payload formats at AVT meetings in DC and LA,
we've talked about putting similar information there.  The format has
not been nailed down, though.
							-- Steve




From rem-conf Wed Sep 16 00:37:35 1998 
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From rem-conf Wed Sep 16 08:47:17 1998 
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To: Christoph Kuhmuench <cjk@pi4.informatik.uni-mannheim.de>
cc: rem-conf@es.net, mjh@east.isi.edu, 
    Volker Hilt <hilt@pi4.informatik.uni-mannheim.de>, 
    Martin Mauve <mauve@pi4.informatik.uni-mannheim.de>
Subject: Re: Guidelines for RTP Payload Format specifications?
In-reply-to: Your message of "Tue, 15 Sep 1998 19:37:05 +0200." <Pine.OSF.3.95.980915183901.3275A-100000@pi4>
Date: Wed, 16 Sep 1998 16:39:17 +0100
Message-ID: <1717.905960357@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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--> Christoph Kuhmuench writes:
>We are currently looking into the development of an RTP profile for
>"interactive media".  This profile specifies a foundation for payloads
>like animations in VRML or Java simulations.  In preparation we read the
>guidelines for writers of RTP payload format specifications
>(draft-ietf-avt-rtp-format-guidelines-00).
[...]

This seems like a good opportunity to remind people that we're soliciting
feedback from the AVT community to ensure that the "guidelines for writers
of RTP payload format specifications" draft accurately reflects real-world
experience.

I'd encourage everyone to read and comment on this document...

Colin



From rem-conf Wed Sep 16 13:53:12 1998 
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From: subbiah@NASBPD01BS.ntc.nokia.com (Subbiah Baranitharan NRC/Bos)
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Organization: Nokia Telecommunications
Date: Wed, 16 Sep 1998 16:33:09 -0400
Subject: Multiplexing in RTP payload
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Hello all,


This is regarding the user multiplexing in RTP payload, which was discussed=
 =
during the last Chicago IETF meeting. To understand this problem more, I =
thought the following message could be a starting point.

Based on the four proposals presented at the Chicago meeting, it seems to m=
e =
that there are two different problems need to be solved.

1. Transporting non RTP sources between gateways through multiplexing ( =
Lucent and Nokia proposals).

2. Transporting RTP sources through multiplexing (Hitachi and ISI/USC)

Is the above assumption acceptable within the working group?


Barani Subbiah
Nokia Research Center




From rem-conf Wed Sep 16 14:32:15 1998 
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Sender: "Vahe Balabanian" <balabani@nortel.ca>
To: subbiah@NASBPD01BS.ntc.nokia.com
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From: "Vahe Balabanian" <balabani@nortel.ca>
Subject: re:Multiplexing in RTP payload
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In message "Multiplexing in RTP payload", subbiah@NASBPD01BS.ntc.nokia.com writes:

>
>Hello all,
>
>
>This is regarding the user multiplexing in RTP payload, which was discussed=
> =
>during the last Chicago IETF meeting. To understand this problem more, I =
>thought the following message could be a starting point.
>
>Based on the four proposals presented at the Chicago meeting, it seems to =
>=
>me that there are two different problems need to be solved.
>
>1. Transporting non RTP sources between gateways through multiplexing =
> (Lucent and Nokia proposals).
>
>2. Transporting RTP sources through multiplexing (Hitachi and ISI/USC)
>
>Is the above assumption acceptable within the working group?
>
My reading was that the 2nd solution was going to be effectively
encompassing the first. Only one solution would be required for 
interoperability reasons.

Vahe
Nortel



From rem-conf Wed Sep 16 14:32:15 1998 
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From: Mark Handley <mjh@east.isi.edu>
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To: subbiah@NASBPD01BS.ntc.nokia.com (Subbiah Baranitharan NRC/Bos)
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Subject: Re: Multiplexing in RTP payload 
In-reply-to: Your message of "Wed, 16 Sep 1998 16:33:09 EDT."
             <1998Sep16.163000.1991.326207@rhino.ntc.nokia.com> 
Date: Wed, 16 Sep 1998 17:24:07 -0400
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>Based on the four proposals presented at the Chicago meeting, it seems to me 
>that there are two different problems need to be solved.
>
>1. Transporting non RTP sources between gateways through multiplexing (
>Lucent and Nokia proposals).
>
>2. Transporting RTP sources through multiplexing (Hitachi and ISI/USC)
>
>Is the above assumption acceptable within the working group?

In the short term, you might view them as different.  In the longer
term though, they're the same problem.  Some of the sources in
1. might eventually become RTP sources, or some of the sinks might
become RTP sinks.  Also a type-1 gateway might end up talking to a
type-2 gateway.

So I view them as really different aspects of the same problem.

Cheers,
	Mark



From rem-conf Thu Sep 17 00:29:15 1998 
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Subject: Collecting Audio & Video packets ?
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------=_NextPart_000_004C_01BDE257.7BE24360--




From rem-conf Thu Sep 17 02:09:23 1998 
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Date:	Thu, 17 Sep 1998 11:01:30 +0200 (MESZ)
From:	Daniel Koehler <koehlerd@informatik.tu-muenchen.de>
To:	rem-conf@es.net
Subject: Anybody tried vic with SGI Octane and Personal Video?
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Hello,

hope i got the right list :-)

Did anyone already try wether vic 2.8 works with SGI Octane and Personal
Video option ???

Thanks,

Daniel





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Date: Thu, 17 Sep 1998 09:22:56 -0400
From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
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To: Subbiah Baranitharan NRC/Bos <subbiah@NASBPD01BS.ntc.nokia.com>
CC: "'rem-conf'" <rem-conf@es.net>
Subject: Re: Multiplexing in RTP payload
References: <1998Sep16.163000.1991.326207@rhino.ntc.nokia.com>
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Subbiah Baranitharan NRC/Bos wrote:
> 
> Hello all,
> 
> This is regarding the user multiplexing in RTP payload, which was discussed during the last Chicago IETF meeting. To understand this problem more, I thought the following message could be a starting point.
> 
> Based on the four proposals presented at the Chicago meeting, it seems to me that there are two different problems need to be solved.
> 
> 1. Transporting non RTP sources between gateways through multiplexing ( Lucent and Nokia proposals).
> 
> 2. Transporting RTP sources through multiplexing (Hitachi and ISI/USC)
> 
> Is the above assumption acceptable within the working group?

I think there was consensus that there were these two problems, but not
really consensus that there should be one or two solutions. Mark's
proposal was to do something that solves both, though, not just #2.

-Jonathan R.

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



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From: subbiah@NASBPD01BS.ntc.nokia.com (Subbiah Baranitharan NRC/Bos)
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 ------------
From: Vahe Balabanian
To: Subbiah Baranitharan (NRC/Bos)
Cc: rem-conf
Subject: re:Multiplexing in RTP payload
Date: Wednesday, September 16, 1998 5:19PM



In message "Multiplexing in RTP payload", subbiah@NASBPD01BS.ntc.nokia.com
writes:

>
>Hello all,
>
>
>This is regarding the user multiplexing in RTP payload, which was =
discussed=3D
> =3D
>during the last Chicago IETF meeting. To understand this problem more, I =
=3D
>thought the following message could be a starting point.
>
>Based on the four proposals presented at the Chicago meeting, it seems to =
=3D
>=3D
>me that there are two different problems need to be solved.
>
>1. Transporting non RTP sources between gateways through multiplexing =3D
> (Lucent and Nokia proposals).
>
>2. Transporting RTP sources through multiplexing (Hitachi and ISI/USC)
>
>Is the above assumption acceptable within the working group?
>
My reading was that the 2nd solution was going to be effectively
encompassing the first. Only one solution would be required for
interoperability reasons.

Vahe
Nortel


 ----------------------------------------
Thanks for the comments.

For various reasons I think the second solution is not suitable for the =
first one. To start with:

1. Why we need to convert a non RTP source speech packet into a RTP packet =
=
and then multiplex when the source and destination does not use the header =
info.

2. In my view the bottleneck is in the access links. If a RTP source is abl=
e =
send a packet without any problem in the access link then I don't see why we=
 =
need to multiplex them. I rather use the RTP/UDP/IP header compression from =
=
the source so that there is no complex multiplexing to save bandwidth =
between gateways.

3. The efficiency proposed in the second method is equal or better only whe=
n =
gateway co-operates. But preliminary studies have indicated that this is not=
 =
the case. We need to use atleast 5 bytes when there is multiple codecs =
between multiplexing end points. I am referring to Mark Handly's proposal ( =
=
CID=3D8, SN, PT  are needed and rest can be compressed).

4. In a mobile environment there are multiple codecs and as the trend =
continues, majority of the calls will be between mobile users. I am not sure=
 =
if it is advantages to use RTP over the air interface for speech only calls.=
 =
Even if the remote is a RTP source these things can be taken care at some =
gateway.

5. There is a need for intermediate switching (multiplexing and =
demultiplexing) so that overhead can be kept low on the entire transport. If=
 =
compression is applied then it adds further processing overhead and delays.

I understand the need for a single solution. However, my proposal would be

1. For RTP sources  --- use RTP/UDP/IP header compression from the source

2. For non RTP sources - Multiplex between gateways as much as possible.

Your comments are welcome.

Best regards,

Barani Subbiah




From rem-conf Thu Sep 17 08:59:23 1998 
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Subject: 9/23 Berkeley, Multimedia, Interfaces, and Graphics   Seminar 
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<center>Electronic Notebooks for Scientific Collaborative Work 

</center><center>Sonia Sachs 

Lawrence Berkeley Laboratories 


Wednesday September 23, 1998

 12:30-2:00 PDT 

405 Soda Hall

</center>


Paper notebooks are regularly used by scientists, engineers, and
technicians to keep records of their observations,

analysis and conclusions. As researchers remotely collaborate in
performing experiments, controlling instruments of

on-line facilities, designing large-scale systems, or in a variety of
laboratory and scientific discovery processes, they

face the management of large amounts of data, personal paper notebooks,
electronic mail, phone conferencing

discussions, and collections of documents and multimedia files. In
addition, collaborating researchers may use

collected data to produce images and plots that are shared with
colleagues in discussions of obtained results. 


Electronic notebooks are used to capture, store, and manage a variety of
data obtained through data acquisition and

data analysis processes. In collaborative work, electronic notebooks
offer remote collaborators the capability of

sharing data and of creating annotations on the shared data, both
synchronously and asynchronously. Electronic

notebooks are also important in collaborative work because they are used
to manage the persistent historical record

of group discussions, meetings, decisions, and conclusions reached by
project team members. Moreover, electronic

notebooks enhance science and engineering inquiry processes by providing
powerful searching capabilities of

diverse database information related to a collaborative project. 


In this talk we will summarize the main ideas regarding electronic
notebooks for collaborative work, review related

work, and then focus on the development of an experimental electronic
notebook for the Spectro-Microscopy

Collaboratory at the Advanced Light Source (ALS) in LBNL. We will discuss
the architecture of the electronic

notebook system, its current implementation, lessons learned, and future
directions. 


The slides for this seminar are available for viewing at

http://www-itg.lbl.gov/~ssachs/TALKS.dir/UCB.dir/ppframe.htm. 


The MBone addresses

are video 224.2.163.7/57076 and audio 224.2.147.61/27202. 

____________________________________________


Chris Williams

Berkeley Multimedia Research Center (BMRC)

http://bmrc.berkeley.edu/

626 Soda Hall #1776

University of California

Berkeley, CA 94720-1776

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

Email: cwilliams@bmrc.berkeley.edu



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OPENSIG '98 Workshop

Open Signaling for ATM, Internet and Mobile Networks

October 5-6 1998, University of Toronto, Toronto, Ontario    

http://www.comet.columbia.edu/opensig/activities/opensig98.html

------------------------------------------
P R O G R A M    A T    A     G L A N C E
------------------------------------------
KEYNOTE on Geoplex

PANEL on Industry's View of Open Signaling 

SIX SESSIONS:
 
-Views of Differentiated Services 
-Enabling Virtual Networking 
-Active Network Interfaces 
-Programmable Network Management 
-OPEN Session 
-Mobile Agents and Open Signaling 

----------------------------------------------------
F U L L   P R O G R A M    A T    A     G L A N C E
----------------------------------------------------

Program: First Day, October 5th, 1998

Welcoming Remarks 
Irene Katzela, University of Toronto

Session 1 - Enabling Virtual Networking 
Organizer and Chair: Andrew T. Campbell, Columbia University 
       
     1. Paul Ferguson, Cisco, 
        What is a VPN ? 

     2. John Vincente, Intel and Columbia University, 
        Towards Programmable Virtual Networking 

     3. Ken Calvert, University of Kentucky, 
	Composing Services in CANES 
      
     4. Jens-Peter Redlich, NEC, 
	How Virtual Networks can support QoS-sensitive Internet Applications 

     5. Anindo Banarjea, ISI, 
	The XBONE: Building Overlay Networks
                
Session 2 - Active Network Interfaces 
Organizer and Chair:  Ken Calvert, University of Kentucky

     1. Bob Braden, ISI ,  
	Active Reservation Protocol 

     2. Dan Nessett, 3Com, 
	Experience with Java on a Router 

     3. Andy Bavier, Princeton University, 
	The Scout/Jout Active Network Node


Session 3 - Programmable Network Management 
Organizer and Chair: Gisli Hjalmtysson, AT&T Research

     1. Pawan Goyal, AT&T - Research,  
	PRONTO: A Platform for Programmable Networking 

     2. Yuval Shavitt, Lucent Bell Labs, 
	An Active Network Approach to  Efficient Network Management 

     3. Dan Ionescu, University of Ottawa, 
	A Corba Environment for Network Management

Session 4 - OPEN Session 
Chair: Nikos Anerousis, AT&T Labs

     Want a 15 minute slots to voice an opinion? Then contact Nikos  
     for on-demand reservation at nikos@research.att.com 
       
     1. Andrew T. Campbell, Columbia University, 
        "Open" Wireless Plaforms: Why Bother? 

     2. Jit Biswas, Kentridge Labs, Secretary, IEEE P1520 Working Group, 
	IEEE Programming Interfaces for Networks (to be confirmed)
       
     3. Radhakrishna Pillai, Kentridge Labs, 
	Mobile Binding Interface Base

OPENSIG Reception  -  Faculty House 

Program: Second Day, October 6th, 1998 

Keynote Address 
Session Chair: Aurel A. Lazar, Columbia University and Xbind Inc.
      
     Nelu Mihai, AT&T West Labs, 
     On Geoplex

Session 5 - Views of Differentiated Services 
Organizer and Chair: Kathleen Nichols, Cisco
*to be on mbone*

     1. John Wroclawski, MIT, Diff Services: 
	Magic Bullet or Cure for Cancer ?

     2. Yoram Bennet, Microsoft, 
	No title (to be confirmed) 

     3. Fran Reichmeyer, Bay Networks, 
	Two-Tier Resource Management 

     4. Scott Bradner, Harvard University, 
	No Title Yet (to be confirmed) 

     5. Kathleen Nichols, Cisco, 
	Using Differentiated Services 

Session 6 - Industry's View of Open Signaling 
Organizer and Chair:  Al. Leon-Garcia, University of Toronto 

     Panelists: SBC, ATT, Nortel, Newbridge

Session 7 - Mobile Agents and Open Signaling 
Organizer and Chair: Roger Imprey, NRC 

     1. Hermann de Meer, U. of Hamburg and Columbia Univeristy,
        Tunnel Agents for QoS Enhancement in the Internet 

     Other Speakers to be confirmed

17:30   Closing Remarks and OPENSIG'99 Announcement 

For further information please send email to opensig@comm.utoronto.edu 

-------------------------------------------------
Organizing Committee: 
-------------------------------------------------
     Irene Katzela, Chair, (University of Toronto)
     Anindo Banerjea (ISI)
     Andrew Campbell (Columbia University) 
     Kitty Chan, Local Arrangements, (University of Toronto)
     Raymond Liao, Webmaster, (Columbia University) 
-------------------------------------------------
Program Committee: 
-------------------------------------------------
     Anindo Banerjea (ISI), 
     Andrew T. Campbell (Columbia University) 
     Simon Crosby (University of Cambridge and CPlane Inc.) 
     Irene Katzela (University of Toronto) 
     Aurel A. Lazar (Columbia University) and Xbind Inc.) 
     Ian Leslie (University of Cambridge and CPlane Inc.) 
-------------------------------------------------



--
Andrew T. Campbell  
http://comet.columbia.edu/~campbell



From rem-conf Fri Sep 18 06:38:07 1998 
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help



From rem-conf Fri Sep 18 09:57:18 1998 
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To: Koskelainen Petri <petkos@cs.tut.fi>
cc: rem-conf@es.net
Subject: Re: RTP payload type names
In-reply-to: Your message of "Wed, 16 Sep 1998 10:04:50 +0300." <199809160704.KAA23847@harakka.cs.tut.fi>
Date: Fri, 18 Sep 1998 17:41:39 +0100
Message-ID: <11105.906136899@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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--> Koskelainen Petri writes:
>Another related point, it makes no sense to announce only H.263(+)
>without any knowledge about the codec options or parameters.
>In draft 
>http://search.ietf.org/internet-drafts/draft-koskelainen-sdp263-02.txt
>we have suggested such format for H.263+ SDP syntax using a=fmtp attribute.
>
>It seems that static payload type 34 must be changed to dynamic one.
>So, is the correct form now as follows ?
>
>m=video 8888 RTP/AVP 96
>a=rtpmap:96 H263-1998
>a=fmtp 96 CIF=4 QCIF=2/MaxBR=1000/E F
>
>(assuming H263-1998 is the RTPmap name)

I don't know enough about H.263 to comment on the parameters you're passing
on the a=fmtp line, but the basic idea is correct for H.263+ streams

If using H.263 with the old packetisation scheme, you would announce 
	m=video 8888 RTP/AVP 34
	a=fmtp 34 ...
to give parameters for the codec. No dynamic payload type is needed in this
case.

Colin



From rem-conf Fri Sep 18 10:37:46 1998 
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References: "Your message of Wed, 16 Sep 1998 10:04:50 +0300."
 <199809160704.KAA23847@harakka.cs.tut.fi>
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Date: Fri, 18 Sep 1998 10:18:20 -0700
To: rem-conf@es.net
From: Dave Singer <singer@apple.com>
Subject: Re: RTP payload type names
Cc: Koskelainen Petri <petkos@cs.tut.fi>
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>--> Koskelainen Petri writes:
>>Another related point, it makes no sense to announce only H.263(+)
>>without any knowledge about the codec options or parameters.
>>In draft
>>http://search.ietf.org/internet-drafts/draft-koskelainen-sdp263-02.txt
>>we have suggested such format for H.263+ SDP syntax using a=fmtp attribute.
>>
>>It seems that static payload type 34 must be changed to dynamic one.
>>So, is the correct form now as follows ?
>>
>>m=video 8888 RTP/AVP 96
>>a=rtpmap:96 H263-1998
>>a=fmtp 96 CIF=4 QCIF=2/MaxBR=1000/E F
>>
>>(assuming H263-1998 is the RTPmap name)
>


I believe the H263 and H263+ bitstreams are self-describing. Why is it
thought necessary to give receivers more format information in the sdp
info? Or is it merely desirable (I'd still like to know why)?

David Singer
Apple Computer/QuickTime 408-974-3162





From rem-conf Fri Sep 18 10:57:22 1998 
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Subject: Re: RTP payload type names
In-Reply-To: <v04003a0db22845b52be6@[17.255.20.102]> from Dave Singer at "Sep 18, 98 10:18:20 am"
To: singer@apple.com (Dave Singer)
Date: Fri, 18 Sep 1998 20:54:53 +0300 (EET DST)
Cc: rem-conf@es.net, petkos@cs.tut.fi
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> >>Another related point, it makes no sense to announce only H.263(+)
> >>without any knowledge about the codec options or parameters.
> 
> I believe the H263 and H263+ bitstreams are self-describing. Why is it
> thought necessary to give receivers more format information in the sdp
> info? Or is it merely desirable (I'd still like to know why)?

Because H.263 standard includes optional features which are not supported
by the basic implementations.
 
Some of the decoders support only very basic features while some are 
advanced decoders which are capable of decoding optional features (like 
arbitrary picture sizes, arithmetic coding, scalability etc).

These basic (simple) decoders are unable to decode advanced H.263 features.
Therefore it makes no sense to send (even standard compliant) H.263 stream 
which includes non-decodeable options.
  
H.263 stream which includes advanced options are usually useless
to receiver. So, the capability set must be agreed somehow beforehand.

This kind of scheme is useful (necessity) at least in SIP.
 
--
Petri Koskelainen
Nokia Research Center



From rem-conf Fri Sep 18 11:22:25 1998 
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To: mbone@isi.edu, rem-conf@es.net
Subject: Version of vic to support new PCI-based SunVideo card?
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Dear Colleagues,

I'm in the midst of upgrading my UltraSPARC 2 to a 60, which is a
PCI-based machine.  The version of vic that I have  (2.8, which appears
to be the latest) does not appear to recognize this new board.  Is
there a version in the offing for the near future that will support
the SunVideo 2 card?

Thanks and Best Regards, Rick Rodgers



From rem-conf Fri Sep 18 11:22:37 1998 
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From: subbiah@NASBPD01BS.ntc.nokia.com (Subbiah Baranitharan NRC/Bos)
To: balabani@nortel.ca (Vahe Balabanian), rem-conf@es.net ('rem-conf')
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Organization: Nokia Telecommunications
Date: Fri, 18 Sep 1998 14:16:22 -0400
Subject: re:Multiplexing in RTP payload
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Hi,

Re-distributing this mail with reply due to some local mail problem. Please=
 =
ignore this if you have received earlier.

Barani Subbiah
 ------------
From: Vahe Balabanian
To: Subbiah Baranitharan (NRC/Bos)
Cc: rem-conf
Subject: re:Multiplexing in RTP payload
Date: Wednesday, September 16, 1998 5:19PM



In message "Multiplexing in RTP payload", subbiah@NASBPD01BS.ntc.nokia.com
writes:

>
>Hello all,
>
>
>This is regarding the user multiplexing in RTP payload, which was =
discussed=3D
> =3D
>during the last Chicago IETF meeting. To understand this problem more, I =
=3D
>thought the following message could be a starting point.
>
>Based on the four proposals presented at the Chicago meeting, it seems to =
=3D
>=3D
>me that there are two different problems need to be solved.
>
>1. Transporting non RTP sources between gateways through multiplexing =3D
> (Lucent and Nokia proposals).
>
>2. Transporting RTP sources through multiplexing (Hitachi and ISI/USC)
>
>Is the above assumption acceptable within the working group?
>
>>My reading was that the 2nd solution was going to be effectively
>>encompassing the first. Only one solution would be required for
>>interoperability reasons.

>>Vahe
>>Nortel


 ----------------------------------------
Thanks for the comments.

For various reasons I think the second solution is not suitable for the =
first
one. To start with:

1. Why we need to convert a non RTP source speech packet into a RTP packet =
=
and
then multiplex when the source and destination does not use the header info=
.

2. In my view the bottleneck is in the access links. If a RTP source is abl=
e
send
a packet without any problem in the access link then I don't see why we nee=
d
to
multiplex them. I rather use the RTP/UDP/IP header compression from the =
source
so that there is no complex multiplexing to save bandwidth between gateways=
.

3. The efficiency proposed in the second method is equal or better only whe=
n
gateway
co-operates. But preliminary studies have indicated that this is not the =
case.
We need to use atleast 5 bytes when there is multiple codecs between
multiplexing end points. I am referring to Mark Handly's proposal ( CID=3D8=
, =
SN,
PT  are needed and rest can be compressed).

4. In a mobile environment there are multiple codecs and as the trend
continues,
majority of the calls will be between mobile users. I am not sure if it is
advantages to use RTP over the air interface for speech only calls. Even if=

the
remote is a RTP source these things can be taken care at some gateway.

5. There is a need for intermediate switching (multiplexing and
demultiplexing)
so that overhead can be kept low on the entire transport. If compression is=

applied then it adds further processing overhead and delays.

I understand the need for a single solution. However, my proposal would be

1. For RTP sources  --- use RTP/UDP/IP header compression from the source

2. For non RTP sources - Multiplex between gateways as much as possible.

Your comments are welcome.

Best regards,

Barani Subbiah






From rem-conf Sun Sep 20 15:21:22 1998 
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From: 71288840@24841.com
Date: Mon, 31 Aug 98 15:10:34 EST
To: niomi@hisbe.com
Subject: Market Timing
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MT is delivered to our subscribers via an automated list 
service; unsubscribing instructions are at the end of this
letter.


September 20, 1998


Table of Contents - Market Timing - Sept 20 edition


1. MT Interviews NuOncology Management
2. Market Commentary 
3. MT in Europe
4. Departments


1. MT Interviews NuOncology Management

NuOncology Labs Inc. Symbol: NLAB - OTC Bulletin Board

We apologize for not getting this letter out a few days ago 
but there is a lot going on with this company - for instance, 
only Tuesday of this week MT received a request to be kept updated 
about NuOncology's progress, from an Asian based pharmaceutical 
concern.

MT met with the Senior Management of NuOncology Labs Inc. 
on September 10, 1998, at the offices of their investment 
relations firm in Washington, D.C.

MT met with Mr. Robert S. Thomas, Chairman and President, Mr. 
Philip F. Enlow, Executive Vice President and Mr. Joe Kane, a 
financial consultant to NuOncology Labs Inc.

As you can appreciate in a meeting such as this, there are 
legal and regulatory restrictions on what information can be 
shared with MT and our readers, but from the information we 
did receive there is no doubt this company is making significant 
progress.

We received confirmation that the doctors at the Baker-Sanger 
Laboratory in Houston, Texas are in the final phases of testing 
and validating Arglabin and determining its method of action. 
We also learned that one of the original doctors from Kazakstan 
is in Houston and assisting in this process.

The completion of these tests will be followed by the publishing 
of a paper, in a recognized medical journal, discussing the 
findings of NuOncology's research team. This paper will be co-authored 
by at least four physicians with extensive experience and 
credentials in oncological research, including Dr. R. Michael 
Williams, NuOncology's Medical Director and Chief Scientific Officer.

Our impression was that this paper would be completed very soon 
and if the demeanor of the people we met with is any indication, 
the results are what NuOncology and their shareholders want to see - 
our feeling is that Arglabin is an FTI.

SN will immediately publish a synopsis of the paper when it is 
available.

We also gained an appreciation of how far ahead of other 
biopharmaceutical companies NuOncology is at the present time.

To the best of MT's knowledge no one has published information 
on clinical (human) trials of FTI's. NuOncology Labs Inc, on the other
hand, has all of the clinical results from the Institute of Phytochemistry, in Kazakstan, for the treatment of over 200 patients with various types of cancers. 

In addition, on a visit to take place between October 19-29, 
1998, further results will be available from clinical trials 
involving 40 liver cancer and 60 breast cancer patients. These 
tests have been conducted so as to conform to the international 
protocols for medical research and should help to add credence 
to the effectiveness of Arglabin. For this reason alone, NuOncology 
Labs Inc could be 1-2 years ahead of their many industry competitors 
who are trying to develop  FTI's. (There are at least 8 major 
pharmaceutical companies conducting research in this area.)

We were also pleased to confirm that the patents for the lead 
compound, Arglabin, and the broader patents for its related and 
derivative compounds are held in the United States and controlled 
by NuOncology Labs Inc.

One of the key questions to our readers is "What is all this 
worth?" The best answer MT could come up with was by making 
an analogy to another cancer treatment, Taxol,  that seems to 
work in a similar way to Arglabin.

Taxol is a $1 billion a year drug today. It has one serious 
problem in that it is highly toxic. Arglabin, conversely, has 
minimal toxic side effects and in clinical trials appears to 
work as well and in some cases better than Taxol, depending on 
the type of cancer being treated. 

Assuming that the clinical trials continue to confirm Arglabin's 
efficacy as a cancer treatment, one way of looking at the 
valuation of Arglabin would be as a less toxic Taxol. This could
easily lead to Arglabin becoming the treatment of choice and 
thereby giving it a value at least equal to Taxol.

In our next letter we will discuss a number of other developments
underway in the predictive and chemosensitivity testing areas of
NuOncology's business. MT wishes to thank the management of 
NuOncology Labs Inc. for taking the time for this meeting and 
providing ourselves and our readers with these insights into 
their company and their business. 

MT continues to feel very bullish about this company and encourage 
our readers to closely follow this rapidly developing situation. 
We believe the shares of NuOncology Labs Inc. (NLAB) to be very 
attractive at prices up to $10.



2. Market Commentary

If the market rally can be credited to the strong hints 
of an interest rate cut late last week and Monday, and 
the slide in the market in the middle of last week can be 
blamed on Mr. Clinton's personal problems then this market 
is not in bad shape and is likely to move higher.

We believe that Mr. Greenspan's guidance of the economy is 
key to how the U.S. economic future will unfold. In fact, the 
future of a number of large Asian and South American economies 
are dependent on his guidance. 

The "doom and gloom" crowd project the present set of trends 
and parameters into the future and get a self fulfilling 
prophecy of economic disaster. This is where we differ from 
those that are predicting a more negative outlook - we do not 
assume the monetary authorities of the major economic powers 
are going to do nothing.

MT believes that the position that is now being taken by the 
Fed in the U.S. is that America will have to try and ward off a 
global recession by initially lowering interest rates and 
thereby encouraging domestic growth. They will also lend 
enough money, through the IMF, to reflate many of the world's 
troubled economies. 

This scenario is only possible because the US economy is strong,
unemployment is low, interest rates are at historic lows (and 
probably heading lower), there is no apparent inflationary 
pressures, and there is no external threat to the U.S..

Mr. Clinton is less than 2 months away from what most pundits 
will soon be referring to as the "lame duck" period of his 
Presidency. Although we do not think it likely that he will 
be impeached, if it does occur, we do not think it will be 
a factor in the economic equation.

The bottom line is that the U.S. will need to be the engine to 
keep many of the world's economies afloat and resuming the type 
of growth they have shown in the past. 

For this reason MT believes that the future for the U.S. economy 
and stock market is positive and there will continue to be 
opportunities to profit for the savvy investor.


3. MT in Europe

MT is expanding its readership in certain 'key' economic centers 
in Europe.  For the convenience of our readers, MT will now be 
available in German. For those subscribers who would like to receive 
their issues of MT in German simply type "German" in the subject
and send an e-mail to :  

<a href="mailto:Wetpapiere@USA.net">Wetpapiere@USA.net</a>

There is no need to write a letter as you 
will automatically be sent future issues of MT in German. 
 

4. Departments


A. Stock Quotes
B. MT Privacy Policy
C. Disclaimer
D. How to Unsubscribe


A. Stock Quotes

To get a stock quote please go to:  
 
<a href="http://www.StockGuide.com">http://www.StockGuide.com</a>
 

B. MT Privacy Policy

At MT we respect your privacy and will NEVER release your e-mail 
address to a third party for any reason.


C. Disclaimer

Please read our disclaimer at:

http://206.132.163.167/krichx/mtdis.htm
 

D. How to Unsubscribe 
 
 
To unsubscribe to MT simply put 'unsubscribe' in the subject and return to:
 
<a href="UnsubscribeMT@USA.net">UnsubscribeMT@USA.net</a>
 
You will be unsubscribed instantly.




From rem-conf Mon Sep 21 01:22:04 1998 
From rem-conf-request@es.net Mon Sep 21 01:22:01 1998
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From: "RZ Dietmar Rahlfs" <RAHLFS@hugo.rz.fh-ulm.de>
Organization:  Fachhochschule Ulm
To: rem-conf@es.net
Date:          Mon, 21 Sep 1998 10:06:49 +0100
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Subject:       VIC 2.7 under Linux: Create Session Problem
X-Confirm-Reading-To: "RZ Dietmar Rahlfs" <RAHLFS@hugo.rz.fh-ulm.de>
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Hello,

I am trying to make a video session under Linux with
vic 2.7a38 and the Matrox Meteor. I can receive
sessions from the M-Bone, and when I am doing the
same under Solaris (SunOS 5.5) I am successful.

Here's what I've done:

sdr &
  New / Create advertised session

Now the session is in the sdr list.

I join my new session, vic starts up, I choose Menu, click
on "Transmit", and a uniformly green image appears where
in other sessions I can see a professor saying something or
another scene.

The Linux kernel version is 2.0.33.

Some more info from VIC:

After clicking on Menu/Transmission:
    Rate Control: about 8 f/s, 5 kb/s

Menu/Encoder:
    Device: Matrox Meteor 1             (which is /dev/mmetfgrab[0])
    Port:   RCA                         (correct: "phone connector" in
                                         the Matrox Meteor manual)
    Options: -
    h261
    normal
    
Menu/Display:
    Options: -
    Tile:    single
    Ordered

info...
    RTP Statistics: all zero but
        Kilobytes   5000
        Frames      6600
        Packets     6600
        (at some moment)
    Decoder Stats: all zero.
    
I can distribute this uniform green image to another Linux PC,
and that PC reacts on Protocol change (h261->nv and so on).

Is Linux with the meteor well supported at all nowadays?    

Thank you for answering

Dietmar Rahlfs



From rem-conf Mon Sep 21 02:02:56 1998 
From rem-conf-request@es.net Mon Sep 21 02:02:46 1998
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From: David PATE <pate@dpt-info.u-strasbg.fr>
Organization: ULP - Strasbourg
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CC: rem-conf@es.net
Subject: Re: VIC 2.7 under Linux: Create Session Problem
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RZ Dietmar Rahlfs wrote:

> Hello,
>
> I am trying to make a video session under Linux with
> vic 2.7a38 and the Matrox Meteor. I can receive
> sessions from the M-Bone, and when I am doing the
> same under Solaris (SunOS 5.5) I am successful.
>
> Here's what I've done:
>
> sdr &
>   New / Create advertised session
>
> Now the session is in the sdr list.
>
> I join my new session, vic starts up, I choose Menu, click
> on "Transmit", and a uniformly green image appears where
> in other sessions I can see a professor saying something or
> another scene.
>
> The Linux kernel version is 2.0.33.
>
> Some more info from VIC:
>
> After clicking on Menu/Transmission:
>     Rate Control: about 8 f/s, 5 kb/s
>
> Menu/Encoder:
>     Device: Matrox Meteor 1             (which is /dev/mmetfgrab[0])
>     Port:   RCA                         (correct: "phone connector" in
>                                          the Matrox Meteor manual)
>     Options: -
>     h261
>     normal
>
> Menu/Display:
>     Options: -
>     Tile:    single
>     Ordered
>
> info...
>     RTP Statistics: all zero but
>         Kilobytes   5000
>         Frames      6600
>         Packets     6600
>         (at some moment)
>     Decoder Stats: all zero.
>
> I can distribute this uniform green image to another Linux PC,
> and that PC reacts on Protocol change (h261->nv and so on).
>
> Is Linux with the meteor well supported at all nowadays?
>

I don't know why you've got a green image, but I just can say that the
meteor is well supported. Perhaps, you have to switch to vic 2.8 that is
the last "official" release.


--

 DAVID
 -----

----------------------------------------------------------------------
David PATE'  ==>> http://dpt-info.u-strasbg.fr/~pate





From rem-conf Mon Sep 21 15:21:40 1998 
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From: Ken Wong <kenwong@nih.gov>
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Subject: NIH Public Participation
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The NIH will be multicasting this event.  Please join the session "NIH 
Public Paricipation" on Wed, Sept 23 from 9am-3:30pm EST.

--- Begin Forwarded Message ---

From:	Vazquez, Laura (OD) 
Sent:	Monday, September 21, 1998 4:42 PM
To:	Wong, Ken (CIT)
Subject:	multicasting Sept. 23 meeting

On Wednesday, September 23, Dr. Harold Varmus, NIH Director, will hold a special
meeting on enhancing public participation in NIH activities.  This meeting is
open to the public and will run from 9 a.m. until 3:30 p.m. on the NIH campus.
At the meeting, individual public participants invited by the NIH will discuss
future activities and responsibilities of the proposed NIH Director's Council of
Public Representatives and the NIH Offices of Public Liaison, initiatives
recommended in the National Academy of Sciences' Institute of Medicine (IOM)
report, "Scientific Opportunities and Public Needs: Improving Priority Setting
and Public Input at the National Institutes of Health," released July 8, 1998.

--- End Forwarded Message ---


__________________________________________________

"I believe I am the most fortunate sentient in
 this sector of the galaxy."

                                      - Data

Ken Wong
National Institutes of Health
kenwong@nih.gov




From rem-conf Tue Sep 22 05:56:51 1998 
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From: Anoop Kulkarni <anoop@sasi.com>
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Hi,
Are there any standardising bodies working on fields such as 'TCP over
wireless media', 'mobile IP on wireless media'? If so, which are they?
Am sorry if this is not the right place to ask these questions.

Thanks in advance,
Regards,
Anoop
--
----------------------------+---------------------------------------------
Home : Anoop Kulkarni, PhD  | Office: Anoop Kulkarni, Silicon Automation
Flat A2, Pioneer Residency, | Systems (I) Pvt Ltd., 3008, 12th B Main, 8th
HAL III Stage, Bangalore 75 | Cross, Indiranagar, Bangalore - 560008 INDIA
Phone: +091-080-528 5749    | Fax   : +091-080-528 4396
Email: anoop@sasi.com       | Phone : +091-080-528 1461 ext 4205
----------------------------+---------------------------------------------



From rem-conf Tue Sep 22 07:48:25 1998 
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From: Forrest  Blair <fblair@s-vision.com>
To: potsag@imtc.org, h323implementors@imtc.org, sg16.lbc@research.kpn.com, 
	rem-conf@es.net
Subject: H.263 Picture Type Bits
Date: Tue, 22 Sep 1998 08:43:37 -0600
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Please excuse the multiple copies for those on multiple lists.

I'm looking for more detailed information about H.263 picture type bits 3
and 4.  Bit 3 is defined as "Split screen indicator" and bit 4 as "Document
camera indicator".

The standard does not clearly define the purpose of these bits and how they
should be handled by the decoder if they are turned on.  Any information you
have would be most helpful.

Also, I'm curious to know if any encoders are actively using these features.

Thanks in advance.

Forrest Blair
Sorenson Vision, Inc.



From rem-conf Tue Sep 22 07:51:02 1998 
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Date:	Tue, 22 Sep 1998 16:52:01 +0200
From:	Rafal Maszkowski <rzm@icm.edu.pl>
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September 24th
18 GMT (20 MET)
RAT GSM+LPC

Warsaw Autumn is a modern music festival - more:
http://www.warsaw-autumn.art.pl/		- the festival
http://www.warsaw-autumn.art.pl/WHO-HOW/	- Internet concert

Sorry for annoucing very late, we have a lot of technical problems with MBONE
and MBONE tools. Whe may have big packet losses too. Is the secondary RAT
encoding a problem for anybody who would like to listen?

R.



From rem-conf Tue Sep 22 08:19:26 1998 
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From: spike@8x8.COM (Daniel Helman)
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To: potsag@imtc.org, h323implementors@imtc.org, sg16.lbc@research.kpn.com,
        rem-conf@es.net, fblair@s-vision.com
Subject: Re: H.263 Picture Type Bits
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The document camera bit is just that; it is on when the image is coming from
a second camera pointed at a document.  We use a 1->0 transition on this bit
to indicate that the previous image should be stored for later viewing.  If
you have this storage capability your user interface can switch between live
video and stored documents.

We don't use the split screen bit.

Dan Helman
spike@8x8.com

> Date: Tue, 22 Sep 1998 08:43:37 -0600
> From: Forrest Blair <fblair@s-vision.com>
> 
> I'm looking for more detailed information about H.263 picture type bits 3
> and 4.  Bit 3 is defined as "Split screen indicator" and bit 4 as "Document
> camera indicator".
> 
> The standard does not clearly define the purpose of these bits and how they
> should be handled by the decoder if they are turned on.  Any information you
> have would be most helpful.
> 
> Also, I'm curious to know if any encoders are actively using these features.
> 
> Thanks in advance.
> 
> Forrest Blair
> Sorenson Vision, Inc.
> 



From rem-conf Tue Sep 22 10:48:50 1998 
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Date: 22 Sep 1998 13:37 EDT
Sender: "Vahe Balabanian" <balabani@nortel.ca>
To: C.Perkins@cs.ucl.ac.uk
Cc: rem-conf@es.net
From: "Vahe Balabanian" <balabani@nortel.ca>
Subject: re:Draft AVT minutes
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Colin, I know I missed the deadline you specified, but I think 
one word is missing:

In message "Draft AVT minutes", C.Perkins@cs.ucl.ac.uk writes:

>I enclose a draft of the minutes for the Chicago AVT meeting. In order to
>meet the submission deadline, I'd be grateful if you could send comments
>or corrections to me by Thursday evening.
>
>Colin
>
>
>===============================================================================
>Minutes of the Audio/Video Transport Working Group
>
...............
>The next IETF meeting coincides with the MPEG4 decision meeting, so it is
>therefore urgent that these issues are resolved. The last chance to make
>changes to MPEG4 version 1 will be at the MPEG meeting on 12-16 October.
>Since it is clear that only limited progress can be made in the short time
>period available it was decided to continue with the development of the
>payload format for MPEG4 elementary streams and to resolve the issues
                   ^single              
>discussed. Once this is done and further experience has been gained with
>actual implementations the group will revisit these issues.
>
....


Vahe



From rem-conf Wed Sep 23 07:10:49 1998 
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To: Vahe Balabanian <balabani@nortel.ca>
cc: rem-conf@es.net
Subject: Re: Draft AVT minutes
In-reply-to: Your message of "22 Sep 1998 13:37:00 EDT."
Date: Wed, 23 Sep 1998 14:51:44 +0100
Message-ID: <6359.906558704@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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Vahe Balabanian writes:
>Colin, I know I missed the deadline you specified, but I think 
>one word is missing:
...
>>Minutes of the Audio/Video Transport Working Group
>...............
>>The next IETF meeting coincides with the MPEG4 decision meeting, so it is
>>therefore urgent that these issues are resolved. The last chance to make
>>changes to MPEG4 version 1 will be at the MPEG meeting on 12-16 October.
>>Since it is clear that only limited progress can be made in the short time
>>period available it was decided to continue with the development of the
>>payload format for MPEG4 elementary streams and to resolve the issues
>                   ^single              
>>discussed. Once this is done and further experience has been gained with
>>actual implementations the group will revisit these issues.

Vahe, 

That's a useful clarification. Once we understand how to efficiently carry
a single MPEG4 elementary stream in RTP and have some experience with the
typical usage of MPEG4, the group should work on the issues of multiplexing
and multiple streams. We should solve the problem for the simple case first
though.

Colin



From rem-conf Thu Sep 24 09:59:15 1998 
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From: Chris Williams <cwilliams@bmrc.berkeley.edu>
Subject: 9/30 Berkeley, Multimedia, Interfaces, and Graphics   Seminar 
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<center>Use What You Teach: Teaching CSCW Using CSCW Tools 


</center><center>James Landay 

Computer Science Division - EECS 

UC Berkeley 


Wednesday September 30, 1998 

12:30-2:00 PDT 405 Soda Hall

</center>

Distance learning technology has repeatedly been cited as a way to reduce
the cost of advanced education and to bring training to the masses.
Drawbacks aside, this technology will be used in many educational
settings and it is therefore important for educators to understand its
advantages and limitations. Researchers in the field of
computer-supported cooperative work (CSCW) have been building research
systems for years that support distance learning. To expose students to
the difficult issues in this area and to learn more myself, I taught a
course on CSCW in the Fall 1997 semester in which we used CSCW tools. In
this talk I describe the course structure, the technology we used, and
the outcomes of the course. 


The course website is at:
http://bmrc.berkeley.edu/courseware/cscw/fall97/index.html. 


The Mbone addresses are video 224.2.163.7/57076 and audio
224.2.147.61/27202. 




____________________________________________


Chris Williams

Berkeley Multimedia Research Center (BMRC)

http://bmrc.berkeley.edu/

626 Soda Hall #1776

University of California

Berkeley, CA 94720-1776

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

Email: cwilliams@bmrc.berkeley.edu



From rem-conf Fri Sep 25 07:17:09 1998 
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From: vishwas <vishwas@future.futsoft.com>
To: "'Bill Fenner'" <fenner@parc.xerox.com>
Cc: rem-conf <rem-conf@es.net>, mbone <mbone@ISI.EDU>
Subject: Setsockopt
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Hi Bill,
 Have a small problem in my multimedia apllication.
When I do setsockopt for an IP addr which is a POINTOPOINT link running   
SLIP running on Linux 1.2.1. I get an error message saying "No Such   
Device".. Is setsock opt is supported for serial interfaces..?

Regards
vishwa



From rem-conf Fri Sep 25 10:01:27 1998 
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Reply-to: mbone@ISI.EDU
Subject: Re: Setsockopt 
In-reply-to: Your message of "Fri, 25 Sep 1998 19:24:00 PDT."
             <360C505C@msgate.future.futsoft.com> 
Date: Fri, 25 Sep 1998 09:45:46 PDT
From: Bill Fenner <fenner@parc.xerox.com>
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I assume you mean the IP_ADD_MEMBERSHIP or IP_MULTICAST_IF socket options.
In order to support "unnumbered" interfaces, where the point-to-point
interface has the same address as another interface, some OS's use the
remote address of the interface.  Others use the local address so there
is no way to specify an "unnumbered" interface.

You could use the route to 224/4 (or, in fact, any group) to specify
the default interface for binding or sending multicast.  If you point
the route towards an interface and then use INADDR_ANY for the interface
specification, you should get the same results.

  Bill



From rem-conf Fri Sep 25 21:05:03 1998 
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To: <rem-conf@es.net>
From: <Bretheren_Coop@powercom.org>
Subject: Brethren CoOp
Date: Fri, 25 Sep 1998 19:14:47
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Brethren Long Distance CoOp

Hi, Dino Layton here...

You can now help missionaries with every long distance call you 
make.  With the strength of the Brethren CoOp, we can now offer 
you, your family, and your business a rate of 8.9 cents per 
minute on all interstate calls (24 hours / 7days), unbeatable 
in-state rates, and unheard of international rates.  AND, 4% of 
all your phone bills will go towards helping missionaries obtain 
discounted goods and services!  This service includes:

No sign-up fees
No monthly minimums
No service charges
6 second increments
Calling cards at 17 cents per minute (No other charges)
Toll-free numbers at 8.9 cents per minute ($2.00 monthly fee for 
this only)
Reimbursement for switching your long distance service

If you would like to save money on your phone bill while helping 
missionaries, reply to Brethren_coop@powercom.org with your name 
and postal address.  We will e-mail you complete information and 
send you applications for service.

Please feel free to call us toll-free if you wish.  Our number is 
(888) 545-4746.

Sincerely,
Dino J. Layton

Dino J. Layton
2900 Emily
Nixa, MO.  65714
Toll-Free (888) 545-4746
Phone: (417) 581-9292
Fax: (417) 581-9393

 
 
 
 
 
 
 
 
 



From rem-conf Sat Sep 26 07:04:43 1998 
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Date: Sat, 26 Sep 1998 15:56:45 +0200
From: Roberto Zicari <zicari@ltt.de>
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Organization: Logon Technology Transfer
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To: "KOBRYN, Cris" <ckobryn@shl.com>,
        "'commsoft@cc.bellcore.com'" <commsoft@cc.bellcore.com>,
        "'odp@dstc.edu.au'" <odp@dstc.edu.au>,
        "'dbworld@cs.wisc.edu'" <dbworld@cs.wisc.edu>,
        "'enternet@BBN.COM'" <enternet@BBN.COM>,
        "'JavaCORBA@luke.org'" <JavaCORBA@luke.org>,
        "'ISWORLD@Danann.hea.ie'" <ISWORLD@Danann.hea.ie>,
        "'comsoc-chapters@IEEE.ORG'" <comsoc-chapters@IEEE.ORG>,
        "'comsoc-gicb@IEEE.ORG'" <comsoc-gicb@IEEE.ORG>,
        "'comsoc.tac@tab.ieee.org'" <comsoc.tac@tab.ieee.org>,
        "'comswtc@gmu.edu'" <comswtc@gmu.edu>,
        "'ctc-members@redbank.tinac.com'" <ctc-members@redbank.tinac.com>,
        "'end2end-interest@isi.edu'" <end2end-interest@isi.edu>,
        "'f-troup@codex.cis.upenn.edu'" <f-troup@codex.cis.upenn.edu>,
        "'fokus-user@fokus.gmd.de'" <fokus-user@fokus.gmd.de>,
        "'g-troup@ccrc.wustl.edu'" <g-troup@ccrc.wustl.edu>,
        "'giga@tele.pitt.edu'" <giga@tele.pitt.edu>,
        "'hipparch@sophia.inria.fr'" <hipparch@sophia.inria.fr>,
        "'ieee_rtc_list@cs.tamu.edu'" <ieee_rtc_list@cs.tamu.edu>,
        "'ieeetcpc@ccvm.sunysb.edu'" <ieeetcpc@ccvm.sunysb.edu>,
        "'isadsoc@fokus.gmd.de'" <isadsoc@fokus.gmd.de>,
        "'itc@fokus.gmd.de'" <itc@fokus.gmd.de>,
        "'modern-heuristics@uk.ac.mailbase'" <modern-heuristics@uk.ac.mailbase>,
        "'multicomm@cc.bellcore.com'" <multicomm@cc.bellcore.com>,
        "'osimcast@BBN.COM'" <osimcast@BBN.COM>,
        "'rem-conf@es.net'" <rem-conf@es.net>,
        "'reres@laas.fr'" <reres@laas.fr>,
        "'sb.all@IEEE.ORG'" <sb.all@IEEE.ORG>,
        "'sc6wg4@ntd.comsat.com'" <sc6wg4@ntd.comsat.com>,
        "'sigmedia@bellcore.com'" <sigmedia@bellcore.com>,
        "'tccc@IEEE.ORG'" <tccc@IEEE.ORG>,
        "'xtp-relay@cs.concordia.ca'" <xtp-relay@cs.concordia.ca>
Subject: last call
References: <c=US%a=MCI%p=SHL%l=DALFW03-980506020821Z-9862@dalmsdoc01.shl.com>
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OBJECT WORLD London`98  -- LAST CALL

Dear Colleague,

The OBJECT WORLD London`98 conference, sponsored by the Object Management
Group (OMG), will take place Oct.5-8, 1998 at Earls Court in London.

The full conference program is available at http://www.comdex.de/OWUK/  
with on-line registration.

** Bona fide educational institutions can claim a 50% discount on the
conference prices.

Please help posting this info as you see proper.

If you wish to receive a printed copy of the conference brochure, please be so
kind and tell us your postal address.

E-mail: Logon@ltt.de, or fax +49-6173- 94 04 20

Thank you for your cooperation!

Regards

Roberto Zicari
PC Chair
OMG Central & Northern Europe



KOBRYN, Cris wrote:
> 
> [Please forward this message to colleagues who might be interested.
> We apologize if you receive multiple copies.]
> 
> The following is the final Call for Papers for the 2nd International
> Enterprise Distributed Object Computing Workshop (EDOC '98).
> 
> A HTML version of the CFP can be found at:
>     http://uml.systemhouse.mci.com/events/edoc98/home.htm
> A PDF version of the CFP can be found at:
>     http://uml.systemhouse.mci.com/events/edoc98/docs/edoc98-cfp.pdf
> 
> Invited speakers for the workshop include:
> 
>     Ivar Jacobson, VP of Business Engineering, Rational Software Corp.
>     Phil Bernstein, Repository Architect, Microsoft Corp.
> 
> ======================================================================
> 
> ***** CALL FOR PAPERS *****
> 
> 2nd International Enterprise Distributed Object Computing Workshop (EDOC
> '98)
> 
> November 3-5, 1998
> 
> Hyatt Regency La Jolla, San Diego, California, USA
> 
> http://uml.systemhouse.mci.com/events/edoc98/home.htm
> 
> ======================================================================
> 
> Hosted by:
>     MCI Systemhouse Corporation
> 
> Technical Co-Sponsors:
>     Object Management Group (OMG)
>     IEEE Communications Society, Enterprise Networking Technical
> Committee
> 
> Endorsed by:
>     Institute of Electronics, Information and Communication Engineers of
> Japan
>     Information Processing Society of Japan (IPSJ)
> 
> Corporate sponsors:
>     Distributed Systems Technology Centre (DSTC)
>     Unisys Corp.
>     Fujitsu Corp.
> 
> ======================================================================
> 
> SCOPE:
> ~~~~~~
> The software industry is in the midst of a major shift in technology
> from
> static, monolithic systems to the era of distributed objects and
> components.
> The remarkable level of interest in the first Enterprise Distributed
> Object
> Computing (EDOC '97) Workshop demonstrated that this transformation is
> well
> underway. However, there remain some major challenges to be overcome if
> distributed objects and components are to fulfill their full potential
> for
> solving enterprise problems.
> 
> The goal of the EDOC workshops is to provide a leading forum for the
> exchange of ideas and results in this rapidly developing field of
> computing,
> and to establish a new community of researchers and industry experts to
> address the enterprise issues associated with distributed object
> technology.
> Building on the success of the first EDOC workshop, EDOC '98 will focus
> on
> securing high-quality papers and tutorials from around the world, and
> will
> provide ample opportunities for discussion and debate. In addition,
> there
> will be relevant panel sessions with substantial time for open
> discussions.
> 
> EDOC '98 will target two interrelated categories of topics. The first
> group
> will deal with the specification of enterprise models for distributed
> systems. This includes issues related to the specification of roles,
> policies, business objects and their relationships in open distributed
> systems. The second group will address the problems of implementing
> systems
> specified by enterprise models using currently available distributed
> object
> technologies (e.g., CORBA, DCOM, Java-RMI, TINA, RM-ODP). This includes
> issues related to architectural models, technology bridges and the
> general
> problem of middleware selection and application.
> 
> TOPICS:
> ~~~~~~~
> The program committee seeks research results, experience reports and
> case
> studies related to all aspects of enterprise distributed object
> computing,
> including such topics as:
> 
> - Distributed object architectures and frameworks
> - Distributed business objects
> - Analysis and design methods
> - Metamodeling architectures
> - Collaborative and workflow systems
> - ODP enterprise language and business modeling
> - UML for enterprise applications
> - Enterprise policy, management and security
> - Enterprise Quality of Service
> - Traders, brokers, and agents for enterprise solutions
> - Application of the above technologies in vertical application
>   domains such as telecomunnications, business and financial systems
> 
> Papers must be of high quality and will be evaluated for originality,
> significance, insight, clarity and their relevance to the enterprise
> aspects
> of distributed object computing.
> 
> IMPORTANT DATES:
> ~~~~~~~~~~~~~~~~
> June 1st, 1998: Papers due
> August 10, 1998: Notification of acceptance
> September 11, 1998: Final papers due
> November 3-5, 1998: Workshop
> 
> INFORMATION FOR AUTHORS:
> ~~~~~~~~~~~~~~~~~~~~~~~~
> PAPERS:
> Authors are invited to submit full papers according to the submission
> guidelines below. Papers will be refereed by an international committee
> of experts. Presented papers will be published by the IEEE
> Computer Society Press.
> 
> SUBMISSIONS:
> The papers should not exceed 12 pages, according to IEEE format
> (double column, single space, 10 point Times font). The preferred method
> for submission is electronic mail (Word, RTF, PostScript or PDF format).
> Please send your submissions to the following address:
>     edoc98-submissions@omg.org
> 
> Enquiries should be directed to:
>     edoc98-info@omg.org
> 
> WORKSHOP ORGANIZERS:
> ~~~~~~~~~~~~~~~~~~~~
> General Chair:
>     Cris Kobryn (MCI Systemhouse, USA)
> Program Chairs:
>      Colin Atkinson (IESE, Germany)
>      Zoran Milosevic (DSTC, Australia)
> Program Committee:
>      Nigel Baker (University of West England, UK)
>      Phil Bernstein (Microsoft, USA)
>      Eng Chew (Optus, Australia)
>      Jim Coplien (Bell Labs, USA)
>      Fred Cummins (EDS, USA)
>      Takeo Hamada (Fujitsu Laboratories of America, USA)
>      Sridar Iyengar (Unisys, USA)
>      Peter Linington (University of Kent at Canterbury, UK)
>      Claudia Linnhoff-Popien (RWTH Aachen, Germany)
>      George Karetsos (National Technical University of Athens, Greece)
>      Subrata Mazumdar (Bell Labs, USA)
>      Osamu Miyagishi (NTT, Japan)
>      Luke O'Connor (IBM Zurich, Switzerland)
>      Maria Orlowska (The University of Queensland, Australia)
>      Jong-Tae Park (Kyungpook National University, Korea)
>      Thomas Preuss (Brandenburg University of Technology, Germany)
>      Kerry Raymond (DSTC, Australia)
>      Roberto Saracco (CSELT, Italy)
>      Marc-Thomas Schmidt (IBM, USA)
>      Morris Sloman (Imperial College, UK)
>      Richard Mark Soley (OMG, USA)
>      Jean-Bernard Stefani (CNET France Telecom, France)
>      Keith Swenson (Netscape, USA)
>      Sandy Tyndale-Biscoe (Birchquest Ltd, UK)
>      Kevin Tyson (Enterprise Engineering, USA)
>      Sanya Uehara  (Fujitsu, Japan)
>      Karl-Heinz Weiss (Public Administration Berlin, Germany)
>      Edward White (MCI, USA)
>      Bryan Wood (Open-IT, UK)
>      Douglas Zuckerman (Bellcore, USA)
> 
> Local Organizing Committee:
>      Cheryl Bissonnette (OMG, USA)
> 
> VENUE:
> ~~~~~~
> EDOC '98 will be held in San Diego, California at one of Southern
> California's premier venues: the Hyatt Regency La Jolla at Aventine. The
> hotel is a San Diego landmark destination and a Mobil 4-Star and AAA
> 4-Diamond Award winner. It provides the perfect complement of business
> and
> recreation facilities.
> 
> The Aventine complex includes the Hyatt Regency La Jolla, a 32,000 sq.
> ft.
> world-class Sporting Club and Spa, a restaurant village with
> internationally
> themed restaurants, lighted tennis courts and a heated pool. The hotel
> is
> five minutes from golf and the beach and 15 minutes from downtown San
> Diego.
> 
> ENQUIRIES AND REGISTRATION DETAILS:
> ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
> Accommodation at a discounted conference rate is available at the
> workshop
> venue. Registration and accommodation details will be available in the
> near
> future.
> For enquiries, please email to Cheryl Bissonette at:
>     cheryl@omg.org
> Postal address:
>     Object Management Group
>     492 Old Connecticut Path
>     Framingham, MA 01701
>     USA
> 
> RELATED EVENTS:
> ~~~~~~~~~~~~~~~
> November 9-13, 1998     OMG Technical Committee meeting, Burlingame, CA,
> USA
> 
> WWW PAGE:
> ~~~~~~~~~
> Further details about the EDOC'98 event and this Call For Papers can be
> found at:
>     http://uml.systemhouse.mci.com/events/edoc98/home.htm



From rem-conf Sun Sep 27 04:43:20 1998 
From rem-conf-request@es.net Sun Sep 27 04:43:19 1998
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Message-ID: <360E20B8.1426@ew.mimos.my>
Date: Sun, 27 Sep 1998 19:25:44 +0800
From: JY Luke <jyluke@ew.mimos.my>
Reply-To: jyluke@ew.mimos.my
Organization: MIMOS Berhad
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To: rem-conf@es.net
Subject: Multimedia Asia 1998 - Virtual Commonwealth
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Dear all,

Sorry for this late annoucement as I can only confirm my MBone
connection just a moment ago.

In conjunction with Multimedia Asia, the annual conference on the
Multimedia Super Corridor, the National Information Technology Council
(NITC) presents the VIRTUAL COMMONWEALTH. 

The Virtual Commonwealth at MMA'98 is a dialogue session where issues of
governance in the Information Age will be discussed by prominent local
and international personalities. With the theme "Governance in an
Internetworked World: Towards a Global Civilized Society", the Virtual
Commonwealth session presents an opportunity for people to discuss
solutions to the problems of a globalized and internet-worked world.

Follow this url for more information:
http://streamworks.jaring.my/mma98.html or
http://propagation.jaring.my/mma98.html

Your are welcome to send your questions to the panelists prior to the
live session or during the Q&A session. For an overall view of the
session content and topics to be discussed at the Virtual Commonwealth
session, follow this link:
http://streamworks.jaring.my/vircomm-content.html or
http://propagation.jaring.my/vircomm-content.html

The live session will be broadcasted on 29th September 1998, 2100MYT
(GMT+8). 
Audio: DVI   224.2.227.121/21586 TTL=127
Video: H.261 224.2.211.189/49792 TTL=127

See you there.

Regards,
JY Luke
MIMOS Berhad,
Malaysia



From rem-conf Mon Sep 28 19:31:11 1998 
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Date: Mon, 28 Sep 1998 21:31:05 -0400 (EDT)
From: Henning Schulzrinne <hgs@cs.columbia.edu>
Message-Id: <199809290131.VAA29424@erlang.cs.columbia.edu>
To: rem-conf@es.net
Subject: CFP: Internet Computing/IEEE Network on Internet Telephony
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Call for Papers for a Joint Issue of 
  IEEE Internet Computing  (http://www.computer.org/internet)
  IEEE Network Magazine (http://www.comsoc.org/socstr/techcom/ntwrk/)
---------------------------------------------------------------------

Special Issue on Internet Telephony (May/June 1999)
---------------------------------------------------

Submission deadline: November 25, 1998

Guest Editor: Henning Schulzrinne
              Department of Computer Science
              Columbia University
              New York, NY
              Phone: 212 939 7042
              Fax:   212 666 0140
              WWW:   http://www.cs.columbia.edu/~hgs
              email: hgs@cs.columbia.edu

Internet telephony, that is, using the Internet as a communications
infrastructure to replace and enhance existing telephone services, is
promising to fundamentally change one of the most basic and familiar
communications services.  This special issue covers the technology
needed to make Internet telephony a reality and the challenges that
need to be addressed.

This is a joint issue of the IEEE Internet Computing and IEEE Network
Magazines.

Topics of interest for technical papers include:

* quality-of-service specific to large-scale telephony services
* charging, billing and customer care
* web-centric telephony service enhancements
* service creation: applications, languages, tools
* interworking with the global switched telephone network (GSTN):
  gateways, service location, directories
* hybrid Internet-PSTN architectures
* directory services
* signaling
* novel applications 
* embedded systems
* management of QOS, services and 
* security and privacy issues
* economic and regulatory impact

We are especially interested in research papers that describe real
services, software systems and experiments.  Work-in-progress papers
describing the state of on-going research projects in Internet telephony
are also encouraged. 

Papers should explicate the technical issues related to Internet
telephony, rather than just generic multimedia applications.  
Research papers should demonstrate the feasibility of the approach and
describe the state of realization.  Case studies and applied papers
should discuss the key factors that made the system work and should also
mention the pitfalls and problems encountered and how they may be
overcome. 

Submission, Approval, Review, and Acceptance. 
---------------------------------------------
Authors are requested to send e-mail to the guest-editor, Henning
Schulzrinne (hgs@cs.columbia.edu), giving a URL where the guest-editor
can review the article, preferably in HTML format with GIF artwork
(postscript or pdf format is also accepted).

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

Articles for review should be in HTML or a common format easily read by
reviewers.  Authors will send all files to an anonymous ftp site
provided by the guest editor.  When an article consists of a collection
of HTML and GIF files, all internal links pointing to this file should
be relative. 

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

Submission Deadline:                    November 25, 1998
Acceptance Notification:                February 1, 1999
Final Manuscripts Due:                  March 1, 1999
Publication of Completed Special Issue: May/June 1999

http://www.comsoc.org/socstr/techcom/ntwrk/special/schulzrinne.html



From rem-conf Tue Sep 29 04:45:38 1998 
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To: IETF-Announce: ;
Cc: RFC Editor <rfc-editor@isi.edu>
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Cc: rem-conf@es.net
From: The IESG <iesg-secretary@ietf.org>
Subject: Protocol Action: RTP Payload Format for the 1998 Version of
	 ITU-T Rec. H.263 Video (H.263+) to Proposed Standard
Date: Tue, 29 Sep 1998 07:36:53 -0400
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The IESG has approved the Internet-Draft 'RTP Payload Format for the
1998 Version of ITU-T Rec. H.263 Video (H.263+)'
<draft-ietf-avt-rtp-h263-video-02.txt> as a Proposed Standard.  This
document is the product of the Audio/Video Transport Working Group.
The IESG contact persons are Scott Bradner and Vern Paxson.


Technical Summary
 
  This protocol specifies an RTP payload header format for carrying
  video streams transmitted using the 1998 version of ITU-T 
  Recommendation H.263. The format is especially designed for resiliency 
  in the face of packet loss.

Working Group Summary

  This draft was presented at the AVT working group meeting in 
  Washington, DC, and received extended discussion.  The draft was 
  revised based on that feedback and presented in Los Angeles where 
  there was consensus for going to Proposed.  The draft was subsequently 
  given a working group last call.

  There was one issue that generated a number of comments, specifically 
  the relationship between this draft and RFC 2190 on the earlier 
  version of the H.263 encoding standard.  It was agreed to add a 
  paragraph in the new spec explaining its relationship to RFC 2190, 
  which has been done in this version.

  There is at least one implementation of the protocol.

Protocol Quality

  Vern Paxson reviewed the protocol for the IESG.  Note that this 
  protocol does not replace RFC 2190, which continues to be used by 
  existing implementations, and may be required for backward 
  compatibility in new implementations.




From rem-conf Tue Sep 29 10:28:19 1998 
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Date: Tue, 29 Sep 1998 10:18:58 -0700
To: rem-conf@es.net
From: Florissa Colina <florissac@bmrc.berkeley.edu>
Subject: 9/30 Berkeley, Multimedia, Interfaces, and Graphics   Seminar  
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<center>Use What You Teach: Teaching CSCW Using CSCW Tools 


James Landay 

Computer Science Division - EECS 

UC Berkeley 


Wednesday September 30, 1998 

12:30-2:00 PDT 405 Soda Hall

</center>

Distance learning technology has repeatedly been cited as a way to reduce
the cost of advanced education and to bring training to the masses.
Drawbacks aside, this technology will be used in many educational
settings and it is therefore important for educators to understand its
advantages and limitations. Researchers in the field of
computer-supported cooperative work (CSCW) have been building research
systems for years that support distance learning. To expose students to
the difficult issues in this area and to learn more myself, I taught a
course on CSCW in the Fall 1997 semester in which we used CSCW tools. In
this talk I describe the course structure, the technology we used, and
the outcomes of the course. 


The course website is at:
<underline><color><param>0000,0000,fefe</param>http://bmrc.berkeley.edu/courseware/cscw/fall97/index.html</color></underline>. 


The Mbone addresses are video 224.2.163.7/57076 and audio
224.2.147.61/27202. 



____________________________________________


Florissa Colina

Berkeley Multimedia Research Center (BMRC)

http://bmrc.berkeley.edu/

626 Soda Hall #1776

University of California

Berkeley, CA 94720-1776

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

Email: florissac@bmrc.berkeley.edu



From rem-conf Tue Sep 29 13:51:13 1998 
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Date: Tue, 29 Sep 1998 16:44:40 -0400 (EDT)
From: Steve Hanna <shanna@bcn.East.Sun.COM>
Reply-To: Steve Hanna <shanna@bcn.East.Sun.COM>
Subject: Re: Video and Java
To: sukeshgg@caip.rutgers.edu
Cc: rem-conf@es.net
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Sun has some hot "Java on Solaris" technology that is available as "early 
access" through the Solaris Developer connection.  You have to register, but 
registration is free.

Check out http://www.sun.com/solaris/java/ and look for the following blurb:

  Download the JDK 1.2 Early Access 4 (Production Release)

  As the leader in SPECjvm98 performance, this early access release
  (now supporting Solaris 2.5.1 and 2.6 - SPARC/Intel) provides
  significantly improved performance and scalability due to an enhanced
  JVM with improved JIT Compiler, fast thread synchronization and
  enhanced memory system.

Sun also ships a reference platform designed to facilitate portability,
but it has less performance, so make sure you are getting something
labelled "Production Release."

> Date: Thu, 10 Sep 1998 12:12:22 -0400 (EDT)
> From: Sukesh Garg <sukeshgg@caip.rutgers.edu>
> To: rem-conf@es.net
> Subject: Video and Java
> MIME-Version: 1.0
> 
>  Hi,
>  I am working on a project that involves sending real-time data over the
> n/w using Java. 
>  I have built an application and found the my application to be very slow.
> 
> Is there any implementation available that is fast....
> or any pointers as to increase the speed...(using java)
> 
> Thanks
> 
> Sukesh
> 
> 
> 




From rem-conf Tue Sep 29 14:54:16 1998 
From rem-conf-request@es.net Tue Sep 29 14:54:15 1998
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From: Forrest  Blair <fblair@s-vision.com>
To: potsag@imtc.org, h323implementors@imtc.org, sg16.lbc@research.kpn.com, 
	rem-conf@es.net
Subject: H.263 Full PIcture Freezes
Date: Tue, 29 Sep 1998 15:51:31 -0600
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Does anyone use H.263 "Full Picture Freeze" and if so what is the
application?

Thanks in advance,

Forrest Blair
Sorenson Vision, Inc.




From rem-conf Wed Sep 30 12:03:10 1998 
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Date: Wed, 30 Sep 1998 11:49:36 -0700
From: lalwaney@procy.gi.com (poornima lalwaney)
Subject: CFP - Multimedia, Services, Applications and Technologies
To: rem-conf@es.net
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> Multimedia Applications, Services, & Technologies '99 (MAST-99)
> in conjunction with ICC 1999, Vancouver, Canada 
> Sponsor: IEEE Communication Society, Technical Committee on Multimedia
> Communications 
> 
> The miniconference will include a keynote speech, panel discussions, paper
> session, and 
> applications/demo session by leaders, recognized experts and active
> researchers in the field. 
> 
> The past few years have seen the Internet explode with multimedia content
> and applications 
> taking advantage of the world-wide-web. High-speed access technologies
> (e.g., xDSL and 
> cable-modems) and content distribution methodologies (e.g., web caching
> and mirroring) 
> are now becoming available which will improve perceived network
> performance. The 
> Internet protocol suite and open standards such as HTML, JAVA, etc. are
> emerging as 
> leading-edge technologies for publishing and delivering multimedia content
> (e.g, 
> video/audio streaming over IP multicast and unicast protocols, images,
> text) to business and 
> residential customers worldwide. 
> 
> This mini-conference is intended to serve as an open forum for discussing
> state-of-the-art 
> developments in the exciting field of multimedia applications, services
> and technologies. The 
> emphasis will be on emerging services and applications designed over the
> open Internet protocol 
> suite and web authoring tools. 
> 
> http://www.comsoc.org/confs/icc/99/mast.html


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