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Date: Thu, 1 Jul 1999 01:09:55 -0700 (Pacific Daylight Time)
From: Stephen Casner <casner@cisco.com>
To: rem-conf@es.net
Subject: Four drafts for AVT
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I submitted four Internet-Drafts for AVT which should be in the
pipeline.  They are also available from:

 ftp://ftpeng.cisco.com/casner/outgoing/draft-ietf-avt-rtp-new-04.ps
 ftp://ftpeng.cisco.com/casner/outgoing/draft-ietf-avt-rtp-new-04.txt

 ftp://ftpeng.cisco.com/casner/outgoing/draft-ietf-avt-profile-new-06.ps
 ftp://ftpeng.cisco.com/casner/outgoing/draft-ietf-avt-profile-new-06.txt

 ftp://ftpeng.cisco.com/casner/outgoing/draft-ietf-avt-rtp-mime-00.txt

 ftp://ftpeng.cisco.com/casner/outgoing/draft-ietf-avt-rtcp-bw-00.txt

The first one is the RTP spec, where the changes were only minor
fixes.  This time Section 12 is not missing from the .txt version.

The second is the RTP Audio/Video Profile.  The main change there was
to the wording of the reference to the separate draft on registering
payload format (encoding) names as MIME subtypes.  That reference
needs to be non-normative since the other draft won't be ready to go
to Draft Standard yet.  See if the wording still conveys the right
message.  I also removed some encoding names/descriptions for which we
really did not have any specification of the payload format, and
probably no implementations.

The rtp-mime draft is a significant revision of the very rough draft
for the MIME subtype registration that we discussed in Minneapolis.  I
think this draft is more or less complete now, and would greatly
appreciate your comments.

The last one is a writeup of the SDP extension to add RTCP bandwidth
modifier keywords for the b= line.  There was no draft in Minneapolis,
but I described what was needed.  We decided there that the value for
these keywords should be in b/s rather than a fractional kb/s (either
of which would be in conflict with the SDP spec).  Again, comments
appreciated.
							-- Steve




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To: rem-conf@es.net
Subject: AVT agenda for Oslo
cc: agenda@ietf.org
Date: Thu, 01 Jul 1999 14:23:30 +0100
Message-ID: <3079.930835410@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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Folks,

Enclosed is the preliminary agenda for the AVT working group meeting in
Oslo. Please send comments/corrections to Steve and I as soon as possible.
Colin





			 Audio/Video Transport WG
				     
				A G E N D A


Wednesday, 14 July at 09:00-11:30
=================================

- Status and progress since last meeting		(Casner)	10
	- SSRC sampling 
	- Guidelines 
	- QCELP

- Update on RTP specification and audio/video profile	(Casner)	20
	draft-ietf-avt-rtp-new-04.txt, .ps
	draft-ietf-avt-profile-new-06.txt, .ps
	draft-ietf-avt-rtp-mime-00.txt
	draft-ietf-avt-rtcp-bw-00.txt

- RTP interoperability statement and testing strategies
  	draft-ietf-avt-rtp-interop-00.txt		(Perkins)	10
  	draft-ietf-avt-rtptest-00.txt			(Perkins)	10
	draft-ietf-avt-rtcptest-01.txt			(Rosenberg)	10
	Discussion							10

- RTP MIB
	draft-ietf-avt-rtp-mib-05.txt			(Baugher)	 5

- RTP Payload for DTMF tones
	draft-ietf-avt-tones-01.txt			(Petrack)	10

- RTP Payload for Generic FEC
	draft-ietf-avt-fec-06.txt			(Rosenberg)	 5

- MPEG4 payload format
	Report from New York meeting

- RTP Payload for MPEG-4 with Scaleable & Flexible Error Resiliency
	draft-guillemot-genrtp-01.txt			(Wesner)	20

- RTP Payload for DV video 
	draft-ietf-avt-dv-video-00.txt			(Kobayashi)	10

- RTP encapsulation for 12-bit DV audio 
	draft-kobayashi-dv-audio12-00.txt		(Kobayashi)	10



Thursday, 15 July at 15:30-17:30
================================

- RTP Payload for AAC audio
	draft-kretschmer-mpeg2aac-00.txt (?)		(Kretschmer)	10

- RTP Payload for real-time pointers
	draft-civanlar-rtp-pointer-00.txt		(Civanlar)	10

- RTP Payload for Text Conversation
							(Hellström)	10

- Header compression for RTP/UDP/IP over cellular links
	draft-degermark-crtp-cellular-00.txt, .ps	(Degermark)	20

- Multiplexing header-compressed RTP over PPP
	draft-pazhyannur-avt-pppmux-00.txt		(Pazhyannur)	10

- Tunneled Compressed RTP	
	draft-wing-avt-tcrtp-00.txt			(Koren)		10

- A Taxonomy of Feedback for Multicast					10
	draft-hnrs-rmt-avt-feedback-00.txt		(Rosenberg)	

- Profile for unicast RTCP 						10
	<no internet draft>				(Casner)

- Discussion of new profiles/feedback mechanisms			10

- RTP payload for virtual worlds
	draft-stewart-avt-00.txt			(Stewart)	15




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We have submitted an Internet draft which is of interest to the AVT and the
PPPext groups. We will be making a 10 minute presentation about the draft to
the AVT group on Thursday, 15th July during the 15:30-17:30 session.

Comments on the draft are welcomed.

Details:

Title:   Multiplexing Compressed RTP/UDP Packets in a PPP Frame
Author:   Rajesh Pazhyannur and Irfan Ali
Filename: draft-pazhyannur-avt-pppmux-00.txt
Pages:    7
Date:     22-June-99
Abstract

   This draft proposes a scheme to increase the capacity of low-speed
   transmission links, T-1/E-1 or smaller, to transport low-bit rate
   voice applications over PPP. Given the relatively small payload size
   of the voice packets (average of 8 bytes for certain cellular
   networks) protocol overheads due to PPP, IP, UDP, and RTP are
   significant in determining the link capacity. The scheme proposed
   here is to multiplex compressed RTP/UDP packets into a single PPP
   frame. As a result, the PPP overhead (5-7 bytes) is distributed over
   multiple compressed RTP/UDP packets.

You can get the draft at

http://www.ietf.org/internet-drafts/draft-pazhyannur-avt-pppmux-00.txt


Irfan Ali & Rajesh Pazhyannur



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Date: Fri, 02 Jul 1999 00:36:37 +0200
To: rem-conf@es.net
From: Gunnar Hellstrom <gunnar.hellstrom@omnitor.se>
Subject: RTP for text conversation for discussion in Oslo
Cc: textphone@lsv.pi.se (Text Telephone Forum)
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I am sorry to clobber your mail, but here is an IETF draft for discussion
on the Thursday AVT session in Oslo, attached to this mail.=20
The topic is RTP format for text conversation data, needed for H.323 Annex=
 G.

I did not succeed to put it in any FTP area this time.

This is a later version of a draft that was distributed a week ago. The
main difference is that when redundant data is used, it is now coded
according to RFC 2198. It is better than invent a new mechanism.

A Security section is also added.

Best regards
Gunnar Hellstr=F6m
--=====================_930861397==_
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Internet Engineering Task Force                                   AVT WG
Internet Draft                                               Gunnar=
 Hellstr=F6m
draft-hellstrom-avt-rtp-text-00.txt                            L M Ericsson
June 25, 1999
Expires: Dec 25, 1999


                      RTP Payload for Text Conversation

STATUS OF THIS MEMO

   This document is an Internet-Draft and it is in full conformance with all
   provisions of section 10 of RFC2026. Internet-Drafts are working
   documents of the Internet Engineering Task Force (IETF), its areas,
   and its working groups.  Note that other groups may also distribute
   working documents as Internet-Drafts.

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

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

   The list of Internet-Draft Shadow Directories can be accessed at=20
   http://www.ietf.org/shadow.html

   Distribution of this document is unlimited.


ABSTRACT

   This memo describes how to carry text conversation session contents in=
 RTP
   packets. Text conversation session contents is specified in ITU-T
   Recommendation T.140 [1].=20

   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.

1 Introduction

   This memo defines a payload type for carrying text conversation session=
=20
   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. Text in text conversation sessions=
 is
=20
Hellstr=F6m                                                   [Page 1]



Internet Draft


   sent as soon as it is available, or with a small delay of up to=20
   0.5 seconds for buffering.
   The text is supposed to be entered by human users from a keyboard,
   handwriting recognition, voice recognition or any other input method. The
   rate of character entry is usually at a level of a few characters per=
 second
   or less. Therefore, the expected number of characters to transmit is low.
   Only one or a few new characters are expected to be transmitted with each
   packet.

   T.140 specifies that text and other T.140 elements shall be transmitted=
 in =20
   ISO 10 646-1 code with UTF-8 transformation. That makes it easy to=
 implement
   internationally useful applications, and to handle the text in modern =20
   information technology environments.=20
=20
   T.140 requires the transport channel to provide characters without
   duplication and in original order.=20
   Text conversation users expect that text will be delivered with no or a=
 low
   level of lost information. If lost information can be replaced with a=
 special=20
   marker, the willingness to accept loss is expected to be higher.
=20
   Therefore a mechanism based on RTP is specified here. It gives text=
 arrival  =20
   in correct order, without duplications, with an optional possibility to =
=20
   repeat data for redundancy to lower the risk of loss, and a mechanism=
 that=20
   reveals loss and therefore can insert a marker for lost text in the=
 received
   stream. Since packet overhead is usually much larger than the T.140=
 contents,
   the increase in channel load by the redundancy scheme is minimal.=20

2. Usage of RTP
   When an unreliable transport of T.140 text session data in the Internet=
 is
   selected, RTP with payload as described in this specification should be=
 used.
   T.140 data is submitted for transmission in a numer of complete protocol=
 data
   units (a T140block consisting of one or more T140 PDU:s). The most common
   T140 PDU is a character of ISO 10646 text, UTF-8 coded, submitted from=
 T.140
   without any extra framing.
  =20
   A text conversation RTP packet contains an RTP header, a Payload
   Header, optional redundant data fields, and the new (primary) T.140 data
   field.
  =20
2.1 RTP packet header
  =20
   Each RTP packet starts with a fixed RTP header. The following fields of=
 the
   RTP fixed header are used for T.140 text streams:

   Payload Type (PT): If redundancy is used, the Payload Type shall indicate=
 redundancy according to RFC 2198. If no redundnacy is used, the Payload=
 Type shall specify the primary T.140 text payload
   format.=20

   Sequence number:  The Sequence Number shall be increased with one for=
 each
   new transmitted packet. It is used for detection of packet loss and=
 packets
   out of order, and can be used in the process of retrieval of redundant=
 text,=20
   reordering of text and marking missing text.

Hellstr=F6m                                                   [Page 2]



Internet Draft                                 =20



   Timestamp: The RTP Timestamp encodes the approximate sampling instance of=
=20
   the primary text in the packet. No packets should use the same timestamp.

2.2 Additional headers
When redundant data is used, additional headers specify the redundant text=
 fields. They are specified according to RFC 2198 and use payload type=
 "redundant T.140 data".

2.3 T.140 Text structure

   T.140 text is UTF-8 coded as specified in T.140 with no extra framing.=
 The
   primary T.140 data is identified with the payload type "primary T.140=
 data".=20
   The transmitting station may select a number of T.140 block generations=
 to
   retransmit in each packet. A higher number introduces better protection
   against loss of text. If network conditions are not known, it is=
 recommended
   to use two generations. A maximum of 6 generations may be specified.

3. Procedures
   Based on the information in the received packets, the receiver can:
      -reorder received text out of order.
      -mark where text is missing because of packet loss.
      -compensate for lost packets by using redundant data.

---note-procedure description for insertion in a protocol spec--------

Procedure  =20
   On reception, the RTP sequence number is compared with the sequence=
 number of
   the last correctly received packet.=20
   If they are consecutive, only the most recent t140block is submitted from=
 the
   packet to T.140.=20

Compensation for lost packets.
 =20
   If there is a gap in the RTP sequence numbers, older t140blocks are=
 retrieved
   from the packet up to the point where the sequence number is consecutive=
 to
   the last correctly received packet or no more t140block is available in=
 the
   received packet.
   If there is still a gap in the sequence, one T.140 missing data marker
   (  Unicode character 2607 "Lightning") is inserted, UTF-8 coded,=
 replacing
   each missing t140block.
=20
   All t140blocks retrieved in this way are submitted to the receiving T.140
   instance.










Hellstr=F6m                                                   [Page 3]
Internet Draft                  			Expires 25 Dec 1999

Compensation for packets out of order.
   For protection against packets arriving out of order, the following=
 procedure
   may be implemented in the receiver.
   If analysis of a received packet reveals gaps in the sequence, the=
 received
   packet can be kept in a buffer before submission to the T.140 layer.=20
   If a packet arrives with a t140block belonging to the gap, the gap is=
 filled
   with the contents of this block.=20
   If all gaps are filled or the packet is kept in the buffer for 0.5=
 seconds,
   valid T140blocks in the buffered packet are retrieved and submitted to=
 the
   T.140 layer. For each t140block still missing, a T.140 missing data=
 marker
   (Unicode character 2607 "Lightning") is inserted. =20
--------- end of note ------------------------------------------------


4.  Security Considerations
    The same security considerations as for RFC 2198 apply.

5   Authors Address

   Gunnar Hellstr=F6m
   L M Ericsson
   Sweden

   e-mail: gunnar.hellstrom@omnitor.se
   Tel:    +46 708 204 288
   Fax:     +46 8 556 002 06


6 Bibliography

[1] ITU-T Recommendation T.140 (1998) - Text conversation protocol for
    multimedia application.






















Hellstr=F6m                                                   [Page 4]

--=====================_930861397==_
Content-Type: text/plain; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable


------------------------------------------------
Gunnar Hellstr=F6m
Omnitor AB
Alsn=F6gatan 7, 4tr
S-116 41 Stockholm
SWEDEN

Tel +46 751 100 501
Video +46 8 556 002 05
Text (V.18) +46 8 556 002 05
Fax +46 8 556 002 06

E-mail: gunnar.hellstrom@omnitor.se
WWW:    www.omnitor.se
------------------------------------------------
--=====================_930861397==_--




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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-2 AAC Streams
	Author(s)	: M. Kretschmer, A. Basso, M. Civanlar,
                          S. Quackenbush, J. Snyder
	Filename	: draft-ietf-avt-rtp-mpeg2aac-00.txt
	Pages		: 8
	Date		: 01-Jul-99
	
This document describes a payload format for transporting MPEG-2 AAC
encoded data using RTP. MPEG-2 AAC is a recent standard from ISO/IEC
for the coding of multi-channel audio data. Several services provided
by RTP are beneficial for MPEG-2 AAC encoded data transport over the
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synchronize MPEG-2 AAC data with other real-time data types.

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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-00.txt
	Pages		: 23
	Date		: 01-Jul-99
	
This memo describes how to carry dual-tone multifrequency (DTMF)
signaling, other tone signals and telephony events in RTP packets.

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	Title		: Guidelines for Writers of RTP Payload Format 
                          Specifications
	Author(s)	: M. Handley, C. Perkins
	Filename	: draft-ietf-avt-rtp-format-guidelines-03.txt,.ps
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	Date		: 01-Jul-99
	
This document provides general guidelines aimed at
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tions in deciding on good formats. These guidelines
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From rem-conf Fri Jul 02 07:07:25 1999 
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Folks,

We've looked at the problem of doing IP telephony over cellular links, and
will present some work on header compression over cellular links in the
AVT WG in Oslo.
 
On cellular links, bandwidth is very expensive, and it is necessary to 
compress headers to be able to compete with circuit-switched systems.
However, CRTP does not appear to work well over cellular links as we 
show in 

	http://www.ietf.org/internet-drafts/draft-degermark-crtp-cellular-00.txt
	http://www.ietf.org/internet-drafts/draft-degermark-crtp-cellular-00.ps

In fact, CRTP does not appear to deliver performance high enough to make
IP telephony to cellular phones economically feasible. To alleviate this
problem we played around with other header compression schemes and came up
with one that performs very well. It is described in 

	http://www.ietf.org/internet-drafts/draft-jonsson-robust-hc-00.txt
	http://www.ietf.org/internet-drafts/draft-jonsson-robust-hc-00.ps

Comments are welcome!

Cheers

Mikael Degermark



http://www.cdt.luth.se/~micke



From rem-conf Fri Jul 02 07:28:27 1999 
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Subject: Working group last call on generic FEC payload format
Date: Fri, 02 Jul 1999 15:26:54 +0100
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From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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This is to announce a two week working group last call on the RTP payload
format for generic forward error correction (draft-ietf-avt-fec-06.txt).
If no comments requiring substantive changes are received in the next two 
weeks (by 16th July), this draft will be submitted for publication as a 
standards track RFC.

Note that Jonathan Rosenberg will make a brief presentation on this draft
in the AVT meeting in Oslo on Wednesday 14th July.

Colin



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Thanks for your comments, =


From: Tobias =D6brink <nv91-tob@nada.kth.se>
Subject: Re: I-D ACTION:draft-ietf-avt-dv-video-00.txt
Date: Wed, 30 Jun 1999 19:11:57 +0200 (MET DST)
Message-ID: <Pine.GSO.3.95.990630185538.1098F-100000@mumrik.nada.kth.se>=


nv91-tob> =

nv91-tob> In section 4.1 RTP header usage:
nv91-tob> =

nv91-tob>    Timestamp: 32-bit 90 kHz timestamp representing the time at=
 which
nv91-tob>    the first data in the frame was sampled.  =

nv91-tob> =

nv91-tob> This statement may lead to the conclusion that the sampling ti=
me for
nv91-tob> the first frame (maybe obtained from the CIP header as
nv91-tob> described in appendix A) should be used as the initial value o=
f the
nv91-tob> timestamp field. To avoid unneccessary confusion I would like =
to add
nv91-tob> the following paragraph from the new RTP draft:
nv91-tob> =

nv91-tob>    The initial value of the timestamp SHOULD be random, as for=
 the
nv91-tob>    sequence number.
nv91-tob> =


We will mention the initial timestamp value issue in the next ID.

nv91-tob> =

nv91-tob> In section 4.2 DV data encapsulation into RTP payload:
nv91-tob> =

nv91-tob>    Therefore, if VAUX/AAUX information is not
nv91-tob>    transmitted in the stream, the equivalent parameters essent=
ial to
nv91-tob>    playout MUST be provided by some out of band means beyond t=
he
nv91-tob>    scope of this document.
nv91-tob>    :
nv91-tob>    :<snip>
nv91-tob>    :
nv91-tob>    Therefore, if the RTP receiver
nv91-tob>    is feeding the DV stream to a device that requires AAUX
nv91-tob>    information and VAUX DIF blocks, the receiver MUST be able =
to
nv91-tob>    generate AAUX within audio DIF blocks and VAUX DIF blocks f=
or the
nv91-tob>    device using the parameters provided out of band.
nv91-tob> =

nv91-tob> I interpret this as, to comply with the spec, the receiver
nv91-tob> implementation MUST retrieve, by some unknown out-of-band-mech=
anism,
nv91-tob> the information needed to recreate the VAUX and AAUX DIF block=
s. =

nv91-tob> If that's the case won't we end up with non-interoperable
nv91-tob> implementations using proprietary out-of-band-mechanisms? =

nv91-tob> Maybe we should at least state a default out-of-band-mechanism=
 as a
nv91-tob> minimal requirement that have to be used by all implementation=
s. The
nv91-tob> dynamic payload type assignment using SDP in section 5 comes t=
o mind.
nv91-tob> =


Does the comment mean you think the profile document should assume the
default out-of-band session description mechanism ?
 =

nv91-tob> =

nv91-tob> In section 5.2 SDP description for bundled stream: =

nv91-tob> =

nv91-tob> The audio format parameter seems quite detailed. How to find o=
ut the =

nv91-tob> values for those parameters if you're about to announce a sess=
ion in
nv91-tob> sdr or writing a SDP-file for a program that haven't yet start=
ed?

Do you think RTP session can be started with an insufficient session
description information or without the session description ?
Or you think the draft should describes as:

  The host must not start RTP session without the sufficient session
  information exchange.
 =

nv91-tob> Aren't there any standardized combinations of those
nv91-tob> parameters? =

nv91-tob> At least <quantization>, <sampling rate> and the number of cha=
nnels =

nv91-tob> only can occur in a few different combinations.
nv91-tob> Maybe the same applies to some of the other parameters as well=
?
nv91-tob> =


Blue book (DV specification) offers some limited combinations as
"basic standard". But, I think the payload description should be designe=
d
that will accord any combination of coding parameters.

                                              ikob@koganei.wide.ad.jp




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From rem-conf Fri Jul 02 12:26:05 1999 
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Date: Fri, 02 Jul 1999 15:18:54 -0400
To: a00khl00@nchc.gov.tw
From: John Jamison <jj@evl.uic.edu>
Subject: Re: request for Mbone feed
Cc: Markus Buchhorn <markus@acsys.anu.edu.au>,
        Jan-Ming Ho <hoho@iis.sinica.edu.tw>, rem-conf@es.net, jj@startap.net,
        Sheng-Hsiung Huang <huangk@gate.sinica.edu.tw>,
        chinfu@iis.sinica.edu.tw, Chia-Hui Wang <chwang@iis.sinica.edu.tw>,
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Ken,

The STAR TAP is planning on setting up an IPv4 service later this month and
should have a multicast service up and running latter this summer. Your
best bet is to get a multicast feed from the vBNS now and when the STAR TAP
router's multicast service is up and running you can cut over. I have
spoken to Kevin Thompson (kthomp@mci.net) about this and he is more than
happy to work with you to set up multicast peering beteen TANnet and vBNS.
Please contact Kevin via email at: kthomp@mci.net.

Regards,

JJ

At 04:39 PM 6/28/99 +0800, Kenny Huang wrote:
>Dear ALL and JJ,
>
>I met JJ at the STARTAP Advisory Committee and briefed the mbone
>issue to him already. Ken Lin at NCHC (a00khl00@nchc.gov.tw) will
>be the primary contact in Taiwan. Please contact Ken Lin for=20
>mbone implementation, and keep me in the communication loop.
>
>Regards
>
>--=20
>     Kenny Huang =B6=C0=B3=D3=B6=AF                      Tel=
 886-2-2789-9492
>     Head, Communication & Network Services  Fax 886-2-2783-6444
>     Academia Sinica                         huangk@sinica.edu.tw
>     http://www.sinica.edu.tw
>
>
>
>Markus Buchhorn wrote:
>>=20
>> At 09:19 21/06/99 +0800, Jan-Ming Ho wrote:
>> >
>> >We are representing Taiwan to seek for Mbone feed to our Academic=
 Network,
>> >Taiwan Academic Network, or TANET for short.  We had long been suffering
>> >a low bandwidth connecting to international internet traffic.  But, we
>> >recently upgrade our connection to the US by a T3 lease line.  We
>> >appreciate it very much if we could have any institute here to provide
>> >us with permanent MBONE feed.  We will run a permanent mbone root for
>> >the academic family including universities in Taiwan.   We are also
>> >willing to transit this MBONE traffic to nearby countries based on the
>> >availability of physical connections and bandwidth.
>>=20
>> Since Taiwan/TANet now has a direct connection to the STARTAP in Chicago
>> I'd suggest getting an MBONE feed from there - contact John Jamison
>> (jj@startap.net), who may or may not be the right person, (sorry John!=
 :-))
>> but should be able to point you to the right person. APAN and SingaREN=
 also
>> get their MBONE feeds off the vBNS through STARTAP.
>>=20
>> Cheers,
>>         Markus
>>=20
>> >Jan-Ming Ho, PhD
>> >Associate Editor, IEEE Transaction on Multimedia
>> >Research Fellow                        (O) 886-2-2788-3799 x1803
>> >IIS, Academis Sinica           (F) 886-2-2788-3799 x1606, =
 886-2-2782-4814
>> >Nankang, Taipei                        email:  hoho@iis.sinica.edu.tw
>> >Taiwan                         URL:   =
 http://www.iis.sinica.edu.tw/~hoho
>> >                                       http://www.iis.sinica.edu.tw/CCL
>> >                                       http://twstudy.sinica.edu.tw
>> >
>>
>=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=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=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>> =3D=3D=3D
>> >
>> >
>> >
>> >
>> >
>> Markus Buchhorn,  Advanced Computational Systems CRC     | Ph: +61 2
62798810
>> email: markus@acsys.anu.edu.au, snail: ACSys, RSISE Bldg,|Fax: +61 2
62798602
>> Australian National University, Canberra 0200, Australia |Mobile: 0417
281429
>=20




From rem-conf Fri Jul 02 12:38:03 1999 
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From: "Tamer Ragab" <ragab@wowmail.com>
To: <rem-conf@es.net>
Subject: 
Date: Fri, 2 Jul 1999 21:22:27 +0300
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Dear Sir/Madam
I have the pleasure to introduce myself
Name : Tamer M.Ragab
Present Occupation : Demonstrator at Faculty of Education,Suez Canal =
University=20
El-Arish, EGYPT
Qualification : B. Sc. in Math & Computer Science (1996) ,Science =
Faculty
Interests : Distance Learning
I kindly ask you if I could get more details about the Distance Learning =
program you offer=20
Hoping to hear from you as soon as possible
Thanking you in advance
My e-mail : Tamer00@hotmail.com



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<DIV><FONT color=3D#000000 face=3DArial size=3D2>Dear Sir/Madam<BR>I =
have the pleasure=20
to introduce myself<BR>Name : Tamer M.Ragab<BR>Present Occupation : =
Demonstrator=20
at Faculty of Education,Suez Canal University <BR>El-Arish,=20
EGYPT<BR>Qualification : B. Sc. in Math &amp; Computer Science (1996) =
,Science=20
Faculty<BR>Interests : Distance Learning<BR>I kindly ask you if I could =
get more=20
details about the Distance Learning program you offer <BR>Hoping to hear =
from=20
you as soon as possible<BR>Thanking you in advance<BR>My e-mail : <A=20
href=3D"mailto:Tamer00@hotmail.com">Tamer00@hotmail.com</A></FONT></DIV>
<DIV><FONT color=3D#000000 face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#000000 face=3DArial =
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From rem-conf Fri Jul 02 14:24:42 1999 
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Date: Fri, 2 Jul 1999 23:21:23 +0200 (MET DST)
From: =?ISO-8859-1?Q?Tobias_=D6brink?= <nv91-tob@nada.kth.se>
To: ikob <ikob@koganei.wide.ad.jp>
cc: rem-conf@es.net
Subject: Re: I-D ACTION:draft-ietf-avt-dv-video-00.txt
In-Reply-To: <19990702234942L.ikob@koganei.wide.ad.jp>
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On Fri, 2 Jul 1999, ikob wrote:

> From: Tobias =D6brink <nv91-tob@nada.kth.se>
>
> nv91-tob> In section 4.2 DV data encapsulation into RTP payload:
> nv91-tob>=20
> nv91-tob>    Therefore, if VAUX/AAUX information is not
> nv91-tob>    transmitted in the stream, the equivalent parameters essen=
tial to
> nv91-tob>    playout MUST be provided by some out of band means beyond =
the
> nv91-tob>    scope of this document.
> nv91-tob>    :
> nv91-tob>    :<snip>
> nv91-tob>    :
> nv91-tob>    Therefore, if the RTP receiver
> nv91-tob>    is feeding the DV stream to a device that requires AAUX
> nv91-tob>    information and VAUX DIF blocks, the receiver MUST be able=
 to
> nv91-tob>    generate AAUX within audio DIF blocks and VAUX DIF blocks =
for the
> nv91-tob>    device using the parameters provided out of band.
> nv91-tob>=20
> nv91-tob> I interpret this as, to comply with the spec, the receiver
> nv91-tob> implementation MUST retrieve, by some unknown out-of-band-mec=
hanism,
> nv91-tob> the information needed to recreate the VAUX and AAUX DIF bloc=
ks.=20
> nv91-tob> If that's the case won't we end up with non-interoperable
> nv91-tob> implementations using proprietary out-of-band-mechanisms?=20
> nv91-tob> Maybe we should at least state a default out-of-band-mechanis=
m as a
> nv91-tob> minimal requirement that have to be used by all implementatio=
ns. The
> nv91-tob> dynamic payload type assignment using SDP in section 5 comes =
to mind.
> nv91-tob>=20
>=20
> Does the comment mean you think the profile document should assume the
> default out-of-band session description mechanism ?

I don't want to change anything in the RTP audio/video profile, but I
think there should be a default out-of-band mechanism specified in=20
the payload specification document. According to the RTP audio/video
profile there are two main mechanisms for dynamic payload type
assignment, so I agree it's not a good idea to force all DV
payload implementations to use one of them.=20

> nv91-tob>=20
> nv91-tob> In section 5.2 SDP description for bundled stream:=20
> nv91-tob>=20
> nv91-tob> The audio format parameter seems quite detailed. How to
> nv91-tob> find out the values for those parameters if you're about
> nv91-tob> to announce a session in sdr or writing a SDP-file for a
> nv91-tob> program that haven't yet started?
>=20
> Do you think RTP session can be started with an insufficient session
> description information or without the session description ?
> Or you think the draft should describes as:
>=20
>   The host must not start RTP session without the sufficient session
>   information exchange.

Only in the case "when the RTP receiver is feeding the DV stream to a
device that requires AAUX information and VAUX DIF blocks, because=20
then the receiver MUST be able to generate AAUX within audio DIF
blocks and VAUX DIF blocks for the device. I don't know how to do
that without getting all the parameters (that I assume is a part of
the session information).

Since there's a MUST in there, I'd also like to have a default
out-of-band mechanism to turn to to get those parameters from, but
it's not necessary ofcourse.

In other cases I see no problem to start a RTP session complying with
the current draft RTP payload specification for DV format video,
without sufficient session information (i.e. relying on default
parameters if we can agree on a set of them).

Also, if I know that the sender doesn't omit VAUX and AAUX
information I don't need to generate AAUX and VAUX and I won't
need any out-of-band mechanism other than for the usual RTP/UDP/IP
parameters (address, port, payload type, ttl,...).

> nv91-tob> Aren't there any standardized combinations of those
> nv91-tob> parameters?=20
> nv91-tob> At least <quantization>, <sampling rate> and the number of ch=
annels=20
> nv91-tob> only can occur in a few different combinations.
> nv91-tob> Maybe the same applies to some of the other parameters as wel=
l?
>=20
> Blue book (DV specification) offers some limited combinations as
> "basic standard". But, I think the payload description should be design=
ed
> that will accord any combination of coding parameters.

I respect that. Does those "basic standard" combinations include
values on language, audio mode et.c.?=20

/Tobias
.........................................................................=
...
Tobias =D6brink                 Grad. Stud. Dept. of Teleinformatics, KTH
Phone numbers:				Paper mail address:
	+46 8 752 14 75 (IT, work)		KTH-ELECTRUM/204
        +46 8 751 17 93 (Fax)			S-164 40 Kista
						Sweden
E-mail: tobias@it.kth.se	URL: http://www.it.kth.se/~nv91-tob/
See if I'm logged in: http://www.it.kth.se/~nv91-tob/notice.html




From rem-conf Fri Jul 02 15:39:18 1999 
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Date: Fri, 02 Jul 1999 18:45:31 -0700
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Dear Colleagues:

Please find attached the 2nd CFP for the 6th IEEE*=20
International Workshop on Mobile Multimedia=20
Communications (MOMUC99), November 15-17, 1999

        http://cwc.ucsd.edu/momuc99/

Please note that the submission deadline is fast
approaching:

        Full papers are due *July 25*, 1999
        Position papers are due: September 1, 1999

MOMUC99 will be held in San Diego, a city with=20
vibrant wireless industries; it follows the very successful=20
and interactive workshops held in Tokyo '93, Bristol '95,=20
Princeton '96, Seoul'97, and Berlin' 98. =20

Join us for this exiting event!

--=20
Andrew
http://comet.columbia.edu/~campbell



                        2nd CALL FOR PAPERS AND DEMOS

   6th IEEE* International Workshop on Mobile Multimedia Communications=20
               =20
                                MOMUC99

                      http://cwc.ucsd.edu/momuc99/

                          November 15-17, 1999
                    =20
                          Hyatt Islandia on=20
                       San Diego's Mission Bay
                         San Diego, California

                 Sponsored by IEEE Communications Society*=20
               Technical Committees on Computer Communications=20
                        Multimedia Communications=20
                  Internet and Personal Communications=20

                        In co-operation with the=20
                      ACM SIGCOMM*, ACM SIGMOBILE*=20
                     Center for Wireless Communications,=20
                  University of California, San Diego and the=20
                     Center for Telecommunications Research,=20
                          Columbia University
               =20
Objectives

The Sixth IEEE* International Workshop on Mobile Multimedia=20
Communications (MOMUC'99)  will provide an international forum=20
for the discussion of research in wireless and mobile multimedia=20
communications. The workshop, to be held in San Diego, a city with=20
vibrant wireless industries, will bring together researchers,=20
developers and practitioners working in all facets of advanced=20
mobile multimedia communications. The scope of MOMUC'99 will=20
include broadband wireless networking for data and multimedia,=20
mobile multimedia systems and applications together with associated=20
mobile computing devices, video processing and enabling software=20
technologies.=20

In the past the workshop has been cross-disciplinary, small and=20
well focused, consisting of keynote speakers, panels and solicited
contributions with the emphasis on innovation. As a result, a=20
considerable amount of time is devoted to informal discussion. The
Sixth IEEE International Workshop on Mobile Multimedia=20
Communications will be limited to 150 participants.=20

Previous events include: Tokyo '93, Bristol '95, Princeton '96, Seoul
'97, and Berlin' 98.=20

Best student paper prize of $500 from=20
C&C Research Laboratories, NEC USA=20

Scope

Authors are invited to submit previously unpublished full research=20
papers, short position papers and outlines for demonstration prototypes.=20
Suggested focus areas include:=20

        =B7 High-speed wireless modem and radio techniques=20
        =B7 Medium access/data link control=20
        =B7 Handoff signaling systems=20
        =B7 Power management issues=20
        =B7 Wireless network management & service control=20
        =B7 Signal processing techniques=20
        =B7 Software radios=20
        =B7 Next generation mobile Internet=20
        =B7 Multimedia satellite communications=20
        =B7 Mobile multimedia ad hoc networks=20
        =B7 Advances in cellular systems=20
        =B7 Piconet technologies
        =B7 Quality of Service in wireless and mobile networks
        =B7 Experimental systems=20
        =B7 Mobile computing devices=20
        =B7 Internetworking of wired and wireless networks
        =B7 Mobile multimedia applications and platforms
        =B7 Distributed systems and mobile multimedia communications
        =B7 Adaptive mobile systems
        =B7 Enabling software technologies for mobile systems=20
        =B7 Programmable and active mobile networks=20
        =B7 Wireless transport services and protocols=20
        =B7 Video compression, including MPEG-4=20
        =B7 Mobile agent technology

Submissions

We request authors to submit their draft papers in electronic=20
format. Accepted full papers and position papers will=20
be published by IEEE as workshop proceedings.=20

Full papers due: July 25, 1999.=20
Position papers due: September 1, 1999.=20
Notification of acceptance: September 13, 1999.=20
Camera Ready Due: October 1, 1999=20

Full papers are limited to 25 pages=20
(11 pt. fonts and double space).
Position papers are limited to 3 pages=20
(11 pt. fonts and single space).=20

If electronic submission is not possible, please follow the=20
instructions at the end of this page. =20

The submission procedure requires that (a) you send your=20
complete draft paper by ftp and (b) you send the cover page=20
information accompanying your submission by e-mail to=20

        momuc99@comet.columbia.edu.=20

Please read all instructions on this page before=20
attempting a submission. =20

-Draft Paper Format

Your paper must consist of one single file,=20
including the figures. When preparing the document,
please use common fonts and, if possible,=20
US letter page format to facilitate the printing.  =20

The following file formats are being accepted: Postscript (PS),=20
Portable Document Format (PDF), and  MS Word.=20
Compression of the postscript files using the compress=20
or gzip commands is advisable.  =20

-Draft Paper Naming Scheme =20

When transferring the file to our host use the=20
following naming scheme:  =20

<last name of managing author>_<first name>.<HHMM>.<file format>[.gz |
.Z]=20

Examples:  =20

Smith_Adam.1205.ps    (Postscript file transferred at 12:05 your time) =20
Smith_Adam.2012.ps.gz (gzip-compressed Postscript file transferred at
20:12) =20
Smith_Adam.0747.pdf   (PDF file transferred at 7:47 your time) =20
Smith_Adam.0841.doc   (MS Word file transferred at 8:41 your time) =20

-Submitting the Draft Paper=20

Transfer your draft paper to our server=20
(ftp.comet.columbia.edu) using anonymous ftp as follows:  =20

        1.ftp ftp.comet.columbia.edu=20
        2.enter anonymous as user name=20
        3.enter <your email address> as password=20
        4.cd pub/momuc/incoming=20
        5.binary=20
        6.put <file name>=20
        7.quit=20

Note that the directory pub/momuc/incoming is write only=20
(i.e., it cannot be listed).  =20

-Preparing the Cover Page=20

Prepare a cover page in plain text (ASCII) containing=20
he following information:  =20

 Managing author's name, affiliation, postal=20
   address, phone number, fax number, e-mail address=20
 Paper Title=20
 Name and affiliation of author(s)=20
 Keywords=20
 Format of the submitted paper (Postscript, PDF, etc.)=20
 Statement, if you would like to participate in the=20
   contention for the best student paper award.=20

-Submitting the Cover Page =20

Send the cover page accompanying  your submission by e-mail to=20
momuc99@comet.columbia.edu. =20

- If Electronic Submission is not Possible =20

In case electronic submission is not possible,=20
authors should send six copies of their draft paper to : =20

   Andrew T. Campbell
   MOMUC99=20
   Department of Electrical Engineering
   and Center for Telecommunications Research=20
   1312 Seeley W. Mudd Bldg.
   Columbia University, New York, NY 10025-6699, USA=20
   campbell@comet.columbia.edu

Workshop Committee

Steering Committee:
D. Goodman (Rutgers University)
D. Raychaudhuri (NEC, Princeton)
H. Tominaga (Waseda University Tokyo)
=20
General Chair:         =20
Anthony Acampora (UC San Diego)

Program Co-Chairs:
Andrew T. Campbell (Columbia University)
George C. Polyzos  (UC San Diego)

Program Committee:
Yoshihiko Akaiwa  (Kyushu University)
B. R. Badrinath (Rutgers)
Victor Bahl (Microsoft Research)
Hari Balakrishnan (MIT)
Vaduvur Bharghavan (UIUC)
Giuseppe Bianchi (Palermo University)
Kalyan Basu (Nortel Networks)
Ernst Bonek (TU Wien)
Shih-Fu Chang (Columbia University)
Franco Davoli (U. di Genoa)
Julio Escobar, (Senacyt)
James Freebersyser (ONR)
Mario Gerla (UCLA)=20
Zygmunt Haas (Cornell University)
Jon Inoyue (Intel)
Rajeev Jain (UCLA)
Anthony Joseph (UC Berkeley)
Gunnar Karlsson (Royal Inst. of Tech.)
Mark Karol (Lucent)
K. S. Kwak (University Seoul)
Khiem Le  (Nokia)
Petri Mahonen (VTT)
Gerald Maguire (KTH Stockholm)
Maximilian Ott (NEC)
Steve Pink (Lulea University)
Tom La Porta (Lucent)
Albert Stienstra (Phillips)
Masahiro Umehira (NTT)
Adam Wolisz (TU Berlin)

Local Organization Committee:
Michael Kounavis and Javier Gomez, Columbia University
Carol K. Tohsaku, San Diego State University

Correspondence

momuc99@comet.columbia.edu
=20
Location

IEEE MOMUC99 will be held in the Hyatt Islandia on San Diego's Mission
Bay

(*pending)



From rem-conf Fri Jul 02 20:12:20 1999 
From rem-conf-request@es.net Fri Jul 02 20:12:19 1999
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Date: Fri, 2 Jul 1999 20:07:32 -0700 (Pacific Daylight Time)
From: Stephen Casner <casner@cisco.com>
To: rem-conf@es.net
Subject: Last call on draft-ietf-avt-rtp-format-guidelines-03.txt
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The the AVT working group:

In April, after the Minneapolis meeting, there was a two-week working
group last call on draft-ietf-avt-rtp-format-guidelines-01.txt for
publication as Best Current Practice.  No comments were received, but
the draft did not include the required Security Considerations
section.  The draft has been revised to include that:

	Title		: Guidelines for Writers of RTP Payload Format 
                          Specifications
	Author(s)	: M. Handley, C. Perkins
	Filename	: draft-ietf-avt-rtp-format-guidelines-03.txt,.ps
	Pages		: 9
	Date		: 01-Jul-99

(Not sure what happened to -02, and the .ps wasn't there when I
looked, but never mind.)

To give time for comments on the Security Considerations section, the
working group last call will continue until our meeting in Oslo on
July 14.  If there are no objections, the request for IESG Last Call
for publication as BCP will be issued from Oslo.
							-- Steve




From rem-conf Sat Jul 03 18:56:05 1999 
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Date: Sat, 03 Jul 1999 20:10:23 -0500
To: "MMC7@IRMA.Anchorage":;
From: Mahbubur Syed <msyed@badlands.nodak.edu>
Subject: Multimedia Computing and Information Management - CFP
Mime-Version: 1.0
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--=====================_931068623==_
Content-Type: text/plain; charset="us-ascii"

Dear Colleague:

My apologies if you received this email more than once.

Please find attached a call for paper. 

I am also attching a pdf version of the CFP.

It will be highly appreciated, if you take the trouble also to circulate
among interested faculty and your network.

Best Regards.

Thanks.


Syed M Rahman


*****CALL FOR PAPERS************

MULTIMEDIA COMPUTING (MMC) track
IRMA 2000 International Conference
May 21-24, 2000
at Anchorage Hilton Hotel Anchorage, Alaska, USA

***TRACK CHAIR*** 
Syed Mahbubur Rahman
Department of Electrical and Computer Engineering, 
North Dakota State University, Fargo, USA
(On secondment from Gippsland School of Computing and Information
Technology, 
Monash University, Australia.)
Email: msyed@badlands.nodak.edu, syed.rahman@infotech.monash.edu.au

***TRACK URL***
For details please visit the following URL:
http://www.ece.ndsu.nodak.edu/~msyed/irmamm/index.html

***IMPORTANT DATES***
Submission Deadline                   October 8,   1999
Notification of Acceptance/Rejection  November 19, 1999
Final Submission Due		           January 14,  2000
Early Registration Ends	           April 3,     2000

***TOPICS***
Multimedia technology has the potential to evolve the paradigm of end
user computing, from the interactive text and graphics model that has
developed since 1950s, into one more compatible with the digital
electronic world of the next century. This track will present papers
and panels designed to help professionals gain an understanding and
perspective on MMC.

The Multimedia Computing Track encourages the submission of quality
papers and panel workshop proposals dealing with (but not limited to)
the following topics:

* Multimedia-specific technologies (e.g., DVI, CD-I, JPEG, MPEG)
* Multimedia platforms and peripherals
* Multimedia and communications (e.g., WAN, LAN, ISDN, fibre optics)
* Graphics, audio, and video hardware and software technologies
* Operating systems
* Authoring tools, languages
* Design and development (software engineering, CASE, multi/cross
platform consideration)
* Media preparation and integration
* Research issues (interface design, end user performance, etc.)
* MMC and information systems
* Training and education issues related to MMC
* MMC applications (e.g., commercial, home, entertainment, etc.)
* Issues concerned with legal, copyright, standard matters
* Related emerging technologies (e.g., HDTV, parallel processing,
virtual systems, hypermedia)
* Global issues (e.g., development, distribution, manufacturing, etc.)

***SUBMISSION CATEGORIES***
# Full Length Submissions
# Reaseach-in-progress Submission
# Panel, Workshop, Tutorial and symposium submissions

Please visit Track URL for more detail information

***SUBMISSION GUIDELINES***
# Manuscript must not exceed 4000 words for full length submissions;
500-1000 words for research-in-progress summary/proposal; and
1000-2000 words for panel, workshop, tutorial and symposium proposals.
# Must be accompanied by a separate cover letter with the title
'SUBMISSION FOR MULTIMEDIA TRACK', EVERY AUTHOR(S) NAME, ADDRESS,
TEL. AND FAX NUMBERS, E-MAIL AND FULL AFFILIATION. All correspondence
will be sent to the first author unless otherwise specified.
# Submitted papers must not currently be under review by any other
publication or conference.
# Submitters must provide their e-mail address where the
acknowledgment will be forwarded.
# The number of submissions by an author (including joint authorship)
is limited to a maximum of two submissions.
# All submissions should be submitted electronically in either MS
Word '97 or rich text format (RTF). (Note: If you do not receive an
e-mail acknowledgment one week after your electronic submissions,
please contact the program chair.

submissions (indicate MULTIMEDIA COMPUTING Track in your submission) to:

Mehdi Khosrowpour, Program Chair
2000 IRMA INTERNATIONAL CONFERENCE
1331 E. Chocolate Avenue, Hershey, 
PA 17033-1117, U.S.A.
Tel: (717) 533-8879 * Fax: (717) 533-8661  
Email: mehdi@irma-international.org


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AAAAAAAAAAAABJEAADIAAAAOAAAADgAAAAAAAAAAAAAAFgABAP///////////////wAAAAAAAAAA
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//////////////////////////////////////////////////////////////////////8BAP7/
AwoAAP////8GCQIAAAAAAMAAAAAAAABGGAAAAE1pY3Jvc29mdCBXb3JkIERvY3VtZW50AAoAAABN
U1dvcmREb2MAEAAAAFdvcmQuRG9jdW1lbnQuOAD0ObJxAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA
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--=====================_931068623==_--




From rem-conf Tue Jul 06 01:08:31 1999 
From rem-conf-request@es.net Tue Jul 06 01:08:30 1999
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Subject: CERN MBone Announcement: Next LHCC
To: rem-conf@es.net (List Mbone-WW)
Date: Tue, 6 Jul 1999 09:47:44 +0200 (MET DST)
From: "Daniel Davids CERN/IT" <Daniel.Davids@cern.ch>
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	CERN is pleased to announce the MBONE broadcast of the

		    LARGE HADRON COLLIDER COMMITTEE
		    ===============================
			 Thursday 8 July 1999



Place: CERN Council Chamber


*** Times are UTC+2 ***


09.00-10.00     ALICE Inner Tracking System TDR


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

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




From rem-conf Tue Jul 06 04:22:59 1999 
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Reply-to: Internet-Drafts@ietf.org
Subject: I-D ACTION:draft-ietf-avt-rtp-mime-00.txt
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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		: MIME Type Registration of RTP Payload Formats
	Author(s)	: S. Casner, P. Hoschka
	Filename	: draft-ietf-avt-rtp-mime-00.txt
	Pages		: 17
	Date		: 02-Jul-99
	
This document defines the procedure to register RTP Payload Formats
as audio or video MIME subtype names.  This is useful in a text-
based format or control protocol to identify the type of an RTP
transmission.  This document also registers all the RTP payload
formats defined in the RTP Profile for Audio and Video Conferences
as MIME subtypes.  Some of these may also be used for transfer
modes other than RTP.

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

Internet-Drafts are also available by anonymous FTP. Login with the username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-ietf-avt-rtp-mime-00.txt".

A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html 
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
	mailserv@ietf.org.
In the body type:
	"FILE /internet-drafts/draft-ietf-avt-rtp-mime-00.txt".
	
NOTE:	The mail server at ietf.org can return the document in
	MIME-encoded form by using the "mpack" utility.  To use this
	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
	a MIME-compliant mail reader.  Different MIME-compliant mail readers
	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
	up into multiple messages), so check your local documentation on
	how to manipulate these messages.
		
		
Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.

--NextPart
Content-Type: Multipart/Alternative; Boundary="OtherAccess"

--OtherAccess
Content-Type: Message/External-body;
	access-type="mail-server";
	server="mailserv@ietf.org"

Content-Type: text/plain
Content-ID:	<19990702133034.I-D@ietf.org>

ENCODING mime
FILE /internet-drafts/draft-ietf-avt-rtp-mime-00.txt

--OtherAccess
Content-Type: Message/External-body;
	name="draft-ietf-avt-rtp-mime-00.txt";
	site="ftp.ietf.org";
	access-type="anon-ftp";
	directory="internet-drafts"

Content-Type: text/plain
Content-ID:	<19990702133034.I-D@ietf.org>

--OtherAccess--

--NextPart--





From rem-conf Tue Jul 06 04:29:30 1999 
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From: Internet-Drafts@ietf.org
Reply-to: Internet-Drafts@ietf.org
Subject: I-D ACTION:draft-ietf-avt-rtcp-bw-00.txt
Date: Tue, 06 Jul 1999 07:28:01 -0400
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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		: SDP Bandwidth Modifiers for RTCP Bandwidth
	Author(s)	: S. Casner
	Filename	: draft-ietf-avt-rtcp-bw-00.txt
	Pages		: 6
	Date		: 02-Jul-99
	
This document defines an extension to the Session Description Pro-
tocol (SDP) to specify two additional modifiers for the bandwidth
attribute.  These modifiers may be used to specify the bandwidth
allowed for RTCP packets in a Real-Time Transport Protocol (RTP)
session.

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

Internet-Drafts are also available by anonymous FTP. Login with the username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-ietf-avt-rtcp-bw-00.txt".

A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html 
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
	mailserv@ietf.org.
In the body type:
	"FILE /internet-drafts/draft-ietf-avt-rtcp-bw-00.txt".
	
NOTE:	The mail server at ietf.org can return the document in
	MIME-encoded form by using the "mpack" utility.  To use this
	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
	a MIME-compliant mail reader.  Different MIME-compliant mail readers
	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
	up into multiple messages), so check your local documentation on
	how to manipulate these messages.
		
		
Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.

--NextPart
Content-Type: Multipart/Alternative; Boundary="OtherAccess"

--OtherAccess
Content-Type: Message/External-body;
	access-type="mail-server";
	server="mailserv@ietf.org"

Content-Type: text/plain
Content-ID:	<19990702133142.I-D@ietf.org>

ENCODING mime
FILE /internet-drafts/draft-ietf-avt-rtcp-bw-00.txt

--OtherAccess
Content-Type: Message/External-body;
	name="draft-ietf-avt-rtcp-bw-00.txt";
	site="ftp.ietf.org";
	access-type="anon-ftp";
	directory="internet-drafts"

Content-Type: text/plain
Content-ID:	<19990702133142.I-D@ietf.org>

--OtherAccess--

--NextPart--





From rem-conf Tue Jul 06 04:29:31 1999 
From rem-conf-request@es.net Tue Jul 06 04:29:31 1999
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To: IETF-Announce: ;
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From: Internet-Drafts@ietf.org
Reply-to: Internet-Drafts@ietf.org
Subject: I-D ACTION:draft-ietf-avt-rtp-new-04.txt,.ps
Date: Tue, 06 Jul 1999 07:27:55 -0400
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--NextPart

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

	Title		: RTP: A Transport Protocol for Real-Time Applications
	Author(s)	: H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson
	Filename	: draft-ietf-avt-rtp-new-04.txt,.ps
	Pages		: 100
	Date		: 02-Jul-99
	
This memorandum is a revision of RFC 1889 in preparation for
advancement from Proposed Standard to Draft Standard status. Readers
are encouraged to use the PostScript form of this draft to see where
changes from RFC 1889 are marked by change bars.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-new-04.txt

Internet-Drafts are also available by anonymous FTP. Login with the username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-ietf-avt-rtp-new-04.txt".

A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html 
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
	mailserv@ietf.org.
In the body type:
	"FILE /internet-drafts/draft-ietf-avt-rtp-new-04.txt".
	
NOTE:	The mail server at ietf.org can return the document in
	MIME-encoded form by using the "mpack" utility.  To use this
	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
	a MIME-compliant mail reader.  Different MIME-compliant mail readers
	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
	up into multiple messages), so check your local documentation on
	how to manipulate these messages.
		
		
Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.

--NextPart
Content-Type: Multipart/Alternative; Boundary="OtherAccess"

--OtherAccess
Content-Type: Message/External-body;
	access-type="mail-server";
	server="mailserv@ietf.org"

Content-Type: text/plain
Content-ID:	<19990702133126.I-D@ietf.org>

ENCODING mime
FILE /internet-drafts/draft-ietf-avt-rtp-new-04.txt

--OtherAccess
Content-Type: Message/External-body;
	name="draft-ietf-avt-rtp-new-04.txt";
	site="ftp.ietf.org";
	access-type="anon-ftp";
	directory="internet-drafts"

Content-Type: text/plain
Content-ID:	<19990702133126.I-D@ietf.org>

--OtherAccess--

--NextPart--





From rem-conf Tue Jul 06 04:29:31 1999 
From rem-conf-request@es.net Tue Jul 06 04:29:31 1999
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To: IETF-Announce: ;
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From: Internet-Drafts@ietf.org
Reply-to: Internet-Drafts@ietf.org
Subject: I-D ACTION:draft-ietf-avt-profile-new-06.txt,.ps
Date: Tue, 06 Jul 1999 07:27:49 -0400
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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 Profile for Audio and Video Conferences
			  with Minimal Control
	Author(s)	: H. Schulzrinne
	Filename	: draft-ietf-avt-profile-new-06.txt,.ps
	Pages		: 37
	Date		: 02-Jul-99
	
This memorandum is a revision of RFC 1890 in preparation
for advancement from Proposed Standard to Draft Standard
status. Readers are encouraged to use the PostScript form
of this draft to see where changes from RFC 1890 are
marked by change bars.
This document describes a profile called 'RTP/AVP' for
the use of the real-time transport protocol (RTP),
version 2, and the associated control protocol, RTCP,
within audio and video multiparticipant conferences with
minimal control. It provides interpretations of generic
fields within the RTP specification suitable for audio
and video conferences. In particular, this document
defines a set of default mappings from payload type
numbers to encodings.
This document also describes how audio and video data may
be carried within RTP. It defines a set of standard
encodings and their names when used within RTP. The
descriptions provide pointers to reference
implementations and the detailed standards. This document
is meant as an aid for implementors of audio, video and
other real-time multimedia applications.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avt-profile-new-06.txt

Internet-Drafts are also available by anonymous FTP. Login with the username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-ietf-avt-profile-new-06.txt".

A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html 
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
	mailserv@ietf.org.
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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           : Multiplexing Scheme for RTP Flows between
			  Access Routers
	Author(s)	: K. El-Khatib, G. Luo, G. Bochmann
	Filename	: draft-ietf-avt-multiplexing-rtp-00.txt
	Pages		: 12
	Date		: 02-Jul-99
	
This draft proposes a light-weight data driven approach for 
multiplexing low bit rate RTP streams at the edge router of the 
Internet in order to reduce the RTP/UDP/IP header overhead associated 
with each RTP stream. Audio packets from different sources in a local 
access network destined to different users in the same remote access 
network are multiplexed into one packet, with the original RTP/UDP/IP 
header of each packet replaced with a mini-header (2 bytes), resulting 
in a reduction of the overhead. The de-multiplexing capability of the 
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Subject: inband dtmf/tones proposal
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I'm still trying to get a handle on this new dtmf/tones proposal.  It is
only the redundancy issue that is perplexing.  There appears to be too much
freedom in the spec to guarantee that a receiver can unravel what is
missing and apply the redundancy to fill it in.
 If there is only one train of sequence numbers then things are probably
okay, but the spec seems to allow more than one train (that is one for
voice and another for dtmf).  If there will be synchronization between
voice packets and dtmf/tone packets so that they use a common period, then
again, there appears to be no problem, but, again, the spec seems to allow
different periods (should instead of shall).  (The redundancy proposal
suffers from additional problems such as ignoring the different look-ahead
specs of the various vocoders but that is a separate issue.)  The spec is
also not clear on whether voice and dtmf/tone can overlap in time.  If this
happens, what is done with the sequence numbers and how is a receiver to
know what has been missed and fill it in?
It appears that voice redundancy does not have to be used in conjunction
with dtmf/tone redundancy.  That is to say, only dtmf/tone packets would
incorporate redundancy and voice packets would not be required to do so.
Is this your intention?
Are all of these sorts of parameters (number of sequence number trains for
example) to be negotiated prior to the start of communication?

Any thoughts on these issues would be appreciated because we are
considering architecting our products to be consistent with these
proposals.





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From: pv2000 <pv2000@diee.unica.it>
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=====================================================================
                    PRELIMINARY CALL FOR PAPERS
======================================================================


                         PACKET VIDEO 2000

           The 10th International Packet Video Workshop




Date: 1-2 May 2000
Venue: Forte Village Resort, Cagliari, Italy
Web site: http://www.diee.unica.it/pv2000/

Organization:
Francesco G.B. De Natale and Daniele D. Giusto, University of Cagliari

Technical sponsors: EURASIP, IEEE Signal Processing Society (IMDSP and MMSP
Technical Committees) 

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

The tenth edition of the International Packet Video Workshop (PV 2000) will
be held at the Forte Village Resort, Cagliari, Italy.
The workshop is devoted to present technological advancements and
innovations in video communications over packet networks, in particular,
the Internet. Packet Video Workshops have been unique in providing a common
ground for people from video coding and networking fields. Presentations on
theory and practice, standards activities, and business and consumer
applicatons are encouraged.
We cordially invite you to take part in this workshop by submitting your
work and look forward to seeing you in Sardinia in May 2000 for what will
be a most rewarding and exciting experience!  

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

TECHNICAL PROGRAM

Chairs:
Leonardo Chiariglione, CSELT, Turin
Maurizio Decina, Cefriel and Polytechnic of Milan

The technical program of Packet Video 2000 will consist of invited talks,
submitted paper presentations, poster sessions and demos

Topics of interest include, but are not limited to:

Video processing and transmission
Video streaming over the Internet
Network adaptive video coding and transport
Packetized video for home LANs
Packetized video for wireless/mobile systems
Layered coding for error resilience and heterogeneous networks
Packet loss resilient coding and transport 
Terminal and server architectures for Internet TV
Efficient transcoding for heterogeneous networks 
Congestion control
Error concealment
Pre and post-processing for picture quality enhancement
Statistical multiplexing for greater network and terminal utilization
Traffic shaping for efficient network and terminal utilization 
Interstream synchronization for multiple video presentations 
Performance modeling and evaluation
Picture quality evaluation
Rate control for VBR video
International Standards: MPEG-4, MPEG-7, H.263, RTP, RTSP, SIP, SDP
Multicasting, MBONE applications
Implementations and commercial applications

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

Best Paper Award

The author of the best paper will receive a $250 prize and a diploma
suitable for framing.

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

Please submit an electronic manuscript written in HTML, not exceeding 7
Mbytes and 10 printed pages. We will produce a CD-ROM containing all the
accepted papers. Electronic components accompanying your submission are
strongly encouraged.

Submit your work in ONE of the following forms: 

1) a set of HTML files organized in a single directory OR
2) as a URL (you must have the rights to all the material there,
and guarantee stability of the files until the conference).

to:

pv2000@diee.unica.it

Detailed instructions for authors and help with manuscript preparation with
HTML can be found on the conference web page.

Deadlines:

paper submission: December 15, 1999, 
notification of acceptance: February 29, 2000
final paper delivery: March 31, 2000

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

Publication and local arrangements

Luigi Atzori
Dept. of Electrical and Electronic Engineering
University of Cagliari
Piazza d'Armi, 19
09123 Cagliari, Italy
gigi@diee.unica.it




From rem-conf Wed Jul 07 15:14:09 1999 
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From: tgl@mtgbcs.ho.lucent.com (Terry G Lyons)
To: rem-conf@es.net
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Subject: Re: draft-ietf-avt-tones-00.txt
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There are inconsistencies and ambiguities in draft-ietf-avt-tones-00.txt
that should be repaired.  They show up in a consideration of DTMF events,
MIME type "audio/dtmf", sections 1 to 2.7, on pages 1-9.  Apologies for
not having scrutinized the rest of the draft carefully.


#1.  Delete section 2.6 "Compact Reliability Scheme"

This reads like an alternative that was considered but eventually rejected,
"A more compact representation could be achieved ...".  It is the payload
format of 2.3, not 2.6, that is used in the examples of sections 2.5 and 5.

This particular text -- "... a fixed sampling rate, so choosing a value that
depends on frame interval of the surrounding codec is not recommended" --
contradicts what is said in section 2.1, 3rd paragraph -- "SHOULD use the
same sequence number and time-stamp base as the regular audio channel".
(Mention of duration and offset covering 8 seconds shows that 2.6 is really
talking about time-stamp units, not sampling rates).

It is time to remove section 2.6 from the draft. 


#2.  Change "digit" to "event" in section 2.5 figure and general text

"event" is how the field is labeled in the figures of sections 2.3 and 5.
"event" is also used in the headings and titles of Tables 1 and 3-6.
Uses of "digit" in the text should mostly be generalized to "event".


#3.  Ambiguity about the end of a DTMF event

This is a long topic.  How well can a receiver reproduce the tones that
were detected at the transmitter and converted to DTMF payload packets?

Pauses between tones are significant.  It is important not only to start
playing out tones accurately but also to end each tone punctually.  As it
stands now there is no explicit indication of the end of an event and the
beginning of a pause.  This must be inferred from two packets in sequence
(modulo packet losses), in ways that are not documented in the draft, hence
subject to inconsistent interpretations and possible misoperation.

Why does it matter?  See section 1, 3rd paragraph, concerning the "RTP trunk"
application:  "This is particularly relevant where duration and timing matter,
as in the carriage of DTMF signals."  And the last paragraph of section 2.3:
"Some applications, such as gateways into the GSTN, care about both delays
and digit duration."

The ambiguity occurs after the end of one event and before the next event
begins (since "pause" is not considered an event in Table 1).  Section 2.3
next to last paragraph says: "If there has been no new digit in the last
interval, the digit SHOULD be retransmitted three times (or until the next
event is recognized) to ensure some measure of reliability for the last event."
[Note: RE-transmitted 3 times means transmitted a total of 4 times, right?]

The 3 repetitions of one event have the SAME time-stamp and duration values.
All that differs is the packet sequence number (adjusted for any audio packets
being redundantly transmitted in the same stream, per section 2.1, 6th para;
such audio packets would have advancing time-stamps, although the interleaved
DTMF packets would be stuck at the time-stamp when the last event began).

Receiving only the first of the 4 copies, can it be concluded that an event
has ended?  - Not if its duration extends all the way to the packet interval
boundary.  - Nor if the receiver is uncertain about the interval, which is
left rather vague by section 2.3's "... SHOULD start transmitting event packets
as soon as it recognizes an event and every 50 ms thereafter or the packet
interval for the audio codec used for this session, if known. (Precise
spacing between event packets is not necessary.)"  - Nor if a new event may
begin before the interval expires, because then a transmitter not using the
redundancy mechanism of RFC 2198 would have to send out two packets quickly
in one interval (separately: end-event1, start-event2); this also makes the
time of end-event1 indeterminate.

Taking full advantage of 4 transmissions for redundancy would require a
receiver to add extra delay equal to 3 intervals at the end of each event.
To keep durations accurate would then require equal delay to be inserted
at the start of each event.  There is also an impact on synchronizing to
the decoding and playout of audio, if that matters.

PROPOSAL:

    A more robust design would give receivers an explicit indication in
    a single packet that the event within it had ended.  This can be done by
    setting the marker bit of the RTP header to 1 at event end, otherwise 0.
    (And keeping the marker set to 1 for the 3 retransmissions.)

As RFC 1889 section 5.1 says, the use of the marker bit is for each profile
to define.  Tones is a large enough topic to rank as a profile.


#4.  State clearly that section 2.4 reliability is optional

It now begins:  "To achieve reliability even when the network loses packets,
the audio redundancy mechanism described in RFC 2198 [6] is used."  This should
be changed to "MAY BE used".

If, on the contrary, RFC 2198 were always to be required, why bother to mention
the alternative reliability mechanism of 3 retransmissions in section 2.3?


- terry.g.lyons@lucent.com  +1 732 817-6016




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To: rem-conf@es.net
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The new version, on the road to Draft Standard, makes an unnecessary
change in the time-stamp unit for the audio encoding G.722, from 8000
to 16000 Hz.  This is not listed in section 9 "Changes from RFC 1890".
The change is inconsistent with prior implementations that comply with
RFC 1890, such as Table B-2/H.225.0(1998) part of the H.323 suite.

PROPOSAL:  Restore the clock rate of G.722 to 8000 Hz as it was.

The raw sampling rate at a microphone is a distinct concept from the
time-stamp unit for encoded audio.  The two need not be the same, as
G.722 demonstrates.  Its encoder spits out one octet every 1/8000 sec,
which cannot be further subdivided in time.

---------------------------------------------------
Quoted from Request for Comments: 1890
    RTP Profile for Audio and Video Conferences with Minimal Control

6.  Payload Type Definitions

   Table 2 defines this profile's static payload type values for the PT
   field of the RTP data header. ...

      PT         encoding      audio/video    clock rate    channels
                 name          (A/V)          (Hz)          (audio)
      _______________________________________________________________
      9          G722          A              8000          1

   Table 2: Payload types (PT) for standard audio and video encodings

---------------------------------------------------
Quoted from draft-ietf-avt-profile-new-06.txt
    RTP Profile for Audio and Video Conferences with Minimal Control

6 Payload Type Definitions

   Tables 4 and 5 define this profile's static payload type values for
   the PT field of the RTP data header.

            PT   encoding    media type  clock rate  channels
                 name                    (Hz)
            ___________________________________________________
            9    G722        A           16000       1

   Table 4: Payload types (PT) for audio encodings

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

- terry.g.lyons@lucent.com  +1 732 817-6016




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To: casner@cisco.com, schulzrinne@cs.columbia.edu, rem-conf@es.net
Reply-To: O.Hodson@cs.ucl.ac.uk
Subject: comment on draft-ietf-avt-profile-new-06.txt
Date: Wed, 07 Jul 1999 16:37:36 -0700
From: Orion Hodson <hodson@aciri.org>
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The latest avt profile draft again has no payload formats are
specified for the 2,3,5 bits per sample coding of G726 ADPCM.  There
are only two issues - the packetization interval and the sample
packing order.  Both are trivial but should be stated for the sake
interoperability.

20 ms packetization interval is common for most sample based coders
and results in all G726 encoding formats being byte aligned.  

The sample packing order is slightly complicated because the existing
G726-32 packs samples in the order s1-s0-s3-s2 and there is no obvious
extension to 3 and 5 bit per sample schemes since they do not align a
single octet boundary.  The other sample based encodings pack samples
in the order with which they were generated.  If there is no objection
>from the list I would like to propose that the 2,3, and 5 bit per
sample encodings are packed in the order of sample generation and we
leave the 4 bits per sample as is.

At least one application already incorporates these codecs, not
because they are cutting edge, but because source code is publicly
available and (relatively) unencumbered.

- Orion




From rem-conf Wed Jul 07 17:36:01 1999 
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From: "Chan Gyun Jeong" <cgjeong@oslab.kyunghee.ac.kr>
To: <rem-conf@es.net>
Subject: Some questions about RTSP
Date: Thu, 8 Jul 1999 09:41:33 +0900
Organization: Kyung Hee Univ.
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Hello, all.

I am Chan Gyun Jeong who is a master student majoring in
Computer Engineering, Kyung Hee University, Seoul, South Korea.

Nowadays, I am interested in Real-time Transport Protocol(RTP)
and Real-Time Streaming Protocol(RTSP).
I also have been developing a VOD system using RTP and
RTSP.

I have some problems that I can't understand. That is how  VOD server
could notify to clients when the media file reaches end of the file?

I think that this notification makes the client change  playing state, so the
client can report the playing state to the end user like Real Player. I really
need this functions.

However, RTSP does not provides this notification method according to
RFC 2326. Is there any notificaiton method that I haven't found yet in
RTSP? Otherwise, do I have to solve this problem with RTCP BYE
message?

please let me know how to solve this problem.
Thank you in advance.

-------------------------------------------------------------------
 Jeong, Chan Gyun (Dept. of Computer Engineering at KyungHee Univ.)
 [Web page] http://oslab.kyunghee.ac.kr/~cgjeong/
 [E-mail] cgjeong@oslab.kyunghee.ac.kr
 < For more information about me, contact to me. >




From rem-conf Wed Jul 07 18:41:15 1999 
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Date: Wed, 07 Jul 1999 18:31:39 -0700
From: Anup Rao <anrao@cisco.com>
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To: Chan Gyun Jeong <cgjeong@oslab.kyunghee.ac.kr>
CC: rem-conf@es.net, confctrl@isi.edu
Subject: Re: Some questions about RTSP
References: <002f01bec8da$a0f83780$6d81b4a3@kyunghee.ac.kr>
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Chan Gyun Jeong wrote:
> 
> Hello, all.
> 
> I am Chan Gyun Jeong who is a master student majoring in
> Computer Engineering, Kyung Hee University, Seoul, South Korea.
> 
> Nowadays, I am interested in Real-time Transport Protocol(RTP)
> and Real-Time Streaming Protocol(RTSP).
> I also have been developing a VOD system using RTP and
> RTSP.

First things first, this question is best addressed to the list :
confctrl@isi.edu.

> I have some problems that I can't understand. That is how  VOD server
> could notify to clients when the media file reaches end of the file?
> 
> I think that this notification makes the client change  playing state, so the
> client can report the playing state to the end user like Real Player. I really
> need this functions.

The way to do this is to have the client issue a DESCRIBE request first, to get
a (SDP) description of the clip. The SDP description has a range parameter - see
Section C.1.5. This parameter specifies the total length of the clip.

The client should keep an mapping between RTP timestamps and NPT(normal play
time). (Appendix B explains all this). 
 So, when a received RTP timestamp maps to a NPT value >=  whats asked for(or >
than the total NPT range of the clip), the end has been reached.

In case the presentation is a "live" one, SDP can specify the range in absolute
time, so the client knows when the feed is over.

Finally, in the case where it is not known beforehand how long the feed is - a
RTCP BYE seems a reasonable way of doing it. Relying on the RTSP control channel
may not always work, since it may be long gone.

-Anup Rao.



From rem-conf Thu Jul 08 15:15:09 1999 
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Reply-To: "Andrea Basso" <basso@research.att.com>
From: "Andrea Basso" <basso@research.att.com>
To: <rem-conf@es.net>
Cc: "Mathias Kretschmer" <mathias@research.att.com>
Subject: AAC-RTP payload format
Date: Thu, 8 Jul 1999 18:06:33 -0400
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This is a multi-part message in MIME format.

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Please find attached the correct version of the I-D AAC MPEG-2  RTP payload
format specification.
The version on the IETF repository ( draft-ietf-avt-rtp-mpeg2aac-00.txt ) is
not the correct one.

all the best

-Andrea

Andrea Basso, AT&T Labs Research
Room 3-219
100 Shultz Drive, Red Bank,NJ 07701
ph:  (732) 345-3302
fax  (732) 345-3033
basso@research.att.com


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Internet Engineering Task Force          Kretschmer-AT&T/Basso-AT&T
INTERNET DRAFT                           Civanlar-AT&T/Quackenbush-AT&T
File:draft-kretschmer-mpeg2aac-01.txt    Snyder-AT&T
                                         June 25, 1999
                                         Expires: December 25, 1999
                                                                       =20


                RTP Payload Format for MPEG-2 AAC Streams


                         STATUS OF THIS MEMO

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

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

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

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

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



                                 Abstract

This document describes a payload format for transporting MPEG-2 AAC
encoded data using RTP. MPEG-2 AAC is a recent standard from ISO/IEC
for the coding multi-channel audio data. Several services provided by
RTP are beneficial for MPEG-2 AAC encoded data transport over the
Internet. Additionally, the use of RTP makes it possible to
synchronize MPEG-2 AAC data with other real-time streams .
















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1. Introduction

The ISO/IEC MPEG-2 Advanced Audio Coding (AAC) [1] technology delivers
unsurpassed multichannel audio quality at rates at or below 64
kbps/channel. It has a flexible bitstream syntax that supports from 1
to 48 audio channels, up to 16 subwoofer channels and up to 16
embedded data channels.  AAC supports a wide range of sampling
frequencies (from 16 kHz to 96 kHz) which enables it to have an
extremely wide range of bitrates.  This permits it to support
applications ranging from professional or home theater sound systems
to Internet music broadcast systems.

The benefits of using RTP for MPEG-2 AAC data stream transport include:

    i. Ability to synchronize MPEG-2 AAC streams with other RTP payloads

    ii. Monitoring MPEG-2 AAC delivery performance through RTCP

    iii. Combining MPEG-2 AAC and other real-time data streams received
    from multiple end-systems into a set of consolidated streams
    through RTP mixers =20

    iv. Converting data types, etc. through the use of RTP translators.

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 [3].


1.1 Overview of MPEG-2 AAC

AAC combines the coding efficiencies of a high resolution filter bank,
a powerful model of audio perception, backward-adaptive prediction,
joint channel coding, and Huffman coding to delivering excellent
signal compression. In 1998 the MPEG Audio subgroup tested the family
of MPEG audio coders (see http://www.tnt.uni-hannover.de/project/mpeg/
audio/public/w2006.pdf). The test results indicate that for a stereo
signal, AAC at 96 kb/s has audio quality comparable to MPEG-3 Layer 3
("mp3") at 128 kb/s.  Therefore at equivalent quality levels, AAC
offers approximately 1/3 greater compression than Layer 3.

AAC is a block oriented, variable rate coding algorithm, which means
that the AAC encoder reads 1024 samples of the input signal file and
writes a variable number of compressed output bits that represent that
block of input data. A sample can be one or more channels. Rate
control can be used in the encoder such that the output bit rate is
averaged to a predetermined rate, as would be required for
constant-rate communication channels. Each block of AAC compressed
bits is called a "raw data block", and it has the nice property that
it can be decoded "stand-alone", that is, without knowledge of
information in prior bitstream blocks. This is ideal for packet
communication channels, in that if the payload of a packet is a single


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raw data block, packet framing facilitates encoder and decoder
synchronization and, most importantly, loss of a single packet does
not impair the decodability of adjacent packets.

1.2 Bitstream Syntax

As already stated, a raw data block represents audio data for a time
period of 1024 samples and may also contain related information and
other data. The syntax of an AAC bitstream is as follows:

<bitstream>        =3D> <raw_data_block><bitstream>=20
<raw_data_block>   =3D> [<element>]<END><PAD>

where <bitstream> indicates the AAC bitstream, <lowercase> indicates
intermediate tokens, <UPPERCASE> indicates terminal tokens and []
indicates one or more occurance. <END> is a token that indicates the
end of a raw_data_block and <PAD> is a variable length token that
forces the total length of a raw_data_block to be an integral number
of byes. In general, intermediate tokens are not an integral number of
bytes in length.

The <element> tokens are a string of bits of varying length, and can
be any of the following:

<single_channel_element>     represent a single audio channel
<channel_pair_element>       represent a stereo presentation (2 =
channels)
<coupling_channel_element>   a mechanism for multi-channel compression
<lfe_channel_element>        represent a special effects channel
<data_stream_element>        represent "user data"
<program_config_element>     a mechanism for describing the bitstream=20
                             content
<fill_element>               a mechanism to use bits (for constant rate
                             channels)

The <elements> above can occur several times in a single
raw_data_block. For example, the raw_data_block for a 5.1 surround
sound signal would be:
 =20
<single_channel_element><channel_pair_element>...
<channel_pair_element><lfe_channel_element><END>

corresponding to the center, left and right, left surround and right
surround and effects channels. Multiple occurances of the
<channel_pair_element> are dis-ambiguated by means of a unique 4-bit
id inside the <channel_pair_element>.









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2. Issues covered by this Payload Format

2.1 Repair Information to reconstruct lost AAC Frames

A smart AAC decoder can mitigate the effects of lost packets using
techniques such as interpolation in the spectral domain. However if
the raw_data_block in a packet is perceptually very meaningful and
also highly unpredictable (e.g. the onset of a symbol crash) then the
encoder may choose to send repair information associated with that
raw_data_block. We will call RepairData the variable size array
containing such information.  The RepairData in a given packet is
typically associated with a raw_data_block that will be decoded in the
future. The association between the raw_data_block and the RepairData
can obtained by means of a specific field called RSEQ.

The syntax of the RepairData and the AAC raw_data_block is the same.
In practice, the RepairData can be a highly compressed monophonic
version of the signal being transmitted. For example, an AAC stereo
signal coded to an average rate of 96 kb/s corresponds to a
raw_data_block size of 279 bytes.  A RepairData version of that block,
compressed to 16 kb/s would be 46 bytes.  Given that perceptually
critical blocks might occur only once per 100 or more blocks, the
average rate imposed by the RepairData is very low.

The usage of the RepairData information is similar to the one proposed =
in[4].
RepairData MAY be provided for every frame but its provision is
OPTIONAL, in general.


2.2 Fragmentation of AAC Frames

Since it is advantagous to put one AAC raw_data_block per packet, it
is desirable to try to limit the size of the AAC raw_data_block to
less than the path-MTU. If this is not possible, the raw_data_block
can be fragmented across several packets. In this case, the
raw_data_block can be fragmented at <element> boundaries and the LEN
field used to indicate the length of the <element> to within a byte
and the UBITS field used to indicate the length of the <element> to a
bit. The LEN and UBITS information allows re-assembly of the
raw_data_block without knowledge of the syntax of the bits within each
<element> in the raw_data_block.













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=0CINTERNET-DRAFT    RTP Payload Format for MPEG-2 AAC Streams     June =
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2.3 Priority of AAC Frames

Priority information is very important for AAC streaming over lossy
channels since it allows to handle adaptively packet losses and/or
given bandwidth constraints. Four priority levels are defined: 0 1 2
3. The priority level expresses the perceptual entropy of the AAC
frame.

Priority information is coded for every AAC frame in the Priority
Quantifier (PQ) which is 2 bit in length. For a given RTP packet such
PQs are organized in a Priority vector.


2.4 Interleaving of AAC Frames

Instead of using a static interleaving scheme (i.e. 7x7) only frames
with the same priority MUST be grouped.  The sequence numbers SEQ of
the AAC frames and RSEQ of REPAIRDATA are used to restore the actual
order on the receiver side. Hence, the interleaving scheme does not
have to be defined rigidly.


2.5 Example RTP Packet Sequence

The example below shows how a sequence of AAC frames (a...p) with
assigned priorities (0=3Dlow, 3=3Dhigh) MAY be grouped.
 =20
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| a | b | c | d | e | f | g | h | i | j | k | l | m | n | o | p |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| 0 | 0 | 0 | 2 | 3 | 1 | 0 | 0 | 0 | 0 | 0 | 0 | 0 | 2 | 2 | 3 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


Proposed interleaving/grouping of AAC frames and assigned RepairData
R(x) being sent within the following RTP packet:

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|a g j|b h k|c i l|  d  |  e  |  f  | m q |  n  |  o  |  p  |
+=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+
|     |R(d) |R(e) |R(f) |     |R(n) |R(o) |R(p) |     |     |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+












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3. RTP AAC Payload Format

The AAC specific RTP payload consists of a 32 or 64 bit header, a
RepairData array which is variable in size containing information
needed to reconstruct lost AAC frames and a variable number of AAC
frames.

The header contains a vector of Priority Quantizers (PQ) specifying
the priority of the current and previous packets to the decoder to
reconstruct the original signal.

The X bit specifies if the header contains 12 or 28 PQs.=20

REPAIRLEN specifies the length of the RepairData array expressed in
32bit words.  REPAIRLEN MUST be set to 0 if the RepairData array is
empty. Every REPAIRDATA array (AAC frame) is preceded by a sequence
number SEQ (RSEQ) and a length specifier LEN (RLEN). In case of
fragmented AAC frames UBITS specifies the number of unused bits in the
last byte since frame fragments may not be byte aligned.

=20
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|X|REPAIRLEN    |PRI VECTOR                                     | Header
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|PRI VECTOR (continued), if X=3D=3D1                                |
+=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+
|RSEQ           |RLEN           |REPAIRDATA 1                   | =20
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+                               |
|				.                               | Repair
|				.                               | Data
|				.	                        |
|               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|               |RSEQ           |RLEN           |REPAIRDATA N   |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+               |
|                                                               |
|                                                               |
|                                                               |
+=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+
|SEQ            |LEN                    |UBITS  |AAC FRAME 1    |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+               |
|				.                               |
|				.                               |
|				.                               |  AAC
|               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+  =
Frames
|               |SEQ            |LEN                    |UBITS  |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+=20
|AAC FRAME N                                                    |
|                                                               |
|                                                               |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


Kretschmer/Basso/Civanlar/Quackenbush/Snyder                     [Page =
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PRI VECTOR: Priority vector. It contains either 12 or 28=20
	    Priority Quantifiers (PQ). The size of a PQ element is 2 bits.
	    Hence, four different priority levels can be assigned to=20
	    an RTP packet. 0 means low and 3 means high priority. =20
	    The first PQ refers to the current packet. The following
	    PQs refer to the most recent previous packets.=20
	    So, the vector looks like this: {PQ(t), PQ(t-1), PQ(t-2)...}

X:          Vector Extension, the priority vector uses 56 instead of 24
	    bits. Hence, another 32bit word is required.


REPAIRLEN:  The total number of 32bit words containing Repair=20
	    Data for previous frames. If REPAIRLEN=3D0 then
	    there is no repair information.=20

RSEQ:	    The SEQ number of the AAC frame REPAIRDATA belongs to.

RLEN:	    The length in bytes of REPAIRDATA.
         =20
REPAIRDATA: An 8bit aligned data array containing repair information.=20
	    This information can be ignored and is not mandatory. It
	    should contain information that helps the decoder to=20
	    reconstruct a lost frame as close to the original as=20
	    possible.
 =20
SEQ:	    8 bit. The sequence number of the AAC frame.=20
	    The application has to make sure that the sequence numbers of=20
	    interleaved frames to not overlap.

LEN:        12 bit. The length of the actual AAC frame

UBITS:	    4 bit. The number of unused bits in the last byte of the AAC
	    frame if the frame is fragmented.=20


3.1 RTP Header Fields Usage:

The RTP header fields are used as follows:

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

Marker (M) bit: Set to one to mark the last fragment (or only
fragment) of an AAC frame.





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

Timestamp (TS): 32-bit 90K Hz timestamp representing presentation time
of the AAC frame.  Same for all packets that make up the fragmented
AAC frame.Timestamps are recommended to start at a random value for
security reasons.

SSRC: set as described in RFC1889 [2].

CC and CSRC fields are used as described in RFC 1889 [2].


4. References

  [1] ISO/IEC 13818-7 Advanced Audio Coding (AAC)

  [2] Schulzrinne, Casner, Frederick, Jacobson RTP: A
  Transport Protocol for Real Time Applications  RFC 1889,
  Internet Engineering Task Force, January 1996.

  [3] S. Bradner, Key words for use in RFCs to Indicate
  Requirement Levels, RFC 2119, March 1997.

  [4] Perkins,Kouvelas,Hodson,Hardman,Handley,Bolot,Vega-Garcia,
  Fosse-Parisis RTP Payload for Redundant Audio Data=20
  draft-ietf-avt-redundancy-revised-00.txt




























Kretschmer/Basso/Civanlar/Quackenbush/Snyder                     [Page =
8]
=0CINTERNET-DRAFT    RTP Payload Format for MPEG-2 AAC Streams     June =
1999


5. Authors' Addresses

Mathias Kretschmer
AT&T Labs - Research
180 Park Ave.
Florham Park, NJ 07932
USA
e-mail: mathias@research.att.com

Andrea Basso
AT&T Labs - Research
100 Schultz Drive
Red Bank, NJ 07701
USA
e-mail: basso@research.att.com

M. Reha Civanlar
AT&T Labs - Research
100 Schultz Drive
Red Bank, NJ 07701
USA
e-mail: civanlar@research.att.com

Schuyler R. Quackenbush
AT&T Labs - Research
180 Park Ave.
Florham Park, NJ 07932
USA
e-mail: srq@research.att.com

James H. Snyder
AT&T Labs - Research
180 Park Ave.
Florham Park, NJ 07932
USA
e-mail: jhs@research.att.com


















Kretschmer/Basso/Civanlar/Quackenbush/Snyder                     [Page =
9]

------=_NextPart_000_028B_01BEC96C.9CB7E160--




From rem-conf Fri Jul 09 08:08:34 1999 
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From: "Lyons, Terry G (Terry)" <tlyons@lucent.com>
To: rem-conf@es.net
Cc: "Lyons, Terry G (Terry)" <tlyons@lucent.com>
Subject: G.722 frame -- Re: draft-ietf-avt-profile-new-06.txt
Date: Fri, 9 Jul 1999 11:02:54 -0400 
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There is one other point to be corrected, which may be the source
of the confusion: Table 1 should list G.722 as "frame" not "sample".

G.722 operates on 2 audio samples at a time, producing 6 bits for
the lower frequency subband and 2 bits for the upper subband.
The G.722 frame is a single octet.
Each encodes a fixed-length 125us block of audio.

- terry.g.lyons@lucent.com  +1 732 817 6016

> ----------
> From: 	tgl@mtgbcs.ho.lucent.com[SMTP:tgl@mtgbcs.ho.lucent.com]
> Sent: 	Wednesday, July 07, 1999 6:27 PM
> To: 	rem-conf@es.net
> Cc: 	Lyons, Terry G (Terry)
> Subject: 	Re: draft-ietf-avt-profile-new-06.txt
> 
> The new version, on the road to Draft Standard, makes an unnecessary
> change in the time-stamp unit for the audio encoding G.722, from 8000
> to 16000 Hz.  This is not listed in section 9 "Changes from RFC 1890".
> The change is inconsistent with prior implementations that comply with
> RFC 1890, such as Table B-2/H.225.0(1998) part of the H.323 suite.
> 
> PROPOSAL:  Restore the clock rate of G.722 to 8000 Hz as it was.
> 
> The raw sampling rate at a microphone is a distinct concept from the
> time-stamp unit for encoded audio.  The two need not be the same, as
> G.722 demonstrates.  Its encoder spits out one octet every 1/8000 sec,
> which cannot be further subdivided in time.
> 
> ---------------------------------------------------
> Quoted from Request for Comments: 1890
>     RTP Profile for Audio and Video Conferences with Minimal Control
> 
> 6.  Payload Type Definitions
> 
>    Table 2 defines this profile's static payload type values for the PT
>    field of the RTP data header. ...
> 
>       PT         encoding      audio/video    clock rate    channels
>                  name          (A/V)          (Hz)          (audio)
>       _______________________________________________________________
>       9          G722          A              8000          1
> 
>    Table 2: Payload types (PT) for standard audio and video encodings
> 
> ---------------------------------------------------
> Quoted from draft-ietf-avt-profile-new-06.txt
>     RTP Profile for Audio and Video Conferences with Minimal Control
> 
> 6 Payload Type Definitions
> 
>    Tables 4 and 5 define this profile's static payload type values for
>    the PT field of the RTP data header.
> 
>             PT   encoding    media type  clock rate  channels
>                  name                    (Hz)
>             ___________________________________________________
>             9    G722        A           16000       1
> 
>    Table 4: Payload types (PT) for audio encodings
> 
> ---------------------------------------------------
> 
> - terry.g.lyons@lucent.com  +1 732 817-6016
> 
> 



From rem-conf Fri Jul 09 08:55:34 1999 
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RTP does not care what the internal implementation of the codec is.
G.722 states (pg. 1) that it transmit an "audio signal which is coded
using 14 bits with 16 kHz sampling". That it happens to be converted
into an octet emitted at 8,000 bytes/second is not relevant, as then any
codec emitting 8,000 bytes/second would have an RTP clockrate of 8,000,
which is clearly not intended.

G.722, by any definition that I have heard, is a sample-based codec, as
it operates on individual samples, not frames. Is is a SB-ADPCM codec,
and ADPCM codec are classic examples of sample-based codecs.

Thus, I believe that the old value of 8,000 Hz is plain technically
wrong. If there's a large installed base, then we need to deal with this
on an exception basis, but otherwise, the correct value is 16 kHz,
sample-based.

Henning



From rem-conf Fri Jul 09 08:56:33 1999 
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From: Ethendranath Bommaiah <ethen@dnrc.bell-labs.com>
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Hi,

I was wondering if Real G2 server supports RTP for real media (.rm and
.ra) ? We have noticed that it does support RTP for other media like
.au and .mpeg. However, with .rm and .ra the server returns "Protocol
not supported" error for the SETUP message with transport set to
RTP/AVP/TCP. We are using Real G2 server 6.0.

Does Real plan to support RTP for .rm and .ra media in the future ?
Or, is there a newer version that supports RTP already ?

Any pointers will be greatly appreciated.

Thanks,
Ethen




From rem-conf Fri Jul 09 09:21:23 1999 
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Ethendranath Bommaiah wrote:
> 
> I was wondering if Real G2 server supports RTP for real media (.rm and
> .ra) ? We have noticed that it does support RTP for other media like
> .au and .mpeg. However, with .rm and .ra the server returns "Protocol
> not supported" error for the SETUP message with transport set to
> RTP/AVP/TCP. We are using Real G2 server 6.0.
> 
> Does Real plan to support RTP for .rm and .ra media in the future ?
> Or, is there a newer version that supports RTP already ?
> 

>from what i have seen, and heard from real, the answer to your question
is "no". real does not support RTP for realmedia files. from talking to
them, i get the feeling that they do not plan to support realmedia over
RTP. the reason they cite is that RDT (their own data transport protocol
now referred to as RDP i think) is tuned for the special needs of
realmedia files.

	--ravi



From rem-conf Fri Jul 09 09:25:29 1999 
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From: Carsten Bormann <cabo@informatik.uni-bremen.de>
To: Henning Schulzrinne <hgs@cs.columbia.edu>
Cc: "Lyons, Terry G (Terry)" <tlyons@lucent.com>, rem-conf@es.net
Subject: Re: G.722 frame -- Re: draft-ietf-avt-profile-new-06.txt
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Henning Schulzrinne writes:
> G.722, by any definition that I have heard, is a sample-based codec

My dim memory seems to say the ADPCM processing is done at an 8 kHz
rate on each of the two sub-bands (produced by quadrature mirror
filtering).  Which I think is what Terry was talking about.

On the other hand, we are not using a 50 Hz clock for GSM, neither are
we using 13.5 MHz for video.  So who cares?

> Thus, I believe that the old value of 8,000 Hz is plain technically
> wrong. If there's a large installed base, then we need to deal with this
> on an exception basis, but otherwise, the correct value is 16 kHz,
> sample-based.

This is a major, interoperability-damaging change, so put the document
back to field 1 (proposed standard), then.  Or strike G.722 and put it
into a separate standards track document -- then you can even
introduce a G722-Terry and a G722-Henning.  Or just put in a footnote*)
that you think 8000 is kind of wrong, but not wrong enough to be
changed.

Gruesse, Carsten

*) There are no footnotes in RFCs.  Never mind.



From rem-conf Fri Jul 09 09:25:46 1999 
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From: "Lyons, Terry G (Terry)" <tlyons@lucent.com>
To: Henning Schulzrinne <hgs@cs.columbia.edu>
Cc: rem-conf@es.net, "Lyons, Terry G (Terry)" <tlyons@lucent.com>
Subject: RE: G.722 frame -- Re: draft-ietf-avt-profile-new-06.txt
Date: Fri, 9 Jul 1999 12:25:27 -0400 
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Henning,

My concern is to maintain alignment with H.225.0
as well as the original RFC 1890.
If specs are published with different clock rates,
there is bound to be confusion and misoperation.
8 kHz works fine; let's stick with it.

The full text from which you quote is:

	Figure 1/G.722 identifies the main functional parts of the
	64 kbit/s (7 kHz) audio codec as follows:
	i)	64 kbit/s (7 kHz) audio encoder comprising:
			-	a transmit audio part which converts an
audio signal
			to a uniform digital signal which is coded using 14
bits with 16 kHz sampling;
			-	a SB-ADPCM encoder which reduces the bit
rate to 64 kbit/s.

The "transmit audio part" is an internal functional block
that feeds the "SB-ADPCM encoder" functional block.
The two together make up the G.722 audio encoder.

It is not possible to map individual 16 kHz samples 1-1 to outputs.
Samples are processed in pairs by the SB-ADPCM encoder.
That is why the description as a "frame" encoder fits better.

In the encoder principles in section 3, you'll see that there is
a distinction of indexes "j" for 16 kHz and indexes "n" for 8 kHz.
To get output x-out[n] you need to compute on pairs x-in[j-1] and x-in[j].

I don't dispute your description of plain ADPCM, eg. G.726, as "sample".
But G.722 SB-ADPCM is different because of the frequency transform
to separately encode the high and low subbands.

- terry.g.lyons@lucent.com  +1 732 817 6016

> ----------
> From: 	Henning Schulzrinne[SMTP:hgs@cs.columbia.edu]
> Sent: 	Friday, July 09, 1999 11:52 AM
> To: 	Lyons, Terry G (Terry)
> Cc: 	rem-conf@es.net
> Subject: 	Re: G.722 frame -- Re: draft-ietf-avt-profile-new-06.txt
> 
> RTP does not care what the internal implementation of the codec is.
> G.722 states (pg. 1) that it transmit an "audio signal which is coded
> using 14 bits with 16 kHz sampling". That it happens to be converted
> into an octet emitted at 8,000 bytes/second is not relevant, as then any
> codec emitting 8,000 bytes/second would have an RTP clockrate of 8,000,
> which is clearly not intended.
> 
> G.722, by any definition that I have heard, is a sample-based codec, as
> it operates on individual samples, not frames. Is is a SB-ADPCM codec,
> and ADPCM codec are classic examples of sample-based codecs.
> 
> Thus, I believe that the old value of 8,000 Hz is plain technically
> wrong. If there's a large installed base, then we need to deal with this
> on an exception basis, but otherwise, the correct value is 16 kHz,
> sample-based.
> 
> Henning
> 



From rem-conf Fri Jul 09 12:27:40 1999 
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From: almeroth@cs.ucsb.edu (Kevin C. Almeroth)
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>>from what i have seen, and heard from real, the answer to your question
>>is "no". real does not support RTP for realmedia files. from talking to
>>them, i get the feeling that they do not plan to support realmedia over
>>RTP. the reason they cite is that RDT (their own data transport protocol
>>now referred to as RDP i think) is tuned for the special needs of
>>realmedia files.

But Real is supporting the other direction, i.e. you can receive
"MBone" sessions using G2...  it'll even send RTCP.

-Kevin



From rem-conf Fri Jul 09 12:27:50 1999 
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From rem-conf Fri Jul 09 18:29:15 1999 
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First, I agree that to minimize confusion, we may need to keep the old 8
kHz sampling rate. However, I disagree with the technical
characterization of the sample rate, so this should probably be marked
as a historical aberration rather than something that should be done for
other subband codecs.

Frame vs. sample: the QMF operates on the 16 kHz-sampled signal, splits
it into two subbands, which are then subsampled at 8 kHz (since they are
each bw-limited to 8 kHz). There is no requirement that I can see that
the number of input samples is a multple of two, which would be an
indication of a frame-based codec. Indeed, the QMF FIR filter operates
over a range of 22 samples (pg. 15), but in an overlapping fashion, not
as a block.

I could be wrong here, but subband codecs are generally treated in the
literature along with all the other sample-based codecs.

Henning



From rem-conf Sat Jul 10 10:04:44 1999 
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Date: Sat, 10 Jul 1999 10:00:10 -0700 (Pacific Daylight Time)
From: Stephen Casner <casner@cisco.com>
To: rem-conf@es.net
Subject: Modify avg_rtp_size calculation?
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To the AVT working group:

Since RFC 1890, the RTCP transmission interval algorithm in the RTP
spec has included an estimator for the average RTCP packet size,
maintained in the variable avg_rtcp_size.  This average is calculated
across all RTCP packets both sent and received.

If one node sends longer packets, or multiple packets at once, then it
will be consuming more than its fair share of the RTCP bandwidth.
That could be fixed by having each node average only the packets it
sends.

The problem with making that change is that the interval calculated by
different nodes might then vary significantly, causing the nodes with
the longer interval to be timed out by nodes with shorter intervals.

Should we leave the algorithm as is and ignore the potential for
unfairness, or should we try to figure out some way to make the change
work?
							-- Steve





From rem-conf Sat Jul 10 10:04:47 1999 
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Date: Sat, 10 Jul 1999 10:00:46 -0700 (Pacific Daylight Time)
From: Stephen Casner <casner@cisco.com>
To: rem-conf@es.net
Subject: Change G722 codec to 16000 Hz clock?
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To the AVT working group:

In the RTP A/V Profile RFC 1890, and therefore in H.225.0 (part of
H.323) where RFC 1890 is incorporated, the RTP timestamp clock
frequency for the G722 payload format / encoding was listed as 8000
Hz.  In fact, the sampling frequency for G.722 is 16000 Hz since it is
a 0 - 7 kHz codec.  Therefore, in the draft revisions of the profile,
the G722 entry in Table 1 and Table 4 has been changed to 16000.

Terry Lyons has asked that the draft revisions be reverted back to
8000 for consistency with the 1890 and H.225.0 (which is important).
Terry notes that G.722 produces octets at 8000 Hz, where each octet
contains a two-bit high-band sample and a 6-bit low-band sample.  So
in some sense, 8000 would be the natural rate for processing of G722.

On the other hand, Henning Schulzrinne believes this would
fundamentally violate the notion of RTP clock rate = sample rate for
audio codecs (excepting MPEG).  If there are commercial
implementations, we probably have to leave it at 8000, but clearly
mark it as an exception due to a mistake, not because it is the
technically right thing to do.

Also, if you wanted to be able to switch between L16 or
PCMU at 16000 and G.722 depending upon the bandwidth constraint, the
inaccurate switch in clock frequencies might cause trouble.

So, I'd like to hear comments on what we should do in this case.
Has anyone implemented G722 encoding?
							-- Steve





From rem-conf Sat Jul 10 10:04:47 1999 
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Date: Sat, 10 Jul 1999 09:58:46 -0700 (Pacific Daylight Time)
From: Stephen Casner <casner@cisco.com>
To: rem-conf@es.net
Subject: Revision of SSRC collision algorithm?
Message-ID: <Pine.WNT.4.10.9907061412520.-216065@revelstoke.cisco.com>
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To the AVT working group:

There are a few questions raised just recently regarding the RTP spec
and profile which we need to discuss in Oslo.  To preview those
questions, I'm sending a few messages, one question per each.  In
addition, Henning Schulzrinne already sent a message about the static
payload type assigned to the CN (Comfort Noise) payload type, which is
also in this list of recent questions.  [Sorry these are so late -- I
tried to send them when I first arrived in Oslo on Tuesday, but
couldn't get a modem connection to work.]

Henning Schulzrinne proposes that the SSRC collision algorithm be
modified, and that the pseudo-code which describes it be replaced with
pseudo-C.

Henning suggests that a mobile phone sending RTP may just change its
IP address as it moves, rather than using mobile IP mechanisms.  This
could work because the RTP SSRC and SDES information uniquely
identifies the source, even if it changes IP addresses.  (An address
change could also happen at a gateway if, for example, routing or
redundancy fallback occurs and packets come from a different
interface.)

The SSRC collision algorithm requires that the SSRC identifier be
changed if the transport address changes in order to avoid the address
change being considered a collision.

Henning proposes instead that when a host changes its address it
should leave the SSRC identifier unchanged and that the spec should
say that receivers SHOULD keep packets from the new address and
discard packets from the old address when this "collision" of
addresses with the same SSRC identifier is detected.

The spec says an implementation MUST include an algorithm similar to
the one described in Section 8.2.  The algorithm shown in pseudo-code
keeps the packets from the old address and discards those from the new
address which is the "right" approach for a genuine collision, based
on the assumption that one wants to continue hearing the valid source
that one was hearing before.  If this event was an address change, the
new source would be accepted once the old source timed out because no
more packets were received.

This is not intended to require that the policy be to ignore the new
source.  A different policy may be chosen and still have the algorithm
be similar; I added a phrase about this in the most recent draft.

Regarding replacing the existing pseudo-code with pseudo-C, the syntax
I used in writing the existing algorithm was taken from an example
provided at the time by the RFC Editor.  The reason for the pseudo
code is that Internet Standards are not supposed to include code
(because they are supposed to specify what happens on the wire, not
how to implement the protocol).  I can get a new reading on this from
the AD and the RFC Editor.
							-- Steve




From rem-conf Sat Jul 10 10:04:47 1999 
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Date: Sat, 10 Jul 1999 09:59:19 -0700 (Pacific Daylight Time)
From: Stephen Casner <casner@cisco.com>
To: rem-conf@es.net
Subject: Distinct UDP ports for layered encodings?
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To the AVT working group:

The RTP spec says that UDP ports MUST be distinct for layered codings,
but the SDP spec says this is illegal if the addresses differ.  Here
is the section from the RTP spec:

   It is RECOMMENDED that layered encoding applications (see Section
   2.4) use a set of contiguous port numbers.  The port numbers MUST be
   distinct because of a widespread deficiency in existing operating
   systems that prevents use of the same port with multiple multicast
   addresses, and for unicast, there is only one permissible address.
   Thus for layer n, the data port is P + 2n, and the control port is P
   + 2n + 1. When IP multicast is used, the addresses MUST also be
   distinct because multicast routing and group membership are managed
   on an address granularity. However, allocation of contiguous IP
   multicast addresses cannot be assumed because some groups may require
   different scopes and may therefore be allocated from different
   address ranges.

And from the SDP spec:

     It is illegal for both multiple addresses to be specified in the
     "c=" field and for multiple ports to be specified in the "m=" field
     in the same session description.

This is clearly a conflict.  How should we resolve it?

							-- Steve





From rem-conf Sat Jul 10 16:09:15 1999 
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Stephen Casner wrote:
> 

> The SSRC collision algorithm requires that the SSRC identifier be
> changed if the transport address changes in order to avoid the address
> change being considered a collision.
> 
> Henning proposes instead that when a host changes its address it
> should leave the SSRC identifier unchanged and that the spec should
> say that receivers SHOULD keep packets from the new address and
> discard packets from the old address when this "collision" of
> addresses with the same SSRC identifier is detected.

I like this, but I have a question on the new mechanism. Why discard
packets from the old address? Basically, if we don't even look at the
transport address, just the SSRC, no change will be seen at all. Thus,
packets from the "old" address can be used if they haven't come too
late, and same with new ones. This assumes that the SN/TS space does not
change when the transport address changes. For a mobile host, thats
reasonable, no?

On a related note, I think it does make sense to look at the transport
address when an SSRC collision is detected (by means of SDES CNAME). In
that case, the source transport address can be used to determine which
of the two colliding hosts the media is from. The differentiation will
fail if one of the colliding hosts is mobile and its transport address
changes after the collision; but, this is a temporary condition - it
should send a BYE and rejoin with a new SSRC.


> Regarding replacing the existing pseudo-code with pseudo-C, the syntax
> I used in writing the existing algorithm was taken from an example
> provided at the time by the RFC Editor.  The reason for the pseudo
> code is that Internet Standards are not supposed to include code
> (because they are supposed to specify what happens on the wire, not
> how to implement the protocol).  I can get a new reading on this from
> the AD and the RFC Editor.

But doesn't the new draft standard already have C code for the
reconsideration stuff? The FEC draft has C code also...

-Jonathan R.


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



From rem-conf Sat Jul 10 17:32:05 1999 
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It does raise an interesting sort-of-DOS attack - I can ignore the RTCP
rules, and send very large RTCP packets very frequently. This will tend
to skew the average RTCP size estimates, causing the estimate to
approach the size of the large packets I send. With a large estimate,
the interval used by every other participant becomes large. This allows
me, in effect, to slow down RTCP transmissions for everyone else in the
group.

Of course, I can also spoof lots of different SSRC and achieve the same
effect...

The DoS attacks aside, I'd be inclined not to change anything to fix the
problem you describe. Fairness doesn't seem that important in this
context; certainly its not worth any substantial changes to the
algorithms. I suspect that fixing this might require a per-user average
RTCP size to be maintained at each participant, and this will affect all
of our other timing mechanisms.

-Jonathan R.

Stephen Casner wrote:
> 
> To the AVT working group:
> 
> Since RFC 1890, the RTCP transmission interval algorithm in the RTP
> spec has included an estimator for the average RTCP packet size,
> maintained in the variable avg_rtcp_size.  This average is calculated
> across all RTCP packets both sent and received.
> 
> If one node sends longer packets, or multiple packets at once, then it
> will be consuming more than its fair share of the RTCP bandwidth.
> That could be fixed by having each node average only the packets it
> sends.
> 
> The problem with making that change is that the interval calculated by
> different nodes might then vary significantly, causing the nodes with
> the longer interval to be timed out by nodes with shorter intervals.
> 
> Should we leave the algorithm as is and ignore the potential for
> unfairness, or should we try to figure out some way to make the change
> work?
>                                                         -- Steve

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



From rem-conf Sun Jul 11 20:23:38 1999 
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Date: Sun, 11 Jul 1999 18:56:34
To: "Andrea Basso" <basso@research.att.com>
From: Ross Finlayson <finlayson@live.com>
Subject: Re: AAC-RTP payload format
Cc: <rem-conf@es.net>, "Mathias Kretschmer" <mathias@research.att.com>
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Some comments on this draft ("draft-kretschmer-mpeg2aac-01.txt"):

Unfortunately I found the draft rather confusing.  This was due in part to
my unfamiliarity with AAC (compared to regular MPEG audio), but also due to
the fact that the draft doesn't make it particularly clear how the data
that would make up an outgoing RTP packet would be produced by an
implementation.  In particular, which parts of this data would come
directly from the AAC data stream, and which parts would be *derived* from
the AAC data stream (& how)?

The draft should have referenced RFC 2250 - the existing RTP payload format
for MPEG - if only to explain why it is inappropriate for MPEG-2 AAC.
Presumably the main (only?) reason why RFC 2250 cannot be used is that AAC
frames look different from regular MPEG audio frames, and so a receiver
could not reliably distinguish them.

One desireable property of RFC 2250, however, is that an implementation can
send (or receive) RTP-encapsulated MPEG audio data without having to know
*anything* about its structure - only where the frame boundaries are.  This
makes it very easy for a RTP implementation to stream from/to MPEG audio
data (whether streamed, or in a file).

Is this property also true of your proposed format?  The draft does not
make this clear.  How much would an implementation of your proposed format
need to know about the internal structure of AAC data?  In particular,
where does the "repair data" come from?  Does this come directly from the
AAC data, or does it need to be 'generated' from the AAC data in some way?
Again, the draft does not make this clear.

It would be *really* desirable to have a RTP payload format for AAC that
does not require an implementation to have a detailed knowledge of AAC
internals.  Perhaps, however, knowledge of AAC internals is necessary for
including the "repair data"?  If so, then perhaps two payload formats could
be defined: a simple, non-repairing payload format (that doesn't require
knowledge of AAC internals), and more complex, reparing format (that does)??

Other, more specific comments about the draft:

Sections 1. and 1.1 contain too much 'marketing' language inappropriate for
a protocol spec.  In particular, the word "unsurpassed" in section 1.
should be removed, as should the whole first paragraph of section 1.1.

The draft refers to both AAC "raw data blocks" and "frames".  How are these
related (or are they the same)?

Section 1.2, paragraph 2
	"byes" should be "bytes"

Section 1.2
	Is the detailed expansion of the "<element>" token relevant to the RTP
packetization?

Section 2.1
	"symbol crash" should be "cymbal crash"

    Ross.





From rem-conf Mon Jul 12 03:09:30 1999 
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From rem-conf Mon Jul 12 07:12:36 1999 
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Dear All,

I must overlook something, somehow I can't
tunnel multicast between a subnet with another
subnet behind a firewall.

I tested mrouted3.8 between two regular subnets
and ran fine with vic and vat.  But, then when 
I try to do the same setup with a subnet which 
sits behind a firewall, I can't see any activity 
>from either vic, vat, or srd.

According to mrinfo, both subnets do see each
others' tunnel and they are up.

When I do the mtrace, I can see the reverse
path from the remote host back to my host, but
with 0 Lost/Sent and 0 pps for PctRate.

When I do the map-mbone, I do see the tunnel
between the two subnets.

When I do the tcpdump on rtp, I only see the rtp
traffic from my host to the firewall subnet but
not the other way around.

We later enable the multicast on the firewall
router, but still no use.  I tried to use metric
4 (since there are about 4 hops away) with threshold 
64, no go.  Then I tried with metric 1 with threshold 128, 
still no go.

So, am I missing something?

Thanks in advanced for any helps and pointers.

Best regards,

--Wo Chang



From rem-conf Mon Jul 12 08:32:32 1999 
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The G.722 sample rate/time stamp should remain at 8kHz. This is for reasons
of backward compatibility,  and a suitable explanation of the reason should
be included for clarity. 

Of course, the time stamp should have been 16kHz, but this was a simple
error which needs to be noted and maintained.

If it important that the H.323 experts are made aware of this situation.

All future coders which oprerate at a sample rate of 16kHz should also use
16kHz for the time stamp.  Note, the recently Determined ITU wideband coder
at 24 and 32 kbit/s will require a 16kHz time stamp if we follow the spirit
of the RTP specifications.



Tony Crossman




At 01:20 AM 7/10/99 +0000, Henning Schulzrinne wrote:
>
>
>
>
>First, I agree that to minimize confusion, we may need to keep the old 8
>kHz sampling rate. However, I disagree with the technical
>characterization of the sample rate, so this should probably be marked
>as a historical aberration rather than something that should be done for
>other subband codecs.
>
>Frame vs. sample: the QMF operates on the 16 kHz-sampled signal, splits
>it into two subbands, which are then subsampled at 8 kHz (since they are
>each bw-limited to 8 kHz). There is no requirement that I can see that
>the number of input samples is a multple of two, which would be an
>indication of a frame-based codec. Indeed, the QMF FIR filter operates
>over a range of 22 samples (pg. 15), but in an overlapping fashion, not
>as a block.
>
>I could be wrong here, but subband codecs are generally treated in the
>literature along with all the other sample-based codecs.
>
>Henning
>
>
>
>



From rem-conf Mon Jul 12 13:33:52 1999 
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From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
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Stephen Casner wrote:
> 
> To the AVT working group:
> 
> Since RFC 1890, the RTCP transmission interval algorithm in the RTP
> spec has included an estimator for the average RTCP packet size,
> maintained in the variable avg_rtcp_size.  This average is calculated
> across all RTCP packets both sent and received.
> 
> If one node sends longer packets, or multiple packets at once, then it
> will be consuming more than its fair share of the RTCP bandwidth.
> That could be fixed by having each node average only the packets it
> sends.

I agree with Jonathan that fairness isn't a prime consideration here.
Longer RTCP packets would occur in two situations:

- very verbose SDES information; in that case, it would probably be
better for the sender to stagger the SDES information;

- longer receiver list, e.g., because the receiver has been around
longer than the rest or other network weirdness (e.g., TTL induced).
Since this information is valuable for others, I don't see why it should
be sending less frequently.


> 
> The problem with making that change is that the interval calculated by
> different nodes might then vary significantly, causing the nodes with
> the longer interval to be timed out by nodes with shorter intervals.
> 
> Should we leave the algorithm as is and ignore the potential for
> unfairness, or should we try to figure out some way to make the change
> work?
>                                                         -- Steve

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



From rem-conf Mon Jul 12 14:19:40 1999 
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You probably aren't passing IP protocol 4 packets through the
firewall.   You described exactly the symptoms of passing only
IGMP and not IP-in-IP.

  Bill



From rem-conf Mon Jul 12 14:40:34 1999 
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Dear Bill,

Thanks for your response, and I think you're right!

I'll check the router to make sure I have the Protocol
4 enable.

Thanks again!

--Wo Chang

> From rem-conf-request@es.net Mon Jul 12 17:19 EDT 1999
> From: Bill Fenner <fenner@research.att.com>
> MIME-Version: 1.0
> To: wchang@nist.gov
> Subject: Re: tunneling between two subnets
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> 
> 
> You probably aren't passing IP protocol 4 packets through the
> firewall.   You described exactly the symptoms of passing only
> IGMP and not IP-in-IP.
> 
>   Bill
> 
> 



From rem-conf Mon Jul 12 14:41:59 1999 
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I tend to agree that the static payload should remain anomalous.  However,
perhaps there should also be a payload name for G722 and authors encouraged
to map another payload type to that name at 16kHz, as a preferred
alternative.

Perhaps it is even time to encourage explicit mappings for even the current
static types, with a view to permitting over-riding or even retiring the
static types down the road?
David Singer
Apple Computer/QuickTime



From rem-conf Mon Jul 12 14:56:54 1999 
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Date: Mon, 12 Jul 1999 17:55:44 -0400 (EDT)
From: Chinmay Padhye <padhye@eng.usf.edu>
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Hello:

	I had a question regarding the information present in a RTCP
packet. Incase FEC is used to protect a voice stream against loss, then
does the receiver send the loss rate after reconstruction or does it send
loss rate before reconstruction.

	How will a sender know how effective its redundancy sheme is?

Thanks for the help.

Sincerely,

Chinmay V. P.




From rem-conf Mon Jul 12 21:03:40 1999 
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Date: Tue, 13 Jul 1999 00:09:43
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From rem-conf Tue Jul 13 00:41:10 1999 
From rem-conf-request@es.net Tue Jul 13 00:41:08 1999
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To: Chinmay Padhye <padhye@eng.usf.edu>
cc: rem-conf@es.net
Subject: Re: RTCP Question.
In-reply-to: Your message of "Mon, 12 Jul 1999 17:55:44 EDT." <Pine.GSO.4.05.9907121750090.27532-100000@sunblast>
Date: Tue, 13 Jul 1999 08:38:44 +0100
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--> Chinmay Padhye writes:
>	I had a question regarding the information present in a RTCP
>packet. Incase FEC is used to protect a voice stream against loss, then
>does the receiver send the loss rate after reconstruction or does it send
>loss rate before reconstruction.

Before reconstruction.

>	How will a sender know how effective its redundancy sheme is?

I don't think there is a clear way of knowing that from RTCP RR packets
as they stand. One could envisage an extension to RTCP which made this
available, if it became necessary.

Colin



From rem-conf Tue Jul 13 02:17:32 1999 
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From: "Anders Klemets (Exchange)" <anderskl@exchange.microsoft.com>
To: 'Stephen Casner' <casner@cisco.com>, rem-conf@es.net
Subject: RE: Distinct UDP ports for layered encodings?
Date: Tue, 13 Jul 1999 02:13:12 -0700
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> The port numbers MUST be
>    distinct because of a widespread deficiency in existing operating
>    systems that prevents use of the same port with multiple multicast
>    addresses, and for unicast, there is only one permissible address.

I don't think there is any particular reason why one MUST use different port
numbers for unicast.  RTP certainly allows multiple unicast flows to be sent
to the same port number, as long as the SSRC is different for each flow.  I
don't think there are any operating system problems that would force one to
use different ports for unicast.

However, it is quite correct that there are OS reasons why the ports should
be different when using multiple multicast groups.

It does not look like the SDP spec gives a good justification for why
different ports would not be allowed.  So my suggestion is that we consider
revising the SDP spec.

Anders



From rem-conf Tue Jul 13 03:28:29 1999 
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From rem-conf Tue Jul 13 05:45:09 1999 
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To: anderskl@exchange.microsoft.com
Subject: RE: Distinct UDP ports for layered encodings?
Cc: casner@cisco.com, rem-conf@es.net
Date: Tue, 13 Jul 1999 05:39:28 -0700
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>It does not look like the SDP spec gives a good justification for why
>different ports would not be allowed.  So my suggestion is that we consider
>revising the SDP spec.

My recollection is that it was unclear how to interpret multiple ports
with multiple addresses (e.g. do you use all ports on all addresses
or certain ports with certain addresses or ...) so we punted completely.
It may be that a straightforward interpretation is possible (e.g.
if there are the same number of ports specified as addresses then you
pair up ports and addresses 1-1, else you punt).

  Bill



From rem-conf Tue Jul 13 06:04:37 1999 
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Date: Tue, 13 Jul 1999 05:57:56 -0700 (Pacific Daylight Time)
From: Stephen Casner <casner@cisco.com>
To: "Anders Klemets (Exchange)" <anderskl@exchange.microsoft.com>
cc: rem-conf@es.net
Subject: RE: Distinct UDP ports for layered encodings?
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On Tue, 13 Jul 1999, Anders Klemets (Exchange) wrote:
> I don't think there is any particular reason why one MUST use different port
> numbers for unicast.  RTP certainly allows multiple unicast flows to be sent
> to the same port number, as long as the SSRC is different for each flow.  I
> don't think there are any operating system problems that would force one to
> use different ports for unicast.

See section 5.2 for general arguments why not to do multiplexing of
different RTP sessions using the SSRC id.  But specifically for the
layered encodings, the spec says:

   For layered encodings transmitted on separate RTP sessions (see
   Section 2.4), a single SSRC identifier space SHOULD be used across
   the sessions of all layers and the core (base) layer SHOULD be used
   for SSRC identifier allocation and collision resolution. When a
   source discovers that it has collided, it transmits an RTCP BYE
   message on only the base layer but changes the SSRC identifier to the
   new value in all layers.

So the RTP sessions need to be kept unique, which means separate
ports.  Or the above gets changed.

> However, it is quite correct that there are OS reasons why the ports should
> be different when using multiple multicast groups.

OK, given they are different for mcast, then unicast should behave the
same way.

> It does not look like the SDP spec gives a good justification for why
> different ports would not be allowed.  So my suggestion is that we consider
> revising the SDP spec.

Perhaps we can have this be the start of the draft that Mark Handley
suggested to contain SDP spec errors and updates.  Probably this would
say someting following Bill Fenner's suggestion of 1-1 mapping or
punt.
							-- Steve





From rem-conf Tue Jul 13 07:28:33 1999 
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To: rem-conf@es.net
Subject: Revised AVT agenda
Date: Tue, 13 Jul 1999 15:19:16 +0100
Message-ID: <1380.931875556@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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Folks,

Enclosed is a revised agenda for the audio/video transport working group
meeting in Olso. We have an additional presentation on multiplexing in
the Thursday session, and the order of presentation during that session
has changed to accomodate clashes with other working groups. 

Please note that we are very short of time, and would appreciate those
presenting to be as brief as possible.

Colin




			 Audio/Video Transport WG
				     
				A G E N D A


Wednesday, 14 July at 09:00-11:30
=================================

- Status and progress since last meeting		(Casner)	10
	- SSRC sampling 
	- Guidelines 
	- QCELP

- Update on RTP specification and audio/video profile	(Casner)	20
	draft-ietf-avt-rtp-new-04.txt, .ps
	draft-ietf-avt-profile-new-06.txt, .ps
	draft-ietf-avt-rtp-mime-00.txt
	draft-ietf-avt-rtcp-bw-00.txt

- RTP interoperability statement and testing strategies
  	draft-ietf-avt-rtp-interop-00.txt		(Perkins)	10
  	draft-ietf-avt-rtptest-00.txt			(Perkins)	10
	draft-ietf-avt-rtcptest-01.txt			(Rosenberg)	10
	Discussion							10

- RTP MIB
	draft-ietf-avt-rtp-mib-05.txt			(Baugher)	 5

- RTP Payload for DTMF tones
	draft-ietf-avt-tones-00.txt			(Petrack)	10

- RTP Payload for Generic FEC
	draft-ietf-avt-fec-06.txt			(Rosenberg)	 5

- Report on intermin meeting on transport of MPEG-4 	(Civanlar)	 5

- RTP Payload for MPEG-4 with Scaleable & Flexible Error Resiliency
	draft-guillemot-genrtp-01.txt			(Wesner)	20

- RTP Payload for DV video 
	draft-ietf-avt-dv-video-00.txt			(Kobayashi)	10

- RTP encapsulation for 12-bit DV audio 
	draft-kobayashi-dv-audio12-00.txt		(Kobayashi)	10



Thursday, 15 July at 15:30-17:30
================================

- A Taxonomy of Feedback for Multicast
	draft-hnrs-rmt-avt-feedback-00.txt		(Rosenberg)	10

- Profile for unicast RTCP 
	<no internet draft>				(Casner)	10

- Discussion of new profiles/feedback mechanisms			 5

- Header compression for RTP/UDP/IP over cellular links
	draft-degermark-crtp-cellular-00.txt, .ps	(Degermark)	20
	draft-jonsson-robust-hc-00.txt

- Multiplexing header-compressed RTP over PPP
	draft-pazhyannur-avt-pppmux-00.txt		(Pazhyannur)	10

- Tunneled Compressed RTP	
	draft-wing-avt-tcrtp-00.txt			(Koren)		10

- Multiplexing scheme for RTP flows between Access Routers
	draft-ietf-avt-multiplexing-rtp-00.txt		(El-Khatib)	10

- RTP Payload for AAC audio
	draft-kretschmer-mpeg2aac-00.txt (?)		(Kretschmer)	10

- RTP Payload for real-time pointers
	draft-civanlar-rtp-pointer-00.txt		(Civanlar)	10

- RTP Payload for Text Conversation
	draft-hellstrom-avt-rtp-text-00.txt		(Hellström)	10

- RTP payload for virtual worlds
	draft-stewart-avt-00.txt			(Stewart)	15




From rem-conf Tue Jul 13 09:17:07 1999 
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From: "Anders Klemets (Exchange)" <anderskl@exchange.microsoft.com>
To: 'Stephen Casner' <casner@cisco.com>
Cc: rem-conf@es.net
Subject: RE: Distinct UDP ports for layered encodings?
Date: Tue, 13 Jul 1999 09:10:51 -0700
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>    For layered encodings transmitted on separate RTP sessions (see
>    Section 2.4), a single SSRC identifier space SHOULD be used across
>    the sessions of all layers and the core (base) layer SHOULD be used
>    for SSRC identifier allocation and collision resolution. 

OK, since the spec says that one "SHOULD" use the same SSRC on all layers,
it actually means that they "may" be different.  And if the SSRC's are
different, then it is no longer necssary to transmit each layer to a
different port, when using unicast.  

Therefore, I suggest that the text in the RTP spec which says that different
layers "MUST" be sent on different ports, be changed to "SHOULD" for unicast
RTP.  

For multicast RTP, there are indeed specific operating system problems that
require the use of "MUST".  But those problems don't apply to unicast RTP.
I don't think it is good to impose these kinds of restrictions on the usage
of unicast RTP, unless there is a good reason.

Anders



From rem-conf Tue Jul 13 09:22:53 1999 
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From: "Andrea Basso" <basso@research.att.com>
To: "Ross Finlayson" <finlayson@live.com>
Cc: <rem-conf@es.net>,
	"Mathias Kretschmer" <mathias@research.att.com>,
	<jhs@research.att.com>, <mrc@research.att.com>
Subject: Re: AAC-RTP payload format
Date: Tue, 13 Jul 1999 12:21:39 -0400
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Ross,
thanks for your comments.

>Unfortunately I found the draft rather confusing.  This was due in part to
>my unfamiliarity with AAC (compared to regular MPEG audio), but also due to
>the fact that the draft doesn't make it particularly clear how the data
>that would make up an outgoing RTP packet would be produced by an
>implementation.  In particular, which parts of this data would come
>directly from the AAC data stream, and which parts would be *derived* from
>the AAC data stream (& how)?




While the AAC frames, referred also as "AAC raw_data_blocks" in the draft
come from the AAC stream, the piority information and the priority vector is
computed by the encoder/packetizer and are not integral part  of  the AAC
stream.
Note while priority of the AAC frames is an intrinsic property of the AAC
stream, these priorities can be computed differently.
There are four AAC frame types: long (L) , short (S) , start (T) , and stop
(P). Start frames are used to transition smoothly from long frames to short
ones, and stop frames are used to transition smoothly from short frames to
long ones.

An example algorithm for computing the frame priorities is the following. In
the following very simple scheme, we use only three priority levels: low,
mid, and high. High priority frames comprise all shorts, stops, and starts.
Mid priority frames comprise long frames adjacent to start frames or stop
frames. Low priority frames are those adjacent only to other long frames.

An example:

sequence no: 012345678
frames:     LLLTSPLLL
priorities:  llmhhhmll

where:

L == long frame
T == start frame
S == short frame
P == stop frame

l == low priority
m == mid priority
h == high priority


We believe that, just as there are many MP3 implementations with many
different levels of quality, so could different implementors create
different methods for identifying "priority" as well as repair information
in AAC. Thus it is important that the payload format has
placeholders to carry effectively the priority as well as the repair
information leaving
their optional usage to the specific implementation. We plan to add an
appendix to the draft with a description of algorithms (as the one
mentioned) for priority and repair information computation.


>The draft should have referenced RFC 2250 - the existing RTP payload format
>for MPEG - if only to explain why it is inappropriate for MPEG-2 AAC.
>Presumably the main (only?) reason why RFC 2250 cannot be used is that AAC
>frames look different from regular MPEG audio frames, and so a receiver
>could not reliably distinguish them.
>
>One desireable property of RFC 2250, however, is that an implementation can
>send (or receive) RTP-encapsulated MPEG audio data without having to know
>*anything* about its structure - only where the frame boundaries are.  This
>makes it very easy for a RTP implementation to stream from/to MPEG audio
>data (whether streamed, or in a file).
>
>Is this property also true of your proposed format?  The draft does not
>make this clear.  How much would an implementation of your proposed format
>need to know about the internal structure of AAC data?  In particular,
>where does the "repair data" come from?  Does this come directly from the
>AAC data, or does it need to be 'generated' from the AAC data in some way?
>Again, the draft does not make this clear.
>
>It would be *really* desirable to have a RTP payload format for AAC that
>does not require an implementation to have a detailed knowledge of AAC
>internals.  Perhaps, however, knowledge of AAC internals is necessary for
>including the "repair data"?  If so, then perhaps two payload formats could
>be defined: a simple, non-repairing payload format (that doesn't require
>knowledge of AAC internals), and more complex, reparing format (that
does)??

I agree with you on this. A reference with RFC 2250 will be added in the
next version of the draft.
I think we  do not need two separate payloads definitions if in the current
draft the priority information is made optional as the repair information
is.


Andrea Basso
Jim      Snyder









From rem-conf Tue Jul 13 09:31:51 1999 
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From: "Michael F. Speer" <speer@valathar.Eng.Sun.COM>
Reply-To: "Michael F. Speer" <speer@valathar.Eng.Sun.COM>
Subject: RE: Distinct UDP ports for layered encodings?
To: Stephen Casner <casner@cisco.com>
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        rem-conf@es.net
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> On Tue, 13 Jul 1999, Anders Klemets (Exchange) wrote:

> > I don't think there is any particular reason why one MUST use different port
> > numbers for unicast.  RTP certainly allows multiple unicast flows to be sent
> > to the same port number, as long as the SSRC is different for each flow.  I
> > don't think there are any operating system problems that would force one to
> > use different ports for unicast.

By doing this, this leads to the interpretation that one could mix audio
and video on the same RTP session.  This was something that we are trying
to avoid.

> 
> See section 5.2 for general arguments why not to do multiplexing of
> different RTP sessions using the SSRC id.  But specifically for the
> layered encodings, the spec says:
> 
>    For layered encodings transmitted on separate RTP sessions (see
>    Section 2.4), a single SSRC identifier space SHOULD be used across
>    the sessions of all layers and the core (base) layer SHOULD be used
>    for SSRC identifier allocation and collision resolution. When a
>    source discovers that it has collided, it transmits an RTCP BYE
>    message on only the base layer but changes the SSRC identifier to the
>    new value in all layers.
> 
> So the RTP sessions need to be kept unique, which means separate
> ports.  Or the above gets changed.

I like the wording just the way it is.  Steve Mccanne and I thought about
this for awhile when we first proposed the changes.  The simplest and easiest
solution was to use different ports with the same SSRC.  This allowed one
to easily identify which layer and where it belonged without complicating
implementation with alot of complicated code.  You knew exactly which layer
was which.

> 
> > However, it is quite correct that there are OS reasons why the ports should
> > be different when using multiple multicast groups.
> 
> OK, given they are different for mcast, then unicast should behave the
> same way.

For simplicity, I believe multicast and unicast should behave the same way.
This way you have the chance of building interoperable applications. 
Especially, if the specification is written as such.

> 
> > It does not look like the SDP spec gives a good justification for why
> > different ports would not be allowed.  So my suggestion is that we consider
> > revising the SDP spec.
> 
> Perhaps we can have this be the start of the draft that Mark Handley
> suggested to contain SDP spec errors and updates.  Probably this would
> say someting following Bill Fenner's suggestion of 1-1 mapping or
> punt.
> 							-- Steve
> 
> 
> 


Michael F. Speer
Staff Socket Scientist
Networking and Security Group
Sun Microsystems Laboratories
Sun Microsystems, Inc
Phone: +1-650-786-6368 Fax: +1-650-786-6445	email:	speer@eng.sun.com





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At 09:15 AM 7/9/99 , ravi narayan wrote:
>Ethendranath Bommaiah wrote:
>> 
>> I was wondering if Real G2 server supports RTP for real media (.rm and
>> .ra) ? We have noticed that it does support RTP for other media like
>> .au and .mpeg. However, with .rm and .ra the server returns "Protocol
>> not supported" error for the SETUP message with transport set to
>> RTP/AVP/TCP. We are using Real G2 server 6.0.
>> 
>> Does Real plan to support RTP for .rm and .ra media in the future ?
>> Or, is there a newer version that supports RTP already ?
>> 
>
>from what i have seen, and heard from real, the answer to your question
>is "no". real does not support RTP for realmedia files. from talking to
>them, i get the feeling that they do not plan to support realmedia over
>RTP. the reason they cite is that RDT (their own data transport protocol
>now referred to as RDP i think) is tuned for the special needs of
>realmedia files.

That's correct.  We had floated a proposal for an optimized packet format
as part of the original RTSP proposal back in 1996, but the group decided
(rightly) that it didn't belong in the RTSP spec, and should be a separate
proposal.  Given the arguments that were floating around at that time about
the relative merits of compressing RTP headers independently from the
IP/UDP layer, we decided to drop this proposal.

So, this indicates that perhaps we should dust off this proposal (and
update with what we are currently doing).  If this is the consensus, we'll
be glad to follow up with this.

Thoughts?
Rob




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From rem-conf Tue Jul 13 13:07:14 1999 
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Please advise how to be removed.  I have found myself receiving considerable
spamming since being included in this listserv.

Matthew Halsey
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From rem-conf Tue Jul 13 13:48:17 1999 
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Colin Perkins wrote:

> >       How will a sender know how effective its redundancy sheme is?
>
> I don't think there is a clear way of knowing that from RTCP RR packets
> as they stand. One could envisage an extension to RTCP which made this
> available, if it became necessary.
>
> Colin

In unicast, the sender can, given channel characteristics (loss
rate and jitter) and encodings/FEC used in the recent past,
estimate which packets got lost at the destination (both from
channel loss and from delay exceeding max playout time)
and infer from that a measure of audio/video/etc quality.
Things are a bit trickier in multicast...
-Jean

------------------------------------------------
 Jean Bolot
 650 210 0302
 Ensim Corp. -- "Enabling services on your net"
------------------------------------------------





From rem-conf Tue Jul 13 15:00:53 1999 
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Date: Tue, 13 Jul 1999 13:41:31
To: "Andrea Basso" <basso@research.att.com>
From: Ross Finlayson <finlayson@live.com>
Subject: Re: AAC-RTP payload format
Cc: <rem-conf@es.net>, "Mathias Kretschmer" <mathias@research.att.com>,
        <jhs@research.att.com>, <mrc@research.att.com>
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>While the AAC frames, referred also as "AAC raw_data_blocks" in the draft
>come from the AAC stream, the piority information and the priority vector is
>computed by the encoder/packetizer and are not integral part  of  the AAC
>stream.
>Note while priority of the AAC frames is an intrinsic property of the AAC
>stream, these priorities can be computed differently.

Thanks for the note.  While I'm still somewhat confused by this, I imagine
that future revisions of the draft will clarify things.  In any case, I
think it's very important that the following two properties - true for
other MPEG/RTP formats - will also hold true for any AAC/RTP format:

1/ Given any *preencoded* AAC stream (e.g., one that's stored in a file), a
sender always has enough information to be able to generate a compliant
sequence of AAC/RTP packets, and
2/ Given any compliant sequence of AAC/RTP packets, a receiver always has
enough information (from the packets, and perhaps from other side
information such as a SDP description) to be able to generate a valid AAC
stream (for immediate or later decoding).

	Ross.





From rem-conf Tue Jul 13 15:09:40 1999 
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> >
> > I don't think there is a clear way of knowing that from RTCP RR packets
> > as they stand. One could envisage an extension to RTCP which made this
> > available, if it became necessary.
> >
> > Colin
> 
> In unicast, the sender can, given channel characteristics (loss
> rate and jitter) and encodings/FEC used in the recent past,
> estimate which packets got lost at the destination (both from
> channel loss and from delay exceeding max playout time)
> and infer from that a measure of audio/video/etc quality.
> Things are a bit trickier in multicast...
> -Jean
> 

Consider this situation.

-> Network is losing packets randomly at 10% loss rate.
-> The receiver RTCP RR report will contain this information.

-> Now the sender adds redundancy to the voice stream.
-> Lets assume network still loses 10% i.e. no change in the network
   conditions.
-> Since redundancy was added at the sender the receiver will now
experience packet loss much lesser than 10%. Hence at this point will the
receiver send

a) The new loss rate after reconstruction, which will be much lesser than
10% or

b) The loss rate before reconstruction, which will still be around 10%.

For case a) the sender now knows that the redundancy added is beneficial
and can make appropriate choices.
For case b) the sender still does not know how his redundancy scheme is
working. 

Please clarify this for me.

Sincerely,
	
  
Chinmay V. P.




From rem-conf Tue Jul 13 15:12:49 1999 
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Subject: Re: RTCP Question.
In-reply-to: Your message of "Tue, 13 Jul 1999 18:08:48 EDT." <Pine.GSO.4.05.9907131759080.21211-100000@sunblast>
Date: Tue, 13 Jul 1999 23:12:09 +0100
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--> Chinmay Padhye writes:
>
>> >
>> > I don't think there is a clear way of knowing that from RTCP RR packets
>> > as they stand. One could envisage an extension to RTCP which made this
>> > available, if it became necessary.
>> >
>> > Colin
>> 
>> In unicast, the sender can, given channel characteristics (loss
>> rate and jitter) and encodings/FEC used in the recent past,
>> estimate which packets got lost at the destination (both from
>> channel loss and from delay exceeding max playout time)
>> and infer from that a measure of audio/video/etc quality.
>> Things are a bit trickier in multicast...
>> -Jean
>> 
>
>Consider this situation.
>
>-> Network is losing packets randomly at 10% loss rate.
>-> The receiver RTCP RR report will contain this information.
>
>-> Now the sender adds redundancy to the voice stream.
>-> Lets assume network still loses 10% i.e. no change in the network
>   conditions.
>-> Since redundancy was added at the sender the receiver will now
>experience packet loss much lesser than 10%. Hence at this point will the
>receiver send
>
>a) The new loss rate after reconstruction, which will be much lesser than
>10% or
>
>b) The loss rate before reconstruction, which will still be around 10%.
>
>For case a) the sender now knows that the redundancy added is beneficial
>and can make appropriate choices.
>For case b) the sender still does not know how his redundancy scheme is
>working. 
>
>Please clarify this for me.

The receiver sends (b), the loss rate before reconstruction.

Colin



From rem-conf Tue Jul 13 16:17:23 1999 
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	"Mathias Kretschmer" <mathias@research.att.com>,
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Subject: Re: AAC-RTP payload format
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Ross,

I am a bit confused by your concern.


>>While the AAC frames, referred also as "AAC raw_data_blocks" in the draft
>>come from the AAC stream, the piority information and the priority vector
is
>>computed by the encoder/packetizer and are not integral part  of  the AAC
>>stream.
>>Note while priority of the AAC frames is an intrinsic property of the AAC
>>stream, these priorities can be computed differently.
>
>Thanks for the note.  While I'm still somewhat confused by this, I imagine
>that future revisions of the draft will clarify things.  In any case, I
>think it's very important that the following two properties - true for
>other MPEG/RTP formats - will also hold true for any AAC/RTP format:
>
>1/ Given any *preencoded* AAC stream (e.g., one that's stored in a file), a
>sender always has enough information to be able to generate a compliant
>sequence of AAC/RTP packets, and

>2/ Given any compliant sequence of AAC/RTP packets, a receiver always has
>enough information (from the packets, and perhaps from other side
>information such as a SDP description) to be able to generate a valid AAC
>stream (for immediate or later decoding).


Actually given that the priority and the repair information is *optional*
and it is not vital for the decoding of the AAC stream but helps in its
recovery in presence of packet losses a decoder will be always able to
depacketize and decode the stream by ignoring the optional repair/priority
info.






From rem-conf Tue Jul 13 16:52:35 1999 
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Chinmay Padhye wrote:

> Consider this situation.
>
> -> Network is losing packets randomly at 10% loss rate.
> -> The receiver RTCP RR report will contain this information.
>
> -> Now the sender adds redundancy to the voice stream.
> -> Lets assume network still loses 10% i.e. no change in the network
>    conditions.
> -> Since redundancy was added at the sender the receiver will now
> experience packet loss much lesser than 10%. Hence at this point will the
> receiver send
>
> a) The new loss rate after reconstruction, which will be much lesser than
> 10% or
>
> b) The loss rate before reconstruction, which will still be around 10%.
>
> For case a) the sender now knows that the redundancy added is beneficial
> and can make appropriate choices.
> For case b) the sender still does not know how his redundancy scheme is
> working.
>
> Please clarify this for me.

The receiver sends (b). Note that the sender can then use this
information to estimate the loss rate after reconstruction - essentially
it would simulate a channel with loss rate given by (b), and apply that
model to the stream of main+FEC info it is sending. In fact, this is the
idea behind choosing the optimal allocation of main and FEC information:
the optimal allocation is that which yields the largest utility (audio quality
in this case) at the destination given estimated channel characteristics
(loss rate in this case), bandwidth constraints (computed using some
algorithm based on ECN, TCP-friendliness, or whatever), and max
tolerable additional caused by FEC coding. Refer to our paper with
Don Towsley in this year's Infocom for details
ftp://ftp-sop.inria.fr/rodeo/bolot/99.Audio_fec.ps.gz

-Jean

------------------------------------------------
 Jean Bolot
 650 210 0302
 Ensim Corp. -- "Enabling services on your net"
------------------------------------------------





From rem-conf Tue Jul 13 17:38:14 1999 
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> That's correct.  We had floated a proposal for an optimized packet format
> as part of the original RTSP proposal back in 1996, but the group decided
> (rightly) that it didn't belong in the RTSP spec, and should be a separate
> proposal.  Given the arguments that were floating around at that time about
> the relative merits of compressing RTP headers independently from the
> IP/UDP layer, we decided to drop this proposal.
> 
> So, this indicates that perhaps we should dust off this proposal (and
> update with what we are currently doing).  If this is the consensus, we'll
> be glad to follow up with this.
> 
> Thoughts?
> Rob

Thanks for the response. I was wondering if you could also support
plain RTP along with the above mentioned compressed RTP header approach
(which I presume is proprietary). Also, would it be possible for you to
make your current approach public (in the form of a draft proposal,
perhaps) ?

Thanks,
Ethen




From rem-conf Tue Jul 13 17:46:01 1999 
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From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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To: Stephen Casner <casner@cisco.com>
CC: rem-conf@es.net
Subject: Re: Revision of SSRC collision algorithm?
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Stephen Casner wrote:
> 


> 
> Henning proposes instead that when a host changes its address it
> should leave the SSRC identifier unchanged and that the spec should
> say that receivers SHOULD keep packets from the new address and
> discard packets from the old address when this "collision" of
> addresses with the same SSRC identifier is detected.
> 
> The spec says an implementation MUST include an algorithm similar to
> the one described in Section 8.2.  The algorithm shown in pseudo-code
> keeps the packets from the old address and discards those from the new
> address which is the "right" approach for a genuine collision, based
> on the assumption that one wants to continue hearing the valid source
> that one was hearing before.  If this event was an address change, the
> new source would be accepted once the old source timed out because no
> more packets were received.

I should add one motivation to this mobility case: If done
appropriately, hand-off in this manner can be significantly faster than
with mobile IP, even with binding updates.

However, this would make it useless in the mobility case above, since
the timeout can take minutes.

Note that the case of mobility can be readily distinguished, if the
implementation wants to bother: in a collision, you will get both "old"
and "new" IP addresses in some alternation, in the mobility case, you
will get only the "new" address until the next address change, typically
many seconds away. However, given the rarity of this situation, that it
is not needed for interoperability and since it is unlikely to be widely
implemented, I don't think this needs to be specified in the draft.

Given that this revision is for Draft status, relaxing requirements
seems the safer thing to do. I doubt that many (any?) existing
implementations discard packets as is.

Also, for unicast, collisions can't happen under normal circumstances
(if one is mixing different unicast streams, one should choose different
destination ports), so discarding packets seems unnecessary to begin
with.

> 
> This is not intended to require that the policy be to ignore the new
> source.  A different policy may be chosen and still have the algorithm
> be similar; I added a phrase about this in the most recent draft.
> 
> Regarding replacing the existing pseudo-code with pseudo-C, the syntax
> I used in writing the existing algorithm was taken from an example
> provided at the time by the RFC Editor.  The reason for the pseudo
> code is that Internet Standards are not supposed to include code
> (because they are supposed to specify what happens on the wire, not
> how to implement the protocol).  I can get a new reading on this from
> the AD and the RFC Editor.

With both the current pseudo-code and more C-like code, the same
behavior is specified, so the "what happens on the wire" argument
doesn't seem to apply.

However, code has the advantage that you can actually check it. I would
want to include nothing that ties down an implementation to any
operating system, just allows to allow the use of tools. (If it
increases the number of implementations that actually use collision
handling, so much the better.)

>                                                         -- Steve

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



From rem-conf Wed Jul 14 04:23:08 1999 
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From: "Anders Klemets (Exchange)" <anderskl@exchange.microsoft.com>
To: "'Michael F. Speer'" <speer@valathar.Eng.Sun.COM>, Stephen Casner
	 <casner@cisco.com>
Cc: rem-conf@es.net
Subject: RE: Distinct UDP ports for layered encodings?
Date: Wed, 14 Jul 1999 04:13:33 -0700
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> By doing this, this leads to the interpretation that one could mix audio
> and video on the same RTP session.  This was something that we are trying
> to avoid.

Sure, but the RTP spec nevertheless allows this.  I just checked section
5.2, and it exactly supports my argument.

>From section 5.2 in the latest RTP Internet Draft:

   Separate audio and video streams SHOULD NOT be carried in a single
   RTP session and demultiplexed based on the payload type or SSRC
   fields. 

This actually means that it is allowed to send unlayered streams to the same
port.  So, why is it that that the spec says that layered streams "MUST" be
sent to distinct ports?  I think this is inconsistent.  I don't think that
it is reasonable to have different rules for layered streams vs. unlayered
streams.  So, the "MUST" qualifier ought to be changed to a "SHOULD".

Anders



From rem-conf Wed Jul 14 08:59:36 1999 
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Date: Wed, 14 Jul 1999 17:53:08 +0200
From: Lidia Yamamoto <yamamoto@run.montefiore.ulg.ac.be>
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Subject: Re: Distinct UDP ports for layered encodings?
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Hi,

For me it seems that the restriction on multiple multicast addresses using the
same port number is not only an implementation restriction, but it is also
important to ensure 1 to n multiplexing from the network layer to the transport
layer (we may use one IP address (unicast or multicast) with several ports, but
not vice-versa), instead of n to n multiplexing. So maybe it is a feature
rather than a bug. This would be one more argument for keeping the RTP
specification and changing SDP.

I think the main point is that SDP syntax makes it difficult to express media
in a hierarchical way. I think it would be more clear, for example, if we had
something like this (for simplification, the examples do not show all the
details of the SDP syntax):

# example with one hierarchical media (e.g. layered video with 3 layers)
# using the same IP address and TTL for the 3 sub-media
m=hierarchical 3 # meaning three sub-media within one
c=IPaddr1/TTL1   # meaning IPaddress 1 and TTL1 will be used for the three
                 # sub-media
                 # the next 3 media descriptions will describe each sub-media
m=video port1
m=video port2
m=video port3

# example with 2 layers, each using different IP addresses, ports and TTLs.
m=hierarchical 2
m=video port1
c=IPaddr1/TTL1
m=video port2
c=IPaddr2/TTL2

# illegal:
m=hierarchical 2
m=video port1
c=IPaddr1/TTL1
m=video port1
c=IPaddr2/TTL2

# multiple hierarchical levels, for more exotic applications:
m=hierarchical 2
m=audio port1
m=hierarchical 2
m=video port2
m=application port3

Something like this would make the matching of transport addresses with network
addresses clear in SDP, and it could be used in more generic applications,
instead of only patching SDP to work with layered transmission.

Lidia.


"Anders Klemets (Exchange)" wrote:

> > By doing this, this leads to the interpretation that one could mix audio
> > and video on the same RTP session.  This was something that we are trying
> > to avoid.
>
> Sure, but the RTP spec nevertheless allows this.  I just checked section
> 5.2, and it exactly supports my argument.
>
> >From section 5.2 in the latest RTP Internet Draft:
>
>    Separate audio and video streams SHOULD NOT be carried in a single
>    RTP session and demultiplexed based on the payload type or SSRC
>    fields.
>
> This actually means that it is allowed to send unlayered streams to the same
> port.  So, why is it that that the spec says that layered streams "MUST" be
> sent to distinct ports?  I think this is inconsistent.  I don't think that
> it is reasonable to have different rules for layered streams vs. unlayered
> streams.  So, the "MUST" qualifier ought to be changed to a "SHOULD".
>
> Anders




From rem-conf Wed Jul 14 23:42:15 1999 
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From: Mark Llacuna <mllacuna@x.Holontech.com>
To: rem-conf@es.net
Subject: HEader compression questions ( RTP )
Date: Wed, 14 Jul 1999 23:39:10 -0700
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Hello everyone,

	I have  a few questions as to how an implementation of RTP header
compression would behave in case an out-of-order packet is received.  The
RFC (2508) specifies that if for example the first three compressed packets
were OK but instead of the fourth packet, the fifth packet was received by
the decompressor, the decompressor would invalidate the context and send a
CONTEXT_STATE packet.  The question therefore is:

	a) does the decompressor keep the first three packets?
	b) does the compressor send the fourth packet ( based on the
sequence number in the CONTEXT STATE packet )? and resend the fifth packet?
	c) if for example, there are several packets ( say 5,6,7,8 ) that
were received in the interim ( before a new FULL HEADER ) is received, what
is the heuristic used for resending a CONTEXT_STATE packet?  Or do you
actually just set a timer until a FULL_HEADER is received and resend a
CONTEXT-STATE?
	d) in case you use timers for c), what would be a recommended
timeout period?

	Any answers would be really useful.

Regards,
Joey Salanga




From rem-conf Wed Jul 14 23:51:03 1999 
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Hi Joey,

Answre to your problem with your example :

If decompressor receives fifth pkt insted of fourth it needs to invalidate the
context (RFC 2508) by sending CONTEXT_STATE packet. Once the context is invalidated

what is acceptable to this context is a FULL_HEADER packet (or terminate the
context if there is no FULL_HEADER).
If the context is waiting for the FULL_HEADER packet and mean while compressed
packets
arrive, those pkts will be rejected by the decompressor.
or other implementation could be save the other pkts (5,6,7,8) and wait for
pkt 4 for a period of n units. (But there are many issues to resolve then)
If it does not receive pkt 4 within this period then invalidate the context.
Once the context has been invalidated next acceptable pkt is FULL_HEADER.


Second part of the question ..

How long you should wait for a FULL_HEADER pkt ?

For this prblme there can be a timer based soln or counter
of number of compressed pkt based soln.
Second soln may be more attractive. Prblm with timer soln is
you canot predict when the source will send you next RTP pkt.
With the second soln, if you receive n compressed pkts
after invalidating the context you could resend the CONTEXT_STATE pkt.

I donot want to suggest any number for the timeout period ..

Hope my answers will help you ..
More doubts are welcome ..
You could subscibe to rem-conf@es.net also for more answers.

Rgds
GVJ


Anand Kumar Goenka wrote:

> I gues u could answer this question . ...
>
> Anand(K.G.)
>
>   ------------------------------------------------------------------------
>
> Subject: Header Compression Error Recovery
> Date: Wed, 14 Jul 1999 17:28:05 -0700
> From: Mark Llacuna <mllacuna@x.holontech.com>
> To: "'iptel@lists.research.bell-labs.com'" <iptel@lists.research.bell-labs.com>
>
> Hello everyone,
>
>         I have  a few questions as to how an implementation of RTP header
> compression would behave in case an out-of-order packet is received.  The
> RFC (2508) specifies that if for example the first three compressed packets
> were OK but instead of the fourth packet, the fifth packet was received by
> the decompressor, the decompressor would invalidate the context and send a
> CONTEXT_STATE packet.  The question therefore is:
>
>         a) does the decompressor keep the first three packets?
>         b) does the compressor send the fourth packet ( based on the
> sequence number in the CONTEXT STATE packet )? and resend the fifth packet?
>         c) if for example, there are several packets ( say 5,6,7,8 ) that
> were received in the interim ( before a new FULL HEADER ) is received, what
> is the heuristic used for resending a CONTEXT_STATE packet?  Or do you
> actually just set a timer until a FULL_HEADER is received and resend a
> CONTEXT-STATE?
>         d) in case you use timers for c), what would be a recommended
> timeout period?
>
>         Any answers would be really useful.
>
> Regards,
> Joey Salanga
>
> ---------
> This message came from the IETF IPTEL Working Group Mailing List.



--------------AD4A40619275D7F80B7C8C90--




From rem-conf Thu Jul 15 02:00:55 1999 
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Date: Thu, 15 Jul 1999 14:32:59 +0530
From: Venkata Narasimha Murthy Nanduri <nvnmurthy@seeyes.co.in>
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Hi,

1) Do U have any idea about the implementation of RTP/UDP/IP
header compression using Frame relay as the link level protocol?

2) Are there any protocol numbers defined for CRTP compressed
    packets(COMPRESSED_RTP_8, COMPRESSED_UDP_8, etc) for frame relay?

   They have been defined for PPP in RFC 2509..

Answers to any of the questions will be useful..





Thanks in advance
Murthy NVN






From rem-conf Thu Jul 15 02:06:44 1999 
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From: Venkata Narasimha Murthy Nanduri <nvnmurthy@seeyes.co.in>
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Hi,

While sending the COMPRESSED_RTP packet, in RFC2508, it has been
mentioned that an addional byte will be transmitted if MSTI = 1111. The
additional byte is mentioned in the packet format as four bits M', S',
T', I' and 4 bits as CC.
In the explanation it has been given that the first nibble(M', S', T',
I') corresponds to the real values of the four bits MSTI.
What is exactly meant by M', S', T', I'?
What is meant by the real values of MSTI?

Thanks
Murthy NVN




From rem-conf Thu Jul 15 02:19:58 1999 
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Subject: Re: [Fwd: Header Compression Error Recovery]
References: <378DE396.3C42ED62@miel.mot.com> <378DEA71.AF02E76A@miel.mot.com> <378DA29E.2008BD4F@seeyes.co.in>
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Hi,

Answer to your questions ..

Venkata Narasimha Murthy Nanduri wrote:

> Hi John,
>
> If we consider that first three packets have been received by the receiver, fourth one
> missing and fifth onwards fine...
>
> Then as per my understanding, we are planning to do this way. Is it correct?
>
> 1) Since 4th packet is not received by the decompressor, it will invalidate the context
> and send the Context state packet to the compressor, with the CID and the sequence
> number of the latest received packet in sequnce (i.e., 3.) drop the packets with seq.
> number greater than four for that context until a FULL_HEADER packet is received for
> that context.

this is correct

>
>
> 2) Now, as the context state packet is received by the compressor, it should send the
> missing packet
> (i.e., packet with seq. no 4) in FULL_HEADER format and the remaining packets of the
> context as stored in the context table at the compressor table.
>

I dont  know, why you need to resent the pkt which is lost ...
That means CRTP is taking care of  retransmission
Is it correct  ?
rather I would say
when the compressor module receives a CONTEXT_STATE packet
it can send the next pkt of the flow as the FULL_HEADER packet with link
sequence number 4 in the example and all consecutive pkts as compressed pkts.
I think retransmission should  be done by upperlayer protocols.

>
> Could you please let me know whether this email is from Bangalore office or Hyderabad
> office of Motorola?

Bangalore office

>

Rgds

GVJ




From rem-conf Thu Jul 15 02:31:24 1999 
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Date: Thu, 15 Jul 1999 15:02:12 -0300
From: GeeVarghese John <gvjohn@miel.mot.com>
Organization: Motorola India Electronics Ltd.
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Subject: Re: Reg. extension header in CRTP
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Hi,

What I understant is

If we need to send the  CC value to decompressor
we set 1111 for the first MSTI nibble and
keep the extra one byte to encode real delta values of MSTI. and next 4
bits to
store the new CC value.  By setting 1111 for the MSTI bits we are
indicating that
CC value is encoded in the compressed pkt.

Rgds
GVJ

Venkata Narasimha Murthy Nanduri wrote:

> Hi,
>
> While sending the COMPRESSED_RTP packet, in RFC2508, it has been
> mentioned that an addional byte will be transmitted if MSTI = 1111. The
> additional byte is mentioned in the packet format as four bits M', S',
> T', I' and 4 bits as CC.
> In the explanation it has been given that the first nibble(M', S', T',
> I') corresponds to the real values of the four bits MSTI.
> What is exactly meant by M', S', T', I'?
> What is meant by the real values of MSTI?
>
> Thanks
> Murthy NVN




From rem-conf Thu Jul 15 04:01:45 1999 
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Subject: Re: [Fwd: Header Compression Error Recovery]
References: <378DE396.3C42ED62@miel.mot.com> <378DEA71.AF02E76A@miel.mot.com> <378DA29E.2008BD4F@seeyes.co.in> <378E1F91.35B3A7FF@miel.mot.com>
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Hi John,

Yes, As I have mentioned, CRTP does the retransmission.
If we don't retransmit the lost packets, there is no necessity for the compressor to store
the packets that
are transmitted.

Let us take the scenario at compressor

seq no:1;CID:1, FULL_HEADER sent and stored in  a table
seq no:2;CID:1, COMPRESSED_UDP sent and stored in  a table
seq no:3;CID:1, COMPRESSED_RTP  sent and stored in  a table
seq no:4;CID:1, COMPRESSED_UDP sent and stored in  a table
seq no:5;CID:1, COMPRESSED_RTP  sent and stored in  a table
seq no:6;CID:1, COMPRESSED_UDP sent and stored in  a table
seq no:7;CID:1, COMPRESSED_RTP  sent and stored in  a table
seq no:8;CID:1, COMPRESSED_UDP sent and stored in  a table
seq no:9;CID:1, COMPRESSED_RTP  sent and stored in  a table
seq no:10;CID:1, COMPRESSED_UDP sent and stored in  a table

Now If the fourth packet(packet with seq. no 4) could not reach the decompressor,

A context state packet has been received by the compressor indicating that the packet with
seq. no 4 is missing..

Then as per u, the compressor should send the next packet of the flow(seq. no 11 as per above
data ) as full header packet by changing the seq. no from 11 to 4 and transmit the remaining
packets as compressed packets. But because of this, the packes from seq. no 4 to 10 are lost(
Is this what U intended to say?)

But if u take the case, where the compressor, sends the seq.no 4 packet as full header and
remaining packets after that are sent as they are stored in the table at the comressor side.
This way, we will not be losing any packets..

As far as the retransmission by the upper layer protocols are concerned, that can be managed
by the compressor part of CRTP itself like this:

1) Since the compressor stores the compressed and uncompressed packets transmitted in a
table...

It can form an IP packet( like the upper layer protocols) if full header packet has to be
transmitted.
It can form the packet format of COMPRESSED_UDP or COMPRESSED_RTP if required to transmit..

Please send your comments along..

Murthy NVN





GeeVarghese John wrote:

> Hi,
>
> Answer to your questions ..
>
> Venkata Narasimha Murthy Nanduri wrote:
>
> > Hi John,
> >
> > If we consider that first three packets have been received by the receiver, fourth one
> > missing and fifth onwards fine...
> >
> > Then as per my understanding, we are planning to do this way. Is it correct?
> >
> > 1) Since 4th packet is not received by the decompressor, it will invalidate the context
> > and send the Context state packet to the compressor, with the CID and the sequence
> > number of the latest received packet in sequnce (i.e., 3.) drop the packets with seq.
> > number greater than four for that context until a FULL_HEADER packet is received for
> > that context.
>
> this is correct
>
> >
> >
> > 2) Now, as the context state packet is received by the compressor, it should send the
> > missing packet
> > (i.e., packet with seq. no 4) in FULL_HEADER format and the remaining packets of the
> > context as stored in the context table at the compressor table.
> >
>
> I dont  know, why you need to resent the pkt which is lost ...
> That means CRTP is taking care of  retransmission
> Is it correct  ?
> rather I would say
> when the compressor module receives a CONTEXT_STATE packet
> it can send the next pkt of the flow as the FULL_HEADER packet with link
> sequence number 4 in the example and all consecutive pkts as compressed pkts.
> I think retransmission should  be done by upperlayer protocols.
>
> >
> > Could you please let me know whether this email is from Bangalore office or Hyderabad
> > office of Motorola?
>
> Bangalore office
>
> >
>
> Rgds
>
> GVJ




From rem-conf Thu Jul 15 08:11:33 1999 
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From rem-conf Thu Jul 15 09:19:26 1999 
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Date: Thu, 15 Jul 1999 09:13:06 -0700
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RFC1889 defines "Monitor" as "an application that receives RTP packets
sent by participants in an RTP session, in particular the reception
reports, and estimates the current quality of service for distribution
monitoring, fault diagnostics and long-term statistics".

In my master thesis I am working on a RTP Monitor written in Java using
Sun's Java Media Framework.

Could anyone send me references to RTP monitor applications ?

Francisco Afonso
Naval Postgraduate School





From rem-conf Thu Jul 15 09:45:29 1999 
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RealNetwoks is interested in participating in RTP interoperability
testing as was discussed in yesterday's AVT meeting. We have currently
tested G2 with vic/vat and the ICAST server ( which I believe is based
on vic/vat) and ICAST viewer.

I am interested in knowing how, when, where, this event might take
place.

Thanks,
- Virginia




From rem-conf Fri Jul 16 16:35:26 1999 
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Subject: H.GCP draft?
To: rem-conf@es.net
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Does anyone have the ITU.T Draft Recomm. H.GCP?
Anything that relates to, or describes the 
(GCP) protocol is also welcome.

Thanks

Hassan Naser
hnaser@comm.utoronto.ca



From rem-conf Fri Jul 16 22:56:32 1999 
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From: Gunnar Hellstrom <gunnar.hellstrom@omnitor.se>
Subject: draft-hellstrom-avt-rtp-text-01.txt Result of Oslo discussion.
Cc: textphone@lsv.pi.se (Text Telephone Forum), mickey.nasiri@etx.ericsson.se
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The draft rtp specification for text conversation was discussed in IETF AVT
last Thursday in Oslo.=20
It was regarded quite stable.
Before settling it completely, I want to discuss a few issues:

1. Timestamp clock frequency. I had put a low figure 10 Hz in the
specification. It was questioned and proposed that a figure of some
thousands should be selected.=20
I am willing to accept that, and propose that 1000 Hz is used. A
millisecond per tick is a handy figure.=20
This will however lead to the timestamp offset in the redundancy data
headers that use only 14 bits, flip around in only 16 seconds. If correct
timing information is wanted, the data flush procedure described in next
section should be used.

2. Retransmission of text before a pause.
Text does not flow continously, but only when a user enter text. Therefore,
a recommended procedure could specify that for the redundancy scheme, it is
advisable to send the ISO 10646 synch character FEFF after some inactivity
timer to flush out redundant data - see section 3.4.=20

3. Recommended procedures section.
In the draft that accompanies this mail, section 3 documents the
recommended procedures for using the payload format. Does the group want it
to be kept there or shall it be moved to the specification specifying use
of this RTP Payload? ( The first user being H.323 Annex G)=20

Any other comments are welcome.

A new version of the document is attached. ( what habits do you have
regarding marking modifications between versions in the text formatted
documents? )

Gunnar Hellstr=F6m

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Internet Engineering Task Force                                   AVT WG
Internet Draft                                               Gunnar=
 Hellstr=F6m
draft-hellstrom-avt-rtp-text-01.txt                            L M Ericsson
July 17, 1999
Expires: Jan 17, 2000


                      RTP Payload for Text Conversation

STATUS OF THIS MEMO

   This document is an Internet-Draft and it is in full conformance with all
   provisions of section 10 of RFC2026. Internet-Drafts are working
   documents of the Internet Engineering Task Force (IETF), its areas,
   and its working groups.  Note that other groups may also distribute
   working documents as Internet-Drafts.

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

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

   The list of Internet-Draft Shadow Directories can be accessed at=20
   http://www.ietf.org/shadow.html

   Distribution of this document is unlimited.


ABSTRACT

   This memo describes how to carry text conversation session contents in=
 RTP
   packets. Text conversation session contents is specified in ITU-T
   Recommendation T.140 [1].=20

   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.






Hellstr=F6m                                                   [Page 1]



Internet Draft

1 Introduction

   This memo defines a payload type for carrying text conversation session=
=20
   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. Text in text conversation sessions=
 is

   sent as soon as it is available, or with a small delay of up to=20
   0.5 seconds for buffering.
   The text is supposed to be entered by human users from a keyboard,
   handwriting recognition, voice recognition or any other input method. The
   rate of character entry is usually at a level of a few characters per=
 second
   or less. Therefore, the expected number of characters to transmit is low.
   Only one or a few new characters are expected to be transmitted with each
   packet.

   T.140 specifies that text and other T.140 elements shall be transmitted=
 in =20
   ISO 10 646-1 code with UTF-8 transformation. That makes it easy to=
 implement
   internationally useful applications, and to handle the text in modern =20
   information technology environments. The most common T140 PDU is a=
 character
   of ISO 10646 text, UTF-8 coded, submitted from T.140 without any extra=
 framing.=20
=20
   T.140 requires the transport channel to provide characters without
   duplication and in original order.=20
   Text conversation users expect that text will be delivered with no or a=
 low
   level of lost information. If lost information can be replaced with a=
 special=20
   marker, the willingness to accept loss is expected to be higher.
=20
   Therefore a mechanism based on RTP is specified here. It gives text=
 arrival  =20
   in correct order, without duplications, with an optional possibility to =
=20
   repeat data for redundancy to lower the risk of loss, and a mechanism=
 that=20
   reveals loss and therefore can insert a marker for lost text in the=
 received
   stream. Since packet overhead is usually much larger than the T.140=
 contents,
   the increase in channel load by the redundancy scheme is minimal.=20

2. Usage of RTP

   When an unreliable transport of T.140 text session data in a packet=
 network is
   selected, RTP with payload as described in this specification should be=
 used.
   T.140 data is submitted for transmission in a number of complete protocol=
 data
   units (T140 PDU). A T140block contains one or more T140 PDU:s.=20
  =20
   A text conversation RTP packet contains an RTP header, a Payload
   Header, optional redundant data fields, and the new (primary) T.140 data
   field.
  =20
2.1 RTP packet header
  =20
   Each RTP packet starts with a fixed RTP header. The following fields of=
 the
   RTP fixed header are used for T.140 text streams:

Hellstr=F6m                                                   [Page 2]



Internet Draft                                 =20

Payload Type (PT): Payload type allocation is dynamic. If redundancy is=
 used,
   the Payload Type shall indicate redundancy ("RED") according to RFC 2198=
 [3].
   If no redundancy is used, the Payload Type shall specify the primary
   T.140 data "T140" payload format.=20

Sequence number:  The Sequence Number shall be increased with one for each
   new transmitted packet. It is used for detection of packet loss and=
 packets
   out of order, and can be used in the process of retrieval of redundant=
 text,=20
   reordering of text and marking missing text.

Timestamp: The RTP Timestamp encodes the approximate sampling instance of =
=20
   the primary text in the packet. A clock frequency of 1000 Hz shall be=
 used.=20
   No sequential packets should use the same timestamp.=20

2.2 Additional headers

   When redundant data is used, additional headers specify the redundant=20
   text fields. They are specified according to RFC 2198 and use dynamic=20
   payload type "T140".

2.3 T.140 Text structure

   T.140 text is UTF-8 coded as specified in T.140 with no extra framing.=
 When
   using the format with redundant data, the transmitting entity may select=
 a
   number of T.140 block generations to retransmit in each packet. A higher=
=20
   number introduces better protection against loss of text.=20
   A maximum of 6 generations may be used.

3. Proposed procedures

   This section contains proposed procedures for usage of the payload=
 format.
   Based on the information in the received packets, the receiver can:
      -reorder received text out of order.
      -mark where text is missing because of packet loss.
      -compensate for lost packets by using redundant data.

3.1 Proposed basic procedure  =20

   On reception, the RTP sequence number is compared with the sequence=
 number of
   the last correctly received packet.=20
   If they are consecutive, only the most recent t140block is retrieved from=
 the
   received packet to the receiving T.140 instance.=20

3.2 Proposed procedure for compensation for lost packets.

   For reduction of data loss in case of packet loss, redundant data may be
   included in the packets.
   If network conditions are not known, it is recommended to use two=
 generations
   of T.140 blocks in the packets. If there is a gap in the RTP sequence=
 numbers,=20
   and redundancy coded T.140blocks are available in the packet, older=20


Hellstr=F6m                                                   [Page 3]

Internet Draft                  		=09

   t140blocks are retrieved from the packet up to the point where the=20
   sequence number is consecutive to the last correctly received packet=20
   or no more t140block are available in the received packet.

   If there is still a gap in the sequence, one T.140 missing data marker
   ( Unicode character 2607 "Lightning") is inserted, UTF-8 coded, replacing
   each missing t140block.
=20
   All t140blocks retrieved in this way are submitted to the receiving T.140
   instance.

3.3 Proposed procedure for compensation for packets out of order.

   For protection against packets arriving out of order, the following=
 procedure
   may be implemented in the receiver.
   If analysis of a received packet reveals gaps in the sequence, the=
 received
   packet can be kept in a buffer before submission to the T.140 layer.=20
   If a packet arrives with a t140block belonging to the gap, the gap is=
 filled
   with the contents of this block.=20
   If all gaps are filled or the packet is kept in the buffer for 0.5=
 seconds,
   valid T140blocks in the buffered packet are retrieved and submitted to=
 the
   T.140 layer. For each t140block still missing, a T.140 missing data=
 marker
   (Unicode character 2607 "Lightning") is inserted.=20

3.4 Transmission during "silent periods" when redundancy is used.

   When using the redundancy transmission scheme, and there is nothing more
   to transmit from T.140, the latest t140block has a risk of getting old=20
   before it is transmitted as redundant data. The result is less useful
   protection against packet loss at the end of a text input sequence.=20
   For cases where this should be avoided, the ISO 10 646 synchronization
   character FEFF may be transmitted (UTF-8 coded) after some time of
   inactivity.=20

4. Examples

   This is an example of a T140 RTP packet without redundancy.

    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=3D0  |M| T140 dyn PT |   sequence number             |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |              timestamp (1000Hz)                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |           synchronization source (SSRC) identifier            |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   +                T140 encoded data (PT=3Ddynamic)                 +
   |                                   =20                           |
   +                                               +---------------+
   |                                               |              =20
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+              =20

Hellstr=F6m                                                   [Page 4]

Internet Draft                  	=09

    This is an example of an RTP packet with two t140 blocks using
    redundancy coding.
    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=3D0  |M| "RED" PT    |   sequence number of primary  |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |              timestamp  of primary encoding "P"               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |           synchronization source (SSRC) identifier            |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |1| T140 dyn. PT|  timestamp offset of "R"  | "R" block length  |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |0| T140 dyn. PT|                                               |
   +-+-+-+-+-+-+-+-+                                               +
   |                                                               |
   +            "R" T140 encoded redundant data (PT=3Ddynamic T140)  +
   |                (6 bytes, not to scale)                        |
   +                                               +---------------+
   |                                               |               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+               +
   |             "P" T140 encoded primary data (PT=3Ddynamic T140)   |
   +                (4 bytes, not to scale)                        +
   +                                               +---------------+
   |                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+



5.  Security Considerations.
    Since the intention of the described payload format is to carry text in=
 a
    text conversation, security measures in the form of encryption is of
    importance. The amount of data in a text conversation session is low and=
=20
    therefore any encryption method may be selected and applied to T.140
    session contents or to the whole RTP packets.=20
    When redundant data is included, the same security considerations as for
    RFC 2198 apply.=20

6.  Author's Address

   Gunnar Hellstr=F6m
   L M Ericsson
   Sweden

   e-mail: gunnar.hellstrom@omnitor.se
   Tel:    +46 708 204 288
   Fax:     +46 8 556 002 06





Hellstr=F6m                                                   [Page 5]

Internet Draft                  			Expires 17 Jan 2000

7 References

[1] ITU-T Recommendation T.140 (1998) - Text conversation protocol for
    multimedia application.

[2] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson,=20
   "Real Time Transport Protocol", RFC 1889.


[3] C. Perkins, I. Kouvelas, V. Hardman, M. Handley, and J. Bolot,
   "RTP payload for redundant audio data," RFC 2198, Internet
   Engineering Task Force, Sept. 1997.



Hellstr=F6m                                                   [Page 6]

--=====================_932183403==_
Content-Type: text/plain; charset="us-ascii"


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

E-mail gunnar.hellstrom@omnitor.se
Tel +46 751 100 501
fax +46 8 556 002 06

--=====================_932183403==_--




From rem-conf Tue Jul 20 15:17:51 1999 
From rem-conf-request@es.net Tue Jul 20 15:17:49 1999
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Date: Tue, 20 Jul 1999 15:05:04 -0700 (Pacific Daylight Time)
From: Stephen Casner <casner@cisco.com>
To: O.Hodson@cs.ucl.ac.uk
cc: rem-conf@es.net
Subject: Re: comment on draft-ietf-avt-profile-new-06.txt
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Orion,

A somewhat delayed follow-up:

> The latest avt profile draft again has no payload formats are
> specified for the 2,3,5 bits per sample coding of G726 ADPCM.  There
> are only two issues - the packetization interval and the sample
> packing order.  Both are trivial but should be stated for the sake
> interoperability.

Actually, there is a difference in this profile draft.  Since no
payload format has been specified for the 2,3,5 bits per sample coding
of G726 ADPCM, only the name "G726-32" is included and the others have
been removed.  Similarly, G.727 was removed entirely.

> 20 ms packetization interval is common for most sample based coders
> and results in all G726 encoding formats being byte aligned.  

That's fine; it's not all that important a parameter anyway (it is
mostly just a recommendation).

> The sample packing order is slightly complicated because the existing
> G726-32 packs samples in the order s1-s0-s3-s2 and there is no obvious
> extension to 3 and 5 bit per sample schemes since they do not align a
> single octet boundary.  The other sample based encodings pack samples
> in the order with which they were generated.  If there is no objection
> from the list I would like to propose that the 2,3, and 5 bit per
> sample encodings are packed in the order of sample generation and we
> leave the 4 bits per sample as is.

Actually, I believe you are just not recognizing that the G726-32
format is packed in little-endian order.  I don't have the G.726
reference accessible at the moment, but perhaps it specifies this
little-endian packing.  There is a straightforward extension to
this method for packing the samples of other sizes: you start at the
LSB end of a byte, and put in bits from the LSB end of the sample.  If
the whole sample doesn't fit into a byte, the MSB's of the sammple go
into the LSB's of the next byte.

> At least one application already incorporates these codecs, not
> because they are cutting edge, but because source code is publicly
> available and (relatively) unencumbered.

This application includes all of the sample sizes?
							-- Steve





From rem-conf Wed Jul 21 08:54:12 1999 
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Hi,
	I hope someone can help me with the following questions:

1.) Can a compund RTCP packet contain multiple SRs. Only
    multiple RRs are discussed in the spec.

2.) Is the 5 second per RTCP on a per channel basis. i.e.
    if I'm sourcing 25 streams, say, should I be sending
    one RTCP SR packet every 5 seconds, or one every 1/5th
    of a second?

3.) For multiple bi-directional channels, can a compund 
    RTCP packet contain multiple sender info sections
    and report blocks?

	Anxious bosses of inquiring minds want to know ;^)

[\] Robert Wood   

The St. Lawrence river - fresh, warm, visible diving.

mailto:robert_wood@mitel.com http://www.magma.ca/~rgwood/



From rem-conf Wed Jul 21 09:36:35 1999 
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Robert,
Unless something has changed since I last read the RTP specification, RTP
was not intended to demultiplex multiple streams on the same "channel".
What you usually see is each stream is given its own data channel (mcast
address and port) and own control channel (by convention usually the same
mcast address and the +1 port). Therefore if you were sourcing 2 audio
channels and 1 video channel, you would also have 3 RTCP channels, each
one corresponding to one of the data channels. So there would never be a
need for a compound sender report, the source would send a single sender
report on each RTCP channel which would correspond to one and only one of
the data channels. The application or applications sourcing these streams
would do a separate calculation on the frequency of RTCP packets based on
on each individual channel.

I hope that helps!

- David Makofske


On Wed, 21 Jul 1999, Robert Wood wrote:

> Hi,
> 	I hope someone can help me with the following questions:
> 
> 1.) Can a compund RTCP packet contain multiple SRs. Only
>     multiple RRs are discussed in the spec.
> 
> 2.) Is the 5 second per RTCP on a per channel basis. i.e.
>     if I'm sourcing 25 streams, say, should I be sending
>     one RTCP SR packet every 5 seconds, or one every 1/5th
>     of a second?
> 
> 3.) For multiple bi-directional channels, can a compund 
>     RTCP packet contain multiple sender info sections
>     and report blocks?
> 
> 	Anxious bosses of inquiring minds want to know ;^)
> 
> [\] Robert Wood   
> 
> The St. Lawrence river - fresh, warm, visible diving.
> 
> mailto:robert_wood@mitel.com http://www.magma.ca/~rgwood/
> 
> 




From rem-conf Wed Jul 21 22:12:08 1999 
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Robert Wood wrote:
> 
> Hi,
>         I hope someone can help me with the following questions:
> 
> 1.) Can a compund RTCP packet contain multiple SRs. Only
>     multiple RRs are discussed in the spec.

By definition, a sender in an RTP session is generating a single media
stream. It sends RTCP packets for the stream it generates, which means
one SR per RTCP packet.

> 
> 2.) Is the 5 second per RTCP on a per channel basis. i.e.
>     if I'm sourcing 25 streams, say, should I be sending
>     one RTCP SR packet every 5 seconds, or one every 1/5th
>     of a second?

That depends what you mean by channel. If each of the 25 streams is a
separate RTP session (separate session means different address/port),
the RTCP interval is independent for each stream. For each sender, RTCP
would be generated, on average, once every 5 seconds. Note that RTCP
transmission is randomized and is not with a fixed 5 second period.

If, however, there is a single RTP multicast session, and the 25 streams
are 25 separate senders in the same session, the RTCP bandwidth is
divided among them. This means that the average RTCP period from each
increases, possibly by a factor of 25 (it could be less depending on
whether the avg RTCP packet size divided by the RTCP sender bandwidth is
more or less than 5 seconds). 

> 
> 3.) For multiple bi-directional channels, can a compund
>     RTCP packet contain multiple sender info sections
>     and report blocks?

Assuming "multiple bi-directional channels" means multiple unicast
sessions between the same endpoints, no. RTCP feedback is for the
session it is sent on.

-Jonathan R.

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



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--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		: Sampling of the Group Membership in RTP
	Author(s)	: J. Rosenberg, H. Schulzrinne
	Filename	: draft-ietf-avt-rtpsample-05.txt
	Pages		: 10
	Date		: 21-Jul-99
	
In large multicast groups, the size of the group membership table
maintained by RTP (Real Time Transport Protocol) participants may
become unwieldy, particularly for embedded devices with limited
memory and processing power. This document discusses mechanisms for
sampling of this group membership table in order to reduce the memory
requirements. Several mechanisms are proposed, and the performance of
each is considered.
 
 
   NOTE:
 
   This is to inform you that Lucent Technologies has applied for and/or
   has patent(s) that relates to the binning algorithm which is a part
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   This declaration is being made pursuant to the provisions of IETF IPR
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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		: An RTP Payload Format for Generic Forward Error 
                          Correction
	Author(s)	: J. Rosenberg, H. Schulzrinne
	Filename	: draft-ietf-avt-fec-07.txt
	Pages		: 22
	Date		: 21-Jul-99
	
This document specifies a payload format for generic forward error
correction of media encapsulated in RTP. It is engineered for FEC
algorithms based on the exclusive-or (parity) operation. The payload
format allows end systems to transmit using arbitrary block lengths
and parity schemes. It also allows for the recovery of both the pay-
load and critical RTP header fields. Since FEC is sent as a separate
stream, it is backwards compatible with non-FEC capable hosts, so
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Internet-Drafts are also available by anonymous FTP. Login with the username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-ietf-avt-fec-07.txt".

A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html 
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
	mailserv@ietf.org.
In the body type:
	"FILE /internet-drafts/draft-ietf-avt-fec-07.txt".
	
NOTE:	The mail server at ietf.org can return the document in
	MIME-encoded form by using the "mpack" utility.  To use this
	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
	a MIME-compliant mail reader.  Different MIME-compliant mail readers
	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
	up into multiple messages), so check your local documentation on
	how to manipulate these messages.
		
		
Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.

--NextPart
Content-Type: Multipart/Alternative; Boundary="OtherAccess"

--OtherAccess
Content-Type: Message/External-body;
	access-type="mail-server";
	server="mailserv@ietf.org"

Content-Type: text/plain
Content-ID:	<19990721075419.I-D@ietf.org>

ENCODING mime
FILE /internet-drafts/draft-ietf-avt-fec-07.txt

--OtherAccess
Content-Type: Message/External-body;
	name="draft-ietf-avt-fec-07.txt";
	site="ftp.ietf.org";
	access-type="anon-ftp";
	directory="internet-drafts"

Content-Type: text/plain
Content-ID:	<19990721075419.I-D@ietf.org>

--OtherAccess--

--NextPart--





From rem-conf Thu Jul 22 06:33:37 1999 
From rem-conf-request@es.net Thu Jul 22 06:33:36 1999
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Message-ID: <061c01bed446$6fe3ec80$5f1bcf87@research.att.com>
From: "Timur Friedman" <timur@research.att.com>
To: <rem-conf@es.net>, "Robert Wood" <robert_wood@Mitel.COM>
References: <3795EAE6.1505@mitel.com>
Subject: Re: RTCP Sender reports
Date: Thu, 22 Jul 1999 09:30:58 -0400
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Robert Wood wrote:
> 1.) Can a compund RTCP packet contain multiple SRs. Only
>     multiple RRs are discussed in the spec.

As others have pointed out, a sender only reports on itself, qua sender,
once.  Nonetheless, the short definitions of SR and RR (6.1) do make it
reasonable to wonder what a sender should do in order to send more than 31
reception report blocks:

        SR: Sender report, for transmission and reception statistics
             from participants that are active senders

        RR: Receiver report, for reception statistics from participants
             that are not active senders

On its face this means that RRs should not be sent by active senders.  This
appears to contradict the later statements (6.1, 6.4) that RRs may be
stacked after an initial SR.  A close textual reading is necessary to
conclude without contradiction that even though RRs are "for" reception
statistics from participants that are not active servers, they can be "for"
other things as well.

These SR and RR definitions are likely the place most people look to remind
themselves of the purpose of these packets, as they are the first time SRs
and RRs are mentioned in the spec.  It would be good if they were not
misleading in this way.  I suggest the following wording:

        RR: Receiver report, for reception statistics from participants
             that are not active senders, and for supplementary
             reception statistics from any participant

--
Timur Friedman  http://www.cs.umass.edu/~friedman











From rem-conf Thu Jul 22 06:53:31 1999 
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Timur Friedman wrote:
> 
> Robert Wood wrote:
> > 1.) Can a compund RTCP packet contain multiple SRs. Only
> >     multiple RRs are discussed in the spec.
> 
> As others have pointed out, a sender only reports on itself, qua sender,
> once.  Nonetheless, the short definitions of SR and RR (6.1) do make it
> reasonable to wonder what a sender should do in order to send more than 31
> reception report blocks:
> 
>         SR: Sender report, for transmission and reception statistics
>              from participants that are active senders
> 
>         RR: Receiver report, for reception statistics from participants
>              that are not active senders
> 
> On its face this means that RRs should not be sent by active senders.  This
> appears to contradict the later statements (6.1, 6.4) that RRs may be
> stacked after an initial SR.  A close textual reading is necessary to
> conclude without contradiction that even though RRs are "for" reception
> statistics from participants that are not active servers, they can be "for"
> other things as well.
> 
> These SR and RR definitions are likely the place most people look to remind
> themselves of the purpose of these packets, as they are the first time SRs
> and RRs are mentioned in the spec.  It would be good if they were not
> misleading in this way.  I suggest the following wording:
> 
>         RR: Receiver report, for reception statistics from participants
>              that are not active senders, and for supplementary
>              reception statistics from any participant

	So, in the case I'm considering, multiple phone calls, I will
	only generate SRs and never an RR ?

[\] Robert Wood   

The St. Lawrence river - fresh, warm, visible diving.

mailto:robert_wood@mitel.com http://www.magma.ca/~rgwood/



From rem-conf Thu Jul 22 08:34:45 1999 
From rem-conf-request@es.net Thu Jul 22 08:34:45 1999
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From: "Timur Friedman" <timur@research.att.com>
To: "Robert Wood" <robert_wood@Mitel.COM>
Cc: <rem-conf@es.net>
References: <3795EAE6.1505@mitel.com> <061c01bed446$6fe3ec80$5f1bcf87@research.att.com> <37972115.42B1@mitel.com>
Subject: Re: RTCP Sender reports
Date: Thu, 22 Jul 1999 11:31:02 -0400
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Robert Wood wrote:
> So, in the case I'm considering, multiple phone calls, I will
> only generate SRs and never an RR ?

If each participant is a data sender then each will generate an SR.  A
participant would only generate an RR in the case where it had to report on
more than 31 other senders.  Under those circumstances (not, I imagine, what
you are envisioning with respect to telephone calls), it would generate an
SR followed by one or more RRs to handle the additional reports.

--
Timur Friedman  http://www.cs.umass.edu/~friedman





From rem-conf Thu Jul 22 09:07:33 1999 
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References: <3795EAE6.1505@mitel.com> <061c01bed446$6fe3ec80$5f1bcf87@research.att.com> <37972115.42B1@mitel.com> <065401bed457$37b318c0$5f1bcf87@research.att.com>
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So, having looked at this issue, in my application,
	it looks like I only have to bother with SR packets.

	I wonder if anyone is actually using RTCP. I know 
	Microsoft's netwmeeting product generates them, but 
	does even it use them? 

	I can imagine a lot of products coming forth which
	will generate RTCP but not actually do anything with 
	received RTCP packets. 	

[\] Robert Wood   

The St. Lawrence river - fresh, warm, visible diving.

mailto:robert_wood@mitel.com http://www.magma.ca/~rgwood/



From rem-conf Thu Jul 22 09:41:04 1999 
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RTP monitors like UCB's "rtpmon" (ftp://mm-ftp.cs.berkeley.edu/pub/rtpmon/)
use RTCP.

  Bill



From rem-conf Thu Jul 22 15:12:25 1999 
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Date: Thu, 22 Jul 1999 18:09:09 -0400 (EDT)
From: Biswaroop Mukherjee 8-| <bmukherj@hopper.math.uwaterloo.ca>
To: rem-conf@es.net
Subject: RTP Algorithms
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As far as I could tell the RTP and RTCP specs do not specify control
mechanisms or algorithms.

Does someone know of any paper which describes the specific algorithms 
used in an implementation for things like, 
when to send in the next packet of data or what to do when a certain type
of sender report arrives.

--Biswaroop Mukherjee
http://www.cs.uwaterloo.ca/~bmukherj





From rem-conf Thu Jul 22 16:03:47 1999 
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Subject: MGCP source code?
To: rem-conf@es.net
Date: Thu, 22 Jul 1999 19:00:15 -0400 (EDT)
Cc: owner-confctrl@ISI.EDU
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Hi

Is there any MGCP source code available for Linux?

Thanks

Hassan Naser
Dept. of Electrical & Computer Eng.
University of Toronto, Toronto,
Ontario, M5S-3G4 Canada
Work: (416) 978 4829    Fax: (416) 978 4425
Email: hnaser@nal.utoronto.ca



From rem-conf Thu Jul 22 16:51:50 1999 
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Date: Thu, 22 Jul 1999 16:49:47 -0700
From: "Lawrence A. Rowe" <Rowe@bmrc.berkeley.edu>
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Organization: U.C. Berkeley
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Hi -

We have experimented a bit with this technology.  Here are some
interesting facts and/or features.

1. Yes, you can play an Mbone stream (h.261/pcm) using the player.  But
it can be only one video source.
2. You can play two concurrent video streams using the G2/SMIL player.
The two streams can be regular G2 streams or one or both can be Mbone
streams.  I've attached example Smil files below.  I also included the
modified sdp file needed to play the Mbone stream.  Audio works
correctly in the dual streams in that the audio tracks from the two
streams are mixed before being played.
3. A company in Mass, named Bitcasting, has implemented a G2 server side
plug-in that allows live and on-demand broadcasting of MPEG streams.
Specifically, you can play stored MPEG1 streams using the standard G2
player if you have purchased their server plug-in.  To see an amazing
demo, check out www.bitcasting.com.  Bitcasting also sells a live
encoder option but you have to use a particular board that is relatively
expensive.  They will also stream stored MPEG2 streams, but you have to
use a hardware decoder.

We are experimenting with various combinations of Mbone, G2, and G2/SMIL
sessions for the Berkeley MIG Seminar that will begin again on Wed Aug
25, 1999.  Watch this list for further announcements.  My hope is that
later in the fall we will offer MPEG1 on-demand replay of selected talks
along with synchronized slides using either SMIL or possibly a
Javascript based replay client.
	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

--- Dual G2 Streams in SMIL
<smil>
  <head>
    <meta name="title" content="Dual Video Window Demo"/>
    <meta name="author" content="BMRC"/>
    <meta name="copyright" content="(c) 1999"/>
    <layout>
      <root-layout width="676" height="304" background-color="gray" />
      <region id="image_area_1" width="320" height="240" left="2"
top="17" />
      <region id="image_area_2" width="320" height="240" left="340"
top="17" />
      <region id="caption_1" width="320" height="15" left="2" top="274"
/>
      <region id="caption_2" width="320" height="15" left="340"
top="274" />
    </layout>
  </head>

  <body>
    <par>
      <video
src="rtsp://media2.bmrc.berkeley.edu/classes/s1999/ns10/19990119.rm"
region="image_area_1" fill="freeze" />
      <video
src="rtsp://media2.bmrc.berkeley.edu/classes/s1999/ns10/19990121.rm"
region="image_area_2" fill="freeze" />
      <textstream src="test1.rt" region="caption_1" fill="freeze" />
      <textstream src="test2.rt" region="caption_2" fill="freeze" />
    </par>
  </body>
</smil>
----- G2 and Mbone SMIL
<smil>
  <head>
    <meta name="title" content="Dual Video Window Demo"/>
    <meta name="author" content="BMRC"/>
    <meta name="copyright" content="(c) 1999"/>
    <layout>
      <root-layout width="676" height="304" background-color="gray" />
      <region id="image_area_1" width="320" height="240" left="2"
top="17" />
      <region id="image_area_2" width="350" height="270" left="324"
top="2" />
      <region id="caption_1" width="320" height="15" left="2" top="274"
/>
      <region id="caption_2" width="320" height="15" left="340"
top="274" />
    </layout>
  </head>

  <body>
    <par>
      <video
src="rtsp://media2.bmrc.berkeley.edu/classes/s1999/ns10/19990119.rm"
region="image_area_1" fill="freeze" />
      <video
src="http://media2.bmrc.berkeley.edu/people/peterp/dual_smil/G2_and_mbone/dbs.sdp"
region="image_area_2" fill="freeze" />
      <textstream src="test1.rt" region="caption_1" fill="freeze" />
      <textstream src="test2.rt" region="caption_2" fill="freeze" />
    </par>
  </body>
</smil>
---- dbs.sdp file mentioned in G2/Mbone example
v=0
o=pbhuang 1 1 IN IP4 169.229.12.73
s=Satellite
i=The satellite link in Soda Hall, University of California, Berkeley
e=Paul Huang <pbhuang@bmrc.berkeley.edu>
c=IN IP4 239.255.124.168/1
t=0 0
a=type:broadcast
m=video 49256 RTP/AVP 31
c=IN IP4 239.255.124.168/1
m=audio 28366 RTP/AVP 00
c=IN IP4 239.255.53.104/1
---



From rem-conf Fri Jul 23 04:12:47 1999 
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From: "Hung Bui" <hvb@salix.com>
To: "'Hassan Naser'" <hnaser@comm.toronto.edu>, <rem-conf@es.net>
Subject: RE: MGCP source code?
Date: Fri, 23 Jul 1999 07:03:12 -0400
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Please check www.vovida.com.


-----Original Message-----
From:	Hassan Naser [mailto:hnaser@comm.toronto.edu]
Sent:	Thursday, July 22, 1999 7:00 PM

Hi

Is there any MGCP source code available for Linux?

Hassan Naser
Dept. of Electrical & Computer Eng.
University of Toronto, Toronto,
Ontario, M5S-3G4 Canada
Work: (416) 978 4829    Fax: (416) 978 4425
Email: hnaser@nal.utoronto.ca



   =========
  ===========   Hung Bui                   mailto:hvb@salix.com
 ============   Consultant                 http://www.salix.com
 ============   SALIX Technologies, Inc.
 ==  ==== ==    904 Wind River Lane        TEL:  (301) 417-0017
      ==        Suite 101                  FAX:  (301) 990-6400
      ===       Gaithersburg, MD  20878

   The statements contained herein are strictly those of
   their author, and unless otherwise specifically indicated
   should not be attributed to SALIX(r) or its management.







From rem-conf Fri Jul 23 04:25:56 1999 
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	env-from (wesner@rus.uni-stuttgart.de)
From: "Stefan Wesner" <wesner@rus.uni-stuttgart.de>
To: "Jonathan Rosenberg" <jdrosen@dnrc.bell-labs.com>
Cc: "Rem-Conf" <rem-conf@es.net>,
        "Paul Christ" <Paul.Christ@rus.uni-stuttgart.de>,
        "Christine Guillemot" <Christine.Guillemot@irisa.fr>,
        "Jerome. Vieron" <Jerome.Vieron@irisa.fr>
Subject: Problems with Generic FEC and Redundant Audio Encoding ?
Date: Fri, 23 Jul 1999 13:21:46 +0200
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Hi,

I have question concerning the usage of the GenericFEC in conjunction wit=
h
the redundant audio format (section 11 in the draft).

In section 8 it is described that the FEC operation is applied to (beside
others) to the X bit and the CC bits. And that the Extension and CSRC Lis=
t
is supposed to be part of the payload and is therefore protected together
with the payload. So the FEC payload will potentially consists of Xtensio=
n
data, CSRC List and media payload header and data.

In section 11 it is nothing said where to place the X bit and the CC bits
for the case of usage of FEC with redundant audio.

The reconstruction process says that for the reconstructing the FEC packe=
t
the X bit and the CC bit should be copied from the RTP packet that contai=
ned
the primary encoding and the fec as secondary data.

I think it is not appropriate to assume that the CC bits and the X bit ar=
e
the same for this FEC RTP packet and for the RTP packet that contains the
media and the FEC.

Possibly "wrong" CC bits and X bit will lead to wrong results.

I try to make an example:
Assuming the extension is switched "on" for the current RTP packet (with =
the
data+fec) the X bit will be set to 1. The previous RTP packets had the X =
bit
set to 0. So the result of a XOR operation would have been also 0. Doing =
the
proposed procedure will result in a FEC packet with X bit set to 1. In th=
e
recovery array the X bit will be also "1". The interpretation of the byte
array will assume a header extension and will therefore not be able to
interpret the recovery array correctly.

So the mechanism for redundant audio works only for the case CC bits are
"0000" and X bit =3D "0".

Do I miss something ?

Kind regards,

Stefan Wesner.
--
                            University Stuttgart computer centre
Stefan  Wesner 	     Communication Systems & BelW=FC development
Allmandring 3a                mailto:wesner@rus.uni-stuttgart.de
70550 Stuttgart  Tel.: +49 711 685 4275   Fax.: +49 711 678 8363




From rem-conf Fri Jul 23 20:43:43 1999 
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Message-ID: <3798F63D.B470827E@dnrc.bell-labs.com>
Date: Fri, 23 Jul 1999 19:09:49 -0400
From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
Organization: Bell Laboratories
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To: Stefan Wesner <wesner@rus.uni-stuttgart.de>
CC: Rem-Conf <rem-conf@es.net>, Paul Christ <Paul.Christ@rus.uni-stuttgart.de>,
        Christine Guillemot <Christine.Guillemot@irisa.fr>,
        "Jerome. Vieron" <Jerome.Vieron@irisa.fr>
Subject: Re: Problems with Generic FEC and Redundant Audio Encoding ?
References: <000e01bed4fd$8ca60ee0$1a0d4581@ksat22.rus.uni-stuttgart.de>
Content-Type: text/plain; charset=iso-8859-1
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You are correct. As defined, the mechanism for encapsulating the FEC in
the redundant encodings payload could cause the CC and X bits to be
mis-constructed if they are not all zeroes. 

I see two ways to fix this:

1. Specify that when used with redundant encodings, the FEC must be used
across packets without extension or CC bits set.

2. Specify that when used with redundant encodings, packets with
extension and CC bits set must have the extension and CSRC list removed,
and the bits set to zero, before computing the FEC. However, the packet
itself can be sent with the extension and CSRC list included. This
basically means that if recovered, it will be recovered without the
extension and CSRC list.

I tend to prefer (2). Whatever the change we decide, I will fold it in
as part of the overall revision to reflect comment received during IESG
last call, which this draft is now in.

Thanks,
Jonathan R.

Stefan Wesner wrote:
> 
> Hi,
> 
> I have question concerning the usage of the GenericFEC in conjunction with
> the redundant audio format (section 11 in the draft).
> 
> In section 8 it is described that the FEC operation is applied to (beside
> others) to the X bit and the CC bits. And that the Extension and CSRC List
> is supposed to be part of the payload and is therefore protected together
> with the payload. So the FEC payload will potentially consists of Xtension
> data, CSRC List and media payload header and data.
> 
> In section 11 it is nothing said where to place the X bit and the CC bits
> for the case of usage of FEC with redundant audio.
> 
> The reconstruction process says that for the reconstructing the FEC packet
> the X bit and the CC bit should be copied from the RTP packet that contained
> the primary encoding and the fec as secondary data.
> 
> I think it is not appropriate to assume that the CC bits and the X bit are
> the same for this FEC RTP packet and for the RTP packet that contains the
> media and the FEC.
> 
> Possibly "wrong" CC bits and X bit will lead to wrong results.
> 
> I try to make an example:
> Assuming the extension is switched "on" for the current RTP packet (with the
> data+fec) the X bit will be set to 1. The previous RTP packets had the X bit
> set to 0. So the result of a XOR operation would have been also 0. Doing the
> proposed procedure will result in a FEC packet with X bit set to 1. In the
> recovery array the X bit will be also "1". The interpretation of the byte
> array will assume a header extension and will therefore not be able to
> interpret the recovery array correctly.
> 
> So the mechanism for redundant audio works only for the case CC bits are
> "0000" and X bit = "0".
> 
> Do I miss something ?
> 
> Kind regards,
> 
> Stefan Wesner.
> --
>                             University Stuttgart computer centre
> Stefan  Wesner       Communication Systems & BelWü development
> Allmandring 3a                mailto:wesner@rus.uni-stuttgart.de
> 70550 Stuttgart  Tel.: +49 711 685 4275   Fax.: +49 711 678 8363

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



From rem-conf Sat Jul 24 23:32:35 1999 
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Date: 7/21/99 8:22:32 PM Pacific Daylight Time
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Subject: >>> Warning - Don't Get Ripped Off !!! <<<
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From rem-conf Sun Jul 25 04:32:40 1999 
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From rem-conf Mon Jul 26 08:21:33 1999 
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From rem-conf Mon Jul 26 14:03:48 1999 
From rem-conf-request@es.net Mon Jul 26 14:03:47 1999
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Date: Mon, 26 Jul 1999 22:56:00 +0200
To: rem-conf@es.net
From: Gunnar Hellstrom <gunnar.hellstrom@omnitor.se>
Subject: draft-hellstrom-avt-rtp-text-01.txt Result of Oslo discussion.
Cc: textphone@lsv.pi.se (Text Telephone Forum), mickey.nasiri@etx.ericsson.se,
        itu-sg16@mailbag.cps.intel.com
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The draft rtp specification for text conversation that is discussed in the
IETF AVT group is now available in the following URL.=20

www.omnitor.se/english/standards/rtp/draft-hellstrom-avt-rtp-text-01.txt

It is referenced in the H.323 Annex G Text Conversation draft that is
entered as a contribution to the ITU-T Q13/16 meeting next week as document
APC1561.
I would like to receive comments on this draft.
I am aware of the following points that might need discussion.

1. Timestamp clock frequency. I had put a low figure 10 Hz in the
specification. It was questioned and proposed that a figure of some
thousands should be selected.=20
I am willing to accept that, and propose that 1000 Hz is used. A
millisecond per tick is a handy figure.=20
This will however lead to the timestamp offset in the redundancy data
headers that use only 14 bits, flip around in only 16 seconds. If correct
timing information is wanted, the data flush procedure described in next
section should be used.

2. Retransmission of text before a pause.
Text does not flow continously, but only when a user enter text. Therefore,
a recommended procedure could specify that for the redundancy scheme, it is
advisable to send the ISO 10646 synch character FEFF after some inactivity
timer to flush out redundant data - see section 3.4.=20

3. Recommended procedures section.
In the draft that accompanies this mail, section 3 documents the
recommended procedures for using the payload format. Does the group want it
to be kept there or shall it be moved to the specification specifying use
of this RTP Payload? ( The first user being H.323 Annex G)=20

Any other comments are welcome.


Gunnar Hellstr=F6m

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

E-mail gunnar.hellstrom@omnitor.se
Tel +46 751 100 501
fax +46 8 556 002 06




From rem-conf Mon Jul 26 14:22:35 1999 
From rem-conf-request@es.net Mon Jul 26 14:22:34 1999
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	id 118sAo-0007SF-00; Mon, 26 Jul 1999 14:20:38 -0700
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	id 118sAA-0007Jt-00; Mon, 26 Jul 1999 14:20:07 -0700
From: <greenmail@hispeednet.com>
To: rem-conf@es.net
Subject: advertisment  *FREE LEATHERMAN TOOL
Date: Mon, 26 Jul 1999 17:20:08
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From rem-conf Mon Jul 26 16:23:11 1999 
From rem-conf-request@es.net Mon Jul 26 16:23:11 1999
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DATE: Mon, 26 Jul 1999 19:23:47 -0500
Message-ID: <sjfoseoq>
Subject: HOT STOCK PICK                                        (adv.)
From:     ENCR@fridaysStockPicks.com
To:     $user@es.net
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Encounter.com, Inc.  (OTC BB:  ENCR)http://fridaysstockpicks.com/
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___________________________________________________________
transmission error - please ignore





From rem-conf Wed Jul 28 12:14:01 1999 
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From: "Cain, Michael E." <mcain@mediaone.com>
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Subject: Load generator for RTP/RTCP/RTSP-based server
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This may not be the appropriate forum to ask this question -- if so, I
apologize in advance.

I'm going to be testing an Apple streaming QuickTime server and want to be
able to generate some amount of load on the box.  Rather than trying to get
enough PCs and/or Macs together, each running a QuickTime player, and
attempting to script those players in some fashion, I would like to write a
pseudo-client that asks for the stream, receives and simply discards the RTP
packets, generates appropriate RTCP responses, etc.  Since no actual
decoding or display of the media would be done, even a low-end machine
should be able to absorb several streams' worth of data.

Because Apple has based their server on open RTP/RTCP/RTSP standards,
writing such a pseudo-client should be straightforward.  Before I do it all
>from scratch, though, it seemed appropriate to ask if anyone has already
done it or if some pieces at least are available (perhaps Columbia's RTP
toolkit would be helpful?).

Thanks in advance,
Michael Cain
MediaOne Labs
mcain@mediaone.com




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you said
"This may not be the appropriate forum to ask this question -- if so, I
apologize in advance.

I'm going to be testing an Apple streaming QuickTime server and want to be
able to generate some amount of load on the box.  Rather than trying to get
enough PCs and/or Macs together, each running a QuickTime player, and
attempting to script those players in some fashion, I would like to write a
pseudo-client that asks for the stream, receives and simply discards the
RTP
packets, generates appropriate RTCP responses, etc.  Since no actual
decoding or display of the media would be done, even a low-end machine
should be able to absorb several streams' worth of data.

Because Apple has based their server on open RTP/RTCP/RTSP standards,
writing such a pseudo-client should be straightforward.  Before I do it all
>from scratch, though, it seemed appropriate to ask if anyone has already
done it or if some pieces at least are available (perhaps Columbia's RTP
toolkit would be helpful?).

Thanks in advance,
Michael Cain
MediaOne Labs
mcain@mediaone.com

"

My attempt at an answer:
 I believe that the RTP tool kit might indeed be helpful.
Our experiments indicate that:
The rtptools on H.Schulzrinne's web site can run multiple copies on one pc.
So, if you configure your server to pump rtp out to a set of port numbers,
and configure the test pc to run rtpdump in a separate window for each of
those port numbers, you will have a test.  rtp does not directly request
service, that is handled by other protocols so you must configure your
server to spew out data on its own for this test.
Hope this helps.  Let us know.





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From: Rebecca Morales <Rebecca.Morales@prism.uvsq.fr>
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Subject: NETWORKING 2000
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NETWORKING 2000

IFIP - TC6/ European Union

Broadband Communications (BC),
High Performance Networking (HPN),
Performance of Communication Networks (PCN)


Paris, France
Cit=E9 des Sciences, La Villette
May 14 =96 19, 2000
Main Sponsors
IFIP TC6, European Union

National Main Sponsor
RNRT and SEE

Supporters (pending)
Alcatel, CISCO, Thomson-CSF Detexis, EDF, France Telecom

Organizer
University of Paris VI , University of Versailles, ENST

General Chair
Guy Pujolle - France

Organization Committee
Serge Fdida - France
Jean-Alain Hernandez - France
Eric Horlait - France
Ren=E9 Joly - France
Guy Pujolle - France

Steering Committee
 Augusto Casaca - Portugal
Andr=E9 Danthine - Belgium
Olli Martikainen - Finland
Harry Perros - USA
Jan Slavik - Czech Republic
Otto Spaniol - Germany
Y. Takahashi - Japan
S. Tohme - France

General Program committee chair
Harry Perros - USA

Program committee chair for BB track
Ulf Korner - Sweden

Program committee chair for HPN track
Serge Fdida - France

Program committee chair for PCN track
Ioannis Stavrakakis - USA


 Program committee members
 Augusto Albuquerque - EC
Harmen van As - Austria
Augusto Casaca - Portugal
Paolo Castelli - Italy
Imrich Chlamtac -USA
Jean-Pierre Coudreuse - France
Nelson L. S. Fonseca - Brazil
Luigi Fratta - Italy
Giorgio Gallassi - Italy
Andre Girard - Canada
Villy B. Iversen - Denmark
Bijan Jabbari -USA
Konosuke Kawashima - Japan
Peter Key - USA
Daniel Kofman - France
Paul Kuehn - Germany
Helmut Leopold - Austria
John Luetchford - Canada
Lorne Mason - Canada
Serafim Nunes - Portugal
Guido Petit - Belgium
Sathya Rao - Switzerland
Joao Rodrigues - Portugal
Catherine Rosenberg - UK
Tadao Saito - Japan
Amardeo Sarnea - Germany
Marion Schreinemachers - Netherl.
Jan Slavik - Czech Republic
Samir Tohme - France
Danny Tsang - Hong Kong

Finn Arve Aagesen - Norway
Andres Albanese - USA
Chase Bailey - USA
Ermanno Berruto - Italy
Andrew Campbell - USA
Jon Crowcroft - UK
Andre Danthine - Belgium
Walid Dabbous - France
Michel Diaz - France
Sarolta Dibuz - Hungary
Christophe Diot - USA
Otto Duarte - Brazil
Wolfgang Effelsberg - Germany
Nicolas D. Georganas - Canada
Enrico Gregori - Italy
Roch Guerin - USA
Christian Huitema =96 USA
David Hutchinson -UK
Marjory Johnson - USA
Farouk Kamoun - Tunisia
Koos Koen - RSA
Jacques Labetoulle - France
Jean-Yves Le Boudec - Switzerl.
Guy Leduc - Belgium
Olli Martikainen - Finland
Steve Pink - Sweden
Nina Taft Plotkin - USA
Radu Popescu-Zeletin - Germany
Luigi Rizzo - Italy
Aruna Seneviratne -Australia
Otto Spaniol - Germany
Ralf Steinmetz - Germany
Chris Blondia - Belgium
Miklos Boda - Hungary
Herwig Bruneel - Belgium
Olga Casals - Spain
Prosper Chemouil - France
Giovanni Colombo - Italy
Tony Ephremides - USA
Andras Farago - USA
Erol Gelenbe - USA
Mario Gerla - USA
Fabrice Guillemin - France
Gerard Hebuterne - France
Demetres Kouvatsos - UK
Jim Kurose - USA
Karl Lindberger -Sweden
Jon Mark - Canada
Marco Marsan - Italy
Lazaros Merakos - Greece
Jouni Mikkonen - Finland
Debasis Mitra - USA
Naotama Morita - Japan
Arne Nilsson - USA
Raif Onvural - USA
Ramon Puigjaner - Spain
James Roberts - France
Yutaka Takahashi - Japan
Ahmed Tantawy - USA
Phouk Tran-Gia - Germany
Jorma Virtamo - Finland
Adam Wolisz - Germany


Tutorials Committee Chair
Eric Horlait -France

Publicity Committee Chair
Raouf Boutaba - Canada

Networking 2000 Conference
Networking 2000 Conference will provide an international technical forum
for experts from industry and academia to exchange ideas and present
results of ongoing research in networking. It is a joint conference of
the following three series of conferences :
Broadband Communications (BB)
High Performance Networking (HPN)
Performance of Communication Networks (PCN).




Topics
 Switching Design
Routing Design
Switching and Routing
Internet/Intranet
LAN, WAN Global Network Interconnection
High Performance Networking and Protocols
Service Provider Interworking
Control and Optimization of Communication Systems
Quality of Service
Multicast
Active Networks
Terrestrial Radio Systems
Wireless and Mobile Communications
Satellite and Space Communications
Low Earth Orbit Satellite Communication Systems
Optical Communications
Access Networks
Video Coding and Distribution
Object Orientation
Network Operations and Management
Signalling
Tariffing
Network Reliability, Availability and Survivability
Internet Services and Applications
Network Design Problems in Gigabit
Terabit Networks
Performance Evaluation of Telecommunication Systems
Traffic Models and Measurements
Traffic and Congestion Control

KEY DATES
* September 1st,, 1999 Deadline for submitting Short Course Proposals
* September 15, 1999  Deadline for submitting papers
* January 15, 2000  Notification of acceptance of papers
* February 15, 2000  Papers received in camera ready form
* May 14-15, 2000  Tutorials
* May 16-19, 2000  Networking 2000 Conference

SUBMISSION OF PAPERS
Paper can be submitted by mail or email. Monitor the web site for
details: www.noc.uoa.gr/net2000


Papers are invited on the conference theme and related topics. Authors
must state that their paper have neither been published before nor
currently being submitted elsewhere.
Full papers should be no longer than 15 pages, including tables,
diagrams and pictures. The font size must be 12 points or larger. The
cover page must contain an abstract of about 150 words, name and
affiliation of author(s) as well as the lead author's postal address,
telephone number, fax number and e-mail.
Accepted papers will be included in the conference proceedings and will
be distributed to the attendees at the conference.
The language of the conference is English and papers must be in this
language.

SHORT COURSES
We solicit proposals for half/full day short courses outlining the
subject area and a short biography of the organizer(s).

Short courses can be submitted by mail or email to
Prof. E. Horlait
LIP6,
4 place Jussieu
75252 Paris Cedex 05

* by email to Eric.Horlait@lip6.fr

INFORMATION
www.noc.uoa.gr/net2000
www.prism.uvsq.fr/~net2000



Mini conferences and workshops will take place during the conference.
The following proposals are being considered: (specific call for papers
will be provided)

MWCN - Mobile and Wireless Communication Networks (Chairman Guy Omidyar
=96 USA) gomidyar@csc.com

IATE - Intelligent agents for telecommunication environments
(Chairpersons Dominique Gaiti - France and Olli Martikainen - Finland)
Dominique.Ga=EFti@lip6.fr, Olli.Martikainen@hut.fi

Quality of Service and Multimedia Applications (Chairman Eric Horlait -
France)
Eric.Horlait@lip6.fr

Constellation of satellites (Chairwomen C. Rosenberg - Canada)
C.Rosenberg@nortel.co.uk

Resource allocation for mobile networks (Chairman Sami Tabbane -
Tunisia)
sami.tabbane@supcom.rnu.tn

Internet and the Web (Chairman Keith Ross- France)
ross@eurecom.fr

Performance of Adaptive Intelligent Networks (Chairman E. Gelenbe - USA
erol@cs.ucf.edu

Programmable Networks and Active Networks (Chairman R. Boutaba =96 Canada=
,
Andrew Campbell - USA, Rolf Stadler =96 USA and Ian Wakeman - USA)
rboutaba@jupiter.nal.utoronto.ca   campbell@comet.columbia.edu

Multimedia Management (Chairman Jose Neuman de Souza =96 Brazil)
neuman@ufc.br






From rem-conf Thu Jul 29 09:34:06 1999 
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To: IETF-Announce: ;
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From: The IESG <iesg-secretary@ietf.org>
SUBJECT: Last Call: An RTP Payload Format for Generic Forward Error
	 Correction to Proposed Standard
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The IESG has received a request from the Audio/Video Transport Working
Group to consider An RTP Payload Format for Generic Forward Error
Correction <draft-ietf-avt-fec-07.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 August 12, 1999.

Files can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-avt-fec-07.txt



From rem-conf Thu Jul 29 10:02:45 1999 
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From: The IESG <iesg-secretary@ietf.org>
SUBJECT: Last Call: Guidelines for Writers of RTP Payload Format
	 Specifications to BCP
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The IESG has received a request from the Audio/Video Transport Working
Group to consider Guidelines for Writers of RTP Payload Format
Specifications <draft-ietf-avt-rtp-format-guidelines-03.txt> as a BCP.

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 August 12, 1999.

Files can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-format-guidelines-03.txt




From rem-conf Thu Jul 29 22:33:23 1999 
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From: <CybrMrkting99@job4u.com>
Subject: I thought you might be interested
Reply-To: CybrMrkting99@job4u.com
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Comments: Authenticated Sender is <CybrMrkting99@job4u.com>
Message-Id: <13888471_91174256>
content-length: 11370
Date: Fri, 30 Jul 1999 01:11:46 -0400
X-Mailing-List: <rem-conf@es.net> 
X-Loop: rem-conf@es.net
Precedence: list


Hi,
Would you like to earn an extra $700 a week... $2,800 a month just by
mailing our business circulars from your home? You can make this kind
of money without even giving up your present job. We have created the
most risk-free way to do this, and all you have to do is mail out our
business circulars and get paid for your work. This exciting new home
employment opportunity is so effective - yet quick and easy that your
success is absolutely GUARANTEED!

We Publish, sell and distribute information booklets, guides, reports,
manuals and computer software all across Canada and the United States.
Since we do the majority of our business by mail, we in turn send out
thousands of our sales circulars each week.Our company circulars are
the sales letters/product offers that are sent out in response to
customer inquiries. When you mail our circulars, you'll be greatly
helping us by getting our offers out to more customers.

You'll be taking part in the most remarkable opportunity available.
Our system of mailing circulars is very easy to operate. All you will
be doing is taking copies of the standard, letter-sized (8 1/2"x11")
circulars that we provide you with and fold them to fit into the
envelopes you will receive. After you have folded and inserted the
circulars into the envelopes, you just have to seal the envelopes and
deposit them in your mail. You will not have to spend any time
addressing envelopes or pay any postage costs to mail out our
circulars, because all envelopes will arrive pre-addressed and with
postage already in place. It should only take you a few hours a week.
It's as simple as that! Circular mailing is easy and pleasant work
that is very profitable!

We have developed a legitimate and realistic money-making business
system that is practical and uncomplicated. Simple enough that
anyone can take part in it regardless of education, age, physical
ability or disability. Our program is easy to understand, with
step-by-step instructions that are sure to get you started quickly
and with confidence.

This is a highly fool-proof, tried, tested and proven method that
you can run from the comfort and privacy of your own home - without
any personal contact with anyone at any time. You can take part in
our program any hour of the day... any day of the week. We do not
require that you have any related experience to take part in our
program. All that we want are serious minded people who can read and
write simple English, and who are able to put in a few extra hours
each week towards earning a great income. Although you don't need 
any experiance, it is important that you be ambitious and motivated
because you will be working on your own - without supervision.
It will be your responsibility to get your work done.

You do not have to meet a certain quota each week, and we do not 
impose any restrictions on the amount of work that you choose to do.
Our Circular Mailing Program allows you the complete flexibility to
organize and choose your own work load and work schedule. You can
work part-time or full-time, and you are always free to take a break
>from your work - planning your own time off. Furthermore, you may
quit the program at any time, since you will be an independent mailer
and have no obligation to our company what-so-ever.

Home employment is wonderful. It can provide you with a great sense
of accomplishment, pride and freedom - but you must remember to
treat your work seriously and with respect. What you get out of our
program is exactly what you put into it. Earn as much or as little
as you like - it's all up to you. You can start the same day you 
receive our supplies and information package, and begin receiving
money within 2 weeks time, and every week from then on for as long
as you desire to participate in our program.

Thousands of people all over Canada and the United States are making
excellent money mailing circulars from their homes. You can join our
successful network, of circular mailers and get your share of this
money too! It makes no difference if you live in a small town or a
large city. As long as you can get to a mailbox to mail the circulars,
you can participate in this great home income opportunity.

Anyone with a little common sense and a desire to succeed can take
part in our program and earn excellent income for themselves in a
very short period of time. One of the nicest things about our circular
mailing program is how quickly it works. You can start the same day
you receive our supplies and information package, and begin receiving
money within 2 weeks time, and then for as long as you decide to
participate in our program. Imagine never having to leave your home
while making more money in a few hours than a lot of people earn
after a full week's work! In fact, you can be making great money in
as little as 10 hours a week!

And remember, you don't need any special education of experience.
This program can work for anyone - regardless of background, age or
location. Just mail the circulars and spend the rest of the day
enjoying yourself. Imagine being able to work in the comfort of
your own home, at your own pace and in your leisure with a simple
system to earn $700.00 a week working only a few hours a week... it's
a great place to start.

You really don't have to work hard to get ahead in life - you just
have to work SMART. By following our easy steps, you will connect
with making $700 a week - every week. We'll even show you how you
can increase an income of $700.00 a week into as much as $1,000.00
a week or more with ease. It's very simple and it's very realistic.
With the basic details we've outlined, you'll easily see the
incredible income potential right away. This program is so well
thought out and developed that YOU CANNOT FAIL to make great money!

You'll be able to set up your operation so that you can have all the
free time in the world to do with as you please, without anyone
looking over your shoulder. If you choose , you'll only have to work
at your affairs for about an hour or two a day, and these will turn
out to be extremely easy, fun hours, with no boss snooping around and
no one to answer to. Believe this, after you spend the small amount
of up-front time getting organized, setting up your own system,
you'll soon realize that nothing could be easier, or offer you more
privacy and personal freedom!

Unlike others who you might see promoting the same old useless
stuff - year in and year out, we have developed a unique approach
which has never been released to the public by anyone else.
Cybermarketing is the only source for this valuable program. So
please don't confuse it with any of the other get-rich-quick schemes
or ads you might see. If you're searching for an honest to goodness,
legitimate, legal and spare time work at home opportunity, then your
search has finally ended. This is a 100% proven Money-Making
Program! PROVEN to make good money for everyone who uses it!

If you're like most folks, you're going to absolutely love our 
Circular Mailing Program because it's the most legitimate, 
"on-the-level", easy to start, profitable work-from-home opportunity
ever created! We say this honestly because IT REALLY WORKS! You 
won't find any gimmicks, surprises or silly schemes. Only valuable
information that you'll need so you can quickly learn exactly WHAT
TO DO and HOW TO DO IT; and our proven , professionally written
circulars.

Our Circular Mailing Program can bring you all the money you need.
When you receive our start up package in the mail, you will get all
the supplies you need to get started right away including your 
Personal Information Kit (your complete instructional handbook) and
our business circulars. You will receive everything as promised.
And don't forget you will not have to pay postage costs to mail our
circulars because envelopes will arrive completely addressed and
with postage already in place. Just mail out the circulars and
receive pay cheques that are yours to spend any way you wish! You can
take part in our program for as long as you want... earn $700 a week
for the rest of your life.

We can only accommodate a limited number of people in our unique
program. So if you are interested, please do not delay. Send in 
your acceptance form as soon as you can. You have our guarantee that
this program can change your life practically overnight. We know of
no other home employment opportunity with the potential to make the
great amounts of money that this program does.

This is a complete home-based opportunity that REALLY WORKS. Our
only requirement is a one time only, fully refundable payment of
only $27.00 . This payment covers the cost of your supplies and
the processing of your membership. This is a one-time payment, you
will not have to pay us any other costs to get additional materials.
And because we're so sure that we have the right home employment
opportunity for you, we are backing up our promises with our
exclusive guarantee...

   $33,600.00 Guarantee

You can easily earn $33,600.00 in the next year with our program.
In fact, we are so confident that you can make over $700 a week 
mailing out our circulars that we are going to offer you the most 
air-tight guarantee in existence. As soon as you receive our start
up package in the mail, send out our circulars right away. If you
don't start earning a minimum of $700 a week within 30 days, simply
return our materials for a COMPLETE REFUND. You either make $700 a
week or your money back!

Join our network of circular mailers today. We truly want to help
you get started as quickly and as easily as possible. This program
is designed for people who are serious about earning a substantial 
income. We are convinced that you will be absolutely thrilled when
you see just how much money you can make with our program. We've
got your start-up kit all packaged up and ready to go. Just give
us the word and we'll have it out the door and on its way to you.
If you follow our instructions, you will be well on your way to
earning $700 per week by mailing circulars from home. Just print 
and fill out the Exclusive Membership Form at the bottom of this 
page and mail it in with your remittance and your order will be 
rushed to you right away by first class mail.

We hope that you allow us the honor of being the ones who helped 
you to achieve long-term financial success and personal freedom.

 Most sincerely,
  The Staff at CyberMarketing
------------------------------------------------------------------
             HOME MAILERS PROGRAM ORDER-FORM
------------------------------------------------------------------
Please RUSH me my Package of the HOME MAILERS PROGRAM and the HOME
BUSINESS DIRECTORY right away!!! Send $27.00 in U.S Funds. 
(Includes Postage & Handling)
                                  
                         NAME:______________________________________
MAIL TO: CYBERMARKETING
P.O.Box #563             ADDRESS:___________________________________
Lindsay, Ontario,
Canada, K9V 4S5          CITY:_________________STATE/PROVINCE:______

	                 ZIP/POSTAL CODE:___________________________
                 
	                 E-MAIL ADDRESS:____________________________
	             
     (Orders payable by cheque please allow 4-6 weeks for delivery.)
07/30/99
--------------------------------------------------------------------



From rem-conf Fri Jul 30 05:41:12 1999 
From rem-conf-request@es.net Fri Jul 30 05:41:10 1999
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To: you@yourdomain.net
Bcc: renee.r@mailcity.com, renee.cohn@astd.noli.com, renee.cockrell@mailexcite.com, renee.chavis@snet.net, rened@evansville.net, renecuellar@mailcity.com, renecose@mail.imagine.com.ar, renebarba@eudoramail.com, reneb@n2mail.com, rene1@lucent.com, rene°@globalserve.net, rene@propane.ca, rene@ntcnet.com, rene@netnoir.net, rene@inforamp.net, rene@globalserve.net, rene@creativeprofile-netcards.co.nz, rene.hamberg@icu.nl, rene.gagnon@fmc.ulaval.ca, rene.brazeau@eudoramail.com, rene.beddok@hol.fr, rene.becker@eudoramail.com, rene.barthelemy@cern.ch, rendler@access.digex.net, rendlai@geocities.com, rendengail@eudoramail.com, rendam@telkom.co.za, renda@starting-point.com, renbau@televar.com, renatosenna@mailcity.com, renato@passosnet.com.br, renato@dgx.wimsey.com, renate.teuschler@ac.com, renate.sonnenberg@t-online.de, renata.bassi@mailexcite.com, renascaince@angelfire.com, renasat@mailcity.com, renapro@intex.net, rena-m@usa.net, renaissance-woman@mailcity.com, renaisncmn@mailcity.com, renae@ionet.net, rena@foxcomm.net, ren3gade@mailexcite.com, ren@thecia.net, ren@ntl.com, ren@england.com, ren@carlsbadnm.com, remy92@remycapital.com, remy@scinfo.u-nancy.fr, remus3@aol.com, removals@eudoramail.com, remote.control@mailexcite.com, remondo@geocities.com, remodel@qni.com, remo@sunworks.com, remo@mailexcite.com, remo@lunet.it, remnant005@mailexcite.com, remmaps@mailexcite.com, remman@thedoghousemail.com, remlover@geocities.com, remley@angelfire.com, remington_chief@hotmail.com, remington@chickmail.com, remind_me@mailexcite.com, remil@mailcity.com, remicote@odyssee.net, remi_gagnon@mailcity.com, remi@gis.net, remi@gazeta.pl, remi.berland@enst-bretagne.fr, remer@mailexcite.com, remek@mailcity.com, remdrec@mailexcite.com, remdrec@mailcity.com, rem-conf@es.net, remccoll@julian.uwo.ca, rembrandt.colgan@mailexcite.com, rembol@indigo.arch.pw.edu.pl, rembol@arch.pw.edu.pl, remaxkb@trib.com, remaxdaley@aol.com, remax@tiac.net, remarket@interpath.com, remark@unicomp.net, remarck@aol.com, remarc-able@mailexcite.com, remarc@mailcity.com, remapi@mbox.vol.it, reman@smart.net, remailer@replay.com, rem@btr.com, rem.desc@mailexcite.com, relyon@accnorwalk.com, relucu@mailexcite.com, reloader@mailcity.com, rellinger@mailexcite.com, --rellil--@mailcity.com, rellceh_j14@hotmail.com
From: <CybrMrkting99@job4u.com>
Subject: I thought you might be interested
Reply-To: CybrMrkting99@job4u.com
X-PMFLAGS: 20720340.50
X-UIDL: 20720340_201230.501
Comments: Authenticated Sender is <CybrMrkting99@job4u.com>
Message-Id: <87119827_44197276>
content-length: 11370
Date: Fri, 30 Jul 1999 08:36:57 -0400
X-Mailing-List: <rem-conf@es.net> 
X-Loop: rem-conf@es.net
Precedence: list


Hi,
Would you like to earn an extra $700 a week... $2,800 a month just by
mailing our business circulars from your home? You can make this kind
of money without even giving up your present job. We have created the
most risk-free way to do this, and all you have to do is mail out our
business circulars and get paid for your work. This exciting new home
employment opportunity is so effective - yet quick and easy that your
success is absolutely GUARANTEED!

We Publish, sell and distribute information booklets, guides, reports,
manuals and computer software all across Canada and the United States.
Since we do the majority of our business by mail, we in turn send out
thousands of our sales circulars each week.Our company circulars are
the sales letters/product offers that are sent out in response to
customer inquiries. When you mail our circulars, you'll be greatly
helping us by getting our offers out to more customers.

You'll be taking part in the most remarkable opportunity available.
Our system of mailing circulars is very easy to operate. All you will
be doing is taking copies of the standard, letter-sized (8 1/2"x11")
circulars that we provide you with and fold them to fit into the
envelopes you will receive. After you have folded and inserted the
circulars into the envelopes, you just have to seal the envelopes and
deposit them in your mail. You will not have to spend any time
addressing envelopes or pay any postage costs to mail out our
circulars, because all envelopes will arrive pre-addressed and with
postage already in place. It should only take you a few hours a week.
It's as simple as that! Circular mailing is easy and pleasant work
that is very profitable!

We have developed a legitimate and realistic money-making business
system that is practical and uncomplicated. Simple enough that
anyone can take part in it regardless of education, age, physical
ability or disability. Our program is easy to understand, with
step-by-step instructions that are sure to get you started quickly
and with confidence.

This is a highly fool-proof, tried, tested and proven method that
you can run from the comfort and privacy of your own home - without
any personal contact with anyone at any time. You can take part in
our program any hour of the day... any day of the week. We do not
require that you have any related experience to take part in our
program. All that we want are serious minded people who can read and
write simple English, and who are able to put in a few extra hours
each week towards earning a great income. Although you don't need 
any experiance, it is important that you be ambitious and motivated
because you will be working on your own - without supervision.
It will be your responsibility to get your work done.

You do not have to meet a certain quota each week, and we do not 
impose any restrictions on the amount of work that you choose to do.
Our Circular Mailing Program allows you the complete flexibility to
organize and choose your own work load and work schedule. You can
work part-time or full-time, and you are always free to take a break
>from your work - planning your own time off. Furthermore, you may
quit the program at any time, since you will be an independent mailer
and have no obligation to our company what-so-ever.

Home employment is wonderful. It can provide you with a great sense
of accomplishment, pride and freedom - but you must remember to
treat your work seriously and with respect. What you get out of our
program is exactly what you put into it. Earn as much or as little
as you like - it's all up to you. You can start the same day you 
receive our supplies and information package, and begin receiving
money within 2 weeks time, and every week from then on for as long
as you desire to participate in our program.

Thousands of people all over Canada and the United States are making
excellent money mailing circulars from their homes. You can join our
successful network, of circular mailers and get your share of this
money too! It makes no difference if you live in a small town or a
large city. As long as you can get to a mailbox to mail the circulars,
you can participate in this great home income opportunity.

Anyone with a little common sense and a desire to succeed can take
part in our program and earn excellent income for themselves in a
very short period of time. One of the nicest things about our circular
mailing program is how quickly it works. You can start the same day
you receive our supplies and information package, and begin receiving
money within 2 weeks time, and then for as long as you decide to
participate in our program. Imagine never having to leave your home
while making more money in a few hours than a lot of people earn
after a full week's work! In fact, you can be making great money in
as little as 10 hours a week!

And remember, you don't need any special education of experience.
This program can work for anyone - regardless of background, age or
location. Just mail the circulars and spend the rest of the day
enjoying yourself. Imagine being able to work in the comfort of
your own home, at your own pace and in your leisure with a simple
system to earn $700.00 a week working only a few hours a week... it's
a great place to start.

You really don't have to work hard to get ahead in life - you just
have to work SMART. By following our easy steps, you will connect
with making $700 a week - every week. We'll even show you how you
can increase an income of $700.00 a week into as much as $1,000.00
a week or more with ease. It's very simple and it's very realistic.
With the basic details we've outlined, you'll easily see the
incredible income potential right away. This program is so well
thought out and developed that YOU CANNOT FAIL to make great money!

You'll be able to set up your operation so that you can have all the
free time in the world to do with as you please, without anyone
looking over your shoulder. If you choose , you'll only have to work
at your affairs for about an hour or two a day, and these will turn
out to be extremely easy, fun hours, with no boss snooping around and
no one to answer to. Believe this, after you spend the small amount
of up-front time getting organized, setting up your own system,
you'll soon realize that nothing could be easier, or offer you more
privacy and personal freedom!

Unlike others who you might see promoting the same old useless
stuff - year in and year out, we have developed a unique approach
which has never been released to the public by anyone else.
Cybermarketing is the only source for this valuable program. So
please don't confuse it with any of the other get-rich-quick schemes
or ads you might see. If you're searching for an honest to goodness,
legitimate, legal and spare time work at home opportunity, then your
search has finally ended. This is a 100% proven Money-Making
Program! PROVEN to make good money for everyone who uses it!

If you're like most folks, you're going to absolutely love our 
Circular Mailing Program because it's the most legitimate, 
"on-the-level", easy to start, profitable work-from-home opportunity
ever created! We say this honestly because IT REALLY WORKS! You 
won't find any gimmicks, surprises or silly schemes. Only valuable
information that you'll need so you can quickly learn exactly WHAT
TO DO and HOW TO DO IT; and our proven , professionally written
circulars.

Our Circular Mailing Program can bring you all the money you need.
When you receive our start up package in the mail, you will get all
the supplies you need to get started right away including your 
Personal Information Kit (your complete instructional handbook) and
our business circulars. You will receive everything as promised.
And don't forget you will not have to pay postage costs to mail our
circulars because envelopes will arrive completely addressed and
with postage already in place. Just mail out the circulars and
receive pay cheques that are yours to spend any way you wish! You can
take part in our program for as long as you want... earn $700 a week
for the rest of your life.

We can only accommodate a limited number of people in our unique
program. So if you are interested, please do not delay. Send in 
your acceptance form as soon as you can. You have our guarantee that
this program can change your life practically overnight. We know of
no other home employment opportunity with the potential to make the
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This is a complete home-based opportunity that REALLY WORKS. Our
only requirement is a one time only, fully refundable payment of
only $27.00 . This payment covers the cost of your supplies and
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will not have to pay us any other costs to get additional materials.
And because we're so sure that we have the right home employment
opportunity for you, we are backing up our promises with our
exclusive guarantee...

   $33,600.00 Guarantee

You can easily earn $33,600.00 in the next year with our program.
In fact, we are so confident that you can make over $700 a week 
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up package in the mail, send out our circulars right away. If you
don't start earning a minimum of $700 a week within 30 days, simply
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Join our network of circular mailers today. We truly want to help
you get started as quickly and as easily as possible. This program
is designed for people who are serious about earning a substantial 
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If you follow our instructions, you will be well on your way to
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and fill out the Exclusive Membership Form at the bottom of this 
page and mail it in with your remittance and your order will be 
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We hope that you allow us the honor of being the ones who helped 
you to achieve long-term financial success and personal freedom.

 Most sincerely,
  The Staff at CyberMarketing
------------------------------------------------------------------
             HOME MAILERS PROGRAM ORDER-FORM
------------------------------------------------------------------
Please RUSH me my Package of the HOME MAILERS PROGRAM and the HOME
BUSINESS DIRECTORY right away!!! Send $27.00 in U.S Funds. 
(Includes Postage & Handling)
                                  
                         NAME:______________________________________
MAIL TO: CYBERMARKETING
P.O.Box #563             ADDRESS:___________________________________
Lindsay, Ontario,
Canada, K9V 4S5          CITY:_________________STATE/PROVINCE:______

	                 ZIP/POSTAL CODE:___________________________
                 
	                 E-MAIL ADDRESS:____________________________
	             
     (Orders payable by cheque please allow 4-6 weeks for delivery.)
07/30/99
--------------------------------------------------------------------



From rem-conf Sat Jul 31 18:32:16 1999 
From rem-conf-request@es.net Sat Jul 31 18:32:15 1999
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Date: 7/26/99 3:35:10 PM Pacific Daylight Time
From: spy@gohip.com
Reply-To:spy@gohip.com
To: spy@gohip.com
Subject:  Don't Get Ripped Off!!! There's only ONE Internet Spy and You Software...
Message-Id: <E11AkBU-0007i7-00@mail1.es.net>
X-Mailing-List: <rem-conf@es.net> 
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Precedence: list

 
Don't mistake this software for other $15 and $22 programs
being advertised on the net... you get what you pay for, and 
there's only ONE "Internet Spy & You" program!!!
 
Please see disclaimers at the end of this mailing.
 
<><><><><><><><><><><><><><><><><><><><><>
Welcome to another... ListBott "Opt-In" broadcast.
<><><><><><><><><><><><><><><><><><><><><>
 
Dear Subscriber, 
 
WE ARE BEING FORCED TO DISCONTINUE SELLING THIS INFORMATION.
Don't miss this very limited time opportunity to get the software that will  
allow you to become the ultimate "Super Snoop" on anyone !!!
 
Get The SOFTWARE They Want BANNED In All COUNTRIES!!!
 
"THE INTERNET SPY AND YOU"  SHOWS YOU 
      HOW TO GET THE FACTS ON ANYONE!!!
 
You now have a few  days left to get the software they want 
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intended to reach your eyes... Plus, this is a VERY LIMITED TIME OFFER !!! 
 
Get the facts on anyone using the Internet!
 
Locate Missing Persons, find Lost Relatives, obtain Addresses
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Beat Spouses.  This is not a Private Investigator, but a 
sophisticated SOFTWARE program DESIGNED to automatically
CRACK YOUR CASE with links to thousands of Public Record databases. 
 
Find out SECRETS about your relatives, friends, enemies,
and everyone else! -- even your spouse with the new,
INTERNET SPY AND YOU!!!
 
It's absolutely astounding!  Here's what you can learn:
 
License plate number!
Get anyone's name and address with just a license plate number!
(Find that girl you met in traffic!)
 
Driving record! 
Get anyone's driving record
 
Social security number!
Trace anyone by social security number!
 
Address! 
Get anyone's address with just a name!
 
Unlisted phone numbers! 
Get anyone's phone number with just a name - even unlisted numbers!
 
Locate! 
Long lost friends, relatives, a past lover who broke your heart!
 
E-mail! 
Send anonymous e-mail completely untraceable!
 
Dirty secrets!
Discover dirty secrets your in-laws don't want you to know!
 
Investigate anyone! 
Use the sources that private investigators use (all on the Internet)
secretly!
 
Ex-spouse!
Learn how to get information on an ex-spouse that will help you 
win in court!  (Dig up old skeletons)
 
Criminal search-background check!
Find out about your daughters boyfriend! 
(or her husband)
 
Find out! 
If you are being investigated!
 
Neighbors!
Learn all about your mysterious neighbors!  Find out what they 
have to hide!
 
People you work with!
Be astonished by what you'll learn about people you work with!
 
Education verification!
Did he really graduate college?  Find out!
 
Internet Spy and You
Software will help you discover ANYTHING about anyone, with 
clickable hyperlinks, no typing in Internet addresses!  Just
insert the floppy disk and Go!
 
You will be shocked and amazed by the secrets that can be
discovered about absolutely everyone!  Find out the secrets 
they don't want you to know!   About others, about yourself! 
 
It's INCREDIBLE what you can find out using the Internet Spy 
and You and the Internet!  You'll be riveted to your computer screen!
Get the software they're trying to ban!  
 
ACT NOW, Before it's too late!
 
LIMITED TIME OFFER -- THIS SPECIAL SOFTWARE OFFER EXPIRES
IN JUST A FEW DAYS!!!  ORDER TODAY!!!
 
Only $45.95 U.S.  (Postage Paid)
Only $39.95 U.S.  (If paying with Cash or Money Order.)
 
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begin discovering all the secrets you ever wanted to know!
 
You can know EVERYTHING about ANYONE with our Internet Spy and You 
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ORDER TODAY !!! 

Send Only $45.95 U.S.  (Postage Paid) 
OR, Only $39.95 U.S.  (If paying with Cash or Money Order.)
 
Sorry, NO PERSONAL OR BUSINESS CHECKS will be accepted.
 
(ORDERS OUTSIDE THE USA, ADD $25.00)
 
(ALL SOFTWARE PACKAGES SHIPPED WITHIN 7 DAYS)
 
DON'T WAIT TO GET STARTED... here's what to do:
 
STEP 1 - Compose a new message using the order form text below
STEP 2 - Type or print your order information into the order form section
STEP 3 -  Print your completed order, then mail to the address below
 
                       >>> Order Toll-Free <<<
OPTIONAL, you may choose to place your order on our secure voicemail 
system by calling 1-800-242-0363  Ext 2301
 
                        >>> Order By Mail <<<
(Must be called-in or postmarked "on" or "before" 8/6/99)
 
Send to:
KMM
P.O. Box 910652
St. George, UT  84791
USA

>>>>>>>>>>>>>>>>>>>>>>>><<<<<<<<<<<<<<<<<<<<<<<<
We have been forced to limit the distribution of this resource.
Orders received or postmarked AFTER 8/6/99, will be 
returned or deleted.  Sorry, there will be NO exceptions!!!
>>>>>>>>>>>>>>>>>>>>>>>><<<<<<<<<<<<<<<<<<<<<<<<
 
 
                        >>> Mail-in Order Form <<<
 
Auth: 4586
 
Name:
 
Address:
 
City:
 
State:
 
Zip:
 

Method of payment: 
 
[   ] Visa [   ] MasterCard [   ] American Express 
 
Credit Card#:
Exp Date:
YOUR SIGNATURE HERE:_______________________________
We cannot process your credit card payment without your signature!
  
[   ] Credit Card Orders $45.95!!!
[   ] Money Order mail in orders -- only $39.95!!!
[   ] Cash mail in orders -- only $39.95!!!
 
WE DO NOT ACCEPT PERSONAL CHECKS!!!
 
"Products & Services For Doing Business On The Net"
 
(SORRY, NO MAC VERSION AVAILABLE AT THIS TIME.)
 
NOTE: THIS PROGRAM WILL NOT WORK ON WINDOWS 3.11 AND OLDER
 
NOTE: Because this is software that can be copied, all sales are final, there will 
be no refunds on this offer.
 
DISCLAIMER:  The seller of this powerful software resource will not be held 
responsible for how the purchaser chooses to use its resources.
 
**************************************************************************************************
This mailing is done by an independent marketing company. We apologize 
if this message has reached you in error.  Save the Planet, Save the Trees!  
Advertise via E-mail.  No wasted paper!   
 
To be removed from this listserver, address book, please call us toll-free at 
1-800-242-0363 Ext 2301 and we'll promptly honor your remove request.







