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CC: Claude Ghaoui <C.GHAOUI@livjm.ac.uk>
Subject: ,        Collaboration in Multimedia NA and Groupware (Deadline is 
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				CALL FOR PAPERS
  
	  Special Session on Collaboration in Multimedia 
		Networked Applications and GroupWare
Session Co-ordinators: Michel Diaz, Khalil Drira, Thierry Villemur ({diaz,drira,villemur}@laas.fr)

	at the EUROMICRO WORKSHOP on Multimedia and Telecommunication
		Maastricht, Netherlands, Sept. 4-7, 2000
	Workshop Programme Chair: Claude Ghaoui (c.ghaoui@livjm.ac.uk)
 
		http://www.laas.fr/~khalil/cmnag.html


Important Dates 
Deadline for Paper Submission and Compositions: February 28th, 2000 
Notification of acceptance: May 2nd, 2000 
Camera-ready paper due: June 15th, 2000 
Deadline for proposals(of Other Submissions): October 15th, 1999 
Proposal acceptance: October 30th, 1999 
  
Overview 
The Euromicro workshop is a forum for interdisciplinary research and practice in computer systems, interfaces, 
architectures and applications. As part of the year 2000 conference in the Netherlands there will be a special session 
looking at . This session is organised as part of the Workshop on Multimedia and Telecommunication. 
  
Background 
The purpose of this session is to bring together researchers, and practitioners working on cooperation, coordination 
and communication in Multimedia Networked Applications and GroupWare. The session serves as a forum enabling 
experience exchange between academia and industry as well as between researchers working in the different 
multimedia and groupware  research branches. 
Topics of interest include but are not limited to: 
+ Formal and semi-formal methods for groupware and multimedia  applications modeling and analysis
+ architecture of groupware and multimedia  applications
+ object oriented design and implementation of cooperation and coordination protocols
+ coordination models and applications for CSCW using multimedia applications
+ group membership protocols
+ applications in  Java and CORBA

Queries 
Any questions regarding submissions for this session should be made (ideally via email) to one of the session 
coordinators , but full papers should be sent to the Programme Chair directly (Claude Ghaoui). 
The Call for Papers for the whole workshop can be viewed in pdf format here. 

Submission of Papers 
There is no restriction on the number or type of submissions. For technical papers, both regular papers 
and short contributions are invited. Authors should send their full paper to the Programme 
Chair directly as: 1) a PostScript version by Email, and 2) four copies by Post. The papers 
should not exceed 4000 words and should include an abstract of up to 150 words. The title page 
should be separate and clearly show the name, full mailing address, e-mail and fax number of the 
author to contact, as well as the topic areas of the submitted paper. Papers will be blind-refereed 
by three reviewers who are expert in the domain area. Accepted papers will appear in the 
proceedings which will be published by the IEEE Computer Society and be available at the 
conference. Papers exceeding 8 pages will be charged NLG 100 per page in excess. 
The following signed statement should be included on the cover page: "Neither this paper nor 
any version close to it has been or is being offered elsewhere for publication. All necessary 
clearances have been obtained for the publication of this paper. If accepted, the paper will be 
made available in Camera-ready forms by June 15th, 2000, and it will be personally presented at 
the EUROMICRO 2000 Conference by the author or one of the co-authors. The presenting 
author(s) will pre-register for EUROMICRO 2000 before the due date of the Camera-ready 
paper." 
  


Journal Publication for Best Papers 
In addition, selected best papers will be considered for journal publication. Authors of these papers will be advised 
on submitting extended versions of their originals for this purpose. Journals offered are: 
+ Special issue, Journal of Network and Computer Applications (JNCA), Guest Editor is Claude Ghaoui, 
'Support for Open and Distance Learning on the WWW', Academic Press pub. 
+ Special issue, J. `Digital Creativity', Swets pub. 
+ European J. Engineering Educ. (EJEE), SEFI pub. 
+ J.Interactive Learning Environments, Intellect pub. 
+ International J. of Knowledge and Information Systems (KAIS), Springer-Verlag pub. 
+ Real Time Graphics & ACM SIGBIO Newsletters 
  


Other Submissions  
Special Sessions, Panels, Half-day or 1-day tutorials and Short Courses (90-180 minutes) on 
related topics are welcomed. Companies and students working on Projects of different levels 
(undergrad., postgrad. or industrially-oriented) are invited to participate in a Project Showcase. 
Submission and reviewing will follow the same procedure of the regular papers. Email the 
proposal to the programme chair, including: your submission title and aims, targeted audience, 
your plan of how to make it a success, short CV and full contact address of the proposer (with 
email) and any other supporting information. For special sessions or panels, please include a 
detailed Call for Papers/Participation. 
  


Keynote Speaker 
Prof. H. Maurer, Institute for Information Processing and Computer Supported Media, Technical 
University Graz, Austria. His main research areas are: multimedia systems; E. publishing, Web based 
learning; telematic; computer networks; CAL; computer supported media & social implications. He is 
the author of fourteen books, over 450 scientific contributions and dozens of multimedia products. He is 
the project manager of a number of multimillion-dollar projects. 
(see http://www.iicm.edu/maurer) 
  


Programme Chair 
Dr Claude Ghaoui 
School of Computing and Mathematical Sciences, 
Liverpool John Moores University, 
Byrom Street, 
Liverpool L3 3AF 
United Kingdom 
Tel: +44 151 .231 2276 
Fax: +44 151 207 4594 
C.GHAOUI@LIVJM.AC.UK 

Session Coordinators 
Michel Diaz , Khalil Drira , Thierry Villemur
Communication Tools and Software Group 
LAAS-CNRS 
7, Av. Colonel Roche 
31077 Toulouse Cedex 4 
FRANCE 
e-mail: diaz@laas.fr ,  drira@laas.fr ,  villemur@laas.fr 
Tel: +33 5 61 33 62 56 / +33 5 61 33 63 22 / +33 5 61 33 69 55
Fax: +33 5 61 33 64 11 



General Information 
The workshop is an integral part of the EUROMICRO 2000 conference. 
UPDATED INFORMATION 
See the EUROMICRO WWW : http://www.euromicro.org 


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From rem-conf Tue Feb 01 11:08:14 2000 
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From: Florissa Colina <florissa@bmrc.berkeley.edu>
Subject: 2/2/2000 Estimation of Web Video Multiplicity -- Samson
  Cheung, EECS Department, U.C. Berkeley 
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Berkeley Multimedia, Graphics and Interfaces Seminar

Estimation of Web Video Multiplicity

Wednesday February 2, 2000  
1:10 - 2:30 PST 
Fujitsu Seminar Room (405 Soda Hall)


Sen-ching Samson Cheung
EECS Department, U.C. Berkeley 

With ever more popularity of video web-publishing, many popular contents
are being mirrored, modified
and republished, resulting in excessive content duplication. While such
redundancy provides fault
tolerance for continuous availability of information, it could potentially
make multimedia web search
results repetitious and cluttered. As such, developing techniques for
detecting similarity and duplication
is important to multimedia search engines. Content providers might also be
interested in identifying
duplicates of their content for business related reasons. In this talk, we
will present an efficient algorithm,
called video signature, to detect similar video sequences for large
databases. The idea is to first form a
"signature" for each video sequence by selecting a small number of its
frames that are most similar to a
number of randomly chosen seed images. Then the similarity between any two
video sequences can be
reliably estimated by comparing their respective signatures. Using this
method, we achieve 85% recall
and precision ratios on a test database of 377 video sequences. We have
applied our proposed
algorithm to a collection of 1800 hours of video corresponding to around
45000 clips from the web. Our
results indicate that, on average, every video in our collection has around
five similar copies.

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

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 Wed Feb 02 02:53:56 2000 
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Date: Wed, 02 Feb 2000 15:48:15 +0530
From: "T.V.Rami Reddy" <ramreddy@chiplogic.com>
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Subject: timstamps in rtp and rtcp?
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hi,
can anyone give the detailed difference between the rtp timestamp in the
rtp packet and timestamp in the rtcp SR packet.
suppose if an rtcp SR packet is send
rtcpSR------------ntp timestamp 2nd feb 2000  10:00:05   time stamp
1174(some value)
then let 4 rtp packets be send in the interval before an another rtcp SR
packet is send
rtp                1234
rtp                1394 (234+160 say)
rtp                1454 (1394+160 )
rtp                1514  (1454+160 )
rtcp SR   ------------ntp timestamp 2nd feb 2000 10:00:10   timestamp
will be  1234 or 1514 or ?


thanks,

--------------BB467BA2C8833963F0358A6E
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
hi,
<br>can anyone give the detailed difference between the rtp timestamp in
the<b> rtp packet</b> and timestamp in the<b> rtcp SR packet</b>.
<br>suppose if an rtcp SR packet is send
<br>rtcpSR------------ntp timestamp 2nd feb 2000&nbsp; 10:00:05&nbsp;&nbsp;
time stamp 1174(some value)
<br>then let 4 rtp packets be send in the interval before an another rtcp
SR packet is send
<br>rtp&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
1234
<br>rtp&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
1394 (234+160 say)
<br>rtp&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
1454 (1394+160 )
<br>rtp&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
1514&nbsp; (1454+160 )
<br>rtcp SR&nbsp;&nbsp; ------------ntp timestamp 2nd feb 2000 10:00:10&nbsp;&nbsp;
timestamp will be&nbsp; 1234 or 1514 or ?
<br>&nbsp;
<p>thanks,</html>

--------------BB467BA2C8833963F0358A6E--




From rem-conf Wed Feb 02 05:38:07 2000 
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To: "T.V.Rami Reddy" <ramreddy@chiplogic.com>
cc: rem-conf@es.net
Subject: Re: timstamps in rtp and rtcp?
In-Reply-To: Message from "T.V.Rami Reddy" <ramreddy@chiplogic.com> of "Wed, 02 Feb 2000 15:48:15 +0530." <38980466.2150E178@chiplogic.com>
Date: Wed, 02 Feb 2000 13:23:12 +0000
From: Colin Perkins <c.perkins@cs.ucl.ac.uk>
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--> "T.V.Rami Reddy" writes:
>can anyone give the detailed difference between the rtp timestamp in the
>rtp packet and timestamp in the rtcp SR packet.
>suppose if an rtcp SR packet is send
>rtcpSR------------ntp timestamp 2nd feb 2000  10:00:05   time stamp
>1174(some value)
>then let 4 rtp packets be send in the interval before an another rtcp SR
>packet is send
>rtp                1234
>rtp                1394 (234+160 say)
>rtp                1454 (1394+160 )
>rtp                1514  (1454+160 )
>rtcp SR   ------------ntp timestamp 2nd feb 2000 10:00:10   timestamp
>will be  1234 or 1514 or ?

The RTP timestamp in an SR packet represents the time at which that SR
packet is generated (section 6.4.1 of draft-ietf-avt-rtp-new-05.txt).
Both NTP and RTP clocks are free-running, and are sampled at the instant
the packet is generated.

For example:

	NTP time		RTP time	
2 Feb 2000 13:15:00.000		1000		RTP
2 Feb 2000 13:15:00.020		1160		RTP
2 Feb 2000 13:15:00.030		1240		RTCP SR
2 Feb 2000 13:15:00.040		1320		RTP
2 Feb 2000 13:15:00.060		1480		RTP
2 Feb 2000 13:15:00.080		1640		RTP
2 Feb 2000 13:15:00.100		1800		RTP
2 Feb 2000 13:15:00.120		1960		RTP
2 Feb 2000 13:15:00.140		2120		RTP
...				...		...

The SR packets provide a mapping between the RTP clock and the NTP clock,
so that RTP data packets don't need to include the NTP timestamp.

Colin



From rem-conf Wed Feb 02 10:29:35 2000 
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Reply-To: "Hari Krishna Vuppaladhadiam" <hariv@chiplogic.com>
From: "Hari Krishna Vuppaladhadiam" <hariv@chiplogic.com>
To: <rem-conf@es.net>
Subject: Audio Playout buffer
Date: Wed, 2 Feb 2000 10:09:28 -0800
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Hi,
   This is w.r.to Audio playout issue: I am going through various
implementation issues.

How much delay does the packet jitter Buffer contribute to the overall end
to end delay. Assume that a fixed buffer is used containing of N-packets.

Regards
Hari





From rem-conf Wed Feb 02 11:14:29 2000 
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From: Stephen Casner <casner@cisco.com>
To: "T.V.Rami Reddy" <ramreddy@chiplogic.com>
cc: rem-conf@es.net
Subject: Re: timstamps in rtp and rtcp?
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On Wed, 2 Feb 2000, T.V.Rami Reddy wrote:

> can anyone give the detailed difference between the rtp timestamp in the
> rtp packet and timestamp in the rtcp SR packet.
> suppose if an rtcp SR packet is send
> rtcpSR------------ntp timestamp 2nd feb 2000  10:00:05   time stamp
> 1174(some value)
> then let 4 rtp packets be send in the interval before an another rtcp SR
> packet is send
> rtp                1234
> rtp                1394 (234+160 say)
> rtp                1454 (1394+160 )
> rtp                1514  (1454+160 )
> rtcp SR   ------------ntp timestamp 2nd feb 2000 10:00:10   timestamp
> will be  1234 or 1514 or ?

Essentially the same question was asked on this list two weeks ago.
The RTP spec says:

        NTP timestamp: 64 bits
             Indicates the wallclock time (see Section 4) when this
             report was sent ...

        RTP timestamp: 32 bits
             Corresponds to the same time as the NTP timestamp (above),
             but in the same units and with the same random offset as
             the RTP timestamps in data packets. ...
     -->     Note that in most cases this timestamp
     -->     will not be equal to the RTP timestamp in any adjacent data
     -->     packet. Rather, it MUST be calculated from the
             corresponding NTP timestamp using the relationship between
             the RTP timestamp counter and real time as maintained by
             periodically checking the wallclock time at a sampling
             instant.

Please let me know whether the text above is unclear or hard to find
in the document so we can improve it for the next revision.

The method I recommend to calculate the RTP timestamp for the RTCP SR
packet is as follows.  Periodically, at the time a sample buffer
completes, establish a correspondence between the RTP timestamp that
will correspond to the first sample in that buffer and the time that
sample was taken according to the clock used for the NTP timestamps.
That time is usually the current time minus the duration of the buffer
and some allowance for interrupt or operating system latency.

The correspondence is an offset in RTP timestamp units:

  offset = (int32)(current_NTP_time * RTP_timestamp_clock_rate) - RTP_timestamp

where current_NTP_time is in seconds and fraction.  The calculation
can be done in floating point (double precision) or with
multiple-precision integer arithmetic.  In both cases, there will be a
truncation to get the 32-bit integer offset and RTP timestamp.

You don't need to do this calculation on every sample buffer, just
often enough to accommodate the drift between the sampling clock and
the clock used for the NTP timestamps if these come from different
oscillators (for example, a crystal on the audio board and a crystal
on the motherboard from which the operating system clock is derived).
If both clocks are derived from the same oscillator, you only need to
establish the correspondence once because there will be no drift, but
then the NTP times are probably not synchronized with Network Time
Protocol to absolute time.  Ideally, the capture interrupt routine
would return a timestamp for the capture time from which you could
form the NTP timestamp.

After you have calculated the offset, then when it is time to send an
RTCP SR packet, you get the current NTP time and calculate the RTP
timestamp for the SR packet as:

  RTP_timestamp = (int32)(current_NTP_time * RTP_timestamp_clock_rate) - offset

I think this is pretty easy.

Generating these timestamps accurately is critical for the receivers
to be able to use them for inter-media synchronization.
Unfortunately, if the server does a bad job of it, then it is the
receiver that looks bad because the presentation is out of sync.
(Of course, the receiver can also mess up the synchronization even
when it is given good timestamps.)
							-- Steve




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Subject: 2/9 User Experiences with the Communications Revolution --
  Charles H. House, Intel Dialogic Division 
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Berkeley Multimedia, Graphics and Interfaces Seminar

User Experiences with the Communications Revolution

Wednesday February 9, 2000,=20
1:10-2:30 p.m. PST
Fujitsu Seminar Room (405 Soda Hall)=20


Charles H. House
Intel Dialogic Division=20

After driving fifty miles down a long road in eastern China, and seeing s=
ix
vehicles in an hour, heading to a "town" of 4 million people with only
three "Western" hotels, our driver=92s beeper beeped. He pulled it out of=
 his
pocket, scanned the screen, and said, in perfect English, "ahh, Cisco, up=
 a
=BD point". Not as compelling as the wireless phone call from the top of =
Mt.
Everest to his wife in New Zealand as the stricken climber died, but
compelling enough for the van of travelers that the world of communicatio=
ns
has indeed changed.=20

What will be the impact of our new technologies - ever more complex and
cheaper chips, wireless and glass-based wideband networks, and ubiquitous
databases with XML cross-product "joins"? Everything from e-commerce to
"virtual meetings" to new forms of company, community, and governance wil=
l
happen. The CITS group (Center for Information Technology and Society) at
UCSB has been established to study these impacts from a wide perspective.
Our speaker, who had a key role in founding the Center, will describe som=
e
of the inquiry that will be done there, and some of the major observation=
s
of where this work might be expected to lead.

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

The seminar is broadcast live on the Internet Mbone and as a Real Network=
s
G2 broadcast.
You can connect to the live RealNetworks broadcast at:

http://media2.bmrc.berkeley.edu/bibs/schedule.cfm=20
You'll see a link to cs298-5/real networks/live programs. =20

You can also directly put in this url into the Real Player:=20
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 t=
he
announcement you may be able to receive the session by manually configuri=
ng
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=20




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Opera Portables, Inc. is presenting "by invitation" visits to our web site.  Newly 
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From: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
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Cc: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
Subject: New Internet Draft for MPEG-4 Audio/Visual RTP payload format
Date: Sat, 5 Feb 2000 17:01:40 +0900
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Dear all,

I have submitted a new version of the MPEG-4 Audio/Visual RTP Internet
Draft as follows.

--------

Title : RTP payload format for MPEG-4 Audio/Visual streams
Author(s) : Yoshihiro Kikuchi, Toshiyuki Nomura, Shigeru Fukunaga,
                 Yoshinori Matsui, Hideaki Kimata
Filename : draft-ietf-avt-rtp-mpeg4-es-00.txt
Pages : 14
Date : 01-Feb-2000

This document describes RTP payload formats for the carriage of MPEG-4
Audio and Visual streams, and an RTCP format for MPEG-4 upstream
messages functionalities. In this specification, MPEG-4 Audio/Visual
bitstreams are directly mapped into RTP packets without using MPEG-4
Systems. The RTP header fields usage and the fragmentation rule for
MPEG-4 Visual and Audio bitstreams are specified. It also specifies an
RTCP packet usage to carry the MPEG-4 upstream messages.

--------

Best regards,
Yoshihiro

----
        Yoshihiro Kikuchi

E-mail: kiku@eel.rdc.toshiba.co.jp
Corporate Research and Development Center
TOSHIBA Corporation
TEL: +81 +44 549 2288    FAX: +81 +44 520 1267


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Internet Engineering Task Force                 Yoshihiro Kikuchi - =
Toshiba
Internet Draft                                       Toshiyuki Nomura - =
NEC
Document: draft-ietf-avt-rtp-mpeg4-es-00.txt         Shigeru Fukunaga - =
Oki
                                              Yoshinori Matsui - =
Matsushita
                                                       Hideaki Kimata - =
NTT
                                                           February 1, =
2000


             RTP payload format for MPEG-4 Audio/Visual streams


Status of this Memo

   This document is an Internet-Draft and is in full conformance with =
all
      provisions of Section 10 of RFC2026 [1].

   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.






                                   Abstract

   This document describes RTP payload formats for the carriage of =
MPEG-4
   Audio and Visual streams[2][3], and an RTCP format for MPEG-4 =
upstream
   messages functionalities[4]. In this specification, MPEG-4 =
Audio/Visual
   bitstreams are directly mapped into RTP packets without using MPEG-4
   Systems[6]. The RTP header fields usage and the fragmentation rule =
for
   MPEG-4 Visual and Audio bitstreams are specified. It also specifies =
an
   RTCP packet usage to carry the MPEG-4 upstream messages.












Kikuchi/Nomura/Fukunaga/Kimata/Matsui                           [Page 1]
=0CRTP payload format for MPEG-4 Audio/Visual streams       February =
2000=0D


1. Introduction

1.1 Why MPEG-4 Audio/Visual RTP format needed?

   The RTP payload formats described in this Internet-Draft specify the
   normative way on how MPEG-4 Audio/Visual streams are fragmented and
   mapped directly onto RTP packets. No extra header field is used for =
such
   functionality as error protection or grouping of streams.

   H.323 terminals could be the case.  MPEG-4 Audio/Visual streams are =
not
   managed by Object Descriptors[6] but H.245, and directly mapped into =
RTP
   packets without Sync Layer[6]. The semantics of RTP headers in this =
case
   need to be clearly defined including the association with the MPEG-4
   Audio/Visual data elements.  In addition, it would be beneficial to
   define the fragmentation rule of RTP packets for MPEG-4 Video streams =
to
   enhance error resiliency by utilizing the error resilience tools =
provided
   inside the MPEG-4 Video stream.  However, they have not been studied
   until now.


1.2 Consideration on the MPEG-4 Visual RTP payload format

   MPEG-4 Visual is a visual coding standard with many new =
functionalities:
   high coding efficiency, high error resiliency, multiple arbitrary =
shaped
   object based coding, etc. [2]. It covers a wide range of bitrate from
   several Kbps to many Mbps. It also covers a wide variety of networks =
from
   guarantied with almost error-free to mobile with high error rate by =
its
   error resilience functionalities.

   A normative way of fragmentation of an MPEG-4 visual bitstream into =
RTP
   packets is defined in this Internet draft. Since MPEG-4 Visual is =
used
   for a wide variety of networks, it is not desired to apply too much
   restriction on the fragmentation like a single video packet shall =
always
   be mapped on a single RTP packet. On the other hand, a careless media
   unaware fragmentation may cause degradation of the error resiliency =
and
   the bandwidth efficiency. The fragmentation rule described in this
   Internet draft is flexible but to define the minimum rules to prevent =
the
   meaningless fragmentation of e.g. splitting a header into packets.
   For video coding media such as H.261 or MPEG-1/2, the additional =
media
   specific RTP header works effectively for recovering e.g. a picture
   header corrupt by packet losses. However, there are error resilience
   functionalities inside MPEG-4 Visual to recover corrupt headers. =
These
   functionalities can commonly be used on RTP/IP network as well as =
other
   networks. (H.223/mobile, MPEG-2/TS, etc.) Therefore, no extra RTP =
header
   fields are defined in the MPEG-4 Visual RTP payload format.


1.3 Consideration on the MPEG-4 Audio RTP payload format

   MPEG-4 Audio is a new kind of audio standard that integrates many
   different types of audio coding tools. It also supports a mechanism
   representing synthesized sounds. Low-overhead MPEG-4 Audio Transport

Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 2]
=0CRTP payload format for MPEG-4 Audio/Visual streams       February =
2000=0D


   Multiplex (LATM) manages the sequence of the compressed or the
   represented audio data by MPEG-4 Audio tools with relatively small
   overhead. In audio-only applications, the LATM-based MPEG-4 Audio
   bitstreams, therefore, are desirable to be directly mapped into the =
RTP
   packets without using MPEG-4 Systems.

   Furthermore, if the payload of a packet is a single audio frame, a =
packet
   loss does not impair the decodability of adjacent packets. Therefore, =
a
   payload specific header for MPEG-4 Audio is not required as same as =
one
   for the other audio coders.


1.4 MPEG-4 Audio/Visual upstream messaging on RTCP packets

   In some cases, MPEG-4 Audio/Visual has upstream messaging
   functionalities. These messages are extremely Audio/Visual specific,
   since coders directly use these messages for controlling coding
   parameters. From the point of view of controlling parameters, these
   messages should be transmitted without delay. Therefore these =
messages
   are directly mapped onto some kind of low delay RTCP packets.




2. Conventions used in this document

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED",  "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC-2119 [7].




3. RTP Packetization of MPEG-4 Visual bitstream

   This section specifies the RTP packetization rule for MPEG-4 Visual
   content. An MPEG-4 Visual bitstream is mapped directly onto the RTP
   payload without any addition of extra header fields or removal of any
   Visual syntax elements. The Combined Configuration/Elementary streams
   mode is used so that the configuration information is carried in the =
same
   RTP port as the elementary stream. (see 6.2.1 "Start codes" of =
ISO/IEC
   14496-2 [2][9][4])

   When the short video header mode is used, RTP payload format for =
H.263
   specified in the relevant RFCs or other standards MAY be used.








Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 3]
=0CRTP payload format for MPEG-4 Audio/Visual streams       February =
2000=0D


   0                   1                   2                   3
   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |V=3D2|P|X|  CC   |M|     PT      |       sequence number         | =
RTP
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                           timestamp                           | =
Header
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |           synchronization source (SSRC) identifier            |
   =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+
   |            contributing source (CSRC) identifiers             |
   |                             ....                              |
   =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+
   |                                                               | RTP
   |       MPEG-4 Visual stream (byte aligned)                     | =
Payload
   |                                                               |
   |                               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                               :...OPTIONAL RTP padding        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

        Figure 1 - An RTP packet for MPEG-4 Visual stream




3.1 RTP header fields usage for MPEG-4 Visual

   Payload Type (PT): Distinct payload type should be assigned to =
specify
   MPEG-4 Visual RTP payload format. If the dynamic payload type =
assignment
   is used, it is specified by some out-of-band means (e.g. H.245, SDP,
   etc.) that the MPEG-4 Visual payload format is used for the =
corresponding
   RTP packet.


   Extension (X) bit: Defined by the RTP profile used.


   Sequence Number: Increment by one for each RTP data packet sent. It
   starts with a random initial value for security reasons.


   Marker (M) bit: The marker bit is set to one to indicate the last RTP
   packet (or only RTP packet) of a VOP.


   Timestamp: The timestamp indicates the composition time, or the
   presentation time in a no-compositor decoder by adding a constant =
random
   offset for security reasons. For a video object plane, it is defined =
by
   vop_time_increment (in units of 1/vop_time_increment_resolution =
seconds)
   plus the cumulative number of whole seconds specified by =
module_time_base
   and time_code of Group_of_VideoObjectPlane() if present. In the case =
of
   interlaced video, a VOP consists of lines from two fields and the
   timestamp indicates the composition time of the first field. If the =
RTP

Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 4]
=0CRTP payload format for MPEG-4 Audio/Visual streams       February =
2000=0D


   packet contains only configuration information and/or
   Group_of_VideoObjectPlane(), the composition time of the subsequent =
VOP
   in the coding order is used. If the RTP packet contains only
   visual_object_sequence_end_code, the composition time of the =
immediately
   preceding VOP in the coding order is used.

   Unless specified by an out-of-band means, the resolution of the =
timestamp
   is set to its default (90KHz).


   SSRC, CC and CSRC fields are used as described in RFC 1889 [8].


3.2 Fragmentation of MPEG-4 Visual bitstream

   A fragmented MPEG-4 Visual bitstream is mapped directly onto the RTP
   payload without any addition of extra header fields or removal of any
   Visual syntax elements. The Combined Configuration/Elementary streams
   mode is used. The following rules apply for the fragmentation.

   (1) The configuration information and Group_of_VideoObjectPlane() =
SHALL
   be placed at the beginning of the RTP payload (just after the RTP =
header)
   or just after the header of the syntactically upper layer function.

   (2) If one or more headers exist in the RTP payload, the RTP payload
   SHALL begin with the header of the syntactically highest function.
   Note: The visual_object_sequence_end_code is regarded as the lowest
   function.

   (3) A header SHALL NOT be split into a plurality of RTP packets.

   (4) Two or more VOPs SHALL be fragmented into different RTP packets =
so
   that one RTP packet consists of the data bytes associated with an =
unique
   presentation time (that indicated to the timestamp field in the RTP
   packet header).

   (5) A single video packet SHOULD NOT be split into a plurality of RTP
   packets. The size of a video packet SHOULD be adjusted such that the
   resulting RTP packet is not larger than the path-MTU.


   Hear, header means:
   - Configuration information (Visual Object Sequence Header, Visual =
Object
     Header and Visual Object Layer Header)
   - visual_object_sequence_end_code
   - The header of the entry point function for an elementary stream
     (Group_of_VideoObjectPlane() or the header of VideoObjectPlane(),
     video_plane_with_short_header(), MeshObject() or FaceObject())
   - The video packet header (video_packet_header() excluding
     next_resync_marker())
   - The header of gob_layer()


Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 5]
=0CRTP payload format for MPEG-4 Audio/Visual streams       February =
2000=0D


   See 6.2.1 "Start codes" of ISO/IEC 14496-2[2][9][4] for the =
definition of
   the configuration information and the entry point functions.


   The video packet starts with the VOP header or the video packet =
header,
   followed by motion_shape_texture(), and ends with =
next_resync_marker() or
   next_start_code).


3.3 Examples of packetized MPEG-4 Visual bitstream

   Considering that MPEG-4 Visual is used on a wide variety of networks =
from
   several Kbps to many Mbps, from guarantied networks with almost =
error-
   free to mobile networks with high error rate, it is not desired to =
apply
   too much restriction on the fragmentation like a single video packet
   shall always be mapped on a single RTP packet. On the other hand, a
   careless media unaware fragmentation will cause degradation of the =
error
   resiliency and the bandwidth efficiency. The fragmentation criteria
   described in 3.2 are flexible but to define the minimum rules to =
prevent
   the meaningless fragmentation of e.g. splitting a header into =
packets.

   For video coding media such as H.261 or MPEG-1/2, the additional =
media
   specific RTP header works effectively for recovering e.g. a picture
   header corrupt by packet losses. However, there is an error =
resilience
   functionality inside MPEG-4 Visual to recover corrupt headers. This
   functionality can commonly be used on RTP/IP network as well as other
   networks. (H.223/mobile, MPEG-2/TS, etc.) Therefore, there is no =
strong
   reason to define MPEG-4 Visual specific extra RTP header fields.


   Figure 2 shows examples of RTP packets generated based on the =
criteria
   described in 3.2

   (a) is an example of the first RTP packet or the random access point =
of
   an MPEG-4 visual bitstream. This RTP packet contains the =
configuration
   information. According to the criterion (1), the Visual Object =
Sequence
   Header(VS header) is placed at the beginning of the RTP payload, and =
the
   Visual Object Header and the Visual Object Layer Header(VO header, =
VOL
   header) follow it. Since the fragmentation rule defined in 3.2 =
guaranties
   that the configuration information, starting with
   visual_object_sequence_start_code, is always placed at the beginning =
of
   the RTP payload, RTP receivers can detect the random access point by
   checking if the first 32-bit field of the RTP payload is
   visual_object_sequence_start_code.

   (b) is an example the RTP packet that contains
   Group_of_VideoObjectPlane(GOV). Following the criterion (1), the GOV =
is
   placed at the beginning of the RTP payload. It is a waste of RTP/IP
   header overhead to generate a RTP packet containing only a GOV whose
   length is 7 bytes. Therefore, (a part of) the following VOP can be =
placed
   in the same RTP packet as shown in (b).


Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 6]
=0CRTP payload format for MPEG-4 Audio/Visual streams       February =
2000=0D


   (c) is an example that one video packet is packetized into one RTP
   packet. When the packet-loss rate of the underlying network is high, =
this
   kind of packetization is recommended. It is strongly recommended to =
set
   resync_marker_disable to 0 in the VOL header to enable adjustment of =
the
   video packet size. Even when the RTP packet containing the VOP header =
is
   discarded by a packet loss, the other RTP packets can be decoded by =
using
   the HEC(Header Extension Code) information in the video packet =
header. No
   extra RTP header field is necessary.

   (d) is an example that more than one video packets are packetized =
into
   one RTP packet. This kind of packetization is effective to save the
   overhead of RTP/IP headers if the bit-rate of the underlying network =
is
   low. However, it will decrease the packet-loss resiliency because
   multiple video packets are discarded by a single RTP packet loss. The
   adequate number of video packets in a RTP packet and the RTP packet
   length depend the packet-loss rate and the bit-rate of the underlying
   network.


   Figure 3 shows examples of RTP packets prohibited by the criteria of =
3.2.

   Fragmentation of a header into multiple RTP packets, like (a), will =
not
   only increase the overhead of RTP/IP headers but also decrease the =
error
   resiliency. Therefore, it is prohibited by the criterion (3).

   When concatenating more than one video packets into a RTP packet, VOP
   header or video_packet_header() shall not be placed in the middle of =
the
   RTP payload. The packetization like (b) is not allowed by the =
criterion
   (2). This is because of the error resiliency. Comparing this example =
with
   Figure 2(c), two video packets are mapped onto two RTP packets in =
both
   cases. However, there is a difference between the packet-loss =
resiliency.
   When the second RTP packet is lost, both video packets 1 and 2 are =
lost
   in the case of Figure 3(b) whereas only video packet 2 is lost in the
   case of Figure 2(c).

   A RTP packet containing more than one VOPs, like (c), is not allowed.

















Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 7]
=0CRTP payload format for MPEG-4 Audio/Visual streams       February =
2000=0D


       +------+------+------+------+
   (a) | RTP  |  VS  |  VO  | VOL  |
       |header|header|header|header|
       +------+------+------+------+

       +------+-----+------------------+
   (b) | RTP  | GOV |Video Object Plane|
       |header|     |                  |
       +------+-----+------------------+

       +------+------+------------+  +------+------+------------+
   (c) | RTP  | VOP  |Video Packet|  | RTP  |  VP  |Video Packet|
       |header|header|    (1)     |  |header|header|    (2)     |
       +------+------+------------+  +------+------+------------+

       =
+------+------+------------+------+------------+------+------------+
   (d) | RTP  |  VP  |Video Packet|  VP  |Video Packet|  VP  |Video =
Packet|
       |header|header|     (1)    |header|    (2)     |header|    (3)    =
 |
       =
+------+------+------------+------+------------+------+------------+

        Figure 2 - Examples of RTP packetized MPEG-4 Visual bitstream


       +------+-------------+  +------+------------+------------+
   (a) | RTP  |First half of|  | RTP  |Last half of|Video Packet|
       |header|  VP header  |  |header|  VP header |            |
       +------+-------------+  +------+------------+------------+

       +------+------+----------+  =
+------+---------+------+------------+
   (b) | RTP  | VOP  |First half|  | RTP  |Last half|  VP  |Video =
Packet|
       |header|header| of VP(1) |  |header| of VP(1)|header|    (2)     =
|
       +------+------+----------+  =
+------+---------+------+------------+

       +------+------+------------------+------+------------------+
   (c) | RTP  | VOP  |Video Object Plane| VOP  |Video Object Plane|
       |header|header|        (1)       |header|       (2)        |
       +------+------+------------------+------+------------------+

   Figure 3 - Examples of prohibited RTP packetization for MPEG-4 Visual
   bitstream













Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 8]
=0CRTP payload format for MPEG-4 Audio/Visual streams       February =
2000=0D


4. RTP Packetization of MPEG-4 Audio bitstream

   When tools defined in MPEG-4 Systems are not used MPEG-4 Audio stream =
is
   formatted by LATM (Low-overhead MPEG-4 Audio Transport Multiplex)
   format[5], and then mapped onto RTP packets as described the =
subsequent
   section.

4.1 RTP Packet Format

   The LATM consists of the sequence of audioMuxElements that include =
one or
   more audio frames. A complete audioMuxElement or the part of
   audioMuxElements SHALL be mapped directly onto the RTP payload =
without
   removal of any audioMuxElement syntax elements as shown in Figure 4. =
The
   first byte of each audioMuxElement SHALL be located at the first =
payload
   location of an RTP packet.


   0                   1                   2                   3
   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |V=3D2|P|X|  CC   |M|     PT      |       sequence number         =
|RTP
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                           timestamp                           =
|Header
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |           synchronization source (SSRC) identifier            |
   =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+
   |            contributing source (CSRC) identifiers             |
   |                             ....                              |
   =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+
   |                                                               |RTP
   :                 audioMuxElement (byte aligned)                =
:Payload
   |                                                               |
   |                               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                               :...OPTIONAL RTP padding        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
                Figure 4 - An RTP packet for MPEG-4 Audio

   It is required for the audioMuxElement to indicate the following
   muxConfigPresent information by an out-of-band means.

   muxConfigPresent: If this information is set to 1, the =
audioMuxElement
   SHALL include an indication bit "useSameStreamMux" and MAY include =
the
   configuration information for audio compression "StreamMuxConfig". =
The
   useSameStreamMux bit indicates whether the StreamMuxConfig element in =
the
   previous frame is applied in the current frame.

4.2 RTP Header Fields Usage

   Payload Type (PT): Distinct payload type should be assigned to =
specify
   MPEG-4 Audio RTP payload format. If the dynamic payload type =
assignment
   is used, it is specified by some out-of-band means (e.g. H.245, SDP,


Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 9]
=0CRTP payload format for MPEG-4 Audio/Visual streams       February =
2000=0D


   etc.) that the MPEG-4 Audio payload format is used for the =
corresponding
   RTP packet.

   Marker (M) bit: The marker bit indicates audioMuxElement boundaries. =
This
   bit is set to one to mark the RTP packet contains a complete
   audioMuxElement or the last fragment of an audioMuxElement.

   Timestamp: The timestamp indicates the composition time, or the
   presentation time in a no-compositor decoder. Timestamps are =
recommended
   to start at a random value for security reasons.

   Unless specified by an out-of-band means, the resolution of the =
timestamp
   is set to its default (90 kHz).

   Sequence Number: Increment by one for each RTP packet sent. It starts
   with a random value for security reasons.

   SSRC, CC and CSRC fields are used as described in RFC 1889 [8].

4.3 Fragmentation of MPEG-4 Audio bitstream

   It is desirable to put one audioMuxElement per RTP packet. The size =
of an
   audioMuxElement is tried to be adjusted such that the resulting RTP
   packet is not larger than the path-MTU. If this is not possible, the
   audioMuxElement MAY be fragmented across several packets based on the
   following rules.

   (1) "payloadMux" which consists of payload elements MAY be fragmented
   into several RTP packets so that one RTP packet consists of one or =
more
   payload elements. A payload element SHOULD NOT be fragmented.

   (2) If the audioMuxElement includes StreamMuxConfig, StreamMuxConfig
   SHALL be included into the RTP packet containing the first payload
   element.




5. RTCP Packetization of MPEG-4 upstream messages

   This section specifies the usage of particular RTCP packets to carry =
the
   upstream messages generated using the MPEG-4 Audio/Visual upstream
   messaging functionalities, e.g. NEWPRED[4]. RTCP packets specified in
   this section SHALL ONLY be used when it is indicated by the profile =
and
   level indication of MPEG-4 the codecs have such functionalities. =
(e.g.
   Advanced Real Time Simple Visual Profile[4])


   The MPEG-4 upstream messages are transmitted on particular RTCP =
packets,
   like H.261 RTCP control packets [10].



Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 10]
=0CRTP payload format for MPEG-4 Audio/Visual streams       February =
2000=0D


   In the case that the RTP session uses a multicast address, the MPEG-4
   upstream message packets are not transmitted to the normal RTCP
   destination transport address. Instead, these upstream message =
packets
   are sent directly via unicast from the decoder to the coder.  The
   destination port number of these upstream message packets is always =
same
   to the port number of the normal RTCP address.

   As a consequence, these upstream message packets may only be used =
when no
   RTP mixers or translators intervene in the path from the coder to the
   decoder.  If such intermediate systems do intervene, the address of =
the
   coder would no longer be present as the network-level source address =
in
   packets received by the decoder, and in fact, it might not be =
possible
   for the decoder to send packets directly to the coder.

   Some reliable multicast protocols use similar NACK control packets
   transmitted over the normal multicast distribution channel, however =
they
   typically use random delays to prevent a NACK implosion problem. The =
goal
   of such protocols is to provide reliable multicast packet delivery at =
the
   expense of delay, which is appropriate for applications such as a =
shared
   whiteboard.

   On the other hand, real-time Audio/Visual transmission is more =
sensitive
   to delay and does not require full reliability.  For Audio/Visual
   applications it is more effective to send the MPEG-4 upstream message
   packets as soon as possible, i.e. as soon as a loss is detected, =
without
   adding any random delays.

5.1.  MPEG-4 Visual upstream message packets definition

        0                   1                   2                   3
        0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       |V=3D2|P|   UMT   | PT=3DRTCP_MP4UM |           length            =
  |
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       |                              SSRC                             |
       =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+
       |       MPEG-4 Upstream Messages Payload (byte aligned)         |
       |                               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       |                               :             padding           |
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   version (V): 2 bits
        Identifies the version of RTP, which is the same in RTCP packets =
as
       in RTP data packets.

   padding (P): 1 bit
       If the padding bit is set, this RTCP packet contains some =
additional
       padding octets at the end which are not part of the control
       information. The last octet of the padding is a count of how many
       padding octets should be ignored. In the case several upstream
       messages are mapped onto one RTCP packet, padding should only be
       required on the last individual message.

Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 11]
=0CRTP payload format for MPEG-4 Audio/Visual streams       February =
2000=0D



   upstream message type (UMT): 5 bits
       Identifies the type of the MPEG-4 upstream messages.
       0:       forbidden
       1:       MPEG-4 Visual NEWPRED
       2-63:    reserved
       In this internet-draft, only NEWPRED is assigned as the candidate =
of
       the UMT for the moment.  Some other MPEG-4 Audio/Visual =
applications
       using the upstream messages may be assigned in the future.

   packet type (PT): 8 bits
       The value of the packet type (PT) identifier is the constant
       RTCP_MP4UM (TBD).

   SSRC: 32 bits
       SSRC is the synchronization source identifier for the sender of =
this
       packet.

   MPEG-4 Upstream Message Payload: variable
        The syntax and semantics of the MPEG-4 upstream messages are =
defined
        in the ISO/IEC 14496-2/3[4][5]. All messages are byte aligned.
        Normally one message is mapped onto one RTCP packet, and several
        messages with same UMT could be continuously mapped onto one =
RTCP
        packet.  One message SHALL NOT be fragmented into different RTCP
        packets.





6. Security Considerations

   RTP packets using the payload format defined in this specification =
are
   subject to the security considerations discussed in the RTP =
specification
   [8]. This implies that confidentiality of the media streams is =
achieved
   by encryption. Because the data compression used with this payload =
format
   is applied end-to-end, encryption may be performed on the compressed =
data
   so there is no conflict between the two operations.

   This payload type does not exhibit any significant non-uniformity in =
the
   receiver side computational complexity for packet processing  to =
cause a
   potential denial-of-service threat.


7. References


   1  Bradner, S., "The Internet Standards Process -- Revision 3", BCP =
9,
      RFC 2026, October 1996.




Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 12]
=0CRTP payload format for MPEG-4 Audio/Visual streams       February =
2000=0D



   2 ISO/IEC 14496-2:1999, "Information technology - Coding of =
audio-visual
      objects - Part2: Visual", December 1999.

   3 ISO/IEC 14496-3:1999, "Information technology - Coding of =
audio-visual
      objects - Part3: Audio", December 1999.

   4 ISO/IEC 14496-2:1999/FDAM1:2000, December 1999.

   5 ISO/IEC 14496-3:1999/FDAM1:2000, December 1999.

   6 ISO/IEC 14496-1:1999, "Information technology - Coding of =
audio-visual
      objects - Part1: Systems", December 1999.

   7  Bradner, S., "Key words for use in RFCs to Indicate Requirement
      Levels", BCP 14, RFC 2119, March 1997

   8 H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson "RTP: A =
Transport
      Protocol for Real Time Applications",  RFC 1889, Internet =
Engineering
      Task Force, January 1996.

   9  ISO/IEC 14496-2/DCOR1, October 1999.

   10 T. Turletti, C. Hitema, "RTP Payload Format for H.261 Video =
Streams",
      RFC 2032, Octover 1996.






8. Author's Addresses


   Yoshihiro Kikuchi
   Toshiba corporation
   1, Komukai Toshiba-cho, Saiwai-ku, Kawasaki, 212-8582, Japan
   Email: kiku@eel.rdc.toshiba.co.jp

   Yoshinori Matsui
   Matsushita Electric Industrial Co., LTD.
   1006, Kadoma, Kadoma-shi, Osaka, Japan
   Email: matsui@drl.mei.co.jp

   Toshiyuki Nomura
   NEC Corporation
   4-1-1,Miyazaki,Miyamae-ku,Kawasaki,JAPAN
   Email: t-nomura@ccm.cl.nec.co.jp

   Shigeru Fukunaga
   Oki Electric Industry Co., Ltd.
   1-2-27 Shiromi, Chuo-ku, Osaka 540-6025 Japan.

Kikuchi/Nomura/Fukunaga/Kimata/Matsui                        [Page 13]
=0CRTP payload format for MPEG-4 Audio/Visual streams       February =
2000=0D


   Email: fukunaga444@oki.co.jp

   Hideaki Kimata
   Nippon Telegraph and Telephone Corporation
   1-1, Hikari-no-oka, Yokosuka-shi, Kanagawa, Japan
   Email: kimata@nttvdt.hil.ntt.co.jp
=0CRTP payload format for MPEG-4 Audio/Visual streams       February =
2000=0D



Full Copyright Statement

   "Copyright (C) The Internet Society (date). All Rights Reserved.
   This document and translations of it may be copied and furnished to
   others, and derivative works that comment on or otherwise explain it
   or assist in its implmentation may be prepared, copied, published
   and distributed, in whole or in part, without restriction of any
   kind, provided that the above copyright notice and this paragraph
   are included on all such copies and derivative works. However, this
   document itself may not be modified in any way, such as by removing
   the copyright notice or references to the Internet Society or other
   Internet organizations, except as needed for the purpose of
   developing Internet standards in which case the procedures for
   copyrights defined in the Internet Standards process must be
   followed, or as required to translate it into
=0C
------=_NextPart_000_01CF_01BF6FFA.AC3C3FC0--




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--NextPart

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

	Title		: RTP Payload for Text Conversation
	Author(s)	: G. Hellstrom
	Filename	: draft-ietf-avt-rtp-text-04.txt
	Pages		: 9
	Date		: 07-Feb-00
	
This memo describes how to carry text conversation session contents
in RTP packets. Text conversation session contents are specified in
ITU-T Recommendation T.140 [1]. 
Text conversation is used alone or in connection to other
conversational facilities such as video and voice, to form multimedia
conversation services.
This RTP payload description contains an optional possibility to
include redundant text from already transmitted packets in order to
reduce the risk of text loss caused by packet loss. The redundancy
coding follows RFC 2198.

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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 format for MPEG-4 Audio/Visual streams
	Author(s)	: Y. Kikuchi, T. Nomura, S. Fukunaga, Y. Matsui, 
                          H. Matsui
	Filename	: draft-ietf-avt-rtp-mpeg4-es-00.txt
	Pages		: 13
	Date		: 07-Feb-00
	
This document describes RTP payload formats for the carriage of MPEG-4
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messages functionalities[4]. In this specification, MPEG-4 Audio/Visual
bitstreams are directly mapped into RTP packets without using MPEG-4
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From rem-conf Tue Feb 08 11:00:37 2000 
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Subject: 2/9 User Experiences with the Communications Revolution --
  Charles H. House, Intel Dialogic Division 
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Berkeley Multimedia, Graphics and Interfaces Seminar

User Experiences with the Communications Revolution

Wednesday February 9, 2000,=20
1:10-2:30 p.m. PST
Fujitsu Seminar Room (405 Soda Hall)=20


Charles H. House
Intel Dialogic Division=20

After driving fifty miles down a long road in eastern China, and seeing s=
ix
vehicles in an hour, heading to a "town" of 4 million people with only
three "Western" hotels, our driver=92s beeper beeped. He pulled it out of=
 his
pocket, scanned the screen, and said, in perfect English, "ahh, Cisco, up=
 a
=BD point". Not as compelling as the wireless phone call from the top of =
Mt.
Everest to his wife in New Zealand as the stricken climber died, but
compelling enough for the van of travelers that the world of communicatio=
ns
has indeed changed.=20

What will be the impact of our new technologies - ever more complex and
cheaper chips, wireless and glass-based wideband networks, and ubiquitous
databases with XML cross-product "joins"? Everything from e-commerce to
"virtual meetings" to new forms of company, community, and governance wil=
l
happen. The CITS group (Center for Information Technology and Society) at
UCSB has been established to study these impacts from a wide perspective.
Our speaker, who had a key role in founding the Center, will describe som=
e
of the inquiry that will be done there, and some of the major observation=
s
of where this work might be expected to lead.

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

The seminar is broadcast live on the Internet Mbone and as a Real Network=
s
G2 broadcast.
You can connect to the live RealNetworks broadcast at:

http://media2.bmrc.berkeley.edu/bibs/schedule.cfm=20
You'll see a link to cs298-5/real networks/live programs. =20

You can also directly put in this url into the Real Player:=20
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 t=
he
announcement you may be able to receive the session by manually configuri=
ng
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=20




From rem-conf Tue Feb 08 12:15:02 2000 
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Date: Tue, 8 Feb 2000 12:01:27 -0800
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Subject: Audio playout issues


Hi,

Can anyone guide me on Audio Playout algorithms.

How much delay does the packet jitter Buffer contribute to the overall end
to end delay? Assume that a fixed buffer is used containing of N-packets. 

Are the Adaptive playout schemes really implemented for VoIP?

If so what kind of Speech codecs does it apply?

Regards
Hari






From rem-conf Wed Feb 09 08:49:03 2000 
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From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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Subject: Re: Question about media desinations
References: <20000208140607.A17103@div8.net>
		<000301bf727b$358f7080$ad9423a6@mcit.com> <14497.36973.637899.472650@thomasm-u1.cisco.com>
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Michael Thomas wrote:
> 
> The packetcable security spec specifies a 2 or 4
> byte MAC (using MMH) to be appended to the RTP

Is this simply hash(RTP packet | shared secret)?

Could the bad guy record a conversation, including the hash, and then
mix in the old data with the "real" one?

> packet itself. I have argued that RFC 2198 might
> actually be a cleaner way to append the MAC
> itself, but so long as the parties agree to the
> payload format, a relatively weak MAC should make
> barge in sufficiently difficult that even if an
> attacker managed to get through every once in a
> while, they probably wouldn't do anything
> especially exciting.

Since this is an RTP issue, I wonder if this shouldn't be written up for
avt. Another possibility is an RTP header extension. That, at least,
would be backward compatible with non-MMH implementations.

> 

>  >
>  > >   So, what's stopping a haxor like me from strobbing a gateway with random
>  > > audio... ??

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



From rem-conf Wed Feb 09 09:24:57 2000 
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Henning Schulzrinne writes:
 > Michael Thomas wrote:
 > > 
 > > The packetcable security spec specifies a 2 or 4
 > > byte MAC (using MMH) to be appended to the RTP
 > 
 > Is this simply hash(RTP packet | shared secret)?

Yes. In packetcable, the payload is encrypted with
RC4, but the MAC is applied to the entire RTP
packet.
 
 > Could the bad guy record a conversation, including the hash, and then
 > mix in the old data with the "real" one?

Not if you use the RTP timestamp. Built in
replay protection :-)

 > > packet itself. I have argued that RFC 2198 might
 > > actually be a cleaner way to append the MAC
 > > itself, but so long as the parties agree to the
 > > payload format, a relatively weak MAC should make
 > > barge in sufficiently difficult that even if an
 > > attacker managed to get through every once in a
 > > while, they probably wouldn't do anything
 > > especially exciting.
 > 
 > Since this is an RTP issue, I wonder if this shouldn't be written up for
 > avt. Another possibility is an RTP header extension. That, at least,
 > would be backward compatible with non-MMH implementations.

Whatever way has the least number of bytes is
probably the right answer. Packetcable's only
incurs the cost of the MAC itself since there is
an agreement between the sender and receiver that
there be the MAC tacked on to the end. Since it's
a trailer, it ought not effect a naive
implementation (though it would certainly violate
the "conservative in what you send" principle).

I do sort of like the idea of the MAC being a
payload not entirely unlike a codec payload,
though I'm not wedded to the notion. It does have
the advantage that we can offer up the MAC "codec"
up in SDP m= as being something that the endpoint
is willing to receive. With RFC 2198, the sender
would then oblige by sending several different
"codecs", namely the real codec and the MAC at the
same time.

	       Mike



From rem-conf Wed Feb 09 09:43:36 2000 
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Michael Thomas wrote:
> 

>  > Since this is an RTP issue, I wonder if this shouldn't be written up for
>  > avt. Another possibility is an RTP header extension. That, at least,
>  > would be backward compatible with non-MMH implementations.
> 
> Whatever way has the least number of bytes is
> probably the right answer. Packetcable's only
> incurs the cost of the MAC itself since there is
> an agreement between the sender and receiver that
> there be the MAC tacked on to the end. Since it's
> a trailer, it ought not effect a naive
> implementation (though it would certainly violate
> the "conservative in what you send" principle).

Since RTP doesn't have a length field, a "naive" implementation would
treat the MAC bytes as G.711 samples. Not good.

> 
> I do sort of like the idea of the MAC being a
> payload not entirely unlike a codec payload,
> though I'm not wedded to the notion. It does have
> the advantage that we can offer up the MAC "codec"
> up in SDP m= as being something that the endpoint
> is willing to receive. With RFC 2198, the sender
> would then oblige by sending several different
> "codecs", namely the real codec and the MAC at the
> same time.

My major concern with this is that there's a fair amount of overhead
just to get 2 or 4 bytes across.

> 
>                Mike

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



From rem-conf Wed Feb 09 10:06:23 2000 
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To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
cc: Michael Thomas <mat@cisco.com>, 
    'SIP List' <sip@lists.research.bell-labs.com>, rem-conf@es.net
Subject: Re: Question about media desinations
In-reply-to: Your message of "Wed, 09 Feb 2000 12:27:20 EST." <38A1A378.469B6FE1@cs.columbia.edu>
Date: Wed, 09 Feb 2000 18:03:27 +0000
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--> Henning Schulzrinne writes:
>Michael Thomas wrote:
>> I do sort of like the idea of the MAC being a
>> payload not entirely unlike a codec payload,
>> though I'm not wedded to the notion. It does have
>> the advantage that we can offer up the MAC "codec"
>> up in SDP m= as being something that the endpoint
>> is willing to receive. With RFC 2198, the sender
>> would then oblige by sending several different
>> "codecs", namely the real codec and the MAC at the
>> same time.
>
>My major concern with this is that there's a fair amount of overhead
>just to get 2 or 4 bytes across.

It's one more byte than using the RTP header extension.....

Still, it may be better to define a new RTP profile which includes the 
MAC as part of the fixed RTP header. That's probably the solution with 
the least overhead, and the protocol definition can be straightforward
RFC1890 + 4 bytes of MAC header.

Colin



From rem-conf Wed Feb 09 11:10:08 2000 
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From: Michael Thomas <mat@cisco.com>
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To: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
Cc: Henning Schulzrinne <schulzrinne@cs.columbia.edu>,
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        "'SIP List'" <sip@lists.research.bell-labs.com>, rem-conf@es.net
Subject: Re: Question about media desinations
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Colin Perkins writes:
 > --> Henning Schulzrinne writes:
 > >Michael Thomas wrote:
 > >> I do sort of like the idea of the MAC being a
 > >> payload not entirely unlike a codec payload,
 > >> though I'm not wedded to the notion. It does have
 > >> the advantage that we can offer up the MAC "codec"
 > >> up in SDP m= as being something that the endpoint
 > >> is willing to receive. With RFC 2198, the sender
 > >> would then oblige by sending several different
 > >> "codecs", namely the real codec and the MAC at the
 > >> same time.
 > >
 > >My major concern with this is that there's a fair amount of overhead
 > >just to get 2 or 4 bytes across.
 > 
 > It's one more byte than using the RTP header extension.....
 > 
 > Still, it may be better to define a new RTP profile which includes the 
 > MAC as part of the fixed RTP header. That's probably the solution with 
 > the least overhead, and the protocol definition can be straightforward
 > RFC1890 + 4 bytes of MAC header.

Would that not lead to a combinatorial explosion
of name space with rtpmap in SDP? Ie, G711-MMH2,
etc?  I'm all for having zero overhead, but not at
the expense of making a mess of SDP. That is sort
of why I was attracted to RFC 2198. If somebody
else has a better way of doing the same thing
without requiring, for example, structured names
in SDP that would be great too.
  
	Mike



From rem-conf Wed Feb 09 11:29:17 2000 
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Michael Thomas wrote:
> 

> Would that not lead to a combinatorial explosion
> of name space with rtpmap in SDP? Ie, G711-MMH2,

No, not really, this designation would replace the RTP/AVP in SDP with
something like RTP/AVP-MMH. The codec names would stay the same, since
they are unaffected by this change. .

> etc?  I'm all for having zero overhead, but not at
> the expense of making a mess of SDP. That is sort
> of why I was attracted to RFC 2198. If somebody
> else has a better way of doing the same thing
> without requiring, for example, structured names
> in SDP that would be great too.
> 
>         Mike

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



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--NextPart

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

	Title		: RTP Payload for Text Conversation
	Author(s)	: G. Hellstrom
	Filename	: draft-ietf-avt-rtp-text-04.txt
	Pages		: 9
	Date		: 07-Feb-00
	
This memo describes how to carry text conversation session contents
in RTP packets. Text conversation session contents are specified in
ITU-T Recommendation T.140 [1]. 
Text conversation is used alone or in connection to other
conversational facilities such as video and voice, to form multimedia
conversation services.
This RTP payload description contains an optional possibility to
include redundant text from already transmitted packets in order to
reduce the risk of text loss caused by packet loss. The redundancy
coding follows RFC 2198.

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--NextPart

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

	Title		: RTP Payload Format for Real-Time Pointers
	Author(s)	: M. Civanlar, G. Cash
 	Filename	: draft-ietf-avt-pointer-01.txt,.ps
	Pages		: 5
	Date		: 09-Feb-00
	
This document describes an RTP [1] payload format for transporting the
coordinates of a dynamic pointer that may be used during a
presentation. Although a mouse can be used as the pointer, this
payload format is not intended and may not have all functionalitiesp
needed to implement a  general mouse event transmission mechanism.
This version of the draft includes MIME type registration information
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--NextPart

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

	Title		: RTP payload format for MPEG-4 Audio/Visual streams
	Author(s)	: Y. Kikuchi, T. Nomura, S. Fukunaga, Y. Matsui, 
                          H. Matsui
	Filename	: draft-ietf-avt-rtp-mpeg4-es-00.txt
	Pages		: 13
	Date		: 07-Feb-00
	
This document describes RTP payload formats for the carriage of MPEG-4
Audio and Visual streams[2][3], and an RTCP format for MPEG-4 upstream
messages functionalities[4]. In this specification, MPEG-4 Audio/Visual
bitstreams are directly mapped into RTP packets without using MPEG-4
Systems[6]. The RTP header fields usage and the fragmentation rule for
MPEG-4 Visual and Audio bitstreams are specified. It also specifies an
RTCP packet usage to carry the MPEG-4 upstream messages.

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Subject: Multicast Seminar 2/18: Vernor Vinge - The Digital Gaia
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Vernor Vinge, "The Digital Gaia"

NGI Distinguished Lecture - 18 February 2000, 1 PM EST

    In this talk, I assume that Moore's Law continues to be valid for
    another quarter century, and consider the consequences of that
    progress for VR/wearable computing, fine-grained distributed systems,
    collaborations, and central-site technology: How far can we take
    ubiquitous computing? What will our lives and work be like as we
    soar along this trajectory of hardware and software improvement?

The DARPA NGI Program is proud to announce the start of a new series
of live and telecast seminars on Next Generation Internet
technologies.  The first NGI Distinguished Lecturer will be the
Hugo Award-winning science fiction author, Vernor Vinge.  Numerous
Internauts have said their imaginations were expanded by Vinge's
works, including True Names (1981), often considered to be the
introduction of the fictional concept of cyberspace (three years
before William Gibson's Neuromancer), The Peace War (1984), and A Fire
upon the Deep (1992).  A Deepness in the Sky (1999) is the prequel to
A Fire upon the Deep; both of these novels feature a richly imagined
intergalactic email infrastructure.  A collection entitled True Names
And the Opening of the Cyberspace Frontier, reprinting the novella and
including essays by Richard Stallman and Danny Hillis, among others,
will appear in April 2000.  Vinge is also an associate professor of 
computer science at San Diego State University, where his interests
are in networks and distributed systems.

Location:

   National Center for Supercomputing Applications
   ACCESS Facility:
   Ballston Metro Center Office Tower, Suite 800
   901 North Stuart Street
   Arlington, Virginia 22203

Attendance:

   Space is limited.  Please RSVP to ngi-seminar-rsvp@isi.edu.
   Directions for getting to the Access Facility can be found at:
   
    http://www.ncsa.uiuc.edu/VEG/homepages/tcoffin/ACCESS/directions.html

Multicast/Webcast:

  The NGI Distinguished Lecture Series will be multicast live and also 
  webcast using RealMedia.  The live multicast will allow for questions and 
  comments from the remote audience.  To tune into the Internet multicasts,
  look for the announcement in your MBONE session directory program
  ('sdr').  If the announcement is not present there, it still may
  be possible to receive the session by manually configuring the client
  programs ('vic', and 'rat' or 'vat') with session addresses.  The
  session addresses, webcast details, and information on non-live access 
  to the seminar will be posted on:

    http://www.ngi-supernet.org/conferences.html

  During the seminar, any technical difficulties about the multicast can
  be addressed to dartnoc@isi.edu.


NGI Distinguished Lecture Series Future Speakers:

  23 March 2000   Kim Claffy, CAIDA (date tentative)
  Date TBA        Vint Cerf, Interplanetary Internet




From rem-conf Fri Feb 11 02:24:47 2000 
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--NextPart

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

	Title		: RTP Payload Format for Real-Time Pointers
	Author(s)	: G. Cash, M. Civanlar
	Filename	: draft-ietf-avt-pointer-02.txt,.ps
	Pages		: 5
	Date		: 10-Feb-00
	
This document describes an RTP [1] payload format for transporting the
coordinates of a dynamic pointer that may be used during a
presentation. Although a mouse can be used as the pointer, this
payload format is not intended and may not have all functionalities
needed to implement a general mouse event transmission mechanism.

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From rem-conf Fri Feb 11 13:43:27 2000 
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From: Stephen Casner <casner@cisco.com>
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cc: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Subject: URGENT question on draft-ietf-avt-tones-07
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To the IETF AVT working group and ITU SG16:

--> Interested parties need to respond by end of day PST Monday 2/14 <--

The following has come up during IESG discussion of the
draft-ietf-avt-tones-06.txt document:

> In section 3.11 there are two events (CRd and MRd) where the signals
> are different for the initiator and the responder.
> This implies that the entity doing the translation between audio and events
> need to be aware of the higher layer roles of the communicating peers.
> Is this a reasonable assumption?

This appears to be a valid concern that we did not realize in the
working group review of this document.  One would want the gateway
that is doing the translation from an event to a tone to not be
required to know the higher layer roles.

It would be possible to give two different codepoints to each of these
events.  I don't know if there has already been enough implementation
of this protocol that this would cause a problem, and the purpose of
this message is to find out.  Given that the modem and fax tones were
added fairly late in the development of the draft, I suspect is would
not be a problem to add two codepoints in that section (32-47) without
changing the other sections.

The author (Henning Schulzrinne) agreed:

> This is at least ambiguous, particularly for a "pass-through" gateway
> that's pretty clueless as to call state. There are effectively four tone
> sequences
> 
> 1375/2002 followed by 1150
> 1529/2225 followed by 1150
> 1357/2002 followed by 1900
> 1529/2225 followed by 1900
> 
> My suggestion would be to split the Crd and Mrd into Crdi and Crdr for
> the initiating and responding side. Same for the other pair.

The draft has been revised to make this split, so the codepoints 41-49
are now different.  The revised draft has been submitted but is
available in the meantime as:

http://www.cs.columbia.edu/~hgs/rtp/drafts/draft-ietf-avt-tones-07.txt
http://www.cs.columbia.edu/~hgs/rtp/drafts/draft-ietf-avt-tones-07.ps

I believe that, subject to this change, the draft has been approved by
the IESG.  If there are no objections from the working group by the
end of the day PST on Monday (2/14), then I will give our OK for the
-07 draft be sent to the RFC editor.  This is a very short response
time, but our goal is to get the RFC number assigned before the end of
the SG16 meeting on 2/18.  To those at the SG16 meeting, please call
this issue to the attention of interested parties that might not have
seen this message.
						-- Steve Casner




From rem-conf Mon Feb 14 13:25:03 2000 
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From rem-conf Mon Feb 14 20:01:05 2000 
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From: "Chopra, Rajeev" <rchopra@lucent.com>
To: "'rem-conf@es.net'" <rem-conf@es.net>
Cc: "Chopra, Rajeev" <rchopra@lucent.com>
Subject: Request for information on Gateways supporting "ietf-avt-tones-06
	.txt"
Date: Mon, 14 Feb 2000 11:34:15 -0500
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Hi,

I am looking for some information on current Gateways supporting the "RTP
payload tokens for
DTMF Signaling" as specified in "ietf-avt-tones-06.txt".

I need the following information

a.	Which all GWs out there currently support this spec, for use with
G.729A encoding.
b.	GWs with plans to support this mechanism in near term (6-12 months)
c.	The spec proposes two ways of conveying DTMF signals and events
i)	Named signals
ii)	Tone representation
		Any information on whether GWs currently supporting the spec
use I) or ii) or both ways. 
d. 	Does the  GW strip the DTMF signals for the RTP stream, if it uses
the proposed RTP token mechanism(s)?
		I know it is optional, as per the specs.

I am working on incorporation of G.729A in a Media server, with the proposed
RTP token mechanism 
 for conveying the DTMF signaling and events. Any responses to above
questions will be highly appreciated.

Thanks
Rajeev 
732-949-9059



From rem-conf Tue Feb 15 03:40:57 2000 
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--NextPart

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

	Title		: RTP Payload for DTMF Digits, Telephony Tones and 
                          Telephony Signals
	Author(s)	: H. Schulzrinne, S. Petrack
	Filename	: draft-ietf-avt-tones-07.txt,.ps
	Pages		: 29
	Date		: 14-Feb-00
	
This memo describes how to carry dual-tone multifrequency (DTMF)
signaling, other tone signals and telephony events in RTP packets.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avt-tones-07.txt

Internet-Drafts are also available by anonymous FTP. Login with the username
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From rem-conf Tue Feb 15 19:26:08 2000 
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Date: Wed, 16 Feb 2000 11:08:28 +0800
From: wmshen <wmshen@public.wh.hb.cn>
Subject: question on video transcoder or gateway
To: rem-conf@es.net
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Hello, all experts,
=20
I am a Ph.D student working on video communication. I am wondering about =
how the video transcoder/gateway works between H.261 and H.263, and =
between IP network and PSTN or ISDN, in paticular i wonder whether there =
is a method by which bitstreams can be translated directly between H.261 =
and H.263 format.
=20
Who knows the technique details on above topics or where may i get some =
useful reference infomation?
=20
Any help will be approciated.
=20
=20
Weiming Shen
=20

--Boundary_(ID_hLiUvVG1KajeUGQ/7I4TRw)
Content-type: text/html; charset=gb2312
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</HEAD>
<BODY bgColor=3D#ffffff>
<DIV>
<DIV><FONT color=3D#000000 face=3D"Times New Roman CE" size=3D2>Hello, =
all=20
experts,</FONT></DIV>
<DIV><FONT color=3D#000000 face=3D"Times New Roman CE" =
size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#000000 face=3D"Times New Roman CE" size=3D2>I am a =
Ph.D student=20
working on video communication. I am wondering about how the video=20
transcoder/gateway works between H.261 and H.263, and between IP network =
and=20
PSTN or ISDN, in paticular i wonder whether there is a method by which=20
bitstreams can be translated directly between H.261 and H.263=20
format.</FONT></DIV>
<DIV><FONT color=3D#000000 face=3D"Times New Roman CE" =
size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#000000 face=3D"Times New Roman CE" size=3D2>Who =
knows the=20
technique details on above topics or where may i get some useful =
reference=20
infomation?</FONT></DIV>
<DIV><FONT color=3D#000000 face=3D"Times New Roman CE" =
size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#000000 face=3D"Times New Roman CE" size=3D2>Any help =
will be=20
approciated.</FONT></DIV>
<DIV><FONT color=3D#000000 face=3D"Times New Roman CE" =
size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#000000 face=3D"Times New Roman CE" =
size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#000000 face=3D"Times New Roman CE" size=3D2>Weiming=20
Shen</FONT></DIV>
<DIV><FONT color=3D#000000 face=3D"Times New Roman CE"=20
size=3D2>&nbsp;</FONT></DIV></DIV></BODY></HTML>

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From rem-conf Wed Feb 16 02:15:42 2000 
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From: Abdul Sadka <a.sadka@eim.surrey.ac.uk>
To: wmshen <wmshen@public.wh.hb.cn>
cc: rem-conf@es.net
Subject: Re: question on video transcoder or gateway
In-Reply-To: <002b01bf782b$19953940$501f67ca@dhx60.telware>
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Video transcoding between two DCT-based compression algorithms consists of
syntactical mapping of output video parameters from one bitstream to
another. For the case of ITU-T H.261 and H.263, the layering structure and
the syntax are very similar except for few differences in half-pixel
accuracy and motion compensation that make the parameter-mapping a bit
more complicated. For instance, H.261 uses one MV predictor to compensate
for motion in the current MB as opposed to 3 MV predictors in H.263
whereby one median predictor in both x and y dimensions is considered for
motion compensation. This will induce certain complications in the
associated video transcoding algorithm.

However, the main focus in heterogeneous video transcoding today is on how
to map the output bitstreams of 2 video coders which have completely
different layering structures and syntactical properties. For example, it
is not very straightforward to transcode from MPEG-4 to H.263 since MPEG-4
employs the arbitrary shape VOP structure as opposed to the full picture
MB-Grid in H.263. To achieve this mapping, you will need to map the VOP
defined in MPEG-4 onto its precise location in the picture area of H.263
and extrapolate the pixels that are off the contour of the VOP but
inside the tightest box that encloses the arbirary shape object.

Perhaps, to have an introductory step into the area, you could refer to
the following publication:

S. Dogan, A. H. Sadka, A.M. Kondoz
"Efficient MPEG-4/H.263 Video Transcoder for Interoperability of
Heterogeneous Multimedia Networks" 
IEE Electronic Letters, Vol. 35, No. 11, pp 863-864, 27th May 1999.

Regards.
_________________________________________
      Dr Abdul H. Sadka (Lecturer)       |
Multimedia Communications Research Group |
Centre for Communication Systems Research|
Uni. of Surrey, GU2 5XH, Surrey, England |
Tel:         +44 (01483) 259490          |
Fax:         +44 (01483) 259504          |
WWW: www.ee.surrey.ac.uk/Personal/A.Sadka|
_________________________________________|


On Wed, 16 Feb 2000, wmshen wrote:

> Hello, all experts,
>  
> I am a Ph.D student working on video communication. I am wondering about how the video transcoder/gateway works between H.261 and H.263, and between IP network and PSTN or ISDN, in paticular i wonder whether there is a method by which bitstreams can be translated directly between H.261 and H.263 format.
>  
> Who knows the technique details on above topics or where may i get some useful reference infomation?
>  
> Any help will be approciated.
>  
>  
> Weiming Shen
>  
> 




From rem-conf Wed Feb 16 08:12:32 2000 
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To: rem-conf@es.net
Subject: RTP interoperability testing
Date: Wed, 16 Feb 2000 16:05:06 +0000
Message-ID: <3141.950717106@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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To the AVT working group,

As noted in the AVT meeting at the Washington DC IETF, it is necessary 
to collect interoperability reports for RTP implementations before we 
can progress the RTP specification and audio/video profile to draft
standard.  Two internet-drafts describe the type of interoperability
statement required:
	draft-ietf-avt-rtp-interop-02.txt
	draft-ietf-avt-profile-interop-00.txt
and a third draft (draft-ietf-avt-rtptest-02.txt) describes possible
testing strategies.

This message is a plea for vendors with RTP implementations to make
available results of their interoperability tests, in order that we can
advance the specifications. Results of the tests to date are available from
	http://www-mice.cs.ucl.ac.uk/multimedia/misc/avt/RTPinterop/ 
Please send additions to this table to me (if you have a problem with
making this information publically available, please contact me privately
and we can arrange alternatives).

Note that we cannot advance the revised RTP specification or audio/video
profile to become RFCs unless this information is received. Your help in
compiling the interoperability statements is vital.

Thanks,
Colin



From rem-conf Wed Feb 16 08:12:33 2000 
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From: Jori <jori@lumumba.luc.ac.be>
To: rem-conf@es.net
Subject: JRTPLIB v2.2
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Hello everybody,

I just wanted to say a couple of things
 - for those of you who have already downloaded jrtplib (version 2.1 or
   2.2). It contained a bug in rtpsourcedata.cpp. You can download the
   corrected file at 
   http://lumumba.luc.ac.be/jori/rtp/rtp.html
 - About JRTPLIB: It's an object-oriented RTP library, written in C++. It
   has been known to work on Linux, Solaris and Windows platforms.
   However, it should also work on other unix-like platforms...
   The download site is the same as above...

Bye,
Jori




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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		: Real-Time Transport Protocol Management 
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From: "Ludovic Havet" <Ludovic.Havet@fr.thomcast.thomson-csf.com>
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Subject: info on MPEG2 packet audio/video player ?
Date: Thu, 17 Feb 2000 17:47:43 +0100
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Hello,

I work on RTP with MPEG2 and search a way to visualize an MPEG2 packet
audio/video (or elementary) program. In this view I search a PC based board
or an STB. Also it would be great to have an API to feed the board and to
decide where and how to get the result.

If you have any information / offer...

Thanks in advance for your kind help,

Ludovic Havet




From rem-conf Thu Feb 17 12:58:27 2000 
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From: Gary Sullivan <garysull@microsoft.com>
To: 'Baldine Paul' <baldine@research.att.com>, cabo@tzi.org, 
	lscline@jf.intel.com, thomas.r.gardos@intel.com, Gary Sullivan
	 <garys@pictel.com>, stewe@cs.tu-berlin.de
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Subject: RE: H.263 RFC 2429
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To Baldine Paul re RFC 2429,
 
Just came across this old question -- I don't know why I didn't see it
earlier.
 
Section 5.1.1 was meant to say that UFEP could be 1 in the redundant header
copy
even if it was 0 in the payload stream.  Whether you actually do that or not
was
intentionally left up to the implementer.  For example, nothing is ever
changing in
the entire stream for hours on end, it is probably safe to not send full
headers.
 
Best Wishes,
 
Gary Sullivan
 
 

-----Original Message-----
From: Baldine Paul [mailto:baldine@research.att.com]
Sent: Friday, October 22, 1999 1:42 PM
To: cabo@tzi.org; lscline@jf.intel.com; thomas.r.gardos@intel.com; Gary
Sullivan; stewe@cs.tu-berlin.de
Cc: rem-conf@es.net
Subject: H.263 RFC 2429


I have a comment regarding the packetization scheme described in 
Section 5.1.2 of the RFC. It specifies that a redundant copy of the picture
header may be attached to the RTP header for packets that begin with GBSC or
SSC.
What is not obvious is whether that redundant picture header is full length
(with UFEP=001) or
not. 
Although, it could be left to the implementer to decide whether to include a
complete picture
header or not, I suggest, to be consistent with the recommendation of
Section 5.1.1, that
all redundant picture headers(for packets beginning with a Picture Start
Code,
a GBSC, a SSC or even a Follow-on packet), be complete, with UFEP=001.
 
Baldine-Brunel Paul
AT&T Labs-Research
Room 3-225, 100 Schultz Drive
Red Bank, NJ 07701
phone: 732-345-3312
baldine@research.att.com <mailto:baldine@research.att.com> 
http://www.research.att.com/info/baldine
<http://www.research.att.com/info/baldine> 
 


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<META content="MSHTML 5.00.2920.0" name=GENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=#ffffff>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=979575519-17022000>To 
Baldine Paul re RFC 2429,</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=979575519-17022000></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=979575519-17022000>Just 
came across this old question -- I don't know why I didn't see it 
earlier.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=979575519-17022000></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=979575519-17022000>Section 5.1.1 was meant to say that UFEP could be 1 in 
the redundant header copy</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=979575519-17022000>even 
if it was 0 in the payload stream.&nbsp; Whether you actually do that or not 
was</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=979575519-17022000>intentionally left up to the implementer.&nbsp; For 
example, nothing is ever changing in</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=979575519-17022000>the 
entire stream for hours on end, it is probably safe to not send full 
headers.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=979575519-17022000></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=979575519-17022000>Best 
Wishes,</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=979575519-17022000></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=979575519-17022000>Gary 
Sullivan</SPAN></FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<BLOCKQUOTE 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Baldine Paul 
  [mailto:baldine@research.att.com]<BR><B>Sent:</B> Friday, October 22, 1999 
  1:42 PM<BR><B>To:</B> cabo@tzi.org; lscline@jf.intel.com; 
  thomas.r.gardos@intel.com; Gary Sullivan; stewe@cs.tu-berlin.de<BR><B>Cc:</B> 
  rem-conf@es.net<BR><B>Subject:</B> H.263 RFC 2429<BR><BR></DIV></FONT>
  <DIV><FONT face=Arial size=2>I have a comment regarding the packetization 
  scheme described in </FONT></DIV>
  <DIV><FONT face=Arial size=2>Section 5.1.2 of the RFC. It specifies that a 
  redundant copy of the picture</FONT></DIV>
  <DIV><FONT face=Arial size=2>header may be attached to the RTP header for 
  packets that begin with GBSC or SSC.</FONT></DIV>
  <DIV><FONT face=Arial size=2>What is not obvious is whether that redundant 
  picture header is full length (with UFEP=001) or</FONT></DIV>
  <DIV><FONT face=Arial size=2>not. </FONT></DIV>
  <DIV><FONT face=Arial size=2>Although, it could be left to the implementer to 
  decide whether to include a complete picture</FONT></DIV>
  <DIV><FONT face=Arial size=2>header or not, I suggest, to be consistent with 
  the recommendation of Section 5.1.1, that</FONT></DIV>
  <DIV><FONT face=Arial size=2>all redundant picture headers(for packets 
  beginning with a Picture Start Code,</FONT></DIV>
  <DIV><FONT face=Arial size=2>a GBSC, a SSC or even a Follow-on 
  packet),</FONT><FONT face=Arial size=2> be complete, with 
  UFEP=001.</FONT></DIV>
  <DIV><FONT face=Arial size=2>&nbsp;</FONT></DIV>
  <DIV><FONT face=Arial size=2>Baldine-Brunel Paul<BR>AT&amp;T 
  Labs-Research<BR>Room 3-225, 100 Schultz Drive<BR>Red Bank, NJ 07701<BR>phone: 
  732-345-3312<BR><A 
  href="mailto:baldine@research.att.com">baldine@research.att.com</A><BR><A 
  href="http://www.research.att.com/info/baldine">http://www.research.att.com/info/baldine</A></FONT></DIV>
  <DIV>&nbsp;</DIV></BLOCKQUOTE></BODY></HTML>

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From rem-conf Thu Feb 17 20:26:27 2000 
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From: "Matthew Clay" <mclay@westga.edu>
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To: rem-conf@es.net
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Does anyone have experience using EnVision Desktop Video Conferencing?

Matthew Clay, Director
Education Technology Services
College of Education
State University of West Georgia





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From: Internet-Drafts@ietf.org
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Subject: I-D ACTION:draft-ietf-avt-rtp-text-05.txt
Date: Fri, 18 Feb 2000 06:34:15 -0500
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--NextPart

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

	Title		: RTP Payload for Text Conversation
	Author(s)	: G. Hellstrom
	Filename	: draft-ietf-avt-rtp-text-05.txt
	Pages		: 9
	Date		: 17-Feb-00
	
This memo describes how to carry text conversation session contents
in RTP packets. Text conversation session contents are specified in
ITU-T Recommendation T.140 [1]. 
Text conversation is used alone or in connection to other
conversational facilities such as video and voice, to form multimedia
conversation services.
This RTP payload description contains an optional possibility to
include redundant text from already transmitted packets in order to
reduce the risk of text loss caused by packet loss. The redundancy
coding follows RFC 2198.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-text-05.txt

Internet-Drafts are also available by anonymous FTP. Login with the username
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From rem-conf Fri Feb 18 09:52:09 2000 
From rem-conf-request@es.net Fri Feb 18 09:52:08 2000
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From: The IESG <iesg-secretary@ietf.org>
Subject: Protocol Action: RTP Payload for Text Conversation to Proposed
	 Standard
Date: Fri, 18 Feb 2000 12:21:03 -0500
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The IESG has approved the Internet-Draft 'RTP Payload for Text
Conversation' <draft-ietf-avt-rtp-text-05.txt> as a Proposed Standard.
This document is the product of the Audio/Video Transport Working
Group.  The IESG contact persons are Scott Bradner and Vern Paxson.


 
Technical Summary
 
 This document defines a way to carry low-rate text in RTP packets,
 in accordance with ITU-T Recommendation T.140.  The payload format
 provides for non-duplicated, in-order delivery, with loss detection
 and optional limited loss recovery via redundancy.

Working Group Summary

 The initial draft was revised as a result of input from the working
 group and there is good working group consensus for publishing the
 document.

Protocol Quality

 Vern Paxson reviewed the the spec for the IESG.


Note to RFC Editor:

The IESG requests that the Revision History section at the end of the
document be removed prior to publication as an RFC.



From rem-conf Fri Feb 18 11:15:20 2000 
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Message-ID: <38AD9A53.1C64A416@viewgraphics.com>
Date: Fri, 18 Feb 2000 11:15:31 -0800
From: Wei Li <wei@viewgraphics.com>
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To: Ludovic Havet <Ludovic.Havet@fr.thomcast.thomson-csf.com>
CC: rem-conf@es.net
Subject: Re: info on MPEG2 packet audio/video player ?
References: <NDBBIFIMNPJMAPFMEENKEEEMCBAA.Ludovic.Havet@fr.thomcast.thomson-csf.com>
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Hi Ludovic,

Viewgraphics Inc. makes PCI boards which can perform MPEG2 ASI input/output
>from a standard PC with NT4.0 or Windows 2000. There exists an SDK as well with
which you gain full access of the card. The system mainly works with transport
stream and seems to fit your application.

For more information, please take a look at: http://www.viewgraphics.com. Under
Products, look for "MediaPump XL with MediaSplice".


--Wei
------------------------------------------------
Wei Li
Viewgraphics Inc.
1340 Space Park Way
Mountain View, CA 94043
Tel: 650-903-4900(main)
       650-903-1481 (direct)
Fax:650-969-6388
E-mail: wei@viewgraphics.com
Web: www.viewgraphics.com
------------------------------------------------


Ludovic Havet wrote:

> Hello,
>
> I work on RTP with MPEG2 and search a way to visualize an MPEG2 packet
> audio/video (or elementary) program. In this view I search a PC based board
> or an STB. Also it would be great to have an API to feed the board and to
> decide where and how to get the result.
>
> If you have any information / offer...
>
> Thanks in advance for your kind help,
>
> Ludovic Havet




From rem-conf Fri Feb 18 14:41:10 2000 
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From rem-conf Sat Feb 19 12:05:16 2000 
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From rem-conf Sun Feb 20 21:39:01 2000 
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From: hIY0wQF37@mail.esshelp.com
DATE: 20 Feb 00 11:11:57 PM
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SUBJECT: Re:  It's Not Too Late!!!                                                                                                       sasfd
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From rem-conf Sun Feb 20 22:06:24 2000 
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From:     EZ <freemp3@altavista.com>
To:        <rem-conf@es.net>
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Hi
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From rem-conf Mon Feb 21 05:21:46 2000 
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From rem-conf Mon Feb 21 05:32:17 2000 
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From: The IESG <iesg-secretary@ietf.org>
Subject: Protocol Action: RTP Payload for DTMF Digits, Telephony Tones
	 and Telephony Signals to Proposed Standard
Date: Mon, 21 Feb 2000 08:28:31 -0500
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The IESG has approved the Internet-Draft 'RTP Payload for DTMF Digits,
Telephony Tones and Telephony Signals' <draft-ietf-avt-tones-07.txt> as
a Proposed Standard.  This document is the product of the Audio/Video
Transport Working Group.  The IESG contact persons are Scott Bradner
and Vern Paxson.

 
Technical Summary
 
 This document defines how to carry both particular telephony control tones
 and general multi-frequency tones in RTP packets.  Tones are explicitly
 encoded so that (1) they can be reproduced without loss of quality even by
 low-rate codecs, and (2) their meaning can be unambiguously determined.

Working Group Summary

 There's been solid working group consensus for the payload format for some
 time.  Discussion focussed on which events in particular to add, and how
 to represent the set of supported events in SDP and in a MIME registration.

Protocol Quality

 The document was reviewed for the IESG by Vern Paxson.



From rem-conf Tue Feb 22 06:28:39 2000 
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To: rem-conf@es.net
Subject: AVT agenda for Adelaide
Date: Tue, 22 Feb 2000 14:10:51 +0000
Message-ID: <6089.951228651@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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The AVT working group has requested two slots at the IETF meeting in
Adelaide, and have been allocated 
	15:30-17:30 on Wednesday 29th March
and	13:00-15:00 on Thursday 30th March
on the draft agenda. Anyone wanting to time to discuss their work at the
meeting should contact me and Steve, and we'll arrange the agenda.

Colin



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From: Joe Touch <touch@ISI.EDU>
Reply-To: sigcomm2000-info@acm.org
Organization: Sigcomm 2000
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Subject: reminder - Sigcomm 2000 tutorial deadline approaching
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The tutorial deadline for Sigcomm 2000 is Feb. 28, 2000.
Please see the website below for further information on
how to submit a proposal.

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


			   Call for Papers
		     ACM SIGCOMM 2000 Conference

		    August 28 - September 1, 2000
		    Grand Hotel, Stockholm, Sweden
		http://www.acm.org/sigcomm/sigcomm2000
		       sigcomm2000-info@acm.org

>>  Tutorial proposals:  February 28, 2000

The SIGCOMM 2000 conference seeks papers describing significant
research contributions to the field of computer and data communication
networks. Authors are invited to submit full papers concerned with
both theory and practice. Proposals for tutorials, information on
student paper award eligibility and procedures, and nominations for
the SIGCOMM Award are also sought at this time.

General Co-Chairs
	Per Gunningberg, Uppsala U., Sweden (perg@docs.uu.se)
	Steve Pink, Lulea U. Tech., Sweden (steve@cdt.luth.se)
Program Co-Chairs
	Christophe Diot, Sprint ATL, USA (cdiot@sprintlabs.com)
	Jim Kurose, U. Massachusetts, USA (kurose@cs.umass.edu)
Publicity Chair
	Joe Touch, USC/ISI, USA (touch@isi.edu, or sigcomm2000-info@acm.org)
Tutorials Chair
	Steve Pink, Lulea U Tech., Sweden (steve@cdt.luth.se)



From rem-conf Wed Feb 23 07:17:25 2000 
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From: "LMM IIT" <lmmiit@mail.ru>
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Subject: invitation
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Invitation to Participate in a Research Project to Develop the Key Encoding Module under the MPEG-4 Video Coding Standard


The new content-oriented video coding standard MPEG-4 adopted in 1999 specifies formats for coding and combining video objects present in video sequences. MPEG-4 is expected to open the way for advanced 21-century digital video coding. This standard, however, does not specify the mechanism or the procedure for locating (segmenting) such objects in a video sequence based on their motion and/or color. This is a subject of active research in many laboratories worldwide.

The problem of spatiotemporal segmentation of video sequences is a priority subject in the Semantic Segmentation research project started at the end of 1997 and conducted by the Laboratory of Mathematical Methods, Institute of Information Technologies, Russian Research Center, Kurchatov Institute, Moscow. This project was supported by SAIT (Samsung Advances Institute of Technology), Korea.

This Laboratory has been doing work in the field of computer vision for two decades. Its researchers have fulfilled a number of projects in theoretical and applied computer vision, medical and industrial diagnostics, automatic form entry and processing, and target detection and recognition.

Working on the Semantic Segmentation project, our computer vision group has been able to find conclusive solutions to several important problems inherent in trying to decompose a video sequence into individual objects. We believe that we have correctly formulated the problems that need be tackled to come closer to extracting video objects from natural video sequences (like sports or drama).

The purpose of this invitation is to get additional financial support that will make it possible to expand the team of researchers working on the project and to obtain the results in a shorter period.

With the additional financial support of about $100,000 over a period of 1.5 to 2 years, we would be able to address and solve most of essential problems in spatiotemporal object segmentation and develop workable technologies for general purpose video sequences.

On your request, we would be ready to provide additional information including Demos on the work done at the previous stages of the Semantic Segmentation project and a detailed technical proposal for the future research.


A.P. Aleksandrov, 
Director of the Institute of Information Technologies. 
lmmiit@mail.ru






From rem-conf Wed Feb 23 07:52:21 2000 
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From: Euchner Martin <Martin.Euchner@mchp.siemens.de>
To: rem-conf@es.net, Euchner Martin <Martin.Euchner@mchp.siemens.de>
Subject: proposal for RTP header authentication method / anti-spamming
Date: Wed, 23 Feb 2000 16:48:01 +0100
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Dear experts,

the following contribution attached below proposes an RTP header
authentication method to protect against denial-of-service attacks using RTP
packet flooding on discovered UDP/RTP ports.

First, some background information may be helpful for the reader's
understanding in general in addition to the contribution:
-	I am the editor in ITU-T Study Group 16 of draft recommendation
H.235 version 2, that deals with "Security and encryption for H-Series
(H.323 and other H.245-based) multimedia terminals".

-	Basically, H.235 describes the framework for security in H.323 based
multimedia systems. It offers a variety of security services such as
authentication, access control, integrity, confidentiality, non-repudiation
with digital signatures and integrated key-management with security
capability negotiation for the secured call signaling and secure media
transport over RTP. H.235v2 features also interoperable security profiles
for various purposes and scenarios.

-	H.235 basically offers two methods for protection of the RTP media
streams:
-	network layer security using IPSEC,
-	or application-level security where the RTP media payload data is
encrypted. This is the preferred method that is currently of major interest.
Several encryption algorithms could be applied, among them are DES,
Triple-DES and RC2.
		Note, that ITU-T SG16 currently does not see necessities to
integrity protect the RTP media stream, this is an issue left for further
study, once there would be a requirement.

-	As the most recent successful and severe denial-of-service attacks
on major ISPs and Internet services shows, sophisticated attacking tools are
already available in the hacker community. Such attacking tools could simply
be misused (with none or slight modifications perhaps) also against the RTP
protocol and thus thwarting any system that uses the RTP protocol including
H.323 Voice-over-IP systems.

-	While approved recommendation H.235v1 does not provide explicit
countermeasures against denial-of-service attacks, it is recognized that
H.235v2 should provide a simple, light-weight security means that is
sufficiently effective without the necessity to integrity-protect complete
media packet for performance reasons. The proposed light-weight security
improvement for RTP interworks both works with H.235 encrypted media streams
but also simply with unencrypted RTP payload data. H.235v2 cares for the
necessary key-management with automatic, integrated session
key-distribution, key update/synchronization and security capability
negotiation using H.245 open logical channel signaling means. H.235 allows
also manual key-management with statically configured keys. RTP does not
define any key-management and defines key-management tasks as by "non-RTP
means". Thus, our proposal nicely fits already in the current RTP approach
and concept.

-	At the most recent ITU-T Study Group 16 meeting in Geneva, CH, 7-18
February 2000; the attached contribution D.335 was accepted for H.235v2.
However, it was pointed out by some delegates, that the IETF AVT working
group currently discusses a revised version of the RTP protocol. As the
fundamental problem is more related to RTP than H.323, we feel, that it
would be wise to solve the problem on an RTP level and provide the solution
to other RTP applications as well. Thus, we see good opportunities to
discuss our ideas with you and that we hopefully could come to an agreement
of a common procedure.

-	Our workplan in ITU-T SG16 is as follows: H.235v2 has recently been
determined. The final version of the draft is scheduled for
approval/decision in November 2000. Our next Rapporteurs meeting will be
held in May. We see good chances to come to a resolution of these common
matters until then.

-	The idea that is described in the proposal, cryptographically
protects the changing portion of the RTP header (i.e. timestamp and sequence
number) with a message digest. The result is put into the modified RTP
padding fields in the end of the RTP packet. This should be backwards
compatible with existing implementations. The message digest could be
computed using encryption techniques (DES-MAC) or using a secure hash
algorithm (SHA-1). These algorithms are not part of D.335 and have been
recently added to the draft of H.235v2. The receiver could verify the
authenticity of the packet and drop it if the digest is not correct. This
operation consumes only low effort and saves any significant processing
effort or resources of the application on top of RTP that would occur
otherwise.

-	Another option could be to deploy the RTP header extension method.
It is not clear, whether this had significant benefits and how that exactly
would work.

-	At present the enhanced RTP protocol format is still very flexible;
the length of the AUTH and padlen fields are not fully fixed since H.235 can
negotiate these parameters using H.245. When the protocol extension became
part of RTP, some further considerations should be given on defining
appropriate packet formats of course.

-	The contribution is the ASCII-converted delayed contribution D.335
with some minor modifications. I've decided to have the H.245 and H.235
stuff still included for better understanding of the entire procedures even
the RTP issues in the current draft would likely not be affected by this.
Those of you who are interested in the original Word97 version of the
document, may obtain it on request of course.


I am looking forward to hearing your comments, opinions and questions on
this.

With kind regards,

Martin Euchner.

Here is the proposal:

TITLE:	RTP anti-spamming method in H.235v2
____________________________________________

This contribution describes a security enhancement to H.235v2 to counter
denial-of service attacks on UDP/RTP ports when sending spam media packets
against the recipient. The described improvement will increase the security
and reliability of H.323 Voice-over-IP-systems.

1. Introduction

		Current H.323 and H.235 based multimedia systems do not
provide explicit means against denial-of-service attacks when flooding
discovered RTP/UDP media ports with spam  packets. This is of concern in
mission-critical VoIP systems that require high availability and have to
resist basic attacks.

		Some simple, lightweight security means are necessary to
counter the flooding attack on a discovered UDP port. As a result of a
successful attack, the end system (terminal, gateway, MCU,...) may no longer
provide its service or function appropriately due to caused overload
situation while considerably wasting resources (processing, memory,..). Such
an attack using IP-masquerading techniques is basically possible on any
H.323 system, regardless of whether H.235 media (voice/video) encryption is
implemented or not. While firewalls usually cannot counter this kind of
attack once the UDP ports have been connected through, the countermeasure
could be implemented in the end-system. For an attacker listening to the
media port negotiation (H.245 or H.225.0), the denial-of service attack is
quite easy to perform. The attacker either sends arbitrary media packets
with meaningful RTP header information or even replays intercepted RTP media
packets on the discovered port. The attacker may even probe the destination
ports without listening to a signaling connection.

		Present H.235 does not provide adequate security means to
protect against such threats. While media integrity in general would provide
means to detect such denial-of-service attacks too, the required
countermeasures are usually quite CPU-cycle intensive as the entire media
packet would have to be cryptographically integrity protected. Actually,
media integrity would not only counter any content manipulation of the data
in transit but also provide the media anti-spamming as a side effect.
However, as the need for true media integrity is not obvious at the moment,
section H.235/B.3 lists integrity and replay protection of the RTP stream as
for further study.
		We provide an appropriate, yet simple lightweight
countermeasure, which is suitable for H.323 systems both with and without
implemented media encryption. Thus, the security can be enhanced
step-by-step.

		The basic idea is to utilize the end-to-end secret that can
be negotiated by means of H.235 anyway and apply it cryptographically to a
short, non-constant portion of the RTP packet and compute some digest upon;
actually this is a cryptographic message authentication code. Rapid checking
of the digest allows the receiver to filter out unauthorized packets from
unknown sources before spending any further considerable resources
processing the malicious packet. This shall thwart the denial-of-service
attacks of RTP packet flooding. We therefore call this security mechanism
the "media anti spamming method". The cryptographic digest may be computed
using a symmetric crypto algorithm such as DES or SHA1.


2. Proposal

		We propose to use the new media anti-spamming method as an
option in H.235v2 (summarized in a new subsection 11.2 of H.235v2) and adopt
the following modifications to H.245 as well.
		It is proposed to use the DES algorithm for the digest
computation; SHA1 may be used as an alternative when the system does not
offer media encryption.

2.1. H.245 modifications:
{this subsection is provided just for your information, it is subject to
H.245 and H.235}

		The capability of a terminal to support anti-spamming is
signaled in H.245 as follows. The anti-spaming capability is thus subject to
negotiation as well:
		EncryptionAuthenticationAndIntegrity offers
AuthenticationCapability, which is extended as follows for media
anti-spamming:

Enhancements to H.245 ANNEX A:

AuthenticationCapability	::=SEQUENCE
{
	nonStandard            	NonStandardParameter OPTIONAL,
	...
	antiSpamAlgorithm	OBJECT IDENTIFIER OPTIONAL
}

		Enhancements to H.245 Annex B.2.2.8 (Encryption,
Authentication and Integrity Capabilities)
		EncryptionCapabilities, if present, indicate the encryption
capabilities of a terminal for each media type where the capabilities are
present. The scope of encryption indicates whether the encryption is applied
to the entire bit stream, a portion of the bit stream in a standard way, or
a portion of the stream in a non standard way. The algorithm selects the
encryption algorithm. 
		AuthenticationCapabilities if present, indicate that the
authentication components of H.235 [16] are supported by the terminal.
antiSpamAlgorithm indicates the method and algorithm how to provide
countermeasures against flooding and denial-of-service attacks.

2.2. H.235 additions

		H.235v2 provides a new subsection "11.2 Media anti-spamming"
with the following content:

		The receiver of an RTP media stream may wish to counter
denial-of-service and flooding attacks on discovered RTP/UDP ports.
Receivers, when having implemented the anti-spam capability, can quickly
determine whether an obtained RTP packet stems from an unauthorized source
and discard it.

		The anti-spamming capability when set indicates use of the
anti-spamming mechanism either
		- for plaintext media data without media encryption (see
case 1 below) or
		- in combination with encrypted media data when
EncryptionCapability features an encryption algorithm (see case 2 below).

		Both options provide a lightweight RTP packet authentication
on selected fields through a computed message authentication code (MAC). The
MAC may be computed using an encryption algorithm (e.g. DES in MAC mode, see
ISO/IEC 9797) or using a cryptographic one-way function (e.g. SHA1). The MAC
algorithm is indicated in the object identifier of antiSpamAlgorithm. The
algorithm OID implicitly indicates also the size of the MAC; e.g. 1 block =
64 bits for DES MAC. In order to save bandwidth, the MAC could be truncated
albeit sacrificing some security; e.g., to a 32-bit MAC; this requires a
different object identifier then. The anti-spam method is independent of any
additional payload encryption (see cases 1 and 2 below).

		Anti-spamming uses the following RTP packet format (see
Figure 1) where the RTP padding sequence is interpreted as follows (see
H.225.0 Annex A.5).
		- The P bit in the RTP header shall be set to 1.
		- Padding bytes shall be appended at the end of the payload
with the following meaning:

 <------------------------------- RTP packet
----------------------------------------------->
 <--        RTP Header            --> <-RTP payload-> <--         RTP
padding             -->

+----------+------+-----------+------+---------------+---------------+------
--------+--------+
|... P ... | SEQ# | timestamp | .... | media payload | padding bytes |
AUTH    | padlen |
+----------+------+-----------+------+---------------+---------------+------
--------+--------+

\____    e.g. 64 bits     ___/        \__ possibly encrypted      __/
\e.g.64 bits / \1 byte/

 
^
 
|
           MAC_k(...SEQ#,timestamp)
----------------------------------------+

		Figure 1: General RTP packet format for media anti-spamming

		Note: If anti-spamming is not used, then the AUTH and padlen
fields are not used either and the usual RTP packet format applies.

1. Case anti-spamming-only:
		This case applies when the media data are not encrypted; see
Figure 2. The last octet of the RTP padding contains a count of how many
padding octets should be ignored at the end of the RTP packet. The other
padding bytes carry the MAC; see Figure 3. The MAC shall be computed over
the first crypto block of the RTP header including the varying timestamp and
sequence number using the negotiated MAC algorithm of antiSpamAlgorithm and
applying the symmetric secret. A static or manually configured shared secret
or a dynamically negotiated shared secret may be used according to the
procedures of H.235. For larger block sizes (more than 64 bits), some
sufficient additional bits of the RTP header or even the first media payload
shall be taken.

		As key for the MAC computation, it is recommended to use the
key that is obtained from the H.235 media session key distribution; although
the session key applied is not used for payload encryption. Secure fast
connect with key establishment (see H.323 Annex J) or manual keying may be
used for key management. The sender computes the MAC as described above and
includes the result in the MAC field in the RTP padding AUTH field. Sender
and receiver know the size of the AUTH field and the length of the MAC by
the antiSpamAlgorithm.

		The MAC verification at the receiver side should be done as
early as possible, if possible already within the RTP stack or at latest
before decryption or decompressing the payload. The receiver first
recomputes the MAC in the same way as the sender did and compares the
computed MAC with the delivered MAC in the RTP padding. If the MACs
mismatch, the RTP header has been modified in transit or was sent by an
unauthorized entity that does not possess the key. Thus, the
mis-authenticated RTP packet shall be discarded, the event may be logged;
this likely indicates an attempted denial-of-service attack. Otherwise, the
authenticated RTP packet can be processed further, the RTP padding is
removed and the payload is fed through the codec.

		Note: The lightweight MAC computation/verification with DES
encryption involves only a single encryption operation; alternatively, SHA1
MAC is computed on a short part of the packets of fixed length, thus the
crypto operations consume absolutely minimal processing resources.

 <------------------------------- RTP packet
-------------------------------->
 <--        RTP Header            --> <-RTP payload-> <--   RTP padding
-->

+----------+------+-----------+------+---------------+---------------+------
--+
|... P ... | SEQ# | timestamp | .... | media payload |     MAC       |
padlen |
+----------+------+-----------+------+---------------+---------------+------
--+

\___ e.g. 64 bits         ___/                        \e.g.64 bits  / \1
byte/

                                                             ^
                                                             |
           MAC_k(...SEQ#,timestamp) -------------------------+

		Figure 2: RTP packet format for media anti-spamming without
encryption


 <----- RTP padding ----------->

+------------------+------------+
|       MAC        |  padlen    |
+------------------+------------+
 \_ e.g. 64 bits _/ \_ 1 byte _/

Figure 3: RTP padding for anti-spamming-only

2. Case anti-spam method and payload encryption:
		This case applies when the media data are encrypted and the
anti-spamming method is invoked; see Figure 4. When the payload does not
fall on even block boundaries, some additional padding bytes have to be
appended to the payload in front of the MAC. The media payload encryption is
according to chapter 11/H.235.



 <------------------------------- RTP packet
----------------------------------------------->
 <--        RTP Header            --> <-RTP payload-> <--         RTP
padding             -->

+----------+------+-----------+------+---------------+---------------+------
--------+--------+
|... P ... | SEQ# | timestamp | .... | media payload | padding bytes |
AUTH    | padlen |
+----------+------+-----------+------+---------------+---------------+------
--------+--------+

\____    e.g. 64 bits     ___/        \_ encrypted in CBC mode    __/
\e.g.64 bits / \1 byte/

 
^
 
|
           MAC_k(...SEQ#,timestamp)
----------------------------------------+

		Figure 4: RTP padding for anti-spamming with payload
encryption

		EncryptionCapability defines the payload encryption
algorithm while antiSpamAlgorithm defines the anti-spamming method. For
security reasons, the media encryption and the MAC shall use different
session keys. The MAC key k is computed by feeding the encryption key K
through the SHA1 one-way hash function;
		k = SHA1(K); sufficient bits shall be taken from the hashed
result in network byte order. When antiSpamAlgorithm indicates an encryption
algorithm, then the collected bits shall be made a correct encryption key;
e.g. setting DES parity bits.

		After the receiver successfully verified the authenticity of
the RTP packet, the payload is decrypted and then the RTP padding be
discarded. The general procedure is according to case 1 above.




3. References

- Draft H.235v2: Security and Encryption for H-series (H.323 and other
H.245-based) Multimedia Terminals
- Draft H.323 Annex J: Security for H.323 Annex F
- H.245v7: Control Protocol for Multimedia Communication
- ISO/IEC 9797: Data cryptographic techniques - Data integrity mechanism
using a cryptographic check function employing an n-bit algorithm with
truncation.


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| Martin Euchner                 Fax  : +49 89 636-48000
| Siemens AG
| ZT IK 3                        mailto:Martin.Euchner@mchp.siemens.de
|                                mailto:martin.euchner@ties.itu.int
| Otto-Hahn-Ring 6               Intranet:
http://zt-security.mchp.siemens.de/de/Competence/Standardization/ITU-T_SG16/
index.html
| D-81730 Muenchen               Internet: http://www.siemens.de
| __________________
| Germany     
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From rem-conf Wed Feb 23 11:41:59 2000 
From rem-conf-request@es.net Wed Feb 23 11:41:58 2000
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Message-ID: <38B43691.188D0986@BMRC.Berkeley.EDU>
Date: Wed, 23 Feb 2000 11:35:45 -0800
From: "Lawrence A. Rowe" <Rowe@bmrc.berkeley.edu>
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Organization: U.C. Berkeley
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To: rem-conf <rem-conf@es.net>
Subject: "The Future of Interactive Television is Internet Webcasting"
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Hi -

I gave a talk on monday at the U of Illinois, Urbana-Champaign that
folks on this list might be interested in seeing.  I've included the
title and abstract below with a link to the archive that you can replay
(Windows Media Player).

Comments appreciated.
	Larry
-- 
Professor Lawrence A. Rowe          Internet:  Rowe@BMRC.Berkeley.EDU
Computer Science Division - EECS       Phone: 510-642-5117
University of California, Berkeley       Fax: 510-642-5615
Berkeley, CA 94720-1776            URL: http://bmrc.berkeley.edu/~larry

---
"The Future of Interactive Television is Internet Webcasting"
(archived at http://www.cs.uiuc.edu/dls/rowe/rowe.html)

Lawrence A. Rowe
Computer Science Division - EECS
University of California at Berkeley

Traditional broadcast television programs are composed of one video 
stream with no interaction, fixed image size, and constrained picture
quality. The development of media distribution networks such as 
Broadcast.com and Real Broadcast Networks and single-source multicast 
protocols suggests that Internet Webcasting will be yet another 
distribution channel for television programming with the same 
constraints imposed by current television distribution technologies 
(e.g., wireless, satellite, and cable).

Internet Webcasting, on the other hand, can support multiple video
streams, interaction between participants, and variable quality
streams.  In spite of these technical capabilities, most webcasts today
are produced using traditional television technologies and are 
constrained to the limitations imposed by that technology.

This talk describes research on developing computer-based webcasting 
technology that exploits capabilities of this new medium including
broadcast management, video-effects processing, and live production
control. The principle application is distance/asynchronous learning,
but the technology applies to many more applications including
distributed collaboration and entertainment.



From rem-conf Thu Feb 24 00:35:16 2000 
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Message-ID: <025901bf7e9f$dbb2ef20$3b1d7da6@hei.co.kr>
From: "Nam Kyu Kim" <batman@hei.co.kr>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
Cc: <rem-conf@es.net>
References: <Pine.LNX.4.10.10001181701060.5877-100000@canine.wipinfo.soft.net> <388480A9.50636D98@dynamicsoft.com> <200001181626.IAA29242@windsor.research.att.com> <38849B62.CA372ADB@dynamicsoft.com> <200001181724.JAA29905@windsor.research.att.com> <3884B6E6.76E311F2@dynamicsoft.com> <200001181914.LAA00776@windsor.research.att.com> <3884C503.FAEB6A54@dynamicsoft.com>
Subject: Re: rtp time stamp in RTCP packet....
Date: Thu, 24 Feb 2000 17:18:50 +0900
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From rem-conf Thu Feb 24 01:33:49 2000 
From rem-conf-request@es.net Thu Feb 24 01:33:48 2000
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Date: Thu, 24 Feb 2000 10:28:46 +0100
From: Martin Mauve <mauve@pi4.informatik.uni-mannheim.de>
Organization: University of Mannheim
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Hi all,

please excuse if this mail is slightly off topic. 

We have submitted an Internet Draft specifying a Real Time Application
Level Protocol for Distributed Interactive Media (RTP/I).

Here is the abstract of the RTP/I Internet Draft:

This document specifies RTP/I, an application level real-time
protocol for distributed interactive media. Typical examples of
distributed interactive media are shared whiteboards, networked
computer games and distributed virtual environments. RTP/I defines
a standardized framing for the transmission of data and provides
mechanisms that are universally needed for this media class.
Thereby RTP/I enables the development of reusable functionality and
generic services that can be employed for multiple distributed
interactive media. Examples for this kind of functionality are the
ability to record sessions, to support late coming participants,
and to provide security services. RTP/I is a protocol that follows
the ideas of application level framing and integrated layer
processing. It has been designed to be independent of the
underlying network and transport layers. To a large extend RTP/I
has been inspired by the real-time transport protocol (RTP), which
is used for continuous non-interactive media.

This document is intended to stimulate the discussion on how to
transport distributed interactive media over the Internet. 

There exists an RTP/I mailing list. Instructions on how to subscribe 
to this list can be found at the RTP/I homepage
(http://www.informatik.uni-mannheim.de/informatik/pi4/projects/RTPI/index.html). 
Feedback on this document should be addressed to the RTP/I mailing 
list or directly to the authors.

The RTP/I Internet Draft can be downloaded from the IETF directory or
>from our RTP/I homepage (see above for the URL).

We would welcome any kind of feedback!

cheers,

Martin
-- 
---------------------------------------------------------------------------
Martin Mauve                                         University of Mannheim
Praktische Informatik IV                            Phone: +49-621-181-2616
L 15,16                                             Fax  : +49-621-181-2601
68131 Mannheim                         mauve@pi4.informatik.uni-mannheim.de
Germany
---------------------------------------------------------------------------



From rem-conf Thu Feb 24 08:08:26 2000 
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          id <g.08060-0@bells.cs.ucl.ac.uk>; Thu, 24 Feb 2000 15:49:54 +0000
Message-ID: <010801bf7eda$e9399aa0$edc666c3@tenaga>
From: Farez <f.abdulrahman@cs.ucl.ac.uk>
To: rem-conf <rem-conf@es.net>, agents <agents@cs.umbc.edu>, 
    mobile-systems <mobile-systems@cs.ucl.ac.uk>
Subject: reminder - AMOC tutorial deadline approaching
Date: Thu, 24 Feb 2000 15:21:15 -0000
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1 March, 2000 is the tutorial proposal deadline for AMOC so there is 1 week
to submit your proposals. Please see the CFT below for detais.

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

                    Call for Tutorial Proposals

         FIRST ASIAN INTERNATIONAL MOBILE COMPUTING CONFERENCE
                           (AMOC 2000)
                1-3 November, 2000. Penang, Malaysia
                     Tutorials: 31 October, 2000
                   http://www.fsktm.um.edu.my/amoc
          http://www.cs.ucl.ac.uk/staff/F.AbdulRahman/amoc
           low bandwidth: append /lowband to URL above


        *** Proposal submission deadline: 1 March, 2000 ***

The AMOC Programme Committee invites proposals for tutorials to be held on
the 31st of October, 2000 in Penang, Malaysia.

We invite proposals for three-hour tutorials on topics relating to the
theory and/or practice of mobile and wireless computing and communications.
The aim is to offer conference delegates both tutorials on up-to-date
technologies, and case study tutorials on the application of the
technologies to real-world problems. A list of suggested topics related to
mobile/wireless computing/communications is given below as a guide; other
topics will naturally be considered:

o Quality of Service(QoS)           o handover & location management
o systems infrastructure            o satellite technology
o Internet access                   o power management
o data services                     o lower layer protocols
o rural wireless communications     o data management
o billing                           o security
o e-commerce                        o agent technology
o multimedia                        o hardware
o Personal Communication Systems    o terminal design & ergonomics

Submission Procedures
---------------------
All tutorial proposals should contain the following:
- A brief description of the tutorial, suitable for inclusion in the
  conference registration brochure.
- A detailed outline of the tutorial, including why the proposed topic is an
  important one and the content of the tutorial.
- The necessary background and the potential target audience for the
  tutorial.
- A description of why the tutorial topic is of interest to a substantial
  part of the AMOC audience.
- A brief resumé of the presenter(s), which should include name, postal
  address, phone and fax numbers, email address if available, background in
  the tutorial area, references to any published work in the area and
evidence
  of teaching experience, if any. This information may be included in the
  conference registration brochure.
- Any audio-visual and room requirements.

Tutorial proposals should be sent by email, in plain text format, to
amoc-submission@fsktm.um.edu.my by 1 March, 2000. For more information,
please
visit http://www.fsktm.um.edu.my/amoc/ or email
amoc-submission@fsktm.um.edu.my


Important Dates
---------------
Proposal submission deadline: 1 March, 2000
Notification of acceptance:   1 April, 2000
Tutorial abstract deadline:   1 May, 2000
Camera ready tutorial notes:  1 August, 2000

Additional Notes
----------------
Tutorial instructors will have their conference registration fees to AMOC
waived. However, a maximum of two presenters' fees can be waived if there
are two or more instructors.

The AMOC organising committee reserves the right to cancel any tutorial if
deadlines are missed or if the number of attendees is too low to support the
costs related to organising and running the tutorial.







From rem-conf Sun Feb 27 01:39:27 2000 
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	for <rem-conf@es.net>; Sun, 27 Feb 2000 12:19:36 +0300 (GMT)
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Date: 27 Feb 00 12:14:00 +-0300
From: "CA$HFLOWEB"<webmaster@cashfloweb.freeservers.com>
To: <rem-conf@es.net>
Subject: Do You Want to Be a Millionaire?
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Dear ,
 
Have you ever sat in front of your PC and thought that there must be some way to use this powerful piece of technology to make extra money, but didn't have slightest idea how to go about doing it? 
 
Welcome to the world of Mail Order!  Now before you get the idea that this will involve a lot of stamp-licking and trips to the post office, let me say that this business is not like the traditional mail-ordering business.  Instead of a lot of paperwork, your PC will be accomplishing the bulk of the work much faster and easier than if it was done manually.  The total investment for this opportunity is $5.00 (no kidding!) and the cost of five postage stamps.  The rest of the work will be done by your PC, and will only involve a small amount of time behind the keyboard.  But once again, I'd like to ask that you follow the instructions carefully.  The success or failure of this plan depends on the honesty and integrity of its participants.  If the plan is followed to the letter, it cannot fail.
 
go to the following link http://www.cashfloweb.freeservers.com. Once you are there, download the program from the main page and start making money.
 
Best Regards,
CA$HFLOWEB
 




From rem-conf Sun Feb 27 06:10:24 2000 
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Date: Sun, 27 Feb 2000 08:05:00 -0600
From: Ken Hopper <khopper@uchicago.edu>
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Organization: The University of Chicago
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"CA$HFLOWEB" wrote:
> 
> Dear ,
> 
> Have you ever sat in front of your PC and thought that there must be some way to use this powerful piece of technology to make extra money, but didn't have slightest idea how to go about doing it?
> 
> Welcome to the world of Mail Order!  Now before you get the idea that this will involve a lot of stamp-licking and trips to the post office, let me say that this business is not like the traditional mail-ordering business.  Instead of a lot of paperwork, your PC will be accomplishing the bulk of the work much faster and easier than if it was done manually.  The total investment for this opportunity is $5.00 (no kidding!) and the cost of five postage stamps.  The rest of the work will be done by your PC, and will only involve a small amount of time behind the keyboard.  But once again, I'd like to ask that you follow the instructions carefully.  The success or failure of this plan depends on the honesty and integrity of its participants.  If the plan is followed to the letter, it cannot fail.
> 
> go to the following link http://www.cashfloweb.freeservers.com. Once you are there, download the program from the main page and start making money.
> 
> Best Regards,
> CA$HFLOWEB
>



From rem-conf Sun Feb 27 10:10:41 2000 
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From: Falkor Technologies Inc. <info@falkortech.com>
To: Subscribers@es.net
Subject: All stock-related operations on your own machine! How could that be ?
Date: Sun, 27 Feb 2000 12:00:41 PDT
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From rem-conf Mon Feb 28 07:15:53 2000 
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From: mmaeda <mmaeda@darpa.mil>
To: "'ngi-list@ngi-supernet.org'" <ngi-list@ngi-supernet.org>
Subject: ngi seminar telcast: Vint Cerf - Interplanetary Internet
Date: Mon, 28 Feb 2000 09:52:43 -0500
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Folks,

Please pass along to your project team and to others who may be interested.

Mari
------

Vint Cerf, "Interplanetary Internet"

NGI Distinguished Lecture - 1 March 2000, 1 PM EST

  In order to expand the potential of Internet protocol beyond
  our realm and into space, I recently began working with NASA's
  Jet Propulsion Laboratory on a new project. The project
  entails developing an interplanetary communications system
  based on a special set of protocols that would carry
  transmissions between planets.

The DARPA and NSF NGI Programs are proud to announce the second
event in the NGI Distinguished Lecturer series of live and telecast
seminars on Next Generation Internet technologies.  Vint Cerf is
senior vice president of Internet Architecture and Technology for
MCI Worldcom.  Cerf is the co-designer of the TCP/IP protocol, the
communications protocol that gave birth to the Internet and which
is commonly used today. In December 1997, President Clinton
presented the U.S. National Medal of Technology to Cerf and his
partner, Robert E. Kahn, for founding and developing the Internet.
During his tenure from 1976-1982 with the U.S. Department of
Defense's Advanced Research Projects Agency (DARPA), Cerf played a
key role leading the development of Internet and Internet-related
data packet and security technologies.  Cerf served as founding
president of the Internet Society from 1992-1995 and as the
chairman of the Board from 1998-1999. He is a fellow of the IEEE,
ACM, American Association for the Advancement of Science, the
American Academy of Arts and Sciences and the National Academy of
Engineering.  He is on the Board of Directors of the Internet
Corporation for Assigned Names and Numbers.

Location:

   National Center for Supercomputing Applications
   ACCESS Facility:
   Ballston Metro Center Office Tower, Suite 800
   901 North Stuart Street
   Arlington, Virginia 22203

Attendance:

   Space is limited.  Please RSVP to ngi-seminar-rsvp@isi.edu.
   Directions for getting to the Access Facility can be found at 
   the seminar's listing on
   
    http://www.ngi-supernet.org/conferences.html

Multicast/Webcast:

  The NGI Distinguished Lecture Series will be multicast live and also 
  webcast using RealMedia.  The live multicast will allow for questions and 
  comments from the remote audience.  To tune into the Internet multicasts,
  look for the announcement in your MBONE session directory program
  ('sdr').  If the announcement is not present there, it still may
  be possible to receive the session by manually configuring the client
  programs ('vic', and 'rat' or 'vat') with session addresses.  The
  session addresses, webcast details, and information on non-live access 
  to the seminar will be posted on:

    http://www.ngi-supernet.org/conferences.html

  During the seminar, any technical difficulties about the multicast can
  be addressed to dartnoc@isi.edu.


NGI Distinguished Lecture Series Future Speakers:

  23 March 2000   Kim Claffy, CAIDA 

========================

If you would like your name removed from this mailing list, 
pls send a note to ngi-list-owner@ngi-supernet.org







From rem-conf Mon Feb 28 14:50:39 2000 
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From: The IESG <iesg-secretary@ietf.org>
SUBJECT: Last Call: RTP Payload Format for Real-Time Pointers to
	 Proposed Standard
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The IESG has received a request from the Audio/Video Transport Working
Group to consider RTP Payload Format for Real-Time Pointers
<draft-ietf-avt-pointer-02.txt> as a Proposed Standard.

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send any comments to the
iesg@ietf.org or ietf@ietf.org mailing lists by March 13, 2000.

Files can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-avt-pointer-02.txt




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To: Euchner Martin <Martin.Euchner@mchp.siemens.de>
Cc: rem-conf@es.net
Subject: proposal for RTP header authentication method / anti-spamming
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Euchner,

You might want to take a look at the Packetcable
security spec (http://www.packetcable.com) which
also defines RTP payload encryption and
authentication. The basic synopsis of the
packetcable media stream security scheme is as
follows:


1) Derive keying from upper layer signaling

2) Use RC-4 for the media encryption of the
   payload only. Since RC-4 is a stream cipher,
   potential packet drops need to be taken into
   account which the spec goes into quite a bit of
   detail about.

3) Use a 2 or 4 byte MAC which is appended to the
   end of the RTP packet itself. The packetcable
   spec does not use the pad bit, though in
   discussions on the SIP mailing list, this was
   also brought up as an option. Others think that
   the packetcable approach but using a different
   RTP profile (say, RTP/AVP-AUTH) ala RFC 1890
   might be the right answer. I tend toward that
   solution because it consumes the fewest bytes,
   but the pad method is also a potential.

   One other difference is that the packetcable
   spec runs the MAC algorithm over the entire
   RTP packet, instead of just selected fields.
   In addition to being more secure, I suspect
   that you will find that running the MAC
   algorithm over the entire RTP datagram is
   cheaper CPUwise since it is more regular
   (we're only talking about a few bytes
   difference, right?).

I have done some preliminary testing and the
overall results look very promising (this was
compiled on my Linux box at home with a P150 in
it):

code size: ~1kb
ram size: ~530 bytes/full duplex (ie send and receive) 
speed: 2MB/sec for the MAC+Encryption 

In other words, it is *very* well suited to
implementation on both large and small gateways,
and very DSP friendly.

		  Mike


Euchner Martin writes:
 > Dear experts,
 > 
 > the following contribution attached below proposes an RTP header
 > authentication method to protect against denial-of-service attacks using RTP
 > packet flooding on discovered UDP/RTP ports.
 > 
 > First, some background information may be helpful for the reader's
 > understanding in general in addition to the contribution:
 > -	I am the editor in ITU-T Study Group 16 of draft recommendation
 > H.235 version 2, that deals with "Security and encryption for H-Series
 > (H.323 and other H.245-based) multimedia terminals".
 > 
 > -	Basically, H.235 describes the framework for security in H.323 based
 > multimedia systems. It offers a variety of security services such as
 > authentication, access control, integrity, confidentiality, non-repudiation
 > with digital signatures and integrated key-management with security
 > capability negotiation for the secured call signaling and secure media
 > transport over RTP. H.235v2 features also interoperable security profiles
 > for various purposes and scenarios.
 > 
 > -	H.235 basically offers two methods for protection of the RTP media
 > streams:
 > -	network layer security using IPSEC,
 > -	or application-level security where the RTP media payload data is
 > encrypted. This is the preferred method that is currently of major interest.
 > Several encryption algorithms could be applied, among them are DES,
 > Triple-DES and RC2.
 > 		Note, that ITU-T SG16 currently does not see necessities to
 > integrity protect the RTP media stream, this is an issue left for further
 > study, once there would be a requirement.
 > 
 > -	As the most recent successful and severe denial-of-service attacks
 > on major ISPs and Internet services shows, sophisticated attacking tools are
 > already available in the hacker community. Such attacking tools could simply
 > be misused (with none or slight modifications perhaps) also against the RTP
 > protocol and thus thwarting any system that uses the RTP protocol including
 > H.323 Voice-over-IP systems.
 > 
 > -	While approved recommendation H.235v1 does not provide explicit
 > countermeasures against denial-of-service attacks, it is recognized that
 > H.235v2 should provide a simple, light-weight security means that is
 > sufficiently effective without the necessity to integrity-protect complete
 > media packet for performance reasons. The proposed light-weight security
 > improvement for RTP interworks both works with H.235 encrypted media streams
 > but also simply with unencrypted RTP payload data. H.235v2 cares for the
 > necessary key-management with automatic, integrated session
 > key-distribution, key update/synchronization and security capability
 > negotiation using H.245 open logical channel signaling means. H.235 allows
 > also manual key-management with statically configured keys. RTP does not
 > define any key-management and defines key-management tasks as by "non-RTP
 > means". Thus, our proposal nicely fits already in the current RTP approach
 > and concept.
 > 
 > -	At the most recent ITU-T Study Group 16 meeting in Geneva, CH, 7-18
 > February 2000; the attached contribution D.335 was accepted for H.235v2.
 > However, it was pointed out by some delegates, that the IETF AVT working
 > group currently discusses a revised version of the RTP protocol. As the
 > fundamental problem is more related to RTP than H.323, we feel, that it
 > would be wise to solve the problem on an RTP level and provide the solution
 > to other RTP applications as well. Thus, we see good opportunities to
 > discuss our ideas with you and that we hopefully could come to an agreement
 > of a common procedure.
 > 
 > -	Our workplan in ITU-T SG16 is as follows: H.235v2 has recently been
 > determined. The final version of the draft is scheduled for
 > approval/decision in November 2000. Our next Rapporteurs meeting will be
 > held in May. We see good chances to come to a resolution of these common
 > matters until then.
 > 
 > -	The idea that is described in the proposal, cryptographically
 > protects the changing portion of the RTP header (i.e. timestamp and sequence
 > number) with a message digest. The result is put into the modified RTP
 > padding fields in the end of the RTP packet. This should be backwards
 > compatible with existing implementations. The message digest could be
 > computed using encryption techniques (DES-MAC) or using a secure hash
 > algorithm (SHA-1). These algorithms are not part of D.335 and have been
 > recently added to the draft of H.235v2. The receiver could verify the
 > authenticity of the packet and drop it if the digest is not correct. This
 > operation consumes only low effort and saves any significant processing
 > effort or resources of the application on top of RTP that would occur
 > otherwise.
 > 
 > -	Another option could be to deploy the RTP header extension method.
 > It is not clear, whether this had significant benefits and how that exactly
 > would work.
 > 
 > -	At present the enhanced RTP protocol format is still very flexible;
 > the length of the AUTH and padlen fields are not fully fixed since H.235 can
 > negotiate these parameters using H.245. When the protocol extension became
 > part of RTP, some further considerations should be given on defining
 > appropriate packet formats of course.
 > 
 > -	The contribution is the ASCII-converted delayed contribution D.335
 > with some minor modifications. I've decided to have the H.245 and H.235
 > stuff still included for better understanding of the entire procedures even
 > the RTP issues in the current draft would likely not be affected by this.
 > Those of you who are interested in the original Word97 version of the
 > document, may obtain it on request of course.
 > 
 > 
 > I am looking forward to hearing your comments, opinions and questions on
 > this.
 > 
 > With kind regards,
 > 
 > Martin Euchner.
 > 
 > Here is the proposal:
 > 
 > TITLE:	RTP anti-spamming method in H.235v2
 > ____________________________________________
 > 
 > This contribution describes a security enhancement to H.235v2 to counter
 > denial-of service attacks on UDP/RTP ports when sending spam media packets
 > against the recipient. The described improvement will increase the security
 > and reliability of H.323 Voice-over-IP-systems.
 > 
 > 1. Introduction
 > 
 > 		Current H.323 and H.235 based multimedia systems do not
 > provide explicit means against denial-of-service attacks when flooding
 > discovered RTP/UDP media ports with spam  packets. This is of concern in
 > mission-critical VoIP systems that require high availability and have to
 > resist basic attacks.
 > 
 > 		Some simple, lightweight security means are necessary to
 > counter the flooding attack on a discovered UDP port. As a result of a
 > successful attack, the end system (terminal, gateway, MCU,...) may no longer
 > provide its service or function appropriately due to caused overload
 > situation while considerably wasting resources (processing, memory,..). Such
 > an attack using IP-masquerading techniques is basically possible on any
 > H.323 system, regardless of whether H.235 media (voice/video) encryption is
 > implemented or not. While firewalls usually cannot counter this kind of
 > attack once the UDP ports have been connected through, the countermeasure
 > could be implemented in the end-system. For an attacker listening to the
 > media port negotiation (H.245 or H.225.0), the denial-of service attack is
 > quite easy to perform. The attacker either sends arbitrary media packets
 > with meaningful RTP header information or even replays intercepted RTP media
 > packets on the discovered port. The attacker may even probe the destination
 > ports without listening to a signaling connection.
 > 
 > 		Present H.235 does not provide adequate security means to
 > protect against such threats. While media integrity in general would provide
 > means to detect such denial-of-service attacks too, the required
 > countermeasures are usually quite CPU-cycle intensive as the entire media
 > packet would have to be cryptographically integrity protected. Actually,
 > media integrity would not only counter any content manipulation of the data
 > in transit but also provide the media anti-spamming as a side effect.
 > However, as the need for true media integrity is not obvious at the moment,
 > section H.235/B.3 lists integrity and replay protection of the RTP stream as
 > for further study.
 > 		We provide an appropriate, yet simple lightweight
 > countermeasure, which is suitable for H.323 systems both with and without
 > implemented media encryption. Thus, the security can be enhanced
 > step-by-step.
 > 
 > 		The basic idea is to utilize the end-to-end secret that can
 > be negotiated by means of H.235 anyway and apply it cryptographically to a
 > short, non-constant portion of the RTP packet and compute some digest upon;
 > actually this is a cryptographic message authentication code. Rapid checking
 > of the digest allows the receiver to filter out unauthorized packets from
 > unknown sources before spending any further considerable resources
 > processing the malicious packet. This shall thwart the denial-of-service
 > attacks of RTP packet flooding. We therefore call this security mechanism
 > the "media anti spamming method". The cryptographic digest may be computed
 > using a symmetric crypto algorithm such as DES or SHA1.
 > 
 > 
 > 2. Proposal
 > 
 > 		We propose to use the new media anti-spamming method as an
 > option in H.235v2 (summarized in a new subsection 11.2 of H.235v2) and adopt
 > the following modifications to H.245 as well.
 > 		It is proposed to use the DES algorithm for the digest
 > computation; SHA1 may be used as an alternative when the system does not
 > offer media encryption.
 > 
 > 2.1. H.245 modifications:
 > {this subsection is provided just for your information, it is subject to
 > H.245 and H.235}
 > 
 > 		The capability of a terminal to support anti-spamming is
 > signaled in H.245 as follows. The anti-spaming capability is thus subject to
 > negotiation as well:
 > 		EncryptionAuthenticationAndIntegrity offers
 > AuthenticationCapability, which is extended as follows for media
 > anti-spamming:
 > 
 > Enhancements to H.245 ANNEX A:
 > 
 > AuthenticationCapability	::=SEQUENCE
 > {
 > 	nonStandard            	NonStandardParameter OPTIONAL,
 > 	...
 > 	antiSpamAlgorithm	OBJECT IDENTIFIER OPTIONAL
 > }
 > 
 > 		Enhancements to H.245 Annex B.2.2.8 (Encryption,
 > Authentication and Integrity Capabilities)
 > 		EncryptionCapabilities, if present, indicate the encryption
 > capabilities of a terminal for each media type where the capabilities are
 > present. The scope of encryption indicates whether the encryption is applied
 > to the entire bit stream, a portion of the bit stream in a standard way, or
 > a portion of the stream in a non standard way. The algorithm selects the
 > encryption algorithm. 
 > 		AuthenticationCapabilities if present, indicate that the
 > authentication components of H.235 [16] are supported by the terminal.
 > antiSpamAlgorithm indicates the method and algorithm how to provide
 > countermeasures against flooding and denial-of-service attacks.
 > 
 > 2.2. H.235 additions
 > 
 > 		H.235v2 provides a new subsection "11.2 Media anti-spamming"
 > with the following content:
 > 
 > 		The receiver of an RTP media stream may wish to counter
 > denial-of-service and flooding attacks on discovered RTP/UDP ports.
 > Receivers, when having implemented the anti-spam capability, can quickly
 > determine whether an obtained RTP packet stems from an unauthorized source
 > and discard it.
 > 
 > 		The anti-spamming capability when set indicates use of the
 > anti-spamming mechanism either
 > 		- for plaintext media data without media encryption (see
 > case 1 below) or
 > 		- in combination with encrypted media data when
 > EncryptionCapability features an encryption algorithm (see case 2 below).
 > 
 > 		Both options provide a lightweight RTP packet authentication
 > on selected fields through a computed message authentication code (MAC). The
 > MAC may be computed using an encryption algorithm (e.g. DES in MAC mode, see
 > ISO/IEC 9797) or using a cryptographic one-way function (e.g. SHA1). The MAC
 > algorithm is indicated in the object identifier of antiSpamAlgorithm. The
 > algorithm OID implicitly indicates also the size of the MAC; e.g. 1 block =
 > 64 bits for DES MAC. In order to save bandwidth, the MAC could be truncated
 > albeit sacrificing some security; e.g., to a 32-bit MAC; this requires a
 > different object identifier then. The anti-spam method is independent of any
 > additional payload encryption (see cases 1 and 2 below).
 > 
 > 		Anti-spamming uses the following RTP packet format (see
 > Figure 1) where the RTP padding sequence is interpreted as follows (see
 > H.225.0 Annex A.5).
 > 		- The P bit in the RTP header shall be set to 1.
 > 		- Padding bytes shall be appended at the end of the payload
 > with the following meaning:
 > 
 >  <------------------------------- RTP packet
 > ----------------------------------------------->
 >  <--        RTP Header            --> <-RTP payload-> <--         RTP
 > padding             -->
 > 
 > +----------+------+-----------+------+---------------+---------------+------
 > --------+--------+
 > |... P ... | SEQ# | timestamp | .... | media payload | padding bytes |
 > AUTH    | padlen |
 > +----------+------+-----------+------+---------------+---------------+------
 > --------+--------+
 > 
 > \____    e.g. 64 bits     ___/        \__ possibly encrypted      __/
 > \e.g.64 bits / \1 byte/
 > 
 >  
 > ^
 >  
 > |
 >            MAC_k(...SEQ#,timestamp)
 > ----------------------------------------+
 > 
 > 		Figure 1: General RTP packet format for media anti-spamming
 > 
 > 		Note: If anti-spamming is not used, then the AUTH and padlen
 > fields are not used either and the usual RTP packet format applies.
 > 
 > 1. Case anti-spamming-only:
 > 		This case applies when the media data are not encrypted; see
 > Figure 2. The last octet of the RTP padding contains a count of how many
 > padding octets should be ignored at the end of the RTP packet. The other
 > padding bytes carry the MAC; see Figure 3. The MAC shall be computed over
 > the first crypto block of the RTP header including the varying timestamp and
 > sequence number using the negotiated MAC algorithm of antiSpamAlgorithm and
 > applying the symmetric secret. A static or manually configured shared secret
 > or a dynamically negotiated shared secret may be used according to the
 > procedures of H.235. For larger block sizes (more than 64 bits), some
 > sufficient additional bits of the RTP header or even the first media payload
 > shall be taken.
 > 
 > 		As key for the MAC computation, it is recommended to use the
 > key that is obtained from the H.235 media session key distribution; although
 > the session key applied is not used for payload encryption. Secure fast
 > connect with key establishment (see H.323 Annex J) or manual keying may be
 > used for key management. The sender computes the MAC as described above and
 > includes the result in the MAC field in the RTP padding AUTH field. Sender
 > and receiver know the size of the AUTH field and the length of the MAC by
 > the antiSpamAlgorithm.
 > 
 > 		The MAC verification at the receiver side should be done as
 > early as possible, if possible already within the RTP stack or at latest
 > before decryption or decompressing the payload. The receiver first
 > recomputes the MAC in the same way as the sender did and compares the
 > computed MAC with the delivered MAC in the RTP padding. If the MACs
 > mismatch, the RTP header has been modified in transit or was sent by an
 > unauthorized entity that does not possess the key. Thus, the
 > mis-authenticated RTP packet shall be discarded, the event may be logged;
 > this likely indicates an attempted denial-of-service attack. Otherwise, the
 > authenticated RTP packet can be processed further, the RTP padding is
 > removed and the payload is fed through the codec.
 > 
 > 		Note: The lightweight MAC computation/verification with DES
 > encryption involves only a single encryption operation; alternatively, SHA1
 > MAC is computed on a short part of the packets of fixed length, thus the
 > crypto operations consume absolutely minimal processing resources.
 > 
 >  <------------------------------- RTP packet
 > -------------------------------->
 >  <--        RTP Header            --> <-RTP payload-> <--   RTP padding
 > -->
 > 
 > +----------+------+-----------+------+---------------+---------------+------
 > --+
 > |... P ... | SEQ# | timestamp | .... | media payload |     MAC       |
 > padlen |
 > +----------+------+-----------+------+---------------+---------------+------
 > --+
 > 
 > \___ e.g. 64 bits         ___/                        \e.g.64 bits  / \1
 > byte/
 > 
 >                                                              ^
 >                                                              |
 >            MAC_k(...SEQ#,timestamp) -------------------------+
 > 
 > 		Figure 2: RTP packet format for media anti-spamming without
 > encryption
 > 
 > 
 >  <----- RTP padding ----------->
 > 
 > +------------------+------------+
 > |       MAC        |  padlen    |
 > +------------------+------------+
 >  \_ e.g. 64 bits _/ \_ 1 byte _/
 > 
 > Figure 3: RTP padding for anti-spamming-only
 > 
 > 2. Case anti-spam method and payload encryption:
 > 		This case applies when the media data are encrypted and the
 > anti-spamming method is invoked; see Figure 4. When the payload does not
 > fall on even block boundaries, some additional padding bytes have to be
 > appended to the payload in front of the MAC. The media payload encryption is
 > according to chapter 11/H.235.
 > 
 > 
 > 
 >  <------------------------------- RTP packet
 > ----------------------------------------------->
 >  <--        RTP Header            --> <-RTP payload-> <--         RTP
 > padding             -->
 > 
 > +----------+------+-----------+------+---------------+---------------+------
 > --------+--------+
 > |... P ... | SEQ# | timestamp | .... | media payload | padding bytes |
 > AUTH    | padlen |
 > +----------+------+-----------+------+---------------+---------------+------
 > --------+--------+
 > 
 > \____    e.g. 64 bits     ___/        \_ encrypted in CBC mode    __/
 > \e.g.64 bits / \1 byte/
 > 
 >  
 > ^
 >  
 > |
 >            MAC_k(...SEQ#,timestamp)
 > ----------------------------------------+
 > 
 > 		Figure 4: RTP padding for anti-spamming with payload
 > encryption
 > 
 > 		EncryptionCapability defines the payload encryption
 > algorithm while antiSpamAlgorithm defines the anti-spamming method. For
 > security reasons, the media encryption and the MAC shall use different
 > session keys. The MAC key k is computed by feeding the encryption key K
 > through the SHA1 one-way hash function;
 > 		k = SHA1(K); sufficient bits shall be taken from the hashed
 > result in network byte order. When antiSpamAlgorithm indicates an encryption
 > algorithm, then the collected bits shall be made a correct encryption key;
 > e.g. setting DES parity bits.
 > 
 > 		After the receiver successfully verified the authenticity of
 > the RTP packet, the payload is decrypted and then the RTP padding be
 > discarded. The general procedure is according to case 1 above.
 > 
 > 
 > 
 > 
 > 3. References
 > 
 > - Draft H.235v2: Security and Encryption for H-series (H.323 and other
 > H.245-based) Multimedia Terminals
 > - Draft H.323 Annex J: Security for H.323 Annex F
 > - H.245v7: Control Protocol for Multimedia Communication
 > - ISO/IEC 9797: Data cryptographic techniques - Data integrity mechanism
 > using a cryptographic check function employing an n-bit algorithm with
 > truncation.
 > 
 > 
 > -----------------------------------------------------------------------
 > | Dipl.-Inf.                     Phone: +49 89 636-46201
 > | Martin Euchner                 Fax  : +49 89 636-48000
 > | Siemens AG
 > | ZT IK 3                        mailto:Martin.Euchner@mchp.siemens.de
 > |                                mailto:martin.euchner@ties.itu.int
 > | Otto-Hahn-Ring 6               Intranet:
 > http://zt-security.mchp.siemens.de/de/Competence/Standardization/ITU-T_SG16/
 > index.html
 > | D-81730 Muenchen               Internet: http://www.siemens.de
 > | __________________
 > | Germany     
 > -----------------------------------------------------------------------
 > 
 > 
 > 



From rem-conf Mon Feb 28 18:12:41 2000 
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Subject: Re: proposal for RTP header authentication method / anti-spamming
References: <ADA3BBFEC390D111A6230060979B34E6A7E241@mchp950a.mch.sbs.de> <14523.6487.46685.713776@thomasm-u1.cisco.com>
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Michael Thomas wrote:
> 
> Euchner,
> 
> You might want to take a look at the Packetcable
> security spec (http://www.packetcable.com) which
> also defines RTP payload encryption and
> authentication. The basic synopsis of the
> packetcable media stream security scheme is as
> follows:
> 
> 1) Derive keying from upper layer signaling
> 
> 2) Use RC-4 for the media encryption of the
>    payload only. Since RC-4 is a stream cipher,
>    potential packet drops need to be taken into
>    account which the spec goes into quite a bit of
>    detail about.
> 
> 3) Use a 2 or 4 byte MAC which is appended to the
>    end of the RTP packet itself. The packetcable
>    spec does not use the pad bit, though in
>    discussions on the SIP mailing list, this was
>    also brought up as an option. Others think that
>    the packetcable approach but using a different
>    RTP profile (say, RTP/AVP-AUTH) ala RFC 1890
>    might be the right answer. I tend toward that
>    solution because it consumes the fewest bytes,
>    but the pad method is also a potential.
> 
>    One other difference is that the packetcable
>    spec runs the MAC algorithm over the entire
>    RTP packet, instead of just selected fields.
>    In addition to being more secure, I suspect
>    that you will find that running the MAC
>    algorithm over the entire RTP datagram is
>    cheaper CPUwise since it is more regular
>    (we're only talking about a few bytes
>    difference, right?).

In a sense, 12 bytes of RTP header is more "regular" than the whole
payload. I agree that picking and choosing bits and pieces of the RTP
header is probably a waste of time.

However, one consideration is that this is needed not just for audio,
but also for video, in which case running MAC over the whole packet is
probably a good idea (2 Mb/s may not do the trick there, particularly
since you don't want to dedicate your whole CPU to this).

Thus, as I would hope that PacketCable looks ahead to more than just
audio, it appears that a header-only approach may be preferable.

> 
> I have done some preliminary testing and the
> overall results look very promising (this was
> compiled on my Linux box at home with a P150 in
> it):
> 
> code size: ~1kb
> ram size: ~530 bytes/full duplex (ie send and receive)
> speed: 2MB/sec for the MAC+Encryption
> 
> In other words, it is *very* well suited to
> implementation on both large and small gateways,
> and very DSP friendly.
> 


-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



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Michael Thomas wrote:
> 
> Henning Schulzrinne writes:
>  > However, one consideration is that this is needed not just for audio,
>  > but also for video, in which case running MAC over the whole packet is
>  > probably a good idea (2 Mb/s may not do the trick there, particularly
>  > since you don't want to dedicate your whole CPU to this).
> 
>    So, I'm confused: you're saying that the MAC
>    *should* be run over the RTP header + payload?
>    If so, I agree, and that is in fact what the
>    packetcable spec mandates.

No, it should only use the first 12 bytes RTP header (i.e., without
worrying about CSRC lists, although this is debatable), since video
packets are generally quite long and the overall video rate can well
approach your 2Mb/s capacity, sucking up all CPU just with doing
integrity checks. 

> 
>    It wasn't clear given my cursory reading of their
>    spec whether the MAC was run over the payload or
>    not. Assuming that it is not, the obvious
>    complaint is that it is susceptible to man in
>    the middle attacks which tamper with the
>    payload. While I guess this comes down to how
>    rigorous you want to be, it seems there should
>    be a huge amount of motivation to *not* do the
>    more secure thing. I'm not convinced at all for
>    audio that there is any such motivation, and
>    I'm pretty doubtful about video as well.

The primary motivation is data handling. In your scheme, a PC would do
nothing but generate the outgoing MAC and verify the incoming MAC, with
no cycles left for anything else useful. A nice denial-of-service attack
right there.

> 
>               Mike

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



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Henning Schulzrinne writes:
 > However, one consideration is that this is needed not just for audio,
 > but also for video, in which case running MAC over the whole packet is
 > probably a good idea (2 Mb/s may not do the trick there, particularly
 > since you don't want to dedicate your whole CPU to this).

   So, I'm confused: you're saying that the MAC
   *should* be run over the RTP header + payload?
   If so, I agree, and that is in fact what the 
   packetcable spec mandates.

   It wasn't clear given my cursory reading of their
   spec whether the MAC was run over the payload or
   not. Assuming that it is not, the obvious
   complaint is that it is susceptible to man in
   the middle attacks which tamper with the
   payload. While I guess this comes down to how
   rigorous you want to be, it seems there should
   be a huge amount of motivation to *not* do the
   more secure thing. I'm not convinced at all for
   audio that there is any such motivation, and
   I'm pretty doubtful about video as well.

	      Mike



From rem-conf Tue Feb 29 10:56:04 2000 
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my impression obtained by looking shortly at the CableLabs Spec (thanks for
providing the pointer) is that CableLabs might address other (i.e. higher)
security requirements than Henning and I have in mind. This however seems to
indicate, that RTP is being used by different applications, in different
scenarios with different security requirements. BTW, I do not know the
reason why CableLabs considered media integrity that important; is this
described somewhere? 
Thus, it seems evident, that RTP cannot and probably does not have to
satisfy them all with the same security mechanism. My guess is, that the
protocol should support the approaches in some basic manner while
maintaining enough degree of flexibility and find some common mechanism (is
this the padding method? or could we agree to something like that?) that
suits all purposes.

As I mentioned in my initial email, H.235 until now has not identified any
security need for provisioning media integrity. We left this issue for
further study and I believe this will stay so for some more time.

People have already concerns about media encryption performance and this
will become even worse when media integrity should be done as well. Please
remember that ITU-T SG 16 uses RTP not only in terminals, where performance
is not that issue but also in large scale voice gateways with thousands of
parallel media streams. Then each CPU cycle counts and unless there is a
real need for integrity, the encryption provided should probably already be
sufficient.

Manipulation on the encrypted media content would probably yield annoying
sounds at the receiver when trying to decode the manipulated data, but this
"denial-of-service" attack could be countered already when using the
proposed integrity check just over the header parts. So I consider media
integrity as a low(er) priority security requirement. I also doubt whether
media integrity is really appropriate just for countering denial-of-service
attacks with sent spam packets: this sounds a bit of over-engineering to me.

On the other hand, throwing away manipulated RTP packets does not result in
a "denial-of-service attack"; although from a user's perspective this might
look so. This is so, as RTP does not provide a guaranteed delivery of data
(because of unreliable UDP) and packets may get dropped anyway somewhere in
network elements.

For your information, I'm preparing already an internet-draft that goes
beyond my initial ideas. I expect it to be ready early next week and you
will get it as soon as it is ready. The paper will take into account media
integrity as an option as well and will describe also means how efficient
media integrity could be achieved without the need for extra CPU-intensive
integrity computation.

Martin
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| Martin Euchner                 Fax  : +49 89 636-48000
| Siemens AG
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	-----Original Message-----
	From:	Michael Thomas [SMTP:mat@cisco.com]
	Sent:	Tuesday, February 29, 2000 6:38 PM
	To:	Henning Schulzrinne
	Cc:	Michael Thomas; Euchner Martin; rem-conf@es.net
	Subject:	Re: proposal for RTP header authentication method /
anti-spamming

	Henning Schulzrinne writes:
	 > However, one consideration is that this is needed not just for
audio,
	 > but also for video, in which case running MAC over the whole
packet is
	 > probably a good idea (2 Mb/s may not do the trick there,
particularly
	 > since you don't want to dedicate your whole CPU to this).

	   So, I'm confused: you're saying that the MAC
	   *should* be run over the RTP header + payload?
	   If so, I agree, and that is in fact what the 
	   packetcable spec mandates.

	   It wasn't clear given my cursory reading of their
	   spec whether the MAC was run over the payload or
	   not. Assuming that it is not, the obvious
	   complaint is that it is susceptible to man in
	   the middle attacks which tamper with the
	   payload. While I guess this comes down to how
	   rigorous you want to be, it seems there should
	   be a huge amount of motivation to *not* do the
	   more secure thing. I'm not convinced at all for
	   audio that there is any such motivation, and
	   I'm pretty doubtful about video as well.

		      Mike



From rem-conf Tue Feb 29 16:00:53 2000 
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        Euchner Martin <Martin.Euchner@mchp.siemens.de>, rem-conf@es.net
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Henning Schulzrinne writes:
 > Michael Thomas wrote:
 > > 
 > > Henning Schulzrinne writes:
 > >  > However, one consideration is that this is needed not just for audio,
 > >  > but also for video, in which case running MAC over the whole packet is
 > >  > probably a good idea (2 Mb/s may not do the trick there, particularly
 > >  > since you don't want to dedicate your whole CPU to this).
 > > 
 > >    So, I'm confused: you're saying that the MAC
 > >    *should* be run over the RTP header + payload?
 > >    If so, I agree, and that is in fact what the
 > >    packetcable spec mandates.
 > 
 > No, it should only use the first 12 bytes RTP header (i.e., without
 > worrying about CSRC lists, although this is debatable), since video
 > packets are generally quite long and the overall video rate can well
 > approach your 2Mb/s capacity, sucking up all CPU just with doing
 > integrity checks. 

   Uh, I don't think we should get all hung up
   about actual speeds. My P150 can do 2-3 M*bytes*,
   not bits, and if done on a DSP with single
   cycle multiplies, it would be closer to 10MB.

   But this ignores the larger question though: do
   you actually consider man in the middle attacks
   to be of no concern? With the way you're
   proposing we do this, I can intercept your 700
   Club video packets, and adulter the payload to
   play out the Hustler Channel (or more
   frightening still, the reverse). Your RTP
   receiver would be none the wiser.

   As far as DoS attacks, I can just as easily
   send you lots of MTU sized packets with ESP
   headers with a valid SPI and spoofed IP headers 
   to and cause you do lots and lots of useless
   SHA-1 computations; it sort of goes with the
   territory. Making RTP marginally better for
   DoS attacks doesn't seem like it's worth much
   in the grander scheme of things, especially
   when you're opening yourself up to well known 
   (and serious, IMO) attacks.

		  Mike



From rem-conf Tue Feb 29 16:05:40 2000 
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Michael Thomas wrote:
> 
> Henning Schulzrinne writes:
>  > Michael Thomas wrote:
>  > >
>  > > Henning Schulzrinne writes:
>  > >  > However, one consideration is that this is needed not just for audio,
>  > >  > but also for video, in which case running MAC over the whole packet is
>  > >  > probably a good idea (2 Mb/s may not do the trick there, particularly
>  > >  > since you don't want to dedicate your whole CPU to this).
>  > >
>  > >    So, I'm confused: you're saying that the MAC
>  > >    *should* be run over the RTP header + payload?
>  > >    If so, I agree, and that is in fact what the
>  > >    packetcable spec mandates.
>  >
>  > No, it should only use the first 12 bytes RTP header (i.e., without
>  > worrying about CSRC lists, although this is debatable), since video
>  > packets are generally quite long and the overall video rate can well
>  > approach your 2Mb/s capacity, sucking up all CPU just with doing
>  > integrity checks.
> 
>    Uh, I don't think we should get all hung up
>    about actual speeds. My P150 can do 2-3 M*bytes*,
>    not bits, and if done on a DSP with single
>    cycle multiplies, it would be closer to 10MB.
> 
>    But this ignores the larger question though: do
>    you actually consider man in the middle attacks
>    to be of no concern? With the way you're
>    proposing we do this, I can intercept your 700
>    Club video packets, and adulter the payload to
>    play out the Hustler Channel (or more
>    frightening still, the reverse). Your RTP
>    receiver would be none the wiser.
> 
>    As far as DoS attacks, I can just as easily
>    send you lots of MTU sized packets with ESP
>    headers with a valid SPI and spoofed IP headers
>    to and cause you do lots and lots of useless
>    SHA-1 computations; it sort of goes with the
>    territory. Making RTP marginally better for
>    DoS attacks doesn't seem like it's worth much
>    in the grander scheme of things, especially
>    when you're opening yourself up to well known
>    (and serious, IMO) attacks.

Practically speaking, I can launch a DOS attack from just about
anywhere. It requires a fair amount of network access to intercept your
packets (since I need to be able to snoop them and then inject them
again).

That said, I agree we'd all be better off if we can agree on one method
within ITU/PacketCable/etc., rather than have half a dozen. Preferably a
method that then has an SDP negotiation mechanism associated with it;
would be even nicer to have one which is backward-compatible with
existing systems, as that would speed its deployment significantly.
Padding offers this, under the probably mistaken assumption that a
certain common RTP client made by a famous maker actually doesn't choke
on these (or header extensions).

> 
>                   Mike

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



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Michael Thomas wrote:
> 
> Henning Schulzrinne writes:
>  > However, one consideration is that this is needed not just for audio,
>  > but also for video, in which case running MAC over the whole packet is
>  > probably a good idea (2 Mb/s may not do the trick there, particularly
>  > since you don't want to dedicate your whole CPU to this).
> 
>    So, I'm confused: you're saying that the MAC
>    *should* be run over the RTP header + payload?
>    If so, I agree, and that is in fact what the
>    packetcable spec mandates.

I think the problem is that Henning left out a "not" in the sentence
above, so it should rather read:


However, one consideration is that this is needed not just for audio,
but also for video, in which case running MAC over the whole packet is
probably *NOT* a good idea (2 Mb/s may not do the trick there,
particularly
since you don't want to dedicate your whole CPU to this).


>    It wasn't clear given my cursory reading of their
>    spec whether the MAC was run over the payload or
>    not. Assuming that it is not, the obvious
>    complaint is that it is susceptible to man in
>    the middle attacks which tamper with the
>    payload.

Actually, you don't need man in the middle to tamper. You can simply
snoop packets, make a copy, remove the payload, insert your own payload,
and then send those. The receiver will get one or the other first; if
the faked ones are received first you will be able to insert your forged
audio stream. 

 While I guess this comes down to how
>    rigorous you want to be, it seems there should
>    be a huge amount of motivation to *not* do the
>    more secure thing. I'm not convinced at all for
>    audio that there is any such motivation, and
>    I'm pretty doubtful about video as well.

Just to make sure we're not missing a not here also - you're arguing for
the less secure mechanism? 

-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




