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Subject: Protocol Action: Guidelines for Writers of RTP Payload Format
	 Specifications to BCP
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The IESG has approved the Internet-Draft 'Guidelines for Writers of RTP
Payload Format Specifications'
<draft-ietf-avt-rtp-format-guidelines-04.txt> as a BCP.  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 document provides guidelines for how to design efficient
 formats for carrying payloads in RTP packets.  The document
 emphasizes the importance of keeping each packet as self-contained
 as possible, so that packets remain useful even in the presence
 of loss and reordering.  The discussion is in terms of codec
 payloads, but the principles apply to most other forms of RTP
 payloads, too.

Working Group Summary

 The goal of this document was to capture discussion which occurred over
 the course of many AVT meetings regarding functions that an RTP pyaload
 format should provide.  The document was revised twice to address working
 group comments.  There were no controversial issues remaining regarding
 the present draft.

Protocol Quality

 The document was reviewed for the IESG by Scott Bradner and Vern Paxson.



From rem-conf Mon Nov 01 08:40:35 1999 
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From rem-conf Mon Nov 01 10:03:55 1999 
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		    Audio/Video Transport Working Group

				  Agenda


Wednesday 10th November 1999, 15:30-17:30
=========================================

- Introduction and status update					10
	draft-ietf-avt-fec-08.txt
	draft-ietf-avt-rtp-format-guidelines-04.txt 
	draft-ietf-avt-rtpsample-05.txt
	draft-ietf-avt-rtp-mib-07.txt
	draft-ietf-avt-tones-02.txt

- RTP and the audio/video profile					30
	draft-ietf-avt-rtp-new-05.txt
	draft-ietf-avt-profile-new-07.txt
	draft-ietf-avt-rtp-mime-01.txt
	draft-ietf-avt-rtcp-bw-00.txt
	draft-ietf-avt-rtp-interop-02.txt
	draft-ietf-avt-rtptest-02.txt

- RTP Payload Format for Real-Time Pointers
	draft-ietf-avt-pointer-00.txt			(Civanlar)	 5

- RTP Payload for text conversation		
	draft-ietf-avt-rtp-text-02.txt			(Hellstrom)	 5

- RTP Payload Format for DV Format Video
	draft-ietf-avt-dv-video-01.txt			(Kobayashi)	10

- RTP Payload Format for 12 bits, 20bits and 24 bits DV Audio
	draft-ietf-avt-dv-audio-00.txt			(Kobayashi) 	10

- A RTCP-based Retransmission Protocol for Unicast RTP Streaming Multimedia
	draft-podolsky-avt-rtprx-00.txt 		(Yano)		10

- RTP Payload Format for MPEG-2 AAC Streams
	draft-ietf-avt-rtp-mpeg2aac-01.txt		(Basso)		10

- A More Loss-Tolerant RTP Payload Format for MP3 Audio
	draft-ietf-avt-rtp-mp3-01.txt			(Finlayson)	10

- Extension of RTP payload Type for Multiple Program MPEG Transport Stream
	draft-ietf-avt-rtp-mp2t-00.txt (sent to list)	(Liu)		10


Thursday 11th November 1999, 09:00-11:30
========================================

- Transport of MPEG-4
	draft-ietf-avt-rtp-mpeg4-02.txt			(Basso)		 5
	draft-guillemot-genrtp-01.txt			(Christ)	 5
	draft-jnb-mpeg4av-rtp-00.txt			(Kikuchi)	 5
	Liason statement from MPEG (29n3290.zip)
	draft-singer-rtp-qtfile-01.txt
	Discussion							45

- Multiplexing
	draft-koren-avt-crtp-enhance-00.txt 		(Thompson)	10
	draft-tulai-avt-dynamic-nx64-00.txt		(Tulai)		10
	Discussion							30

- Charter Bashing							10


The documents mentioned are available from:
  http://www.ietf.org/html.charters/avt-charter.html
  http://www-mice.cs.ucl.ac.uk/multimedia/misc/avt/IETF46/internet-drafts/

NOTE to presenters: The purpose of these slots on the AVT schedule is
not to present what's in the draft, but to identify what problems were
resolved, or what issues still remain open, and to make it clear what
action from the working group is requested so we can discuss it. Please
confirm with us that you will be able to make the presentations listed.

NOTE to attendees: Since the presenters aren't supposed to present their
drafts, it follows that the attendees are responsible for having read the
drafts already or they may be disappointed!

At the Oslo meeting, we had too many brand new drafts to cover so we did
not have adequate time for discussion.  We'll try to manage that more
carefully this time.  

We also plan to hold a separate design team meeting to discuss transport of 
MPEG-4. Those who would be interested in attending such a meeting should
contact us so we can arrange for a room. This is for those who have been
active in the discussion of MPEG-4 and the MPEG-4 payload format - we will
report back to the rest of the group during the Thursday session.

Colin & Steve



From rem-conf Mon Nov 01 13:05:32 1999 
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To: rem-conf@es.net
Subject: RTP A/V profile interoperability draft
Date: Mon, 01 Nov 1999 20:45:54 +0000
From: Colin Perkins <c.perkins@cs.ucl.ac.uk>
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Folks,

Attached is draft-ietf-avt-profile-interop-00.txt, the interoperability
statement for the RTP audio/video profile which missed the I-D deadline.
This complements draft-ietf-avt-rtp-interop-02.txt. 

As Steve has noted previously:
  >In addition, this is a request to RTP implementors to fill in the
  >interoperability checklist that we will need to accompany the drafts
  >for advancement to Draft Standard.  We've made the same request
  >informally before, but this time we're asking that you do this in the
  >next two weeks so that we can have at least a first-pass tally at the
  >meeting.  If it looks like there are features for which we don't have
  >two interoperable implementations, we may have to remove those
  >features from the draft, or do some additional implementation and
  >interop testing!
We also need implementation reports for the a/v profile, as detailed by
this draft. I would encourage implementors to complete this checklist
and the checklist for the main RTP specification.

Colin

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


INTERNET-DRAFT                               28 October 1999


                                               Colin Perkins
                                   University College London



      RTP Audio/Video Profile Interoperability Statement
            draft-ietf-avt-profile-interop-00.txt


                    Status of this memo

This document is an Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC2026.

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

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

The list of current Internet-Drafts can be accessed at 
http://www.ietf.org/ietf/1id-abstracts.txt

The list of Internet-Draft Shadow Directories can be accessed at
http://www.ietf.org/shadow.html.

Distribution of this document is unlimited.

Comments are solicited and should be addressed to the author and/or the
IETF Audio/Video Transport working group's mailing list at rem-conf@es.net.


                         Abstract

    It is required to demonstrate interoperability of implementations
    in order to move the RTP audio/video profile to draft standard.
    This memo outlines those features to be tested, as the first
    stage of an interoperability statement.


1  Introduction

The Internet standards process [1] places a number of requirements
on a standards track protocol specification.  In particular, when
advancing a protocol from proposed standard to draft standard it
is necessary to demonstrate at least two independent and interoperable
implementations, from different code bases, of all options and features
of that protocol.  Further, in cases where one or more options or
features have not been demonstrated in at least two interoperable


Perkins                                            Page 1


INTERNET-DRAFT                             28 October 1999


implementations, the specification may advance to the draft standard level
only if those options or features are removed.  The RTP Profile for Audio
and Video Conferences with Minimal Control was originally specified in
RFC1890 as a proposed standard [2].  The revision of this specification for
draft standard status is now well underway, so it has become necessary to
conduct such an interoperability demonstration.

This memo describes the set of features and options of the RTP Audio/Video
profile which need to be tested as a basis for this demonstration.  This
memo is for information only and does not specify a standard of any kind.


2  Features and options required to demonstrate interoperability


In order to demonstrate interoperability it is required to produce
a statement of interoperability for each feature noted below.  Such
a statement should note the pair of implementations tested, including
version numbers, and a pass/fail statement for each feature.  It
is not expected that every implementation will implement every feature,
but each feature needs to be demonstrated by some pair of applications.
The RTP specification [3] enumerates a number of items that can be
specified or modified by a profile.  Those modified from the defaults
by the audio/video profile are as follows, and interoperable behaviour 
must be demonstrated:

  1.Exchange of RTCP packets with the default RTCP reporting interval.

  2.Exchange of RTCP packets with a modified RTCP reporting interval
    as selected by a different fraction of the session bandwidth
    (the means by which this interval is signalled are outside of
    the scope of this memo, but one such mechanism should be demonstrated).

  3.Implementations must send RTCP packets containing an SDES CNAME
    in every reporting interval.

  4.Other SDES items must be sent every third interval, with NAME
    every 7 of 8 times within that slot and any other SDES items
    cycically taking up the 8th slot.

  5.Interoperable selection of `pass phrase' for encryption and exchange
    of media using DES encryption.  This must include mapping of
    the pass phrase to the canonical form.

  6.Interoperable transport of RTP via unicast UDP

  7.Interoperable transport of RTP via multicast UDP

  8.Interoperable transport of RTP via TCP using the encapsulation
    defined in section 7 of [2].


Perkins                                            Page 2


INTERNET-DRAFT                             28 October 1999


The primary purpose of the audio/video profile is to define the mapping
>from payload format to payload type number.  Accordingly, the majority
of the work needed to demonstrate interoperability consists of testing
that media data is exchanged in an interoperable manner using the
full range of codecs enumerated in the profile.

The following codecs are assigned static payload types.  It should
be verified that interoperable implementations exist for each static
payload type:

  1.The 8kHz PCMU codec (payload type 0)

  2.The 8kHz 1016 codec (payload type 1)

  3.The 8kHz G726-32 codec (payload type 2)

  4.The 8kHz GSM codec (payload type 3)

  5.The 8kHz G723 codec (payload type 4)

  6.The 8kHz DVI4 codec (payload type 5)

  7.The 16kHz DVI4 codec (payload type 6)

  8.The 8kHz LPC codec (payload type 7)

  9.The 8kHz PCMA codec (payload type 8)

 10.The 8kHz G722 codec (payload type 9)

 11.The 44.1kHz stereo L16 codec (payload type 10)

 12.The 44.1kHz mono L16 codec (payload type 11)

 13.The 8kHz QCELP codec (payload type 12)

 14.The 8kHz CN codec (payload type 13)

 15.The MPA codec (payload type 14)

 16.The 8kHz G728 codec (payload type 15)

 17.The 11.025kHz DVI4 codec (payload type 16)

 18.The 22.050kHz DVI4 codec (payload type 17)

 19.The 8kHz G729 codec (payload type 18)

The following codecs use dynamic payload types.  It should be verified
that some non-RTP means can be used to assign a dynamic payload type
to be used by implementations, and that those implementations can
then interwork.

  1.The GSM-HR codec

  2.The GSM-EFR codec

Perkins                                            Page 3


INTERNET-DRAFT                             28 October 1999

  3.The L8 codec

  4.The RED codec

  5.The VDVI codec

Similarly, interoperable implementation of the following video codecs
with static payload type numbers must be demonstrated:

  1.The CelB codec (payload type 25)

  2.The JPEG codec (payload type 26)

  3.The nv codec (payload type 28)

  4.The H261 codec (payload type 31)

  5.The MPV codec (payload type 32)

  6.The MP2T codec (payload type 33)

  7.The H263 codec (payload type 34)

The following codecs use dynamic payload types.  It should be verified
that some non-RTP means can be used to assign a dynamic payload type
to be used by implementations, and that those implementations can
then interwork.

  1.The BT656 codec

  2.The H263-1998 codec

  3.The MP1S codec

  4.The MP2P codec

  5.The BMPEG codec

Implementations must also demonstrate interoperable use of the marker
bit.  That is, the M bit is set on the first packet of each talkspurt
for audio and on the last packet of each frame for video.


3  Author's Address

Colin Perkins
Department of Computer Science
University College London
Gower Street
London WC1E 6BT
United Kingdom

Email:  c.perkins@cs.ucl.ac.uk

Perkins                                            Page 4


INTERNET-DRAFT                             28 October 1999


4  References

[1] S. Bradner, ``The Internet Standards Process -- Revision 3'',
    RFC2026, Internet Engineering Task Force, October 1996.

[2] H. Schulzrinne, S. Casner, ``RTP Profile for Audio and Video
    Conferences with Minimal Control'', RFC1890, Internet Engineering
    Task Force, January 1996.

[3] H. Schulzrinne, S. Casner, R. Frederick and V. Jacobson, ``RTP:
    A Transport Protocol to Real-Time Applications'', RFC1889, Internet
    Engineering Task Force, January 1996.







































Perkins                                            Page 5



From rem-conf Mon Nov 01 16:08:08 1999 
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From: Matt Podolsky <podolsky@CS.Berkeley.EDU>
To: rem-conf@es.net
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        Steve McCanne <mccanne@eecs.berkeley.edu>
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Hello-

I'd like to announce the availability of
draft-podolsky-avt-rtprx-00.txt, written by Koichi
Yano, Steve McCanne, and myself.  It's a first cut at
creating a standard format for doing retransmissions of
unicast RTP streams.  We're very interested in getting
feedback from the group and improving it.

It can be found on the IETF website at:

http://search.ietf.org/internet-drafts/draft-podolsky-avt-rtprx-00.txt

The abstract is below.  Koichi and I will also be at
the IETF next week, and we have a slot in Wednesday's
session for discussion.

Thanks,
Matt

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

Abstract:

   This note defines a retransmission request protocol for use with
   unicast Real Time Protocol (RTP) multimedia sessions.  We refer
   to this protocol as RTPRx.  The protocol is defined as a new RTCP
   type and is flexible enough to allow a variety of retransmission
   request and response schemes.  This note describes the basic format
   for retransmission requests, and provides examples of three different
   retransmission request-response algorithms which use different
   methods to determine if retransmissions should be requested and
   whether the requests should be served.






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           CERN is pleased to announce the MBONE broadcast of the

                          LEP Experiments Committee
                          =========================
                           Tuesday 9 November 1999


Place: CERN Main Auditorium


*** Times are UTC+1 ***


                  Reports on the LEP machine
                  --------------------------
  09:00 - 09:30   Status of LEP (Andy Butterworth)

                  Reports on the LEP experiments
                  ------------------------------
  09:30 - 10:00   OPAL (Patricia Ward)
  10:00 - 10:30   ALEPH (Alain Blondel)

  10:30 - 11:00   Coffee break

  11:00 - 11:30   DELPHI (Jesus Marco)
  11:30 - 12:00   L3 (Ghita Rahal-Callot)



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

        The MBONE Broadcast is Announced via 'sdr' as  "CERN LEPC"
        The 'vat'&'vic' Applications will be Used with a 'ttl=127'

                  Audio   Addr 224.2.174.8     Port 31974
                  Video   Addr 224.2.234.43    Port 53004

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

          The Sessions are Recorded Using the 'wrtp' Applications
            They can be Downloaded from the following Web-Site:
            "http://csvod1.cern.ch/cgi-bin/nph-MBone-sessions/"

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

  In Case of Questions or Problems Please Contact: "multicast@noc.cern.ch"





From rem-conf Wed Nov 03 08:41:27 1999 
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Date: Wed, 03 Nov 1999 11:29:29 -0500
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From: Robert Zopf <zopf@lucent.com>
Subject: CNG
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Dear Colleagues,

The main concerns regarding a new CNG scheme have been the following:
* the creation of a variant of G.711 would reduce the interoperability of 
different implementations, both within H.323 and within other applications.
* a CNG scheme that remains generic for use with any codec would be more 
useful than if it is specific to G.711.
* the implications of breaking the existing CNG mechanism and proposing a 
new one should be carefully considered before moving forward.
* defining a CNG scheme may in some way limit the flexibility of the design 
of VAD/DTX systems that it would operate with.
* due ITU procedures should be in place and enough time should be given for 
interested parties to advance their proposals.
* the IETF is currently advancing the draft RTP Profile to Draft Standard 
at the meeting next week in Washington, DC.

The main objective of this CNG work proposed by Lucent is to define a means 
for H.323 to provide an acceptable level of quality using Comfort Noise in 
an interoperable fashion for codecs (especially G.711) without built-in 
CNG.  In VoIP, the 3 main codecs being deployed are G.711, G.723.1, and 
G.729A.  For sessions using G.723.1, or G.729A, the built-in CNG provides 
an acceptable level of quality during DTX operation.  However, for G.711, 
there is no built-in CNG, and the mechanism within RTP must be used to 
signal DTX and describe the background noise.  In this case, currently only 
the energy is available, and as a result, a predictable, reliable, 
acceptable level of quality across different background noise conditions is 
not possible, especially between different vendors equipment.  Of course 
this problem also exists for other codecs without built-in CNG, such as 
G.722, G.726, G.727, G.728, etc.

In light of all  of the above, our proposal is to define an extension to 
the current "energy only" scheme.  This extended definition may be used to 
further describe the background noise.  However, it's full or partial use 
is up to the implementation.  The first byte would represent the energy in 
exactly the same manner it is now in RTP.  The presence of additional bytes 
would signal additional information, the meaning of which and decoding of 
its parameters we would define.  Ideally, this additional information would 
not be codec or sampling rate specific so that the scheme could continue to 
be used as flexibly as it is now.  The current method of signalling DTX and 
updating the CN information would remain the same as it is now (that is, 
the first CN packet identifies the start of the inactive segment, and the 
update rate of this information after this is implementation specific).

Interested parties would propose bit-stream formats (and part of this would 
be to define how to properly encode/decode the bit-stream to and from the 
parameter set), and a "preferred" means of analysis and synthesis.  This 
preferred means would be tested against the requirements set out in the ToR 
to verify that at least one implementation exists that meets the ToR.  We 
will obtain guidance from SG12 on how to best test the CNG.  Based on the 
results, the working party group would select the best candidate, and the 
bit-stream format and preferred implementation would make up an appendix 
that could be referenced for use in protocols such as RTP.  Ideally, the 
Audio/Video Transport Working Group of the IETF could adopt the 
"extended"  CN in the future.

Please provide any comments by the end of the week.  If this is an 
acceptable means of moving forward, we will send out a draft ToR reflecting 
this approach.

Best Regards,

Rob.

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Rob Zopf
Principal Engineer
Speech Technology Group
Lucent Technologies, INS

Phone 	: (732) 578-3207
Fax		: (732) 578-3213
Email		: zopf@lucent.com
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~



From rem-conf Wed Nov 03 09:11:10 1999 
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Colleagues,

 	The beauty of the G.711 is that almost no processing
	is required and packetization delay can be minimized 
	if desired. 

	My issue with any further complication of VAD/CNG
	operation in G.711 is the increased complexity 
	required to implement.

mailto:rwood@mosaid.com



From rem-conf Wed Nov 03 10:10:39 1999 
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Date: Wed, 03 Nov 1999 13:07:56 -0500
To: Robert Wood <rwood@mosaid.com>, zopf@lucent.com
From: Gerard Aguilar <Gerard.Aguilar@ascend.com>
Subject: Re: CNG
Cc: wp3audio@ctd.comsat.com, rem-conf@es.net
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It is true that G.711 is very low complexity and
adding any additional processing needed in a channel
must be highly scrutinized.  However, in real-world VoIP
systems each "Channel" is actually running much more
than G.711.  In fact listed below is what typically runs
within each G.711 channel with estimates of actual DSP
complexity.

0 - 1 MIPS  -- G.711 (HW/SW)
1 - 3 MIPS  -- Fax/Modem Detect
1- 3 MIPS -- DTMF Detect/Regeneration
7 - 15 MIPS -- G.165/G.168 LEC
0.25 - 2 MIPS -- VAD/CNG
0.25 - 3 MIPS -- Packet Loss Recovery
----------------------------------------------------------
10 MIPS - 27 MIPS


Therefore, if one ELECTED to incorporate an enhanced 1 MIP CNG
over what exists today within RFC 1890 it would only take between
4% - 10% of the complexity required to run an actual G.711 Channel.

Gerard





At 12:08 PM 11/03/1999 -0500, Robert Wood wrote:

>Colleagues,
>
>         The beauty of the G.711 is that almost no processing
>         is required and packetization delay can be minimized
>         if desired.
>
>         My issue with any further complication of VAD/CNG
>         operation in G.711 is the increased complexity
>         required to implement.
>
>mailto:rwood@mosaid.com




From rem-conf Wed Nov 03 11:41:39 1999 
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From: Stephen Casner <casner@cisco.com>
To: Robert Zopf <zopf@lucent.com>
cc: wp3audio@ctd.comsat.com, rem-conf@es.net
Subject: Re: CNG
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I thank Rob Zopt for his message on Comfort Noise Generator (CNG)
enhancements.  Let me try to provide a bit of context for the AVT
working group.

Rob contacted me as a result of the working group last call
announcement on advancing the revision of the RFC 1890 RTP Audio/Video
Profile to Draft Standard.  One of the additions in that revision is a
Comfort Noise (CN) payload format which is a simple one-octet message
specifying the noise level.  Rob asked whether changes to the comfort
noise specification would still be possible at this point, and if so,
what form that should take.

I had seen the proposal on the ITU SG16 mailing list to provide an
enhanced CNG specification which included spectral shape in addition
to level.  In an earlier reply to messages from Rob and others about
this proposal, I said that if the CN payload format/encoding as
defined in the current A/V Profile draft is not useful as it stands,
then it should be replaced with something that is useful before the
draft revision of RFC 1890 is published as a new RFC.

I agree with Rob that it would not be a good idea to add comfort noise
functions to a modified version of the PCMU/A (G.711) payload format
because that would likely impair interoperability and would require
similar revisions to other payload formats to add comfort noise to
them.

Options include:

  - Extending the current CN defintion as Rob described in his message.
    The current "energy only" specification would still be valid, but
    additional spectral information could be added.  Receivers would
    know from the payload length whether the additional info was
    present.  Does this seem like a "safe" extension that would not
    cause problems for existing implementations of CN as in the draft?
    Note that CN has not been published in an RFC, but it has been in
    the sequence of draft revisions for at least two years.

  - Leaving the existing CN as is, but adding another CNG payload
    format for the enhanced generator.  This would avoid any
    compatibility issues, and would provide an opportunity to express
    the noise energy in a different way.  However, in H.323 it would
    require negotiation of both CN and CNG payload formats for
    backward compatibility with endpoints that only implement the
    existing CN.

For either of these options, there is a question of whether it is
possible to change the comfort noise specification in the revised
profile draft, and if so, how.  The answer is certainly yes because
the working group can decide that this function should be addressed
before the draft is submitted to the IESG for IETF-wide Last Call.

On the other hand, the advancement of RTP to Draft Standard is long
overdue, so we'd like to avoid introducing additional delay.  There
may be a compromise: ask for IESG Last Call on the main RTP spec, but
hold off on the profile.  The RTP spec cannot be published as an RFC
until the profile is also ready, but my concern is that the RTP spec
is large and somewhat significant and therefore the IESG review may
take some time.  I'd like to get that process started.

A question for the ITU folks is how much time would be required to
allow the enhanced CNG proposal to be completed and tested by the
speech experts to make sure it works correctly before we standardize
it.  This notion is well accepted within IETF.

If the consensus is that a new CNG payload format should be added
(leaving the existing CN unchanged), then it would be possible for the
new payload format to be defined in a separate RFC just as we expect
the continuing development of additional payload format RFCs for new
audio and video encodings.  Perhaps we could/should add words to the
CN definition in the current draft to say that enhanced comfort noise
payload formats are in development and will be published in future
RFCs.

I'd like to discuss these questions as part of the review of the RTP
spec and profile status in the AVT meeting at IETF next Wednesday.
Rob will be attending to participate in that discussion.

							-- Steve




From rem-conf Wed Nov 03 11:52:27 1999 
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From: Stephen Casner <casner@cisco.com>
To: Robert Wood <rwood@mosaid.com>, Gerard Aguilar <Gerard.Aguilar@ascend.com>
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On Wed, 3 Nov 1999, Robert Wood wrote:

>  	The beauty of the G.711 is that almost no processing
> 	is required and packetization delay can be minimized 
> 	if desired. 
> 
> 	My issue with any further complication of VAD/CNG
> 	operation in G.711 is the increased complexity 
> 	required to implement.

Note that the "MBone tools" vat and RAT have implemented PCMU (G.711)
encoding for a long time without any comfort noise mechanism at all.
Silence suppression is still possible -- the receiver does not require
an indicator at the start of silence, it just runs out of data.

With the definition of the CN payload format or an enhanced CNG
payload format, it is still possible for simple systems to just
process the PCMU payload format packets and ignore the CN packets if
that level of performance is acceptable.  This high level of
interoperability is an advantage of keeping the existing audio
encoding payload formats separate from CN.


On Wed, 3 Nov 1999, Gerard Aguilar wrote:

> Therefore, if one ELECTED to incorporate an enhanced 1 MIP CNG
> over what exists today within RFC 1890 it would only take between
> 4% - 10% of the complexity required to run an actual G.711 Channel.

Please note that RFC 1890 did not define any comfort noise scheme,
only the simple silence suppression I described above.

							-- Steve




From rem-conf Thu Nov 04 08:00:08 1999 
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I would like to make three points concerning CN:

1) Who has ever implemented the CN in a real system? Whether you completely
squelch audio
in between talk spurts or you add in an artificial static representation of
white or colored noise makes
no difference on audio quality.  CN was and is an ill-conceived notion.
Holding on to it for legacy reasons
when nobody will likely defend it use doesn't make sense.  I think that the
CN type should be redefined.
 Anyone who does use it - please respond and defend its use.

2) One of the respondants to this string indicates that the current
VAD/DTX/CNG in G.729 and G.723.1
work pretty well.  While they seem to work ok under nominal conditions,
they suffer from some flaws:
     a) The g.723.1 DTX scheme will stream SID packets when you send all
very low energy (no energy)
     Careful analysis of equation A-10 in G.723.1, Annex A will show this
to be true.
     b) Both G.729 and G.723.1 VAD algorithms were not adequately tested
with lower levels of speech
     The consequences are that an annoying "swooshing" sound occurs when
the VAD misdetects
     speech as nonspeech and codes a SID.
     c) The DTX decision logic is flawed.  Low levels of non-stationary
noise (power hum) can almost
     completely defeat the intended packet rate reduction during nonspeech
periods.
If there was to be a standard for a VAD/DTX/CNG scheme for G.711, it should
be mindful of these flaws.

3) Should there be a similar VAD for G.726 as well?  As  we begin to
understand that VoIP does not really
mean "Internet" telephony, it means Internet Protocol Telephony, we see
that people are moving away from the
low bit rate vocoders because they are not robust to ambient or non-speech
sounds, they do not transport DTMF
they degrade in speakerphone calls, and they do not tandem well at all
(conferencing). G.726 is low
complexity and half the bit rate of G.711.  No, it is not 6.4 or 8.0 kbps,
but I don't believe that people will put up
with the poor robustness of these speech coders unless they get something
for it in return (like mobility as in
digital cell phones).

Kevin Bruemmer






Stephen Casner <casner@cisco.com> on 11/03/99 02:39:23 PM

To:   Robert Zopf <zopf@lucent.com>
cc:   wp3audio@ctd.comsat.com, rem-conf@es.net (bcc: Kevin Bruemmer)
Subject:  Re: CNG




I thank Rob Zopt for his message on Comfort Noise Generator (CNG)
enhancements.  Let me try to provide a bit of context for the AVT
working group.

Rob contacted me as a result of the working group last call
announcement on advancing the revision of the RFC 1890 RTP Audio/Video
Profile to Draft Standard.  One of the additions in that revision is a
Comfort Noise (CN) payload format which is a simple one-octet message
specifying the noise level.  Rob asked whether changes to the comfort
noise specification would still be possible at this point, and if so,
what form that should take.

I had seen the proposal on the ITU SG16 mailing list to provide an
enhanced CNG specification which included spectral shape in addition
to level.  In an earlier reply to messages from Rob and others about
this proposal, I said that if the CN payload format/encoding as
defined in the current A/V Profile draft is not useful as it stands,
then it should be replaced with something that is useful before the
draft revision of RFC 1890 is published as a new RFC.

I agree with Rob that it would not be a good idea to add comfort noise
functions to a modified version of the PCMU/A (G.711) payload format
because that would likely impair interoperability and would require
similar revisions to other payload formats to add comfort noise to
them.

Options include:

  - Extending the current CN defintion as Rob described in his message.
    The current "energy only" specification would still be valid, but
    additional spectral information could be added.  Receivers would
    know from the payload length whether the additional info was
    present.  Does this seem like a "safe" extension that would not
    cause problems for existing implementations of CN as in the draft?
    Note that CN has not been published in an RFC, but it has been in
    the sequence of draft revisions for at least two years.

  - Leaving the existing CN as is, but adding another CNG payload
    format for the enhanced generator.  This would avoid any
    compatibility issues, and would provide an opportunity to express
    the noise energy in a different way.  However, in H.323 it would
    require negotiation of both CN and CNG payload formats for
    backward compatibility with endpoints that only implement the
    existing CN.

For either of these options, there is a question of whether it is
possible to change the comfort noise specification in the revised
profile draft, and if so, how.  The answer is certainly yes because
the working group can decide that this function should be addressed
before the draft is submitted to the IESG for IETF-wide Last Call.

On the other hand, the advancement of RTP to Draft Standard is long
overdue, so we'd like to avoid introducing additional delay.  There
may be a compromise: ask for IESG Last Call on the main RTP spec, but
hold off on the profile.  The RTP spec cannot be published as an RFC
until the profile is also ready, but my concern is that the RTP spec
is large and somewhat significant and therefore the IESG review may
take some time.  I'd like to get that process started.

A question for the ITU folks is how much time would be required to
allow the enhanced CNG proposal to be completed and tested by the
speech experts to make sure it works correctly before we standardize
it.  This notion is well accepted within IETF.

If the consensus is that a new CNG payload format should be added
(leaving the existing CN unchanged), then it would be possible for the
new payload format to be defined in a separate RFC just as we expect
the continuing development of additional payload format RFCs for new
audio and video encodings.  Perhaps we could/should add words to the
CN definition in the current draft to say that enhanced comfort noise
payload formats are in development and will be published in future
RFCs.

I'd like to discuss these questions as part of the review of the RTP
spec and profile status in the AVT meeting at IETF next Wednesday.
Rob will be attending to participate in that discussion.

                                   -- Steve









From rem-conf Thu Nov 04 08:55:27 1999 
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From: Richard V Cox <rvc@research.att.com>
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To: Kevin_Bruemmer@nmss.com
Cc: casner@cisco.com, Robert Zopf <zopf@lucent.com>,
	wp3audio@ctd.comsat.com, rem-conf@es.net
Subject: Re: CNG
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Comfort Noise has been used in Circuit Multiplication Equipment for
years.  Its precise implementation is left up to vendors of the
equipment.  I believe they just supply the noise level and insert
white noise, but since this information is not public, I can only
speculate.  Listening to speech over a loudspeaker vs on a handset
is what distinguishes whether CN is needed.  In an office like
mine, with lots of fan noise from computers and AC/heating system,
I would not need CN for listening to speech on a loudspeaker
unless the background noise level at the other end was very
high.  On the other hand, I believe that on a handset I would
draw the opposite conclusion unless the background noise
was very low at the other end.
    The issue of the VADs in G.723.1 and G.729 not working
well for low level speech is immaterial as to whether CN is
needed.  If they really don't work well enough for real world
conditions, they would need to be improved.
    So, put me down as being in favor of CN generally.
        Rich Cox

Kevin_Bruemmer@nmss.com wrote:

> I would like to make three points concerning CN:
>
> 1) Who has ever implemented the CN in a real system? Whether you completely
> squelch audio
> in between talk spurts or you add in an artificial static representation of
> white or colored noise makes
> no difference on audio quality.  CN was and is an ill-conceived notion.
> Holding on to it for legacy reasons
> when nobody will likely defend it use doesn't make sense.  I think that the
> CN type should be redefined.
>  Anyone who does use it - please respond and defend its use.
>
> 2) One of the respondants to this string indicates that the current
> VAD/DTX/CNG in G.729 and G.723.1
> work pretty well.  While they seem to work ok under nominal conditions,
> they suffer from some flaws:
>      a) The g.723.1 DTX scheme will stream SID packets when you send all
> very low energy (no energy)
>      Careful analysis of equation A-10 in G.723.1, Annex A will show this
> to be true.
>      b) Both G.729 and G.723.1 VAD algorithms were not adequately tested
> with lower levels of speech
>      The consequences are that an annoying "swooshing" sound occurs when
> the VAD misdetects
>      speech as nonspeech and codes a SID.
>      c) The DTX decision logic is flawed.  Low levels of non-stationary
> noise (power hum) can almost
>      completely defeat the intended packet rate reduction during nonspeech
> periods.
> If there was to be a standard for a VAD/DTX/CNG scheme for G.711, it should
> be mindful of these flaws.
>
> 3) Should there be a similar VAD for G.726 as well?  As  we begin to
> understand that VoIP does not really
> mean "Internet" telephony, it means Internet Protocol Telephony, we see
> that people are moving away from the
> low bit rate vocoders because they are not robust to ambient or non-speech
> sounds, they do not transport DTMF
> they degrade in speakerphone calls, and they do not tandem well at all
> (conferencing). G.726 is low
> complexity and half the bit rate of G.711.  No, it is not 6.4 or 8.0 kbps,
> but I don't believe that people will put up
> with the poor robustness of these speech coders unless they get something
> for it in return (like mobility as in
> digital cell phones).
>
> Kevin Bruemmer
>
> Stephen Casner <casner@cisco.com> on 11/03/99 02:39:23 PM
>
> To:   Robert Zopf <zopf@lucent.com>
> cc:   wp3audio@ctd.comsat.com, rem-conf@es.net (bcc: Kevin Bruemmer)
> Subject:  Re: CNG
>
> I thank Rob Zopt for his message on Comfort Noise Generator (CNG)
> enhancements.  Let me try to provide a bit of context for the AVT
> working group.
>
> Rob contacted me as a result of the working group last call
> announcement on advancing the revision of the RFC 1890 RTP Audio/Video
> Profile to Draft Standard.  One of the additions in that revision is a
> Comfort Noise (CN) payload format which is a simple one-octet message
> specifying the noise level.  Rob asked whether changes to the comfort
> noise specification would still be possible at this point, and if so,
> what form that should take.
>
> I had seen the proposal on the ITU SG16 mailing list to provide an
> enhanced CNG specification which included spectral shape in addition
> to level.  In an earlier reply to messages from Rob and others about
> this proposal, I said that if the CN payload format/encoding as
> defined in the current A/V Profile draft is not useful as it stands,
> then it should be replaced with something that is useful before the
> draft revision of RFC 1890 is published as a new RFC.
>
> I agree with Rob that it would not be a good idea to add comfort noise
> functions to a modified version of the PCMU/A (G.711) payload format
> because that would likely impair interoperability and would require
> similar revisions to other payload formats to add comfort noise to
> them.
>
> Options include:
>
>   - Extending the current CN defintion as Rob described in his message.
>     The current "energy only" specification would still be valid, but
>     additional spectral information could be added.  Receivers would
>     know from the payload length whether the additional info was
>     present.  Does this seem like a "safe" extension that would not
>     cause problems for existing implementations of CN as in the draft?
>     Note that CN has not been published in an RFC, but it has been in
>     the sequence of draft revisions for at least two years.
>
>   - Leaving the existing CN as is, but adding another CNG payload
>     format for the enhanced generator.  This would avoid any
>     compatibility issues, and would provide an opportunity to express
>     the noise energy in a different way.  However, in H.323 it would
>     require negotiation of both CN and CNG payload formats for
>     backward compatibility with endpoints that only implement the
>     existing CN.
>
> For either of these options, there is a question of whether it is
> possible to change the comfort noise specification in the revised
> profile draft, and if so, how.  The answer is certainly yes because
> the working group can decide that this function should be addressed
> before the draft is submitted to the IESG for IETF-wide Last Call.
>
> On the other hand, the advancement of RTP to Draft Standard is long
> overdue, so we'd like to avoid introducing additional delay.  There
> may be a compromise: ask for IESG Last Call on the main RTP spec, but
> hold off on the profile.  The RTP spec cannot be published as an RFC
> until the profile is also ready, but my concern is that the RTP spec
> is large and somewhat significant and therefore the IESG review may
> take some time.  I'd like to get that process started.
>
> A question for the ITU folks is how much time would be required to
> allow the enhanced CNG proposal to be completed and tested by the
> speech experts to make sure it works correctly before we standardize
> it.  This notion is well accepted within IETF.
>
> If the consensus is that a new CNG payload format should be added
> (leaving the existing CN unchanged), then it would be possible for the
> new payload format to be defined in a separate RFC just as we expect
> the continuing development of additional payload format RFCs for new
> audio and video encodings.  Perhaps we could/should add words to the
> CN definition in the current draft to say that enhanced comfort noise
> payload formats are in development and will be published in future
> RFCs.
>
> I'd like to discuss these questions as part of the review of the RTP
> spec and profile status in the AVT meeting at IETF next Wednesday.
> Rob will be attending to participate in that discussion.
>
>                                    -- Steve




From rem-conf Thu Nov 04 09:50:20 1999 
From rem-conf-request@es.net Thu Nov 04 09:50:19 1999
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Subject: 11/10 Application Outsourcing: The Next Big Thing on the
  Internet -- Peter Newman, Ensim Corporation 
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Berkeley Multimedia, Graphics and Interfaces Seminar

Application Outsourcing: The Next Big Thing on the Internet

Wednesday, November 10, 1999, 
1:00-2:30 p.m. PST
Fujitsu Seminar Room (405 Soda Hall) 


Peter Newman 
Ensim Corporation 

The current model for application distribution is deeply flawed. Customers
are forced to keep track of the latest version of each software, to size
their desktop machines to have sufficient power to deal with anticipated
application load, and to painfully recover from hardware failures. These
problems disappear overnight when applications are outsourced to
Application Service Providers (ASPs). Potentially, the shift of
applications from desktops to ASPs is as powerful and disruptive as the
introduction of the World Wide Web. However, ASPs too face severe scaling
problems in providing outsourced services to tens of thousands of
customers. In this talk, we will outline these problems and describe a
novel approach to solving them.

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

The seminar is broadcast live on the Internet Mbone and as a RealNetworks
G2 broadcast. You can connect to the live RealNetworks broadcast at:
http://media2.bmrc.berkeley.edu/bibs/schedule.cfm 

You'll see a link to cs298-5/real networks/live programs.  

You can also directly put in this url into the Real Player:
rtsp://media2.bmrc.berkeley.edu/encoder/cs298-5.rm
To do so you will need to have the "Real Player G2" installed, which is
available from:
http://www.real.com/products/player/downloadrealplayer.html

To tune into the Internet MBone broadcast, look for the announcement in
your MBone session directory program ('sdr').  If you are not receiving the
announcement you may be able to receive the session by manually configuring
the client programs ('vic', and 'vat') with the session addresses:

low bit rate
	video: 233.0.25.1/22334
	audio: 233.0.25.2/22446

medium bit rate
	video: 233.0.25.129/22334
	audio: 233.0.25.2/22446

You can get further information about this seminar, and access to replays
of previous seminars at the MIG Seminar web page:
http://bmrc.berkeley.edu/298/index.html

Sponsored by the Berkeley Multimedia Research Center 




From rem-conf Thu Nov 04 11:57:02 1999 
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From: Kevin_Bruemmer@nmss.com
To: Richard V Cox <rvc@research.att.com>
cc: casner@cisco.com,
	Robert Zopf <zopf@lucent.com>,
	wp3audio@ctd.comsat.com,
	rem-conf@es.net
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Richard:

Perhaps I didn't do a good enough job in the first part of my email.  The
"CN" type, as it exists in RTP/RTCP is ill-conceived - not the notion of
comfort noise.
As far as DCME, you are perhaps more aware of how much this equipment is
used anymore; my impression is that it is not.  And - since the
implementations are proprietary and each end of the equipment is designed
by the same people, the notion of specifying it in a standard, doesn't
apply.


Kevin




Richard V Cox <rvc@research.att.com> on 11/04/99 11:49:45 AM

To:   Kevin Bruemmer
cc:   casner@cisco.com, Robert Zopf <zopf@lucent.com>,
      wp3audio@ctd.comsat.com, rem-conf@es.net
Subject:  Re: CNG




Comfort Noise has been used in Circuit Multiplication Equipment for
years.  Its precise implementation is left up to vendors of the
equipment.  I believe they just supply the noise level and insert
white noise, but since this information is not public, I can only
speculate.  Listening to speech over a loudspeaker vs on a handset
is what distinguishes whether CN is needed.  In an office like
mine, with lots of fan noise from computers and AC/heating system,
I would not need CN for listening to speech on a loudspeaker
unless the background noise level at the other end was very
high.  On the other hand, I believe that on a handset I would
draw the opposite conclusion unless the background noise
was very low at the other end.
    The issue of the VADs in G.723.1 and G.729 not working
well for low level speech is immaterial as to whether CN is
needed.  If they really don't work well enough for real world
conditions, they would need to be improved.
    So, put me down as being in favor of CN generally.
        Rich Cox

Kevin_Bruemmer@nmss.com wrote:

> I would like to make three points concerning CN:
>
> 1) Who has ever implemented the CN in a real system? Whether you
completely
> squelch audio
> in between talk spurts or you add in an artificial static representation
of
> white or colored noise makes
> no difference on audio quality.  CN was and is an ill-conceived notion.
> Holding on to it for legacy reasons
> when nobody will likely defend it use doesn't make sense.  I think that
the
> CN type should be redefined.
>  Anyone who does use it - please respond and defend its use.
>
> 2) One of the respondants to this string indicates that the current
> VAD/DTX/CNG in G.729 and G.723.1
> work pretty well.  While they seem to work ok under nominal conditions,
> they suffer from some flaws:
>      a) The g.723.1 DTX scheme will stream SID packets when you send all
> very low energy (no energy)
>      Careful analysis of equation A-10 in G.723.1, Annex A will show this
> to be true.
>      b) Both G.729 and G.723.1 VAD algorithms were not adequately tested
> with lower levels of speech
>      The consequences are that an annoying "swooshing" sound occurs when
> the VAD misdetects
>      speech as nonspeech and codes a SID.
>      c) The DTX decision logic is flawed.  Low levels of non-stationary
> noise (power hum) can almost
>      completely defeat the intended packet rate reduction during
nonspeech
> periods.
> If there was to be a standard for a VAD/DTX/CNG scheme for G.711, it
should
> be mindful of these flaws.
>
> 3) Should there be a similar VAD for G.726 as well?  As  we begin to
> understand that VoIP does not really
> mean "Internet" telephony, it means Internet Protocol Telephony, we see
> that people are moving away from the
> low bit rate vocoders because they are not robust to ambient or
non-speech
> sounds, they do not transport DTMF
> they degrade in speakerphone calls, and they do not tandem well at all
> (conferencing). G.726 is low
> complexity and half the bit rate of G.711.  No, it is not 6.4 or 8.0
kbps,
> but I don't believe that people will put up
> with the poor robustness of these speech coders unless they get something
> for it in return (like mobility as in
> digital cell phones).
>
> Kevin Bruemmer
>
> Stephen Casner <casner@cisco.com> on 11/03/99 02:39:23 PM
>
> To:   Robert Zopf <zopf@lucent.com>
> cc:   wp3audio@ctd.comsat.com, rem-conf@es.net (bcc: Kevin Bruemmer)
> Subject:  Re: CNG
>
> I thank Rob Zopt for his message on Comfort Noise Generator (CNG)
> enhancements.  Let me try to provide a bit of context for the AVT
> working group.
>
> Rob contacted me as a result of the working group last call
> announcement on advancing the revision of the RFC 1890 RTP Audio/Video
> Profile to Draft Standard.  One of the additions in that revision is a
> Comfort Noise (CN) payload format which is a simple one-octet message
> specifying the noise level.  Rob asked whether changes to the comfort
> noise specification would still be possible at this point, and if so,
> what form that should take.
>
> I had seen the proposal on the ITU SG16 mailing list to provide an
> enhanced CNG specification which included spectral shape in addition
> to level.  In an earlier reply to messages from Rob and others about
> this proposal, I said that if the CN payload format/encoding as
> defined in the current A/V Profile draft is not useful as it stands,
> then it should be replaced with something that is useful before the
> draft revision of RFC 1890 is published as a new RFC.
>
> I agree with Rob that it would not be a good idea to add comfort noise
> functions to a modified version of the PCMU/A (G.711) payload format
> because that would likely impair interoperability and would require
> similar revisions to other payload formats to add comfort noise to
> them.
>
> Options include:
>
>   - Extending the current CN defintion as Rob described in his message.
>     The current "energy only" specification would still be valid, but
>     additional spectral information could be added.  Receivers would
>     know from the payload length whether the additional info was
>     present.  Does this seem like a "safe" extension that would not
>     cause problems for existing implementations of CN as in the draft?
>     Note that CN has not been published in an RFC, but it has been in
>     the sequence of draft revisions for at least two years.
>
>   - Leaving the existing CN as is, but adding another CNG payload
>     format for the enhanced generator.  This would avoid any
>     compatibility issues, and would provide an opportunity to express
>     the noise energy in a different way.  However, in H.323 it would
>     require negotiation of both CN and CNG payload formats for
>     backward compatibility with endpoints that only implement the
>     existing CN.
>
> For either of these options, there is a question of whether it is
> possible to change the comfort noise specification in the revised
> profile draft, and if so, how.  The answer is certainly yes because
> the working group can decide that this function should be addressed
> before the draft is submitted to the IESG for IETF-wide Last Call.
>
> On the other hand, the advancement of RTP to Draft Standard is long
> overdue, so we'd like to avoid introducing additional delay.  There
> may be a compromise: ask for IESG Last Call on the main RTP spec, but
> hold off on the profile.  The RTP spec cannot be published as an RFC
> until the profile is also ready, but my concern is that the RTP spec
> is large and somewhat significant and therefore the IESG review may
> take some time.  I'd like to get that process started.
>
> A question for the ITU folks is how much time would be required to
> allow the enhanced CNG proposal to be completed and tested by the
> speech experts to make sure it works correctly before we standardize
> it.  This notion is well accepted within IETF.
>
> If the consensus is that a new CNG payload format should be added
> (leaving the existing CN unchanged), then it would be possible for the
> new payload format to be defined in a separate RFC just as we expect
> the continuing development of additional payload format RFCs for new
> audio and video encodings.  Perhaps we could/should add words to the
> CN definition in the current draft to say that enhanced comfort noise
> payload formats are in development and will be published in future
> RFCs.
>
> I'd like to discuss these questions as part of the review of the RTP
> spec and profile status in the AVT meeting at IETF next Wednesday.
> Rob will be attending to participate in that discussion.
>
>                                    -- Steve









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Date: Thu, 04 Nov 1999 18:49:52 -0500
To: Kevin_Bruemmer@nmss.com, casner@cisco.com
From: Robert Zopf <zopf@lucent.com>
Subject: Re: CNG
Cc: wp3audio@ctd.comsat.com, rem-conf@es.net
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See my comments below.

At 10:45 AM 11/04/1999 -0500, Kevin_Bruemmer@nmss.com wrote:


>I would like to make three points concerning CN:
>
>1) Who has ever implemented the CN in a real system? Whether you completely
>squelch audio
>in between talk spurts or you add in an artificial static representation of
>white or colored noise makes
>no difference on audio quality.  CN was and is an ill-conceived notion.
>Holding on to it for legacy reasons
>when nobody will likely defend it use doesn't make sense.  I think that the
>CN type should be redefined.
>  Anyone who does use it - please respond and defend its use.

I too interpreted the above as CNG in general, but now from your latest 
email I understand that you mean the CN as it exists in the profile draft.

Your question then is similar to Steve's, so we now have a two part 
question that is very relevant :  Does anyone make use of the CN today as 
it is defined in the draft profile, and if so, would the proposed extension 
cause any problems ?

So far, it seems the answer to both of parts is no.


>2) One of the respondants to this string indicates that the current
>VAD/DTX/CNG in G.729 and G.723.1
>work pretty well.  While they seem to work ok under nominal conditions,
>they suffer from some flaws:
>      a) The g.723.1 DTX scheme will stream SID packets when you send all
>very low energy (no energy)
>      Careful analysis of equation A-10 in G.723.1, Annex A will show this
>to be true.
>      b) Both G.729 and G.723.1 VAD algorithms were not adequately tested
>with lower levels of speech
>      The consequences are that an annoying "swooshing" sound occurs when
>the VAD misdetects
>      speech as nonspeech and codes a SID.
>      c) The DTX decision logic is flawed.  Low levels of non-stationary
>noise (power hum) can almost
>      completely defeat the intended packet rate reduction during nonspeech
>periods.
>If there was to be a standard for a VAD/DTX/CNG scheme for G.711, it should
>be mindful of these flaws.

I agree that this VAD causes some artifacts.  However, I made this 
reference as a comparison for the quality of CNG that may be 
acceptable.  If the G.729 VAD/DTX operates correctly, the comfort noise 
produced by the CNG algorithm is adequate.  Remember, the proposal is for a 
means to communicate the CN information, not the VAD/DTX used.


>3) Should there be a similar VAD for G.726 as well?  As  we begin to
>understand that VoIP does not really
>mean "Internet" telephony, it means Internet Protocol Telephony, we see
>that people are moving away from the
>low bit rate vocoders because they are not robust to ambient or non-speech
>sounds, they do not transport DTMF
>they degrade in speakerphone calls, and they do not tandem well at all
>(conferencing). G.726 is low
>complexity and half the bit rate of G.711.  No, it is not 6.4 or 8.0 kbps,
>but I don't believe that people will put up
>with the poor robustness of these speech coders unless they get something
>for it in return (like mobility as in
>digital cell phones).

Again, the VAD that is used is left to the implementation.  The redefined 
CN could certainly be used with G.726 as well.


Rob.








>Stephen Casner <casner@cisco.com> on 11/03/99 02:39:23 PM
>
>To:   Robert Zopf <zopf@lucent.com>
>cc:   wp3audio@ctd.comsat.com, rem-conf@es.net (bcc: Kevin Bruemmer)
>Subject:  Re: CNG
>
>
>
>
>I thank Rob Zopt for his message on Comfort Noise Generator (CNG)
>enhancements.  Let me try to provide a bit of context for the AVT
>working group.
>
>Rob contacted me as a result of the working group last call
>announcement on advancing the revision of the RFC 1890 RTP Audio/Video
>Profile to Draft Standard.  One of the additions in that revision is a
>Comfort Noise (CN) payload format which is a simple one-octet message
>specifying the noise level.  Rob asked whether changes to the comfort
>noise specification would still be possible at this point, and if so,
>what form that should take.
>
>I had seen the proposal on the ITU SG16 mailing list to provide an
>enhanced CNG specification which included spectral shape in addition
>to level.  In an earlier reply to messages from Rob and others about
>this proposal, I said that if the CN payload format/encoding as
>defined in the current A/V Profile draft is not useful as it stands,
>then it should be replaced with something that is useful before the
>draft revision of RFC 1890 is published as a new RFC.
>
>I agree with Rob that it would not be a good idea to add comfort noise
>functions to a modified version of the PCMU/A (G.711) payload format
>because that would likely impair interoperability and would require
>similar revisions to other payload formats to add comfort noise to
>them.
>
>Options include:
>
>   - Extending the current CN defintion as Rob described in his message.
>     The current "energy only" specification would still be valid, but
>     additional spectral information could be added.  Receivers would
>     know from the payload length whether the additional info was
>     present.  Does this seem like a "safe" extension that would not
>     cause problems for existing implementations of CN as in the draft?
>     Note that CN has not been published in an RFC, but it has been in
>     the sequence of draft revisions for at least two years.
>
>   - Leaving the existing CN as is, but adding another CNG payload
>     format for the enhanced generator.  This would avoid any
>     compatibility issues, and would provide an opportunity to express
>     the noise energy in a different way.  However, in H.323 it would
>     require negotiation of both CN and CNG payload formats for
>     backward compatibility with endpoints that only implement the
>     existing CN.
>
>For either of these options, there is a question of whether it is
>possible to change the comfort noise specification in the revised
>profile draft, and if so, how.  The answer is certainly yes because
>the working group can decide that this function should be addressed
>before the draft is submitted to the IESG for IETF-wide Last Call.
>
>On the other hand, the advancement of RTP to Draft Standard is long
>overdue, so we'd like to avoid introducing additional delay.  There
>may be a compromise: ask for IESG Last Call on the main RTP spec, but
>hold off on the profile.  The RTP spec cannot be published as an RFC
>until the profile is also ready, but my concern is that the RTP spec
>is large and somewhat significant and therefore the IESG review may
>take some time.  I'd like to get that process started.
>
>A question for the ITU folks is how much time would be required to
>allow the enhanced CNG proposal to be completed and tested by the
>speech experts to make sure it works correctly before we standardize
>it.  This notion is well accepted within IETF.
>
>If the consensus is that a new CNG payload format should be added
>(leaving the existing CN unchanged), then it would be possible for the
>new payload format to be defined in a separate RFC just as we expect
>the continuing development of additional payload format RFCs for new
>audio and video encodings.  Perhaps we could/should add words to the
>CN definition in the current draft to say that enhanced comfort noise
>payload formats are in development and will be published in future
>RFCs.
>
>I'd like to discuss these questions as part of the review of the RTP
>spec and profile status in the AVT meeting at IETF next Wednesday.
>Rob will be attending to participate in that discussion.
>
>                                    -- Steve
>
>
>
>
>


~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Rob Zopf
Principal Engineer
Speech Technology Group
Lucent Technologies, INS

Phone 	: (732) 578-3207
Fax		: (732) 578-3213
Email		: zopf@lucent.com
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~



From rem-conf Thu Nov 04 15:52:16 1999 
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>
>
>A question for the ITU folks is how much time would be required to
>allow the enhanced CNG proposal to be completed and tested by the
>speech experts to make sure it works correctly before we standardize
>it.  This notion is well accepted within IETF.


The target is to have an appendix in February, but the schedule needs to be 
worked out.  It may take longer.

Rob.




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The primary need for a CN or CNG packet is for G711 which has (even for
5ms) a relatively large voice packet.  Therefore, even a relatively large
CN packet would allow a significant data reduction.  This means that, after
the energy info, a large type or version field could be attached allowing
a variety of algorithms, each more complex than the one before, to be
announced without significantly diiminshing the usefulness of the packet
for
data reduction.  It seems to me that this means we don't have to agonize
too much or too long on the issue, just come up with one or two relatively
simple algorithms of a spectral nature that some early adopters can agree
on and start reaping the data reduction benefits.





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From rem-conf Sat Nov 06 19:09:03 1999 
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Dear colleagues,

I would like to answer a few questions related to CNG.

1) Why ONLY a noise energy level mechanism has been
so far proposed as CNG for G.711 (both ATM and IP)?

Simpler and more complex VAD and CNG algorithms have
been used for a long time, so clearly not a lack of
of knowledge in the field prevented more sophisticated
mechanism to be adopted.
a/u-law PCM coders are the most popular in the world.
The a/u-law PCM coders are very cheap and no royalties
have to be paid by the end user.
The use of sophisticated speech compression algorithms
(for example those in the ITU-T series G.72x) has been
limited to applications where bandwidth was at a high
premium. Even so , their use has been somehow limited
by the the higher cost to the end user and the
reduced capability of transporting tones (DTMF, modem,
etc.). More recent speech compression algorithms have
been adopted with their own VAD and CNG mechanism and
the ratio between the computational power required by
the coder and that required by the VAD/CNG is around 5:1
or higher.
To implement a PCM to linear decoder or a linear to
PCM coder requires around 30 cycles/frame on a popular
C5x machine. This is roughly equivalent to 0.48 MIPS
for a full Compander. Some DSPs have the capability to do
these operations in 1 cycle (for example Mitel's Miliwat
DSP core).
To implement a VAD+CNG mechanism that requires around
2 MIPS leads us to the abberation where the ratio between
the coder CPU requirements and the VAD/CNG requirements
are around 1:4 or worst!!
If someone has 2 MIPS to spare probably has 6 as well.
And if 6 MIPS are available, why wouldn't one use a toll
quality 32 kbps ADPCM algorithm with no silence supression ?
RFC 1889 not only allows ADPCM, but it allows for changing
coders on the fly to accomodate various bandwidth needs.
So, now we see that a simple VAD/CNG was used simply
because it made a lot of sense to use something on the same
degree of complexity (i.e. low) with the PCM coders.

2) Does anybody use the CN level cell/packet mechanism ?
Yes, we do.
We support the existent draft-ietf-avt-profile-new-07.txt
document and we want to see it approved as an RFC without
any delay.

3) How about a more sophisticated CN, for example one based
on noise spectral information ?
We believe there is no need for standardizing such an algorithm
neither by IETF or ITU-T.
However, if enough support for the adoption of such an algorithm
is voiced at the next ITU-T meeting in Geneva, we are prepared
to participate in the activites for defining the terms of
reference of a CNG algorithm for G.711 as well as  making proposals
for a candidate algorithm.
I would be suprised if we would see at ITU-T just two proposals.

4) How fast should the work be done ?
Well , there is no question dealing with CN for G.711 .
I believe G.711 is too important for everybody. We need
a separate question under SG16.
The ITU-T procedures for standardization are strict and well
defined. In this particular case many companies may have a high
interest in following the work and contribuing to it.
Work can be progressed through reflector groups but the
Telecommuncation community cannot be properly informed without
a general meeting in Geneva.
We want to make sure that everybody in this world using PCM coders
has a chance to take notice of the new work proposed and a chance
to contribute.

Regards,
Alexander Tulai
Mitel Corporation
Ph: (613)592-2122
Fax: (613) 592-4784
alexander_tulai@mitel.com







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[sorry for any duplication of this message]

Dear Colleague:

Please find below a call for chapters on Multimedia Systems.

I invite you to submit a chapter.

Please also circulate the CFC to your colleagues and network.

Thanks for your cooperation.

Mahbubur Rahman Syed


--------------------------><----------------------------------

                 CALL FOR BOOK CHAPTERS

Submissions are invited for inclusion in a forthcoming book:

   "Design and Management of Multimedia 
    Information Systems: Opportunities and Challenges"

to be published by Idea Group Publishing, Hershey, USA, in 2000.

Editor:  
Syed. M. Rahman, North Dakota State University, USA
(on secondment from Monash University, Australia)

The objective of this book is to compile together the state-of-the-art
research, design, development, implementation and management experiences
in multimedia systems both in stand alone and distributed environment. 
This book is aimed to promote the exchange of information and ideas 
between academia and industry, faculty, practitioners and students. 
It should also be useful to students as a text for courses in the area 
of multimedia systems. 

It will include chapters from professionals, researchers and the 
business community. 

Information for authors, including topic areas, chapter organisation
guidelines and the submission procedure may be found at the 
following locations:

http://venus.ece.ndsu.nodak.edu/~msyed/mmbook/index.html
http://www.gscit.monash.edu.au/~rahman/mmbook/index.html

                    IMPORTANT DATES
Expression of Interest/proposal by:   15 December 1999
Feedback on the proposal:             22 December 1999
Full Chapter due by:                  28 February 2000
Notification of Review Status:  	    1 May 2000
Revised Chapter due by:               5 June 2000 

You may send email queries to:

syed.rahman@infotech.monash.edu.au
msyed@badlands.nodak.edu





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        itc@fokus.gmd.de, itc@ieee.org, kia@usl.edu,
        MariaLuisa.Rossari@cselt.it, members@hpovua.org, members@tmforum.org,
        mobile-ip@tadpole.com, modern-heuristics@uk.ac.mailbase.ucdavis.edu,
        multicomm@cc.bellcore.com, multicomm@research.panasonic.com,
        nm-australia@amazon.postech.ac.kr, nmkorea@amazon.postech.ac.kr,
        Omar.Elloumi@enst-bretagne.fr, opensig-announce@ctr.columbia.edu,
        osimcast@bbn.com, performance@haven.epm.ornl.gov, prs@masi.ibp.fr,
        qos-iso@mci.org.uk, qosws@dstc.edu.au, rem-conf@es.net, reres@laas.fr,
        s.low@ee.mu.oz.au, sb.all@ieee.org, sc6wg4@ntd.comsat.com,
        sig-dce@opengroup.org, sigmedia@bellcore.com, tccc@ieee.org,
        tchen@seas.smu.edu, testnet@canarie.ca, tm_member@ns.ieice.or.jp,
        wwlu@ieee.org, xtprelay@cs.concordia.ca, lyko@kt.agh.edu.pl
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You can see the Final Program and to register on-line via Web.

**************************************************************
*       You are very welcome to participate in the           *
*                                                            *
*                      IEEE LANOMS99                         *
*                                                            *
* LATIN AMERICAN NETWORK OPERATIONS AND MANAGEMENT SYMPOSIUM *
*                                                            *
*            http://www.lrg.ufsc.br/~lanoms99/               *
*                                                            *
*                   Rio Atlantica Hotel                      *
*                Rio de Janeiro - Brazil                     *
*                   December 3-5, 1999                       *
**************************************************************




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Subject: CFP: Conference on High Performance Switching Routing in June 2000
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My sincere apology if you receive multiple copies of this CFP.
Please feel free to pass the CFP to anyone who might be interested.

Kind regards,
Sibille GralkaCall For Papers

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

Conference on High Performance Switching & Routing

Joint

IEEE ATM Workshop 2000 and
3rd International Conference on ATM (ICATM'2000)

June 26-29, 2000
Crowne Plaza Hotel, Heidelberg, Germany

Sponsored by the IEEE Communication Society

Supported by:
VDE, EC Information Society Technologies, EURESCOM GmbH,
Alcatel SEL AG, Cisco Systems GmbH, Deutsche Telekom AG,
Ericsson Eurolab Deutschland GmbH,  IBM Deutschland GmbH,
Lucent Technologies GmnH, NEC Europe Ltd., Siemens AG

For a nicely formatted version of this Call for Papers as
well as for any other information related to this event
please check the conference website under

URL: http://www.ccrle.nec.de/heidelberg/atm2000


Conference Scope

A wide variety of ATM products have been developed for local
and wide-area, private and public networks. First-generation
ATM public services have been deployed on a global basis and
efforts are underway to implement full-service, large-scale
ATM networks. Wired (xDSL, HFC) and wireless (W-ATM) access
networks enabling end-to-end ATM services are becoming available.
Integration between ATM and IP evolves rapidly. Researchers and
practitioners are studying issues like network design, network
performance and traffic engineering, network reliability and
survivability, charging and tariffing, quality of service and
multicast support. At the same time next generation switching
and routing technologies including IP over SONET/SDH or IP over
WDM are being developed, converging cell and packet switching
architectures. Driving applications for integrated networks
become visible, namely Virtual Private Networks (VPN),
Voice over Packet (VOP) and group communication services like
conferencing and video on demand (VOD). They create new
requirements on network architecture and management.

The purpose of the conference is to share ideas, experiences,
and information among researchers, developers, and service
providers in the field of data, voice and multimedia
communications using ATM and other high-speed switching and
routing technologies, including Gigabit/Terabit switch/routers
for optical networks.

Original papers are hereby solicited on such topics as
(but not limited to):

1. ATM Networks
1.1. ATM/WDM Networks
1.2. Broadband Access Networks (xDSL,HFC,PON)
1.3. Wireless / Mobile ATM
1.4. Satellite-Based ATM
1.5. ATM and UMTS/IMT2000
1.6. Interworking

2. ATM Switching
2.1. ATM Switch Architectures
2.2. Large-Scale Switch Implementations and Performance
2.3. Broadcasting/Multicasting

3. Signaling and Control
3.1. Broadband Signaling
3.2. Large-scale Call Processing Architectures.
3.3. PNNI, I-PNNI
3.4. QoS Routing
3.5. Signaling Interworking (ATM, PSTN, VoIP,...)

4. Traffic Engineering
4.1. ATM Traffic Modeling
4.2. Cell and Packet Level Scheduling
4.3. UBR,VBR, ABR, CBR Performance
4.4. Traffic Engineering
4.5. Network Design and Dimensioning
4.6. Network Optimization

5. ATM/IP Integration
5.1. IP/ATM Integration: MPLS, MPOA, ...
5.2. IP Multicasting over ATM
5.3. IPv6 over ATM
5.4. TCP over ATM
5.5. IP over ATM vs. native IP

6. Integrated Services, Multimedia
6.1. Native ATM Applications and  Interfaces
6.2. QoS and CoS
6.3. IntServ and Diffserv
6.4. Video Coding and Transmission
6.5. VToA, VoP, VoIP
6.6. Multimedia Traffic Characteristics

7. Management and Control
7.1. Software Architectures for Switch/Router Control
7.2. Traffic Management Functions (UPC,CAC,...)
7.3. Tariffing, Charging and Accounting
7.4. Virtual Private Networks, VLANs,...
7.5. Network/VPN Security
7.6. Survivability, Fault Tolerance, Self-Healing,...
7.7. Network and Service Management
7.8. Service Provisioning

8. Gigabit/Terabit Routers
8.1. High-Speed Packet Switching and Routing
8.2. Switching & Routing for WDM
8.3. IP over SONET/SDH and IP over WDM
8.4. MPLS over WDM
8.5. Multicast Routing

9. Real User Networks
9.1. Operational Experience
9.2. User Experiences
9.3. Business Aspects
9.4. Network Evolution

10. Alternative Technologies
10.1. DTM
10.2. Photonic Switching
10.3. others


Instructions for Authors:

Send an electronic version of your submission to the address
below. Submissions should be extended abstracts (2000 - 3000
words) summarizing original work. All the manuscripts must be
written in English. The first page of each paper should contain
the title of the paper, the authors' name(s), affiliation,
address, telephone and fax numbers, e-mail of the author
responsible for correspondence,  a list of four keywords and
categories from the above list as well as a summary of up
to 100 words of the main achievements in your contribution.

Electronic submissions are strongly encouraged.
Acceptable formats are PDF, Postscript Level 2, RTF,
FrameMaker Vs.5,  Word 97. Please use A4 paper format
when formatting your submission!

Authors of accepted papers will later be required to submit
an IEEE copyright form, please assure that you have all
necessary authorizations in place. Typing instructions
for accepted full papers can be downloaded from
http://www.vde-verlag.de.

All submitted papers should be sent to the following address:

Dr. Heinrich J. St=FCttgen,
General Chair IEEE ATM Workshop 2000 & ICATM 2000
Computer and Communcation Research Laboratories
NEC Europe Ltd.
Adenauerplatz 6
D-69115 Heidelberg, Germany
Phone: +49 6221 905 11-0
Fax:     +49 6221 905 11-55
E-mail: ATM2000@ccrle.nec.de

Important Dates:

Submission of extended abstract due:             January 10th, 2000
(Ext. Abstracts of approx. 5 pages or 2500 words)

Authors notified:                                March 15th, 2000
Final camera ready papers due:                   April 17th, 2000
(Final papers of max. 5000 Words or 10 Pages)


Tutorials:

On Monday, June 26th, 2000 the workshop will begin with
two half day tutorials focusing on emerging technologies
in the areas of ATM, WDM Switching, IP over WDM, Gigabit Routing,
IP/ATM in UMTS/IMT2000, Multimedia Communication or related
subjects. Researchers interested in proposing/presenting a
tutorial at ATM 2000 should contact the workshop chair via email
under ATM2000@ccrle.nec.de to inquire for details.


Workshop Committees

General Chair (IEEE ATM Workshop)
Heinrich J. St=FCttgen, C&C Research Laboratories,
NEC Europe Ltd., Heidelberg, Germany

General Co-Chair (ICATM)
Pascal Lorenz, University Haute Alsace, Colmar, France

Technical Co-Chairs
Jonathan Turner, Washington University, St.Louis, USA
Naoaki Yamanaka, NTT Network System Laboratories, Tokyo, Japa


Organizing Committee

H. Besier, Deutsche Telekom AG, Germany
H. Brueggemann, EURESCOM GmbH, Germany
C. Carrelli, EURESCOM GmbH, Germany
W. Frohberg, Alcatel SEL AG, Germany
B. Jabbari, George Mason University, USA
P. Kuehn, Stuttgart University, Germany
R. Rompel (Treasurer), VDE, Germany
F. Sass, Siemens AG & ATM Forum, Germany
S. Schaller, NEC Europe Ltd., Germany


IEEE ATM Workshop Advisory Board

A. Casaca, IST/INESC, Portugal
G. Copeland, CSC, USA
J. P. Coudreuse, Mitsubishi, France
M. Decina, CEFRIEL, Italy
G. Dobrowski, Ficon Technologies, USA
D. Dorman, Nortel Networks, Australia
B. Goode, IBM, USA
R. Guerin, Univ. of Pennsylvania, USA
Y. Inoue, NTT, Japan
B. Jabbari, G.Mason University, USA
P. K=FChn, Univ. Stuttgart, Germany
A. Leon-Garcia, Univ, of Toronto, Canada
L. Mason, INRS-Telecom, Canada
G. Pujolle, Lab. PRISM, France
J. Roberts, France Telecom, France
M. Schwartz, Columbia University, USA
H. Stuettgen, NEC Europe, Germany
S. Suzuki, NTT, Japan
S. Tohme, ENST, France
R. Vickers, NORTEL, Canada
S. Walters, Telcordia, USA
S. Weinstein, NEC America, USA


Joint Technical Program Committee

D. Awduche, UUNET, USA
A. Baiocchi, Univ. Rome, Italy
K. Begain,  Mu'tah Univ., Jordan
A. Benslimane, Tech. Univ. of Belfort, France
B. Bing, University of Maryland, USA
C. Blondia, Univ. of Antwerp, Belgium
D. Boettle, Alcatel SEL, Germany
D. Bonjour, France Telecom CNET, France
T. Braun, Berne University,Switzerland
B. Butscher, GMD FOKUS, Germany
A. Choudhury, PMC-Sierra Inc., USA
O. Casals, Univ. Politecnica Catalonia, Spain
P. de Sousa, DG XIII, European Commission
J. Ebersp=E4cher, Tech.Univ Munich, Germany
W. Fischer, Cisco, Germany
K.D. Grohs, Deutsche Telekom AG, Germany
J. Hayes, Concordia University, Canada
R.G. Herrtwich, DaimlerChrysler, Germany
B. Hirosaki, NEC, Japan
Z. Hulicki, Univ. of Cracow, Poland
A. Jajszczyk, ITTI Ltd. and AGH, Poland
M. Karol, Bell Labs, USA
R. Keller, Ericsson Eurolab, Germany
U. Killat, TU Hamburg Harburg, Germany
D. Kofman, ENST, France
S. Komandur, Lucent Technologies, USA
S. Kumar, DARPA, USA
G.S. Kuo, National Central Univ., Taiwan
M.M. Lee, Dongshin University, Korea
F. Le Faucheur, Cisco, France
P. Lorenz, Univ. Haute Alsace, France
Z. Mammeri, IRIT/Univ. P.Sabatier, France
N. Mastorakis, Military Institute of Univ. Educ., Greece
P. Morreale, Stevens Inst. of Techn., USA
M. Murata, Osaka Univ., Japan
M. Nunes, IST/INESC, Portugal
G. Omidyar, CSC, USA
M. Pullen, G.Mason Univ., USA
M. Potts, Martel, Switzerland
S. Rao, Telscom AG, Switzerland
E. Rathgeb, Univ. Essen, Germany
G. Reali, Univ. di Perugia, Italy
S. Ritzenthaler, Newbridge, France
H. Saito, NTT, Japan
K. Sauer, Bosch AG, Germany
D. Serpanos, ICS FORTH, Greece
V. Trecordi, CEFRIEL, Italy
D. Tsang, Hong Kong University Hong Kong
P. van Mieghem, Delft University,Netherlands
R. Wille-Fier, Siemens AG, Germany
L. Wolf, Tech. Univ. Darmstadt, Germany
A. Wolisz, Tech. Univ. Berlin, Germany
F. Yegenoglu, COMSAT Labs, USA
M. Zitterbart, TU Braunschweig, Germany




From rem-conf Mon Nov 08 06:25:10 1999 
From rem-conf-request@es.net Mon Nov 08 06:25:10 1999
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TOOLS Europe 2000
   "Enterprise Architecture - Patterns - Components"

Mont St Michel & St Malo, France
    June 5-8, 2000

http://www.tools.com/europe


CALL FOR CONTRIBUTIONS (deadline 31st January 2000)

(Please accept our apologies if you receive more than one copy. Please
forward this message to colleagues you think might be interested.)

TOOLS is the major international conference series devoted to
applications of object-oriented technology.  TOOLS Europe 2000 will be
held in Le Mont St Michel, at the borderline of Normandy and Brittany
in the north-west of France, and will continue the commitment to
excellence of earlier TOOLS conferences in Europe, Australia, Asia and
the USA.

The proceedings will be published world-wide by the IEEE Computer
Society.


PAPERS

TOOLS Europe 2000 is now soliciting papers on all aspects of
object-oriented technology. All submitted papers will be refereed and
assessed for technical quality and usefulness to practitioners and
applied researchers.

TOOLS Europe particularly welcomes papers that present general
findings based upon industrial experience. Such papers will be judged
by the quality of their contribution to industrial best-practice
(rather than to the body of research knowledge).


TUTORIALS and WORKSHOPS

Tutorials and workshops form an important part of TOOLS
conferences. TOOLS Europe 2000 is keen to receive proposals for
tutorials, workshops and panels on topics.


MORE DETAILS

Please visit the conference website for more details.
http://www.tools.com/europe





From rem-conf Mon Nov 08 19:25:59 1999 
From rem-conf-request@es.net Mon Nov 08 19:25:58 1999
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Date: Mon, 8 Nov 1999 19:18:23 -0800 (Pacific Standard Time)
From: Stephen Casner <casner@cisco.com>
To: Bruce_Levens@nmss.com
cc: rem-conf@es.net
Subject: Re: CNG
In-Reply-To: <85256820.004D4AFE.00@notes4.nmss.com>
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On Fri, 5 Nov 1999 Bruce_Levens@nmss.com wrote:

> The primary need for a CN or CNG packet is for G711 which has (even for
> 5ms) a relatively large voice packet.  Therefore, even a relatively large
> CN packet would allow a significant data reduction.

True.  The main value of VAD is not sending any packets at all, except
perhaps for a comfort noise packet of some kind at the start of
silence and occasionally during the silence.

>  This means that, after
> the energy info, a large type or version field could be attached allowing
> a variety of algorithms, each more complex than the one before, to be
> announced without significantly diiminshing the usefulness of the packet
> for data reduction.

Here I don't agree.  Interoperability is a key requirement, and
allowing a plethora of algorithms would most likely impair
interoperability.  For H.323, one would probably need/want to
negotiate compatibility of the comfort noise payload format.
Negotiating the payload type to be used seems like the right way to do
that.  It would be harder to negotiate the embedded version value you
suggest.  If the requirement for negotiation is valid, then there
would be no additional freedom to create new methods via that path
than through dynamic RTP payload type binding to new algorithm
descriptors (in whatever for is used for H.323 negotiation).

							-- Steve




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All,

I would like to respond to Alexander's comments below in the context of the 
proposal on the table and the application it is intended for.
Please see my comments below.

At 09:52 PM 11/06/1999 -0500, Alexander Tulai wrote:

>Dear colleagues,
>
>I would like to answer a few questions related to CNG.
>
>1) Why ONLY a noise energy level mechanism has been
>so far proposed as CNG for G.711 (both ATM and IP)?
>
>More recent speech compression algorithms have
>been adopted with their own VAD and CNG mechanism and
>the ratio between the computational power required by
>the coder and that required by the VAD/CNG is around 5:1
>or higher.
>To implement a PCM to linear decoder or a linear to
>PCM coder requires around 30 cycles/frame on a popular
>C5x machine. This is roughly equivalent to 0.48 MIPS
>for a full Compander. Some DSPs have the capability to do
>these operations in 1 cycle (for example Mitel's Miliwat
>DSP core).
>To implement a VAD+CNG mechanism that requires around
>2 MIPS leads us to the abberation where the ratio between
>the coder CPU requirements and the VAD/CNG requirements
>are around 1:4 or worst!!
>If someone has 2 MIPS to spare probably has 6 as well.
>And if 6 MIPS are available, why wouldn't one use a toll
>quality 32 kbps ADPCM algorithm with no silence supression ?
>RFC 1889 not only allows ADPCM, but it allows for changing
>coders on the fly to accomodate various bandwidth needs.
>So, now we see that a simple VAD/CNG was used simply
>because it made a lot of sense to use something on the same
>degree of complexity (i.e. low) with the PCM coders.

 From a previous email, we have :

It is true that G.711 is very low complexity and
adding any additional processing needed in a channel
must be highly scrutinized.  However, in real-world VoIP
systems each "Channel" is actually running much more
than G.711.  In fact listed below is what typically runs
within each G.711 channel with estimates of actual DSP
complexity.

0 - 1 MIPS  -- G.711 (HW/SW)
1 - 3 MIPS  -- Fax/Modem Detect
1- 3 MIPS -- DTMF Detect/Regeneration
7 - 15 MIPS -- G.165/G.168 LEC
0.25 - 2 MIPS -- VAD/CNG
0.25 - 3 MIPS -- Packet Loss Recovery
----------------------------------------------------------
10 MIPS - 27 MIPS


Therefore, if one ELECTED to incorporate an enhanced 1 MIP CNG
over what exists today within RFC 1890 it would only take between
4% - 10% of the complexity required to run an actual G.711 Channel.

So the computational ratio is actually between 10:1 and 25:1 for the 
application that is of concern.

The second issue is why would someone use G.711 with VAD/CNG instead of G.726 ?

One must consider the application, that being a voice call over H.323 using 
an RTP/UDP/IP stack.  In this situation, the header overhead is generally 
40 bytes. Because of the overhead, the transmission rate plays a larger 
role in overall system bit-rate than the codec bit-rate.  One must also 
consider the network over which the data is sent.  Below is a table 
illustrating the effective bit rate over an ATM link (assuming 48 byte cell 
and 5 byte header for simplicity) for G.711 with/without VAD-CNG and 
G.726.  This is only one example, but it does illustrate the benefits.

Parameters : frame size = 5ms, Voice Activity Factor = 0.6 (very realistic 
for a two-way call), CNG update rate = 200ms, CNG payload size=6 bytes.

Codec   Frames/Packet           Bit-Rate (No Vad)       Bit-Rate (Vad)

G.711           1                       169600                  102608
                 2                       127200                  77168

G726            1                       169600
                 2                       84800

For 1 frame/packet, both codecs require 2 ATM cells and the effective bit 
rate is the same.  For G.711 with Vad, we see significant bit-rate 
savings.  At 2 frames/packet, G726 provides a significant saving over G711 
without Vad, but G.711 with Vad-CNG still provides a savings over G.726.

Again, this is a simplified example, but it does show the value of using a 
VAD-CNG scheme.

In conclusion, when G.711 with VAD-CNG is considered in VoIP, it can 
provide significant bit-rate savings, better quality, and lower complexity 
compared to G.726.



>2) Does anybody use the CN level cell/packet mechanism ?
>Yes, we do.
>We support the existent draft-ietf-avt-profile-new-07.txt
>document and we want to see it approved as an RFC without
>any delay.

The use of the proposed "enhanced" CN payload is not mandatory, and 
defaults back to the existing definition.  If desired, the use of the 
enhancement  provides additional quality with little or no extra payload 
overhead, at a fractional increase in overall complexity, and remains 
generic to be used with any codec, not just G.711.  If Mitel has an RTP 
application in mind that uses only G.711 with a super fast VAD-DTX scheme 
and uses only energy for Comfort Noise, then it is still free to do 
so.  The proposal in no way changes this.

The enhanced CN payload has many advantages over the existing CN. There is 
still the opportunity for it to be part of the RTP profile, and it should 
be considered.


>3) How about a more sophisticated CN, for example one based
>on noise spectral information ?
>We believe there is no need for standardizing such an algorithm
>neither by IETF or ITU-T.
>However, if enough support for the adoption of such an algorithm
>is voiced at the next ITU-T meeting in Geneva, we are prepared
>to participate in the activites for defining the terms of
>reference of a CNG algorithm for G.711 as well as  making proposals
>for a candidate algorithm.
>I would be suprised if we would see at ITU-T just two proposals.

We look forward to proposals from all interested parties.


>4) How fast should the work be done ?
>Well , there is no question dealing with CN for G.711 .
>I believe G.711 is too important for everybody. We need
>a separate question under SG16.
>The ITU-T procedures for standardization are strict and well
>defined. In this particular case many companies may have a high
>interest in following the work and contribuing to it.
>Work can be progressed through reflector groups but the
>Telecommuncation community cannot be properly informed without
>a general meeting in Geneva.
>We want to make sure that everybody in this world using PCM coders
>has a chance to take notice of the new work proposed and a chance
>to contribute.

The issue was first brought up at the Q12-14/16 Rapporteur's meeting in 
Berlin in August 1999.  Here is a summary of the issue after discussion by 
the complete group of experts :

It was noted that RFC 1890 is ill-defined; it will be the second version of 
RFC 1890 that should be referred to. Its current status is pre-RFC (see 
"draft-ietf-avt-profile-new-06.ps").  It was noted that the IETF work is 
general to all sample based coders, while the proposal focused on 
G.711.  There may be a need for non-G.711 work.

It was agreed:

1) To communicate a liaison to the IETF indicating that an issue exists and 
that more work is needed.
2) To communicate our concerns to Q19.

The general goal would be to work with the RTP editors and not produce 
additional standards unless necessary.

A liaison statement was generated from this meeting and sent to Q19.  This 
liaison was considered at the Q19-21/16 Rapporteur's meeting in Geneva in 
Sept. 1999.  Here is the summary from the Q19 Rapporteur about the issue 
and what was decided by the group :

         Initiating study for CNG under DTX condition
Following the request by the Rapporteur Q.13/16, the meeting agreed to 
start study on the Comfort Noise Generator for the speech codec without 
built-in CNG such as G.711. A target of this work will be an Appendix to 
G.711 that defines the CNG function. The first step was to draft a terms of 
reference to define the CNG algorithm. The meeting formed an ad-hoc group 
with chairman of Mr. Robert Zopf (Lucent Technologies), which proposed a 
draft Performance Requirement s and Objectives in AC-99-46. With respect to 
the quality, the chairman of Q.14/12 (Mr. Mark Parkins, AT&T) kindly 
offered co-operation in defining the quality assessment method for coding 
speech under DTX/CNG operation during the interim period after the 
September meeting by correspondence.
The meeting agreed that the work should be proceeded in the correspondence 
group of chairmanship by Mr. Zopf under the meeting of Q.19/16, with help 
of Q.14/12 for quality issue.  The correspondence group should continue 
following works:
(1)     Complete the Performance Requirement s and Objectives
(2)     Draft preliminary test methodology and ask Q.14/12 for comments
(3)     Complete the Test Methodology
(4)     Collect candidate algorithms for CNG
(5)     Performance assessment for the candidates
(6)     Select a candidate and define the algorithm (Draft Appendix to G.711) .

The time schedule should be established as a part of Terms of Reference 
taking into account that the Appendix is strongly desired by WP2/16 and 
IETF as soon as possible. Expectation for the Appendix may be approval at 
the next SG16 meeting in February 2000.

The issue has been properly addressed within the ITU, and the work is 
following the path laid out at the meeting by the Q19 Rapporteur and the 
WP3 Chairman.

I am sorry to everyone for the long email.  I hope that the work can now 
progress in a timely fashion.

Best Regards,

Rob.

--=====================_194611276==_.ALT
Content-Type: text/html; charset="us-ascii"
Content-Transfer-Encoding: quoted-printable
X-MIME-Autoconverted: from 8bit to quoted-printable by drawbridge.ascend.com id UAA02031

<html>
All,<br>
<br>
I would like to respond to Alexander's comments below in the context of
the proposal on the table and the application it is intended for.<br>
Please see my comments below.<br>
<br>
At 09:52 PM 11/06/1999 -0500, Alexander Tulai wrote:<br>
<br>
<blockquote type=3Dcite cite>Dear colleagues,<br>
<br>
I would like to answer a few questions related to CNG.<br>
<br>
1) Why ONLY a noise energy level mechanism has been<br>
so far proposed as CNG for G.711 (both ATM and IP)?<br>
<br>
More recent speech compression algorithms have<br>
been adopted with their own VAD and CNG mechanism and<br>
the ratio between the computational power required by<br>
the coder and that required by the VAD/CNG is around 5:1<br>
or higher.<br>
To implement a PCM to linear decoder or a linear to<br>
PCM coder requires around 30 cycles/frame on a popular<br>
C5x machine. This is roughly equivalent to 0.48 MIPS<br>
for a full Compander. Some DSPs have the capability to do<br>
these operations in 1 cycle (for example Mitel's Miliwat<br>
DSP core).<br>
To implement a VAD+CNG mechanism that requires around<br>
2 MIPS leads us to the abberation where the ratio between<br>
the coder CPU requirements and the VAD/CNG requirements<br>
are around 1:4 or worst!!<br>
If someone has 2 MIPS to spare probably has 6 as well.<br>
And if 6 MIPS are available, why wouldn't one use a toll<br>
quality 32 kbps ADPCM algorithm with no silence supression ?<br>
RFC 1889 not only allows ADPCM, but it allows for changing<br>
coders on the fly to accomodate various bandwidth needs.<br>
So, now we see that a simple VAD/CNG was used simply<br>
because it made a lot of sense to use something on the same<br>
degree of complexity (i.e. low) with the PCM coders.<br>
</blockquote><br>
 From a previous email, we have :<br>
<br>
It is true that G.711 is very low complexity and<br>
adding any additional processing needed in a channel<br>
must be highly scrutinized.&nbsp; However, in real-world VoIP<br>
systems each &quot;Channel&quot; is actually running much more<br>
than G.711.&nbsp; In fact listed below is what typically runs<br>
within each G.711 channel with estimates of actual DSP<br>
complexity.<br>
<br>
0 - 1 MIPS&nbsp; -- G.711 (HW/SW)<br>
1 - 3 MIPS&nbsp; -- Fax/Modem Detect<br>
1- 3 MIPS -- DTMF Detect/Regeneration<br>
7 - 15 MIPS -- G.165/G.168 LEC<br>
0.25 - 2 MIPS -- VAD/CNG<br>
0.25 - 3 MIPS -- Packet Loss Recovery<br>
----------------------------------------------------------<br>
10 MIPS - 27 MIPS<br>
<br>
<br>
Therefore, if one ELECTED to incorporate an enhanced 1 MIP CNG<br>
over what exists today within RFC 1890 it would only take between<br>
4% - 10% of the complexity required to run an actual G.711 Channel.<br>
<br>
So the computational ratio is actually between 10:1 and 25:1 for the
application that is of concern.<br>
<br>
The second issue is why would someone use G.711 with VAD/CNG instead of
G.726 ?<br>
<br>
One must consider the application, that being a voice call over H.323
using an RTP/UDP/IP stack.&nbsp; In this situation, the header overhead
is generally 40 bytes. Because of the overhead, the transmission rate
plays a larger role in overall system bit-rate than the codec
bit-rate.&nbsp; One must also consider the network over which the data is
sent.&nbsp; Below is a table illustrating the effective bit rate over an
ATM link (assuming 48 byte cell and 5 byte header for simplicity) for
G.711 with/without VAD-CNG and G.726.&nbsp; This is only one example, but
it does illustrate the benefits.<br>
<br>
Parameters : frame size =3D 5ms, Voice Activity Factor =3D 0.6 (very
realistic for a two-way call), CNG update rate =3D 200ms, CNG payload
size=3D6 bytes.<br>
<br>
Codec<x-tab>&nbsp;&nbsp;&nbsp;</x-tab>Frames/Packet<x-tab>&nbsp;&nbsp;&nb=
sp;</x-tab><x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab=
>Bit-Rate
(No Vad)<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>Bit-Rate
(Vad)<br>
<br>
G.711<x-tab>&nbsp;&nbsp;&nbsp;</x-tab><x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;</x-tab>1<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;</x-tab><x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-t=
ab><x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>169600<=
x-tab>&nbsp;&nbsp;</x-tab><x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;</x-tab><x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</=
x-tab>102608<br>
<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab><x-tab>&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>2<x-tab>&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab><x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;</x-tab><x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;</x-tab>127200<x-tab>&nbsp;&nbsp;</x-tab><x-tab>&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab><x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;</x-tab>77168<br>
<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab><x-tab>&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab><br>
G726<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;</x-tab><x-tab>&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>1<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;</x-tab><x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
</x-tab><x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>16=
9600<br>
<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab><x-tab>&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>2<x-tab>&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab><x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;</x-tab><x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;</x-tab>84800<br>
<br>
For 1 frame/packet, both codecs require 2 ATM cells and the effective bit
rate is the same.&nbsp; For G.711 with Vad, we see significant bit-rate
savings.&nbsp; At 2 frames/packet, G726 provides a significant saving
over G711 without Vad, but G.711 with Vad-CNG still provides a savings
over G.726.<br>
<br>
Again, this is a simplified example, but it does show the value of using
a VAD-CNG scheme.<br>
<br>
In conclusion, when G.711 with VAD-CNG is considered in VoIP, it can
provide significant bit-rate savings, better quality, and lower
complexity compared to G.726. <br>
<br>
<br>
<br>
<blockquote type=3Dcite cite>2) Does anybody use the CN level cell/packet
mechanism ?<br>
Yes, we do.<br>
We support the existent draft-ietf-avt-profile-new-07.txt<br>
document and we want to see it approved as an RFC without<br>
any delay.<br>
</blockquote><br>
The use of the proposed &quot;enhanced&quot; CN payload is not mandatory,
and defaults back to the existing definition.&nbsp; If desired, the use
of the enhancement&nbsp; provides additional quality with little or no
extra payload overhead, at a fractional increase in overall complexity,
and remains generic to be used with any codec, not just G.711.&nbsp; If
Mitel has an RTP application in mind that uses only G.711 with a super
fast VAD-DTX scheme and uses only energy for Comfort Noise, then it is
still free to do so.&nbsp; The proposal in no way changes this.<br>
<br>
The enhanced CN payload has many advantages over the existing CN. There
is still the opportunity for it to be part of the RTP profile, and it
should be considered.<br>
<br>
<br>
<blockquote type=3Dcite cite>3) How about a more sophisticated CN, for
example one based<br>
on noise spectral information ?<br>
We believe there is no need for standardizing such an algorithm<br>
neither by IETF or ITU-T.<br>
However, if enough support for the adoption of such an algorithm<br>
is voiced at the next ITU-T meeting in Geneva, we are prepared<br>
to participate in the activites for defining the terms of<br>
reference of a CNG algorithm for G.711 as well as&nbsp; making
proposals<br>
for a candidate algorithm.<br>
I would be suprised if we would see at ITU-T just two proposals.<br>
</blockquote><br>
We look forward to proposals from all interested parties.<br>
<br>
<br>
<blockquote type=3Dcite cite>4) How fast should the work be done ?<br>
Well , there is no question dealing with CN for G.711 .<br>
I believe G.711 is too important for everybody. We need<br>
a separate question under SG16.<br>
The ITU-T procedures for standardization are strict and well<br>
defined. In this particular case many companies may have a high<br>
interest in following the work and contribuing to it.<br>
Work can be progressed through reflector groups but the<br>
Telecommuncation community cannot be properly informed without<br>
a general meeting in Geneva.<br>
We want to make sure that everybody in this world using PCM coders<br>
has a chance to take notice of the new work proposed and a chance<br>
to contribute.<br>
</blockquote><br>
The issue was first brought up at the Q12-14/16 Rapporteur's meeting in
Berlin in August 1999.&nbsp; Here is a summary of the issue after
discussion by the complete group of experts :<br>
<br>
<font face=3D"TIMES" size=3D4><i>It was noted that RFC 1890 is ill-define=
d;
it will be the second version of RFC 1890 that should be referred to. Its
current status is pre-RFC (see =93draft-ietf-avt-profile-new-06.ps=94).&n=
bsp;
It was noted that the IETF work is general to all sample based coders,
while the proposal focused on G.711.&nbsp; There may be a need for
non-G.711 work. <br>
<br>
It was agreed:<br>
<br>
1) To communicate a liaison to the IETF indicating that an issue exists
and that more work is needed.<br>
2) To communicate our concerns to Q19.<br>
<br>
The general goal would be to work with the RTP editors and not produce
additional standards unless necessary.<br>
<br>
</font></i>A liaison statement was generated from this meeting and sent
to Q19.&nbsp; This liaison was considered at the Q19-21/16 Rapporteur's
meeting in Geneva in Sept. 1999.&nbsp; Here is the summary from the Q19
Rapporteur about the issue and what was decided by the group :<br>
<br>
<font face=3D"Times New Roman, Times"><i><x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;</x-tab>Initiating
study for CNG under DTX condition<br>
Following the request by the Rapporteur Q.13/16, the meeting agreed to
start study on the Comfort Noise Generator for the speech codec without
built-in CNG such as G.711. A target of this work will be an Appendix to
G.711 that defines the CNG function. The first step was to draft a terms
of reference to define the CNG algorithm. The meeting formed an ad-hoc
group with chairman of Mr. Robert Zopf (Lucent Technologies), which
proposed a draft Performance Requirement s and Objectives in AC-99-46.
With respect to the quality, the chairman of Q.14/12 (Mr. Mark Parkins,
AT&amp;T) kindly offered co-operation in defining the quality assessment
method for coding speech under DTX/CNG operation during the interim
period after the September meeting by correspondence. <br>
The meeting agreed that the work should be proceeded in the
correspondence group of chairmanship by Mr. Zopf under the meeting of
Q.19/16, with help of Q.14/12 for quality issue.&nbsp; The correspondence
group should continue following works:<br>
(1)<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>Complete the Performance
Requirement s and Objectives<br>
(2)<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>Draft preliminary test
methodology and ask Q.14/12 for comments<br>
(3)<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>Complete the Test
Methodology<br>
(4)<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>Collect candidate
algorithms for CNG<br>
(5)<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>Performance assessment
for the candidates<br>
(6)<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>Select a candidate and
define the algorithm (Draft Appendix to G.711) .<br>
<br>
The time schedule should be established as a part of Terms of Reference
taking into account that the Appendix is strongly desired by WP2/16 and
IETF as soon as possible. Expectation for the Appendix may be approval at
the next SG16 meeting in February 2000.<br>
<br>
</font></i>The issue has been properly addressed within the ITU, and the
work is following the path laid out at the meeting by the Q19 Rapporteur
and the WP3 Chairman.<br>
<br>
I am sorry to everyone for the long email.&nbsp; I hope that the work can
now progress in a timely fashion.<br>
<br>
Best Regards,<br>
<br>
Rob.<br>
</html>

--=====================_194611276==_.ALT--




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Subject: Robust Header Compression BOF agenda
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This is a revised agenda for the robust header compression 
BOF to be held today Tuesday. As this is the night of the
social event, we will ignore the break and instead aim to
finish 17.45 at the latest. 

-------------
Robust Header compression BOF (robhc)

Tuesday, November 9 at 1545-1800
==========

CHAIRS: 
Mikael Degermark <micke@cdt.luth.se>
Stephen Pink <steve@sics.se>

DESCRIPTION:
The cellular community is now standardizing the next generation cellular
systems. These systems will use IP to deliver telephony to coming mobile
phones, to allow end-to-end IP telephony. As cellular bandwith is expensive,
radio spectrum efficiency is of outmost importance and thus the IP/UDP/RTP
headers must be compressed over the air. 3GPP has specified the requirements
for the needed header compression scheme and it has been shown that no current
header compression scheme in the IETF can satisfy these requirements. CRTP
[RFC-2508] is the closest candidate but is not sufficiently robust against
errors on the link, i.e., CRTP cannot deliver acceptable performance at the
anticipated error rates. It is likely that the number of users of cellular
IP telephony will be larger than the current number of people using TCP/IP,
therefore it is important that a suitable header compression scheme is 
developed.

The BOF will outline the relevant properties of the cellular links, the
requirements, and will explain why CRTP fails. One or more proposals for
a suitable header compression scheme will be presented, and finally we 
will discuss if and how the IETF will work on standardizing a suitable
header compression scheme.

AGENDA

    0. Introduction
        Mikael Degermark

    1. 3GPP timeline
        Mats Nilsson

    2. 3GPP requirements on robust header compression.
        Khiem Le, Nokia
        draft-degermark-hc-requirements-00.txt

    3. Why CRTP is not good enough
        Mikael Degermark
        draft-degermark-crtp-eval-00.{ps,txt}

    4. Cellular link properties affecting header compression - 
                delays, error rates & error distributions for
                EDGE, WCDMA, GSM, etc. 
        Krister Svanbro, Ericsson
          
    5. RObust Checksum-based header COmpression - ROCCO
        Lars-Erik Jonsson, Ericsson
        draft-jonsson-robust-hc-02.{ps,txt}

    6. Should we propose that a WG is formed? Outline of charter presented.
        Mikael Degermark





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From: "Peter Lewis" <peter.lewis@upperside.fr>
To: <Undisclosed-Recipient:@s2.smtp.oleane.net;>
Subject: Media Gateway Control Conference
Date: Tue, 9 Nov 1999 14:18:59 +0100
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MGCP, Megaco, H.248: performing seamless interoperation between IP and =
PSTN
networks. The international rendez-vous in Paris, 15-17 December 1999.

More infos:=20
http://www.upperside.fr/bamgc.htm


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Diso-8859-1" =
http-equiv=3DContent-Type>
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<DIV><FONT size=3D3>MGCP, Megaco, H.248: performing seamless =
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between IP and PSTN<BR>networks. The international rendez-vous in Paris, =
15-17=20
December 1999.<BR><BR>More infos: </FONT></DIV>
<DIV><FONT color=3D#800080 size=3D3><A=20
href=3D"http://www.upperside.fr/bamgc.htm">http://www.upperside.fr/bamgc.=
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From rem-conf Tue Nov 09 06:39:43 1999 
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To: micke@cdt.luth.se, ipng@sunroof.eng.sun.com, rem-conf@es.net,
        pilc@grc.nasa.gov, iptel@lists.research.bell-labs.com
Cc: micke@basil.cdt.luth.se
Subject: RE: Robust Header Compression BOF agenda
Date: Tue, 9 Nov 1999 16:18:36 +0200 
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Mikael,

I think you forgot an item on the agenda. As we discussed and agreed, we are
going to present a robust RTP header compression scheme. This is separate
>from the 3GPP requirements item. Also, due to my schedule constraints, as
discussed last week, would it be possible to reshuffle the items on the
agenda as follows (items 1 and 2 back-to-back will save some time, i.e.
together they should take less than 30 min):

AGENDA

    0. Introduction
        Mikael Degermark

    1. 3GPP requirements on robust header compression.
        Chris Clanton, Nokia
        draft-degermark-hc-requirements-00.txt

    2. A robust and efficient RTP header compression scheme
	Khiem Le, Nokia

    3. Why CRTP is not good enough
        Mikael Degermark
        draft-degermark-crtp-eval-00.{ps,txt}

	4. 3GPP timeline
        Mats Nilsson

    5. Cellular link properties affecting header compression - 
                delays, error rates & error distributions for
                EDGE, WCDMA, GSM, etc. 
        Krister Svanbro, Ericsson
          
    6. RObust Checksum-based header COmpression - ROCCO
        Lars-Erik Jonsson, Ericsson
        draft-jonsson-robust-hc-02.{ps,txt}

    7. Should we propose that a WG is formed? Outline of charter presented.
        Mikael Degermark

Thanks

Khiem

-----Original Message-----
From: EXT Mikael Degermark [mailto:micke@cdt.luth.se]
Sent: Tuesday, November 09, 1999 7:48 AM
To: ipng@sunroof.eng.sun.com; rem-conf@es.net; pilc@grc.nasa.gov;
iptel@lists.research.bell-labs.com
Cc: micke@basil.cdt.luth.se
Subject: Robust Header Compression BOF agenda


This is a revised agenda for the robust header compression 
BOF to be held today Tuesday. As this is the night of the
social event, we will ignore the break and instead aim to
finish 17.45 at the latest. 

-------------
Robust Header compression BOF (robhc)

Tuesday, November 9 at 1545-1800
==========

CHAIRS: 
Mikael Degermark <micke@cdt.luth.se>
Stephen Pink <steve@sics.se>

DESCRIPTION:
The cellular community is now standardizing the next generation cellular
systems. These systems will use IP to deliver telephony to coming mobile
phones, to allow end-to-end IP telephony. As cellular bandwith is expensive,
radio spectrum efficiency is of outmost importance and thus the IP/UDP/RTP
headers must be compressed over the air. 3GPP has specified the requirements
for the needed header compression scheme and it has been shown that no
current
header compression scheme in the IETF can satisfy these requirements. CRTP
[RFC-2508] is the closest candidate but is not sufficiently robust against
errors on the link, i.e., CRTP cannot deliver acceptable performance at the
anticipated error rates. It is likely that the number of users of cellular
IP telephony will be larger than the current number of people using TCP/IP,
therefore it is important that a suitable header compression scheme is 
developed.

The BOF will outline the relevant properties of the cellular links, the
requirements, and will explain why CRTP fails. One or more proposals for
a suitable header compression scheme will be presented, and finally we 
will discuss if and how the IETF will work on standardizing a suitable
header compression scheme.

AGENDA

    0. Introduction
        Mikael Degermark

    1. 3GPP timeline
        Mats Nilsson

    2. 3GPP requirements on robust header compression.
        Khiem Le, Nokia
        draft-degermark-hc-requirements-00.txt

    3. Why CRTP is not good enough
        Mikael Degermark
        draft-degermark-crtp-eval-00.{ps,txt}

    4. Cellular link properties affecting header compression - 
                delays, error rates & error distributions for
                EDGE, WCDMA, GSM, etc. 
        Krister Svanbro, Ericsson
          
    5. RObust Checksum-based header COmpression - ROCCO
        Lars-Erik Jonsson, Ericsson
        draft-jonsson-robust-hc-02.{ps,txt}

    6. Should we propose that a WG is formed? Outline of charter presented.
        Mikael Degermark




From rem-conf Tue Nov 09 10:40:00 1999 
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From: Florissa Colina <florissa@bmrc.berkeley.edu>
Subject: 11/10 Application Outsourcing: The Next Big Thing on the
  Internet -- Peter Newman, Ensim Corporation 
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Berkeley Multimedia, Graphics and Interfaces Seminar

Application Outsourcing: The Next Big Thing on the Internet

Wednesday, November 10, 1999, 
1:00-2:30 p.m. PST
Fujitsu Seminar Room (405 Soda Hall) 


Peter Newman 
Ensim Corporation 

The current model for application distribution is deeply flawed. Customers
are forced to keep track of the latest version of each software, to size
their desktop machines to have sufficient power to deal with anticipated
application load, and to painfully recover from hardware failures. These
problems disappear overnight when applications are outsourced to
Application Service Providers (ASPs). Potentially, the shift of
applications from desktops to ASPs is as powerful and disruptive as the
introduction of the World Wide Web. However, ASPs too face severe scaling
problems in providing outsourced services to tens of thousands of
customers. In this talk, we will outline these problems and describe a
novel approach to solving them.

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

The seminar is broadcast live on the Internet Mbone and as a RealNetworks
G2 broadcast. You can connect to the live RealNetworks broadcast at:
http://media2.bmrc.berkeley.edu/bibs/schedule.cfm 

You'll see a link to cs298-5/real networks/live programs.  

You can also directly put in this url into the Real Player:
rtsp://media2.bmrc.berkeley.edu/encoder/cs298-5.rm
To do so you will need to have the "Real Player G2" installed, which is
available from:
http://www.real.com/products/player/downloadrealplayer.html

To tune into the Internet MBone broadcast, look for the announcement in
your MBone session directory program ('sdr').  If you are not receiving the
announcement you may be able to receive the session by manually configuring
the client programs ('vic', and 'vat') with the session addresses:

low bit rate
	video: 233.0.25.1/22334
	audio: 233.0.25.2/22446

medium bit rate
	video: 233.0.25.129/22334
	audio: 233.0.25.2/22446

You can get further information about this seminar, and access to replays
of previous seminars at the MIG Seminar web page:
http://bmrc.berkeley.edu/298/index.html

Sponsored by the Berkeley Multimedia Research Center 




From rem-conf Wed Nov 10 02:53:25 1999 
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Subject: Isn't It Time To Wipe Out Your Bills?
Date: Tue, 09 Nov 1999 02:56:37 +0000
From: giants8@pmail.net
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From rem-conf Wed Nov 10 05:29:55 1999 
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Date: Wed, 10 Nov 1999 14:24:49 +0100
From: Nieves Pedreira Souto <nieves@udc.es>
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Organization: Universidade da Coru=F1a

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TTL > 48        =20

Bandwidth: 128 Kbps video + 39 Kbps audio=20

Session type: Audio and Video

Contact: nieves@udc.es

URL: http://rnasa.dc.fi.udc.es
=20
--=20
Nieves Pedreira Souto
Servicios Inform=E1ticos de Apoio =E1 Investigaci=F3n
Telf.: 981.167000 ext.:1308



From rem-conf Wed Nov 10 15:13:50 1999 
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Subject: Slides from AVT meeting
Date: Wed, 10 Nov 1999 21:39:40 +0000
From: Colin Perkins <c.perkins@cs.ucl.ac.uk>
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Could those presenters who have not already done so, please send me a copy
of their slides, for inclusion in the minutes.

Thanks,
Colin



From rem-conf Wed Nov 10 22:50:38 1999 
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From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: "Colin Perkins" <c.perkins@cs.ucl.ac.uk>, <rem-conf@es.net>
Cc: "Mickey Nasiri (ECS)" <mickey.nasiri@ecs.ericsson.se>,
        "Text Telephone Forum" <textphone@lsv.pi.se (Text Telephone Forum)>
Subject: SV: Slides from AVT meeting RTP-Text discussion
Date: Thu, 11 Nov 1999 11:41:13 +0100
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Colin and all,
I attach the slides for the RTP-Text discussion in two forms, Word and tx=
t.
Regards
Gunnar

-----Ursprungligt meddelande-----
Fr=E5n: Colin Perkins [mailto:c.perkins@cs.ucl.ac.uk]
Skickat: den 10 november 1999 22:40
Till: rem-conf@es.net
=C4mne: Slides from AVT meeting


Could those presenters who have not already done so, please send me a cop=
y
of their slides, for inclusion in the minutes.

Thanks,
Colin

------------------------------------------
Gunnar Hellstrom
LM Ericsson

E-mail gunnar.hellstrom@omnitor.se
Tel +46 8 556 002 03
Mob +46 708 204 288
fax +46 8 556 002 06

------=_NextPart_000_0000_01BF2C39.A8A050E0
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	name="RTP-text status nov-99.txt"
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Internet Engineering Task Force				AVT WG
IETF Washington DC 				Gunnar Hellstr=F6m
Nov 8-12 1999.				Ian Rytina
					LM Ericsson



Progress of draft-ietf-avt-rtp-text since the Oslo meeting

Draft-ietf-avt-rtp-text-02.txt is the latest version of the RTP Payload =
description for ITU-T T.140 Text Conversation.

The draft was entered on Oct 22. The specification seems now stable and =
the author now asks for the last call procedure to be initiated.



Summary of discussions that has been held, and modifications done:

Since July, the following discussions and modifications have taken place

1. Type of redundancy to use, FEC or RED? Agreed that it should be open =
and not in the normative part, but RED recommended because of its =
simplicity and the need to encourage interoperability. That is now =
reflected in section 2.3 and in the "Recommended procedures" =
section.3.2.

2. The need for this specification when IRC and TELNET CHAT are =
available? Answer: There is a user need for an end-to-end =
character-by-character internationally useful text conversation protocol =
to flow directly between terminals and possible to use in multimedia =
conversation environments like H.323 and SIP.

3. Proper procedure for MIME registration? The MIME registration section =
(6.) added, and the whole draft sent for MIME registration as text/t140.

4. Clock frequency? Irrelevant for text - it is collected as the user =
inputs text. But 1000 Hz specified now in section 2.1


5. Order of redundant data? For the recovery procedures to work, =
redundant data must be inserted in age order and with one redundant =
block from each transmitted packet. That is now clarified in section =
2.3.

6. Avoid double specification of fill character for missing data.
The character 0x2607 "Lightning" is to be used in T.140 to mark a spot =
where data is missing. In order to not cause double requirements, this =
is now only mentioned in a should clause in RTP-Text, section 3.2.

7. Transmission during silent periods: To give the last characters =
before a pause the same chance to redundancy recovery, redundant =
transmission with empty primary data is recommended as many times as the =
redundancy generations. (In one version, the Unicode sync character =
0xFEFF was specified.) See section 3.4.




Author: Gunnar Hellstrom,  gunnar.hellstrom@omnitor.se

------=_NextPart_000_0000_01BF2C39.A8A050E0
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////////////////////////////////////////////////////////////////////////////
////////////////////////////////////////////////////////////////////////////
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------=_NextPart_000_0000_01BF2C39.A8A050E0--




From rem-conf Thu Nov 11 11:38:36 1999 
From rem-conf-request@es.net Thu Nov 11 11:38:36 1999
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Message-ID: <000032ac4cc1$000039fd$0000086a@>
To: <Undisclosed Recipients>
From: los321@email.com
Subject: Get your Free Mortgage Quote!!
Date: Thu, 11 Nov 1999 13:12:15 -0500
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Dear Homeowner,
From: los321@email.com
Bcc:
Return-Path: los321@email.com
X-OriginalArrivalTime: 11 Nov 1999 19:19:39.0185 (UTC) FILETIME=[B2C91210:01BF2C79]
Date: 12 Nov 1999 06:19:39 +1100

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From rem-conf Thu Nov 11 16:18:43 1999 
From rem-conf-request@es.net Thu Nov 11 16:18:43 1999
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Date: Thu, 11 Nov 1999 19:12:48 -0500
To: wp3audio@ctd.comsat.com, rem-conf@es.net
From: Robert Zopf <zopf@lucent.com>
Subject: CNG ToR
Cc: meperkins@att.com
Mime-Version: 1.0
Content-Type: multipart/mixed;
	boundary="=====================_33968754==_"
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--=====================_33968754==_
Content-Type: text/plain; charset="us-ascii"; format=flowed

Colleagues,

Find attached an updated ToR.  The main changes from the original draft 
include:

* it now notes sampling rates other than 8.0 kHz
* car noise is added as a background noise source
* a complexity objective of less than 1.0 MIP is added to reflect the 
concerns about complexity.
* compatibility with the current RTP profile is added (first byte 
represents the energy).
* removal of a 16 bit fixed point bit exact implementation
* removal of a frame size specification (the CNG should work with all frame 
sizes)
* removal of CNG update rate - this is implementation specific
* removal of bit-rate since this depends on the CNG update rate.

The schedule is left blank for now.  It can be addressed after the 
requirements have been agreed upon.  In this way, the final requirements 
can be sent to Q.14/12 in order to finalize the test plan.

Please provide comments so we can have the ToR finalized by the end of next 
week.

Regards,

Rob.

--=====================_33968754==_
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--=====================_33968754==_--




From rem-conf Fri Nov 12 05:40:25 1999 
From rem-conf-request@es.net Fri Nov 12 05:40:24 1999
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	id 11mGqc-0000jl-00; Fri, 12 Nov 1999 05:34:38 -0800
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From: sgidshfds@mailr.com
Message-Id: <199911121341.OAA03462@diaserv.diakonie.de>
To: dsdaufds@hotbot.com
Subject: The answer to cancer has been known!!!
Date: Fri, 12 Nov 1999 04:00:53 -0700
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<FONT size=3> There are many web sites out there that are showing that the answer to cancer has been known. The evidence shows that when a person adds bitter seeds to their diet which contain the vitamin B17, the possibilities of contracting cancer is very slim. The evidence also shows that even after a person has been diagnosed with cancer, simply adding seeds and vitamin B17 to their diet can shrink the tumors and will protect the rest of the persons body. Some of these companies have hundreds of success records posted right on their web site with phone numbers and addresses for others to contact. In almost all of the cases that these websites record, blood pressure is lowered and blood sugar levels improve.<BR>
The websites show that a person should not only depend on the B17 and Seeds, but also clearly states that individuals should include other nutritious immune system building foods in their diet.\~It also shows that a person who still decides to stay on chemo and takes the apricot seeds live many times longer then if they took the chemo alone.<BR>
The FDA has made it very difficult for companies to advertise information concerning Vitamin B17 and the Apricot seeds. <BR>
If this is your first time ordering, order one of the packages which includes one bottle of the B17, one pound of the apricot seeds, the book, the video, the journal and complete instructions<BR>
<BR>
Call our office for the web site location.<BR>
 <BR>
You may order by credit card by calling 800-395-7379<BR>
<BR>
Book: World Without Cancer, by Edward Griffin<BR>
Contains most of what's mentioned in the videotape 'World Without Cancer' and much more on studies that were covered up and great scientist who where arrested when they began telling others about the truth with vitamin B17. Highly annotated for the serious student who wants to research the information for himself. Contains studies on Amygdalin (vitamin B17, laetrile) that were conducted at the famous cancer institute, Sloan Kettering and covered up. Full explanation of the politics of cancer from Mr. Edward Griffin.<BR>
<BR>
Videotape: "World Without Cancer" Length 55 min.<BR>
By Edward Griffin. Features how the answer to cancer has always been known and was pushed out of the U.S. This video will blow you away. It brings to light in an organized and simple to understand fashion the politics of cancer, the scandal, the biologics scientifics, the cultures that have never had a case of cancer and some success stories...including my own which was featured on Inside Edition's Extra in 1996. $19.95<BR>
<BR>
Apricot Kernels<BR>
These seeds have the highest content of B-17 on earth. These are not the same as the ones that are sometimes found in the health food stores. The ones in the health food store are sun- dried. Sun dried kernels are not effective, that's why they're allowed to be sold in U.S. stores without problem. Our kernels are fresh, chewable and bitter and a necessity in out diet. Dr. Krebs says 7 per day for absolute prevention. As therapy for existing cancer, between 20 and 50 per day is recommended. Although they are bitter, they must be integrated into our diets. Stores do not sell fresh apricot seeds because of what the FDA did years ago. $12.95 per pound, approx. 860 seeds per pound already cracked out of the shell.<BR>
<BR>
Vitamin B-17<BR>
Fruit seed extract is called Amygdalin, Vitamin B17 or Laetrile. We carry this vitamin in tablet form in 100mg and 500mg quantities. They are 99% pure fruit seed extract from Cyto Pharm of Mexico. Cyto Pharma of Mexico are the pioneers of the seed extraction of Vitamin B17 for over 20 years.. They are the only ones who have followed Dr. Krebs exact specifications for the production of vitamin B17. For prevention the 100mg capsules are recommended by Dr. Krebs, one or two per day; If existing cancer is the case, between 2000-4000mg for the first 30 days is recommended along with 2-5 seeds per hour (waking hours). $95 per 100 500 mg tablets. $34 per 100 of the 100 mg tablets.<BR>
<BR>
Call if you have questions and we'll give you a web site that has all the info.<BR>
<BR>
To complain to UUNET about blocking this information on the web please complain to the security dept. of UUNET at 800-488-6384.  Please ask for the security dept. and for John Bradshore.  He has unethically censored out the web sites that talk of apricot seeds.  Or e-mail abuse@uunet.net<BR>
<BR>
<BR>
<BR>
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Under Bill s.1618 TITLE III passed by the 105th US<BR>
Congress this letter cannot be considered spam as long as the sender<BR>
includes contact information & a method of removal.<BR>
To be removed mailto:<A HREF="mailto:None4me@angelfire.com">None4me@angelfire.com <BR>
and type your email address in the subject line.OR call (877) 810-1218<BR>
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From rem-conf Sat Nov 13 04:51:53 1999 
From rem-conf-request@es.net Sat Nov 13 04:51:52 1999
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Date: Sat, 13 Nov 1999 10:34:45 -0200 (EDT)
From: "LANOMS'99" <lanoms99@lrg.ufsc.br>
Subject: IEEE LANOMS99 - Special prices for Academics
To: activenets_all@ittc.ukansas.edu, ActiveNets_Wire@ittc.ukans.edu,
        af-all@atmforum.com, alg@comm.toronto.edu,
        amlist@takilab.k.dendai.ac.jp, apc@ee.nthu.edu.tw, apc@eee.nthu.edu.tw,
        apc_members@hornbill.ee.nus.sg, apnoms-all@nile.postech.ac.kr,
        apnoms-oc@amazon.postech.ac.kr, apnoms-oc@nile.postech.ac.kr,
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        cnon@maestro.bellcore.com, commsoft@cc.belcore.com,
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        comsoc.tac@tab.ieee.org, comsoc-chapters@ieee.org,
        comsoc-gicb@ieee.org, comswtc@gmu.edu,
        cost237-transport@comp.lancs.ac.uk, cost237-transport@comp.lancs.at,
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From rem-conf Sat Nov 13 10:42:47 1999 
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this question is best addressed by avt..

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com
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From: Ramana Yarlagadda <ramana@chiplogic.com>
Subject: RTP Packetization
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Hi all,

In section 6.2.1/H225v2 it is mentioned as :
For audio algorithms such as G.723 which use more than one size of audio

frame, audio frame boundaries within each packet shall be signalled
in-band to the audio channel.

If we consider it from RTP point of view, then in a RTP packet
there can be various combinations of the silence and voice frames.
Let's consider one RTP packet like this:
<voice><voice><SID><voice><SID>.
so, audio frame boundaries can be
1. between two voice frames or
2. between voice frame and a non-voice frame.
3. or both 1 and 2.
My question is exactely which of the above mentioned audio boundaries,
according to the section 6.2.1/h225v2 needs to be signalled in the RTP
packet?

And, the boubndaries are to be signalled in-band, it means we have to
add the boundary signalling in the RTP payload itself.
am I correct?

If we are going to put the boundary signalling inside the RTP payload,
and this sort of signalling is H323 specific thing. then I think it will
create some interoperability problems with RTP implementations other
than
H323.

-Thanks in advance.
-ramana


---------
This message came from the IETF IPTEL Working Group Mailing List.


--------------4959C43537771C97BD87CE67--




From rem-conf Sun Nov 14 17:35:34 1999 
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From: 62630496@awwww.com
Date: Sun, 14 Nov 99 18:33:24 EST
To: nbvnvb@ghgfhfghfgh.com
Subject: Low price dental optical plan for you or your family
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Hello, 

We work with a group of your local doctors and dentists 
and are offering a Dental - Optical Plan that runs 
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From rem-conf Mon Nov 15 10:26:40 1999 
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From: Florissa Colina <florissa@bmrc.berkeley.edu>
Subject: 11/17 Operational Multicast Network Management -- Radhika
  Malpani, HP Labs 
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Berkeley Multimedia, Graphics and Interfaces Seminar

Operational Multicast Network Management

Wednesday November 17, 1999 
1:00-2:30 p.m. PST
Fujitsu Seminar Room (405 Soda Hall) 

Radhika Malpani 
HP Labs 

Despite the increased need for IP multicast, driven by the popularity of
applications like multimedia events and
data distribution services, its deployment has not been widespread.
Surprisingly, the lack of appropriate
network management tools for IP multicast has proven to be a major barrier
for its deployment. While a few
diagnotic tools do exist today, these usually require expert knowledge and
understanding of multicast
protocols.Unlike unicast, where domain knowledge and management experience
is considerable, in the
multicast domain few people have the requisite multicast knowledge. This is
especially true in the NOCs
(Network Operations Center), where multicast management experience is
practically non-existent 

This talk will describe the results of research at HP Laboratories into the
management needs of multicast
network operators. It will describe a prototype multicast network
management system we've built that is aimed
at network operators and discuss how it can be used to manage multicast
networks. This prototype is currently
used at HP and other sites to manage the multicast networks and some of our
preliminary experiences with the
prototype will be discussed.
---------------

The seminar is broadcast live on the Internet Mbone and as a Real Networks
G2 broadcast.

You can connect to the live RealNetworks broadcast at:

http://media2.bmrc.berkeley.edu/bibs/schedule.cfm 
You'll see a link to cs298-5/real networks/live programs.  

You can also directly put in this url into the Real Player: 
rtsp://media2.bmrc.berkeley.edu/encoder/cs298-5.rm

To do so you will need to have the "Real Player G2" installed, which is
available from:
http://www.real.com/products/player/downloadrealplayer.html

To tune into the Internet MBone broadcast, look for the announcement in
your MBone session directory program ('sdr').  If you are not receiving the
announcement you may be able to receive the session by manually configuring
the client programs ('vic', and 'vat') with the session addresses:

low bit rate
	video: 233.0.25.1/22334
	audio: 233.0.25.2/22446

medium bit rate
	video: 233.0.25.129/22334
	audio: 233.0.25.2/22446

You can get further information about this seminar, and access to replays
of previous seminars at the MIG Seminar web page:

http://bmrc.berkeley.edu/298/index.html

Sponsored by the Berkeley Multimedia Research Center 




From rem-conf Mon Nov 15 11:59:26 1999 
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From: <wendyhilton@thehelm.com>
Subject: We need book,software,prod reviewers, keep after review post - iNet News
Date: Mon, 15 Nov 1999 11:28:58
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iNet News Nov 1999 (for reviewers and prospects only)

INet Reviews needs book, software & product reviewers. You may 
request any virtually title or consumer product. Titles are 
shipped directly to you with the freight paid from the publisher 
or manufacturer. You write a brief review and post it to our web 
site. After you write brief review you keep all review titles.

For your review starter information kit send an email message 
with the words STARTER 1114G39 in the message subject line to  
wendyhilton@thehelm.com The starter kit contains complete 
information plus a 70 page list of review products others have 
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From rem-conf Mon Nov 15 13:32:36 1999 
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If the subject is about how to pack and unpack one or more frames of
encoded media in a single RTP payload, then I think that the RTP audio
profile addressed this already.  The method varies across vocoders; for
G.723.1 there is a dibit in the first byte of each encoded frame which
indicates the frame's length.  G729A is different; the profile specifies
the constraint that G729A payloads must consist of one or more active
frames followed by zero or one SIDs, which also permits a unique parsing of
the payload.  These may be the in-band codings to which H225 refers.  Seems
to me, however, that they are more out-of-band -- in-band instead to the
audio profile.  It is hard to imagine how this would break any robust RTP
implementations.

Vince Virgilio




Jonathan Rosenberg <jdrosen@dynamicsoft.com> on 11/13/99 12:38:13 PM

To:   rem-conf@es.net
cc:    (bcc: Vincent Virgilio)
Subject:  [Fwd: RTP Packetization]




this question is best addressed by avt..

-Jonathan R.
--
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From: Ramana Yarlagadda <ramana@chiplogic.com>
Subject: RTP Packetization
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Hi all,

In section 6.2.1/H225v2 it is mentioned as :
For audio algorithms such as G.723 which use more than one size of audio

frame, audio frame boundaries within each packet shall be signalled
in-band to the audio channel.

If we consider it from RTP point of view, then in a RTP packet
there can be various combinations of the silence and voice frames.
Let's consider one RTP packet like this:
<voice><voice><SID><voice><SID>.
so, audio frame boundaries can be
1. between two voice frames or
2. between voice frame and a non-voice frame.
3. or both 1 and 2.
My question is exactely which of the above mentioned audio boundaries,
according to the section 6.2.1/h225v2 needs to be signalled in the RTP
packet?

And, the boubndaries are to be signalled in-band, it means we have to
add the boundary signalling in the RTP payload itself.
am I correct?

If we are going to put the boundary signalling inside the RTP payload,
and this sort of signalling is H323 specific thing. then I think it will
create some interoperability problems with RTP implementations other
than
H323.

-Thanks in advance.
-ramana


---------
This message came from the IETF IPTEL Working Group Mailing List.











From rem-conf Tue Nov 16 03:14:59 1999 
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Subject: Firewalls and Mbone...
Date: Tue, 16 Nov 1999 12:00:03 +0100
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Hi;

Has anyone had any experience trying to get Mbone applications functioning 
behind a Firewall. Especially applications such as SDR, VIC and RAT.
I'm interested in information dealing with Cisco's PIX and Checkpoint's 
Firewall-1.

Thanks
  Chris




From rem-conf Tue Nov 16 04:00:20 1999 
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--NextPart

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

	Title		: RTP Audio/Video Profile Interoperability Statement
	Author(s)	: C. Perkins
	Filename	: draft-ietf-avt-profile-interop-00.txt
	Pages		: 5
	Date		: 15-Nov-99
	
It is required to demonstrate interoperability of implementations
in order to move the RTP audio/video profile to draft standard.
This memo outlines those features to be tested, as the first
stage of an interoperability statement.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avt-profile-interop-00.txt

Internet-Drafts are also available by anonymous FTP. Login with the username
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From rem-conf Tue Nov 16 11:13:56 1999 
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Date: Tue, 16 Nov 1999 14:10:25 -0500
To: wp3audio@ctd.comsat.com, rem-conf@es.net
From: Robert Zopf <zopf@lucent.com>
Subject: CNG Complexity
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All,

In order to accurately measure the complexity of candidate algorithms, 
Lucent proposes that all algorithms be submitted using the basic operators 
used in G.729.  There is an updated basic operator library with the ability 
to count WMOPS (sent out on the reflector on 08/09/99 by Simao) that could 
be used to measure the complexity of candidate algorithms.


We propose that if more than one candidate meets the requirements, that the 
selection then be based on the WMOPS.  The worst case WMOPS would be used 
for a 5ms frame with the CN analysis/synthesis operating every frame 
(VAD=0, and CN updated every frame).

Other proposals ?

Since the complexity requirement is currently stated in MIPS, we should 
change it to WMOPS.  Does anyone have a mapping from WMOPS (as in the basic 
op library) to MIPS on a typical fixed-point DSP ?

Rob.




From rem-conf Tue Nov 16 12:59:12 1999 
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Date: Tue, 16 Nov 1999 15:38:26 -0500
To: rem-conf@es.net
From: Robert Zopf <zopf@lucent.com>
Subject: Please see ITU reflector for CNG work
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Following Steve's advice, I will continue the discussion of CNG only on the 
ITU reflector.

Regards,

Rob.


>Date: Tue, 16 Nov 1999 11:51:29 -0800 (Pacific Standard Time)
>From: Stephen Casner <casner@cisco.com>
>To: Robert Zopf <zopf@lucent.com>
>Subject: Re: CNG Complexity
>Sender: casner@cisco.com
>
>Rob,
>
>It isn't a problem, but I would advise that you need not keep rem-conf
>in the loop for your discussion of candidate algorithms and complexity
>measures, etc.  I suspect that the folks with expertise in this area
>might already be participating in ITU as well as IETF.
>
>It would be useful to keep us posted on progress and when questions
>about packet format or compatibility arise.
>                                                         -- Steve




From rem-conf Tue Nov 16 14:43:51 1999 
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	IEEE INTERNATIONAL CONFERENCE ON MULTIMEDIA AND EXPO

	       JULY 30 -- AUGUST 2, 2000, NEW YORK CITY

	        http://www.ieee-multimedia.poly.edu


-----------------CALL FOR PAPERS--------------------

International Conference on Multimedia & Expo 2000 is the first IEEE
multimedia conference that will take place in joint collaboration with
four IEEE societies. The same four IEEE societies that have recently
launched the IEEE Transactions on Multimedia, namely the Circuits and
Systems Society, the Computer Society, the Signal Processing Society
and the Communications Society have again come together in making this
one-of-a-kind conference a reality.

The goal is to unify all IEEE as well as non-IEEE multimedia related
activities under the common umbrella of ICME 2000 -- the millennium
multimedia meeting. It is expected that this will become a yearly
event as the flagship Multimedia conference of the IEEE.  New York
City (the Big Apple) is uniquely suited for this first-time event in
view of its large concentration of high tech Industries, first-rate
academic institutions and numerous small multimedia companies. The
intent is to blend the traditional IEEE conference framework into an
Expo by providing the industry an opportunity to showcase products.

Authors should submit a four-page manuscript in double-column format
including authors' names, affiliations and a short abstract. Only
electronic submission (in postscript format) will be accepted.
Electronic submission procedure will be made available at the
conference web site:

	http://www.ieee-multimedia.poly.edu

Topics covered include but are not limited to the following:
	* Signal processing for media integration
	* Components and technologies for multimedia systems
	* Human-machine interface and interaction
	* Multimedia databases and file systems
	* Multimedia communication and networking
	* System integration
	* Applications
	* STDS standards and related issues

Once accepted, the authors will be asked to prepare a four-page camera ready
paper to appear in the conference proceedings and CD-ROM proceedings.

Proposals for special sessions and half-day or full-day tutorial courses
may be submitted to the respective chairs before the respective
deadlines. Visit the conference web site for further information.


			DEADLINES

	December 1, 1999	Submission of regular papers
	February 1, 2000	Proposals for special sessions, panels, tutorials, demos
	March 1, 2000	Notification of acceptance for papers
	March 15, 2000	Notification of acceptance for proposals
	April 1, 2000	Paper camera-ready copy due




		CONFERENCE COMMITTEE

	General Chair:
		Sankar Basu, IBM
		sankar@watson.ibm.com

	Finance Chair:
		Peter Westerink, IBM

	Technical Program Co-Chairs:
		Tsuhan Chen, Carnegie Mellon
		Jeff H. Derby, IBM
		Peiya Liu, Siemens
		Chung-Sheng Li, IBM

	Special Sessions:
		Raj Acharya, SUNY, Buffalo
		Chalapathy Neti, IBM

	Tutorials:
		Rama Chellappa, U. Maryland
		Alexandros Eleftheriadis, Columbia

	Plenaries:
		S. Y. Kung, Princeton U.
		Shih-Fu Chang, Columbia U.

	Demos/Exhibits:
		Yao Wang, NY Polytechnic U.

	Publicity:
		Jelena Kovacevic, Bell Labs
		Ed Wong, NY Polytechnic

	Publications:
		Y. Q. Shi, NJIT
		Bhuvana Ramabhadran, IBM

	Registration:
		Junehwa Song, IBM

	Local Arrangements:
		Goran Djuknic, Lucent Tech.
		Ravi Rao, IBM
		Atul Puri, AT&T


      Electronic Media:
           Raj Kumar Rajendran, Columbia University
           Nasir Memon, Polytechnic University

	European Liaison:
		Alberto Del Bimbo, U. Florence
		Martin Vetterli, EPFL

	Far East Liaison:
		Masahito Hirakawa, Hiroshima U.
		Sadaoki Furui, Tokyo Inst. Tech.

	Steering Committee:
		Tadao Ichikawa, Hiroshima U.
		Rama Chellappa, U. Maryland
		Gary W. Cobb, Dell
		Alberto Del Bimbo, U. Florence
		Jeff H. Derby, IBM
		Alexander D. Gelman, Panasonic
		Yih-Fang Huang, U. Notre Dame
		Chung-Sheng Li, IBM
		Peiya Liu, Siemens
		Antonio Ortega, USC
		Dipankar Raychaudhuri, NEC
		Bing Sheu, Avanti Corporation
		John A. Sorensen, TU Denmark

	Advisory Committee:
		Richard Cox, AT&T
		Ed Deprettere, Delft U. Tech.
		Cesar Gonzales, IBM
		Scott Kirkpatrick, IBM
		Michael Picheny, IBM








From rem-conf Tue Nov 16 16:33:01 1999 
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From: Florissa Colina <florissa@bmrc.berkeley.edu>
Subject: 11/17 Operational Multicast Network Management -- Radhika
  Malpani, HP Labs 
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Berkeley Multimedia, Graphics and Interfaces Seminar

Operational Multicast Network Management

Wednesday November 17, 1999 
1:00-2:30 p.m. PST
Fujitsu Seminar Room (405 Soda Hall) 

Radhika Malpani 
HP Labs 

Despite the increased need for IP multicast, driven by the popularity of
applications like multimedia events and
data distribution services, its deployment has not been widespread.
Surprisingly, the lack of appropriate
network management tools for IP multicast has proven to be a major barrier
for its deployment. While a few
diagnotic tools do exist today, these usually require expert knowledge and
understanding of multicast
protocols.Unlike unicast, where domain knowledge and management experience
is considerable, in the
multicast domain few people have the requisite multicast knowledge. This is
especially true in the NOCs
(Network Operations Center), where multicast management experience is
practically non-existent 

This talk will describe the results of research at HP Laboratories into the
management needs of multicast
network operators. It will describe a prototype multicast network
management system we've built that is aimed
at network operators and discuss how it can be used to manage multicast
networks. This prototype is currently
used at HP and other sites to manage the multicast networks and some of our
preliminary experiences with the
prototype will be discussed.
---------------

The seminar is broadcast live on the Internet Mbone and as a Real Networks
G2 broadcast.

You can connect to the live RealNetworks broadcast at:

http://media2.bmrc.berkeley.edu/bibs/schedule.cfm 
You'll see a link to cs298-5/real networks/live programs.  

You can also directly put in this url into the Real Player: 
rtsp://media2.bmrc.berkeley.edu/encoder/cs298-5.rm

To do so you will need to have the "Real Player G2" installed, which is
available from:
http://www.real.com/products/player/downloadrealplayer.html

To tune into the Internet MBone broadcast, look for the announcement in
your MBone session directory program ('sdr').  If you are not receiving the
announcement you may be able to receive the session by manually configuring
the client programs ('vic', and 'vat') with the session addresses:

low bit rate
	video: 233.0.25.1/22334
	audio: 233.0.25.2/22446

medium bit rate
	video: 233.0.25.129/22334
	audio: 233.0.25.2/22446

You can get further information about this seminar, and access to replays
of previous seminars at the MIG Seminar web page:

http://bmrc.berkeley.edu/298/index.html

Sponsored by the Berkeley Multimedia Research Center 




From rem-conf Thu Nov 18 10:05:59 1999 
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To: GeeVarghese John <gvjohn@miel.mot.com>, casner@cisco.com,
        micke@cdt.luth.se
From: Tmima Koren <tmima@cisco.com>
Subject: Re: Extensions to CRTP
Cc: brucet@cisco.com, dwing@cisco.com, pruddy@cisco.com, rem-conf@es.net
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At 02:13 PM 11/18/99 -0500, you wrote: 
>
> Steve, Tmima  & all, 
>
> This is regarding the ietf draft - Extensions to CRTP  :
> draft-koren-avt-crtp-enhance-00.txt 
> I have few doubts and could you please clarrify these points .. 
>
> In the section 1. COMPRESSED_UDP packet "T bit" .. 
>
> It is stated that CUDP packet reset the deltaT value to 0 and 
> therefore T bit avoids the need to send deltaT in the next CRTP. 
>
> Q1) My doubt is If the entire original RTP header is the in the CUDP packet
> why to 
> send the delta also with the original header .. 
> why can't the decompressor findout the new delta to be used 
> from the RTP header in the  CUDP packet itself, Is it not redundancy ? 
>
> Also You have stated that 
>
> >> Multiple COMPRESSED_UDP packets may be sent by the compressor when changes
> in the 
> >> compressor state need to be synchronized. Sending multiple packets will
> increase the 
> >> probability of delivery of the synchronized state in the event of packet
> loss. 
>
> Why do we need to send multiple packet  and how it will increase the
> probability even 
> if you send more CUDP packets, as long as the decomperssor look at the seq
> number 
> and detect  packet loss  it should send a CONTEXT_STATE packet. 
>
> If the draft reference you made was for loss of more than 16 packets from the
> same flow, 
> and the possible wrong decompression of headers .. 
> twice algorithm should detect this (But twice algo is not used if checksum is
> 0) 
> So here what exactly the limit could be , the sequence number space 1..16 
> Since this draft address the use of this algo over ATM/AAL5 links  I think 
> sequence number space 1..16 may not be sufficient. 
>  


Some notations:
dt         deltaT
FH        FULL_HEADER
CR       COMPRESSED_RTP
CR+     COMPRESSED_RTP with deltaT (dt)
CU       COMPRESSED_UDP
CU+     COMPRESSED_UDP with dt
CS       CONTEXT_STATE     

I'll use an example
The sender sends a sample every 10 ms
Using CRTP, the sequence of packets will be:

seq#   Time     pkt type
1        10         FH
2        20         CR+    dt=10
3        30         CR
...
10      100       CR

At this point this side stops talking. Talking resumes at time 200

11      200       CR+    dt=100
12      210       CR+    dt=10
13      220       CR
....

With the extension of T-bit in CU, we can send:

11'     200       CU+    dt=10
12'     210       CR
...

We are trying to adopt CRTP to links that have some loss of packets and a long
round trip time (RT) 
We want to avoid the decompressor getting out of sync and having to send CS
because this will cause loss of many packets for this stream due to the long
RT.

Let's check what happens if some packets are lost
1. packet 4 was lost
    The decompressor will use the 'twice' and will stay in sync
2. packet 11 was lost
    The decompressor is out of sync
3. packet 11'  was lost
    The decompressor is out of sync

We see that there is a weak point in the compression at the beginning of a talk
spurt
We try to correct this by sending multiple CU+ 

11"    200       CU+    dt=10
12"    210       CU+    dt=10 
13"    220       CR
...

If the decompressor received any of 11"  or 12" ,  he is still in sync

How many CU+ do we want to send at the beginning of a talk spurt?
This depends on the quality of the link (which is good but not perfect).
If we know that on this link we may lose at most 2 packets in a row, we'll send
3 CU+ at the beginning of a talk spurt
The decompressor will stay in sync as long as he received at least one of the
CU+  


>
>  
>
> Q2) Can you explain the use of  "The NON-RTP stream flag" for this algorithm 


The compressor knows that this stream is a non-RTP stream, but he still wants
to compress the stream. The compressor will have to use CU only. If the
compressor can pass a hint to the decompressor that this is not an RTP stream,
the decompressor will save only the IP and UDP headers for this context, and
will not save the extra context of the RTP header (that does not exist here)
and will not try to calculate the deltas and other RTP-related calculations


>
> Q3) Section III : Acknowledgement to a new compressed stream 
>
> Context Invalidation was one point which I had pointed out to Steve/tmima
> long time back .. 
> I am having those mails with me .. 
>
> But, (i) Do we need ACCEPT and REJECT packet  ?? 
> Having a NACK mechnism sounds reasonable .. 
> I feel there shouldnt be both .. 
> But what compressor is supposed to do after sending the first FULL_HEADER
> packet 
> when compressor waiting for the ACCEPT or REJECT 
> should it queue the packets till the reply/or timeout 
> or Should it send the compressed packets assuming the decompressor will drop
> if there are 
> REJECT state at the decompressor side. 
>
> But (ii) what about the backword compatibility issue because this
> ACCEPT/REJECT mechanism has dependancy and it will not work with CRTP-2508 
>
> As per the current proposal we need to send ACCEPT/REJECT packet for every 
> FULL_HEADER packet ... (iii)Is it not adding extra bandwidth consuption .. 
> Draft talks about FULL_HEADER retransmit as well .. 


The REJECT packet is useful when the decompressor runs out of resources. The
compressor initiates compression of a stream by sending a FH. With CRTP today
the decompressor can send a CS to invalidate the context, but the compressor
does not know the reason and will attempt to compress this stream again. A
REJECT will notify the compressor that the decompressor cannot handle
compresstion of additional streams at this moment.
An implementation that was not enhanced to use REJECT will still use CS to
invalidate the context each time the compressor attempts to compress the
stream.
The compressor continues to send the compressed packets of the stream until he
receives CS or REJECT
The intent is to make the ACCEPT/REJECT packet optional - an optimization.

Tmima


>
> More questions after discussing these points .. 
> Please answer the questions ................... 
>   
>
> Regards 
>
> GVJ 
>   
>
> --
> --------------------------------------------------------------------------- 
> Geevarghese John,   DataCom Group, 
> Motorola India Electronics Ltd, 
> "The Senate", No-33 A, 
> Ulsoor Road, 
> Bangalore - 42. 
> Phone  : 91-80-5598615  Xnt-4005 
> E-mail : gvjohn@miel.mot.com 
> --------------------------------------------------------------------------
> ----- 
> --------------------------------------------------------------------------
> ----- 
>  



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<html>
<br>
At 02:13 PM 11/18/99 -0500, you wrote: <br>
<blockquote type=cite cite>Steve, Tmima&nbsp; &amp; all, <br>
<br>
This is regarding the ietf draft - Extensions to CRTP&nbsp; :
draft-koren-avt-crtp-enhance-00.txt <br>
I have few doubts and could you please clarrify these points .. <br>
<br>
In the section 1. COMPRESSED_UDP packet &quot;T bit&quot; .. <br>
<br>
It is stated that CUDP packet reset the deltaT value to 0 and <br>
therefore T bit avoids the need to send deltaT in the next CRTP. <br>
<br>
Q1) <b>My doubt is If the entire original RTP header is the in the CUDP
packet why to</b> <br>
<b>send the delta also with the original header ..</b> <br>
<b>why can't the decompressor findout the new delta to be used</b> <br>
<b>from the RTP header in the&nbsp; CUDP packet itself, Is it not
redundancy ?</b> <br>
<br>
Also You have stated that <br>
<br>
&gt;&gt; Multiple COMPRESSED_UDP packets may be sent by the compressor
when changes in the <br>
&gt;&gt; compressor state need to be synchronized. Sending multiple
packets will increase the <br>
&gt;&gt; probability of delivery of the synchronized state in the event
of packet loss. <br>
<br>
<i>Why do we need to send multiple packet</i>&nbsp; and <i>how it will
increase the probability even</i> <br>
if you send more CUDP packets, as long as the decomperssor look at the
seq number <br>
and detect&nbsp; packet loss&nbsp; it should send a CONTEXT_STATE packet.
<br>
<br>
I<i>f the draft reference you made was for loss of more than 16 packets
>from the same flow,</i> <br>
<i>and the possible wrong decompression of headers ..</i> <br>
<i>twice algorithm should detect this (But twice algo is not used if
checksum is 0)</i> <br>
<i>So here what exactly the limit could be , the sequence number space
1..16</i> <br>
<i>Since this draft address the use of this algo over ATM/AAL5
links&nbsp; I think</i> <br>
<i>sequence number space 1..16 may not be sufficient.</i> <br>
&nbsp;</blockquote><br>
Some notations:<br>
dt&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; deltaT<br>
FH&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; FULL_HEADER<br>
CR&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; COMPRESSED_RTP<br>
CR+&nbsp;&nbsp;&nbsp;&nbsp; COMPRESSED_RTP with deltaT (dt)<br>
CU&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; COMPRESSED_UDP<br>
CU+&nbsp;&nbsp;&nbsp;&nbsp; COMPRESSED_UDP with dt<br>
CS&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CONTEXT_STATE&nbsp;&nbsp;&nbsp;&nbsp; <br>
<br>
I'll use an example<br>
The sender sends a sample every 10 ms<br>
Using CRTP, the sequence of packets will be:<br>
<br>
seq#&nbsp;&nbsp; Time&nbsp;&nbsp;&nbsp;&nbsp; pkt type<br>
1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
10&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; FH<br>
2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
20&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; CR+&nbsp;&nbsp;&nbsp;
dt=10<br>
3&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
30&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; CR<br>
...<br>
10&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 100&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR<br>
<br>
At this point this side stops talking. Talking resumes at time 200<br>
<br>
11&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 200&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR+&nbsp;&nbsp;&nbsp; dt=100<br>
12&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 210&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR+&nbsp;&nbsp;&nbsp; dt=10<br>
13&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 220&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR<br>
....<br>
<br>
With the extension of T-bit in CU, we can send:<br>
<br>
11'&nbsp;&nbsp;&nbsp;&nbsp; 200&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CU+&nbsp;&nbsp;&nbsp; dt=10<br>
12'&nbsp;&nbsp;&nbsp;&nbsp; 210&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR<br>
...<br>
<br>
We are trying to adopt CRTP to links that have some loss of packets and a
long round trip time (RT) <br>
We want to avoid the decompressor getting out of sync and having to send
CS because this will cause loss of many packets for this stream due to
the long RT.<br>
<br>
Let's check what happens if some packets are lost<br>
1. packet 4 was lost<br>
&nbsp;&nbsp;&nbsp; The decompressor will use the 'twice' and will stay in
sync<br>
2. packet 11 was lost<br>
&nbsp;&nbsp;&nbsp; The decompressor is out of sync<br>
3. packet 11'&nbsp; was lost<br>
&nbsp;&nbsp;&nbsp; The decompressor is out of sync<br>
<br>
We see that there is a weak point in the compression at the beginning of
a talk spurt<br>
We try to correct this by sending multiple CU+ <br>
<br>
11&quot;&nbsp;&nbsp;&nbsp; 200&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CU+&nbsp;&nbsp;&nbsp; dt=10<br>
12&quot;&nbsp;&nbsp;&nbsp; 210&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CU+&nbsp;&nbsp;&nbsp; dt=10 <br>
13&quot;&nbsp;&nbsp;&nbsp; 220&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR<br>
...<br>
<br>
If the decompressor received any of 11&quot;&nbsp; or 12&quot; ,&nbsp; he
is still in sync<br>
<br>
How many CU+ do we want to send at the beginning of a talk spurt?<br>
This depends on the quality of the link (which is good but not
perfect).<br>
If we know that on this link we may lose at most 2 packets in a row,
we'll send 3 CU+ at the beginning of a talk spurt<br>
The decompressor will stay in sync as long as he received at least one of
the CU+&nbsp; <br>
<br>
<br>
<blockquote type=cite cite>&nbsp;<br>
<br>
<b>Q2) Can you explain the use of&nbsp; &quot;The NON-RTP stream
flag&quot; for this algorithm</b> </blockquote><br>
The compressor knows that this stream is a non-RTP stream, but he still
wants to compress the stream. The compressor will have to use CU only. If
the compressor can pass a hint to the decompressor that this is not an
RTP stream, the decompressor will save only the IP and UDP headers for
this context, and will not save the extra context of the RTP header (that
does not exist here) and will not try to calculate the deltas and other
RTP-related calculations<br>
<br>
<br>
<b><blockquote type=cite cite>Q3) Section III : Acknowledgement to a new
compressed stream</b> <br>
<br>
Context Invalidation was one point which I had pointed out to Steve/tmima
long time back .. <br>
I am having those mails with me .. <br>
<br>
But, (i) <i>Do we need ACCEPT and REJECT packet&nbsp; ??</i> <br>
Having a NACK mechnism sounds reasonable .. <br>
I feel there shouldnt be both .. <br>
But what compressor is supposed to do after sending the first FULL_HEADER
packet <br>
when compressor waiting for the ACCEPT or REJECT <br>
should it queue the packets till the reply/or timeout <br>
or Should it send the compressed packets assuming the decompressor will
drop if there are <br>
REJECT state at the decompressor side. <br>
<br>
But (ii) <i>what about the backword compatibility issue </i>because this
ACCEPT/REJECT mechanism has dependancy and it will not work with
CRTP-2508 <br>
<br>
As per the current proposal we need to send ACCEPT/REJECT packet for
every <br>
FULL_HEADER packet ... (iii)<i>Is it not adding extra bandwidth
consuption </i>.. <br>
Draft talks about FULL_HEADER retransmit as well .. </blockquote><br>
The REJECT packet is useful when the decompressor runs out of resources.
The compressor initiates compression of a stream by sending a FH. With
CRTP today the decompressor can send a CS to invalidate the context, but
the compressor does not know the reason and will attempt to compress this
stream again. A REJECT will notify the compressor that the decompressor
cannot handle compresstion of additional streams at this moment.<br>
An implementation that was not enhanced to use REJECT will still use CS
to invalidate the context each time the compressor attempts to compress
the stream.<br>
The compressor continues to send the compressed packets of the stream
until he receives CS or REJECT<br>
The intent is to make the ACCEPT/REJECT packet optional - an
optimization.<br>
<br>
Tmima<br>
<br>
<br>
<blockquote type=cite cite>More questions after discussing these points
.. <br>
Please answer the questions ................... <br>
&nbsp; <br>
<br>
Regards <br>
<br>
GVJ <br>
&nbsp; <br>
<br>
--
---------------------------------------------------------------------------
<br>
Geevarghese John,&nbsp;&nbsp; DataCom Group, <br>
Motorola India Electronics Ltd, <br>
&quot;The Senate&quot;, No-33 A, <br>
Ulsoor Road, <br>
Bangalore - 42. <br>
Phone&nbsp; : 91-80-5598615&nbsp; Xnt-4005 <br>
E-mail : gvjohn@miel.mot.com <br>
-------------------------------------------------------------------------------
<br>
-------------------------------------------------------------------------------
<br>
&nbsp;</blockquote><br>
</html>

--=====================_1458145722==_.ALT--




From rem-conf Thu Nov 18 22:58:09 1999 
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Date: Fri, 19 Nov 1999 12:17:09 -0500
From: GeeVarghese John <gvjohn@miel.mot.com>
Organization: Motorola India Electronics Ltd.
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To: Tmima Koren <tmima@cisco.com>
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        pruddy@cisco.com, rem-conf@es.net, gvjohn@miel.mot.com
Subject: Re: Extensions to CRTP
References: <4.1.19991118024030.01aeaa50@jindo.cisco.com>
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Hi,

My comments to you answers are added below  ..
Please have a look at that  //

Tmima Koren wrote:

>
> At 02:13 PM 11/18/99 -0500, you wrote:
>
>> Steve, Tmima  & all,
>>
>> This is regarding the ietf draft - Extensions to CRTP  :
>> draft-koren-avt-crtp-enhance-00.txt
>> I have few doubts and could you please clarrify these points ..
>>
>> In the section 1. COMPRESSED_UDP packet "T bit" ..
>>
>> It is stated that CUDP packet reset the deltaT value to 0 and
>> therefore T bit avoids the need to send deltaT in the next CRTP.
>>
>> Q1) My doubt is If the entire original RTP header is the in the CUDP
>> packet why to
>> send the delta also with the original header ..
>> why can't the decompressor findout the new delta to be used
>> from the RTP header in the  CUDP packet itself, Is it not redundancy
>> ?
>>
>> Also You have stated that
>>
>> >> Multiple COMPRESSED_UDP packets may be sent by the compressor
>> when changes in the
>> >> compressor state need to be synchronized. Sending multiple
>> packets will increase the
>> >> probability of delivery of the synchronized state in the event of
>> packet loss.
>>
>> Why do we need to send multiple packet  and how it will increase the
>> probability even
>> if you send more CUDP packets, as long as the decomperssor look at
>> the seq number
>> and detect  packet loss  it should send a CONTEXT_STATE packet.
>>
>> If the draft reference you made was for loss of more than 16 packets
>> from the same flow,
>> and the possible wrong decompression of headers ..
>> twice algorithm should detect this (But twice algo is not used if
>> checksum is 0)
>> So here what exactly the limit could be , the sequence number space
>> 1..16
>> Since this draft address the use of this algo over ATM/AAL5 links  I
>> think
>> sequence number space 1..16 may not be sufficient.
>
>
> Some notations:
> dt         deltaT
> FH        FULL_HEADER
> CR       COMPRESSED_RTP
> CR+     COMPRESSED_RTP with deltaT (dt)
> CU       COMPRESSED_UDP
> CU+     COMPRESSED_UDP with dt
> CS       CONTEXT_STATE
>
> I'll use an example
> The sender sends a sample every 10 ms
> Using CRTP, the sequence of packets will be:
>
> seq#   Time     pkt type
> 1        10         FH
> 2        20         CR+    dt=10
> 3        30         CR
> ...
> 10      100       CR
>
> At this point this side stops talking. Talking resumes at time 200
>
> 11      200       CR+    dt=100
> 12      210       CR+    dt=10
> 13      220       CR
> ....
>
> With the extension of T-bit in CU, we can send:
>
> 11'     200       CU+    dt=10
> 12'     210       CR
> ...
>
> We are trying to adopt CRTP to links that have some loss of packets
> and a long round trip time (RT)
> We want to avoid the decompressor getting out of sync and having to
> send CS because this will cause loss of many packets for this stream
> due to the long RT.
>
> Let's check what happens if some packets are lost
> 1. packet 4 was lost
>     The decompressor will use the 'twice' and will stay in sync
> 2. packet 11 was lost
>     The decompressor is out of sync
> 3. packet 11'  was lost
>     The decompressor is out of sync
>
> We see that there is a weak point in the compression at the beginning
> of a talk spurt
> We try to correct this by sending multiple CU+
>
> 11"    200       CU+    dt=10
> 12"    210       CU+    dt=10
> 13"    220       CR

Here, I think you assumed that 12"  packet will reach the compessor with

timestamp 210. But, unfortunately if it was lost what you get is 13" and
time stamp 220.
In that case again you have send dt again with your CU+ after sending
CU+ with dt 10.
In this type of situation processing due to delta encoding was double
and you send two
delta encoded bytes in two different packets insteadof one
You might have assumed that MOST of  the case packet loss doesnot happen
..
But I feel that is also purely dependant on the network where this
algorithm is executing ..

>
> ...
>
> If the decompressor received any of 11"  or 12" ,  he is still in sync

I think this is not correct, If the decompressor receives 12" insted of
11",
he is not in sync because of link seq number mismatch and
will send a CONTEXT_STATE packet.
This will lead to FH sending from compressor to decompressor ..

>
>
> How many CU+ do we want to send at the beginning of a talk spurt?
> This depends on the quality of the link (which is good but not
> perfect).
> If we know that on this link we may lose at most 2 packets in a row,
> we'll send 3 CU+ at the beginning of a talk spurt

Looks OK, but How do we decide on the number of packet loss ?
Packet loss can be due to link quality and/or network congestion etc ..
If the loss was due network congestion or similar factor, behaviour vary
>from time to time ..
So here also CU+ is giving disadvantage instead of the benifits
expected.


>
> The decompressor will stay in sync as long as he received at least one
> of the CU+

I think any of the CU+ loss will lead to CONTEXT_STATE packet generation
from
decompressor to compressor because of the link seq number mismatch ..Is
nt it ?

>

I will include rest of the questions in another mail ..
Please reply back ..

Rgds
GVJ




--------------273F322522180A003FD42464
Content-Type: text/html; charset=us-ascii
Content-Transfer-Encoding: 7bit

<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Hi,
<p>My comments to you answers are added below&nbsp; ..
<br>Please have a look at that&nbsp; //
<p>Tmima Koren wrote:
<blockquote TYPE=CITE>&nbsp;
<br>At 02:13 PM 11/18/99 -0500, you wrote:
<blockquote type=cite cite>Steve, Tmima&nbsp; &amp; all,
<p>This is regarding the ietf draft - Extensions to CRTP&nbsp; : draft-koren-avt-crtp-enhance-00.txt
<br>I have few doubts and could you please clarrify these points ..
<p>In the section 1. COMPRESSED_UDP packet "T bit" ..
<p>It is stated that CUDP packet reset the deltaT value to 0 and
<br>therefore T bit avoids the need to send deltaT in the next CRTP.
<p>Q1) <b>My doubt is If the entire original RTP header is the in the CUDP
packet why to</b>
<br><b>send the delta also with the original header ..</b>
<br><b>why can't the decompressor findout the new delta to be used</b>
<br><b>from the RTP header in the&nbsp; CUDP packet itself, Is it not redundancy
?</b>
<p>Also You have stated that
<p>>> Multiple COMPRESSED_UDP packets may be sent by the compressor when
changes in the
<br>>> compressor state need to be synchronized. Sending multiple packets
will increase the
<br>>> probability of delivery of the synchronized state in the event of
packet loss.
<p><i>Why do we need to send multiple packet</i>&nbsp; and <i>how it will
increase the probability even</i>
<br>if you send more CUDP packets, as long as the decomperssor look at
the seq number
<br>and detect&nbsp; packet loss&nbsp; it should send a CONTEXT_STATE packet.
<p>I<i>f the draft reference you made was for loss of more than 16 packets
>from the same flow,</i>
<br><i>and the possible wrong decompression of headers ..</i>
<br><i>twice algorithm should detect this (But twice algo is not used if
checksum is 0)</i>
<br><i>So here what exactly the limit could be , the sequence number space
1..16</i>
<br><i>Since this draft address the use of this algo over ATM/AAL5 links&nbsp;
I think</i>
<br><i>sequence number space 1..16 may not be sufficient.</i></blockquote>

<p><br>Some notations:
<br>dt&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; deltaT
<br>FH&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; FULL_HEADER
<br>CR&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; COMPRESSED_RTP
<br>CR+&nbsp;&nbsp;&nbsp;&nbsp; COMPRESSED_RTP with deltaT (dt)
<br>CU&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; COMPRESSED_UDP
<br>CU+&nbsp;&nbsp;&nbsp;&nbsp; COMPRESSED_UDP with dt
<br>CS&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; CONTEXT_STATE
<p>I'll use an example
<br>The sender sends a sample every 10 ms
<br>Using CRTP, the sequence of packets will be:
<p>seq#&nbsp;&nbsp; Time&nbsp;&nbsp;&nbsp;&nbsp; pkt type
<br>1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 10&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
FH
<br>2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 20&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR+&nbsp;&nbsp;&nbsp; dt=10
<br>3&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 30&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR
<br>...
<br>10&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 100&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR
<p>At this point this side stops talking. Talking resumes at time 200
<p>11&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 200&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR+&nbsp;&nbsp;&nbsp; dt=100
<br>12&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 210&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR+&nbsp;&nbsp;&nbsp; dt=10
<br>13&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 220&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR
<br>....
<p>With the extension of T-bit in CU, we can send:
<p>11'&nbsp;&nbsp;&nbsp;&nbsp; 200&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CU+&nbsp;&nbsp;&nbsp; dt=10
<br>12'&nbsp;&nbsp;&nbsp;&nbsp; 210&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR
<br>...
<p>We are trying to adopt CRTP to links that have some loss of packets
and a long round trip time (RT)
<br>We want to avoid the decompressor getting out of sync and having to
send CS because this will cause loss of many packets for this stream due
to the long RT.
<p>Let's check what happens if some packets are lost
<br>1. packet 4 was lost
<br>&nbsp;&nbsp;&nbsp; The decompressor will use the 'twice' and will stay
in sync
<br>2. packet 11 was lost
<br>&nbsp;&nbsp;&nbsp; The decompressor is out of sync
<br>3. packet 11'&nbsp; was lost
<br>&nbsp;&nbsp;&nbsp; The decompressor is out of sync
<p>We see that there is a weak point in the compression at the beginning
of a talk spurt
<br>We try to correct this by sending multiple CU+
<p>11"&nbsp;&nbsp;&nbsp; 200&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; CU+&nbsp;&nbsp;&nbsp;
dt=10
<br>12"&nbsp;&nbsp;&nbsp; 210&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; CU+&nbsp;&nbsp;&nbsp;
dt=10
<br>13"&nbsp;&nbsp;&nbsp; 220&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; CR</blockquote>
Here, I think you assumed that 12"&nbsp; packet will reach the compessor
with
<br>timestamp 210. But, unfortunately if it was lost what you get is 13"
and time stamp 220.
<br>In that case again you have send dt again with your CU+ after sending
CU+ with dt 10.
<br>In this type of situation <b>processing due to delta encoding was double</b>
and you send <b>two</b>
<br><b>delta encoded bytes in two different packets insteadof one</b>
<br>You might have assumed that MOST of&nbsp; the case packet loss doesnot
happen ..
<br>But I feel that is also purely dependant on the network where this
algorithm is executing ..
<blockquote TYPE=CITE>&nbsp;
<br>...
<p>If the decompressor received any of 11"&nbsp; or 12" ,&nbsp; he is still
in sync</blockquote>
I think this is not correct, If the decompressor receives 12" insted of
11",
<br>he is not in sync because of link seq number mismatch and
<br>will send a CONTEXT_STATE packet.
<br>This will lead to FH sending from compressor to decompressor ..
<blockquote TYPE=CITE>&nbsp;
<p>How many CU+ do we want to send at the beginning of a talk spurt?
<br>This depends on the quality of the link (which is good but not perfect).
<br>If we know that on this link we may lose at most 2 packets in a row,
we'll send 3 CU+ at the beginning of a talk spurt</blockquote>
Looks OK, but How do we decide on the number of packet loss ?
<br>Packet loss can be due to link quality and/or network congestion etc
..
<br>If the loss was due network congestion or similar factor, behaviour
vary from time to time ..
<br>So here also CU+ is giving disadvantage instead of the benifits expected.
<br>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<br>The decompressor will stay in sync as long as he received at least
one of the CU+</blockquote>
I think any of the CU+ loss will lead to CONTEXT_STATE packet generation
from
<br>decompressor to compressor because of the link seq number mismatch
..Is nt it ?
<blockquote TYPE=CITE>&nbsp;</blockquote>
I will include rest of the questions in another mail ..
<br>Please reply back ..
<p>Rgds
<br>GVJ
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;</html>

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Date: Fri, 19 Nov 1999 12:51:02 -0500
From: GeeVarghese John <gvjohn@miel.mot.com>
Organization: Motorola India Electronics Ltd.
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To: Tmima Koren <tmima@cisco.com>, casner@cisco.com, micke@cdt.luth.se,
        brucet@cisco.com, dwing@cisco.com, pruddy@cisco.com, rem-conf@es.net,
        gvjohn@miel.mot.com, pazhynnr@cig.mot.com
Subject: Re: Extensions to CRTP
References: <4.1.19991118024030.01aeaa50@jindo.cisco.com> <38358615.D8679DA0@miel.mot.com>
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Sorry, Please neglect my previous mail, this is the correct mail ..
I pasted my answers at the wrong place which might confuse those who
read ..

GVJ

GeeVarghese John wrote:

> Hi,
>
> My comments to you answers are added below  ..
> Please have a look at that  //
>
> Tmima Koren wrote:
>
>>
>> At 02:13 PM 11/18/99 -0500, you wrote:
>>
>> > Steve, Tmima  & all,
>> >
>> > This is regarding the ietf draft - Extensions to CRTP  :
>> > draft-koren-avt-crtp-enhance-00.txt
>> > I have few doubts and could you please clarrify these points ..
>> >
>> > In the section 1. COMPRESSED_UDP packet "T bit" ..
>> >
>> > It is stated that CUDP packet reset the deltaT value to 0 and
>> > therefore T bit avoids the need to send deltaT in the next CRTP.
>> >
>> > Q1) My doubt is If the entire original RTP header is the in the
>> > CUDP packet why to
>> > send the delta also with the original header ..
>> > why can't the decompressor findout the new delta to be used
>> > from the RTP header in the  CUDP packet itself, Is it not
>> > redundancy ?
>> >
>> > Also You have stated that
>> >
>> > >> Multiple COMPRESSED_UDP packets may be sent by the compressor
>> > when changes in the
>> > >> compressor state need to be synchronized. Sending multiple
>> > packets will increase the
>> > >> probability of delivery of the synchronized state in the event
>> > of packet loss.
>> >
>> > Why do we need to send multiple packet  and how it will increase
>> > the probability even
>> > if you send more CUDP packets, as long as the decomperssor look at
>> > the seq number
>> > and detect  packet loss  it should send a CONTEXT_STATE packet.
>> >
>> > If the draft reference you made was for loss of more than 16
>> > packets from the same flow,
>> > and the possible wrong decompression of headers ..
>> > twice algorithm should detect this (But twice algo is not used if
>> > checksum is 0)
>> > So here what exactly the limit could be , the sequence number space
>> > 1..16
>> > Since this draft address the use of this algo over ATM/AAL5 links
>> > I think
>> > sequence number space 1..16 may not be sufficient.
>>
>>
>> Some notations:
>> dt         deltaT
>> FH        FULL_HEADER
>> CR       COMPRESSED_RTP
>> CR+     COMPRESSED_RTP with deltaT (dt)
>> CU       COMPRESSED_UDP
>> CU+     COMPRESSED_UDP with dt
>> CS       CONTEXT_STATE
>>
>> I'll use an example
>> The sender sends a sample every 10 ms
>> Using CRTP, the sequence of packets will be:
>>
>> seq#   Time     pkt type
>> 1        10         FH
>> 2        20         CR+    dt=10
>> 3        30         CR
>> ...
>> 10      100       CR
>>
>> At this point this side stops talking. Talking resumes at time 200
>>
>> 11      200       CR+    dt=100
>> 12      210       CR+    dt=10
>> 13      220       CR
>> ....
>>
>> With the extension of T-bit in CU, we can send:
>>
>> 11'     200       CU+    dt=10
>> 12'     210       CR
>
Here, I think you assumed that 12'  packet will reach the compessor with

timestamp 210. But, unfortunately if it was lost what you get is 13' and
time stamp 220.
In that case again you have send dt again with your CU+ after sending
CU+ with dt 10.
In this type of situation processing due to delta encoding was double
and you send two
delta encoded bytes in two different packets insteadof one
You might have assumed that MOST of  the case packet loss doesnot happen
..
But I feel that is also purely dependant on the network where this
algorithm is executing ..

And also are you not loosing the bandwidth saving by sending CU+ instead
of CR
because  you send the complete RTP hdr + delta encoded deltaT byte in
CU+.


>>
>> ...
>>
>> We are trying to adopt CRTP to links that have some loss of packets
>> and a long round trip time (RT)
>> We want to avoid the decompressor getting out of sync and having to
>> send CS because this will cause loss of many packets for this stream
>> due to the long RT.
>>
>> Let's check what happens if some packets are lost
>> 1. packet 4 was lost
>>     The decompressor will use the 'twice' and will stay in sync
>> 2. packet 11 was lost
>>     The decompressor is out of sync
>> 3. packet 11'  was lost
>>     The decompressor is out of sync
>>
>> We see that there is a weak point in the compression at the
>> beginning of a talk spurt
>> We try to correct this by sending multiple CU+
>>
>> 11"    200       CU+    dt=10
>> 12"    210       CU+    dt=10
>> 13"    220       CR
>
>>
>> ...
>>
>> If the decompressor received any of 11"  or 12" ,  he is still in
>> sync
>
I think this is not correct, If the decompressor receives 12" insted of
11",
he is not in sync because of link seq number mismatch and
will send a CONTEXT_STATE packet.
This will lead to FH sending from compressor to decompressor ..

>
>
>>
>>
>> How many CU+ do we want to send at the beginning of a talk spurt?
>> This depends on the quality of the link (which is good but not
>> perfect).
>> If we know that on this link we may lose at most 2 packets in a row,
>> we'll send 3 CU+ at the beginning of a talk spurt
>
Looks OK, but How do we decide on the number of packet loss ?
Packet loss can be due to link quality and/or network congestion etc ..
If the loss was due network congestion or similar factor, behaviour vary
>from time to time ..
So here also CU+ is giving disadvantage instead of the benifits
expected.

>>
>> The decompressor will stay in sync as long as he received at least
>> one of the CU+
>
I think any of the CU+ loss will lead to CONTEXT_STATE packet generation
from
decompressor to compressor because of the link seq number mismatch ..Is
nt it ?


>
>
>>
>
> I will include rest of the questions in another mail ..
> Please reply back ..
>
> Rgds
> GVJ
>
>
>



--------------44AB1B42F5C07FDF6CBF0C1E
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Sorry, Please neglect my previous mail, this is the correct mail ..
<br>I pasted my answers at the wrong place which might confuse those who
read ..
<p>GVJ
<p>GeeVarghese John wrote:
<blockquote TYPE=CITE>Hi,
<p>My comments to you answers are added below&nbsp; ..
<br>Please have a look at that&nbsp; //
<p>Tmima Koren wrote:
<blockquote TYPE=CITE>&nbsp;
<br>At 02:13 PM 11/18/99 -0500, you wrote:
<blockquote type=cite cite>Steve, Tmima&nbsp; &amp; all,
<p>This is regarding the ietf draft - Extensions to CRTP&nbsp; : draft-koren-avt-crtp-enhance-00.txt
<br>I have few doubts and could you please clarrify these points ..
<p>In the section 1. COMPRESSED_UDP packet "T bit" ..
<p>It is stated that CUDP packet reset the deltaT value to 0 and
<br>therefore T bit avoids the need to send deltaT in the next CRTP.
<p>Q1) <b>My doubt is If the entire original RTP header is the in the CUDP
packet why to</b>
<br><b>send the delta also with the original header ..</b>
<br><b>why can't the decompressor findout the new delta to be used</b>
<br><b>from the RTP header in the&nbsp; CUDP packet itself, Is it not redundancy
?</b>
<p>Also You have stated that
<p>>> Multiple COMPRESSED_UDP packets may be sent by the compressor when
changes in the
<br>>> compressor state need to be synchronized. Sending multiple packets
will increase the
<br>>> probability of delivery of the synchronized state in the event of
packet loss.
<p><i>Why do we need to send multiple packet</i>&nbsp; and <i>how it will
increase the probability even</i>
<br>if you send more CUDP packets, as long as the decomperssor look at
the seq number
<br>and detect&nbsp; packet loss&nbsp; it should send a CONTEXT_STATE packet.
<p>I<i>f the draft reference you made was for loss of more than 16 packets
>from the same flow,</i>
<br><i>and the possible wrong decompression of headers ..</i>
<br><i>twice algorithm should detect this (But twice algo is not used if
checksum is 0)</i>
<br><i>So here what exactly the limit could be , the sequence number space
1..16</i>
<br><i>Since this draft address the use of this algo over ATM/AAL5 links&nbsp;
I think</i>
<br><i>sequence number space 1..16 may not be sufficient.</i></blockquote>

<p><br>Some notations:
<br>dt&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; deltaT
<br>FH&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; FULL_HEADER
<br>CR&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; COMPRESSED_RTP
<br>CR+&nbsp;&nbsp;&nbsp;&nbsp; COMPRESSED_RTP with deltaT (dt)
<br>CU&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; COMPRESSED_UDP
<br>CU+&nbsp;&nbsp;&nbsp;&nbsp; COMPRESSED_UDP with dt
<br>CS&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; CONTEXT_STATE
<p>I'll use an example
<br>The sender sends a sample every 10 ms
<br>Using CRTP, the sequence of packets will be:
<p>seq#&nbsp;&nbsp; Time&nbsp;&nbsp;&nbsp;&nbsp; pkt type
<br>1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 10&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
FH
<br>2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 20&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR+&nbsp;&nbsp;&nbsp; dt=10
<br>3&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 30&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR
<br>...
<br>10&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 100&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR
<p>At this point this side stops talking. Talking resumes at time 200
<p>11&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 200&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR+&nbsp;&nbsp;&nbsp; dt=100
<br>12&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 210&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR+&nbsp;&nbsp;&nbsp; dt=10
<br>13&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 220&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR
<br>....
<p>With the extension of T-bit in CU, we can send:
<p>11'&nbsp;&nbsp;&nbsp;&nbsp; 200&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CU+&nbsp;&nbsp;&nbsp; dt=10
<br>12'&nbsp;&nbsp;&nbsp;&nbsp; 210&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR</blockquote>
</blockquote>

<p>Here, I think you assumed that 12'&nbsp; packet will reach the compessor
with
<br>timestamp 210. But, unfortunately if it was lost what you get is 13'
and time stamp 220.
<br>In that case again you have send dt again with your CU+ after sending
CU+ with dt 10.
<br>In this type of situation <b>processing due to delta encoding was double</b>
and you send <b>two</b>
<br><b>delta encoded bytes in two different packets insteadof one</b>
<br>You might have assumed that MOST of&nbsp; the case packet loss doesnot
happen ..
<br>But I feel that is also purely dependant on the network where this
algorithm is executing ..
<p>And also are you not loosing the bandwidth saving by sending CU+ instead
of CR
<br>because&nbsp; you send the complete RTP hdr + delta encoded deltaT
byte in CU+.
<br>&nbsp;
<blockquote TYPE=CITE>
<blockquote TYPE=CITE>&nbsp;
<br>...
<p>We are trying to adopt CRTP to links that have some loss of packets
and a long round trip time (RT)
<br>We want to avoid the decompressor getting out of sync and having to
send CS because this will cause loss of many packets for this stream due
to the long RT.
<p>Let's check what happens if some packets are lost
<br>1. packet 4 was lost
<br>&nbsp;&nbsp;&nbsp; The decompressor will use the 'twice' and will stay
in sync
<br>2. packet 11 was lost
<br>&nbsp;&nbsp;&nbsp; The decompressor is out of sync
<br>3. packet 11'&nbsp; was lost
<br>&nbsp;&nbsp;&nbsp; The decompressor is out of sync
<p>We see that there is a weak point in the compression at the beginning
of a talk spurt
<br>We try to correct this by sending multiple CU+
<p>11"&nbsp;&nbsp;&nbsp; 200&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; CU+&nbsp;&nbsp;&nbsp;
dt=10
<br>12"&nbsp;&nbsp;&nbsp; 210&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; CU+&nbsp;&nbsp;&nbsp;
dt=10
<br>13"&nbsp;&nbsp;&nbsp; 220&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; CR</blockquote>

<blockquote TYPE=CITE>&nbsp;
<br>...
<p>If the decompressor received any of 11"&nbsp; or 12" ,&nbsp; he is still
in sync</blockquote>
</blockquote>
I think this is not correct, If the decompressor receives 12" insted of
11",
<br>he is not in sync because of link seq number mismatch and
<br>will send a CONTEXT_STATE packet.
<br>This will lead to FH sending from compressor to decompressor ..
<blockquote TYPE=CITE>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<p>How many CU+ do we want to send at the beginning of a talk spurt?
<br>This depends on the quality of the link (which is good but not perfect).
<br>If we know that on this link we may lose at most 2 packets in a row,
we'll send 3 CU+ at the beginning of a talk spurt</blockquote>
</blockquote>
Looks OK, but How do we decide on the number of packet loss ?
<br>Packet loss can be due to link quality and/or network congestion etc
..
<br>If the loss was due network congestion or similar factor, behaviour
vary from time to time ..
<br>So here also CU+ is giving disadvantage instead of the benifits expected.
<blockquote TYPE=CITE>
<blockquote TYPE=CITE>&nbsp;
<br>The decompressor will stay in sync as long as he received at least
one of the CU+</blockquote>
</blockquote>
I think any of the CU+ loss will lead to CONTEXT_STATE packet generation
from
<br>decompressor to compressor because of the link seq number mismatch
..Is nt it ?
<br>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<blockquote TYPE=CITE>&nbsp;</blockquote>
I will include rest of the questions in another mail ..
<br>Please reply back ..
<p>Rgds
<br>GVJ
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;</blockquote>

<br>&nbsp;</html>

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From: Robert Tan <tjsh@netchina.com.cn>
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Dear Sir: This is my discussion article.
The title:
                   Flash Bandwidth 1KHz to 100MHz
                         Digital Controlled Broadband
                     Anti-alias & Reconstruction Filter
The details:
http://www.cnindex.net/~tjsh/Base_of_Broadband_Access.html
http://www.cnindex.net/~tjsh/Base_of_Broadband_Access.doc

Sincerely,

Robert Tan
11.22


Flash Bandwidth 1KHz to 100MHz
Digital Controlled Broadband
Anti-alias & Reconstruction Filter
The Next Era Web of Multimedia Data-stream
Transmission Solution of the Internet!

By Robert Tan
Update: 1999. 11. 22

For the many more format and definite of standard of digital film and
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such as the Broadband Cable Internet Access or HFC users, one of them
would like to get a
variable data speed according to the applications of his needed. It will
provide a lot of
digital control frequency bandwidth which is separated one another in
HFC transmission system,
and it will more effective for the "hot clients", such as the VOD video
clients or the other
high-code rate users and high-speed data stream user's applications. The
FBW will improve the
speed of the data stream, diminish the amount of the spare resource
seizing of "Cool Clients",
reduce the calculations of the DSP processor of the center web main
switching system and the
branch switching system. Reduce the CPU utilities of the applications in
all of client
communication adapter and its cost of produce is very important. The
"Cool Clients" mains
the IP telephone users or the other lower speed code rate audio
applications. Anyway, the
speed of the up-stream data and down-stream data speed could be changed
to any value you
needed, and the changing degree will be very smoothly and very simply in
writing one or
two bytes in the command system of communication web switcher. By the
high-speed frequency
variable characteristic, if the FBW filter were used in your FTTX or HFC
switching system,
the upward and downward data stream speed could be managed and attempted
by your command
system. So that, the transmission bandwidth will be able to adjusted to
your needed in any
time within one millionth second. The real-time will be very more
improved and more effective
on the communication main roads such as FTTX and HFC system,
transmission controlling system
will completely control the data stream in real-time.



Dear Technology Research and Development Administrator:
   I am an electronic engineer in the field of Analog and Digital
filters researching and
designing. I had been a DSP engineer for ten years. My development field
is electronic circuits
-- digital and analog hardware designing and researching in applications
of Digital Signal
Processing. My works contains the frequency spectrum analysis, digital
audio and digital video
systems, noise spectrum controlling and processing, and the other
military applications. I have
researched and designed several kind of analog and digital anti-alias &
reconstruction filters.
A few years before, in my R&D works, I find the special frequency of the
filters is usually
fixed, it could be very difficult to change the cutoff frequency of the
low-pass anti-alias
& reconstruction filter. But the fixed filter could not suit the target
of my technical
applications. If we want to get several special frequency of the
filters, we must to use
several different fixed filters or instead of fixed filter with
switching capacity filter
or digital filter, such as IIR or FIR filter, and their frequency must
be variable in order
to change the analysis frequency. It is very important for the spectrum
analysis system and
the digital random real time vibrated controlling system. But, in this
way, we must pay for
digital IIR or FIR filter, it is very expensive in the price, take a
large space for the DSP
chips and its outside memory banks to install it. So, we would get into
much troubles and
complex questions from a simple problem.
However, we have the switching capacity technologies, and the product of
up to 8th order
filters is a very normally used. But the switching capacity filter will
have a high noise
and low frequency range, and it has a non-exactly special frequency,
generally. The special
frequency of most of the switching capacity filters is from 1/95 to
1/105 times of its main
switching clock and it is difficult to improve its accuracy. Actually,
the switching capacity
filter circuit is very difficult to adjust and use in applications.
Anywise, its order is
normally below 8th order, and the special frequency and bandwidth of the
switching capacity
filters is below to 50KHz, and the amplitude response is not exactly
yet, the dynamic range
of the amplitude is very low, normally, it is below the 48dB. Except of
frequency range from
0 to 50Hz, in this field, its dynamic range is lower than 30 dB and not
usefully, it is
normally being used in the speech processing circuit and the other lower
frequency band
processing systems. Attention, all of this analysis is not including the
phase noise of the
main switching clock and the noise of clock feed through yet. The
others, it is difficult to
get much more analysis frequency bands or special frequency points for
the switching capacity
filters, it is especially for the higher frequency band which is
approached to the top
frequency point. All of these characteristics of the switching capacity
filters will restrict
its applications in the area of modern digital communications.
The others method of the filters designing is the contact-time analog
filter .I have found
some method. It could make many transfer functions. Such as the
elliptical function,
Chebyshev function and Butterworth function, but there is some
interesting things here,
it is that the same circuit frame could be built into many transfer
functions, and it
would have programmable (digital controlled) special frequency. Changing
special frequency
on line is available. It could get the very wide frequency range, very
quickly to change
the used band, and the digital controlled circuit is very directly and
simply, the special
frequency step is very smoothly (because the Frequency is controlled by
long bit Digital
multiplier). In that points, integrate the FBW filter circuit technology
is very useful
for the digital controlling and processing systems. Because of the
Digital Controlling
Flash Bandwidth Filer (FBW) is depend on a few chips of high quality 8
or 12 bits long
Digital Multiplier and a few chips of high-speed amplifiers. The other,
this filter
technology is used with a few exactitude resistors and exactitude
capacitors also. For
example, an 8th order elliptical filter need 8chips of dual Digital
Multiplier and several
chips high bandwidth amplifiers, 8chips exactitude capacitors and
several chips exactitude
resisters. If it is possible, integrating all the Digital Multiplier and
amplifiers into
one chip or one plastic package (mixed circuit), place all the
exactitude resistors and
capacitors out of the package. It will be finest filter and very easy to
used and could
be re-designed agilely by the customers and the OEMs. The parameter of
the filters is
depending on the value of the resistors and the capacitors. Its value
could be designed
by the customs freely and calculated by constant functions directly, and
that is very
simply.
And then, the FBW filters would be very usefully in the FTTX or HFC
networks. For working
with the 256QAM or 64VSB modulation technology and any narrow-band
transmission technology
systems, such as the Broadband Cable Internet Access webs, HDTV or SDTV
and VOD systems, it
would compress the used frequency bandwidth in mostly extent. Because of
the WAN systems in
any nations will be depending on the "Tree structures" as the main road
in the webs in the
future. The frequency source is very expensive in this web, It would be
welcomed in the
Internet web digital information transmission, exchanging and switching
technology area,
DSP area and digital communications area, such as MAN and WAN coax cable
webs or the FTTX
system applications. Any way, the most of modern networking transmission
mode is TDM. It
is very suitable for the text material or media, because of it is a
fixed media in size,
and it is the fixed continue time in length and generation procedure.
But the most of
multimedia is not to be suitable with that. The data stream of voice and
video will have
no constant size to be forecasted has no constant procedure time. It is
only have a constant
and fixed bandwidth of media data stream. Because of your interesting
points is onto the
generations and the procedures of the video or audio information of thus
material or the
news, such as the high quality digital cinema, digital musicale and so
on. Certainly, you
could take the any multimedia information and any moving picture on you
desktop through the
Internet Web. It will be appeared the true alive world with you in any
site you can go! So
that, the only one of best way of the media data streams transmission
and exchanging is the
FDM mode with the SSB-ASK or VSB-ASK technology, it should be working
with the QAM or
VSB- ASK digital modulation technology in the future Internet webs.
FBW filter would be used in most of switchers and routers, which is
depending on the FDM
technologies. For the cable TV systems STB (set top box), video & audio
servers in Broadband
Cable Internet Access, cable modems or the client-end adapters in the
most of family users
and the most of business cable users in the world will need a very great
deal of broadband
web transmission bandwidth. Because of that, in any modern
home-electronic equipment, such
as the Internet officers, shopping's and the VOD or the other broadband
information
electronics, its using method would be depending on the FDM switching
technologies. So that,
the "Online Home-Electronics" or the other Internet accessing products
which is used in
people's homes would be simply to operate and easy to use. And it must
be depend on the
simply low-layer hardware equipment and protocols of the webs, such as
the VOD systems
and the STB terminals in the Broadband Cable Internet Access or HFC
networks. The FBW
technology would provide us more applicable and useable method in the
physics layer of
the public HFC networks, its FDM applications in a broadband web get us
in a lower building
price, more effective than the other technology. Such as the Broadband
Cable Internet Access
webs, HFC networks, FTTX networks, and the other and the other WAN
public broadband networks
would have a very great applications of the FBW filters.
The others, using a Flash-Band-width filter will help you get in to a
fully data stream
bandwidth management of any expensive data-link channels and the very
expensive communication
data stream bandwidth, such as the communication data-link of the
satellite communication
systems, and the ocean bottom communication systems. FBW filter will
control the any leased
data-stream bandwidth in dynamic mode within a very large frequency
range, a large range of
signal frequency bandwidth and data-stream speed in any time for a
communication administration
and operation system. It will change the bandwidth with you leased data
stream very imminently
within several microsecond, improve your expensive bandwidth of
data-stream transmission channel,
save your leased payment and money cost for any customers of digital
communications and any
Tele-communication operating and service company. It will hold any
important transmission of
data-stream in smoothly and take it freely. So, in this system, any
interrupt of your
communication data stream will be never.



The Digital Control Flash Bandwidth Filter specifications:
(1) General details:
Frequency range:   0.1Hz--100MHz
Useable frequency range:  0.1Hz--100MHz(at least)
Special frequency step:  0.001Hz-1000KHz
Transfer functions:   Elliptical, Butterworth, Chebyshev, and Bassel
function.
       And the others contacted-time filters.
Available filter type:  Low-pass filter, High-pass filter, Notch filter,

       Band-pass filter and All-pass filter.
Noise Level:  -160dB(max) (Only depend on the number of the used
amplifiers.)
Order range: 2nd--12th(Depend on the sensitivity of the transfer
functions.)
Filter switching time:  Less than 300nS (Filter Setting time is 200nS)
Useful Range:    Anti-alias filters, Reconstruction filters.

(2) Pass Band Specifications:
Frequency selectable range:  100Hz--100MHz
(Depend on the performance of the selected amplifiers)
Special Frequency steps:  1Hz-1000KHz
Pass band Dynamic range: 90dB (Depend on the amplifiers and the order of
the filter.)
Ripple:  0.01--1.0dB (Only depend on the filter function and the
passband ripple designed.)

(3) Pass to Stop Band:
Drop speed of Interim-band Cut-off:
180db/oct (8th order elliptical function with 0.05dB pass-band ripple)

(4) Stop Band:
Amplitude min drop:  120db(8th elliptical filter for example)
Frequency range:   0.1Hz--100MHz
(Depend on the amplifiers and the Digital Multiplier specifications)

(5) Digital Control specifications:
Digital component is used (8 bit, 10bit, 12bit, and 16bit is available).

The special frequency is (N1/N2) * Frequency (ref).
(The N1, N2 is the Digital component input byte, from 8bit to 16 bit.)
The Frequency (ref) is form 10Hz to 10MHz.
(This parameter is depends on one RC time constant.)
High speed and voltage feedback high-bandwidth operation amplifiers are
required.

(6) Requirement: (The filter section of 8th order)
Digital component:    8 chips (Selections depend on the frequency range)

High-performance Amplifiers: 12 to 24 chips (Selections depend on the
frequency)
Capacitors or inductors:   8 chips (exactitude degree value is 1% to
0.1%)
Resistors:          16-36 chips (exactitude degree value is 1% to 0.1%)

(7) Group delay:                Depend on the function of the filter
designed, filter
                                phase response designed, and the filter
order.

(8) Filter switching time:  Less than 300nS (Filter Setting time is
200nS)




   The Comparisons of the 4 kinds active Filters

For example:
8th order filter
Switching Capacity Filter
Digital Filter
(FIR or IIR Filter)
Traditional
Fixed Analog Frequency Filter
Digital controlling FBW filter

Noise Level or  (THD+N) Value:

Highest

-48db
Lower

-70db
Very Low

-160db
Very Low

-120db
Filter Dynamic Range
The Value:
Little

50db
Middle

70db
Very Large

160db
Large

100db
Real-time          active frequency
The range of the value:
Narrow

200Hz to 300KHz
Wide

0.001Hz to  50KHz
Very Wide

0.1Hz   to 1MHz
Very wide

0.001Hz    to 100MHz
The Outside in advance  Anti-alias & Reconstruction assistant filter
requirement

The degree of assistant filter Quality
Required





2nd to 4th order  anti-alias & reconstruction assistant outside filter
Required





4th to 6th order anti-alias & reconstruction   assistant outside filter
Not required.





----------
Not required





----------
The Precision of the special  frequency:
The ability of on-line changing the filter special frequency
&Method
Low.

+/-3%

Very easy
(Changing the switching clock)
High

+/-3%

Very difficult
(Change a new programs and
parameters )
Very high

0.1%

Very difficult.
(Changing the system time
constant)
Very high

0.1%

Easy(Writing  digital word to the filter control port)
The complex degree of the filter system designing
&The used space
Simple



Small
Very complex



Very large
Simple



Middle
Middle



Middle
The price for  produce a filter
&The difficult degree of the applications
Very low

$10~50

Easy
Very high

$300~600

Difficult
Low

$30~90

Easy
Middle

$50~100

Middle
Anti-alias &  Reconstruction filter Performance: (1)Pass Band
Ripple &Dynamic Range:
(2)Stop Band
Rejection
(3)Interim Band Dropping Slope:
Low Quality & Low Price.


+/-0.2db

50db

55db

50db/oct
Normal or High Quality & High price

+/-0.01db

70db

70db

130db/oct
Normal Quality  & Low Price


+/-0.01db

120db

120db

180db/oct
High Quality  & & Middle Price.


+/-0.01db

110db

120db

180db/oct
Online/offline Changing of highest/lowest special frequency rate &
Range:
&changing time:
All available
1000 times
300Hz to 300KHz

10mS(Time of PLL tracking & locked)
Offline available
Not available

Not available

Not available
(Reboot DSP & A/D system)
Offline available
1000 times

Not available

Not available

All available
100000 times
1KHz to 100MHz

160nS (TTL 2 bytes writing time)
Filter online reconstruction setting time (The total time  of
filter frequency switched)


10mS


Not available


Not available


300nS
Group delay and the phase response:
Producer design only.
Linear or Producer design only.
Producer design only.
Producer & User design is available.
Equality data stream speed or comported speed of its TxDAC :
&Butterworth 8th filter
(Broadband  channel HFC usable signal dynamic range is  about 50dB):
4.8Kbps to 4.8Mbps
(600SPS to 600KSPS)


8bit TxDAC
Constant 400Kbps
(50KSPS)



8bit TxDAC
Constant 800Mbps
(100MSPS)



8bit TxDAC
16Kbps to 1600Mbps
(2KSPS to 200MSPS)


8bit TxDAC
Suitable with the 8bit TxDAC and used 256QAM (or 64VSB) in  SSB/VSB mode
of technology
(Data speed of Base Band):

Carrier signal RF full-band tie up:
Difficult to be used with the 256QAM and 64VSB.
(High Level Phase Noise)
4.8Kbps(Min)
4.8Mbps(Max)

384Hz to 384KHz
Difficult to be used in variable data-speed transmission or receiving
system



Difficult to be used in variable data-speed transmission or receiving
system
With 256QAM:
8Kbps(Min)
800Mbps
(Max)
With 64VSB:
12Kbps(Min)
1200Mbps (Max).

1.28KHz to 128MHz
For the HFC Specialty signal TV Channel:
40dB Eb/N
6MHz Full-band 64VSB model
@1KHz-6MHz
Data-rate:



With 64VSB:

9.375Kbps
(Min)
56.25Mbps
(Max)

Contact Address:
No.2 Buliding Room1007,
Mudanyuan Beili, Haidian District
Beijing P. R. China,
Post Code: 100083
Name: Tan Junsheng (Robert Tan)
Tele: 8610-82076834, 86-13701070213(Mobile)
E-mail:  Tjsh@netchina.com.cn or tanjun@hotmail.com
Homepage: http://www.cnindex.net/~tjsh
Details Remind:
http://www.cnindex.net/~tjsh/Base_of_Broadband_Access.html






From rem-conf Mon Nov 22 05:01:42 1999 
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To: rem-conf@es.net
Subject: AVT slides from Washington DC
Date: Mon, 22 Nov 1999 12:57:58 +0000
Message-ID: <1304.943275478@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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Copies of the slides presented in the last AVT meeting are now available from
http://www-mice.cs.ucl.ac.uk/multimedia/misc/avt/IETF46/slides/

Colin



From rem-conf Mon Nov 22 10:54:38 1999 
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Date: Mon, 22 Nov 1999 10:48:17 -0800
To: GeeVarghese John <gvjohn@miel.mot.com>, Tmima Koren <tmima@cisco.com>,
        casner@cisco.com, micke@cdt.luth.se, brucet@cisco.com, dwing@cisco.com,
        pruddy@cisco.com, rem-conf@es.net, gvjohn@miel.mot.com,
        pazhynnr@cig.mot.com
From: Tmima Koren <tmima@cisco.com>
Subject: Re: Extensions to CRTP
In-Reply-To: <38358E06.C3F08940@miel.mot.com>
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 <38358615.D8679DA0@miel.mot.com>
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At 12:51 PM 11/19/99 -0500, GeeVarghese John wrote: 
>
> Sorry, Please neglect my previous mail, this is the correct mail .. 
> I pasted my answers at the wrong place which might confuse those who read .. 
>
> GVJ 
>
> GeeVarghese John wrote: 
>>
>> Hi, 
>>
>> My comments to you answers are added below  .. 
>> Please have a look at that  // 
>>
>> Tmima Koren wrote: 
>>>
>>>   
>>> At 02:13 PM 11/18/99 -0500, you wrote: 
>>>>
>>>> Steve, Tmima  & all, 
>>>>
>>>> This is regarding the ietf draft - Extensions to CRTP  :
>>>> draft-koren-avt-crtp-enhance-00.txt 
>>>> I have few doubts and could you please clarrify these points .. 
>>>>
>>>> In the section 1. COMPRESSED_UDP packet "T bit" .. 
>>>>
>>>> It is stated that CUDP packet reset the deltaT value to 0 and 
>>>> therefore T bit avoids the need to send deltaT in the next CRTP. 
>>>>
>>>> Q1) My doubt is If the entire original RTP header is the in the CUDP packet
>>>> why to 
>>>> send the delta also with the original header .. 
>>>> why can't the decompressor findout the new delta to be used 
>>>> from the RTP header in the  CUDP packet itself, Is it not redundancy ? 
>>>>
>>>> Also You have stated that 
>>>>
>>>> >> Multiple COMPRESSED_UDP packets may be sent by the compressor when
>>>> changes in the 
>>>> >> compressor state need to be synchronized. Sending multiple packets will
>>>> increase the 
>>>> >> probability of delivery of the synchronized state in the event of
>>>> packet loss. 
>>>>
>>>> Why do we need to send multiple packet  and how it will increase the
>>>> probability even 
>>>> if you send more CUDP packets, as long as the decomperssor look at the seq
>>>> number 
>>>> and detect  packet loss  it should send a CONTEXT_STATE packet. 
>>>>
>>>> If the draft reference you made was for loss of more than 16 packets from
>>>> the same flow, 
>>>> and the possible wrong decompression of headers .. 
>>>> twice algorithm should detect this (But twice algo is not used if checksum
>>>> is 0) 
>>>> So here what exactly the limit could be , the sequence number space 1..16 
>>>> Since this draft address the use of this algo over ATM/AAL5 links  I think 
>>>> sequence number space 1..16 may not be sufficient.
>>>
>>>
>>>
>>> Some notations: 
>>> dt         deltaT 
>>> FH        FULL_HEADER 
>>> CR       COMPRESSED_RTP 
>>> CR+     COMPRESSED_RTP with deltaT (dt) 
>>> CU       COMPRESSED_UDP 
>>> CU+     COMPRESSED_UDP with dt 
>>> CS       CONTEXT_STATE 
>>>
>>> I'll use an example 
>>> The sender sends a sample every 10 ms 
>>> Using CRTP, the sequence of packets will be: 
>>>
>>> seq#   Time     pkt type 
>>> 1        10         FH 
>>> 2        20         CR+    dt=10 
>>> 3        30         CR 
>>> ... 
>>> 10      100       CR 
>>>
>>> At this point this side stops talking. Talking resumes at time 200 
>>>
>>> 11      200       CR+    dt=100 
>>> 12      210       CR+    dt=10 
>>> 13      220       CR 
>>> .... 
>>>
>>> With the extension of T-bit in CU, we can send: 
>>>
>>> 11'     200       CU+    dt=10 
>>> 12'     210       CR
>>
>
>
> Here, I think you assumed that 12'  packet will reach the compessor with 
> timestamp 210. But, unfortunately if it was lost what you get is 13' and time
> stamp 220. 
> In that case again you have send dt again with your CU+ after sending CU+
> with dt 10. 
> In this type of situation processing due to delta encoding was double and you
> send two 
> delta encoded bytes in two different packets insteadof one 
> You might have assumed that MOST of  the case packet loss doesnot happen .. 
> But I feel that is also purely dependant on the network where this algorithm
> is executing .. 
>
> And also are you not loosing the bandwidth saving by sending CU+ instead of
> CR 
> because  you send the complete RTP hdr + delta encoded deltaT byte in CU+. 
>   
>>
>>>
>>>   
>>> ... 
>>>
>>> We are trying to adopt CRTP to links that have some loss of packets and a
>>> long round trip time (RT) 
>>> We want to avoid the decompressor getting out of sync and having to send CS
>>> because this will cause loss of many packets for this stream due to the
>>> long RT. 
>>>
>>> Let's check what happens if some packets are lost 
>>> 1. packet 4 was lost 
>>>     The decompressor will use the 'twice' and will stay in sync 
>>> 2. packet 11 was lost 
>>>     The decompressor is out of sync 
>>> 3. packet 11'  was lost 
>>>     The decompressor is out of sync 
>>>
>>> We see that there is a weak point in the compression at the beginning of a
>>> talk spurt 
>>> We try to correct this by sending multiple CU+ 
>>>
>>> 11"    200       CU+    dt=10 
>>> 12"    210       CU+    dt=10 
>>> 13"    220       CR
>>>   
>>> ... 
>>>
>>> If the decompressor received any of 11"  or 12" ,  he is still in sync
>>
>
> I think this is not correct, If the decompressor receives 12" insted of 11", 
> he is not in sync because of link seq number mismatch and 
> will send a CONTEXT_STATE packet. 
> This will lead to FH sending from compressor to decompressor .. 
>>
>>   
>>>
>>>   
>>>
>>> How many CU+ do we want to send at the beginning of a talk spurt? 
>>> This depends on the quality of the link (which is good but not perfect). 
>>> If we know that on this link we may lose at most 2 packets in a row, we'll
>>> send 3 CU+ at the beginning of a talk spurt
>>
>
> Looks OK, but How do we decide on the number of packet loss ? 
> Packet loss can be due to link quality and/or network congestion etc .. 
> If the loss was due network congestion or similar factor, behaviour vary from
> time to time .. 
> So here also CU+ is giving disadvantage instead of the benifits expected. 
>>
>>>
>>>   
>>> The decompressor will stay in sync as long as he received at least one of
>>> the CU+
>>
>
> I think any of the CU+ loss will lead to CONTEXT_STATE packet generation from
>
> decompressor to compressor because of the link seq number mismatch ..Is nt it
> ? 


The decompressor should be in sync after receiving a CU+. There is no reason
for a CS
Tmima

>
>   
>>
>>   
>>>
>>>  
>>
>> I will include rest of the questions in another mail .. 
>> Please reply back .. 
>>
>> Rgds 
>> GVJ 
>>   
>>   
>>  
>
>
>  



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Content-Type: text/html; charset="us-ascii"

<html>
At 12:51 PM 11/19/99 -0500, GeeVarghese John wrote: <br>
<blockquote type=cite cite>Sorry, Please neglect my previous mail, this
is the correct mail .. <br>
I pasted my answers at the wrong place which might confuse those who read
.. <br>
<br>
GVJ <br>
<br>
GeeVarghese John wrote: <br>
<blockquote type=cite cite>Hi, <br>
<br>
My comments to you answers are added below&nbsp; .. <br>
Please have a look at that&nbsp; // <br>
<br>
Tmima Koren wrote: <br>
<blockquote type=cite cite>&nbsp; <br>
At 02:13 PM 11/18/99 -0500, you wrote: <br>
<blockquote type=cite cite>Steve, Tmima&nbsp; &amp; all, <br>
<br>
This is regarding the ietf draft - Extensions to CRTP&nbsp; :
draft-koren-avt-crtp-enhance-00.txt <br>
I have few doubts and could you please clarrify these points .. <br>
<br>
In the section 1. COMPRESSED_UDP packet &quot;T bit&quot; .. <br>
<br>
It is stated that CUDP packet reset the deltaT value to 0 and <br>
therefore T bit avoids the need to send deltaT in the next CRTP. <br>
<br>
Q1) <b>My doubt is If the entire original RTP header is the in the CUDP
packet why to</b> <br>
<b>send the delta also with the original header ..</b> <br>
<b>why can't the decompressor findout the new delta to be used</b> <br>
<b>from the RTP header in the&nbsp; CUDP packet itself, Is it not
redundancy ?</b> <br>
<br>
Also You have stated that <br>
<br>
&gt;&gt; Multiple COMPRESSED_UDP packets may be sent by the compressor
when changes in the <br>
&gt;&gt; compressor state need to be synchronized. Sending multiple
packets will increase the <br>
&gt;&gt; probability of delivery of the synchronized state in the event
of packet loss. <br>
<br>
<i>Why do we need to send multiple packet</i>&nbsp; and <i>how it will
increase the probability even</i> <br>
if you send more CUDP packets, as long as the decomperssor look at the
seq number <br>
and detect&nbsp; packet loss&nbsp; it should send a CONTEXT_STATE packet.
<br>
<br>
I<i>f the draft reference you made was for loss of more than 16 packets
>from the same flow,</i> <br>
<i>and the possible wrong decompression of headers ..</i> <br>
<i>twice algorithm should detect this (But twice algo is not used if
checksum is 0)</i> <br>
<i>So here what exactly the limit could be , the sequence number space
1..16</i> <br>
<i>Since this draft address the use of this algo over ATM/AAL5
links&nbsp; I think</i> <br>
<i>sequence number space 1..16 may not be
sufficient.</i></blockquote><br>
<br>
Some notations: <br>
dt&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; deltaT <br>
FH&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; FULL_HEADER <br>
CR&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; COMPRESSED_RTP <br>
CR+&nbsp;&nbsp;&nbsp;&nbsp; COMPRESSED_RTP with deltaT (dt) <br>
CU&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; COMPRESSED_UDP <br>
CU+&nbsp;&nbsp;&nbsp;&nbsp; COMPRESSED_UDP with dt <br>
CS&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; CONTEXT_STATE <br>
<br>
I'll use an example <br>
The sender sends a sample every 10 ms <br>
Using CRTP, the sequence of packets will be: <br>
<br>
seq#&nbsp;&nbsp; Time&nbsp;&nbsp;&nbsp;&nbsp; pkt type <br>
1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
10&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; FH <br>
2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
20&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; CR+&nbsp;&nbsp;&nbsp;
dt=10 <br>
3&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
30&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; CR <br>
... <br>
10&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 100&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR <br>
<br>
At this point this side stops talking. Talking resumes at time 200 <br>
<br>
11&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 200&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR+&nbsp;&nbsp;&nbsp; dt=100 <br>
12&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 210&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR+&nbsp;&nbsp;&nbsp; dt=10 <br>
13&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 220&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR <br>
.... <br>
<br>
With the extension of T-bit in CU, we can send: <br>
<br>
11'&nbsp;&nbsp;&nbsp;&nbsp; 200&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CU+&nbsp;&nbsp;&nbsp; dt=10 <br>
12'&nbsp;&nbsp;&nbsp;&nbsp; 210&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR</blockquote></blockquote><br>
Here, I think you assumed that 12'&nbsp; packet will reach the compessor
with <br>
timestamp 210. But, unfortunately if it was lost what you get is 13' and
time stamp 220. <br>
In that case again you have send dt again with your CU+ after sending CU+
with dt 10. <br>
In this type of situation <b>processing due to delta encoding was
double</b> and you send <b>two</b> <br>
<b>delta encoded bytes in two different packets insteadof one</b> <br>
You might have assumed that MOST of&nbsp; the case packet loss doesnot
happen .. <br>
But I feel that is also purely dependant on the network where this
algorithm is executing .. <br>
<br>
And also are you not loosing the bandwidth saving by sending CU+ instead
of CR <br>
because&nbsp; you send the complete RTP hdr + delta encoded deltaT byte
in CU+. <br>
&nbsp; <br>
<blockquote type=cite cite><blockquote type=cite cite>&nbsp; <br>
... <br>
<br>
We are trying to adopt CRTP to links that have some loss of packets and a
long round trip time (RT) <br>
We want to avoid the decompressor getting out of sync and having to send
CS because this will cause loss of many packets for this stream due to
the long RT. <br>
<br>
Let's check what happens if some packets are lost <br>
1. packet 4 was lost <br>
&nbsp;&nbsp;&nbsp; The decompressor will use the 'twice' and will stay in
sync <br>
2. packet 11 was lost <br>
&nbsp;&nbsp;&nbsp; The decompressor is out of sync <br>
3. packet 11'&nbsp; was lost <br>
&nbsp;&nbsp;&nbsp; The decompressor is out of sync <br>
<br>
We see that there is a weak point in the compression at the beginning of
a talk spurt <br>
We try to correct this by sending multiple CU+ <br>
<br>
11&quot;&nbsp;&nbsp;&nbsp; 200&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CU+&nbsp;&nbsp;&nbsp; dt=10 <br>
12&quot;&nbsp;&nbsp;&nbsp; 210&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CU+&nbsp;&nbsp;&nbsp; dt=10 <br>
13&quot;&nbsp;&nbsp;&nbsp; 220&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR<br>
&nbsp; <br>
... <br>
<br>
If the decompressor received any of 11&quot;&nbsp; or 12&quot; ,&nbsp; he
is still in sync</blockquote></blockquote>I think this is not correct, If
the decompressor receives 12&quot; insted of 11&quot;, <br>
he is not in sync because of link seq number mismatch and <br>
will send a CONTEXT_STATE packet. <br>
This will lead to FH sending from compressor to decompressor .. <br>
<blockquote type=cite cite>&nbsp; <br>
<blockquote type=cite cite>&nbsp; <br>
<br>
How many CU+ do we want to send at the beginning of a talk spurt? <br>
This depends on the quality of the link (which is good but not perfect).
<br>
If we know that on this link we may lose at most 2 packets in a row,
we'll send 3 CU+ at the beginning of a talk
spurt</blockquote></blockquote>Looks OK, but How do we decide on the
number of packet loss ? <br>
Packet loss can be due to link quality and/or network congestion etc ..
<br>
If the loss was due network congestion or similar factor, behaviour vary
>from time to time .. <br>
So here also CU+ is giving disadvantage instead of the benifits expected.
<br>
<blockquote type=cite cite><blockquote type=cite cite>&nbsp; <br>
The decompressor will stay in sync as long as he received at least one of
the CU+</blockquote></blockquote>I think any of the CU+ loss will lead to
CONTEXT_STATE packet generation from <br>
decompressor to compressor because of the link seq number mismatch ..Is
nt it ? </blockquote><br>
The decompressor should be in sync after receiving a CU+. There is no
reason for a CS<br>
Tmima<br>
<br>
<blockquote type=cite cite>&nbsp; <br>
<blockquote type=cite cite>&nbsp; <br>
<blockquote type=cite cite>&nbsp;</blockquote>I will include rest of the
questions in another mail .. <br>
Please reply back .. <br>
<br>
Rgds <br>
GVJ <br>
&nbsp; <br>
&nbsp; <br>
&nbsp;</blockquote><br>
&nbsp;</blockquote><br>
</html>

--=====================_434372375==_.ALT--




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From rem-conf Tue Nov 23 02:27:55 1999 
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From: "Lars-Erik Jonsson" <lars-erik.jonsson@ericsson.com>
To: "GeeVarghese John" <gvjohn@miel.mot.com>, "Tmima Koren" <tmima@cisco.com>,
        <casner@cisco.com>, <micke@cdt.luth.se>, <brucet@cisco.com>,
        <dwing@cisco.com>, <pruddy@cisco.com>, <rem-conf@es.net>,
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Subject: T-bit extension for CRTP
Date: Tue, 23 Nov 1999 11:20:48 +0100
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I have a number of comments both to what is said in the draft
(draft-koren-crtp-enhance-00.txt) and to the previous discussions about i=
t.
All are included below:


- Draft: The purpose

Tmima: In the draft you state that you need this to be able to do periodi=
c
refreshing of the context when CRTP is used over links with large RTT.
However, in the previous discussion you are only talking about what happe=
ns
after silence periods. I think you should clarify what you are really aim=
ing
for, maybe the draft should also talk about the implications after silenc=
e
periods.


- Draft: Packet types for refreshing

In the draft it is stated that there are TWO packet types that can be use=
d
to periodically refresh the context of an IPv4 CRTP decompressor,
FULL_HEADER and COMPRESSED_UDP packets. This is not really correct. For
IPv4, the COMPRESSED_UDP packet does NOT refresh the context completely,
because the IP-ID is communicated as a delta. The COMPRESSED_NON_TCP pack=
et
type should instead be used with IPv4. For IPv6, the two packet
types(C_NON_TCP and C_UDP) carries the same information and are identical=
 in
size.


- Discussion: GeeVarghese Q1

>Q1) My doubt is If the entire original RTP header is the in the CUDP pac=
ket
why to
>send the delta also with the original header ..
>why can't the decompressor findout the new delta to be used
>from the RTP header in the  CUDP packet itself, Is it not redundancy ?

The decompressor can NOT find out the delta for sure, because its context
could have been invalid when the C_UDP was received. So, if the intention=
 is
to have the delta "correctly" set (not reset to 0) after a C_UDP, then yo=
u
must include it. The question is whether this is advantageous compared to
the current solution, sending the delta in subsequent C_RTP.


- Discussion: GeeVarghese about multiple packets

>Why do we need to send multiple packet  and how it will increase the
probability even
>if you send more CUDP packets, as long as the decomperssor look at the s=
eq
number
>and detect  packet loss  it should send a CONTEXT_STATE packet.

Yes, but if the RTT is large, the update procedure would take a long time.
If Twice is used (requires UDP checksum and should be used carefully), lo=
ss
of packets that DO NOT change the decompressor context could be handled
without discarding additional packets. On the other hand, if packets that=
 DO
change the context are lost, Twice would fail and subsequent packets must=
 be
discarded while waiting for a refresh. If context update information was
repeated in more then one packet, a higher tolerance against packet loss
could be achieved.


- Discussion: The example

>Some notations:
>dt         deltaT
>FH        FULL_HEADER
>CR       COMPRESSED_RTP
>CR+     COMPRESSED_RTP with deltaT (dt)
>CU       COMPRESSED_UDP
>CU+     COMPRESSED_UDP with dt
>CS       CONTEXT_STATE

>I'll use an example
>The sender sends a sample every 10 ms
>Using CRTP, the sequence of packets will be:

>seq#   Time     pkt type
>1        10         FH
>2        20         CR+    dt=3D10
>3        30         CR
>...
>10      100       CR

>At this point this side stops talking. Talking resumes at time 200

>11      200       CR+    dt=3D100
>12      210       CR+    dt=3D10
>13      220       CR
>....

>With the extension of T-bit in CU, we can send:

>11'     200       CU+    dt=3D10
>12'     210       CR
>...

When I look at the two different solutions 11+12 and 11'+12' I can not
really see the big advantage of 11'+12'. First of all, the latter require=
s
more overhead. While 11 would have a 3 octet header, the 11' header would=
 be
at least 15 octets large. The idea is certainly to have the context
completely refreshed, but that could not be achieved with CU+ because it
does not include the IP ID in absolute terms. IF you want to refresh the
context with a CU packet as packet #11, you could also still use the old =
CU
and send the dt-value in the next compressed packet, it would require one
octet less in the CU and one more in the CR(+). Also the dt information
could be repeated in more than one CR-packet for use with "sophisticated
Twice".


- Discussion: GeeVarghese about FH

>I think this is not correct, If the decompressor receives 12" insted of
11",
>he is not in sync because of link seq number mismatch and
>will send a CONTEXT_STATE packet.
>This will lead to FH sending from compressor to decompressor ..

Not necessarily a FULL_HEADER, a COMPRESSED_NON_TCP would be a better
choice.


- Discussion: GeeVarghese and Koren about CONTEXT_UPDATE

>>I think any of the CU+ loss will lead to CONTEXT_STATE packet generatio=
n
from
>>decompressor to compressor because of the link seq number mismatch ..Is=
 nt
it ?

>The decompressor should be in sync after receiving a CU+. There is no
reason for a CS

As the CRTP RFC says: "Note, however, that there is a nontrivial risk of =
an
incorrect positive indication. It may be advisable to request a FULL_HEAD=
ER
or COMPRESSED_NON_TCP packet even if the "twice" algorithm succeeds." The
UDP checksum is rather weak, so especially if you use more complex "twice=
"
solutions, the advice above should be followed, meaning that a CONTEXT_ST=
ATE
packet should be sent.


Regards!!
/Lars-Erik

--------------------------------------------------------------
Lars-Erik Jonsson, M.Sc.
Voice Processing and Radio Network Research
Ericsson Erisoft AB
Box 920, S-971 28 Lule=E5, Sweden
E-mail: lars-erik.jonsson@ericsson.com
Phone: +46 920 20 21 07
Mobile: +46 70 365 20 58
Fax: +46 920 20 20 99
Home: +46 920 999 57





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>> >>
>> >>  The decompressor will stay in sync as long as he received at
>> >>  least one of the CU+
>> >
>> I think any of the CU+ loss will lead to CONTEXT_STATE packet
>> generation from
>> decompressor to compressor because of the link seq number mismatch
>> ..Is nt it ?
>
>
> The decompressor should be in sync after receiving a CU+. There is no
> reason for a CS
> Tmima

Tmima,

Could you please explain, how IP id variance should be treated in this
case ?
For IP id field  first order difference is sent as delta in CUDP with I
bit set.
In this case, if there is a CU+ packet loss how IP ID delta encoding
should be treated ..

and also I am not still convinced with the advantage of CU+ over CR+
after
the  silence period. Could you please explain your point  ..(please ref
my previous mail)

Please reply .

Rgds

GVJ

--------------FBCBB89BB0AB9919535664A3
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
&nbsp;
<blockquote TYPE=CITE>
<blockquote type=cite cite>
<blockquote type=cite cite>
<blockquote type=cite cite>&nbsp;
<br>The decompressor will stay in sync as long as he received at least
one of the CU+</blockquote>
</blockquote>
I think any of the CU+ loss will lead to CONTEXT_STATE packet generation
from
<br>decompressor to compressor because of the link seq number mismatch
..Is nt it ?</blockquote>

<p><br>The decompressor should be in sync after receiving a CU+. There
is no reason for a CS
<br>Tmima</blockquote>
Tmima,
<p>Could you please explain, how IP id variance should be treated in this
case ?
<br>For IP id field&nbsp; first order difference is sent as delta in CUDP
with I bit set.
<br>In this case, if there is a CU+ packet loss how IP ID delta encoding
should be treated ..
<p>and also I am not still convinced with the advantage of CU+ over CR+
after
<br>the&nbsp; silence period. Could you please explain your point&nbsp;
..(please ref my previous mail)
<p>Please reply .
<p>Rgds
<p>GVJ</html>

--------------FBCBB89BB0AB9919535664A3--




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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		: Sampling of the Group Membership in RTP
	Author(s)	: J. Rosenberg, H. Schulzrinne
	Filename	: draft-ietf-avt-rtpsample-06.txt
	Pages		: 11
	Date		: 23-Nov-99
	
In large multicast groups, the size of the group membership table
maintained by RTP (Real Time Transport Protocol) participants may
become unwieldy, particularly for embedded devices with limited
memory and processing power. This document discusses mechanisms for
sampling of this group membership table in order to reduce the memory
requirements. Several mechanisms are proposed, and the performance of
each is considered.
 
 
   NOTE:
 
   This is to inform you that Lucent Technologies has applied for and/or
   has patent(s) that relates to the binning algorithm which is a part
   of the specification.
 
   This declaration is being made pursuant to the provisions of IETF IPR
   Policy , Sections 10.3.1 and 10.3.2.
 
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   follows:
 
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From rem-conf Wed Nov 24 11:13:38 1999 
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From: "John Weiler, OBJECTive Technology Group" <john_weiler@TheOTG.com>
Organization: Interoperability Clearinghouse 703-768-0400
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Hi again,

I wanted to let you know that the Interoperability Clearinghouse (ICH)
has greatly accelerated its Architecture Validation Program (AVP) for
interoperable E-Business & Information Infrastructure frameworks.  The
ICH is preparing its initial press release in conjunction with
industry=92s largest IT standards group, ISVs, integrators and domain
implementers.  If your organization is interested in participating in
this launch scheduled for January, 2000, please review the following
materials and get back to us ASAP, so we can discuss engagement
options.

If you are not the appropriate individual to represent your
organization=92s interest for this effort, please forward this ICH notice
to those who should be apprised of this collaborative architecture
initiative.  Below is a summary of some of our recent developments that
will be incorporated into our January 2000 announcement.

- ICH, in cooperation with the AFCEA, AFEI, IAC, and ITAA has
established four E-Business blue ribbon panels for OSD AT&L and OSD
C3I.  The Software Quality and Interoperability working group has
concluded its interim report (www.eccwg.org). Other ICH co-sponsored
working groups include Information Assurance, Performance
Measurements and Incentives.  Participation options are open.
- Due to growing demand for the ICH, OSD C3I recently concluded an
extensive validation of the ICH collaborative initiative.   Copies
are available for prospective ICH partners.
- A Defense Architecture Immersion Program is beginning with Navy,
DIA, DFAS, and DoD solution providers.  Focus is on directory
services, web application servers, and information assurance.
- OSD C3I has commissioned a Defense@OOTS seminar (www.TheOTG.com) set
for  March 2000. Co-sponsors include OMG, ICH, IBM, other
distributed computing experts.  Sponsorships are still
open.
- The Distributed Component-based Architecture Modeling (DCAM) IRAD
team is expanding to address Information Assurance architecture
modeling building on earlier R&D successes sponsored by DARPA.
- AF Scientific Advisory Board references ICH as means of dealing with
COTS software in its annual recommendations to the Secretary of the
AF.
- Dep. Of Commerce Architecture Validation Program: Frameworks
addressing Web based infrastructure, PKI and Enterprise Directory
services
- Health Information Infrastructure (HII):  Developing and Validating
architecture frameworks for Government Wide Computer-based Patient
Record (VA, DoD, HHS).
- DOJ sponsored Justice Architecture Initiative: Cooperative effort
with states to define common infrastructure frameworks.
- Multi-State Architecture Validation Program began with kickoff at
NASIRE conference.
- Resubmission of Interoperability proposal to Federal CIO Council.
- CDC has concluded a Data Warehousing mini-session in cooperation
with TRW with great success.

Due to the confidential nature of our Telecom and Finance Domain
Architecture initiatives, we will only be able to discuss under non-
disclosure.

Participation Options;
To help domain implementers accelerate the adoption of leading edge
technologies, the ICH has instituted its industry support Architecture
Validation Program. Below are ways disparate organizations can
contribute to the refinement of these domain architecture models;
- SDOs: Have standards abstracted and validated through mapping of
specifications to product implementations.  The evidence of viable
standards can be found in real implementations.
- Vendors: Have products offerings modeled and validated into domain
architecture for multi-client applications. Technology focus areas
include: Web development environments, Web Application Servers,
Middleware, Directory Services, PKI, Object Persistence, Data
Management, Component Models and Enterprise Resource Management.
- Integrators: Support technology research and validation efforts.
Document domain and technical experience for best practices
repository.
- Domain Users: Engage Architecture Validation Program.  Co-develop
and validate interoperable frameworks for secure, Internet based
architectures.   Architecture mentoring services available.
- All: Join ICH domain working groups, collaborative IR&D project to
develop knowledge repository, participate in Industry@IT-
Architectures forums.

Those organizations who have already begun to engage this collaborative
architecture validation initiative include;
Component Source, DARPA, Defense Medical, Defense Finance, Defense
Intelligence Agency, Department of Justice, EDS, IBM, IEEE, Lockheed
Martin, Microsoft, Object Management Group, US Navy, Bell Atlantic,
NSTL, Open Applications Group, Open GIS, Office of the Secretary of
Defense, Patent Trademark Office, TINA-C, TRW, State of Missouri,
Federal E-Commerce Consortium (ITAA, CommerceNET, IAC, AFCEA, AFEI)

For those interested in being part of our 2000 press release, submit
your approved quote before December 15, 1999. Below is the set of
quotes already supplied by our prospective partners in this global
effort.

IBM:  " IBM, will continue to be an influential force in promoting open
systems and interoperability, and as such, sees the Interoperability
Clearinghouse as one of the more significant initiatives in defining
the road map to enterprise computing.  Our support of this initiative
is further evidence of our commitment to open, standards based IT".

SUN MICROSYSTEMS: =93Sun applauds the efforts of the OMG, OTG and DARPA
in developing an =91Interoperability Clearinghouse=92 tool.  This ambitio=
us
project will help to evangelize what Sun has been saying for so long.
=91Open systems are the best way to protect IT investments.=92 =94

MICROSOFT: =93The Interoperability Clearinghouse initiative promises to
be a means for IT professionals to make informed decisions in this
complex area. Microsoft welcomes the efforts of the ICH initiative and
looks forward to it becoming a useful tool for understanding
Interoperability of IT products=94

LOCKHEED MARTIN: =93This initiative complements several on going COTS
efforts within Lockheed Martin Corp., and we recognize the potential
value that the (Interoperability Clearinghouse) effort will bring to
industry=94

OPEN APPLICATIONS GROUP: =93We at the Open Applications Group are
interested in this initiative and will incorporate our specifications
and testing results into the ICH once built.=94

OBJECT MANAGEMENT GROUP: =93As one of the leading advocates of open
systems and interoperability, the OMG believes that the
Interoperability Clearinghouse initiative will help users realize the
benefits from our combined efforts.=94

For more information, please call 703-768-4975 or email
info@e-interop.com.

http://www.E-interop.com (Interoperability Clearinghouse link)
http://www.gcn.com/vol18_no24/enterprise/313-1.html
http://www.infoworld.com/cgi-bin/displayStory.pl?981113.whsoft.htm
http://www.fcw.com/pubs/fcw/1999/0208/fcw-newsinterop-2-8-99.html
http://206.144.247.65/archives/gcn/1999/February8/1c.htm
http://206.144.247.65/archives/gcn/1999/March29/16.htm
http://www.healthcare-informatics.com/issues/1999/05_99/standards.htm
http://www.omg.org/techprocess/meetings/ic.html

john





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From rem-conf Wed Nov 24 15:24:44 1999 
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Date: Wed, 24 Nov 1999 15:19:29 -0800
To: GeeVarghese John <gvjohn@miel.mot.com>, Tmima Koren <tmima@cisco.com>
From: Tmima Koren <tmima@cisco.com>
Subject: Re: Extensions to CRTP
Cc: casner@cisco.com, micke@cdt.luth.se, brucet@cisco.com, dwing@cisco.com,
        pruddy@cisco.com, rem-conf@es.net, pazhynnr@cig.mot.com,
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At 10:17 PM 11/23/99 -0500, GeeVarghese John wrote: 
>
>   
>>
>>>
>>>>
>>>>>
>>>>>   
>>>>> The decompressor will stay in sync as long as he received at least one of
>>>>> the CU+
>>>>
>>>
>>> I think any of the CU+ loss will lead to CONTEXT_STATE packet generation
>>> from 
>>> decompressor to compressor because of the link seq number mismatch ..Is nt
>>> it ?
>>
>>
>>
>> The decompressor should be in sync after receiving a CU+. There is no reason
>> for a CS 
>> Tmima
>
> Tmima, 
>
> Could you please explain, how IP id variance should be treated in this case ?
>
> For IP id field  first order difference is sent as delta in CUDP with I bit
> set. 
> In this case, if there is a CU+ packet loss how IP ID delta encoding should
> be treated .. 


Usually the IP id will not increment uniformly in packets of an RTP stream (the
host sends packets other then the RTP streams, silence periods).
This means that if we want to compress a stream, we might have to send a new
deltaID often, maybe in each packet. This will increase the size of the
packets.
To optimize the compression, the host can assign IP id's to the packets of each
stream in a constant increment.
This keeps the deltaID constant for each stream.

The advantage of a CU+ packet is that in one packet we restore all RTP-related
parameters at the decompressor.
The sequence of:  CU followed by a CR+  is very vulnerable: the CU zeroes the
deltaT, so if the CR+ was lost, the decompressor is out of sync.
The CU+ does not restore the IP id. But if the implementation keeps the deltaID
constant..

After a silence period, the sequence number will still increment by 1 and the
IP id by deltaID: there is no reason to skip RTP sequence numbers or IP id's.
deltaT is the only difference that is not correct for the first packet after a
silence period.

To optimize CRTP over lossy links 
1. Use a constant deltaID
2. Check udp checksum
3. Use CU+ as necessary


>
> and also I am not still convinced with the advantage of CU+ over CR+ after 
> the  silence period. Could you please explain your point  ..(please ref my
> previous mail) 


Let's say that the characteristics of the link is that we may lose at most 1
packet in a row
If we use:

11      200       CR+    dt=100
12      210       CR+    dt=10
13      220       CR

If 11 is lost, the decompressor does not have the correct RTP timestamp for
packet 12.
If 12 is lost, the decompressor has a wrong deltaT.
In both cases the decompressor is out of sync. 
This is a very weak sequence to use after a silence period.

If instead we use:

11"    200       CU+    dt=10 
12"    210       CU+    dt=10 
13"    220       CR

In this case If one of 11" or 12" is lost, the decompressor remains in sync. So
this is a very strong sequence

Tmima


>
> Please reply . 
>
> Rgds 
>
> GVJ



--=====================_19711203==_.ALT
Content-Type: text/html; charset="us-ascii"

<html>
At 10:17 PM 11/23/99 -0500, GeeVarghese John wrote: <br>
<blockquote type=cite cite>&nbsp; <br>
<blockquote type=cite cite><blockquote type=cite cite><blockquote type=cite cite><blockquote type=cite cite>&nbsp;
<br>
The decompressor will stay in sync as long as he received at least one of
the CU+</blockquote></blockquote>I think any of the CU+ loss will lead to
CONTEXT_STATE packet generation from <br>
decompressor to compressor because of the link seq number mismatch ..Is
nt it ?</blockquote><br>
<br>
The decompressor should be in sync after receiving a CU+. There is no
reason for a CS <br>
Tmima</blockquote>Tmima, <br>
<br>
Could you please explain, how IP id variance should be treated in this
case ? <br>
For IP id field&nbsp; first order difference is sent as delta in CUDP
with I bit set. <br>
In this case, if there is a CU+ packet loss how IP ID delta encoding
should be treated .. </blockquote><br>
Usually the IP id will not increment uniformly in packets of an RTP
stream (the host sends packets other then the RTP streams, silence
periods).<br>
This means that if we want to compress a stream, we might have to send a
new deltaID often, maybe in each packet. This will increase the size of
the packets.<br>
To optimize the compression, the host can assign IP id's to the packets
of each stream in a constant increment.<br>
This keeps the deltaID constant for each stream.<br>
<br>
The advantage of a CU+ packet is that in one packet we restore all
RTP-related parameters at the decompressor.<br>
The sequence of:&nbsp; CU followed by a CR+&nbsp; is very vulnerable: the
CU zeroes the deltaT, so if the CR+ was lost, the decompressor is out of
sync.<br>
The CU+ does not restore the IP id. But if the implementation keeps the
deltaID constant..<br>
<br>
After a silence period, the sequence number will still increment by 1 and
the IP id by deltaID: there is no reason to skip RTP sequence numbers or
IP id's. deltaT is the only difference that is not correct for the first
packet after a silence period.<br>
<br>
To optimize CRTP over lossy links <br>
1. Use a constant deltaID<br>
2. Check udp checksum<br>
3. Use CU+ as necessary<br>
<br>
<br>
<blockquote type=cite cite>and also I am not still convinced with the
advantage of CU+ over CR+ after <br>
the&nbsp; silence period. Could you please explain your point&nbsp;
..(please ref my previous mail) </blockquote><br>
Let's say that the characteristics of the link is that we may lose at
most 1 packet in a row<br>
If we use:<br>
<br>
11&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 200&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR+&nbsp;&nbsp;&nbsp; dt=100<br>
12&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 210&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR+&nbsp;&nbsp;&nbsp; dt=10<br>
13&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 220&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR<br>
<br>
If 11 is lost, the decompressor does not have the correct RTP timestamp
for packet 12.<br>
If 12 is lost, the decompressor has a wrong deltaT.<br>
In both cases the decompressor is out of sync. <br>
This is a very weak sequence to use after a silence period.<br>
<br>
If instead we use:<br>
<br>
11&quot;&nbsp;&nbsp;&nbsp; 200&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CU+&nbsp;&nbsp;&nbsp; dt=10 <br>
12&quot;&nbsp;&nbsp;&nbsp; 210&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CU+&nbsp;&nbsp;&nbsp; dt=10 <br>
13&quot;&nbsp;&nbsp;&nbsp; 220&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR<br>
<br>
In this case If one of 11&quot; or 12&quot; is lost, the decompressor
remains in sync. So this is a very strong sequence<br>
<br>
Tmima<br>
<br>
<br>
<blockquote type=cite cite>Please reply . <br>
<br>
Rgds <br>
<br>
GVJ</blockquote><br>
</html>

--=====================_19711203==_.ALT--




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Date: Wed, 24 Nov 1999 16:01:49 -0800
To: "Lars-Erik Jonsson" <lars-erik.jonsson@ericsson.com>,
        "GeeVarghese John" <gvjohn@miel.mot.com>,
        "Tmima Koren" <tmima@cisco.com>, <casner@cisco.com>,
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From: Tmima Koren <tmima@cisco.com>
Subject: Re: T-bit extension for CRTP
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At 11:20 AM 11/23/99 +0100, Lars-Erik Jonsson wrote:
>I have a number of comments both to what is said in the draft
>(draft-koren-crtp-enhance-00.txt) and to the previous discussions about it.
>All are included below:
>
>
>- Draft: The purpose
>
>Tmima: In the draft you state that you need this to be able to do periodic
>refreshing of the context when CRTP is used over links with large RTT.
>However, in the previous discussion you are only talking about what happens
>after silence periods. I think you should clarify what you are really=
 aiming
>for, maybe the draft should also talk about the implications after silence
>periods.

Thanks, I'll clarify this point.
The idea is that once the deltas are set at the decompressor and you are=
 sending CR packets, if you lose packets you can recover using 'twice'.
But if you try to change a delta by sending a CR+ or CU, and if this packet=
 is lost, the decompressor cannot recover using 'twice'.
A silence period is just an example of a time when you need to change the=
 deltaT.=20

>
>
>- Draft: Packet types for refreshing
>
>In the draft it is stated that there are TWO packet types that can be used
>to periodically refresh the context of an IPv4 CRTP decompressor,
>FULL_HEADER and COMPRESSED_UDP packets. This is not really correct. For
>IPv4, the COMPRESSED_UDP packet does NOT refresh the context completely,
>because the IP-ID is communicated as a delta. The COMPRESSED_NON_TCP packet
>type should instead be used with IPv4. For IPv6, the two packet
>types(C_NON_TCP and C_UDP) carries the same information and are identical=
 in
>size.

You are right - I'll correct this
A FH does not refresh the context: it starts a new context
And a CU+ restores all RTP-related parameters at the decompressor. The IP id=
 is not restored.

I checked the COMPRESSED_NON_TCP packet and I can't use it to fully refresh=
 the state at the decompressor: it will set the deltaT to zero.

I think I'll propose an additional change to the CU packet:

In a CU packet the M and S bits are always 0.
I propose that if the M bit is set instead of the I bit, then the packet=
 carries the full IP id and not only the delta IP id.

Tmima


>
>
>- Discussion: GeeVarghese Q1
>
>>Q1) My doubt is If the entire original RTP header is the in the CUDP=
 packet
>why to
>>send the delta also with the original header ..
>>why can't the decompressor findout the new delta to be used
>>from the RTP header in the  CUDP packet itself, Is it not redundancy ?
>
>The decompressor can NOT find out the delta for sure, because its context
>could have been invalid when the C_UDP was received. So, if the intention=
 is
>to have the delta "correctly" set (not reset to 0) after a C_UDP, then you
>must include it. The question is whether this is advantageous compared to
>the current solution, sending the delta in subsequent C_RTP.

>
>
>- Discussion: GeeVarghese about multiple packets
>
>>Why do we need to send multiple packet  and how it will increase the
>probability even
>>if you send more CUDP packets, as long as the decomperssor look at the seq
>number
>>and detect  packet loss  it should send a CONTEXT_STATE packet.
>
>Yes, but if the RTT is large, the update procedure would take a long time.
>If Twice is used (requires UDP checksum and should be used carefully), loss
>of packets that DO NOT change the decompressor context could be handled
>without discarding additional packets. On the other hand, if packets that=
 DO
>change the context are lost, Twice would fail and subsequent packets must=
 be
>discarded while waiting for a refresh. If context update information was
>repeated in more then one packet, a higher tolerance against packet loss
>could be achieved.
>
>
>- Discussion: The example
>
>>Some notations:
>>dt         deltaT
>>FH        FULL_HEADER
>>CR       COMPRESSED_RTP
>>CR+     COMPRESSED_RTP with deltaT (dt)
>>CU       COMPRESSED_UDP
>>CU+     COMPRESSED_UDP with dt
>>CS       CONTEXT_STATE
>
>>I'll use an example
>>The sender sends a sample every 10 ms
>>Using CRTP, the sequence of packets will be:
>
>>seq#   Time     pkt type
>>1        10         FH
>>2        20         CR+    dt=3D10
>>3        30         CR
>>...
>>10      100       CR
>
>>At this point this side stops talking. Talking resumes at time 200
>
>>11      200       CR+    dt=3D100
>>12      210       CR+    dt=3D10
>>13      220       CR
>>....
>
>>With the extension of T-bit in CU, we can send:
>
>>11'     200       CU+    dt=3D10
>>12'     210       CR
>>...
>
>When I look at the two different solutions 11+12 and 11'+12' I can not
>really see the big advantage of 11'+12'. First of all, the latter requires
>more overhead. While 11 would have a 3 octet header, the 11' header would=
 be
>at least 15 octets large. The idea is certainly to have the context
>completely refreshed, but that could not be achieved with CU+ because it
>does not include the IP ID in absolute terms. IF you want to refresh the
>context with a CU packet as packet #11, you could also still use the old CU
>and send the dt-value in the next compressed packet, it would require one
>octet less in the CU and one more in the CR(+). Also the dt information
>could be repeated in more than one CR-packet for use with "sophisticated
>Twice".
>
>
>- Discussion: GeeVarghese about FH
>
>>I think this is not correct, If the decompressor receives 12" insted of
>11",
>>he is not in sync because of link seq number mismatch and
>>will send a CONTEXT_STATE packet.
>>This will lead to FH sending from compressor to decompressor ..
>
>Not necessarily a FULL_HEADER, a COMPRESSED_NON_TCP would be a better
>choice.
>
>
>- Discussion: GeeVarghese and Koren about CONTEXT_UPDATE
>
>>>I think any of the CU+ loss will lead to CONTEXT_STATE packet generation
>from
>>>decompressor to compressor because of the link seq number mismatch ..Is=
 nt
>it ?
>
>>The decompressor should be in sync after receiving a CU+. There is no
>reason for a CS
>
>As the CRTP RFC says: "Note, however, that there is a nontrivial risk of an
>incorrect positive indication. It may be advisable to request a FULL_HEADER
>or COMPRESSED_NON_TCP packet even if the "twice" algorithm succeeds." The
>UDP checksum is rather weak, so especially if you use more complex "twice"
>solutions, the advice above should be followed, meaning that a=
 CONTEXT_STATE
>packet should be sent.
>
>
>Regards!!
>/Lars-Erik
>
>--------------------------------------------------------------
>Lars-Erik Jonsson, M.Sc.
>Voice Processing and Radio Network Research
>Ericsson Erisoft AB
>Box 920, S-971 28 Lule=E5, Sweden
>E-mail: lars-erik.jonsson@ericsson.com
>Phone: +46 920 20 21 07
>Mobile: +46 70 365 20 58
>Fax: +46 920 20 20 99
>Home: +46 920 999 57
>
>
>
>




From rem-conf Wed Nov 24 20:49:39 1999 
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Hi,
I think these questions can be better answered on this list.

Regards,
-Rafiq.
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From: "Bogdan Ghita" <b.ghita@jack.see.plym.ac.uk>
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Subject: RTP measuring
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Hello all,

I have three questions related to RTP/RTCP.
1. Are there any  tools for monitoring RTP/RTCP (free or commercial)? I =
am interested especially in RTP analysis, and not RTCP parsing. And, if =
there are, which are their capabilities?
In order to restrict the range of replies, I must mention that I've =
already found:
- Hammer VoIP Test System - analysing all the parameters (drop packets, =
jitter, delay) within a VoIP call
- HP Internet Advisor - similar
(for these two commercial packages, I couldn't get more information =
about what actually is monitored)
- rtpmon, which is free, but, in spite of its name, it is an RTCP =
monitor.

2. Can anybody explain me what is the structure of a compound RTCP =
packet? I know, from the RTP specification, that it looks like this:


if encrypted: random 32-bit integer
|
|[------- packet -------][----------- packet -----------][-packet-]
|
| receiver reports chunk chunk
V item item item item
--------------------------------------------------------------------
|R[SR|# sender #site#site][SDES|# CNAME PHONE |#CNAME LOC][BYE##why]
|R[  |# report #  1 #  2 ][    |#             |#         ][   ##   ]
|R[  |#        #    #    ][    |#             |#         ][   ##   ]
|R[  |#        #    #    ][    |#             |#         ][   ##   ]
--------------------------------------------------------------------
|<------------------ UDP packet (compound packet) --------------->|
#: SSRC/CSRC

The question is, as I couldn't understand the figure: how is the above =
figure packetised in UDP packets (is it 'line-by-line') ?=20

3. Are there any standards for encrypting RTP/RTCP (in relation with =
H.323) ? Besides inserting the 32-bit integer in the RTCP packet and =
using DES there is nothing else specified.


Thank you very much
Best regards
Bogdan Ghita


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	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Diso-8859-1" =
http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.2314.1000" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY background=3D"" bgColor=3D#ffffff>
<DIV><FONT face=3D"Courier New" size=3D2>Hello all,<BR><BR>I =
have&nbsp;three=20
questions related to RTP/RTCP.<BR>1. Are there any&nbsp; tools for =
monitoring=20
RTP/RTCP (free or commercial)? I am interested especially in RTP =
analysis, and=20
not RTCP parsing. And, if there are, which are their capabilities?<BR>In =
order=20
to restrict the range of replies, I must mention that I've already =
found:<BR>-=20
Hammer VoIP Test System - analysing all the parameters (drop packets, =
jitter,=20
delay) within a VoIP call<BR>- HP Internet Advisor - similar<BR>(for =
these two=20
commercial packages, I couldn't get more information about what actually =
is=20
monitored)<BR>- rtpmon, which is free, but, in spite of its name, it is =
an RTCP=20
monitor.<BR><BR>2. Can anybody explain me what is the structure of a =
compound=20
RTCP packet? I know, from the RTP specification, that it looks like=20
this:<BR><BR><BR>if encrypted: random 32-bit integer<BR>|<BR>|[------- =
packet=20
-------][----------- packet -----------][-packet-]<BR>|<BR>| receiver =
reports=20
chunk chunk<BR>V item item item=20
item<BR>-----------------------------------------------------------------=
---<BR>|R[SR|#=20
sender #site#site][SDES|# CNAME PHONE |#CNAME =
LOC][BYE##why]<BR>|R[&nbsp; |#=20
report #&nbsp; 1 #&nbsp; 2 ][&nbsp;&nbsp;&nbsp;=20
|#&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;=20
|#&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ][&nbsp;&nbsp; =
##&nbsp;&nbsp;=20
]<BR>|R[&nbsp; |#&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
#&nbsp;&nbsp;&nbsp;=20
#&nbsp;&nbsp;&nbsp; ][&nbsp;&nbsp;&nbsp;=20
|#&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;=20
|#&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ][&nbsp;&nbsp; =
##&nbsp;&nbsp;=20
]<BR>|R[&nbsp; |#&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
#&nbsp;&nbsp;&nbsp;=20
#&nbsp;&nbsp;&nbsp; ][&nbsp;&nbsp;&nbsp;=20
|#&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;=20
|#&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ][&nbsp;&nbsp; =
##&nbsp;&nbsp;=20
]<BR>--------------------------------------------------------------------=
<BR>|&lt;------------------=20
UDP packet (compound packet) ---------------&gt;|<BR>#: =
SSRC/CSRC<BR><BR>The=20
question is, as I couldn't understand the figure: how is the above =
figure=20
packetised in UDP packets (is it 'line-by-line') ? </FONT></DIV>
<DIV><FONT face=3D"Courier New" size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3D"Courier New" size=3D2>3. Are there any standards for =
encrypting=20
RTP/RTCP (in relation with H.323) ? Besides inserting the 32-bit integer =
in the=20
RTCP packet and using DES there is nothing else specified.</FONT></DIV>
<DIV><FONT face=3D"Courier New" size=3D2><BR><BR>Thank you very =
much</FONT></DIV>
<DIV><FONT face=3D"Courier New" size=3D2>Best regards</FONT></DIV>
<DIV><FONT face=3D"Courier New" size=3D2>Bogdan Ghita</FONT></DIV>
<DIV>&nbsp;</DIV></BODY></HTML>

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From rem-conf Thu Nov 25 00:07:47 1999 
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From: "Lars-Erik Jonsson" <lars-erik.jonsson@ericsson.com>
To: "GeeVarghese John" <gvjohn@miel.mot.com>, "Tmima Koren" <tmima@cisco.com>
Cc: <casner@cisco.com>, <micke@cdt.luth.se>, <brucet@cisco.com>,
        <dwing@cisco.com>, <pruddy@cisco.com>, <rem-conf@es.net>,
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Subject: Re: Extensions to CRTP
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>> =3D GVJ
>  =3D Tmima

>>Could you please explain, how IP id variance should be treated in this
case ?
>>For IP id field  first order difference is sent as delta in CUDP with I
bit set.
>>In this case, if there is a CU+ packet loss how IP ID delta encoding
should be treated ..


>Usually the IP id will not increment uniformly in packets of an RTP stre=
am
(the host sends packets other >then the RTP streams, silence periods).
>This means that if we want to compress a stream, we might have to send a
new deltaID often, maybe in each >packet. This will increase the size of =
the
packets.
>To optimize the compression, the host can assign IP id's to the packets =
of
each stream in a constant >increment.
>This keeps the deltaID constant for each stream.


>The advantage of a CU+ packet is that in one packet we restore all
RTP-related parameters at the >decompressor.
>The sequence of:  CU followed by a CR+  is very vulnerable: the CU zeroe=
s
the deltaT, so if the CR+ was lost, >the decompressor is out of sync.
>The CU+ does not restore the IP id. But if the implementation keeps the
deltaID constant..


Yes, IF the deltaID was constant. But you can never expect it to be.....s=
o
the compression should handle non-constant deltaID's also (without too mu=
ch
overhead).

>After a silence period, the sequence number will still increment by 1 an=
d
the IP id by deltaID: there is no >reason to skip RTP sequence numbers or=
 IP
id's. deltaT is the only difference that is not correct for the first
>packet after a silence period.


Yes, and therefore the additional overhead introduced by sending CU (or C=
U+)
packets seems to me indefensible. What you relly want is to send somethin=
g
more than just ONE packet with a different deltaT and avoid the situation
below:
...
...
N+1: deltaT=3D10
N+2: deltaT=3D100
N+3: deltaT=3D10
N+4: deltaT=3D10
...
...
Because a loss of packet N+2 would NOT be recoverable with Twice. That is
the point? Right?? I think (even if I do not trust Twice to be reliable,
both due to the weak UDP checksum and to the fact that the checksum will =
not
always be present) it would be nice to have a better way to do this, but =
I
think it could be done cheaper than by sending full RTP header++...=3D)

>To optimize CRTP over lossy links
>1. Use a constant deltaID

Do not expect it to be constant. To do so, you either need to change the
IP-stacks OR you must violate the end-to-end integrity of the packets and
reassign the ID "somewhere" in the path.....=3D(

>2. Check udp checksum

The Twice idea makes some sence, BUT you should not expect the checksum t=
o
be enabled. It is up to the application to decide whether to use it (and =
you
can not expect the application to know that it is needed for header
compression purposes).

>3. Use CU+ as necessary


Even if I am a little bit sceptical to some of your arguments, I still th=
ink
it is a good idea to improve the CU packet type and make it more flexible=
,
especially since there are "unused" bits in it.



Cheers!!
/Lars-Erik


--------------------------------------------------------------
Lars-Erik Jonsson, M.Sc.
Voice Processing and Radio Network Research
Ericsson Erisoft AB
Box 920, S-971 28 Lule=E5, Sweden
E-mail: lars-erik.jonsson@ericsson.com
Phone: +46 920 20 21 07
Mobile: +46 70 365 20 58
Fax: +46 920 20 20 99
Home: +46 920 999 57





From rem-conf Thu Nov 25 02:42:53 1999 
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To: "Lars-Erik Jonsson" <lars-erik.jonsson@ericsson.com>
From: Mikael Degermark <micke@cdt.luth.se>
Subject: Re: Extensions to CRTP
Cc: "GeeVarghese John" <gvjohn@miel.mot.com>, "Tmima Koren" <tmima@cisco.com>,
        <casner@cisco.com>, <micke@cdt.luth.se>, <brucet@cisco.com>,
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I also think it is a good idea to make CRTP more robust.

We cannot, however, expect that it can be made good enough
for the more severe cases (e.g., cellular) unless rather radical=20
modifications are done. The place to ponder such modifications
is the robhc WG, which will be announced shortly.=20

Micke Degermark

At 09:01 1999-11-25 +0100, Lars-Erik Jonsson wrote:
>>> =3D GVJ
>>  =3D Tmima
>
>>>Could you please explain, how IP id variance should be treated in this
>case ?
>>>For IP id field  first order difference is sent as delta in CUDP with I
>bit set.
>>>In this case, if there is a CU+ packet loss how IP ID delta encoding
>should be treated ..
>
>
>>Usually the IP id will not increment uniformly in packets of an RTP stream
>(the host sends packets other >then the RTP streams, silence periods).
>>This means that if we want to compress a stream, we might have to send a
>new deltaID often, maybe in each >packet. This will increase the size of=
 the
>packets.
>>To optimize the compression, the host can assign IP id's to the packets of
>each stream in a constant >increment.
>>This keeps the deltaID constant for each stream.
>
>
>>The advantage of a CU+ packet is that in one packet we restore all
>RTP-related parameters at the >decompressor.
>>The sequence of:  CU followed by a CR+  is very vulnerable: the CU zeroes
>the deltaT, so if the CR+ was lost, >the decompressor is out of sync.
>>The CU+ does not restore the IP id. But if the implementation keeps the
>deltaID constant..
>
>
>Yes, IF the deltaID was constant. But you can never expect it to be.....so
>the compression should handle non-constant deltaID's also (without too much
>overhead).
>
>>After a silence period, the sequence number will still increment by 1 and
>the IP id by deltaID: there is no >reason to skip RTP sequence numbers or=
 IP
>id's. deltaT is the only difference that is not correct for the first
>>packet after a silence period.
>
>
>Yes, and therefore the additional overhead introduced by sending CU (or=
 CU+)
>packets seems to me indefensible. What you relly want is to send something
>more than just ONE packet with a different deltaT and avoid the situation
>below:
>...
>...
>N+1: deltaT=3D10
>N+2: deltaT=3D100
>N+3: deltaT=3D10
>N+4: deltaT=3D10
>...
>...
>Because a loss of packet N+2 would NOT be recoverable with Twice. That is
>the point? Right?? I think (even if I do not trust Twice to be reliable,
>both due to the weak UDP checksum and to the fact that the checksum will=
 not
>always be present) it would be nice to have a better way to do this, but I
>think it could be done cheaper than by sending full RTP header++...=3D)
>
>>To optimize CRTP over lossy links
>>1. Use a constant deltaID
>
>Do not expect it to be constant. To do so, you either need to change the
>IP-stacks OR you must violate the end-to-end integrity of the packets and
>reassign the ID "somewhere" in the path.....=3D(
>
>>2. Check udp checksum
>
>The Twice idea makes some sence, BUT you should not expect the checksum to
>be enabled. It is up to the application to decide whether to use it (and=
 you
>can not expect the application to know that it is needed for header
>compression purposes).
>
>>3. Use CU+ as necessary
>
>
>Even if I am a little bit sceptical to some of your arguments, I still=
 think
>it is a good idea to improve the CU packet type and make it more flexible,
>especially since there are "unused" bits in it.
>
>
>
>Cheers!!
>/Lars-Erik
>
>
>--------------------------------------------------------------
>Lars-Erik Jonsson, M.Sc.
>Voice Processing and Radio Network Research
>Ericsson Erisoft AB
>Box 920, S-971 28 Lule=E5, Sweden
>E-mail: lars-erik.jonsson@ericsson.com
>Phone: +46 920 20 21 07
>Mobile: +46 70 365 20 58
>Fax: +46 920 20 20 99
>Home: +46 920 999 57
>
>
>
>
>



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Tmima,

Thanks for the reply and I have added some more comments on the CU+
packets in this
mail. I read the first mail from "Lars-Erik Jonsson", and your reply to
that, Good
you are changing the CU+ with IP ID but you may not need to set M bit
there,
by settting deltaT bit you can encode both deltaT and IP ID, because you
need both values...
I have added whatever comments I have about CU+,  please read that below

According to me CU+ with IP ID also will not give a robust solution.
Reason stated below.

I think we should windup our discussion on CU + with this mail unless
someone else is
interested. If you want to reply to me please ..If you want more
clarrification on my point
I will reply back.

My next mail will address few things about ACCEPT/REJECT
packet use, you & team proposed. Please reply to that mail.

My comments on CU+ are :

Point 1.1:

IP ID can be assigned by an host in three ways
a) sequence number in ascending order with variace 1
b) random number
c) sequence number with variance n (n >1)

I am not aware of any other implementation ..(May be possible)
IF the IP id variance is not in the ascending order,
CU+ will have problem. You expect that, host will implement this
while transmitting the RTP packets. I dont know how many (non compliant)
people
will change this for the new proposal came in "extending CRTP".
Or you are agreed for IP ID in CU+, it is a positive move.

Point # 1.2

Infact looking at the sequence number of CUDP packets helps to  get rid
of
IP id sync problem which we discussed .  (What I meant is if CUDP came
in with a
non expected link seqnumber, decompressor send out a CS packet to
compressor to synchronise the context)
This will take care all issues with the IP id variance (variance in any
manner random, ascending order).
If we extend CRTP with the new extension you proposed,
we may loose this synchronization chance between compressor and
decompressor,
because we are NOT expected to look at CUDP+ (or CU+) link sequence
number
to synchronize anymore as far as IP ID variance is concerned.
Otherwise CU+ packet should always carry IP ID field then we can avoid
looking at sequnce number but not in the case with CU packet.

What is your comment about CUDP with extension co-exist?
Should it look for link seq number match further also  ?

Point #2

One solution to IP ID prolem you stated as "Twics Algo"
Solution to this problem could be using twice algorithm.
But "twice" algorithm is not mandatory. There are implementations
without preserving UDP checksum. In this case I dont have an answer.
May be we have to force a standard that everyone should implement
UDP checksum while transmitting out. I dont know how many people will do
that.

Point #3.1

In our discussion we have addressed two kinds of packet loss

1)  RTP packet loss in the network before reaching compessor (Say L1)
2) Compressed packet loss sent from compressor to decompressor (Say L2)


In a lossy environment (L1), we end up in sending more number of encoded
bytes
with CU+ than what we used to send with CR+.
Because in CU+ we encode deltaT of the next expected RTP packet, with
the current packet.
I can understand your point that source generate with same periodic time
gap for
each packet within a talkburst. But the carrier network is IP datagram
service and
(+ link lossy nature). So should we rely on the possible network
behaviour or should we, on the source behaviour.

If L2 is greater than L1 then you claimed advantage of CU+ sync between
compressor
and decompressor can be achieved.

But What is the likely chance for L2 >  L1 ?
Only for some specific lossy links.

Point # 3.2

As far as I know CRTP use some heuristic mechanism to identify a RTP
flow.
It is not a definite mechanism. In this case  does it make sense to
send the deltaT of the next expected packet. Consider L1 case mentioned
above.
Not only that, assume the time gap between talkburst is (T1) and time
gap between
inter packet arrival of same session is T2. Consider a case of,
If we have identified a non-RTP flow as RTP flow and started compression

with our heuristics how do we determine when to send CU+.
Will we end up in sending continuous CU+ ?  we may ..
There need not be talkburst gap with some what high definite gap
when  compare to interpacket gap of the same session...
(See Pt # 4 on CU+ size comparison with CR+)


Point #4

Size of the CU+ is more than CR+ because of the compressed RTP header ..

in CR+. In CU+ you send complete RTP header as it is (i.e 12 byts).
In old CRTP, Assume that for some situation we have to encode RTP seq
number and
Timestamp, For ideal case encoded byte size will be 2 byts (1 for seq
number
+ 1 for deltaT) or otherwise 4 or max 6 (still less than 12 bytes).
Over a lossy link (L2), which is particular to the link, advantage of
CU+ is missing.

Continuous sending of CU+ for a talk burst may avoid send CS from
decompressor to
compressor. But What kind of loss is more likey L1 or L2.


My personal comment to CU+ is

May be we can claim some advantage only if
the stated assumptions (about packet loss (L1 < L2),  twice algo..etc)
are met ..
That also with the modification of IP ID inside the CU+  packet.

Thanks & Rgds

GVJ

Tmima Koren wrote:

>  At 10:17 PM 11/23/99 -0500, GeeVarghese John wrote:
>
>>
>>
>> >> >>
>> >> >>  The decompressor will stay in sync as long as he received at
>> >> >>  least one of the CU+
>> >> >
>> >>  I think any of the CU+ loss will lead to CONTEXT_STATE packet
>> >>  generation from
>> >>  decompressor to compressor because of the link seq number
>> >>  mismatch ..Is nt it ?
>> >
>> > The decompressor should be in sync after receiving a CU+. There is
>> > no reason for a CS
>> > Tmima
>>
>> Tmima,
>>
>> Could you please explain, how IP id variance should be treated in
>> this case ?
>> For IP id field  first order difference is sent as delta in CUDP
>> with I bit set.
>> In this case, if there is a CU+ packet loss how IP ID delta encoding
>> should be treated ..
>
>
> Usually the IP id will not increment uniformly in packets of an RTP
> stream (the host sends packets other then the RTP streams, silence
> periods).
> This means that if we want to compress a stream, we might have to send
> a new deltaID often, maybe in each packet. This will increase the size
> of the packets.
> To optimize the compression, the host can assign IP id's to the
> packets of each stream in a constant increment.
> This keeps the deltaID constant for each stream.
>
> The advantage of a CU+ packet is that in one packet we restore all
> RTP-related parameters at the decompressor.
> The sequence of:  CU followed by a CR+  is very vulnerable: the CU
> zeroes the deltaT, so if the CR+ was lost, the decompressor is out of
> sync.
> The CU+ does not restore the IP id. But if the implementation keeps
> the deltaID constant..
>
> After a silence period, the sequence number will still increment by 1
> and the IP id by deltaID: there is no reason to skip RTP sequence
> numbers or IP id's. deltaT is the only difference that is not correct
> for the first packet after a silence period.
>
> To optimize CRTP over lossy links
> 1. Use a constant deltaID
> 2. Check udp checksum
> 3. Use CU+ as necessary
>
>
>
>> and also I am not still convinced with the advantage of CU+ over CR+
>> after
>> the  silence period. Could you please explain your point  ..(please
>> ref my previous mail)
>
>
> Let's say that the characteristics of the link is that we may lose at
> most 1 packet in a row
> If we use:
>
> 11      200       CR+    dt=100
> 12      210       CR+    dt=10
> 13      220       CR
>
> If 11 is lost, the decompressor does not have the correct RTP
> timestamp for packet 12.
> If 12 is lost, the decompressor has a wrong deltaT.
> In both cases the decompressor is out of sync.
> This is a very weak sequence to use after a silence period.
>
> If instead we use:
>
> 11"    200       CU+    dt=10
> 12"    210       CU+    dt=10
> 13"    220       CR
>
> In this case If one of 11" or 12" is lost, the decompressor remains in
> sync. So this is a very strong sequence
>
> Tmima
>
>
>
>> Please reply .
>>
>> Rgds
>>
>> GVJ
>
-

--------------3314A3E2131FFCA19337BE73
Content-Type: text/html; charset=us-ascii
Content-Transfer-Encoding: 7bit

<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Tmima,
<p>Thanks for the reply and I have added some more comments on the CU+
packets in this
<br>mail. I read the first mail from "Lars-Erik Jonsson", and your reply
to that, Good
<br>you are changing the CU+ with IP ID but you may not need to set M bit
there,
<br>by settting deltaT bit you can encode both deltaT and IP ID, because
you need both values...
<br><b>I have added whatever comments I have about CU+,&nbsp; please read
that below</b>
<br><b>According to me CU+ with IP ID also will not give a robust solution.</b>
<br><b>Reason stated below.</b>
<p>I think we should windup our discussion on CU + with this mail unless
someone else is
<br>interested. If you want to reply to me please ..If you want more clarrification
on my point
<br>I will reply back.
<p>My next mail will address few things about ACCEPT/REJECT
<br>packet use, you &amp; team proposed. Please reply to that mail.
<p>My comments on CU+ are :
<p><b>Point 1.1:</b>
<p>IP ID can be assigned by an host in three ways
<br>a) sequence number in ascending order with variace 1
<br>b) random number
<br>c) sequence number with variance n (n >1)
<p>I am not aware of any other implementation ..(May be possible)
<br>IF the IP id variance is not in the ascending order,
<br>CU+ will have problem. You expect that, host will implement this
<br>while transmitting the RTP packets. I dont know how many (non compliant)
people
<br>will change this for the new proposal came in "extending CRTP".
<br>Or you are agreed for IP ID in CU+, it is a positive move.
<p><b>Point # 1.2</b>
<p>Infact looking at the sequence number of CUDP packets helps to&nbsp;
get rid of
<br>IP id sync problem which we discussed .&nbsp; (What I meant is if CUDP
came in with a
<br>non expected link seqnumber, decompressor send out a CS packet to
<br>compressor to synchronise the context)
<br>This will take care all issues with the IP id variance (variance in
any manner random, ascending order).
<br>If we extend CRTP with the new extension you proposed,
<br>we may loose this synchronization chance between compressor and decompressor,
<br>because we are NOT expected to look at CUDP+ (or CU+) link sequence
number
<br>to synchronize anymore as far as IP ID variance is concerned.
<br>Otherwise CU+ packet should always carry IP ID field then we can avoid
<br>looking at sequnce number but not in the case with CU packet.
<p>What is your comment about CUDP with extension co-exist?
<br>Should it look for link seq number match further also&nbsp; ?
<p><b>Point #2</b>
<p>One solution to IP ID prolem you stated as "Twics Algo"
<br>Solution to this problem could be using twice algorithm.
<br>But "twice" algorithm is not mandatory. There are implementations
<br>without preserving UDP checksum. In this case I dont have an answer.
<br>May be we have to force a standard that everyone should implement
<br>UDP checksum while transmitting out. I dont know how many people will
do that.
<p><b>Point #3.1</b>
<p>In our discussion we have addressed two kinds of packet loss
<p>1)&nbsp; RTP packet loss in the network before reaching compessor (Say
L1)
<br>2) Compressed packet loss sent from compressor to decompressor (Say
L2)
<br>&nbsp;
<p>In a lossy environment (L1), we end up in sending more number of encoded
bytes
<br>with CU+ than what we used to send with CR+.
<br>Because in CU+ we encode deltaT of the next expected RTP packet, with
the current packet.
<br>I can understand your point that source generate with same periodic
time gap for
<br>each packet within a talkburst. But the carrier network is IP datagram
service and
<br>(+ link lossy nature). So should we rely on the possible network behaviour
or should we, on the source behaviour.
<p>If L2 is greater than L1 then you claimed advantage of CU+ sync between
compressor
<br>and decompressor can be achieved.
<p>But What is the likely chance for L2 >&nbsp; L1 ?
<br>Only for some specific lossy links.
<p><b>Point # 3.2</b>
<p>As far as I know CRTP use some heuristic mechanism to identify a RTP
flow.
<br>It is not a definite mechanism. In this case&nbsp; does it make sense
to
<br>send the deltaT of the next expected packet. Consider L1 case mentioned
above.
<br>Not only that, assume the time gap between talkburst is (T1) and time
gap between
<br>inter packet arrival of same session is T2. Consider a case of,
<br>If we have identified a non-RTP flow as RTP flow and started compression
<br>with our heuristics how do we determine when to send CU+.
<br>Will we end up in sending continuous CU+ ?&nbsp; we may ..
<br>There need not be talkburst gap with some what high definite gap
<br>when&nbsp; compare to interpacket gap of the same session...
<br>(See Pt # 4 on CU+ size comparison with CR+)
<br>&nbsp;
<p><b>Point #4</b>
<p>Size of the CU+ is more than CR+ because of the compressed RTP header
..
<br>in CR+. In CU+ you send complete RTP header as it is (i.e 12 byts).
<br>In old CRTP, Assume that for some situation we have to encode RTP seq
number and
<br>Timestamp, For ideal case encoded byte size will be 2 byts (1 for seq
number
<br>+ 1 for deltaT) or otherwise 4 or max 6 (still less than 12 bytes).
<br>Over a lossy link (L2), which is particular to the link, advantage
of CU+ is missing.
<p>Continuous sending of CU+ for a talk burst may avoid send CS from decompressor
to
<br>compressor. But What kind of loss is more likey L1 or L2.
<br>&nbsp;
<p>My personal comment to CU+ is
<p>May be we can claim some advantage only if
<br>the stated assumptions (about packet loss (L1 &lt; L2),&nbsp; twice
algo..etc) are met ..
<br>That also with the modification of IP ID inside the CU+&nbsp; packet.
<p>Thanks &amp; Rgds
<p>GVJ
<p>Tmima Koren wrote:
<blockquote TYPE=CITE>&nbsp;At 10:17 PM 11/23/99 -0500, GeeVarghese John
wrote:
<blockquote type=cite cite>&nbsp;
<blockquote type=cite cite>
<blockquote type=cite cite>
<blockquote type=cite cite>
<blockquote type=cite cite>&nbsp;
<br>The decompressor will stay in sync as long as he received at least
one of the CU+</blockquote>
</blockquote>
I think any of the CU+ loss will lead to CONTEXT_STATE packet generation
from
<br>decompressor to compressor because of the link seq number mismatch
..Is nt it ?</blockquote>

<p>The decompressor should be in sync after receiving a CU+. There is no
reason for a CS
<br>Tmima</blockquote>
Tmima,
<p>Could you please explain, how IP id variance should be treated in this
case ?
<br>For IP id field&nbsp; first order difference is sent as delta in CUDP
with I bit set.
<br>In this case, if there is a CU+ packet loss how IP ID delta encoding
should be treated ..</blockquote>

<p><br>Usually the IP id will not increment uniformly in packets of an
RTP stream (the host sends packets other then the RTP streams, silence
periods).
<br>This means that if we want to compress a stream, we might have to send
a new deltaID often, maybe in each packet. This will increase the size
of the packets.
<br>To optimize the compression, the host can assign IP id's to the packets
of each stream in a constant increment.
<br>This keeps the deltaID constant for each stream.
<p>The advantage of a CU+ packet is that in one packet we restore all RTP-related
parameters at the decompressor.
<br>The sequence of:&nbsp; CU followed by a CR+&nbsp; is very vulnerable:
the CU zeroes the deltaT, so if the CR+ was lost, the decompressor is out
of sync.
<br>The CU+ does not restore the IP id. But if the implementation keeps
the deltaID constant..
<p>After a silence period, the sequence number will still increment by
1 and the IP id by deltaID: there is no reason to skip RTP sequence numbers
or IP id's. deltaT is the only difference that is not correct for the first
packet after a silence period.
<p>To optimize CRTP over lossy links
<br>1. Use a constant deltaID
<br>2. Check udp checksum
<br>3. Use CU+ as necessary
<br>&nbsp;
<br>&nbsp;
<blockquote type=cite cite>and also I am not still convinced with the advantage
of CU+ over CR+ after
<br>the&nbsp; silence period. Could you please explain your point&nbsp;
..(please ref my previous mail)</blockquote>

<p><br>Let's say that the characteristics of the link is that we may lose
at most 1 packet in a row
<br>If we use:
<p>11&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 200&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR+&nbsp;&nbsp;&nbsp; dt=100
<br>12&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 210&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR+&nbsp;&nbsp;&nbsp; dt=10
<br>13&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 220&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR
<p>If 11 is lost, the decompressor does not have the correct RTP timestamp
for packet 12.
<br>If 12 is lost, the decompressor has a wrong deltaT.
<br>In both cases the decompressor is out of sync.
<br>This is a very weak sequence to use after a silence period.
<p>If instead we use:
<p>11"&nbsp;&nbsp;&nbsp; 200&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; CU+&nbsp;&nbsp;&nbsp;
dt=10
<br>12"&nbsp;&nbsp;&nbsp; 210&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; CU+&nbsp;&nbsp;&nbsp;
dt=10
<br>13"&nbsp;&nbsp;&nbsp; 220&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; CR
<p>In this case If one of 11" or 12" is lost, the decompressor remains
in sync. So this is a very strong sequence
<p>Tmima
<br>&nbsp;
<br>&nbsp;
<blockquote type=cite cite>Please reply .
<p>Rgds
<p>GVJ</blockquote>
</blockquote>

<p>-</html>

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From rem-conf Thu Nov 25 19:23:17 1999 
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From rem-conf Fri Nov 26 04:33:03 1999 
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From: "Norihito Takatori" <takatori@isl.melco.co.jp>
To: <rem-conf@es.net>
Subject: Question about the MPEG-4 SL Payload Format
Date: Fri, 26 Nov 1999 21:26:49 +0900
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Hello all,

I am implementing a RTP system in compliance with I-D
$B!H(Bdraft-ietf-avt-rtp-mpeg4-01.txt$B!I(B. In the draft, it is written that
Sequence Number is derived from the sequenceNumber field of the SL
packet by adding a constant random offset.
I have a question as regards the Sequence Number.  My understanding is
that when a RTP client issues $B!H(BSEEK$B!I(B operation from current position
to a previous position (e.g. 0 position) , the Sequence Number should
return the previous Sequence Number.
On the other hand, it is written that the sequence number increments
by one for each RTP data packet sent in RFC1889.
Which should I comply with ?
Also, I guess that the same problem will occur as regards the
timestamp.

Thanks.

---------------------------------
Norihito Takatori
E-mail: takatori@isl.melco.co.jp
Information$B!!(BTechnology$B!!(BR$B!u(BD$B!!(BCenter
Mitsubishi Electric Corporation
TEL: +81-467-41-2103,  FAX: +81-467-41-2137




From rem-conf Fri Nov 26 11:25:26 1999 
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From: "Ram Kordale" <kordale@relic.engr.sgi.com>
Message-Id: <9911261111.ZM1385@relic.engr.sgi.com>
Date: Fri, 26 Nov 1999 11:11:07 -0800
In-Reply-To: "Norihito Takatori" <takatori@isl.melco.co.jp>
        "Question about the MPEG-4 SL Payload Format" (Nov 26,  9:26pm)
References: <00b801bf3809$83615b40$1f044a0a@isl.melco.co.jp>
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Subject: Re: Question about the MPEG-4 SL Payload Format
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Generally speaking, the sequence number of consecutive RTP packets should
either stay the same (allowing for deliberate repeated packets) or be
incremented. This mechanism is used to gracefully handle problems arising from
"non-ideal" networks.

On the other hand, since the timestamp reflects the sampling instant of the
first octet in the RTP data packet, the timestamp of an RTP packet after
a reposition can indeed be lower.

Ram

On Nov 26,  9:26pm, Norihito Takatori wrote:
> Subject: Question about the MPEG-4 SL Payload Format
> Hello all,
>
> I am implementing a RTP system in compliance with I-D
> $B!H(Bdraft-ietf-avt-rtp-mpeg4-01.txt$B!I(B. In the draft, it is written
that
> Sequence Number is derived from the sequenceNumber field of the SL
> packet by adding a constant random offset.
> I have a question as regards the Sequence Number.  My understanding is
> that when a RTP client issues $B!H(BSEEK$B!I(B operation from current
position
> to a previous position (e.g. 0 position) , the Sequence Number should
> return the previous Sequence Number.
> On the other hand, it is written that the sequence number increments
> by one for each RTP data packet sent in RFC1889.
> Which should I comply with ?
> Also, I guess that the same problem will occur as regards the
> timestamp.
>
> Thanks.
>
> ---------------------------------
> Norihito Takatori
> E-mail: takatori@isl.melco.co.jp
> Information$B!!(BTechnology$B!!(BR$B!u(BD$B!!(BCenter
> Mitsubishi Electric Corporation
> TEL: +81-467-41-2103,  FAX: +81-467-41-2137
>
>-- End of excerpt from Norihito Takatori



-- 
Ram Kordale
Advanced Media Products



From rem-conf Fri Nov 26 20:02:10 1999 
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Dear Sirs,

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We are an internet, intramet and communication products and services
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We request you to register us .

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Fax : 0091- 452 - 726950




From rem-conf Fri Nov 26 20:49:29 1999 
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From rem-conf Sat Nov 27 07:27:55 1999 
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To: Norihito Takatori <takatori@isl.melco.co.jp>
cc: rem-conf <rem-conf@es.net>
Subject: Re: Question about the MPEG-4 SL Payload Format
In-Reply-To: Message from Norihito Takatori <takatori@isl.melco.co.jp> of "Fri, 26 Nov 1999 21:26:49 +0900." <00b801bf3809$83615b40$1f044a0a@isl.melco.co.jp>
Date: Sat, 27 Nov 1999 15:08:25 +0000
From: Colin Perkins <c.perkins@cs.ucl.ac.uk>
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--> Norihito Takatori writes:
>I am implementing a RTP system in compliance with I-D
>draft-ietf-avt-rtp-mpeg4-01.txt. In the draft, it is written that Sequence
>Number is derived from the sequenceNumber field of the SL packet by adding
>a constant random offset.  I have a question as regards the Sequence
>Number.  My understanding is that when a RTP client issues SEEK operation
>from current position to a previous position (e.g. 0 position) , the
>Sequence Number should return the previous Sequence Number.  On the other
>hand, it is written that the sequence number increments by one for each
>RTP data packet sent in RFC1889.  Which should I comply with ?  Also, I
>guess that the same problem will occur as regards the timestamp.

This is actually a generic issue when streaming from a file, rather than an
MPEG-4 specific issue. 

The RTP sequence number must be incremented by one for each packet sent. If
packets are sent with a previous sequence number, the receiver may discard
them as duplicates (and the RTCP statistics will be incorrect). 

Receivers use the RTP timestamp to determine when to playout media data. If
this jumps backwards as a result of a seek operation, this indicates to the
receiver that the data is old - has arrived too late for playout - and
should be discarded. 

The result is that you should remap the sequence number and timestamp to be
a contiguous sequence when seeking, as if the data were a single continuous 
stream.

Colin



From rem-conf Sat Nov 27 17:18:06 1999 
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From rem-conf Mon Nov 29 04:07:09 1999 
From rem-conf-request@es.net Mon Nov 29 04:07:08 1999
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Date: Mon, 29 Nov 1999 01:34:14 -0200 (EDT)
From: "LANOMS'99" <lanoms99@lrg.ufsc.br>
Subject: IEEE LANOMS99 - THANK YOU VERY MUCH
To: activenets_all@ittc.ukansas.edu, ActiveNets_Wire@ittc.ukans.edu,
        alg@comm.toronto.edu, amlist@takilab.k.dendai.ac.jp,
        apc@ee.nthu.edu.tw, apc@eee.nthu.edu.tw,
        apc_members@hornbill.ee.nus.sg, apnoms-all@nile.postech.ac.kr,
        apnoms-oc@amazon.postech.ac.kr, apnoms-oc@nile.postech.ac.kr,
        arpanet-bboard@mc.lcs.mit.edu, atm@bbn.com, atm@sun.com,
        bd_email@inesc.pt, bm-list1@cs.ucdavis.edu, cabernet-events@ncl.ac.uk,
        cabernet-general@newcastle.ac.uk, ccrc@dworkin.wustl.edu,
        celluar@dfv.rwth-aachen.edu, cellular@comnets.rwth-aachen.de,
        cif@injector.ca.sandia.gov, cnom@maestro.bellcore.com,
        cnon@maestro.bellcore.com, commsoft@cc.belcore.com,
        commsoft@cc.bellcore.com, comsoc.bog@tab.ieee.org,
        comsoc.tac@tab.ieee.org, comsoc-chapters@ieee.org,
        comsoc-gicb@ieee.org, comswtc@gmu.edu,
        cost237-transport@comp.lancs.ac.uk, cost237-transport@comp.lancs.at,
        cpmsoc-gicb@ieee.org, ctc-members@redbank.tinac.com,
        dbword@cs.wisc.edu, ekpark@cstp.umkc.edu, end2end-interest@isi.edu,
        engp5273@leonis.nus.sg, enternet@bbn.com, events@teco.edu,
        fokus-user@fokus.gmd.de, f-troup@codex.cis.upenn.edu,
        ftroup@dsl.cis.upenn.edu, gcomlist1@manjaro.ece.iit.edu,
        giga@tele.pitt.edu, g-troup@ccrc.wustl.edu, heads@hpovua.org,
        hipparch@entropy.inria.fr, hipparch@sophia.inria.fr,
        Hossam.Afifi@enst-bretagne.fr, ieee.announce@bellcore.com,
        ieee_rtc_list@cs.tamu.edu, ieeetcpc@ccvm.sunysb.edu, ietf@ietf.org,
        ifip-6-1distr@run.montefiore.ulg.ac.be, im-oc@doc.ic.ac.uk,
        int-serv@isi.edu, isadsoc@fokus.gmd.de, issll@mercury.lcs.mit.edu,
        itc@fokus.gmd.de, itc@ieee.org, kia@usl.edu,
        MariaLuisa.Rossari@CSELT.IT, members@hpovua.org, members@tmforum.org,
        mobile-ip@tadpole.com, modern-heuristics@uk.ac.mailbase.ucdavis.edu,
        multicomm@cc.bellcore.com, multicomm@research.panasonic.com,
        nm-australia@amazon.postech.ac.kr, nmkorea@amazon.postech.ac.kr,
        Omar.Elloumi@enst-bretagne.fr, opensig-announce@ctr.columbia.edu,
        osimcast@bbn.com, performance@haven.epm.ornl.gov, prs@masi.ibp.fr,
        qos-iso@mci.org.uk, qosws@dstc.edu.au, rem-conf@es.net, reres@laas.fr,
        s.low@ee.mu.oz.au, sb.all@ieee.org, sc6wg4@ntd.comsat.com,
        sig-dce@opengroup.org, sigmedia@bellcore.com, tccc@ieee.org,
        tchen@seas.smu.edu, testnet@canarie.ca, tm_member@ns.ieice.or.jp,
        wwlu@ieee.org, xtprelay@cs.concordia.ca, lyko@kt.agh.edu.pl
In-Reply-To: <199910231940.RAA14214@lrg-gw.lrg.ufsc.br>
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The organizers of the IEEE LANOMS99 want to say THANK YOU VERY 
MUCH for your attention and contribution associated with the 
realization of the First IEEE LANOMS.

**************************************************************
*       You are very welcome to participate in the           *
*                                                            *
*                      IEEE LANOMS99                         *
*                                                            *
* LATIN AMERICAN NETWORK OPERATIONS AND MANAGEMENT SYMPOSIUM *
*                                                            *
*            http://www.lrg.ufsc.br/~lanoms99/               *
*                                                            *
*                   Rio Atlantica Hotel                      *
*           Av. Atlantica, 2964 - Copacabana                 *
*                Rio de Janeiro - Brazil                     *
*                   December 3-5, 1999                       *
**************************************************************




From rem-conf Mon Nov 29 06:32:46 1999 
From rem-conf-request@es.net Mon Nov 29 06:32:45 1999
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Date: Mon, 29 Nov 1999 06:04:42 -0800
To: "LANOMS'99" <lanoms99@lrg.ufsc.br>,activenets_all@ittc.ukansas.edu,
 ActiveNets_Wire@ittc.ukans.edu,alg@comm.toronto.edu,
 amlist@takilab.k.dendai.ac.jp,apc@ee.nthu.edu.tw,apc@eee.nthu.edu.tw,
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 apnoms-oc@amazon.postech.ac.kr,apnoms-oc@nile.postech.ac.kr,
 arpanet-bboard@mc.lcs.mit.edu,atm@bbn.com,atm@sun.com,bd_email@inesc.pt,
 bm-list1@cs.ucdavis.edu,cabernet-events@ncl.ac.uk,
 cabernet-general@newcastle.ac.uk,ccrc@dworkin.wustl.edu,
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 commsoft@cc.bellcore.com,comsoc.bog@tab.ieee.org,
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 giga@tele.pitt.edu,g-troup@ccrc.wustl.edu,heads@hpovua.org,
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 members@hpovua.org,members@tmforum.org,mobile-ip@tadpole.com,
 modern-heuristics@uk.ac.mailbase.ucdavis.edu,multicomm@cc.bellcore.com,
 multicomm@research.panasonic.com,nm-australia@amazon.postech.ac.kr,
 nmkorea@amazon.postech.ac.kr,Omar.Elloumi@enst-bretagne.fr,
 opensig-announce@ctr.columbia.edu,osimcast@bbn.com,
 performance@haven.epm.ornl.gov,prs@masi.ibp.fr,qos-iso@mci.org.uk,
 qosws@dstc.edu.au,rem-conf@es.net,reres@laas.fr,s.low@ee.mu.oz.au,
 sb.all@IEEE.ORG,sc6wg4@ntd.comsat.com,sig-dce@opengroup.org,
 sigmedia@bellcore.com,tccc@IEEE.ORG,tchen@seas.smu.edu,
 testnet@canarie.ca,tm_member@ns.ieice.or.jp,wwlu@IEEE.ORG,
 xtprelay@cs.concordia.ca,lyko@kt.agh.edu.pl
From: Ian@NC-Labs.com (Ian Nandhra)
Subject: Re: IEEE LANOMS99 - THANK YOU VERY MUCH
In-Reply-To: <Pine.3.89.9911290147.A17000-0100000@lrg-gw.lrg.ufsc.br>
References: <199910231940.RAA14214@lrg-gw.lrg.ufsc.br>
Mime-Version: 1.0
Content-Type: text/plain; charset="us-ascii"
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Thank you very much for SPAMMING assorted people with your email. Aside
>from the SPAM, you have also included a visible distribution list with the
email.

MAYBE YOU SHOULD ATTEND YOUR OWN SYMPOSIUM AND LEARN AT LEAST **SOMETHING**
ABOUT EMAIL AND THE INTERNET.

Pathetic!

At 01:34 AM 11/29/99 -0200, LANOMS'99 wrote:
>The organizers of the IEEE LANOMS99 want to say THANK YOU VERY 
>MUCH for your attention and contribution associated with the 
>realization of the First IEEE LANOMS.
>
>**************************************************************
>*       You are very welcome to participate in the           *
>*                                                            *
>*                      IEEE LANOMS99                         *
>*                                                            *
>* LATIN AMERICAN NETWORK OPERATIONS AND MANAGEMENT SYMPOSIUM *
>*                                                            *
>*            http://www.lrg.ufsc.br/~lanoms99/               *
>*                                                            *
>*                   Rio Atlantica Hotel                      *
>*           Av. Atlantica, 2964 - Copacabana                 *
>*                Rio de Janeiro - Brazil                     *
>*                   December 3-5, 1999                       *
>**************************************************************
>
>
--
Ian Nandhra                         Phone : 209 533 4559
NC Laboratories Inc.                FAX   : 209 532 1578
Suite 17-107, 18711 Tiffeni Drive,  email : ian@nc-labs.com
Twain Harte, CA 95383, USA.         http//www.nc-labs.com



From rem-conf Mon Nov 29 10:13:31 1999 
From rem-conf-request@es.net Mon Nov 29 10:13:31 1999
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From: Florissa Colina <florissa@bmrc.berkeley.edu>
Subject: 12/1 The Berkeley-MIT Citywalk Project:  Towards Highly
  Interactive, Large-Scale Acquired Models --Richard Bukowski, Computer
  Science Division - EECS,  U.C. Berkeley
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Berkeley Multimedia, Graphics and Interfaces Seminar

The Berkeley-MIT Citywalk Project:  Towards Highly Interactive, Large-Scale
Acquired Models

Wednesday December 1, 1999 
1:00-2:30 p.m. PDT
Fujitsu Seminar Room (405 Soda Hall) 


Richard Bukowski
Computer Science Division - EECS,  U.C. Berkeley 

Over the last year, groups at Berkeley and MIT have been working together
to merge the Berkeley Firewalk/Architectural Walkthrough system with the
MIT Citywalk city scanning project. The result is a system that combines
the highly interactive, simulation-enriched Berkeley visualization tools
with real-world data acquired by the MIT Argus probe and reconstructed into
exterior models of the as-built MIT campus. Our goal is to provide an
integrated system that has the ability to visualize models several orders
of magnitude larger than those we have previously rendered (i.e. the Soda
Hall interior), as well as to allow multiple distributed users to interact
with and run simulations on those models in a shared virtual world, and to
store and index semantic information about those models. Because our models
are derived from real-world data, they can be applied to real-world
problems such as site and building lifecycle management, construction
management, building plan evaluation, and fire safety training. 

This talk will present background and preliminary work on extending the
Walkthrough's database and systems to handle both outdoor environments and
distributed, multi-user database interaction and editing. These extensions
achieve these goals while preserving the viewpoint and data management
systems that allow Walkthrough to present huge, interactive models at
real-time frame rates. 
---------------

The seminar is broadcast live on the Internet Mbone and as a Real
Networks G2 broadcast.

You can connect to the live RealNetworks broadcast at:

http://media2.bmrc.berkeley.edu/bibs/schedule.cfm 
You'll see a link to cs298-5/real networks/live programs.  

You can also directly put in this url into the Real Player:

rtsp://media2.bmrc.berkeley.edu/encoder/cs298-5.rm

To do so you will need to have the "Real Player G2" installed, which is
available from:
http://www.real.com/products/player/downloadrealplayer.html

To tune into the Internet MBone broadcast, look for the announcement in
your MBone session directory program ('sdr').  If you are not receiving
the announcement you may be able to receive the session by manually
configuring the client programs ('vic', and 'vat') with the session
addresses:

low bit rate
	video: 233.0.25.1/22334
	audio: 233.0.25.2/22446

medium bit rate
	video: 233.0.25.129/22334
	audio: 233.0.25.2/22446

You can get further information about this seminar, and access to
replays of previous seminars at the MIG Seminar web page:

http://bmrc.berkeley.edu/298/index.html

Sponsored by the Berkeley Multimedia Research Center 

____________________________________________

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 Nov 30 02:39:00 1999 
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Subject: [ mannatech mailing list ] "Something New & Natural"
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From rem-conf Tue Nov 30 04:08:53 1999 
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: RTP Payload for DTMF Digits, Telephony Tones and 
                          Telephony Signals
	Author(s)	: H. Schulzrinne, S. Petrack
	Filename	: draft-ietf-avt-tones-03.txt,.ps
	Pages		: 28
	Date		: 29-Nov-99
	
This memo describes how to carry dual-tone multifrequency (DTMF)
signaling, other tone signals and telephony events in RTP packets.

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From rem-conf Tue Nov 30 14:10:49 1999 
From rem-conf-request@es.net Tue Nov 30 14:10:48 1999
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Subject: RTP Setup Signaling
Date: Tue, 30 Nov 1999 17:01:17 -0500
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I am working on a project that is required to support RTP and RTCP.  I've 
read a lot about RTP in general, but I still have not found any information 
about setting up the RTP session.

I am assuming (for the sake of this discussion) that RTP is not used with 
any other standards like H.323 or RSVP.  Is there some setup signaling used 
so that the receiver is informed of the port numbers (RTP and RTCP) that 
were decided for the session?

I read some document on the Internet that said Session Description Protocol 
(SDP) was used to communicate the port numbers.  However, I wasn't convince 
that this protocol was being used in applications like LiveMedia or 
Netmeeting.

Any information that you can share concerning the setup signaling will be 
appreciated.



From rem-conf Tue Nov 30 15:32:31 1999 
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To: "foy01@qwestinternet.net" <foy01@qwestinternet.net>
cc: "'rem-conf@es.net'" <rem-conf@es.net>
Subject: Re: RTP Setup Signaling
In-Reply-To: Message from Patrick Foy <foy01@qwestinternet.net> of "Tue, 30 Nov 1999 17:01:17 EST." <01BF3B54.85473030.foy01@qwestinternet.net>
Date: Tue, 30 Nov 1999 23:28:27 +0000
From: Colin Perkins <c.perkins@cs.ucl.ac.uk>
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--> Patrick Foy writes:
>I am working on a project that is required to support RTP and RTCP.  I've 
>read a lot about RTP in general, but I still have not found any information 
>about setting up the RTP session.
>
>I am assuming (for the sake of this discussion) that RTP is not used with 
>any other standards like H.323 or RSVP.  Is there some setup signaling used 
>so that the receiver is informed of the port numbers (RTP and RTCP) that 
>were decided for the session?
>
>I read some document on the Internet that said Session Description Protocol 
>(SDP) was used to communicate the port numbers.  However, I wasn't convince 
>that this protocol was being used in applications like LiveMedia or 
>Netmeeting.
>
>Any information that you can share concerning the setup signaling will be 
>appreciated.

There is no built-in signalling in RTP, some other protocol must be used.
Applications use something like H.323, SIP, RTSP or SAP to setup the RTP
session. These protocols convey session descriptions, often in SDP format.

Colin



From rem-conf Tue Nov 30 17:47:54 1999 
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From rem-conf Tue Nov 30 20:18:59 1999 
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Date: Wed, 1 Dec 1999 09:28:30 +0530 (IST)
From: Ramana Yarlagadda <ramana@chiplogic.com>
To: Patrick Foy <foy01@qwestinternet.net>
cc: "'rem-conf@es.net'" <rem-conf@es.net>
Subject: Re: RTP Setup Signaling
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Patrcik,

In H323 umberalla standards H245 is used to setup an RTP session

-cheers
-ramana

On Tue, 30 Nov 1999, Patrick Foy wrote:

> I am working on a project that is required to support RTP and RTCP.  I've 
> read a lot about RTP in general, but I still have not found any information 
> about setting up the RTP session.
> 
> I am assuming (for the sake of this discussion) that RTP is not used with 
> any other standards like H.323 or RSVP.  Is there some setup signaling used 
> so that the receiver is informed of the port numbers (RTP and RTCP) that 
> were decided for the session?
> 
> I read some document on the Internet that said Session Description Protocol 
> (SDP) was used to communicate the port numbers.  However, I wasn't convince 
> that this protocol was being used in applications like LiveMedia or 
> Netmeeting.
> 
> Any information that you can share concerning the setup signaling will be 
> appreciated.
> 




