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Subject: [AVT] Call for Papers: WiOpt'04
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Apologies if you receive multiple copies of this call!
Please note the deadline extension!

----------------------------------------------------------------------
  
                         DEADLINE EXTENSION

               The submission deadline has been extended
                     to the 12th of November, 2003.


        ***************************************************
        ********       2nd  CALL FOR PAPERS        ********
        ***************************************************
        *                                                 *
        *                   WiOpt '04                     *
        *                                                 *
        *     2nd Workshop on Modeling and Optimization   *
        *      in Mobile, Ad Hoc and Wireless Networks    *
        *                                                 *
        *        March 24 - 26, 2004, Cambridge, UK       *
        *                                                 *
        *                                                 *
        ***************************************************


                         Workshop Website:
           http://www.cl.cam.ac.uk/Research/SRG/wiopt04/


This workshop is intended to bring together leading researchers from
both academia and industry to present both foundations and practical
methods and tools for modeling and optimisation of mobile and ad hoc
networks.  The workshop will bring together researchers studying
optimisation issues of mobile networks from both the network
performance and the mobile networking perspectives:

*  Mobile Network Optimisation: Applications and Algorithms
*  Performance characterisation and optimisation of
   mobile/wireless networks and systems
*  Protocol design (MAC, Routing) for optimal mobile network
   utilisation
*  Tools for network optimisation
*  Energy efficiency in mobile networks
*  Security and co-operation models for mobile networks
*  Pricing and incentives in mobile and ad hoc networks.

Papers to be presented at WiOpt'04 will be selected based on extended
abstracts of up to 8 pages, double-spaced, 12 point font size. The
submission should not exceed 3000 words including tables and figures,
each of which count as 250 words.  Authors of accepted presentations
will have the opportunity to publish either the extended abstract or
the full paper in a proceedings volume. Full papers will further be
considered for publication in a special issue of the MONET journal. 

Plenary Speakers:

Bruce Hajek (University of Illinois at Urbana-Champaign, USA) 
Ian Akyildiz (Georgia Institute of Technology, USA) 


IMPORTANT DATES:
Submission of extended abstract: Nov. 12, 2003 (EXTENDED!).
Notification of acceptance:      Jan. 15, 2004.
Camera ready copy:               Feb. 15, 2004.

(Further details on the workshop website,
http://www.cl.cam.ac.uk/Research/SRG/wiopt04/)



General chairman: Imrich Chlamtac

Vice General Chair: Roger Whitaker 

Program Co-Chairs: Marco Conti and Tamer Basar

Program committee of the conference:

Eitan Altman, Venkat Anantharam, Konstantin Avrachenkov, Andrew
Campbell, X. Cao, A. Chockalingam, Jon Crowcroft, Anthony Ephremides,
Silvia Giordano, Stephen V. Hanly, Jean-Pierre Hubaux, Holger Karl,
Peter Key, P. R. Kumar, Peter Marbach, Derek Mcauley, Archan Misra,
Philippe Nain, Sven Oestring, Yoshikuni Onozato, Ariel Orda, Ramesh
Rao, Catherine Rosenberg, Ness Shroff, Violet Syrotiuk, Thierry
Turletti, Adam Wolisz, Jeff Wieselthier, Moshe Zukerman

Local Arrangements Co-Chair: Sven Oestring, Julian Chesterfield and
	Rajiv Chakravorty  

Treasurer: Meng How Lim

Publicity Chair: Holger Karl


The workshop is co-sponsored by
        * Univ. of Cambridge
        * IIT (Istituto di Informatica e Telematica)
        * IEEE IT Society (technical co-sponsorship)
(ACM sponsorship is pending.)

Additional financial support is provided by:
        * Intel
        * Microsoft Research



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Sat Nov  1 12:42:14 2003
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From: "Magnus Westerlund (KI/EAB)" <magnus.westerlund@ericsson.com>
To: "'hatanaka@av.crl.sony.co.jp '" <hatanaka@av.crl.sony.co.jp>,
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Subject: [AVT] Comments on draft-hatanaka-avt-rtp-atracx-03.txt
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Hi,

Here is my comments on draft-hatanaka-avt-rtp-atracx-03.txt

1. There seem to be some problem with the page length, it varies widely between some pages.

2. I think the introduction chapter is lacking some information. Due to the massive amount of functionality in the payload format there is a need to motivate some of this. Also some more details around the codex is needed, especially that has impact on the transport. For example is the codec using fixed time length slots, and what is there length?

3. In general I think the payload format is suffering of an extreme case of feature creap. Every imaginable possible usage has been given functionality without thinking if RTP or other layers can help provide the functionality. I think more work concering what intended use cases to solve is needed. Thus enable propoer understanding of the requierments. This mail will actually mostly consist of  questions regarding what is necessary.

4. Starting in section 1.2 and figure 1: What is the motivation behind the usage of segments and what are they purpose. As I understand they have multiple of usages and not all seems appropriate for RTP. To my understanding segements can be used to:
- Carry individual mono or stereo encoding blocks, part of a multi block channel combination, for example 5.1.
- Carry a complete encoding block, including 5.1
- Use multiple segments to carry multiple blocks of multi-channel content.
- Carry multiple different sources in one payload? (This is definitly not appropriate)
This latitude in the usage makes it hard to grepp how it should be best implemented and used. I think one should review the segment concept and if it really is appropriate and need, or if other solutions can be found.

5. Section, 2.1, figure 2: This figure is very confusing, when introducing a format it is normal useful to start on high level and few details, I would basically recommend a figure with: RTP header, payload header, data blocks, etc.

6.  The bit-fields drawings are confusing as they are lacking a horisontal divider between different rows of octets. Thus one has a hard time knowing if somehting is spread over multiple lines but consist of only one single octet row.

7. Section 2.3, the version field: Is this the version of the payload format? If it is, its not appropriate. Having the version in the payload format prevents the signaling protocol to know what versions are supported, etc. Thus it is better to have the version handling on the signalling layer, for example thru the usage of different MIME types for different versions that are not compatible.

8. Section 2.3: What is the purpose of the FRSEQNO? I don't see the meaning of this, I assume that one can use RTP TS to recover the slot the data belongs to.

9. Section 2.3: The elementID: I am not certain of the purpose of this identifier. Are they used to identify different segments? IF the purposes is to identify different sources the RTP SSRC exist for this purpose.

10. Section 2.3: The segments are they each having an individual complet RTP payload header as described in figure 3?

11. Figure 4: In this example, are there really need to be able to send 12+2 channels in the same RTP payload? How is the mapping between the different multi-channel accomplished? Each of the individual combinations, like 5.1 has a unique mapping and an assumed speaker setup but what does it mean when one have both 5.1 and 7.1 in a slot?

12. Section 2.3 The combination of the two frame blocks, how is this accomplished? Like when the LFE is combined with another stereo block?

13. Section 4. The meta data: If there isn't already defined a number of given metadata block types why is the metadata ID need. In the interest of interoperability isn't it best that meta data actually is understood and therefore defined. Thus each type could be assigned an ID.

14. What TS rate are you going to support, I woul assume, any of the three sampling rates, as long as you do not envision that one will switch frequently between them.


In my opinion you need to go back to the drawing board and rethink the design. This design seems to be much to complex for the purpose. However as you haven't clearly communicated the purpose I am not certain. I think there exist simple solutions.

If I would do this I would structure the payload format this way.

Remove the segment concept totally. If one needs to send several multichannel encodings in the same session I would use the SSRC instead.

Create a payload format that allows for aggregation of several slots in a nice way, thus enable the redundancy scheme.

The payload would then look like:

[RTP header] [Payload header] [Slot header: TS offset, nr of blocks(n)]  n*[Block: block type, block len, data] [Slot header: TS offset, nr of blocks(m)] m*[Block: block type, block len, data]

The payload would consist of a nested strucuture. First a packet common payload header, carrying any common information (not having analyzed what needs to go here). For each time slot there is a header giving the number of blocks the slot consist of and what the RTP TS offset is in regards to the RTP TS. Within each time slot there can be a number of blocks. Each block carries a type identifier, audio, metadata, etc, and a length field, then is the data of the block. The slot headers block count is used to know when there should be a new slot header if there still are more unparsed bytes of the packet.

The redunancy mechanism does not use any explicit type of data structure unless the codec requries it. IF redundancy is used the receiver may receive multiple sets of blocks for the same slot. The receiver simply selects the most appropriat, usually the one with most channels.

If priority is important to carry, that can be added to the appropriate header, probably block header. As I see it the priority in the payload is only useful if one foresees that one will have gateways or repacking entities when sending from IP to another net with possibly redudeced capabilities. In the IP to IP case the sender needs to perform all bit-rate adaptation prior to packetization.


Editorial comments:

- Read the ID nits and follow them. http://www.ietf.org/ID-nits.html
- Rethink the structure of the draft, introducation, definitions, general functionality, then details to make it more readible.

I am very willing to discuss this in Minneapolis.

Cheers

Magnus


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Sun Nov  2 04:01:19 2003
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From: "Even, Roni" <roni.even@polycom.co.il>
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Date: Sun, 2 Nov 2003 10:55:18 +0200 
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Subject: [AVT] SDP parameters for supporting H.263 and H.261 options
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Hi,
This is a short summary of the presentation we intend to have in Minneapolis

There were two drafts submitted addressing the same topic which is adding
optional SDP parameters to H.261 and H.263 for specifying codec parameters.

draft-even-avt-h263-h261-options-00.txt - uses fmtp
draft-nismail-mmusic-sdp-video-00.txt - uses gpmd



Issues to be discussed during the IETF meeting:

	1.	Can we adopt this work as an AVT working group item
      2.	GPMD is not an RFC and might not be adopted hence authors
prefer to use fmtp. We would like to discuss the process for doing that.
	3.	We are soliciting input on the defined parameters, did we
specify all.


Thanks
Roni Even

*************************************
Roni Even

Polycom Israel

Tel: +972-3-9251200
Cell: +972-55-481099
email:roni.even@polycom.co.il
*******************************************


_______________________________________________
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https://www1.ietf.org/mailman/listinfo/avt



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On Tuesday, Oct 14, 2003, at 22:15 Europe/London, Linus Walleij wrote:
> One minor:
>
>    This document also serves as the MIME type registration according 
> RFC
>                                                                     ^
>                                                                     to
> Apart from that it's ready to roll, IMHO.

I caught a few minor things:

- Section 2 is missing the citation [2]
- Section 3: "A sample rate of 8000 Hz is used" -> "MUST be used"?
- Section 3 needs to indicate that other RTP header fields are used as
   specified in RFC 3550 and any applicable profile.
- Section 6: This is a payload format, not a profile. Need to reword the
   first paragraph appropriately.
- Section 6: Would be appropriate to mention that RTP is not an 
appropriate
   transfer protocol for reliable octet streams, and that TCP is better 
in
   those cases. Should probably also mention the importance of congestion
   control, and reference the discussion in RFC 3551.
- Reference [10] should include the draft filename, so the RFC Editor 
can
   track the dependency

Other than these details, I think it's ready to go. If other members of 
the group have comments, they should be sent to the mailing list by the 
14th of November (the last day of the IETF meeting), so a revised draft 
can be submitted immediately the archives re-open.

-- 
Colin Perkins
http://csperkins.org/


_______________________________________________
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https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Sun Nov  2 13:33:05 2003
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Subject: Re: [AVT] TCP framing update
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On Tuesday, Oct 21, 2003, at 04:32 Europe/London, John Lazzaro wrote:
> 	Just submitted an updated version of the TCP framing
> I-D (draft-lazzaro-avt-rtp-framing-contrans-02.txt) to
> internet-drafts@ietf.org, you can pick it up at:
>
> http://www.cs.berkeley.edu/~lazzaro/sa/pubs/txt/current-contrans.txt
>
> 	Only the name and date have been updated -- the memo had been
> awaiting re-chartering, and was about to expire.  The re-chartering
> text of interest is:
>
>> - to provide a framing mechanism for RTP over TCP and TLS
>
> 	The draft-lazzaro-avt-rtp-framing-contrans-02.txt draft is
> driven from two requirements, which were culled from earlier
> discussions on avt@ietf.org, and from the discussion in San
> Francisco in the spring:
>
>   -1-  Compatibility with the TCP framing format in earlier
>        versions of the RTP RFCs.
>
>   -2-  Compatibility with draft-ietf-mmusic-sdp-comedia-05.txt
>
> 	It would be good to get feedback on whether the document falls
> short on either requirement.
>
> 	My take on the San Francisco discussions is that there is a
> need for a protocol that meets these requirements, to support existing
> codebases.  Based on that assessment, I'd like to request WG Item
> status for draft-lazzaro-avt-rtp-framing-contrans-02.txt.

If anyone objects to this being an AVT work item - to address the 
charter text mentioned above - please comment to the mailing list by 
14th November 2003.

Colin


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Mon Nov  3 01:26:06 2003
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Subject: RE: [AVT] Moving draft-jones-avt-audio-t38-01.txt forward
Date: Mon, 3 Nov 2003 08:23:56 +0200
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Hello,

Together with advantages (encryption for security needs, the same
port/address for RTP+audio and RTP+t38) the proposed encapsulation of
T.38 into RTP has some disadvantages.

1. The protection from and recovery of packet lost in T.38 is usually
stronger than that of RTP. So, using the RTP error correction for T.38
seems to be problematic.

2. The bi-directional switching between Voice-Band-Data and T.38 modes
remains problematic as in current implementations because the RTP
payload of VBD has no ID of fax session.

The more attractive approach is given in delayed contribution 303 of
CISCO (May 2003, ITU-T, Study Group 16) to support RTP encapsulated into
T.38. That idea is inversed to the discussed one, and has more
capabilities for implementation of reliable fax transfer.

Regards,
Vladimir Ulybin
AudioCodes, Ltd.
=20



-----Original Message-----
From: Paul E. Jones [mailto:paulej@packetizer.com]
Sent: Friday, October 31, 2003 11:01 PM
To: avt@ietf.org
Subject: [AVT] Moving draft-jones-avt-audio-t38-01.txt forward


Folks,

I would like to seek agreement from the AVT group to move the document
audio/t38 document forward.  As I have mentioned previously, the ITU
would
like to incorporate this in T.38 in an amendment to be approved in
January
2004.

The motivation for transporting T.38 via RTP is probably obvious to
most.
In fact, I recall several saying that that is the way it should have
been
done from the outset.  Most certainly, this will prove to be a good step
in
leveraging SRTP for fax to address security needs.

The document is relatively short, just seeking the MIME registration.
The
technical details will be specified in the ITU-T Recommendation T.38.

Can I get agreement to make this an official working group item?  Can we
try
to get this approved by the January 2004 ITU-T SG16 meeting?

Thanks,
Paul


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Hi Qiaobing,

See comments inline.

Qiaobing Xie wrote:

> Hi, Magnus,
> 
> Thank you very much for the feedback. Please see my comments/answers
> in-line.
> 
> Magnus Westerlund wrote:
> 
>>Hi Qiaobing,
>>
>>I have looked at your three dsr drafts:
>>  draft-ietf-avt-dsr-es202050-01.txt
>>  draft-xie-avt-xdsr-es202211-00.txt
>>  draft-xie-avt-xadsr-es202212-00.txt
>>
>>1. My first comment is that these three draft could easily be folded
>>into a single draft. To my understanding they have a lot in common. The
>>only real difference is the actual internal frame format and that 050
>>have shorter frames then 211 and 212. Then there is the need define
>>individual MIME types for each. However this seems to be about all.
>>Having a single draft will reduce the workload on IESG and RFC editor
>>slightly. It will also make it less RFC to keep track of.
> 
> 
> I talked with Steve C. about this a year ago. At that time, the
> front-ends were at very different maturity levels in ETSI. So we agreed
> to create separate draft for each front-end. Now things have progressed
> and my understanding is that all three front-ends above are finalized in
> ETSI. So, I think to combine them may be a good thing to do.
> 
> We can either merge 050 and 212 into one draft (since they are more
> logically related) and leave 211 stand-alone, or fold all three into one
> draft. I will double-check with my ETSI colleagues about this and if
> there is no other issues, will release merged version after this IETF
> meeting.
> 

I think they from an RTP transport perspective are all the same and 
therefore could be had in one specification.

> 
>>3. I think the security consideration is a bit thin. Are you really
>>certain that the format does not result in any special considerations?
> 
> 
> Besides defining their specific frame bit-format, these payload
> definitions do not add anything else to the standard RTP functions (they
> may be one of the most straightforward RTP payloads). I really do not
> see any special security issues introduced. I'd appreciate for any input
> on this.
> 

Okay, good. There is no DSR frame that can creates unusual high 
processing requirements on receiver side?

Except for the merger there seem to be few problems with these formats. 
I would some more revisions to get the editorial nits right. I spotted 
some minor editorial things, I didn't comment on them due to that I saw 
the need to merge the document and that process will probably reveal 
them and create some other instead.


Cheers

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Hi Gamze,

Comments inline:

Gamze Seckin wrote:

> 
> 2. Regarding the corruption statistics. For media that uses codec that 
> have prediction any error normally remains for a rather long time. 
> However it is very difficult to determine when the effects has been 
> removed. Therefore this measurement is hard to derive on the receiver 
> side. Combined with the definition it only provides a good, possibly bad 
> indication. Also looking at the reasons for corruption, the only real 
> reason is lost or late packets. A good feeling for the quality can be 
> determined by the server simply by employing packet loss vector 
> reporting and knowing when the buffer is empty.
> 
> [g]packet loss is definitely a way of estimating corruption, but this 
 > gives an algorithm in the decoder (or pre-decoder)a chance to provide
 > better statistics.

Yes, it might be possible for such an algorithm to do this. However the 
measurement will not provide that much information. I think you 
basically can only say this is good enough or not, there will not exist 
any possibility to say how bad things actually are. Otherwise you will 
need to clearly specify how this measurement algorithm shall work.

However one concern here is that the corruption measurement will be 
quite complex and require rather much memory to perform properly. In a 
video decoder it will require the measurement tool to track how the 
error propagates over the screen. For some audio codecs it will be 
impossible to determine when the error is totally removed from the loop. 
To do this you would need to have information about what the correct 
state is.

> 
> 3. The rebuffering measurements makes more sense. However there are at 
> least one client that does not really perform rebuffering, namely 
> quicktime. Instead is assumes that if the transmitted bit-rate is 
> acceptable any short term effects will go away and the client will later 
>   have a buffer fullness given by the initial buffering delay. The 
> interesting thing about this measurement is that in 3GPP they already 
> have a way of letting the server measure when rebuffering takes place, 
> namely the OBSN reporting specified in TS 26.234 REL 6 (only preliminary 
> versions available).
> 
> [g] yes, and I believe those statistics are useful, however they are 
 > tight with rate adaptation (only turned on when rate-adaptation
 > is enabled) and tightly coupled with other rtsp messages.

Yes, they however is another way of measuring this. I think the 
rebuffering measure is possibly more useful then the corruption 
measurement.


> 6. Section 1. Last sentence: "As all the other report blocks
> defined in [1], the streaming report blocks are applicable to
> one-to-one or one-to-many applications that generate
> RTCP SR or RR reports." This sentence is not really correct. The use of 
> the RTCP XR reports is possible in any application sending RTCP.
> 
> [g] himm,i am confused i thought that the first sentence covers everything - what is left out? 
> So [1] needs to be updated too?

What I object against is the distinction around RTCP SR or RR reports 
and the use of generate. I would rather formulate it as:

As all the other report blocks defined in [1], the streaming report 
blocks are applicable to one-to-one or one-to-many applications that 
send RTCP in the desired direction.


> 8. Section 2, Figure 1. This figure shows this report formats potential 
> need to have reporting from all layers on the receiving side. This might 
> be a problem for certain application architectures. I am not saying that 
> it can't be used, its only that some applications may have difficulties 
> to include your reports.
> 
> [g] we believe the statistics reported should be extensible also should be allowed to be ignored. 
> So if an end side cannot support it should put simply a zero - what do you think?

I think one need to consider the meaningfulness of allowing only some of 
the measurement to be given.

> 10. Section 3.3, PAge 5, first bullet after figure 2:
> "- at the AVP module: Packet drop and packet errors can sometimes be
> detected by incorrect or missing sequence number or checksum. " Do you 
> really need to consider bit-errors, the only allowed possibility for 
> bit-errors in RTP is if partial checksums are used. Bit-errors surviving 
> to the transport layer is the meaning to be caught by the UDP, etc., 
> checksum.
> 
> [g] there could be cases where bit errors are propagated (e.g. udp lite).

I think that one should be clearer about that this then. However the 
usage of UDP-Lite is still under a lot of discussion within the IETF. 
There are some very though problems for IP to accept bit-errors 
partially. If one are using another transport of RTP then it might be 
more common. I would still like you to be more clear that this might be 
the case if one uses such mechanisms as UDP-Lite and payload formats + 
codecs that accept it.


Cheers

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Mon Nov  3 13:44:18 2003
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> Magnus Westerlund <magnus.westerlund@ericsson.com> writes:
>
> I think that one should be clearer about that this then. However the
> usage of UDP-Lite is still under a lot of discussion within the IETF.
> There are some very though problems for IP to accept bit-errors
> partially. If one are using another transport of RTP then it might be
> more common. I would still like you to be more clear that this might be
> the case if one uses such mechanisms as UDP-Lite and payload formats +
> codecs that accept it.

It might be time for AVT to set a provisional policy with regards to
what payload format authors should or should not be doing with respect
to UDP-lite. If one is starting with a blank sheet of paper to
design a payload format, the road one takes to fully leverage UDP-lite
is a different road from the norm.  In the RTP MIDI case, for example,
I would need to add in a few checksums words in key places for the
recovery journal system to work well in a world where delivered packets
may have bit-errors in the payload.  This seemed too complicated to
add to a payload format so close to completion, so I decided not to
go that route.  But maybe a payload author starting design work
today should make a different decision ...

-------------------------------------------------------------------------
John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
-------------------------------------------------------------------------


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Mon Nov  3 20:59:35 2003
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Subject: [AVT] Summary changes to draft-ietf-avt-uncomp-video-04.txt.
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Dear all:

The following is a list of changes made to 
draft-ietf-avt-uncomp-video-04.txt:

1. For interlaced video the M bit is set at the end of both fields, and 
it indicates the
end of a field.

2. The qualifier for  RTP timestamps has been changed to SHOULD (from 
MUST):
       "A 90 kHz timestamp SHOULD be used ..."

3.  Two new SDP parameters have introduced:
      - chromo-position:
            designates the position of the chromo values relative to 
luma values
      - top-field-first:
            for interlaced video, defines with which field are the 
chromo samples first transmitted.

We feel with these changes the draft is complete and ready for Last 
Call, and would like to
receive feedback from the community in this regard.


Thank  you,
Ladan




_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Tue Nov  4 09:34:29 2003
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To: Ersue Mehmet <mehmet.ersue@siemens.com>
CC: "'ari.lakaniemi@nokia.com'" <ari.lakaniemi@nokia.com>,
        "'avt@ietf.org'" <avt@ietf.org>, Wenk Stefan <stefan.wenk@siemens.com>
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Subject: [AVT] Re: RFC3267 question regarding not sending of NO_DATA
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Hi Ersue,

This comment actually applies to payload format in general. If you look 
at the Octet-algined mode this section is referenced regarding the 
definition of the FT field. We should probably try to restructure this 
or make it clearer that it applies for both bandwidth-efficient and 
octet-align mode in the draft standard version.

The reasoning that it is very unnecessary to transmit NO_DATA frames 
when they instead can be deducted from the RTP TS is true for all modes, 
except interleaving where it does not hold.

I hope this clarifies it for you.

Cheers

Magnus

Ersue Mehmet wrote:

> Hi Magnus, Ari,
> 
> we are implementing RFC3267 octet aligned mode and during the 
> compliance cross check one question arose:
> 
> On page 18 in chapter 4.3.2, which belongs to chapter 
> "4.3 Bandwidth efficient mode" is written:
> 
>    "Note that packets containing only NO_DATA frames SHOULD NOT be
>    transmitted.  Also, frame-blocks containing only NO_DATA frames at
>    the end of a packet SHOULD NOT be transmitted, except in the case of
>    interleaving."
> 
> First we assumed that packets with only NO_DATA SHOULD NOT be sent 
> in Bandwith efficient mode only, because this is in the "Bandwith 
> efficient" chapter. 
> On the other hand we do not understand the remark in the next sentence, 
> because it mentions interleaving. Interleaving is defined for Octet 
> Aligned only. (See 3.8).
> 
> Are these "SHOULD NOT"s now valid also for "Octet Aligned"?
> 
> 
> Best Regards,
> Mehmet 

-- 

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com


_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Hi Magnus, Ari,

we are implementing RFC3267 octet aligned mode and during the 
compliance cross check one question arose:

On page 18 in chapter 4.3.2, which belongs to chapter 
"4.3 Bandwidth efficient mode" is written:

   "Note that packets containing only NO_DATA frames SHOULD NOT be
   transmitted.  Also, frame-blocks containing only NO_DATA frames at
   the end of a packet SHOULD NOT be transmitted, except in the case of
   interleaving."

First we assumed that packets with only NO_DATA SHOULD NOT be sent 
in Bandwith efficient mode only, because this is in the "Bandwith 
efficient" chapter. 
On the other hand we do not understand the remark in the next sentence, 
because it mentions interleaving. Interleaving is defined for Octet 
Aligned only. (See 3.8).

Are these "SHOULD NOT"s now valid also for "Octet Aligned"?


Best Regards,
Mehmet 


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: [AVT] AVT meeting Slides and Summary
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Hi,

As pointed out in the mail with the preliminary agenda, ALL persons with 
agenda time needs to do two things prior to the meeting:

1. Send a summary of open issues to the AVT mailing list. This needs to 
be done ASAP. The deadline for this has already passed!

2. Send slides in ppt or pdf to BOTH chairs before 12 noon EST (GMT 
-05:00) on Friday the 7th November.

If this is not accomplished we chairs may resolve any scheduling 
difficulties or time constraints by excluding you from the agenda.

Best Regards

Magnus Westerlund & Colin Perkins



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Tue Nov  4 10:36:22 2003
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From: "Jose Rey" <rey@panasonic.de>
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Subject: [AVT] Summary changes draft-rey-avt-3gpp-timed-text-01.txt and Timed Text presentation in Minneapolis
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Hi all,

this is what I would like to discuss on Monday:

- draft-rey-avt-3gpp-timed-text-01.txt presents a payload format for the
3GPP Timed Text media format. With respect to the -00 version in general
all sections were completed:
* completed empty sections from -00 draft.
* abstract and introduction re-arranged. Moved section "Basics of the
3GP File Structure" to end of the document as Annex B.
* SLEN, SIDX and SDUR lengths fixed to 16, 16 and 24 bits, respectively.
* New OPTIONAL header, SPLDESC, added to transport sample description
in-band.
* Section 4 on payload format expanded: text header, fragment header and
sample description header are fully specified.
* SMIL usage section added.

- Generic vs. Specific transport: in Vienna the issue was raised whether
it should be more generic. After the publication of -01 version, little
feedback has been heard about this; we would like to hear evereyone's
opinion so the work can continue without surprises in the future.

- fragmentation headers: the draft also defines complex fragmentation
headers (FHDR, MTSF) that enable to get the most out of each fragment
even when previous fragments were lost.  What is the opinion of the WG
on this? This may be an overkill as it increases complexity for just a
couple of cases in which they will be used...

- the PT-multiplexing will be abandoned because it is complex and gives
problems, for example in SDP and ROHC. Instead additional fields in
fixed header wil be used.

- the sizes of length, index and duration fields of the text samples
(SLEN, SIDX and SDUR) are currently 16, 16 and 24 bits.  We are not
clear if this is enough/too much.  They allow for text samples sizes up
64KBytes, 65536 different indexes and a duration of samples up to four
hours.  When thinking on reducing these lengths one could argue to
reduce SLEN to 1 byte but 256Bytes is too small . Taking a non-multiple
of 8 is not an option because of parsing complexity, so 16bits is OK.
For SIDX, 8 bits, should be enough I don't think there would be more
than 256 sample descriptions in a text sample. And for SDUR, 16 bits
would mean @1Khz ~1 minute, which is too short for streaming, so we
propose to keep 24bits; also to cope with the case of different
clockrate values. Comments?

- resilient transport: when a packet is fragmented into several pieces,
we think *at least* the fragments containing plain text strings shall be
playable regardless of previous arrived or not (i.e. self-contained).
Making the modifier boxes also resilient  introduces complexity, as seen
with the MTSF headers in the -01 draft. Therefore the tradeoff here
would be to keep/adapt the necessary fields in the fragment header
(FHDR) to display isolated text strings but abandon the MTSF headers for
the modifier boxes. A solution for the latter could be to protect this
through retransmissions, FEC or else.

- the -01 draft does not specify how to reassemble fragments. This will
be specified.

- all other comments on the list noted.

That's all, see you in Minneapolis.

thanks,

Jose






____________________________________
Jose Rey
Mobile Multimedia Team -  MCOM Group
Panasonic European Laboratories GmbH
Email: rey@panasonic.de
Tel:   +49(0)6103-766-134
Fax:   +49(0)6103-766-166
http://www.pel.panasonic.de
____________________________________


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Tue Nov  4 13:21:20 2003
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Subject: RE: [AVT] Retransmission draft
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Hi,

I think it would be good to put the current IPR situation into =
perspective. All that we know is that Matsushita have a patent =
application that could be applicable to the retransmission draft =
according to their judgement.
This does not mean that an actual patent will be granted or that the =
retransmission draft would really infringe on such a patent. At this =
point of time, saying that the retransmission draft is patented is only =
speculation. Such speculation should not in my opinion prevent the draft =
from progressing.

David.


> >
> > Taking all the above into account and considering that
> > independent of the patent situation this work is intended to be=20
> > included in
> > the next release of 3GPP, we would like to ask the group to proceed=20
> > with the
> > draft on standards track.
> >
> > cheers,
> > Jose
>=20
> [takes working group chair hat off]
>=20
> 3GPP is only one forum which uses IETF standards, and one=20
> where dealing=20
> with IPR issues is commonplace. Having patented standards is=20
> more of an=20
> issue for open source developers, who are often excluded by the=20
> licensing terms. Would Matsushita be willing to clarify if the=20
> "reasonable, nondiscriminatory" terms allow open source=20
> implementations=20
> to use the retransmission format (this will probably require a=20
> Royalty-Free license)?
>=20
> [puts working group chair hat on again]
>=20
> --=20
> Colin Perkins
> http://csperkins.org/
>=20
>=20
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
>=20

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Tue, 4 Nov 2003 18:20:30 +0000
Subject: Re: [AVT] Moving draft-jones-avt-audio-t38-01.txt forward
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From: Colin Perkins <csp@csperkins.org>
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Vladimir,

On Monday, Nov 3, 2003, at 06:23 Europe/London, Vladimir Ulybin wrote:
> Together with advantages (encryption for security needs, the same
> port/address for RTP+audio and RTP+t38) the proposed encapsulation of
> T.38 into RTP has some disadvantages.
>
> 1. The protection from and recovery of packet lost in T.38 is usually
> stronger than that of RTP. So, using the RTP error correction for T.38
> seems to be problematic.
>
> 2. The bi-directional switching between Voice-Band-Data and T.38 modes
> remains problematic as in current implementations because the RTP
> payload of VBD has no ID of fax session.
>
> The more attractive approach is given in delayed contribution 303 of
> CISCO (May 2003, ITU-T, Study Group 16) to support RTP encapsulated 
> into
> T.38. That idea is inversed to the discussed one, and has more
> capabilities for implementation of reliable fax transfer.

Since the RTP payload format for T.38 is being developed in ITU-T SG16, 
these comments would be better suited to that forum. The MIME type 
registration we are developing here is dependent on that work, and will 
only be published as an RFC when the ITU completes the revision to T.38 
including the payload format.

-- 
Colin Perkins
http://csperkins.org/


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Tue, 4 Nov 2003 18:27:29 +0000
Subject: Re: [AVT] Moving draft-jones-avt-audio-t38-01.txt forward
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To: "Paul E. Jones" <paulej@packetizer.com>
From: Colin Perkins <csp@csperkins.org>
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Paul,

On Friday, Oct 31, 2003, at 21:00 Europe/London, Paul E. Jones wrote:
> I would like to seek agreement from the AVT group to move the document
> audio/t38 document forward.  As I have mentioned previously, the ITU 
> would
> like to incorporate this in T.38 in an amendment to be approved in 
> January
> 2004.
>
> The motivation for transporting T.38 via RTP is probably obvious to 
> most.
> In fact, I recall several saying that that is the way it should have 
> been
> done from the outset.  Most certainly, this will prove to be a good 
> step in
> leveraging SRTP for fax to address security needs.
>
> The document is relatively short, just seeking the MIME registration.  
> The
> technical details will be specified in the ITU-T Recommendation T.38.
>
> Can I get agreement to make this an official working group item?  Can 
> we try to get this approved by the January 2004 ITU-T SG16 meeting?

After discussion, Magnus and I have decided that it is not appropriate 
to make this an official work item of AVT, because the payload format 
is not being developed in AVT. We are, however, willing to review the 
format and recommend its publication to the IESG, assuming there is 
consensus from the working group that this is appropriate.

Accordingly, this message is to solicit comments from the working group 
on the suitability of this MIME type registration for publication 
(assuming a matching payload format is developed by ITU SG16). Please 
send comments by Monday 17th November, so we have time to co-ordinate 
publication schedules with the ITU, IESG and RFC Editor.

-- 
Colin Perkins
http://csperkins.org/


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Tue Nov  4 14:37:26 2003
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From: "Andrea Basso" <Andrea_Basso@nmss.com>
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Subject: [AVT] Summary discussion items form Minneapolis - draft-basso-avt-videoconreq-00.txt
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We are planning to briefly discuss the draft
draft-basso-avt-videoconreq-00.txt in the coming IETF. The draft looks at
the requirements for the transport of video codec control commands.  We are
specifically  interested in discussing:

1- List of commands:
The current draft has a comprehensive list of commands for codec control.
Some of them are popular in particular in video conferencing applications.
Some others are much less popular and almost never used in current
applications. So the list needs trimming. We are looking for comments on
commands that need to be added to the list and the ones that can be safely
removed.

2- Reliability Requirements of codec control commands:
We are looking for comments on the reliability requirements for each of
commands discussed in the list resulting from the discussion of item 1.

3- Next steps:
   Is AVT a good home for the follow-up draft that will describe the
mechanisms of transport video codec control commands?


-Andrea


Andrea Basso
NMS Communications




_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Thread-Topic: [AVT] Draft agenda for Minneapolis
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From: "Ash, Gerald R (Jerry), ALABS" <gash@att.com>
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All,

Here is a brief summary regarding the following agenda item:

> RTP Header Compression over Multi-Hop Paths		(Ash)		30
>	draft-ash-e2e-voip-hdr-comp-rqmts-01.txt
http://ietf.org/internet-drafts/draft-ash-e2e-voip-hdr-comp-rqmts-01.txt

Some service providers (such as AT&T) have plans to migrate large =
amounts of legacy voice traffic to VoIP, as well as to provide VPN =
services enabling VoIP.  VoIP typically uses the encapsulation =
voice/RTP/UDP/IP/.  When MPLS=20
labels are added, this becomes voice/RTP/UDP/IP/MPLS-labels.  For an=20
MPLS VPN, the packet header is at least 48 bytes, while the voice=20
payload is often no more than 30 bytes.  VoIP header compression can =
significantly reduce the VoIP overhead through various compression =
mechanisms.  This is important on access links where bandwidth is =
scarce, and can be important on backbone facilities, especially where =
costs are high (e.g., some global cross-sections).  Providing VoIP =
header compression over multiple hop paths avoids multiple =
compression/decompression cycles at each router, and thereby increases =
the processing scalability of the maximum number of simultaneous VoIP =
flows which use header compression at each router. =20
 =20
I will summarize the problem statement, goals, example scenarios, =
requirements, and work items as described in =
http://ietf.org/internet-drafts/draft-ash-e2e-voip-hdr-comp-rqmts-01.txt.=
 =20

I'll also discuss the following issues:

1. Protocol extensions -- Extensions are proposed to [cRTP] and [ECRTP], =
which involve a new packet type field to identify FULL_HEADER, =
CONTEXT_STATE, etc. packets.  New objects are defined for [RSVP-TE].  =
Extensions are also proposed to RFC2547 VPNs, which create 'SCID routing =
tables' to allow routing based on the session context ID (SCID).  These =
extensions need coordination with other WGs (e.g., MPLS).

2. Resynchronization -- VoIP header compression over multiple-hop paths =
using cRTP header compression might not perform well with frequent =
resynchronizations.  [ECRTP] seems appropriate in this context.  The =
'simple' header compression approach would avoid the need for =
resynchronization, but achieves less efficiency than cRTP.

3. Scalability -- VoIP header compression over multiple-hop paths =
applied between CE-CE would perhaps require a large number of LSPs to be =
created.  There is concern for CE ability to do the necessary processing =
and the scalability of the architecture.

4. LDP application -- It would be desirable to signal the MPLS LSPs with =
LDP, since many RFC2547 VPN implementations use LDP as the underlying =
LSP signaling mechanism.

Please review the document and raise any issues/comments to the list.

Thanks,
Jerry Ash

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Tue Nov  4 21:32:24 2003
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From: "Paul E. Jones" <paulej@packetizer.com>
To: "Colin Perkins" <csp@csperkins.org>
Cc: <avt@ietf.org>
References: <8C68CB19-0EF4-11D8-83F8-000A957FC5F2@csperkins.org>
Subject: Re: [AVT] Moving draft-jones-avt-audio-t38-01.txt forward
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Colin,

It's perfectly fine with me if we take this course of action.  I just want
to ensure that we have the necessary MIME registration in place, so whatever
is necessary to make that happen, I'm willing to do it.

Please advise on the next step to be taken to get the necessary
registration.

Paul

----- Original Message ----- 
From: "Colin Perkins" <csp@csperkins.org>
To: "Paul E. Jones" <paulej@packetizer.com>
Cc: <avt@ietf.org>
Sent: Tuesday, November 04, 2003 1:27 PM
Subject: Re: [AVT] Moving draft-jones-avt-audio-t38-01.txt forward


> Paul,
>
> On Friday, Oct 31, 2003, at 21:00 Europe/London, Paul E. Jones wrote:
> > I would like to seek agreement from the AVT group to move the document
> > audio/t38 document forward.  As I have mentioned previously, the ITU
> > would
> > like to incorporate this in T.38 in an amendment to be approved in
> > January
> > 2004.
> >
> > The motivation for transporting T.38 via RTP is probably obvious to
> > most.
> > In fact, I recall several saying that that is the way it should have
> > been
> > done from the outset.  Most certainly, this will prove to be a good
> > step in
> > leveraging SRTP for fax to address security needs.
> >
> > The document is relatively short, just seeking the MIME registration.
> > The
> > technical details will be specified in the ITU-T Recommendation T.38.
> >
> > Can I get agreement to make this an official working group item?  Can
> > we try to get this approved by the January 2004 ITU-T SG16 meeting?
>
> After discussion, Magnus and I have decided that it is not appropriate
> to make this an official work item of AVT, because the payload format
> is not being developed in AVT. We are, however, willing to review the
> format and recommend its publication to the IESG, assuming there is
> consensus from the working group that this is appropriate.
>
> Accordingly, this message is to solicit comments from the working group
> on the suitability of this MIME type registration for publication
> (assuming a matching payload format is developed by ITU SG16). Please
> send comments by Monday 17th November, so we have time to co-ordinate
> publication schedules with the ITU, IESG and RFC Editor.
>
> -- 
> Colin Perkins
> http://csperkins.org/
>


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Wed, 05 Nov 2003 12:35:48 +0900
To: "Magnus Westerlund (KI/EAB)" <magnus.westerlund@ericsson.com>
Cc: hatanaka@av.crl.sony.co.jp, matthew.romaine@jp.sony.com, avt@ietf.org,
        nishi@av.crl.sony.co.jp, jun@av.crl.sony.co.jp
From: Jun Matsumoto <jun@av.crl.sony.co.jp>
In-Reply-To: <034BEFD03799D411A59F00508BDF754602BF41CA@esealnt448.al.sw.ericsson.se>
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Dear Mr.Westerlund,

>In my opinion you need to go back to the drawing board and rethink the design. 
>This design seems to be much to complex for the purpose. However as you haven't clearly 
>communicated the purpose I am not certain. I think there exist simple solutions.

OK, We agree that the complexity of our proposal.
Reffering your advise, we will re-design a simplified payload format,
and then make a proposal as ***-atracx-04.txt by Seoul meeting.

We have decided we do not go to Minneapolis meeting, so we would like you 
to cancel our presentation time slot.

Thank you very much for your useful advise.

Best Regards,

--------------------------------------
  Jun Matsumoto
   
  Audio Codec Development Department
  Sony Corporation, IMNC  

  jun@av.crl.sony.co.jp
--------------------------------------

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Thread-Topic: RTP payload format for AMR-WB+ audio codec
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Subject: [AVT] RTP payload format for AMR-WB+ audio codec
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hello all,

Here is a very short summary about the agenda item "RTP payload format =
for AMR-WB+ audio codec" (draft-sjoberg-avt-rtp-amrwbplus-00.txt).

As the title suggests, the draft proposes an RTP payload format for the =
AMR-WB+ codec, which is currently being specified by 3GPP TSG SA WG4. =
The main topics of the presentation/discussion are

1. Introduction of the AMR-WB+ codec
- audio extension to the AMR-WB speech codec

2. Proposal for the RTP payload format for AMR-WB+ codec
- based on AMR-WB RTP payload format (RFC 3267)

3. Summary of differences between the AMR-WB RTP format and the proposed =
format for AMR-WB+

4. Next steps
- feedback from the group
- group work item?

---
Ari Lakaniemi
Nokia Research Center

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Folks,

As you may have noticed, the final IETF agenda moves the first AVT 
slot, giving us 30 minutes less discussion time. Attached is a revised 
agenda, taking this into account. Apologies for the change - there were 
scheduling conflicts which have, so far, proved impossible to 
resolvable.

-- 
Colin Perkins
http://csperkins.org/

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Audio/Video Transport Working Group

CHAIRS: Colin Perkins     <csp@csperkins.org>
	Magnus Westerlund <magnus.westerlund@ericsson.com>


Monday, 10th October 2003 at 13:00-15:00
========================================

Introduction and Status Update				(Chairs)         5
	draft-ietf-avt-crtp-enhance-07.txt		=> RFC 3545
	draft-ietf-avt-rtp-new-12.txt			=> RFC 3550
	draft-ietf-avt-profile-new-13.txt		=> RFC 3551
	draft-ietf-avt-rtp-mime-06.txt			=> RFC 3555
	draft-ietf-avt-rtcp-bw-05.txt			=> RFC 3556
	draft-ietf-avt-dsr-05.txt			=> RFC 3557
	draft-ietf-avt-evrc-smv-03.txt			=> RFC 3558
	draft-ietf-avt-rtp-retransmission-09.txt	[AD Evaluation]
	draft-ietf-avt-rtcp-feedback-07.txt		[AD Evaluation]
	draft-burmeister-avt-rtcp-feedback-sim-02.txt	[AD Evaluation]
	draft-ietf-avt-tcrtp-07.txt			[AD Evaluation]
	draft-ietf-srtp-09.txt 				[IESG review]
	draft-ietf-avt-mpeg4-simple-08.txt		[RFC Editor]
	draft-ietf-avt-rtcp-report-extns-06.txt		[RFC Editor]
	draft-ietf-avt-uxp-06.txt
	draft-ietf-avt-rfc2833bis-03.txt	
	draft-ietf-avt-rtp-clearmode-03.txt
	draft-ietf-avt-mpeg1and2-mod-00.txt
	draft-kerr-avt-vorbis-rtp-03.txt
	draft-herlein-speex-rtp-profile-01.txt 
	draft-jones-avt-audio-t38-01.txt
	draft-ramalho-rgl-rtpformat-02.txt
	draft-chen-rtp-bv-00.txt	

Updates on behalf of author				(Chairs)        10
	draft-ietf-avt-ulp-08.txt
	draft-ietf-avt-rtp-jpeg2000-04.txt
	draft-ietf-avt-midi-rtp-00.txt
	draft-ietf-avt-midi-guidelines-00.txt
	draft-lazzaro-avt-rtp-framing-contrans-02.txt

RTP payload format for AMR-WB+ audio codec		(Lakaniemi)     15
	draft-sjoberg-avt-rtp-amrwbplus-00.txt
	
Highlights of VMR-WB RTP Payload and Storage Format 	(Ahmadi)        15 
	draft-ahmadi-avt-rtp-vmr-wb-00.txt

RTP payload formats for ETSI DSR front ends		(Xie)           15
	draft-ietf-avt-dsr-es202050-01.txt
	draft-xie-avt-xdsr-es202211-00.txt
	draft-xie-avt-xadsr-es202212-00.txt

RTP payload format for ATRAC-X				(Hatanaka)      15
	draft-hatanaka-avt-rtp-atracx-03.txt

Internet Low Bit Rate Codec & its RTP payload format	(Duric)         15
	draft-ietf-avt-rtp-ilbc-03.txt
	draft-ietf-avt-ilbc-codec-03.txt
	
RTP Payload for Text Conversation			(Hellstrom)     15	
	draft-hellstrom-avt-rfc2793bis-02.txt

Way forward for the RTP payload format for Timed Text	(Rey)           15
	draft-rey-avt-3gpp-timed-text-01.txt



Wednesday, 12th October 2003 at 09:00-11:30
===========================================

RTP Payload Format for Uncompressed Video 		(Gharai)        10
	draft-ietf-avt-uncomp-video-04.txt

RTP payload Format for H.264 Video			(Wenger)        15
	draft-ietf-avt-rtp-h264-03.txt

RTCP Extensions for SSM with Unicast Feedback		(Ott)           15
	draft-ietf-avt-rtcpssm-05.txt

Extended Secure RTP Profile for RTCP-based Feedback 	(Ott)           15
	draft-ietf-avt-profile-savpf-00.txt

RTCP Streaming Extended Reports				(Tseng)         15
	draft-tseng-avt-rtcp-streaming-extens-00.txt

Proposed additions to RTP MIB				(Clark)         15
	 draft-clark-avt-rtpmibv2-00.txt
	 
Datagram Congestion Control Protocol (DCCP) and RTP	(Kohler)        15
	draft-ietf-dccp-spec-05.txt
	draft-ietf-dccp-ccid2-04.txt
	draft-ietf-dccp-ccid3-04.txt
	draft-ietf-dccp-ccid3-thin-00.txt	 

RTP Header Compression over Multi-Hop Paths		(Ash)           15
	draft-ash-e2e-voip-hdr-comp-rqmts-01.txt

Requirements for transport of video control commands 	(Basso)         15
	draft-basso-avt-videoconreq-00.txt

SDP parameters for supporting H.263 and H.261 options	(Even/Ismail)   20
	draft-even-avt-h263-h261-options-00.txt
	draft-nismail-mmusic-sdp-video-00.txt


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_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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To: Jun Matsumoto <jun@av.crl.sony.co.jp>
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        nishi@av.crl.sony.co.jp
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Dear Jun,

I hope that you are not discouraged to continue the work by my rather 
long mail. As I see it there are some problems that one faces when 
trying to write a payload format for a codec within the AVT WG. The 
first is a matter of learning. One needs to have a rather good 
understanding of the codecs capabilities and working to know what 
requirements it has on the transport, thus it is usually codec people 
that comes to AVT. Then one must learn about how RTP and IP works and 
what the available mechanisms are. This will require one to learn a 
rather big pile of documents. Sorry to say we chairs or other group 
members does not always have the time to early enough point out some 
important details. I hope we in the future can improve this and also 
create some more informational material to help RTP payload designers.

I am also very sorry to hear that you are not coming to Minneapolis, I 
would have very much liked to discuss your payload format's design. 
Having direct communication can help immensely to ensure that the 
important aspects are covered.

I would also recommend that you try to create a new proposal in good 
time prior to the IETF meeting, thus we can make better progress. People 
usually have more time for discussion things when not stressed by 
deadlines or hugh workload prior to a meeting. It also allows you time 
to discuss or ask questions halfway the design process.

When it comes to payload format design I think there are some general 
things that needs to be ensured:

- Follow the application level framing principle, i.e. self contained 
packets.
- Keep it simple.
- Utilize the available mechanism in RTP in a good way.

Again if you have questions regarding design choices, please voice them 
to the list.

Best Regards

Magnus


Jun Matsumoto wrote:

> Dear Mr.Westerlund,
> 
> 
>>In my opinion you need to go back to the drawing board and rethink the design. 
>>This design seems to be much to complex for the purpose. However as you haven't clearly 
>>communicated the purpose I am not certain. I think there exist simple solutions.
> 
> 
> OK, We agree that the complexity of our proposal.
> Reffering your advise, we will re-design a simplified payload format,
> and then make a proposal as ***-atracx-04.txt by Seoul meeting.
> 
> We have decided we do not go to Minneapolis meeting, so we would like you 
> to cancel our presentation time slot.
> 
> Thank you very much for your useful advise.
> 
> Best Regards,
> 
> --------------------------------------
>   Jun Matsumoto
>    
>   Audio Codec Development Department
>   Sony Corporation, IMNC  
> 
>   jun@av.crl.sony.co.jp
> --------------------------------------

-- 

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
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Subject: [AVT] Re: ULP multiplexing schemes
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Hi Adam,

I thought this mail being of such general discussion value that I CC the 
mailing list in my response, I hope you don't feel hurt.

Adam Li wrote:

> Hi Magnus,
> 
> I have a couple of questions about multiplexing. 
> 
> RFC 3550 says "For efficient protocol processing, the number of
> multiplexing points should be minimized, as described in the integrated
> layer processing design principle." It specifically recommends against
> multiplexing multiple stream of different media type in the same session
> based on PT and SSRC. (It does recognize multiplexing media of the same
> type with SSRC.) With this, shouldn't we discourage multiplexing, i.e.,
> putting the media and FEC streams (they should be regarded as different
> media?) in the same address/port and multiplexing them only based on PT
> or SSRC?

Yes, and no. In some sense one might consider it as another media, and 
thus send it in a separate session. Using separate sessions has the 
advantage when one likes to be able do disconnect the ULP media. On the 
other hand the ULP stream is the same media as it encodes over. IT 
shares the timestamp behaviour, delivery requirements, etc. If you look 
in the list of argument in RFC 3550 against multiplexing separate media 
into the a single session you see that they does not apply for this case:

    1. If, say, two audio streams shared the same RTP session and the
       same SSRC value, and one were to change encodings and thus acquire
       a different RTP payload type, there would be no general way of
       identifying which stream had changed encodings.

This does not apply as they after reconstruction are a single stream.

    2. An SSRC is defined to identify a single timing and sequence number
       space.  Interleaving multiple payload types would require
       different timing spaces if the media clock rates differ and would
       require different sequence number spaces to tell which payload
       type suffered packet loss.

As the ULP simply integrates into the timestamp space and does not 
really use it, there is no problem. The sequence number does for most 
payload formats not create any problem, the current exceptions are some 
fragmented video formats and RFC 2793. It is after all if the there is 
needed to know if the lost packet was an ULP packet or real media that 
enables the usage of this or not.

    3. The RTCP sender and receiver reports (see Section 6.4) can only
       describe one timing and sequence number space per SSRC and do not
       carry a payload type field.

This is not a concern for ULP as it does not use the TS space, and 
packet loss reports can just as well be done aggregated then separated.

    4. An RTP mixer would not be able to combine interleaved streams of
       incompatible media into one stream.

An mixer is an user of ULP to recovery the stream for mixing therefore 
it is not a problem.

    5. Carrying multiple media in one RTP session precludes: the use of
       different network paths or network resource allocations if
       appropriate; reception of a subset of the media if desired, for
       example just audio if video would exceed the available bandwidth;
       and receiver implementations that use separate processes for the
       different media, whereas using separate RTP sessions permits
       either single- or multiple-process implementations.

This is a valid reason and in cases where it applies the session 
multiplexing should be used.

    Using a different SSRC for each medium but sending them in the same
    RTP session would avoid the first three problems but not the last
    two.

So in summary, there are two of these points that applies for ULP as I 
see it: 2 and 5. And this is fully dependent on the use cases and media 
transported. I think the concerns around point 2 is the more important 
one and these can be addressed though the usage of RFC 2198 for ULP 
information.


> 
> Another question is about the Session Multiplexing. It seems that we
> usually refer to multiplexing as the practice of putting multiple
> streams in the same socket connection (same address/port pair), and
> separate the streams based on some marks on the packets. If two streams
> are in separate sessions, then maybe it is a little confusing to call
> them multiplexing. 

Yes, however there is a clear binding between the ULP stream and its 
original data stream. Without the capability to bind them together the 
ULP information becomes useless.

> 
> I guess you are more concerned about how we can associate the FEC
> streams to the payload streams under different scenarios, right? Sending
> the FEC stream in another session seems is just a very special case of
> the many other ways to send the FEC stream separately. And it is a quite
> un-natural way. It is not obvious how to associate a stream in one
> session to another stream in another session? I feel it may not be
> necessary to specify all the details for every combinations and
> possibilities, because there are simply way too many of them to send two
> streams separately. 

This is exactly my concerns, and I do not agree that there is not need 
to specify how they are used. I can agree that not all possible needs to 
be specified, however I feel that the different major cases needs to be 
defined.

As I see it we have three different ways that for different usage are 
needed due to different scenarios, these should be clearly specified how 
to use. The three cases are:

- PT multiplexing, restricted for some payload formats that needs it 
sequence number untouched, RFC 2793 and some fragmentation schemes in 
some payload formats, like H.264 will not like it.

- Usage of RFC 2198 to include ULP data as redundant data.

- Sending ULP in separate RTP session, need when one wants to have 
separate treatment of the data streams or avoid sending useless data to 
receivers that can't decode it.

> 
> My suggestion is that we simply mention in the draft that PT
> multiplexing is possible, SSRC multiplex is not possible, and refer to
> the RFC 3550 to discourage multiplexing FEC stream with the payload
> through the same address/port. 
> 

All of the three above mentioned methods has separate concerns regarding 
signalling and usage. For example the session multiplexing scheme needs 
to use grouping of media lines (RFC3388) to bind them together 
explicitly.  There are also SSRC consideration rules for this, see the 
RTP retransmission draft. The usage of RED is already in there, and some 
paragraphs about the PT multiplexing and its concerns are also needed. I 
would also include a motivation why SSRC multiplexing is not a good method.

Cheers

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Wed Nov  5 11:43:24 2003
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Subject: [AVT] Summary for draft-ietf-avt-rtp-h264-03.txt
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Hi

This is the summary of open issues to be discussed for the H.264 RTP 
payload format:

o The security section needs review.

o An optional MIME/SDP parameter, max-don-diff, was added as an option 
to control the receiver buffering in the interleaved packetization mode. 
  max-don-diff is specified similarly to the maximum displacement 
parameter in the draft-ietf-avt-mpeg4-simple Internet Draft, but instead 
of using a maximum difference in terms of RTP timestamps a maximum 
difference in terms of decoding order numbers is used.  Does 
max-don-diff bring benefits that justify the complicated receiver 
description?

o ITU-T H.241 provides a way for decoders to signal capability for 
greater processing speed or memory amount than required in the profile 
and level that is used.  H.241 specifies CustomMaxMBPS, CustomMaxFS, 
CustomMaxDPB, and CustomMaxBRandCPB.  Should similar parameters be 
specified as optional MIME/SDP parameters to enhance the capability 
exchange procedure of SIP-based video conferencing?

Please send any comment to the AVT list.

Best Regards

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: [AVT] RTP Payload and Storage File Formats for VMR-WB
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All,

The following is a summary of the agenda item "Highlights of VMR-WB RTP =
Payload and Storage Format".

A new Internet Draft (ID) has recently been proposed for VMR-WB; i.e., =
the new cdma2000 wideband speech codec standard.=20

Due to algorithmic similarities between VMR-WB and AMR-WB and =
understanding that the two codecs are interoperable, a large part of =
RFC3267 has been reused in VMR-WB payload and storage formats. It was =
ensured that the reuse of the existing formats for AMR-WB has not =
compromised the optimal operation of VMR-WB; given the differences =
between the native systems for which the two codecs have been developed. =
 =20

I will provide a short presentation on 11/10/03, highlighting the =
important aspects of this new ID. The outline of my presentation is as =
follows:

* Background on VMR-WB
* Motivation for the Codec Interoperability
* Interoperability with AMR-WB
* Overview of VMR-WB RTP Payload Format
* Overview of VMR-WB Storage File Format
* Applications of VMR-WB

Your feedback and comments are welcomed.=20

Regards

-Sassan Ahmadi, Ph.D.
 Nokia Inc. USA

_______________________________________________
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Subject: [AVT] RFC 3611 on RTP Control Protocol Extended Reports (RTCP XR)
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--NextPart


A new Request for Comments is now available in online RFC libraries.


        RFC 3611

        Title:      RTP Control Protocol Extended Reports (RTCP XR)
        Author(s):  T. Friedman, R. Caceres, A. Clark, Eds.
        Status:     Standards Track
        Date:       November 2003
        Mailbox:    timur.friedman@lip6.fr, caceres@watson.ibm.com,
                    alan@telchemy.com
        Pages:      55
        Characters: 126736
        Updates/Obsoletes/SeeAlso:    None

        I-D Tag:    draft-ietf-avt-rtcp-report-extns-06.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3611.txt


This document defines the Extended Report (XR) packet type for the
RTP Control Protocol (RTCP), and defines how the use of XR packets
can be signaled by an application if it employs the Session
Description Protocol (SDP).  XR packets are composed of report
blocks, and seven block types are defined here.  The purpose of the
extended reporting format is to convey information that supplements
the six statistics that are contained in the report blocks used by
RTCP's Sender Report (SR) and Receiver Report (RR) packets.  Some
applications, such as multicast inference of network characteristics
(MINC) or voice over IP (VoIP) monitoring, require other and more
detailed statistics.  In addition to the block types defined here,
additional block types may be defined in the future by adhering to
the framework that this document provides.

This document is a product of the Audio/Video Transport Working Group
of the IETF.

This is now a Proposed Standard Protocol.

This document specifies an Internet standards track protocol for
the Internet community, and requests discussion and suggestions
for improvements.  Please refer to the current edition of the
"Internet Official Protocol Standards" (STD 1) for the
standardization state and status of this protocol.  Distribution
of this memo is unlimited.

This announcement is sent to the IETF list and the RFC-DIST list.
Requests to be added to or deleted from the IETF distribution list
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specifically noted otherwise on the RFC itself, all RFCs are for
unlimited distribution.echo 
Submissions for Requests for Comments should be sent to
RFC-EDITOR@RFC-EDITOR.ORG.  Please consult RFC 2223, Instructions to RFC
Authors, for further information.


Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

...

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_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Wed Nov  5 17:50:39 2003
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Subject: RE: [AVT] Moving draft-jones-avt-audio-t38-01.txt forward
Date: Wed, 5 Nov 2003 15:48:59 -0700
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Thread-Topic: [AVT] Moving draft-jones-avt-audio-t38-01.txt forward
Thread-Index: AcOf8i8E3Y9dOALPRq2EPGmu6PnUfQD+NHBA
From: "Jean-Francois Mule" <jf.mule@cablelabs.com>
To: "Paul E. Jones" <paulej@packetizer.com>, <avt@ietf.org>
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Hi Paul,

I did a brief review of draft 01 - glad to see updtl is not the end game
for t.38 fax. Find below some comments.
Hope this helps.=20
Jean-Francois.

--- editorial comment
In section 3, secure RTP should be named or mentioned as one of the
advantages of going to t38/rtp for its overall capabilities. The concern
is that the paragraph in section 3 only refers to confidentiality of fax
streams (encryption) while I think that authentication is going to be as
important.=20
Also consider change "an RTP" to "a RTP"
--
A number of issues were raised with respect to the usage of UDPTL for
   the long-term, though.  Specifically, there were concerns over the
   fact that UDPTL does not provide the same kind of statistics
   reporting as RTCP.  Further, there are no procedures in place for
   encrypting the UDPTL stream.  While the latter could be addressed in
   UDPTL, doing so would require a lot of effort and would largely be a
   duplication of the security work already on-going within the IETF.
--

--- technical comment
The section 6 (security considerations) is rather silent on the security
threats.=20
Looking at the IFP packet format, I would argue that those packets do
expose serious threats to the communication that you don't see in audio
voice. If you insert a bad RTP packet voice payload, oh well, you may
hear some garbage or clipping. With rtp/t38 fax, bad packets may cause
real harm:=20
+ denial of service attacks can be mounted by inserting RTP/t38 packets
with hdlc-sig-end data field can simply terminate the fax control
channel. Same with HDLC-FCS-BAD-Sig-End. And one can easily create those
packets.
+ theft of terminal identity? In some countries, faxes have more legal
value that emails and the access to the rtp packet content could change
the Called Subscriber ID frame (phone numbers), etc.  As you well know,
this info is usually not coded in RTP but in signaling protocols.

I'm sure the Study Group has thought of more. Could some examples of
security threats be given in the security consideration section and
shouldn't this section again reference SRTP?=20
--
   The IFP packets carried via RTP do not carry any particular security
   risks in and of themselves.  However, as with other audio packets,
   there is the potential for unauthorized eavesdropping or manipulation
   of facsimile transmissions if the media stream is not properly
   secured.

   The security considerations discussed in the RTP specification [3]
   and any applicable RTP profile (for example, [8]) are also
   applicable.
--




> -----Original Message-----
> From: Paul E. Jones [mailto:paulej@packetizer.com]=20
> Sent: Friday, October 31, 2003 2:01 PM
> To: avt@ietf.org
> Subject: [AVT] Moving draft-jones-avt-audio-t38-01.txt forward
>=20
>=20
> Folks,
>=20
> I would like to seek agreement from the AVT group to move the=20
> document audio/t38 document forward.  As I have mentioned=20
> previously, the ITU would like to incorporate this in T.38 in=20
> an amendment to be approved in January 2004.
>=20
> The motivation for transporting T.38 via RTP is probably=20
> obvious to most. In fact, I recall several saying that that=20
> is the way it should have been done from the outset.  Most=20
> certainly, this will prove to be a good step in leveraging=20
> SRTP for fax to address security needs.
>=20
> The document is relatively short, just seeking the MIME=20
> registration.  The technical details will be specified in the=20
> ITU-T Recommendation T.38.
>=20
> Can I get agreement to make this an official working group=20
> item?  Can we try to get this approved by the January 2004=20
> ITU-T SG16 meeting?
>=20
> Thanks,
> Paul
>=20
>=20
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
>=20

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Wed Nov  5 17:54:25 2003
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Subject: RE: [AVT] Moving draft-jones-avt-audio-t38-01.txt forward
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From: "Jean-Francois Mule" <jf.mule@cablelabs.com>
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Colin Perkins wrote:
> After discussion, Magnus and I have decided that it is not=20
> appropriate=20
> to make this an official work item of AVT, because the payload format=20
> is not being developed in AVT. We are, however, willing to review the=20
> format and recommend its publication to the IESG, assuming there is=20
> consensus from the working group that this is appropriate.
This is appropriate in my opinion. I also agree with Paul that it is
important to move this forward asap and facilitate the ITU meeting.

> Accordingly, this message is to solicit comments from the=20
> working group=20
> on the suitability of this MIME type registration for publication=20
> (assuming a matching payload format is developed by ITU SG16). Please=20
> send comments by Monday 17th November, so we have time to co-ordinate=20
> publication schedules with the ITU, IESG and RFC Editor.
Sent my comments.

Jean-Francois Mule
jfm@cablelabs.com

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Gamze Seckin" <gamze@vidiator.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>,
        "IETF AVT WG" <avt@ietf.org>
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This is a multi-part message in MIME format.

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Hi,
please see comments inline.
=20
thanks
g

-----Original Message-----
From: Magnus Westerlund [  <mailto:magnus.westerlund@ericsson.com> =
mailto:magnus.westerlund@ericsson.com]
Sent: Monday, November 03, 2003 3:55 AM
To: Gamze Seckin
Cc: Alan Tseng; IETF AVT WG
Subject: Re: Comments on draft-tseng-rtcp-streaming-extns-00.txt


Hi Gamze,

Comments inline:

Gamze Seckin wrote:

>
> 2. Regarding the corruption statistics. For media that uses codec that
> have prediction any error normally remains for a rather long time.
> However it is very difficult to determine when the effects has been
> removed. Therefore this measurement is hard to derive on the receiver
> side. Combined with the definition it only provides a good, possibly =
bad
> indication. Also looking at the reasons for corruption, the only real
> reason is lost or late packets. A good feeling for the quality can be
> determined by the server simply by employing packet loss vector
> reporting and knowing when the buffer is empty.
>
> [g]packet loss is definitely a way of estimating corruption, but this
 > gives an algorithm in the decoder (or pre-decoder)a chance to provide
 > better statistics.

Yes, it might be possible for such an algorithm to do this. However the
measurement will not provide that much information. I think you
basically can only say this is good enough or not, there will not exist
any possibility to say how bad things actually are. Otherwise you will
need to clearly specify how this measurement algorithm shall work.

However one concern here is that the corruption measurement will be
quite complex and require rather much memory to perform properly. In a
video decoder it will require the measurement tool to track how the
error propagates over the screen. For some audio codecs it will be
impossible to determine when the error is totally removed from the loop.
To do this you would need to have information about what the correct
state is.

[Gamze Seckin] I agree that measuring corruption is difficult and is a =
best effort method.=20
However it is a useful metric to asses quality of streaming and =
applications who provide this information should be encouraged to send =
it.=20

>
> 3. The rebuffering measurements makes more sense. However there are at
> least one client that does not really perform rebuffering, namely
> quicktime. Instead is assumes that if the transmitted bit-rate is
> acceptable any short term effects will go away and the client will =
later
>   have a buffer fullness given by the initial buffering delay. The
> interesting thing about this measurement is that in 3GPP they already
> have a way of letting the server measure when rebuffering takes place,
> namely the OBSN reporting specified in TS 26.234 REL 6 (only =
preliminary
> versions available).
>
> [g] yes, and I believe those statistics are useful, however they are
 > tight with rate adaptation (only turned on when rate-adaptation
 > is enabled) and tightly coupled with other rtsp messages.

Yes, they however is another way of measuring this. I think the
rebuffering measure is possibly more useful then the corruption
measurement.
=20
[Gamze Seckin]  i think we agree on the usefulness of measuring =
rebuffering.


> 6. Section 1. Last sentence: "As all the other report blocks
> defined in [1], the streaming report blocks are applicable to
> one-to-one or one-to-many applications that generate
> RTCP SR or RR reports." This sentence is not really correct. The use =
of
> the RTCP XR reports is possible in any application sending RTCP.
>
> [g] himm,i am confused i thought that the first sentence covers =
everything - what is left out?
> So [1] needs to be updated too?

What I object against is the distinction around RTCP SR or RR reports
and the use of generate. I would rather formulate it as:

As all the other report blocks defined in [1], the streaming report
blocks are applicable to one-to-one or one-to-many applications that
send RTCP in the desired direction.

[Gamze Seckin] thanks - agreed.=20

> 8. Section 2, Figure 1. This figure shows this report formats =
potential
> need to have reporting from all layers on the receiving side. This =
might
> be a problem for certain application architectures. I am not saying =
that
> it can't be used, its only that some applications may have =
difficulties
> to include your reports.
>
> [g] we believe the statistics reported should be extensible also =
should be allowed to be ignored.
> So if an end side cannot support it should put simply a zero - what do =
you think?

I think one need to consider the meaningfulness of allowing only some of
the measurement to be given.
[Gamze Seckin] some metrics can be still useful without the complete =
set. I believe this is an application dependent issue.=20

> 10. Section 3.3, PAge 5, first bullet after figure 2:
> "- at the AVP module: Packet drop and packet errors can sometimes be
> detected by incorrect or missing sequence number or checksum. " Do you
> really need to consider bit-errors, the only allowed possibility for
> bit-errors in RTP is if partial checksums are used. Bit-errors =
surviving
> to the transport layer is the meaning to be caught by the UDP, etc.,
> checksum.
>
> [g] there could be cases where bit errors are propagated (e.g. udp =
lite).

I think that one should be clearer about that this then. However the
usage of UDP-Lite is still under a lot of discussion within the IETF.
There are some very though problems for IP to accept bit-errors
partially. If one are using another transport of RTP then it might be
more common. I would still like you to be more clear that this might be
the case if one uses such mechanisms as UDP-Lite and payload formats +
codecs that accept it.
[Gamze Seckin] noted.

thanks,=20
g=20

Cheers

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com



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<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D898001623-05112003>Hi,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D898001623-05112003>please=20
see comments inline.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D898001623-05112003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D898001623-05112003>thanks</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D898001623-05112003>g</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT><BR><FONT =
size=3D2>-----Original=20
Message-----<BR>From: Magnus Westerlund [</FONT><A=20
href=3D"mailto:magnus.westerlund@ericsson.com"><FONT=20
size=3D2>mailto:magnus.westerlund@ericsson.com</FONT></A><FONT =
size=3D2>]<BR>Sent:=20
Monday, November 03, 2003 3:55 AM<BR>To: Gamze Seckin<BR>Cc: Alan Tseng; =
IETF=20
AVT WG<BR>Subject: Re: Comments on=20
draft-tseng-rtcp-streaming-extns-00.txt<BR><BR><BR>Hi =
Gamze,<BR><BR>Comments=20
inline:<BR><BR>Gamze Seckin wrote:<BR><BR>&gt;<BR>&gt; 2. Regarding the=20
corruption statistics. For media that uses codec that<BR>&gt; have =
prediction=20
any error normally remains for a rather long time.<BR>&gt; However it is =
very=20
difficult to determine when the effects has been<BR>&gt; removed. =
Therefore this=20
measurement is hard to derive on the receiver<BR>&gt; side. Combined =
with the=20
definition it only provides a good, possibly bad<BR>&gt; indication. =
Also=20
looking at the reasons for corruption, the only real<BR>&gt; reason is =
lost or=20
late packets. A good feeling for the quality can be<BR>&gt; determined =
by the=20
server simply by employing packet loss vector<BR>&gt; reporting and =
knowing when=20
the buffer is empty.<BR>&gt;<BR>&gt; [g]packet loss is definitely a way =
of=20
estimating corruption, but this<BR>&nbsp;&gt; gives an algorithm in the =
decoder=20
(or pre-decoder)a chance to provide<BR>&nbsp;&gt; better =
statistics.<BR><BR>Yes,=20
it might be possible for such an algorithm to do this. However=20
the<BR>measurement will not provide that much information. I think=20
you<BR>basically can only say this is good enough or not, there will not =

exist<BR>any possibility to say how bad things actually are. Otherwise =
you=20
will<BR>need to clearly specify how this measurement algorithm shall=20
work.<BR><BR>However one concern here is that the corruption measurement =
will=20
be<BR>quite complex and require rather much memory to perform properly. =
In=20
a<BR>video decoder it will require the measurement tool to track how=20
the<BR>error propagates over the screen. For some audio codecs it will=20
be<BR>impossible to determine when the error is totally removed from the =

loop.<BR>To do this you would need to have information about what the=20
correct<BR>state is.<BR><BR><SPAN class=3D898001623-05112003><FONT =
face=3DArial=20
color=3D#0000ff>[Gamze Seckin]&nbsp;I agree that measuring corruption is =
difficult=20
and is a best effort method. </FONT></SPAN></FONT></DIV>
<DIV><FONT size=3D2><SPAN class=3D898001623-05112003><FONT face=3DArial=20
color=3D#0000ff>However it is a useful metric to asses quality of =
streaming and=20
applications who provide this information should be encouraged to send =
it.=20
</FONT></SPAN></FONT></DIV><FONT face=3DArial =
color=3D#0000ff></FONT><FONT=20
face=3DArial color=3D#0000ff></FONT><FONT face=3DArial =
color=3D#0000ff></FONT><FONT=20
face=3DArial color=3D#0000ff></FONT><FONT face=3DArial =
color=3D#0000ff></FONT>
<DIV><BR><FONT size=3D2>&gt;<BR>&gt; 3. The rebuffering measurements =
makes more=20
sense. However there are at<BR>&gt; least one client that does not =
really=20
perform rebuffering, namely<BR>&gt; quicktime. Instead is assumes that =
if the=20
transmitted bit-rate is<BR>&gt; acceptable any short term effects will =
go away=20
and the client will later<BR>&gt;&nbsp;&nbsp; have a buffer fullness =
given by=20
the initial buffering delay. The<BR>&gt; interesting thing about this=20
measurement is that in 3GPP they already<BR>&gt; have a way of letting =
the=20
server measure when rebuffering takes place,<BR>&gt; namely the OBSN =
reporting=20
specified in TS 26.234 REL 6 (only preliminary<BR>&gt; versions=20
available).<BR>&gt;<BR>&gt; [g] yes, and I believe those statistics are =
useful,=20
however they are<BR>&nbsp;&gt; tight with rate adaptation (only turned =
on when=20
rate-adaptation<BR>&nbsp;&gt; is enabled) and tightly coupled with other =
rtsp=20
messages.<BR><BR>Yes, they however is another way of measuring this. I =
think=20
the<BR>rebuffering measure is possibly more useful then the=20
corruption<BR>measurement.<BR><SPAN class=3D898001623-05112003><FONT =
face=3DArial=20
color=3D#0000ff>&nbsp;</FONT></SPAN><BR><SPAN =
class=3D898001623-05112003><FONT=20
face=3DArial color=3D#0000ff>[Gamze Seckin]&nbsp;&nbsp;i think we agree =
on the=20
usefulness of measuring rebuffering.</FONT></SPAN><BR><BR><BR>&gt; 6. =
Section 1.=20
Last sentence: "As all the other report blocks<BR>&gt; defined in [1], =
the=20
streaming report blocks are applicable to<BR>&gt; one-to-one or =
one-to-many=20
applications that generate<BR>&gt; RTCP SR or RR reports." This sentence =
is not=20
really correct. The use of<BR>&gt; the RTCP XR reports is possible in =
any=20
application sending RTCP.<BR>&gt;<BR>&gt; [g] himm,i am confused i =
thought that=20
the first sentence covers everything - what is left out?<BR>&gt; So [1] =
needs to=20
be updated too?<BR><BR>What I object against is the distinction around =
RTCP SR=20
or RR reports<BR>and the use of generate. I would rather formulate it=20
as:<BR><BR>As all the other report blocks defined in [1], the streaming=20
report<BR>blocks are applicable to one-to-one or one-to-many =
applications=20
that<BR>send RTCP in the desired direction.<BR><BR><SPAN=20
class=3D898001623-05112003><FONT face=3DArial color=3D#0000ff>[Gamze=20
Seckin]&nbsp;thanks - agreed.&nbsp;</FONT></SPAN><BR><BR>&gt; 8. Section =
2,=20
Figure 1. This figure shows this report formats potential<BR>&gt; need =
to have=20
reporting from all layers on the receiving side. This might<BR>&gt; be a =
problem=20
for certain application architectures. I am not saying that<BR>&gt; it =
can't be=20
used, its only that some applications may have difficulties<BR>&gt; to =
include=20
your reports.<BR>&gt;<BR>&gt; [g] we believe the statistics reported =
should be=20
extensible also should be allowed to be ignored.<BR>&gt; So if an end =
side=20
cannot support it should put simply a zero - what do you think?<BR><BR>I =
think=20
one need to consider the meaningfulness of allowing only some of<BR>the=20
measurement to be given.<BR><SPAN class=3D898001623-05112003><FONT =
face=3DArial=20
color=3D#0000ff>[Gamze Seckin]&nbsp;some metrics can be still useful =
without the=20
complete set. I believe this&nbsp;is an application=20
dependent&nbsp;issue.&nbsp;</FONT></SPAN><BR><BR>&gt; 10. Section 3.3, =
PAge 5,=20
first bullet after figure 2:<BR>&gt; "- at the AVP module: Packet drop =
and=20
packet errors can sometimes be<BR>&gt; detected by incorrect or missing =
sequence=20
number or checksum. " Do you<BR>&gt; really need to consider bit-errors, =
the=20
only allowed possibility for<BR>&gt; bit-errors in RTP is if partial =
checksums=20
are used. Bit-errors surviving<BR>&gt; to the transport layer is the =
meaning to=20
be caught by the UDP, etc.,<BR>&gt; checksum.<BR>&gt;<BR>&gt; [g] there =
could be=20
cases where bit errors are propagated (e.g. udp lite).<BR><BR>I think =
that one=20
should be clearer about that this then. However the<BR>usage of UDP-Lite =
is=20
still under a lot of discussion within the IETF.<BR>There are some very =
though=20
problems for IP to accept bit-errors<BR>partially. If one are using =
another=20
transport of RTP then it might be<BR>more common. I would still like you =
to be=20
more clear that this might be<BR>the case if one uses such mechanisms as =

UDP-Lite and payload formats +<BR>codecs that accept it.<BR><SPAN=20
class=3D898001623-05112003><FONT face=3DArial color=3D#0000ff>[Gamze=20
Seckin]&nbsp;noted.</FONT></SPAN></FONT></DIV>
<DIV><SPAN class=3D898001623-05112003></SPAN><FONT face=3DArial =
color=3D#0000ff=20
size=3D2></FONT><BR><FONT size=3D2><SPAN =
class=3D898001623-05112003><FONT face=3DArial=20
color=3D#0000ff>thanks,&nbsp;</FONT></SPAN><BR><SPAN=20
class=3D898001623-05112003><FONT face=3DArial=20
color=3D#0000ff>g&nbsp;</FONT></SPAN><BR><BR>Cheers<BR><BR>Magnus=20
Westerlund<BR><BR>Multimedia Technologies, Ericsson Research=20
EAB/TVA/A<BR>------------------------------------------------------------=
----------<BR>Ericsson=20
AB&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;=20
| Phone +46 8 4048287<BR>Torshamsgatan=20
23&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | =
Fax&nbsp;&nbsp;=20
+46 8 7575550<BR>S-164 80 Stockholm, Sweden | mailto:=20
magnus.westerlund@ericsson.com<BR><BR></FONT></DIV></BODY></HTML>

------_=_NextPart_001_01C3A3F3.ECA9E64C--

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: RE: [AVT] RE: Comments on draft-tseng-rtcp-streaming-extns-00.txt
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From: "Gamze Seckin" <gamze@vidiator.com>
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Hi Magnus,

about your comment
"7. Section 2, You talk about AVP, but this extension is not dependent =
on=20
any special profile and seems to be compatible with any of the four=20
available or under development profiles that exist."

Do you also mean AVPF? Since I am confused about the relationship =
between AVPF and RFC3611.

thx
g
-----Original Message-----
From: Gamze Seckin=20
Sent: Friday, October 31, 2003 11:08 AM
To: Magnus Westerlund; Alan Tseng
Cc: IETF AVT WG
Subject: [AVT] RE: Comments on draft-tseng-rtcp-streaming-extns-00.txt



Hi Magnus,=20
thanks for the comments.
See my response inline.

best regards
gamze
-----Original Message-----
From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
Sent: Wednesday, October 29, 2003 8:17 AM
To: Alan Tseng; Gamze Seckin
Cc: IETF AVT WG
Subject: Comments on draft-tseng-rtcp-streaming-extns-00.txt


Hi,

Here is my comments on your draft:

1. First I think the draft isn't very clear on why any metrics are=20
needed. In the 3GPP document it only states: "The objective of the=20
streaming service quality metrics is to offer to the service providers=20
means to evaluate the end user experience.". This doesn't make it very=20
clear what the real needs are. I think further developing of this is=20
needed.

[g]you are right, the draft does not talk about that clearly. We should =
add that.

2. Regarding the corruption statistics. For media that uses codec that=20
have prediction any error normally remains for a rather long time.=20
However it is very difficult to determine when the effects has been=20
removed. Therefore this measurement is hard to derive on the receiver=20
side. Combined with the definition it only provides a good, possibly bad =

indication. Also looking at the reasons for corruption, the only real=20
reason is lost or late packets. A good feeling for the quality can be=20
determined by the server simply by employing packet loss vector=20
reporting and knowing when the buffer is empty.

[g]packet loss is definitely a way of estimating corruption, but this =
gives an algorithm in the decoder (or pre-decoder)a chance to provide =
better statistics.

3. The rebuffering measurements makes more sense. However there are at=20
least one client that does not really perform rebuffering, namely=20
quicktime. Instead is assumes that if the transmitted bit-rate is=20
acceptable any short term effects will go away and the client will later =

  have a buffer fullness given by the initial buffering delay. The=20
interesting thing about this measurement is that in 3GPP they already=20
have a way of letting the server measure when rebuffering takes place,=20
namely the OBSN reporting specified in TS 26.234 REL 6 (only preliminary =

versions available).

[g] yes, and I believe those statistics are useful, however they are =
tight with rate adaptation (only turned on when rate-adaptation is =
enabled) and tightly coupled with other rtsp messages.

4. As state privately before this drafts need to be checked against the=20
ID nits: http://www.ietf.org/ID-nits.html

[g] yes, we should definitely do that thanks.

5. Section 1. The introduction is basically the same as the=20
introduction. I would recommend you to write a better more informative=20
introduction.

[g] agreed.

6. Section 1. Last sentence: "As all the other report blocks
defined in [1], the streaming report blocks are applicable to
one-to-one or one-to-many applications that generate
RTCP SR or RR reports." This sentence is not really correct. The use of=20
the RTCP XR reports is possible in any application sending RTCP.

[g] himm,i am confused i thought that the first sentence covers =
everything - what is left out?=20
So [1] needs to be updated too?

7. Section 2, You talk about AVP, but this extension is not dependent on =

any special profile and seems to be compatible with any of the four=20
available or under development profiles that exist.

[g] agreed.

8. Section 2, Figure 1. This figure shows this report formats potential=20
need to have reporting from all layers on the receiving side. This might =

be a problem for certain application architectures. I am not saying that =

it can't be used, its only that some applications may have difficulties=20
to include your reports.

[g] we believe the statistics reported should be extensible also should =
be allowed to be ignored.=20
So if an end side cannot support it should put simply a zero - what do =
you think?

9. Section 3.1, first paragraph: "All timestamps reported by the XR=20
packet SHALL be RTP timestamps. All durations SHALL be in units of the=20
RTP timebase." This definition is not clear, because in applications=20
allowing for switching between payload types the rate can change. I=20
think that the intention of using the RTP timestamp value for indication =

over what interval which the report covers is a good one. However some=20
more consideration over formulation must be done.

[g] valid point - I guess we need to think more about the corner cases.

10. Section 3.3, PAge 5, first bullet after figure 2:
"- at the AVP module: Packet drop and packet errors can sometimes be
detected by incorrect or missing sequence number or checksum. " Do you=20
really need to consider bit-errors, the only allowed possibility for=20
bit-errors in RTP is if partial checksums are used. Bit-errors surviving =

to the transport layer is the meaning to be caught by the UDP, etc.,=20
checksum.

[g] there could be cases where bit errors are propagated (e.g. udp =
lite).

11. Section 6. The streaming metrics should probably have a value to=20
combine with the parameter to indicate how often the report block should =

be included. This seem to be a block that is not needed very often.

[g] valid point, note taken.

12. Section 8. The security section does not look sufficient. I guess=20
that the consideration for this is rather similar to the VoIP block.=20
However I think there should be made some effort to indicate the main=20
security concerns also in this document.

[g] agreed.

13. Section 12. The references need to be divided into Normative and=20
Informative ones.

[g] agreed.

Cheers

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Jean-Francois,

It has been a long time!  Hope you're doing well...

Comments below:

> --- editorial comment
> In section 3, secure RTP should be named or mentioned as one of the
> advantages of going to t38/rtp for its overall capabilities. The concern
> is that the paragraph in section 3 only refers to confidentiality of fax
> streams (encryption) while I think that authentication is going to be as
> important.

I purposely did not address SRTP, because this brings in a different RTP
profile (SAVP) and I've not fully studied what (if anything) needs to be
done to use that profile.  Further, SRTP is still in draft form.  I really
don't want to have interdependencies.  If SRTP can be used without any
textual changes to this document, then is begs the question as to whether
anything needs to be said.  If textual changes are required here OR to T.38,
then it strongly suggests we need to address it separately.  Advise welcome,
but please keep in mind that Q11/16 would like to have this MIME
registration before the Jan 2004 meeting of SG16.

> Also consider change "an RTP" to "a RTP"

This is the most interesting request I've seen.  Doing a Google search, I
see 18K pages with "an RTP" and 13K pages with "a RTP".  Which is correct?
It looks right to see "a RTP", but it just doesn't "sound" right.  Spelling
out RTP, I'd agree with "a Real..."

> --
> A number of issues were raised with respect to the usage of UDPTL for
>    the long-term, though.  Specifically, there were concerns over the
>    fact that UDPTL does not provide the same kind of statistics
>    reporting as RTCP.  Further, there are no procedures in place for
>    encrypting the UDPTL stream.  While the latter could be addressed in
>    UDPTL, doing so would require a lot of effort and would largely be a
>    duplication of the security work already on-going within the IETF.
> --

What was the specifc change you want made above?  Just calling out the "an
RTP"?

>
> --- technical comment
> The section 6 (security considerations) is rather silent on the security
> threats.
> Looking at the IFP packet format, I would argue that those packets do
> expose serious threats to the communication that you don't see in audio
> voice. If you insert a bad RTP packet voice payload, oh well, you may
> hear some garbage or clipping. With rtp/t38 fax, bad packets may cause
> real harm:
> + denial of service attacks can be mounted by inserting RTP/t38 packets
> with hdlc-sig-end data field can simply terminate the fax control
> channel. Same with HDLC-FCS-BAD-Sig-End. And one can easily create those
> packets.
> + theft of terminal identity? In some countries, faxes have more legal
> value that emails and the access to the rtp packet content could change
> the Called Subscriber ID frame (phone numbers), etc.  As you well know,
> this info is usually not coded in RTP but in signaling protocols.
>
> I'm sure the Study Group has thought of more. Could some examples of
> security threats be given in the security consideration section and
> shouldn't this section again reference SRTP?

These are both valid points.  I fully support the addition of some text that
highlights this fact.  I would not want to call these out explicitly,
necessarily, as there may be other places (e.g., modification or injection
of NSF data, etc.), so I think it's reasonable to speak to this generically.

Would you like to suggest some text?  I'd be happy to incorporate it.

Paul


> --
>    The IFP packets carried via RTP do not carry any particular security
>    risks in and of themselves.  However, as with other audio packets,
>    there is the potential for unauthorized eavesdropping or manipulation
>    of facsimile transmissions if the media stream is not properly
>    secured.
>
>    The security considerations discussed in the RTP specification [3]
>    and any applicable RTP profile (for example, [8]) are also
>    applicable.
> --
>
>
>
>
> > -----Original Message-----
> > From: Paul E. Jones [mailto:paulej@packetizer.com]
> > Sent: Friday, October 31, 2003 2:01 PM
> > To: avt@ietf.org
> > Subject: [AVT] Moving draft-jones-avt-audio-t38-01.txt forward
> >
> >
> > Folks,
> >
> > I would like to seek agreement from the AVT group to move the
> > document audio/t38 document forward.  As I have mentioned
> > previously, the ITU would like to incorporate this in T.38 in
> > an amendment to be approved in January 2004.
> >
> > The motivation for transporting T.38 via RTP is probably
> > obvious to most. In fact, I recall several saying that that
> > is the way it should have been done from the outset.  Most
> > certainly, this will prove to be a good step in leveraging
> > SRTP for fax to address security needs.
> >
> > The document is relatively short, just seeking the MIME
> > registration.  The technical details will be specified in the
> > ITU-T Recommendation T.38.
> >
> > Can I get agreement to make this an official working group
> > item?  Can we try to get this approved by the January 2004
> > ITU-T SG16 meeting?
> >
> > Thanks,
> > Paul
> >
> >
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www1.ietf.org/mailman/listinfo/avt
> >
>


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Thu, 6 Nov 2003 00:02:53 +0000
Subject: Re: [AVT] I-D ACTION:draft-ietf-avt-rtp-ilbc-03.txt
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From: Colin Perkins <csp@csperkins.org>
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On Tuesday, Oct 28, 2003, at 20:23 Europe/London, 
Internet-Drafts@ietf.org wrote:
> 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 iLBC Speech
> 	Author(s)	: A. Duric, S. Andersen
> 	Filename	: draft-ietf-avt-rtp-ilbc-03.txt
> 	Pages		: 12
> 	Date		: 2003-10-28
> 	
> This document describes the RTP payload format for the internet Low
> Bit Rate Coder (iLBC) Speech [1] developed by Global IP Sound
> (GIPS). Also, within the document there are included necessary
> details for the use of iLBC with MIME and SDP.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-ilbc-03.txt

I have a few minor comments - almost all editorial - on this version:

- Note that the codec has changed again? One extra bit for each frame 
size,
   replacing a previously unused but with an empty frame indicator. Is 
the
   bitstream now stable?
- The abstract cannot contain citations
- Section 2: Please remove the word "freeware" from the codec 
description.
   I expect this falls foul of RFC 2026 section 10.3.2 (b), and it will 
be
   easier if all the IPR discussion is left to the codec draft.
- Section 2: when discussing how the codec bitstream is packed into an
   octet sequence, why SHOULD not MUST?
- Section 3: spelling mistake: "bellow" -> "below"
- Section 3: "No addition header specific to this payload format" -> "No
   additional header..."
- Section 3: The packet format diagram needs to be a numbered figure, 
with
   caption.
- As mentioned previously, the comment about CableLabs intending to 
adopt
   iLBC as their standard is not appropriate in the payload format. It's 
not
   appropriate for us to publish a standard saying what another 
standards body
   may do in future.
- A few formatting nits:
    Bad chars at 356
    Bad chars at 357
    Bad chars at 477
    Bad chars at 478
    Bad chars at 529
    draft-ietf-avt-rtp-ilbc-03.txt: 5 lines containing non-US-ASCII 
characters
         - think these are all Microsoft "smart" quotes.
    draft-ietf-avt-rtp-ilbc-03.txt: 12 formfeeds not on a line by 
themselves
- Shouldn't the mode parameter allow "30" as a legal value? Doesn't 
hurt to
   be explicit... Also missing a close ")" in this paragraph too.

Cheers,
-- 
Colin Perkins
http://csperkins.org/


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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On Thursday, Nov 6, 2003, at 00:02 Europe/London, Paul E. Jones wrote:
...
>> Also consider change "an RTP" to "a RTP"
>
> This is the most interesting request I've seen.  Doing a Google 
> search, I
> see 18K pages with "an RTP" and 13K pages with "a RTP".  Which is 
> correct?
> It looks right to see "a RTP", but it just doesn't "sound" right.  
> Spelling
> out RTP, I'd agree with "a Real..."

http://www.google.com/search?q=%22a+or+an%22

Cheers,
Colin


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Thread-Topic: Comments on draft-tseng-rtcp-streaming-extns-00.txt
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From: "Gamze Seckin" <gamze@vidiator.com>
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Hi,
During IETF 58 in Minneapolis we would like to discuss the following =
issues wrt
draft-tseng-rtcp-streaming-extns-00.txt:

- review objectives and motivations for streaming rtcp reports
- review parameter types and packet format
- we would like to also get comments on issues such as:
	* RTP timestamp usage
	* relationship to AVPF
	* reporting frequency
	* security concerns
=09
your comments are appreciated,

best regards,
g

____________________________
Gamze Seckin, Ph.D.
Vidiator Technology (US) Inc.

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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To: "Colin Perkins" <csp@csperkins.org>
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Based on what I could find, the use of "an" or "a" is based on sound, not
spelling.  Thus, "an RTP" is right... but who can trust these documents on
the web? ;-)

pj

----- Original Message ----- 
From: "Colin Perkins" <csp@csperkins.org>
To: "Paul E. Jones" <paulej@packetizer.com>
Cc: "Jean-Francois Mule" <jf.mule@cablelabs.com>; <avt@ietf.org>
Sent: Wednesday, November 05, 2003 7:26 PM
Subject: Re: [AVT] Moving draft-jones-avt-audio-t38-01.txt forward


> On Thursday, Nov 6, 2003, at 00:02 Europe/London, Paul E. Jones wrote:
> ...
> >> Also consider change "an RTP" to "a RTP"
> >
> > This is the most interesting request I've seen.  Doing a Google
> > search, I
> > see 18K pages with "an RTP" and 13K pages with "a RTP".  Which is
> > correct?
> > It looks right to see "a RTP", but it just doesn't "sound" right.
> > Spelling
> > out RTP, I'd agree with "a Real..."
>
> http://www.google.com/search?q=%22a+or+an%22
>
> Cheers,
> Colin
>


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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To: Gamze Seckin <gamze@vidiator.com>
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Subject: Re: [AVT] RE: Comments on draft-tseng-rtcp-streaming-extns-00.txt
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Hi Gamze,

To my knowledge there is no profile dependencies in RFC 3611. Thus it 
can be used for any of the four profiles defined or under definition. My 
comment was to point out this fact and that it seems inappropriate to 
talk about any special profile in this context.

cheers

Magnus

Gamze Seckin wrote:

> Hi Magnus,
> 
> about your comment
> "7. Section 2, You talk about AVP, but this extension is not dependent on 
> any special profile and seems to be compatible with any of the four 
> available or under development profiles that exist."
> 
> Do you also mean AVPF? Since I am confused about the relationship between AVPF and RFC3611.
> 
> thx
> g
> -----Original Message-----
> From: Gamze Seckin 
> Sent: Friday, October 31, 2003 11:08 AM
> To: Magnus Westerlund; Alan Tseng
> Cc: IETF AVT WG
> Subject: [AVT] RE: Comments on draft-tseng-rtcp-streaming-extns-00.txt
> 
> 
> 
> Hi Magnus, 
> thanks for the comments.
> See my response inline.
> 
> best regards
> gamze
> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Wednesday, October 29, 2003 8:17 AM
> To: Alan Tseng; Gamze Seckin
> Cc: IETF AVT WG
> Subject: Comments on draft-tseng-rtcp-streaming-extns-00.txt
> 
> 
> Hi,
> 
> Here is my comments on your draft:
> 
> 1. First I think the draft isn't very clear on why any metrics are 
> needed. In the 3GPP document it only states: "The objective of the 
> streaming service quality metrics is to offer to the service providers 
> means to evaluate the end user experience.". This doesn't make it very 
> clear what the real needs are. I think further developing of this is 
> needed.
> 
> [g]you are right, the draft does not talk about that clearly. We should add that.
> 
> 2. Regarding the corruption statistics. For media that uses codec that 
> have prediction any error normally remains for a rather long time. 
> However it is very difficult to determine when the effects has been 
> removed. Therefore this measurement is hard to derive on the receiver 
> side. Combined with the definition it only provides a good, possibly bad 
> indication. Also looking at the reasons for corruption, the only real 
> reason is lost or late packets. A good feeling for the quality can be 
> determined by the server simply by employing packet loss vector 
> reporting and knowing when the buffer is empty.
> 
> [g]packet loss is definitely a way of estimating corruption, but this gives an algorithm in the decoder (or pre-decoder)a chance to provide better statistics.
> 
> 3. The rebuffering measurements makes more sense. However there are at 
> least one client that does not really perform rebuffering, namely 
> quicktime. Instead is assumes that if the transmitted bit-rate is 
> acceptable any short term effects will go away and the client will later 
>   have a buffer fullness given by the initial buffering delay. The 
> interesting thing about this measurement is that in 3GPP they already 
> have a way of letting the server measure when rebuffering takes place, 
> namely the OBSN reporting specified in TS 26.234 REL 6 (only preliminary 
> versions available).
> 
> [g] yes, and I believe those statistics are useful, however they are tight with rate adaptation (only turned on when rate-adaptation is enabled) and tightly coupled with other rtsp messages.
> 
> 4. As state privately before this drafts need to be checked against the 
> ID nits: http://www.ietf.org/ID-nits.html
> 
> [g] yes, we should definitely do that thanks.
> 
> 5. Section 1. The introduction is basically the same as the 
> introduction. I would recommend you to write a better more informative 
> introduction.
> 
> [g] agreed.
> 
> 6. Section 1. Last sentence: "As all the other report blocks
> defined in [1], the streaming report blocks are applicable to
> one-to-one or one-to-many applications that generate
> RTCP SR or RR reports." This sentence is not really correct. The use of 
> the RTCP XR reports is possible in any application sending RTCP.
> 
> [g] himm,i am confused i thought that the first sentence covers everything - what is left out? 
> So [1] needs to be updated too?
> 
> 7. Section 2, You talk about AVP, but this extension is not dependent on 
> any special profile and seems to be compatible with any of the four 
> available or under development profiles that exist.
> 
> [g] agreed.
> 
> 8. Section 2, Figure 1. This figure shows this report formats potential 
> need to have reporting from all layers on the receiving side. This might 
> be a problem for certain application architectures. I am not saying that 
> it can't be used, its only that some applications may have difficulties 
> to include your reports.
> 
> [g] we believe the statistics reported should be extensible also should be allowed to be ignored. 
> So if an end side cannot support it should put simply a zero - what do you think?
> 
> 9. Section 3.1, first paragraph: "All timestamps reported by the XR 
> packet SHALL be RTP timestamps. All durations SHALL be in units of the 
> RTP timebase." This definition is not clear, because in applications 
> allowing for switching between payload types the rate can change. I 
> think that the intention of using the RTP timestamp value for indication 
> over what interval which the report covers is a good one. However some 
> more consideration over formulation must be done.
> 
> [g] valid point - I guess we need to think more about the corner cases.
> 
> 10. Section 3.3, PAge 5, first bullet after figure 2:
> "- at the AVP module: Packet drop and packet errors can sometimes be
> detected by incorrect or missing sequence number or checksum. " Do you 
> really need to consider bit-errors, the only allowed possibility for 
> bit-errors in RTP is if partial checksums are used. Bit-errors surviving 
> to the transport layer is the meaning to be caught by the UDP, etc., 
> checksum.
> 
> [g] there could be cases where bit errors are propagated (e.g. udp lite).
> 
> 11. Section 6. The streaming metrics should probably have a value to 
> combine with the parameter to indicate how often the report block should 
> be included. This seem to be a block that is not needed very often.
> 
> [g] valid point, note taken.
> 
> 12. Section 8. The security section does not look sufficient. I guess 
> that the consideration for this is rather similar to the VoIP block. 
> However I think there should be made some effort to indicate the main 
> security concerns also in this document.
> 
> [g] agreed.
> 
> 13. Section 12. The references need to be divided into Normative and 
> Informative ones.
> 
> [g] agreed.
> 
> Cheers
> 
> Magnus Westerlund
> 
> Multimedia Technologies, Ericsson Research EAB/TVA/A
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> 
> 
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
> 

-- 

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Nov  6 11:22:22 2003
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I made a bunch of comments, most of them editorial, on 2833bis last 
August.  You'll be pleased to know that I am passing the remainder of my 
editorial comments to Henning offline.  However, I had a couple of 
technical comments buried in my earlier message that I will repeat here:

Section 3.5, description of duration field:

If a softstate duration expires without refresh, what should the 
receiver do?  I would suggest adding words to the effect that if a soft 
state expires, the receiver SHOULD consider the state to be "unknown".

Flash event: can keep the same coding, but should be moved from Table 2 
to Table 5, along with the note that Flash can happen only when the 
state is "offhook".

Tom Taylor


_______________________________________________
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To: rey@panasonic.de, "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>
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I am not attending MPS so I'm going to give my comments here.  I hope 
they're helpful.


My comments on the draft are in roughly three buckets:  (a) whether 
we should have it at all (b) the overall nature of the draft, (c) 
detailed comments and (d) where to go.

(a) Do we want it at all?

I think yes, for the following reasons.

Ideally, we see an authoring language from w3c, for which there is a 
similarly powerful RTP expression, and that these are supersets of 
the 3GPP format.

But the ideal is unlikely.  For a start, the w3c format is designed 
to be able to express things as diverse as 3GPP and EIA sub-titling. 
It is unlikely that any given delivery system will be able to carry 
everything that can be expressed in the w3c format;  profiles will be 
needed for different delivery environments.  A generic timed text 
format, which would need to cover captioning (both for hearing 
disabled and for language, which are different), displayed timed text 
(e.g. titling), karaoke, ticker tapes, and so on.  Having 
participated in the w3c discussions so far, I am realizing how broad 
this question is.  This is akin to searching for a generic 'video' 
payload format, or, dare I say it, a generic MPEG-4 format!

Secondly, I don't see a reason why text (which is a data type) 
shouldn't be treated like video.  That is, different ways to express 
('compress') the text are conveyed in different 'codecs' and 
different payload formats.  We tried a fairly general-purpose format 
in the MPEG-4 'simple' draft and it was not pleasant.  Targetted 
formats for well-defined coding systems work better.

So, I am not convinced that a targetted 3GPP timed text format is 
'wrong'.  As long as we admit that there may (will) be a family of 
text presentation formats (like audio and video), it is not a 
problem, either.


(b) Overall comments.

The 3GPP timed text spec is conceptually pretty simple;  this payload 
format seems pretty complex.  There are several bit-fields, for 
example.  It uses two payload type indicators.  I think that we can 
use one payload type indicator, and one header which tells us (i) is 
the 'static information' present? (ii) is this a single fragment or a 
set of whole samples? (iii) what are the various offsets etc.

I get the idea of giving the explicit offset to the start of a box, 
and also the need to 'place' a fragment in the string, and so on. 
I'm just aware that it comes across complex as is, and wonder if we 
can't skin it slightly differently -- possibly at the expense of a 
few bytes for unusual cases -- for a simpler spec.

I'm willing to spend time in this direction if there is consensus 
that such an exploration take place.

(c) Details.

I have a number of detailed comments:  things talked about before 
they are defined (e.g. T=1 in 4.1);  what is the SIDX field for? and 
so on.  However, I'd rather see us work out the direction in general 
(a) and the approach (b) before dealing with details.

(d) Where to go.

I think we should proceed with this.  Though it is not on the table 
for release 6 of 3GPP officially, no-one there will object if text 
becomes streamable:  it's a useful good idea.  I don't think the W3C 
work is either going to complete soon, or be a direct translation 
into an RTP format, and that's the only 'bigger' idea on the horizon. 
And I don't think this should be shoe-horned into the MPEG-4 payload 
format either;  it has unique needs (and it ought to be possible to 
make it simpler).

A liaison or communication from the chairs to 3GPP and W3C saying 
what the Minneapolis meeting thinks is the direction would be prudent 
and courteous, however.
-- 
David Singer
Apple Computer/QuickTime

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Nov  6 13:44:21 2003
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Subject: RE: [AVT] Moving draft-jones-avt-audio-t38-01.txt forward
Date: Thu, 6 Nov 2003 11:42:54 -0700
Message-ID: <AEE1FD45F580334296FDA24A1167FFA50E2930@srvxchg.cablelabs.com>
Thread-Topic: [AVT] Moving draft-jones-avt-audio-t38-01.txt forward
Thread-Index: AcOkBD5komhnocTTQKCQtoVJurOxwwAkUBLA
From: "Jean-Francois Mule" <jf.mule@cablelabs.com>
To: "Paul E. Jones" <paulej@packetizer.com>,
        "Colin Perkins" <csp@csperkins.org>
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Yes, you're right, forget my french and my comment...
I have been misled, it is "an RTP".

> The "Oxford Companion to the English Language" says, "...the=20
> use of "a"=20
> or "an" indicates whether a writer is thinking of them as=20
> letters or words".
>=20
> Less ambiguously, "A Modern Guide to English Usage" says "The test is=20
> how people say or read such designations", and it gives the=20
> example of "an MA degree" as being proper.

Thx,
Jean-Fran=E7ois=20
> -----Original Message-----
> From: Paul E. Jones [mailto:paulej@packetizer.com]=20
> Sent: Wednesday, November 05, 2003 6:20 PM
> To: Colin Perkins
> Cc: Jean-Francois Mule; avt@ietf.org
> Subject: Re: [AVT] Moving draft-jones-avt-audio-t38-01.txt forward
>=20
>=20
> Based on what I could find, the use of "an" or "a" is based=20
> on sound, not spelling.  Thus, "an RTP" is right... but who=20
> can trust these documents on the web? ;-)
>=20
> pj
>=20
> ----- Original Message -----=20
> From: "Colin Perkins" <csp@csperkins.org>
> To: "Paul E. Jones" <paulej@packetizer.com>
> Cc: "Jean-Francois Mule" <jf.mule@cablelabs.com>; <avt@ietf.org>
> Sent: Wednesday, November 05, 2003 7:26 PM
> Subject: Re: [AVT] Moving draft-jones-avt-audio-t38-01.txt forward
>=20
>=20
> > On Thursday, Nov 6, 2003, at 00:02 Europe/London, Paul E.=20
> Jones wrote:=20
> > ...
> > >> Also consider change "an RTP" to "a RTP"
> > >
> > > This is the most interesting request I've seen.  Doing a Google=20
> > > search, I see 18K pages with "an RTP" and 13K pages with=20
> "a RTP". =20
> > > Which is correct?
> > > It looks right to see "a RTP", but it just doesn't "sound" right.
> > > Spelling
> > > out RTP, I'd agree with "a Real..."
> >
> > http://www.google.com/search?q=3D%22a+or+an%22
> >
> > Cheers,
> > Colin
> >
>=20
>=20

_______________________________________________
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https://www1.ietf.org/mailman/listinfo/avt



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Subject: [AVT] Summary of discussion items for draft-ahmadi-avt-rtp-vmr-wb-00.txt
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All,

During the 58th IETF meetings in Minneapolis we would like to discuss =
the following issues wrt draft-ahmadi-avt-rtp-vmr-wb-00.txt:

-Background on VMR-WB
-Similarities and differences with RFC3267
-Highlights of the proposed draft

Given the large amount of similarities (reuse) between this internet =
draft and RFC3267, I hope that AVT WG can focus mostly on the new items =
and hopefully accelerate the development of a RFC for VMR-WB.

Therefore, I would like to ask AVT WG to adopt this draft as WG work =
item (document) to help faster completion of a RFC for VMR-WB.

Your help and feedback/comments are greatly appreciated.

Regards

-Sassan Ahmadi, Ph.D.
 VMR-WB Project Manager
 Nokia Inc. USA=20

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From exim@www1.ietf.org  Thu Nov  6 13:53:23 2003
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Subject: RE: [AVT] Moving draft-jones-avt-audio-t38-01.txt forward
Date: Thu, 6 Nov 2003 11:51:29 -0700
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Thread-Topic: [AVT] Moving draft-jones-avt-audio-t38-01.txt forward
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From: "Jean-Francois Mule" <jf.mule@cablelabs.com>
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Paul wrote:
> I purposely did not address SRTP, because this brings in a=20
> different RTP profile (SAVP) and I've not fully studied what=20
> (if anything) needs to be done to use that profile.  Further,=20
> SRTP is still in draft form.  I really don't want to have=20
> interdependencies.
I'll let other folks comment on whether it is appropriate to not address =
the draft's connections with SRTP.

> > --
> > A number of issues were raised with respect to the usage of=20
> UDPTL for
> >    the long-term, though.  Specifically, there were=20
> concerns over the
> >    fact that UDPTL does not provide the same kind of statistics
> >    reporting as RTCP.  Further, there are no procedures in place for
> >    encrypting the UDPTL stream.  While the latter could be=20
> addressed in
> >    UDPTL, doing so would require a lot of effort and would=20
> largely be a
> >    duplication of the security work already on-going within=20
> the IETF.
> > --
>=20
> What was the specifc change you want made above?  Just=20
> calling out the "an RTP"?

How about something like:
s/encrypting the UDPTL stream/authenticating or encrypting the UDPTL =
stream


> > --- technical comment
> > The section 6 (security considerations) is rather silent on the=20
> > security threats. Looking at the IFP packet format, I would=20
> argue that=20
> > those packets do expose serious threats to the=20
> communication that you=20
> > don't see in audio voice. If you insert a bad RTP packet voice=20
> > payload, oh well, you may hear some garbage or clipping.=20
> With rtp/t38=20
> > fax, bad packets may cause real harm:
> > + denial of service attacks can be mounted by inserting RTP/t38=20
> > + packets
> > with hdlc-sig-end data field can simply terminate the fax control=20
> > channel. Same with HDLC-FCS-BAD-Sig-End. And one can easily create=20
> > those packets.
> > + theft of terminal identity? In some countries, faxes have=20
> more legal
> > value that emails and the access to the rtp packet content could=20
> > change the Called Subscriber ID frame (phone numbers), etc.  As you=20
> > well know, this info is usually not coded in RTP but in signaling=20
> > protocols.
> >
> > I'm sure the Study Group has thought of more. Could some=20
> examples of=20
> > security threats be given in the security consideration section and=20
> > shouldn't this section again reference SRTP?
>=20
> These are both valid points.  I fully support the addition of=20
> some text that highlights this fact.  I would not want to=20
> call these out explicitly, necessarily, as there may be other=20
> places (e.g., modification or injection of NSF data, etc.),=20
> so I think it's reasonable to speak to this generically.
>=20
> Would you like to suggest some text?  I'd be happy to incorporate it.
I would rather let you wordsmith my comments into something general you =
like on this one.=20

Jean-Fran=E7ois=20

_______________________________________________
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https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Nov  6 14:28:21 2003
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Date: Thu, 06 Nov 2003 11:27:08 -0800
To: "Jean-Francois Mule" <jf.mule@cablelabs.com>
From: Mark Baugher <mbaugher@cisco.com>
Subject: RE: [AVT] Moving draft-jones-avt-audio-t38-01.txt forward
Cc: "Paul E. Jones" <paulej@packetizer.com>, <avt@ietf.org>
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 om>
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At 10:51 AM 11/6/2003, Jean-Francois Mule wrote:
>Paul wrote:
> > I purposely did not address SRTP, because this brings in a
> > different RTP profile (SAVP) and I've not fully studied what
> > (if anything) needs to be done to use that profile.  Further,
> > SRTP is still in draft form.  I really don't want to have
> > interdependencies.
>I'll let other folks comment on whether it is appropriate to not address=20
>the draft's connections with SRTP.

I think that there is _an_ SRTP connection (or is it connexion :?) that=20
should be referred to in the security considerations.  The text that is in=
=20
the current draft could be extended slightly to say that t38 is vulnerable=
=20
to attacks that are common on other types of RTP payloads and that the=20
security considerations and mechanisms of RTP and SRTP apply to t38.

Mark


> > > --
> > > A number of issues were raised with respect to the usage of
> > UDPTL for
> > >    the long-term, though.  Specifically, there were
> > concerns over the
> > >    fact that UDPTL does not provide the same kind of statistics
> > >    reporting as RTCP.  Further, there are no procedures in place for
> > >    encrypting the UDPTL stream.  While the latter could be
> > addressed in
> > >    UDPTL, doing so would require a lot of effort and would
> > largely be a
> > >    duplication of the security work already on-going within
> > the IETF.
> > > --
> >
> > What was the specifc change you want made above?  Just
> > calling out the "an RTP"?
>
>How about something like:
>s/encrypting the UDPTL stream/authenticating or encrypting the UDPTL stream
>
>
> > > --- technical comment
> > > The section 6 (security considerations) is rather silent on the
> > > security threats. Looking at the IFP packet format, I would
> > argue that
> > > those packets do expose serious threats to the
> > communication that you
> > > don't see in audio voice. If you insert a bad RTP packet voice
> > > payload, oh well, you may hear some garbage or clipping.
> > With rtp/t38
> > > fax, bad packets may cause real harm:
> > > + denial of service attacks can be mounted by inserting RTP/t38
> > > + packets
> > > with hdlc-sig-end data field can simply terminate the fax control
> > > channel. Same with HDLC-FCS-BAD-Sig-End. And one can easily create
> > > those packets.
> > > + theft of terminal identity? In some countries, faxes have
> > more legal
> > > value that emails and the access to the rtp packet content could
> > > change the Called Subscriber ID frame (phone numbers), etc.  As you
> > > well know, this info is usually not coded in RTP but in signaling
> > > protocols.
> > >
> > > I'm sure the Study Group has thought of more. Could some
> > examples of
> > > security threats be given in the security consideration section and
> > > shouldn't this section again reference SRTP?
> >
> > These are both valid points.  I fully support the addition of
> > some text that highlights this fact.  I would not want to
> > call these out explicitly, necessarily, as there may be other
> > places (e.g., modification or injection of NSF data, etc.),
> > so I think it's reasonable to speak to this generically.
> >
> > Would you like to suggest some text?  I'd be happy to incorporate it.
>I would rather let you wordsmith my comments into something general you=20
>like on this one.
>
>Jean-Fran=E7ois
>
>_______________________________________________
>Audio/Video Transport Working Group
>avt@ietf.org
>https://www1.ietf.org/mailman/listinfo/avt



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Nov  6 16:43:25 2003
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From: "Paul E. Jones" <paulej@packetizer.com>
To: "Jean-Francois Mule" <jf.mule@cablelabs.com>,
        "Mark Baugher" <mbaugher@cisco.com>
Cc: <avt@ietf.org>
References: <AEE1FD45F580334296FDA24A1167FFA50284CB@srvxchg.cablelabs.com> <6.0.0.22.2.20031106112128.03b0d9b0@mira-sjc5-6.cisco.com>
Subject: Re: [AVT] Moving draft-jones-avt-audio-t38-01.txt forward
Date: Thu, 6 Nov 2003 16:40:21 -0500
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Mark,

Your comments are valid.  In fact, it's SRTP that is one driver (but not =
the
only driver) for wanting to move to RTP.  But, I hesitate to mention SRTP
without discussing the use of SRTP.

Is there specific text that you'd like to propose.  If you propose it and
folks agree, I definitely have no objections to incorporating it.

Paul

----- Original Message -----=20
From: "Mark Baugher" <mbaugher@cisco.com>
To: "Jean-Francois Mule" <jf.mule@cablelabs.com>
Cc: "Paul E. Jones" <paulej@packetizer.com>; <avt@ietf.org>
Sent: Thursday, November 06, 2003 2:27 PM
Subject: RE: [AVT] Moving draft-jones-avt-audio-t38-01.txt forward


> At 10:51 AM 11/6/2003, Jean-Francois Mule wrote:
> >Paul wrote:
> > > I purposely did not address SRTP, because this brings in a
> > > different RTP profile (SAVP) and I've not fully studied what
> > > (if anything) needs to be done to use that profile.  Further,
> > > SRTP is still in draft form.  I really don't want to have
> > > interdependencies.
> >I'll let other folks comment on whether it is appropriate to not addre=
ss
> >the draft's connections with SRTP.
>
> I think that there is _an_ SRTP connection (or is it connexion :?) that
> should be referred to in the security considerations.  The text that is=
 in
> the current draft could be extended slightly to say that t38 is vulnera=
ble
> to attacks that are common on other types of RTP payloads and that the
> security considerations and mechanisms of RTP and SRTP apply to t38.
>
> Mark
>
>
> > > > --
> > > > A number of issues were raised with respect to the usage of
> > > UDPTL for
> > > >    the long-term, though.  Specifically, there were
> > > concerns over the
> > > >    fact that UDPTL does not provide the same kind of statistics
> > > >    reporting as RTCP.  Further, there are no procedures in place =
for
> > > >    encrypting the UDPTL stream.  While the latter could be
> > > addressed in
> > > >    UDPTL, doing so would require a lot of effort and would
> > > largely be a
> > > >    duplication of the security work already on-going within
> > > the IETF.
> > > > --
> > >
> > > What was the specifc change you want made above?  Just
> > > calling out the "an RTP"?
> >
> >How about something like:
> >s/encrypting the UDPTL stream/authenticating or encrypting the UDPTL
stream
> >
> >
> > > > --- technical comment
> > > > The section 6 (security considerations) is rather silent on the
> > > > security threats. Looking at the IFP packet format, I would
> > > argue that
> > > > those packets do expose serious threats to the
> > > communication that you
> > > > don't see in audio voice. If you insert a bad RTP packet voice
> > > > payload, oh well, you may hear some garbage or clipping.
> > > With rtp/t38
> > > > fax, bad packets may cause real harm:
> > > > + denial of service attacks can be mounted by inserting RTP/t38
> > > > + packets
> > > > with hdlc-sig-end data field can simply terminate the fax control
> > > > channel. Same with HDLC-FCS-BAD-Sig-End. And one can easily creat=
e
> > > > those packets.
> > > > + theft of terminal identity? In some countries, faxes have
> > > more legal
> > > > value that emails and the access to the rtp packet content could
> > > > change the Called Subscriber ID frame (phone numbers), etc.  As y=
ou
> > > > well know, this info is usually not coded in RTP but in signaling
> > > > protocols.
> > > >
> > > > I'm sure the Study Group has thought of more. Could some
> > > examples of
> > > > security threats be given in the security consideration section a=
nd
> > > > shouldn't this section again reference SRTP?
> > >
> > > These are both valid points.  I fully support the addition of
> > > some text that highlights this fact.  I would not want to
> > > call these out explicitly, necessarily, as there may be other
> > > places (e.g., modification or injection of NSF data, etc.),
> > > so I think it's reasonable to speak to this generically.
> > >
> > > Would you like to suggest some text?  I'd be happy to incorporate i=
t.
> >I would rather let you wordsmith my comments into something general yo=
u
> >like on this one.
> >
> >Jean-Fran=E7ois
> >
> >_______________________________________________
> >Audio/Video Transport Working Group
> >avt@ietf.org
> >https://www1.ietf.org/mailman/listinfo/avt
>
>
>


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Mark Baugher <mbaugher@cisco.com>
Subject: Re: [AVT] Moving draft-jones-avt-audio-t38-01.txt forward
Cc: "Jean-Francois Mule" <jf.mule@cablelabs.com>, <avt@ietf.org>,
        Mats 
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Paul
   I think the approach that you mentioned in the security consideration is=
=20
good since it relies on the security consideration and security mechanisms=
=20
found in RTP and SRTP.  What is needed is to specify what might be specific=
=20
to the fax payload type with respect to SRTP.  I'd say that a short 32-bit=
=20
tag is a SHOULD NOT for fax and not a MAY as it is with other applications=
=20
owing to the fact that the effects of fraud are so much greater form=20
certain fax applications than is generally true for voice.  I'd offer=20
something like this.
"T38 is vulnerable to attacks that are common on other types of RTP=20
payloads.  The security considerations and mechanisms of RTP and SRTP apply=
=20
to T38.  Regarding SRTP configuration, fax payloads SHOULD NOT use an=20
HMAC-SHA1 authentication tag that is shorter than 80 bits."

Mark
At 01:40 PM 11/6/2003, Paul E. Jones wrote:
>Mark,
>
>Your comments are valid.  In fact, it's SRTP that is one driver (but not=
 the
>only driver) for wanting to move to RTP.  But, I hesitate to mention SRTP
>without discussing the use of SRTP.
>
>Is there specific text that you'd like to propose.  If you propose it and
>folks agree, I definitely have no objections to incorporating it.
>
>Paul
>
>----- Original Message -----
>From: "Mark Baugher" <mbaugher@cisco.com>
>To: "Jean-Francois Mule" <jf.mule@cablelabs.com>
>Cc: "Paul E. Jones" <paulej@packetizer.com>; <avt@ietf.org>
>Sent: Thursday, November 06, 2003 2:27 PM
>Subject: RE: [AVT] Moving draft-jones-avt-audio-t38-01.txt forward
>
>
> > At 10:51 AM 11/6/2003, Jean-Francois Mule wrote:
> > >Paul wrote:
> > > > I purposely did not address SRTP, because this brings in a
> > > > different RTP profile (SAVP) and I've not fully studied what
> > > > (if anything) needs to be done to use that profile.  Further,
> > > > SRTP is still in draft form.  I really don't want to have
> > > > interdependencies.
> > >I'll let other folks comment on whether it is appropriate to not=
 address
> > >the draft's connections with SRTP.
> >
> > I think that there is _an_ SRTP connection (or is it connexion :?) that
> > should be referred to in the security considerations.  The text that is=
 in
> > the current draft could be extended slightly to say that t38 is=
 vulnerable
> > to attacks that are common on other types of RTP payloads and that the
> > security considerations and mechanisms of RTP and SRTP apply to t38.
> >
> > Mark
> >
> >
> > > > > --
> > > > > A number of issues were raised with respect to the usage of
> > > > UDPTL for
> > > > >    the long-term, though.  Specifically, there were
> > > > concerns over the
> > > > >    fact that UDPTL does not provide the same kind of statistics
> > > > >    reporting as RTCP.  Further, there are no procedures in place=
 for
> > > > >    encrypting the UDPTL stream.  While the latter could be
> > > > addressed in
> > > > >    UDPTL, doing so would require a lot of effort and would
> > > > largely be a
> > > > >    duplication of the security work already on-going within
> > > > the IETF.
> > > > > --
> > > >
> > > > What was the specifc change you want made above?  Just
> > > > calling out the "an RTP"?
> > >
> > >How about something like:
> > >s/encrypting the UDPTL stream/authenticating or encrypting the UDPTL
>stream
> > >
> > >
> > > > > --- technical comment
> > > > > The section 6 (security considerations) is rather silent on the
> > > > > security threats. Looking at the IFP packet format, I would
> > > > argue that
> > > > > those packets do expose serious threats to the
> > > > communication that you
> > > > > don't see in audio voice. If you insert a bad RTP packet voice
> > > > > payload, oh well, you may hear some garbage or clipping.
> > > > With rtp/t38
> > > > > fax, bad packets may cause real harm:
> > > > > + denial of service attacks can be mounted by inserting RTP/t38
> > > > > + packets
> > > > > with hdlc-sig-end data field can simply terminate the fax control
> > > > > channel. Same with HDLC-FCS-BAD-Sig-End. And one can easily create
> > > > > those packets.
> > > > > + theft of terminal identity? In some countries, faxes have
> > > > more legal
> > > > > value that emails and the access to the rtp packet content could
> > > > > change the Called Subscriber ID frame (phone numbers), etc.  As=
 you
> > > > > well know, this info is usually not coded in RTP but in signaling
> > > > > protocols.
> > > > >
> > > > > I'm sure the Study Group has thought of more. Could some
> > > > examples of
> > > > > security threats be given in the security consideration section=
 and
> > > > > shouldn't this section again reference SRTP?
> > > >
> > > > These are both valid points.  I fully support the addition of
> > > > some text that highlights this fact.  I would not want to
> > > > call these out explicitly, necessarily, as there may be other
> > > > places (e.g., modification or injection of NSF data, etc.),
> > > > so I think it's reasonable to speak to this generically.
> > > >
> > > > Would you like to suggest some text?  I'd be happy to incorporate=
 it.
> > >I would rather let you wordsmith my comments into something general you
> > >like on this one.
> > >
> > >Jean-Fran=E7ois
> > >
> > >_______________________________________________
> > >Audio/Video Transport Working Group
> > >avt@ietf.org
> > >https://www1.ietf.org/mailman/listinfo/avt
> >
> >
> >
>
>
>_______________________________________________
>Audio/Video Transport Working Group
>avt@ietf.org
>https://www1.ietf.org/mailman/listinfo/avt



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: [AVT] RE: Comments on draft-rey-avt-rtp-3gpp-timed-text-01.txt
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Hi Dave,

thanks for your comments. I also think it is better to stay in the
abstract plane for now, and leave the details for later...and I
appreciate very much your readiness to help. Please see inline.

---cut---
>
> (b) Overall comments.
>
> The 3GPP timed text spec is conceptually pretty simple;  this payload
> format seems pretty complex.  There are several bit-fields, for
> example.  It uses two payload type indicators.  I think that we can
> use one payload type indicator, and one header which tells us (i) is
> the 'static information' present? (ii) is this a single fragment or a
> set of whole samples? (iii) what are the various offsets etc.
>
> I get the idea of giving the explicit offset to the start of a box,
> and also the need to 'place' a fragment in the string, and so on.
> I'm just aware that it comes across complex as is, and wonder if we
> can't skin it slightly differently -- possibly at the expense of a
> few bytes for unusual cases -- for a simpler spec.
>
> I'm willing to spend time in this direction if there is consensus
> that such an exploration take place.
>

Yes, the draft is pretty complex and yes we can make it simpler, of
course by having more overhead. The draft has a lot of functionalities,
which would probably (?) be seldom used. The question is where and how
to cut down.

In particular, the thing that 'bothers' me most is the following: the
draft allows for self-contained packets of the two kinds of text sample
fragments: those containing text strings and those that not.  Apart from
the current composition of the headers, 1) do we want to keep that? 2)
or is it enough to guarantee the display of just the text strings
(plain) in the case of packet losses? Is it correct to assume modifier
boxes would very seldom expand over several RTP packets so that the
effort for specific modifier field fragment headers would not be worth?
or is it preferable to stay in the safe side and contemplate those rare
cases also by introducing overhead? Comments are welcome!


> (c) Details.
>
> I have a number of detailed comments:  things talked about before
> they are defined (e.g. T=1 in 4.1);  what is the SIDX field for? and
> so on.  However, I'd rather see us work out the direction in general
> (a) and the approach (b) before dealing with details.
>

I agree, I prefer to comment this later, it might change.

> (d) Where to go.
>
> I think we should proceed with this.  Though it is not on the table
> for release 6 of 3GPP officially, no-one there will object if text
> becomes streamable:  it's a useful good idea.  I don't think the W3C
> work is either going to complete soon, or be a direct translation
> into an RTP format, and that's the only 'bigger' idea on the horizon.
> And I don't think this should be shoe-horned into the MPEG-4 payload
> format either;  it has unique needs (and it ought to be possible to
> make it simpler).
>
> A liaison or communication from the chairs to 3GPP and W3C saying
> what the Minneapolis meeting thinks is the direction would be prudent
> and courteous, however.

Good idea.

cheers,

Jose



> --
> David Singer
> Apple Computer/QuickTime
>


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: [AVT] Summary for draft-ietf-avt-rtp-ilbc-03.txt and draft-ietf-avt-ilbc-codec-03.txt
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Hi,

during the 58th IETF meeting in Minneapolis there will be discussed
following wrt draft-ietf-avt-rtp-ilbc-03.txt and
draft-ietf-avt-rtp-ilbc-03.txt:

# changes since the last version of (draft-ietf-avt-rtp-ilbc-03.txt)
	- input reflecting AVT chairs input on 02 version (clarified a
number of ID nits and descriptions)
	- changes in file format (2 different file types, empty frame
introduced) 

# changes since the last version of (draft-ietf-avt-ilbc-codec-03.txt)
	- input reflecting AVT chairs input on 02 version (clarified a
number of ID nits and descriptions, 2 compiler errors 	in the source
code due to formatting fixed)
	- empty frame introduction (no effect on existing
implementations)

# Update on running implementations

# WGLC (proposal) for both of the drafts

 	

Your feedback/comments are greatly appreciated.

Regards,
Alan



________________________________________________________________________
____________
Cuius rei demonstrationem mirabilem sane detexi hanc marginis exiguitas
non caperet.  



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Section 4:
In the example "CIF=3D4 QCIF=3D3 SQCIF=3D2 XMAX=3D360 YMAX=3D240 =
MPI=3D2", the text says that the MPI=3D2 parameter goes with SQCIF=3D2.  =
How does it know?


Nits - Section 2
ABNF
Size =3D "QCIF" "=3D" "mpi | "CIF" "=3D" mpi
Size =3D "QCIF" "=3D" mpi | "CIF" "=3D" mpi
                  ^
Section 3 ABNF
"CIF4" "=3D" mpi I "CIF16" "=3D" mpi
"CIF4" "=3D" mpi | "CIF16" "=3D" mpi
               ^

Here #term means comma separated list of term.
Here #term means comma separated list of terms.
                                             ^


_______________________________________________
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From: Qiaobing Xie <Qiaobing.Xie@motorola.com>
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Subject: [AVT] Re: Comments on DSR drafts
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Hi, Magnus,

Magnus Westerlund wrote:
...
> I think they from an RTP transport perspective are all the same and
> therefore could be had in one specification.

Ok, I will merge the three into one draft after the IETF.

> Okay, good. There is no DSR frame that can creates unusual high
> processing requirements on receiver side?

To my knowledge the values in a DSR frame (mostly indexes of a small,
fixed size codebook) are not capable of creating this type of threats.

regards,
-Qiaobing

_______________________________________________
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From: Qiaobing Xie <Qiaobing.Xie@motorola.com>
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Subject: [AVT] Discussion points for "RTP payload formats for ETSI DSR front ends"
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Hi, all,

...
> RTP payload formats for ETSI DSR front ends      (Xie)    15
>         draft-ietf-avt-dsr-es202050-01.txt
>         draft-xie-avt-xdsr-es202211-00.txt
>         draft-xie-avt-xadsr-es202212-00.txt

In the above presentation, I will discuss the following:

 - A review of changes made to es202050-01 draft since it was 
   presented in IETF 56th.

 - An introduction to es202211-00 and es202212-00 drafts and their 
   differences from the 2 previous DSR front ends.

 - A proposal to merge the three drafts into one WG item.

 - There are no other open issues.

regards,
-Qiaobing

_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Even, Roni" <roni.even@polycom.co.il>
To: "'Eric Burger'" <eburger@snowshore.com>, avt@ietf.org, mmusic@ietf.org
Date: Sat, 8 Nov 2003 13:16:16 +0200 
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Subject: [AVT] RE: [MMUSIC] Comments on draft-even-avt-h263-h261-options-00
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Eric,
The 2 in SQCIF=2 is the MPI value. The MPI but it self relates to the custom
format x=360, y=240, MPI=2.
Roni

*************************************
Roni Even
VP Product Planning
Polycom Israel

Tel: +972-3-9251200
Cell: +972-55-481099
email:roni.even@polycom.co.il
*******************************************


-----Original Message-----
From: Eric Burger [mailto:eburger@snowshore.com]
Sent: Friday, November 07, 2003 11:20 PM
To: avt@ietf.org; mmusic@ietf.org
Subject: [MMUSIC] Comments on draft-even-avt-h263-h261-options-00


Section 4:
In the example "CIF=4 QCIF=3 SQCIF=2 XMAX=360 YMAX=240 MPI=2", the text says
that the MPI=2 parameter goes with SQCIF=2.  How does it know?


Nits - Section 2
ABNF
Size = "QCIF" "=" "mpi | "CIF" "=" mpi
Size = "QCIF" "=" mpi | "CIF" "=" mpi
                  ^
Section 3 ABNF
"CIF4" "=" mpi I "CIF16" "=" mpi
"CIF4" "=" mpi | "CIF16" "=" mpi
               ^

Here #term means comma separated list of term.
Here #term means comma separated list of terms.
                                             ^


_______________________________________________
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Dear Jose, all,

I will not be in Minneapolis either but would like to provide some input 
on the issue of transport of 3GPP timed text. I know this input is very 
late; my appologies for that.

As you know also MPEG is working on timed text; currently the Committe 
Draft (CD) is ballotted, and MPEG schedules to enter the next phase in 
about one month and to finish the work in July 2004. The end-result will 
be part 17 of MPEG-4, defining MPEG-4 text streams. Through liaisons, MPEG 
tries to harmonize its work with the work that is already carried out in 
3GPP and also with ongoing work in W3C. Most important in this respect is 
that MPEG focusses at maximum commonality with the timed text format 
defined by 3GPP for download. In summary, the current CD of MPEG-4 part 17 
specifies streaming of the 3GPP timed text format by defining a very 
straightforward way to transcode the 3GPP format into the MPEG-4 structure 
of a decoder configuration and text access units. Please find the 
Committee Draft (a public document) attached to this email. So far, 3GPP 
expressed through liaison statements to be very pleased with the work done 
by MPEG.

Once the work is finished, the timed text format can be streamed using 
"RFC#simple", draft-ietf-avt-mpeg4-simple-08.txt, that is scheduled to be 
published as RFC very soon. In summary this approach would work as 
follows:
- media type: "video",
- MIME subtype name: mpeg4-generic,
- streamType (required parameter in RFC#simple): indicates MPEG-4 text 
stream as defined in ISO 14496-17,
- profile-level-id (required parameter in RFC#simple): specifies the 
functional subset as defined in the current 3GPP timed text specification,
- config (required parameter in RFC#simple): carries the decoder 
configuration specified for text streams in ISO 14496-17,
- the RTP payload contains either (a) one fragment of a text access unit 
or (b) one or more complete text access units,
- the RTP payload carries one AU-header for each carried text access unit,
- in the RTP payload the concatenated AU-headers preceed concatenated data 
of text access units, as defined in RFC#simple,
- each AU-header contains (at least) the following information on the 
associated text access unit: (a) the size of the text access unit, (b) the 
time stamp of the text access unit,
- if deemed necessary, the AU-header may also carry the index of a text 
access unit (fragment) and/or an indication whether the associated text 
access unit is a random access point.
In RFC#simple the fields in the AU-header are configurable, but if desired 
one or two new modes for RFC#simple could be defined with fixed length 
fields, for example one using an 8 bit AU-size field (for most common 
applications) and another using 16 bits AU-size field for applications 
using (very) large text access units.
 
The above "RFC#simple approach" has a high level of commonality with 
draft-rey-avt-rtp-3gpp-timed-text-01.txt. There are some small differences 
that can be fixed either way, but roughly the "RFC#simple approach" is a 
simplified version of draft-rey-avt-rtp-3gpp-timed-text-01.txt. In 
particular, the "RFC#simple" approach does not have the advanced error 
resiliency measures that are in draft-rey-avt-rtp-3gpp-timed-text-01.txt. 

In my view not only in IETF, but also in 3GPP and in MPEG a number of 
basic issues should be discussed, such as:
- are the error resiliency features of 
draft-rey-avt-rtp-3gpp-timed-text-01.txt sufficiently valuable to justify 
their complexity (note that a partly corrupted subtitle may not have much 
more value to the user than a missing one);
- is it desirable to define a 3GPP timed text transport solution based on 
draft-rey-avt-rtp-3gpp-timed-text-01.txt next to the one that will be 
available based on RFC#simple;
- is it desirable that draft-rey-avt-rtp-3gpp-timed-text-01.txt specifies 
a mode or an extension of RFC#simple;
- how can a timing conflict (if any) between release 6 of 3GPP and part 17 
of MPEG-4 be resolved (note that such conflict will only exist if 3GPP 
adopts streaming of timed text as a work item and if 3GPP decides that 
this functionality has to be in release 6).

I hope you will have a fruitful discussion on this subject in Mineapolis. 
But, this is not just an IETF issue. Therefore, as both 3GPP and MPEG are 
main "customers" of a timed text streaming solution, I suggest that IETF 
requests 3GPP and MPEG to advice IETF as to the most desirable direction 
for resolving this issue. Note that both 3GPP and MPEG will have meetings 
where this issue can be discussed well before the end of this year. 

Best regards,

Jan van der Meer



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<br><font size=2 face="sans-serif">Dear Jose, all,</font>
<br>
<br><font size=2 face="sans-serif">I will not be in Minneapolis either but would like to provide some input on the issue of transport of 3GPP timed text. I know this input is very late; my appologies for that.</font>
<br>
<br><font size=2 face="sans-serif">As you know also MPEG is working on timed text; currently the Committe Draft (CD) is ballotted, and MPEG schedules to enter the next phase in about one month and to finish the work in July 2004. The end-result will be part 17 of MPEG-4, defining MPEG-4 text streams. Through liaisons, MPEG tries to harmonize its work with the work that is already carried out in 3GPP and also with ongoing work in W3C. Most important in this respect is that MPEG focusses at maximum commonality with the timed text format defined by 3GPP for download. In summary, the current CD of MPEG-4 part 17 specifies streaming of the 3GPP timed text format by defining a very straightforward way to transcode the 3GPP format into the MPEG-4 structure of a decoder configuration and text access units. Please find the Committee Draft (a public document) attached to this email. So far, 3GPP expressed through liaison statements to be very pleased with the work done by MPEG.</font>
<br>
<br><font size=2 face="sans-serif">Once the work is finished, the timed text format can be streamed using &quot;RFC#simple&quot;, draft-ietf-avt-mpeg4-simple-08.txt, that is scheduled to be published as RFC very soon. In summary this approach would work as follows:</font>
<br><font size=2 face="sans-serif">- media type: &quot;video&quot;,</font>
<br><font size=2 face="sans-serif">- MIME subtype name: mpeg4-generic,</font>
<br><font size=2 face="sans-serif">- streamType (required parameter in RFC#simple): indicates MPEG-4 text stream as defined in ISO 14496-17,</font>
<br><font size=2 face="sans-serif">- profile-level-id (required parameter in RFC#simple): specifies the functional subset as defined in the current 3GPP timed text specification,</font>
<br><font size=2 face="sans-serif">- config (required parameter in RFC#simple): carries the decoder configuration specified for text streams in ISO 14496-17,</font>
<br><font size=2 face="sans-serif">- the RTP payload contains either (a) one fragment of a text access unit or (b) one or more complete text access units,</font>
<br><font size=2 face="sans-serif">- the RTP payload carries one AU-header for each carried text access unit,</font>
<br><font size=2 face="sans-serif">- in the RTP payload the concatenated AU-headers preceed concatenated data of text access units, as defined in RFC#simple,</font>
<br><font size=2 face="sans-serif">- each AU-header contains (at least) the following information on the associated text access unit: (a) the size of the text access unit, (b) the time stamp of the text access unit,</font>
<br><font size=2 face="sans-serif">- if deemed necessary, the AU-header may also carry the index of a text access unit (fragment) and/or an indication whether the associated text access unit is a random access point.</font>
<br><font size=2 face="sans-serif">In RFC#simple the fields in the AU-header are configurable, but if desired one or two new modes for RFC#simple could be defined with fixed length fields, for example one using an 8 bit AU-size field (for most common applications) and another using 16 bits AU-size field for applications using (very) large text access units.</font>
<br><font size=2 face="sans-serif">&nbsp;</font>
<br><font size=2 face="sans-serif">The above &quot;RFC#simple approach&quot; has a high level of commonality with draft-rey-avt-rtp-3gpp-timed-text-01.txt. There are some small differences that can be fixed either way, but roughly the &quot;RFC#simple approach&quot; is a simplified version of draft-rey-avt-rtp-3gpp-timed-text-01.txt. In particular, the &quot;RFC#simple&quot; approach does not have the advanced error resiliency measures that are in draft-rey-avt-rtp-3gpp-timed-text-01.txt. </font>
<br>
<br><font size=2 face="sans-serif">In my view not only in IETF, but also in 3GPP and in MPEG a number of basic issues should be discussed, such as:</font>
<br><font size=2 face="sans-serif">- are the error resiliency features of draft-rey-avt-rtp-3gpp-timed-text-01.txt sufficiently valuable to justify their complexity (note that a partly corrupted subtitle may not have much more value to the user than a missing one);</font>
<br><font size=2 face="sans-serif">- is it desirable to define a 3GPP timed text transport solution based on draft-rey-avt-rtp-3gpp-timed-text-01.txt next to the one that will be available based on RFC#simple;</font>
<br><font size=2 face="sans-serif">- is it desirable that draft-rey-avt-rtp-3gpp-timed-text-01.txt specifies a mode or an extension of RFC#simple;</font>
<br><font size=2 face="sans-serif">- how can a timing conflict (if any) between release 6 of 3GPP and part 17 of MPEG-4 be resolved (note that such conflict will only exist if 3GPP adopts streaming of timed text as a work item and if 3GPP decides that this functionality has to be in release 6).</font>
<br>
<br><font size=2 face="sans-serif">I hope you will have a fruitful discussion on this subject in Mineapolis. But, this is not just an IETF issue. Therefore, as both 3GPP and MPEG are main &quot;customers&quot; of a timed text streaming solution, I suggest that IETF requests 3GPP and MPEG to advice IETF as to the most desirable direction for resolving this issue. Note that both 3GPP and MPEG will have meetings where this issue can be discussed well before the end of this year. </font>
<br><font size=2 face="sans-serif"><br>
Best regards,<br>
<br>
Jan van der Meer<br>
</font>
<br>
<br>
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_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Mon Nov 10 11:16:24 2003
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Cc: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [AVT] Slides for Monday AVT session
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Copies of the presentation materials which will be used in AVT this 
afternoon can be retrieved as 
http://www.dcs.gla.ac.uk/~csp/ietf58-avt-mon.zip

These will migrate onto the IETF server after the meeting.

Colin


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: AW: [AVT] More comments on draft-rey-avt-rtp-3gpp-timed-text-01.txt
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Jan, all,


> -----Urspr=FCngliche Nachricht-----
> Von: Jos=E9 Rey [mailto:rey@panasonic.de]
> Gesendet: Montag, 10. November 2003 18:11
> An: Jose Rey
> Betreff:
>
>
> Dear Jose, all,
>
> I will not be in Minneapolis either but would like to provide
> some input
> on the issue of transport of 3GPP timed text. I know this
> input is very
> late; my appologies for that.
>
> As you know also MPEG is working on timed text; currently the Committe
> Draft (CD) is ballotted, and MPEG schedules to enter the next phase in
> about one month and to finish the work in July 2004. The
> end-result will
> be part 17 of MPEG-4, defining MPEG-4 text streams. Through liaisons,
> MPEG tries to harmonize its work with the work that is already carried
> out in 3GPP and also with ongoing work in W3C. Most important in this
> respect is that MPEG focusses at maximum commonality with the
> timed text
> format defined by 3GPP for download. In summary, the current CD of
> MPEG-4 part 17 specifies streaming of the 3GPP timed text format by
> defining a very straightforward way to transcode the 3GPP format into
> the MPEG-4 structure of a decoder configuration and text access units.
> Please find the Committee Draft (a public document) attached to this
> email. So far, 3GPP expressed through liaison statements to be very
> pleased with the work done by MPEG.
>
> Once the work is finished, the timed text format can be streamed using
> "RFC#simple", draft-ietf-avt-mpeg4-simple-08.txt, that is scheduled to
> be published as RFC very soon. In summary this approach would work as
> follows:
> - media type: "video",
> - MIME subtype name: mpeg4-generic,
> - streamType (required parameter in RFC#simple): indicates MPEG-4 text
> stream as defined in ISO 14496-17,
> - profile-level-id (required parameter in RFC#simple): specifies the
> functional subset as defined in the current 3GPP timed text
> specification,
> - config (required parameter in RFC#simple): carries the decoder
> configuration specified for text streams in ISO 14496-17,
> - the RTP payload contains either (a) one fragment of a text
> access unit
> or (b) one or more complete text access units,
> - the RTP payload carries one AU-header for each carried text access
> unit,
> - in the RTP payload the concatenated AU-headers preceed concatenated
> data of text access units, as defined in RFC#simple,
> - each AU-header contains (at least) the following information on the
> associated text access unit: (a) the size of the text access unit, (b)
> the time stamp of the text access unit,
> - if deemed necessary, the AU-header may also carry the index
> of a text
> access unit (fragment) and/or an indication whether the
> associated text
> access unit is a random access point.
> In RFC#simple the fields in the AU-header are configurable, but if
> desired one or two new modes for RFC#simple could be defined
> with fixed
> length fields, for example one using an 8 bit AU-size field (for most
> common applications) and another using 16 bits AU-size field for
> applications using (very) large text access units.
>
> The above "RFC#simple approach" has a high level of commonality with
> draft-rey-avt-rtp-3gpp-timed-text-01.txt. There are some small
> differences that can be fixed either way, but roughly the "RFC#simple
> approach" is a simplified version of
> draft-rey-avt-rtp-3gpp-timed-text-01.txt. In particular, the
> "RFC#simple" approach does not have the advanced error resiliency
> measures that are in draft-rey-avt-rtp-3gpp-timed-text-01.txt.
>
> In my view not only in IETF, but also in 3GPP and in MPEG a number of
> basic issues should be discussed, such as:
> - are the error resiliency features of
> draft-rey-avt-rtp-3gpp-timed-text-01.txt sufficiently valuable to
> justify their complexity (note that a partly corrupted
> subtitle may not
> have much more value to the user than a missing one);

This is a good question which I will also ask during my presentation on
Monday.  This is a question which certainly needs feedback of the group.
The complexity can be reduced, we have to agree where to cut down: do we
need the MTSF headers? or do we want to just keep resiliency for plain
text fragments?, for example.  PT multiplexing will probably be
abandoned for a simpler scheme.

> - is it desirable to define a 3GPP timed text transport solution based
> on draft-rey-avt-rtp-3gpp-timed-text-01.txt next to the one
> that will be
> available based on RFC#simple;

Allowing both would mean that one does not have to implement RFC#simple
in order to transport timed text, which I think is desireable.

> - is it desirable that draft-rey-avt-rtp-3gpp-timed-text-01.txt
> specifies a mode or an extension of RFC#simple;

Would this mean that the RFC#simple has to be implemented to use the
draft? I think this is not a good idea.

> - how can a timing conflict (if any) between release 6 of
> 3GPP and part
> 17 of MPEG-4 be resolved (note that such conflict will only exist if
> 3GPP adopts streaming of timed text as a work item and if 3GPP decides
> that this functionality has to be in release 6).
>

We are targetting to include this in Release 6 (03/04). According to
your information above MPEG4 timed text will be ready in July'04, which
is out of the deadline. Apart from that, current Release 6 does not use
the RFC#simple for streaming... so one would not be able to use timed
text in Release 6.

Assuming timed text comes into the Release 6 agenda, what I would like
to know is what should, in your opinion, the current draft include in
order to consider the MPEG work and the possible extensions it may
incorporate ?. It is now the time to talk about these issues if the
draft shall consider the MPEG work.


> I hope you will have a fruitful discussion on this subject in
> Mineapolis.

You can dowload the session slides and take  a look at the presentation
which is related to this email.

cheers,

Jose

>But, this is not just an IETF issue. Therefore,
> as both 3GPP
> and MPEG are main "customers" of a timed text streaming solution, I
> suggest that IETF requests 3GPP and MPEG to advice IETF as to the most
> desirable direction for resolving this issue. Note that both 3GPP and
> MPEG will have meetings where this issue can be discussed well before
> the end of this year.
>
> Best regards,
>
> Jan van der Meer
>
>


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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To: =?ISO-8859-1?Q?Jos=E9_Rey?= <rey@panasonic.de>
Cc: "'IETF AVT WG'" <avt@ietf.org>, avt-admin@ietf.org,
        "Colin Perkins" <csp@csperkins.org>,
        "Magnus Westerlund" <magnus.westerlund@ericsson.com>,
        "Yoshinori Matsui" <matsui.yoshinori@jp.panasonic.com>,
        "'Dave Singer'" <singer@apple.com>
Subject: Re: AW: [AVT] More comments on draft-rey-avt-rtp-3gpp-timed-text-01.txt
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Hello Jos=E9,

> Allowing both would mean that one does not have to implement RFC#simple
> in order to transport timed text, which I think is desireable.

Could you please clarify the above statement. It is possible to have a=20
straightforward solution for transport of time text based on RFC#simple=20
and your draft is a second and more complex solution that is more=20
desirable because then there is no need to implement the simpler one ??

Best regards,

Jan










Jos=E9 Rey <rey@panasonic.de>
Sent by:=20
avt-admin@ietf.org
2003-11-10 09:02 PM

=20
        To:     Jan vanderMeer/EHV/CE/PHILIPS@PHILIPS
"'IETF AVT WG'" <avt@ietf.org>
        cc:     "Colin Perkins" <csp@csperkins.org>
"Magnus Westerlund" <magnus.westerlund@ericsson.com>
"'Dave Singer'" <singer@apple.com>
"Yoshinori Matsui" <matsui.yoshinori@jp.panasonic.com>
        Subject:        AW: [AVT] More comments on draft-rey-avt-rtp-3gpp-t=
imed-text-01.txt
        Classification:=20




Jan, all,


> -----Urspr=FCngliche Nachricht-----
> Von: Jos=E9 Rey [mailto:rey@panasonic.de]
> Gesendet: Montag, 10. November 2003 18:11
> An: Jose Rey
> Betreff:
>
>
> Dear Jose, all,
>
> I will not be in Minneapolis either but would like to provide
> some input
> on the issue of transport of 3GPP timed text. I know this
> input is very
> late; my appologies for that.
>
> As you know also MPEG is working on timed text; currently the Committe
> Draft (CD) is ballotted, and MPEG schedules to enter the next phase in
> about one month and to finish the work in July 2004. The
> end-result will
> be part 17 of MPEG-4, defining MPEG-4 text streams. Through liaisons,
> MPEG tries to harmonize its work with the work that is already carried
> out in 3GPP and also with ongoing work in W3C. Most important in this
> respect is that MPEG focusses at maximum commonality with the
> timed text
> format defined by 3GPP for download. In summary, the current CD of
> MPEG-4 part 17 specifies streaming of the 3GPP timed text format by
> defining a very straightforward way to transcode the 3GPP format into
> the MPEG-4 structure of a decoder configuration and text access units.
> Please find the Committee Draft (a public document) attached to this
> email. So far, 3GPP expressed through liaison statements to be very
> pleased with the work done by MPEG.
>
> Once the work is finished, the timed text format can be streamed using
> "RFC#simple", draft-ietf-avt-mpeg4-simple-08.txt, that is scheduled to
> be published as RFC very soon. In summary this approach would work as
> follows:
> - media type: "video",
> - MIME subtype name: mpeg4-generic,
> - streamType (required parameter in RFC#simple): indicates MPEG-4 text
> stream as defined in ISO 14496-17,
> - profile-level-id (required parameter in RFC#simple): specifies the
> functional subset as defined in the current 3GPP timed text
> specification,
> - config (required parameter in RFC#simple): carries the decoder
> configuration specified for text streams in ISO 14496-17,
> - the RTP payload contains either (a) one fragment of a text
> access unit
> or (b) one or more complete text access units,
> - the RTP payload carries one AU-header for each carried text access
> unit,
> - in the RTP payload the concatenated AU-headers preceed concatenated
> data of text access units, as defined in RFC#simple,
> - each AU-header contains (at least) the following information on the
> associated text access unit: (a) the size of the text access unit, (b)
> the time stamp of the text access unit,
> - if deemed necessary, the AU-header may also carry the index
> of a text
> access unit (fragment) and/or an indication whether the
> associated text
> access unit is a random access point.
> In RFC#simple the fields in the AU-header are configurable, but if
> desired one or two new modes for RFC#simple could be defined
> with fixed
> length fields, for example one using an 8 bit AU-size field (for most
> common applications) and another using 16 bits AU-size field for
> applications using (very) large text access units.
>
> The above "RFC#simple approach" has a high level of commonality with
> draft-rey-avt-rtp-3gpp-timed-text-01.txt. There are some small
> differences that can be fixed either way, but roughly the "RFC#simple
> approach" is a simplified version of
> draft-rey-avt-rtp-3gpp-timed-text-01.txt. In particular, the
> "RFC#simple" approach does not have the advanced error resiliency
> measures that are in draft-rey-avt-rtp-3gpp-timed-text-01.txt.
>
> In my view not only in IETF, but also in 3GPP and in MPEG a number of
> basic issues should be discussed, such as:
> - are the error resiliency features of
> draft-rey-avt-rtp-3gpp-timed-text-01.txt sufficiently valuable to
> justify their complexity (note that a partly corrupted
> subtitle may not
> have much more value to the user than a missing one);

This is a good question which I will also ask during my presentation on
Monday.  This is a question which certainly needs feedback of the group.
The complexity can be reduced, we have to agree where to cut down: do we
need the MTSF headers? or do we want to just keep resiliency for plain
text fragments?, for example.  PT multiplexing will probably be
abandoned for a simpler scheme.

> - is it desirable to define a 3GPP timed text transport solution based
> on draft-rey-avt-rtp-3gpp-timed-text-01.txt next to the one
> that will be
> available based on RFC#simple;

Allowing both would mean that one does not have to implement RFC#simple
in order to transport timed text, which I think is desireable.

> - is it desirable that draft-rey-avt-rtp-3gpp-timed-text-01.txt
> specifies a mode or an extension of RFC#simple;

Would this mean that the RFC#simple has to be implemented to use the
draft? I think this is not a good idea.

> - how can a timing conflict (if any) between release 6 of
> 3GPP and part
> 17 of MPEG-4 be resolved (note that such conflict will only exist if
> 3GPP adopts streaming of timed text as a work item and if 3GPP decides
> that this functionality has to be in release 6).
>

We are targetting to include this in Release 6 (03/04). According to
your information above MPEG4 timed text will be ready in July'04, which
is out of the deadline. Apart from that, current Release 6 does not use
the RFC#simple for streaming... so one would not be able to use timed
text in Release 6.

Assuming timed text comes into the Release 6 agenda, what I would like
to know is what should, in your opinion, the current draft include in
order to consider the MPEG work and the possible extensions it may
incorporate ?. It is now the time to talk about these issues if the
draft shall consider the MPEG work.


> I hope you will have a fruitful discussion on this subject in
> Mineapolis.

You can dowload the session slides and take  a look at the presentation
which is related to this email.

cheers,

Jose

>But, this is not just an IETF issue. Therefore,
> as both 3GPP
> and MPEG are main "customers" of a timed text streaming solution, I
> suggest that IETF requests 3GPP and MPEG to advice IETF as to the most
> desirable direction for resolving this issue. Note that both 3GPP and
> MPEG will have meetings where this issue can be discussed well before
> the end of this year.
>
> Best regards,
>
> Jan van der Meer
>
>


=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=
=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



--=_alternative 00336142C1256DDB_=
Content-Type: text/html; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable


<br><font size=3D2 face=3D"sans-serif">Hello Jos=E9,</font>
<br>
<br><font size=3D2 face=3D"Courier New">&gt; Allowing both would mean that =
one does not have to implement RFC#simple<br>
&gt; in order to transport timed text, which I think is desireable.</font><=
font size=3D2 face=3D"sans-serif"><br>
</font>
<br><font size=3D2 face=3D"sans-serif">Could you please clarify the above s=
tatement. It is possible to have a straightforward solution for transport o=
f time text based on RFC#simple and your draft is a second and more complex=
 solution that is more desirable because then there is no need to implement=
 the simpler one ??</font>
<br><font size=3D2 face=3D"sans-serif"><br>
Best regards,<br>
<br>
Jan<br>
</font>
<br>
<br>
<br>
<table width=3D100%>
<tr valign=3Dtop>
<td>
<td>
<br>
<br>
<br>
<br>
<br><font size=3D1 face=3D"sans-serif"><b>Jos=E9 Rey &lt;rey@panasonic.de&g=
t;</b></font>
<p><font size=3D1 face=3D"sans-serif">Sent by: </font>
<br><font size=3D1 face=3D"sans-serif">avt-admin@ietf.org</font>
<p><font size=3D1 face=3D"sans-serif">2003-11-10 09:02 PM</font>
<br>
<td><font size=3D1 face=3D"Arial">&nbsp; &nbsp; &nbsp; &nbsp; </font>
<br><font size=3D1 face=3D"sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; To: &nbs=
p; &nbsp; &nbsp; &nbsp;Jan vanderMeer/EHV/CE/PHILIPS@PHILIPS<br>
&quot;'IETF AVT WG'&quot; &lt;avt@ietf.org&gt;</font>
<br><font size=3D1 face=3D"sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; cc: &nbs=
p; &nbsp; &nbsp; &nbsp;&quot;Colin Perkins&quot; &lt;csp@csperkins.org&gt;<=
br>
&quot;Magnus Westerlund&quot; &lt;magnus.westerlund@ericsson.com&gt;<br>
&quot;'Dave Singer'&quot; &lt;singer@apple.com&gt;<br>
&quot;Yoshinori Matsui&quot; &lt;matsui.yoshinori@jp.panasonic.com&gt;</fon=
t>
<br><font size=3D1 face=3D"sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; Subject:=
 &nbsp; &nbsp; &nbsp; &nbsp;AW: [AVT] More comments on draft-rey-avt-rtp-3g=
pp-timed-text-01.txt</font>
<p><font size=3D1 face=3D"sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; Classific=
ation: &nbsp; &nbsp; &nbsp; &nbsp;</font>
<br></table>
<br>
<br>
<br><font size=3D2 face=3D"Courier New"><br>
Jan, all,<br>
<br>
<br>
&gt; -----Urspr=FCngliche Nachricht-----<br>
&gt; Von: Jos=E9 Rey [mailto:rey@panasonic.de]<br>
&gt; Gesendet: Montag, 10. November 2003 18:11<br>
&gt; An: Jose Rey<br>
&gt; Betreff:<br>
&gt;<br>
&gt;<br>
&gt; Dear Jose, all,<br>
&gt;<br>
&gt; I will not be in Minneapolis either but would like to provide<br>
&gt; some input<br>
&gt; on the issue of transport of 3GPP timed text. I know this<br>
&gt; input is very<br>
&gt; late; my appologies for that.<br>
&gt;<br>
&gt; As you know also MPEG is working on timed text; currently the Committe=
<br>
&gt; Draft (CD) is ballotted, and MPEG schedules to enter the next phase in=
<br>
&gt; about one month and to finish the work in July 2004. The<br>
&gt; end-result will<br>
&gt; be part 17 of MPEG-4, defining MPEG-4 text streams. Through liaisons,<=
br>
&gt; MPEG tries to harmonize its work with the work that is already carried=
<br>
&gt; out in 3GPP and also with ongoing work in W3C. Most important in this<=
br>
&gt; respect is that MPEG focusses at maximum commonality with the<br>
&gt; timed text<br>
&gt; format defined by 3GPP for download. In summary, the current CD of<br>
&gt; MPEG-4 part 17 specifies streaming of the 3GPP timed text format by<br>
&gt; defining a very straightforward way to transcode the 3GPP format into<=
br>
&gt; the MPEG-4 structure of a decoder configuration and text access units.=
<br>
&gt; Please find the Committee Draft (a public document) attached to this<b=
r>
&gt; email. So far, 3GPP expressed through liaison statements to be very<br>
&gt; pleased with the work done by MPEG.<br>
&gt;<br>
&gt; Once the work is finished, the timed text format can be streamed using=
<br>
&gt; &quot;RFC#simple&quot;, draft-ietf-avt-mpeg4-simple-08.txt, that is sc=
heduled to<br>
&gt; be published as RFC very soon. In summary this approach would work as<=
br>
&gt; follows:<br>
&gt; - media type: &quot;video&quot;,<br>
&gt; - MIME subtype name: mpeg4-generic,<br>
&gt; - streamType (required parameter in RFC#simple): indicates MPEG-4 text=
<br>
&gt; stream as defined in ISO 14496-17,<br>
&gt; - profile-level-id (required parameter in RFC#simple): specifies the<b=
r>
&gt; functional subset as defined in the current 3GPP timed text<br>
&gt; specification,<br>
&gt; - config (required parameter in RFC#simple): carries the decoder<br>
&gt; configuration specified for text streams in ISO 14496-17,<br>
&gt; - the RTP payload contains either (a) one fragment of a text<br>
&gt; access unit<br>
&gt; or (b) one or more complete text access units,<br>
&gt; - the RTP payload carries one AU-header for each carried text access<b=
r>
&gt; unit,<br>
&gt; - in the RTP payload the concatenated AU-headers preceed concatenated<=
br>
&gt; data of text access units, as defined in RFC#simple,<br>
&gt; - each AU-header contains (at least) the following information on the<=
br>
&gt; associated text access unit: (a) the size of the text access unit, (b)=
<br>
&gt; the time stamp of the text access unit,<br>
&gt; - if deemed necessary, the AU-header may also carry the index<br>
&gt; of a text<br>
&gt; access unit (fragment) and/or an indication whether the<br>
&gt; associated text<br>
&gt; access unit is a random access point.<br>
&gt; In RFC#simple the fields in the AU-header are configurable, but if<br>
&gt; desired one or two new modes for RFC#simple could be defined<br>
&gt; with fixed<br>
&gt; length fields, for example one using an 8 bit AU-size field (for most<=
br>
&gt; common applications) and another using 16 bits AU-size field for<br>
&gt; applications using (very) large text access units.<br>
&gt;<br>
&gt; The above &quot;RFC#simple approach&quot; has a high level of commonal=
ity with<br>
&gt; draft-rey-avt-rtp-3gpp-timed-text-01.txt. There are some small<br>
&gt; differences that can be fixed either way, but roughly the &quot;RFC#si=
mple<br>
&gt; approach&quot; is a simplified version of<br>
&gt; draft-rey-avt-rtp-3gpp-timed-text-01.txt. In particular, the<br>
&gt; &quot;RFC#simple&quot; approach does not have the advanced error resil=
iency<br>
&gt; measures that are in draft-rey-avt-rtp-3gpp-timed-text-01.txt.<br>
&gt;<br>
&gt; In my view not only in IETF, but also in 3GPP and in MPEG a number of<=
br>
&gt; basic issues should be discussed, such as:<br>
&gt; - are the error resiliency features of<br>
&gt; draft-rey-avt-rtp-3gpp-timed-text-01.txt sufficiently valuable to<br>
&gt; justify their complexity (note that a partly corrupted<br>
&gt; subtitle may not<br>
&gt; have much more value to the user than a missing one);<br>
<br>
This is a good question which I will also ask during my presentation on<br>
Monday. &nbsp;This is a question which certainly needs feedback of the grou=
p.<br>
The complexity can be reduced, we have to agree where to cut down: do we<br>
need the MTSF headers? or do we want to just keep resiliency for plain<br>
text fragments?, for example. &nbsp;PT multiplexing will probably be<br>
abandoned for a simpler scheme.<br>
<br>
&gt; - is it desirable to define a 3GPP timed text transport solution based=
<br>
&gt; on draft-rey-avt-rtp-3gpp-timed-text-01.txt next to the one<br>
&gt; that will be<br>
&gt; available based on RFC#simple;<br>
<br>
Allowing both would mean that one does not have to implement RFC#simple<br>
in order to transport timed text, which I think is desireable.</font>
<br><font size=3D2 face=3D"Courier New"><br>
&gt; - is it desirable that draft-rey-avt-rtp-3gpp-timed-text-01.txt<br>
&gt; specifies a mode or an extension of RFC#simple;<br>
<br>
Would this mean that the RFC#simple has to be implemented to use the<br>
draft? I think this is not a good idea.<br>
<br>
&gt; - how can a timing conflict (if any) between release 6 of<br>
&gt; 3GPP and part<br>
&gt; 17 of MPEG-4 be resolved (note that such conflict will only exist if<b=
r>
&gt; 3GPP adopts streaming of timed text as a work item and if 3GPP decides=
<br>
&gt; that this functionality has to be in release 6).<br>
&gt;<br>
<br>
We are targetting to include this in Release 6 (03/04). According to<br>
your information above MPEG4 timed text will be ready in July'04, which<br>
is out of the deadline. Apart from that, current Release 6 does not use<br>
the RFC#simple for streaming... so one would not be able to use timed<br>
text in Release 6.<br>
<br>
Assuming timed text comes into the Release 6 agenda, what I would like<br>
to know is what should, in your opinion, the current draft include in<br>
order to consider the MPEG work and the possible extensions it may<br>
incorporate ?. It is now the time to talk about these issues if the<br>
draft shall consider the MPEG work.<br>
<br>
<br>
&gt; I hope you will have a fruitful discussion on this subject in<br>
&gt; Mineapolis.<br>
<br>
You can dowload the session slides and take &nbsp;a look at the presentatio=
n<br>
which is related to this email.<br>
<br>
cheers,<br>
<br>
Jose<br>
<br>
&gt;But, this is not just an IETF issue. Therefore,<br>
&gt; as both 3GPP<br>
&gt; and MPEG are main &quot;customers&quot; of a timed text streaming solu=
tion, I<br>
&gt; suggest that IETF requests 3GPP and MPEG to advice IETF as to the most=
<br>
&gt; desirable direction for resolving this issue. Note that both 3GPP and<=
br>
&gt; MPEG will have meetings where this issue can be discussed well before<=
br>
&gt; the end of this year.<br>
&gt;<br>
&gt; Best regards,<br>
&gt;<br>
&gt; Jan van der Meer<br>
&gt;<br>
&gt;<br>
<br>
<br>
=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=
=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F<br>
Audio/Video Transport Working Group<br>
avt@ietf.org<br>
https://www1.ietf.org/mailman/listinfo/avt<br>
</font>
<br>
<br>
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_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: Re: [AVT] Slides for Monday AVT session
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Hello Colin,

Thanks for making the slides available. I can read all of them, except the 
one on Timed Text, which is corrupted when reading it with my Acrobat 
Reader 6.0. Acrobat tells it's a font encoding problem. I would appreciate 
if you could resolve that.

Best regards,

Jan










Colin Perkins <csp@csperkins.org>
Sent by: 
avt-admin@ietf.org
2003-11-10 05:14 PM

 
        To:     avt@ietf.org
        cc:     Magnus Westerlund <magnus.westerlund@ericsson.com>
(bcc: Jan vanderMeer/EHV/CE/PHILIPS)
        Subject:        [AVT] Slides for Monday AVT session
        Classification: 



Copies of the presentation materials which will be used in AVT this 
afternoon can be retrieved as 
http://www.dcs.gla.ac.uk/~csp/ietf58-avt-mon.zip

These will migrate onto the IETF server after the meeting.

Colin


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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


<br><font size=2 face="sans-serif">Hello Colin,</font>
<br>
<br><font size=2 face="sans-serif">Thanks for making the slides available. I can read all of them, except the one on Timed Text, which is corrupted when reading it with my Acrobat Reader 6.0. Acrobat tells it's a font encoding problem. I would appreciate if you could resolve that.<br>
<br>
Best regards,<br>
<br>
Jan<br>
</font>
<br>
<br>
<br>
<table width=100%>
<tr valign=top>
<td>
<td>
<br>
<br>
<br>
<br>
<br><font size=1 face="sans-serif"><b>Colin Perkins &lt;csp@csperkins.org&gt;</b></font>
<p><font size=1 face="sans-serif">Sent by: </font>
<br><font size=1 face="sans-serif">avt-admin@ietf.org</font>
<p><font size=1 face="sans-serif">2003-11-10 05:14 PM</font>
<br>
<td><font size=1 face="Arial">&nbsp; &nbsp; &nbsp; &nbsp; </font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; To: &nbsp; &nbsp; &nbsp; &nbsp;avt@ietf.org</font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; cc: &nbsp; &nbsp; &nbsp; &nbsp;Magnus Westerlund &lt;magnus.westerlund@ericsson.com&gt;<br>
(bcc: Jan vanderMeer/EHV/CE/PHILIPS)</font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; Subject: &nbsp; &nbsp; &nbsp; &nbsp;[AVT] Slides for Monday AVT session</font>
<p><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; Classification: &nbsp; &nbsp; &nbsp; &nbsp;</font>
<br></table>
<br>
<br>
<br><font size=2 face="Courier New">Copies of the presentation materials which will be used in AVT this <br>
afternoon can be retrieved as <br>
http://www.dcs.gla.ac.uk/~csp/ietf58-avt-mon.zip<br>
<br>
These will migrate onto the IETF server after the meeting.<br>
<br>
Colin<br>
<br>
<br>
_______________________________________________<br>
Audio/Video Transport Working Group<br>
avt@ietf.org<br>
https://www1.ietf.org/mailman/listinfo/avt<br>
</font>
<br>
<br>
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_______________________________________________
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https://www1.ietf.org/mailman/listinfo/avt



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On Tuesday, Nov 11, 2003, at 05:13 US/Eastern, 
jan.vandermeer@philips.com wrote:
> Thanks for making the slides available. I can read all of them, except 
> the one on Timed Text, which is corrupted when reading it with my 
> Acrobat Reader 6.0. Acrobat tells it's a font encoding problem. I 
> would appreciate if you could resolve that.

I updated the zip file - maybe it's better now?

Colin


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Colin,

Thanks for your effort, but unfortunately the timed-text presentation is 
still corrupted. Because of the problem I installed the latest version of 
the Acrobat Reader, but that did not help.

Best regards,

Jan










Colin Perkins <csp@csperkins.org>
Sent by: 
avt-admin@ietf.org
2003-11-11 05:11 PM

 
        To:     Jan vanderMeer/EHV/CE/PHILIPS@PHILIPS
        cc:     avt@ietf.org
        Subject:        Re: [AVT] Slides for Monday AVT session
        Classification: 



On Tuesday, Nov 11, 2003, at 05:13 US/Eastern, 
jan.vandermeer@philips.com wrote:
> Thanks for making the slides available. I can read all of them, except 
> the one on Timed Text, which is corrupted when reading it with my 
> Acrobat Reader 6.0. Acrobat tells it's a font encoding problem. I 
> would appreciate if you could resolve that.

I updated the zip file - maybe it's better now?

Colin


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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<br><font size=2 face="sans-serif">Colin,</font>
<br>
<br><font size=2 face="sans-serif">Thanks for your effort, but unfortunately the timed-text presentation is still corrupted. Because of the problem I installed the latest version of the Acrobat Reader, but that did not help.</font>
<br><font size=2 face="sans-serif"><br>
Best regards,<br>
<br>
Jan<br>
</font>
<br>
<br>
<br>
<table width=100%>
<tr valign=top>
<td>
<td>
<br>
<br>
<br>
<br>
<br><font size=1 face="sans-serif"><b>Colin Perkins &lt;csp@csperkins.org&gt;</b></font>
<p><font size=1 face="sans-serif">Sent by: </font>
<br><font size=1 face="sans-serif">avt-admin@ietf.org</font>
<p><font size=1 face="sans-serif">2003-11-11 05:11 PM</font>
<br>
<td><font size=1 face="Arial">&nbsp; &nbsp; &nbsp; &nbsp; </font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; To: &nbsp; &nbsp; &nbsp; &nbsp;Jan vanderMeer/EHV/CE/PHILIPS@PHILIPS</font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; cc: &nbsp; &nbsp; &nbsp; &nbsp;avt@ietf.org</font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; Subject: &nbsp; &nbsp; &nbsp; &nbsp;Re: [AVT] Slides for Monday AVT session</font>
<p><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; Classification: &nbsp; &nbsp; &nbsp; &nbsp;</font>
<br></table>
<br>
<br>
<br><font size=2 face="Courier New">On Tuesday, Nov 11, 2003, at 05:13 US/Eastern, <br>
jan.vandermeer@philips.com wrote:<br>
&gt; Thanks for making the slides available. I can read all of them, except <br>
&gt; the one on Timed Text, which is corrupted when reading it with my <br>
&gt; Acrobat Reader 6.0. Acrobat tells it's a font encoding problem. I <br>
&gt; would appreciate if you could resolve that.<br>
<br>
I updated the zip file - maybe it's better now?<br>
<br>
Colin<br>
<br>
<br>
_______________________________________________<br>
Audio/Video Transport Working Group<br>
avt@ietf.org<br>
https://www1.ietf.org/mailman/listinfo/avt<br>
</font>
<br>
<br>
--=_alternative 005A2F8DC1256DDB_=--

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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	 Tue, 11 Nov 2003 17:27:13 +0100
From: "Jose Rey" <rey@panasonic.de>
To: <jan.vandermeer@philips.com>, "'IETF AVT WG'" <avt@ietf.org>
Cc: "Colin Perkins" <csp@csperkins.org>,
        "Magnus Westerlund" <magnus.westerlund@ericsson.com>,
        "'Dave Singer'" <singer@apple.com>,
        "Yoshinori Matsui" <matsui.yoshinori@jp.panasonic.com>
Subject: AW: [AVT] More comments on draft-rey-avt-rtp-3gpp-timed-text-01.txt
Date: Tue, 11 Nov 2003 17:27:12 +0100
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Hi all,

for those that did not attend the AVT session on Monday, there was a
number of comments and useful feedback during the timed text
presentation.

In particular, it was suggested that the draft sticks too much to the
3gp file format itself and does not look at the particularities of the
RTP transport (Colin please correct me if the language is wrong).
Currently the draft just reads the text samples from the 3gp file and
puts them into RTP packets. If a packet is too big for one RTP packet
then it is cut and a fragment header is used to be able to interpret the
information in that fragment. As long as fragmentation happens rarely
then this could be the best option.

On the other side, there's the option (which I call 'pure ALF approach'
in the slides) to preprocess the data read from the 3gp file before
packetising. This preprocessing would not only cut the text samples to
fix the RTP packet size but also re-arrange the packets so that the
fragments are not only self-contained but contain all the decorations.
This approach requires to define extra headers for the cases in which
the modifier boxes would have to be split into several RTP packets.

You can download some clarifying slides at
http://www.pel.panasonic.de/ietf/docs/tt-design.pdf in pdf format or
from http://www.pel.panasonic.de/ietf/docs/tt-design.ppt in MSPowerPoint
format.

The current approach makes sense if the text samples are regularly small
and one or several of them fit into an RTP packet. Fragmentation would
happen seldom. The 'pure alf approach' is, in general, the most correct
one, and so also the one that requires more header definitions. If
fragmentation is a rare event, this overhead may be not worth (?).  The
question to the group is pretty simple: which approach should be
pursued?


We appreciate your feedback,

cheers,


Jose





> -----Urspr=FCngliche Nachricht-----
> Von: avt-admin@ietf.org [mailto:avt-admin@ietf.org]Im Auftrag von Jos=E9
> Rey
> Gesendet: Montag, 10. November 2003 21:02
> An: jan.vandermeer@philips.com; 'IETF AVT WG'
> Cc: Colin Perkins; Magnus Westerlund; 'Dave Singer'; Yoshinori Matsui
> Betreff: AW: [AVT] More comments on
> draft-rey-avt-rtp-3gpp-timed-text-01.txt
>
>
>
> Jan, all,
>
>
> > -----Urspr=FCngliche Nachricht-----
> > Von: Jos=E9 Rey [mailto:rey@panasonic.de]
> > Gesendet: Montag, 10. November 2003 18:11
> > An: Jose Rey
> > Betreff:
> >
> >
> > Dear Jose, all,
> >
> > I will not be in Minneapolis either but would like to provide
> > some input
> > on the issue of transport of 3GPP timed text. I know this
> > input is very
> > late; my appologies for that.
> >
> > As you know also MPEG is working on timed text; currently
> the Committe
> > Draft (CD) is ballotted, and MPEG schedules to enter the
> next phase in
> > about one month and to finish the work in July 2004. The
> > end-result will
> > be part 17 of MPEG-4, defining MPEG-4 text streams. Through
> liaisons,
> > MPEG tries to harmonize its work with the work that is
> already carried
> > out in 3GPP and also with ongoing work in W3C. Most
> important in this
> > respect is that MPEG focusses at maximum commonality with the
> > timed text
> > format defined by 3GPP for download. In summary, the current CD of
> > MPEG-4 part 17 specifies streaming of the 3GPP timed text format by
> > defining a very straightforward way to transcode the 3GPP
> format into
> > the MPEG-4 structure of a decoder configuration and text
> access units.
> > Please find the Committee Draft (a public document) attached to this
> > email. So far, 3GPP expressed through liaison statements to be very
> > pleased with the work done by MPEG.
> >
> > Once the work is finished, the timed text format can be
> streamed using
> > "RFC#simple", draft-ietf-avt-mpeg4-simple-08.txt, that is
> scheduled to
> > be published as RFC very soon. In summary this approach
> would work as
> > follows:
> > - media type: "video",
> > - MIME subtype name: mpeg4-generic,
> > - streamType (required parameter in RFC#simple): indicates
> MPEG-4 text
> > stream as defined in ISO 14496-17,
> > - profile-level-id (required parameter in RFC#simple): specifies the
> > functional subset as defined in the current 3GPP timed text
> > specification,
> > - config (required parameter in RFC#simple): carries the decoder
> > configuration specified for text streams in ISO 14496-17,
> > - the RTP payload contains either (a) one fragment of a text
> > access unit
> > or (b) one or more complete text access units,
> > - the RTP payload carries one AU-header for each carried text access
> > unit,
> > - in the RTP payload the concatenated AU-headers preceed
> concatenated
> > data of text access units, as defined in RFC#simple,
> > - each AU-header contains (at least) the following
> information on the
> > associated text access unit: (a) the size of the text
> access unit, (b)
> > the time stamp of the text access unit,
> > - if deemed necessary, the AU-header may also carry the index
> > of a text
> > access unit (fragment) and/or an indication whether the
> > associated text
> > access unit is a random access point.
> > In RFC#simple the fields in the AU-header are configurable, but if
> > desired one or two new modes for RFC#simple could be defined
> > with fixed
> > length fields, for example one using an 8 bit AU-size field
> (for most
> > common applications) and another using 16 bits AU-size field for
> > applications using (very) large text access units.
> >
> > The above "RFC#simple approach" has a high level of commonality with
> > draft-rey-avt-rtp-3gpp-timed-text-01.txt. There are some small
> > differences that can be fixed either way, but roughly the
> "RFC#simple
> > approach" is a simplified version of
> > draft-rey-avt-rtp-3gpp-timed-text-01.txt. In particular, the
> > "RFC#simple" approach does not have the advanced error resiliency
> > measures that are in draft-rey-avt-rtp-3gpp-timed-text-01.txt.
> >
> > In my view not only in IETF, but also in 3GPP and in MPEG a
> number of
> > basic issues should be discussed, such as:
> > - are the error resiliency features of
> > draft-rey-avt-rtp-3gpp-timed-text-01.txt sufficiently valuable to
> > justify their complexity (note that a partly corrupted
> > subtitle may not
> > have much more value to the user than a missing one);
>
> This is a good question which I will also ask during my
> presentation on
> Monday.  This is a question which certainly needs feedback of
> the group.
> The complexity can be reduced, we have to agree where to cut
> down: do we
> need the MTSF headers? or do we want to just keep resiliency for plain
> text fragments?, for example.  PT multiplexing will probably be
> abandoned for a simpler scheme.
>
> > - is it desirable to define a 3GPP timed text transport
> solution based
> > on draft-rey-avt-rtp-3gpp-timed-text-01.txt next to the one
> > that will be
> > available based on RFC#simple;
>
> Allowing both would mean that one does not have to implement
> RFC#simple
> in order to transport timed text, which I think is desireable.
>
> > - is it desirable that draft-rey-avt-rtp-3gpp-timed-text-01.txt
> > specifies a mode or an extension of RFC#simple;
>
> Would this mean that the RFC#simple has to be implemented to use the
> draft? I think this is not a good idea.
>
> > - how can a timing conflict (if any) between release 6 of
> > 3GPP and part
> > 17 of MPEG-4 be resolved (note that such conflict will only exist if
> > 3GPP adopts streaming of timed text as a work item and if
> 3GPP decides
> > that this functionality has to be in release 6).
> >
>
> We are targetting to include this in Release 6 (03/04). According to
> your information above MPEG4 timed text will be ready in
> July'04, which
> is out of the deadline. Apart from that, current Release 6
> does not use
> the RFC#simple for streaming... so one would not be able to use timed
> text in Release 6.
>
> Assuming timed text comes into the Release 6 agenda, what I would like
> to know is what should, in your opinion, the current draft include in
> order to consider the MPEG work and the possible extensions it may
> incorporate ?. It is now the time to talk about these issues if the
> draft shall consider the MPEG work.
>
>
> > I hope you will have a fruitful discussion on this subject in
> > Mineapolis.
>
> You can dowload the session slides and take  a look at the
> presentation
> which is related to this email.
>
> cheers,
>
> Jose
>
> >But, this is not just an IETF issue. Therefore,
> > as both 3GPP
> > and MPEG are main "customers" of a timed text streaming solution, I
> > suggest that IETF requests 3GPP and MPEG to advice IETF as
> to the most
> > desirable direction for resolving this issue. Note that
> both 3GPP and
> > MPEG will have meetings where this issue can be discussed
> well before
> > the end of this year.
> >
> > Best regards,
> >
> > Jan van der Meer
> >
> >
>
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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	 Tue, 11 Nov 2003 17:31:38 +0100
From: "Jose Rey" <rey@panasonic.de>
To: <jan.vandermeer@philips.com>, "Colin Perkins" <csp@csperkins.org>
Cc: <avt@ietf.org>
Subject: AW: [AVT] Slides for Monday AVT session
Date: Tue, 11 Nov 2003 17:31:38 +0100
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Jan, you can download the ppt version of it under
http://www.pel.panasonic.de/ietf/docs/ietf58-tt.ppt

cheers,

Jos=E9


  -----Urspr=FCngliche Nachricht-----
  Von: avt-admin@ietf.org [mailto:avt-admin@ietf.org]Im Auftrag von
jan.vandermeer@philips.com
  Gesendet: Dienstag, 11. November 2003 17:23
  An: Colin Perkins
  Cc: avt@ietf.org; avt-admin@ietf.org
  Betreff: Re: [AVT] Slides for Monday AVT session



  Colin,

  Thanks for your effort, but unfortunately the timed-text presentation i=
s
still corrupted. Because of the problem I installed the latest version of
the Acrobat Reader, but that did not help.

  Best regards,

  Jan








        Colin Perkins <csp@csperkins.org>
        Sent by:
        avt-admin@ietf.org

        2003-11-11 05:11 PM


                To:        Jan vanderMeer/EHV/CE/PHILIPS@PHILIPS
                cc:        avt@ietf.org
                Subject:        Re: [AVT] Slides for Monday AVT session
                Classification:





  On Tuesday, Nov 11, 2003, at 05:13 US/Eastern,
  jan.vandermeer@philips.com wrote:
  > Thanks for making the slides available. I can read all of them, excep=
t
  > the one on Timed Text, which is corrupted when reading it with my
  > Acrobat Reader 6.0. Acrobat tells it's a font encoding problem. I
  > would appreciate if you could resolve that.

  I updated the zip file - maybe it's better now?

  Colin


  _______________________________________________
  Audio/Video Transport Working Group
  avt@ietf.org
  https://www1.ietf.org/mailman/listinfo/avt




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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
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http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.2919.6307" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN =
class=3D130253016-11112003>Jan,=20
you can download the ppt version of it under <A=20
href=3D"http://www.pel.panasonic.de/ietf/docs/ietf58-tt.ppt">http://www.p=
el.panasonic.de/ietf/docs/ietf58-tt.ppt</A></SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D130253016-11112003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D130253016-11112003>cheers,</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D130253016-11112003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D130253016-11112003>Jos=E9</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D130253016-11112003></SPAN></FONT>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<BLOCKQUOTE=20
style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; PADDING-LEFT: =
5px">
  <DIV align=3Dleft class=3DOutlookMessageHeader dir=3Dltr><FONT =
face=3DTahoma=20
  size=3D2>-----Urspr=FCngliche Nachricht-----<BR><B>Von:</B> =
avt-admin@ietf.org=20
  [mailto:avt-admin@ietf.org]<B>Im Auftrag von=20
  </B>jan.vandermeer@philips.com<BR><B>Gesendet:</B> Dienstag, 11. =
November 2003=20
  17:23<BR><B>An:</B> Colin Perkins<BR><B>Cc:</B> avt@ietf.org;=20
  avt-admin@ietf.org<BR><B>Betreff:</B> Re: [AVT] Slides for Monday AVT=20
  session<BR><BR></DIV></FONT><BR><FONT face=3Dsans-serif =
size=3D2>Colin,</FONT>=20
  <BR><BR><FONT face=3Dsans-serif size=3D2>Thanks for your effort, but =
unfortunately=20
  the timed-text presentation is still corrupted. Because of the problem =
I=20
  installed the latest version of the Acrobat Reader, but that did not=20
  help.</FONT> <BR><FONT face=3Dsans-serif size=3D2><BR>Best=20
  regards,<BR><BR>Jan<BR></FONT><BR><BR><BR>
  <TABLE width=3D"100%">
    <TBODY>
    <TR vAlign=3Dtop>
      <TD>
      <TD><BR><BR><BR><BR><BR><FONT face=3Dsans-serif size=3D1><B>Colin =
Perkins=20
        &lt;csp@csperkins.org&gt;</B></FONT>=20
        <P><FONT face=3Dsans-serif size=3D1>Sent by: </FONT><BR><FONT=20
        face=3Dsans-serif size=3D1>avt-admin@ietf.org</FONT>=20
        <P><FONT face=3Dsans-serif size=3D1>2003-11-11 05:11 PM</FONT> =
<BR></P>
      <TD><FONT face=3DArial size=3D1>&nbsp; &nbsp; &nbsp; &nbsp; =
</FONT><BR><FONT=20
        face=3Dsans-serif size=3D1>&nbsp; &nbsp; &nbsp; &nbsp; To: =
&nbsp; &nbsp;=20
        &nbsp; &nbsp;Jan vanderMeer/EHV/CE/PHILIPS@PHILIPS</FONT> =
<BR><FONT=20
        face=3Dsans-serif size=3D1>&nbsp; &nbsp; &nbsp; &nbsp; cc: =
&nbsp; &nbsp;=20
        &nbsp; &nbsp;avt@ietf.org</FONT> <BR><FONT face=3Dsans-serif =
size=3D1>&nbsp;=20
        &nbsp; &nbsp; &nbsp; Subject: &nbsp; &nbsp; &nbsp; &nbsp;Re: =
[AVT]=20
        Slides for Monday AVT session</FONT>=20
        <P><FONT face=3Dsans-serif size=3D1>&nbsp; &nbsp; &nbsp; &nbsp;=20
        Classification: &nbsp; &nbsp; &nbsp; &nbsp;</FONT>=20
  <BR></P></TR></TBODY></TABLE><BR><BR><BR><FONT face=3D"Courier New" =
size=3D2>On=20
  Tuesday, Nov 11, 2003, at 05:13 US/Eastern, =
<BR>jan.vandermeer@philips.com=20
  wrote:<BR>&gt; Thanks for making the slides available. I can read all =
of them,=20
  except <BR>&gt; the one on Timed Text, which is corrupted when reading =
it with=20
  my <BR>&gt; Acrobat Reader 6.0. Acrobat tells it's a font encoding =
problem. I=20
  <BR>&gt; would appreciate if you could resolve that.<BR><BR>I updated =
the zip=20
  file - maybe it's better=20
  =
now?<BR><BR>Colin<BR><BR><BR>____________________________________________=
___<BR>Audio/Video=20
  Transport Working=20
  =
Group<BR>avt@ietf.org<BR>https://www1.ietf.org/mailman/listinfo/avt<BR></=
FONT><BR><BR></BLOCKQUOTE></BODY></HTML>

------=_NextPart_000_0006_01C3A879.A9B7A580--


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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To: Colin Perkins <csp@csperkins.org>
Cc: avt@ietf.org, avt-admin@ietf.org
Subject: Re: [AVT] Slides for Monday AVT session
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Hello Colin,

From Jose's latest email:

> You can download some clarifying slides at
> http://www.pel.panasonic.de/ietf/docs/tt-design.pdf in pdf format or
> from http://www.pel.panasonic.de/ietf/docs/tt-design.ppt in MSPowerPoint
> format.

The pdf above does not work for me either, but the ppt does. Perhaps it is 
possible to upload the original ppt ?

Best regards,

Jan










Jan vanderMeer
2003-11-11 05:23 PM


        To:     Colin Perkins <csp@csperkins.org>
        cc:     avt@ietf.org
avt-admin@ietf.org
        Subject:        Re: [AVT] Slides for Monday AVT session
        Classification: Unclassified


Colin,

Thanks for your effort, but unfortunately the timed-text presentation is 
still corrupted. Because of the problem I installed the latest version of 
the Acrobat Reader, but that did not help.

Best regards,

Jan










Colin Perkins <csp@csperkins.org>
Sent by: 
avt-admin@ietf.org
2003-11-11 05:11 PM

 
        To:     Jan vanderMeer/EHV/CE/PHILIPS@PHILIPS
        cc:     avt@ietf.org
        Subject:        Re: [AVT] Slides for Monday AVT session
        Classification: 



On Tuesday, Nov 11, 2003, at 05:13 US/Eastern, 
jan.vandermeer@philips.com wrote:
> Thanks for making the slides available. I can read all of them, except 
> the one on Timed Text, which is corrupted when reading it with my 
> Acrobat Reader 6.0. Acrobat tells it's a font encoding problem. I 
> would appreciate if you could resolve that.

I updated the zip file - maybe it's better now?

Colin


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt





--=_alternative 005B45DCC1256DDB_=
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<br><font size=2 face="sans-serif">Hello Colin,</font>
<br>
<br><font size=2 face="sans-serif">From Jose's latest email:</font>
<br>
<br><font size=2 face="Courier New">&gt; You can download some clarifying slides at<br>
&gt; http://www.pel.panasonic.de/ietf/docs/tt-design.pdf in pdf format or<br>
&gt; from http://www.pel.panasonic.de/ietf/docs/tt-design.ppt in MSPowerPoint<br>
&gt; format.<br>
</font><font size=2 face="sans-serif"><br>
The pdf above does not work for me either, but the ppt does. Perhaps it is possible to upload the original ppt ?</font>
<br><font size=2 face="sans-serif"><br>
Best regards,<br>
<br>
Jan<br>
</font>
<br>
<br>
<br>
<table width=100%>
<tr valign=top>
<td>
<td>
<br>
<br>
<br>
<br>
<br><font size=1 face="sans-serif"><b>Jan vanderMeer</b></font>
<p><font size=1 face="sans-serif">2003-11-11 05:23 PM</font>
<br>
<td>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; To: &nbsp; &nbsp; &nbsp; &nbsp;Colin Perkins &lt;csp@csperkins.org&gt;</font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; cc: &nbsp; &nbsp; &nbsp; &nbsp;avt@ietf.org<br>
avt-admin@ietf.org</font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; Subject: &nbsp; &nbsp; &nbsp; &nbsp;Re: [AVT] Slides for Monday AVT session</font><a href=Notes:///C125672400515F44/38D46BF5E8F08834852564B500129B2C/A190576F657DA492C1256DDB0058E8AE>Link</a>
<p><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; Classification: &nbsp; &nbsp; &nbsp; &nbsp;Unclassified</font>
<br></table>
<br>
<br><font size=2 face="sans-serif">Colin,</font>
<br>
<br><font size=2 face="sans-serif">Thanks for your effort, but unfortunately the timed-text presentation is still corrupted. Because of the problem I installed the latest version of the Acrobat Reader, but that did not help.</font>
<br><font size=2 face="sans-serif"><br>
Best regards,<br>
<br>
Jan<br>
</font>
<br>
<br>
<br>
<table width=100%>
<tr valign=top>
<td>
<td>
<br>
<br>
<br>
<br>
<br><font size=1 face="sans-serif"><b>Colin Perkins &lt;csp@csperkins.org&gt;</b></font>
<p><font size=1 face="sans-serif">Sent by: </font>
<br><font size=1 face="sans-serif">avt-admin@ietf.org</font>
<p><font size=1 face="sans-serif">2003-11-11 05:11 PM</font>
<br>
<td><font size=1 face="Arial">&nbsp; &nbsp; &nbsp; &nbsp; </font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; To: &nbsp; &nbsp; &nbsp; &nbsp;Jan vanderMeer/EHV/CE/PHILIPS@PHILIPS</font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; cc: &nbsp; &nbsp; &nbsp; &nbsp;avt@ietf.org</font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; Subject: &nbsp; &nbsp; &nbsp; &nbsp;Re: [AVT] Slides for Monday AVT session</font>
<p><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; Classification: &nbsp; &nbsp; &nbsp; &nbsp;</font>
<br></table>
<br>
<br>
<br><font size=2 face="Courier New">On Tuesday, Nov 11, 2003, at 05:13 US/Eastern, <br>
jan.vandermeer@philips.com wrote:<br>
&gt; Thanks for making the slides available. I can read all of them, except <br>
&gt; the one on Timed Text, which is corrupted when reading it with my <br>
&gt; Acrobat Reader 6.0. Acrobat tells it's a font encoding problem. I <br>
&gt; would appreciate if you could resolve that.<br>
<br>
I updated the zip file - maybe it's better now?<br>
<br>
Colin<br>
<br>
<br>
_______________________________________________<br>
Audio/Video Transport Working Group<br>
avt@ietf.org<br>
https://www1.ietf.org/mailman/listinfo/avt<br>
</font>
<br>
<br>
<br>
<br>
--=_alternative 005B45DCC1256DDB_=--

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Tue Nov 11 11:53:22 2003
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	 Tue, 11 Nov 2003 17:52:29 +0100
From: "Jose Rey" <rey@panasonic.de>
To: <jan.vandermeer@philips.com>, "Colin Perkins" <csp@csperkins.org>
Cc: <avt@ietf.org>
Subject: AW: [AVT] Slides for Monday AVT session
Date: Tue, 11 Nov 2003 17:52:28 +0100
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Jan,

> You can download some clarifying slides at
> http://www.pel.panasonic.de/ietf/docs/tt-design.pdf in pdf format or
> from http://www.pel.panasonic.de/ietf/docs/tt-design.ppt in MSPowerPoin=
t
> format.

the ppt is also there, as it says above.

cheers,

Jos=E9


  -----Urspr=FCngliche Nachricht-----
  Von: avt-admin@ietf.org [mailto:avt-admin@ietf.org]Im Auftrag von
jan.vandermeer@philips.com
  Gesendet: Dienstag, 11. November 2003 17:35
  An: Colin Perkins
  Cc: avt@ietf.org; avt-admin@ietf.org
  Betreff: Re: [AVT] Slides for Monday AVT session



  Hello Colin,

  From Jose's latest email:

  > You can download some clarifying slides at
  > http://www.pel.panasonic.de/ietf/docs/tt-design.pdf in pdf format or
  > from http://www.pel.panasonic.de/ietf/docs/tt-design.ppt in MSPowerPo=
int
  > format.

  The pdf above does not work for me either, but the ppt does. Perhaps it=
 is
possible to upload the original ppt ?

  Best regards,

  Jan








        Jan vanderMeer
        2003-11-11 05:23 PM


                To:        Colin Perkins <csp@csperkins.org>
                cc:        avt@ietf.org
        avt-admin@ietf.org
                Subject:        Re: [AVT] Slides for Monday AVT sessionLi=
nk
                Classification:        Unclassified




  Colin,

  Thanks for your effort, but unfortunately the timed-text presentation i=
s
still corrupted. Because of the problem I installed the latest version of
the Acrobat Reader, but that did not help.

  Best regards,

  Jan








        Colin Perkins <csp@csperkins.org>
        Sent by:
        avt-admin@ietf.org

        2003-11-11 05:11 PM


                To:        Jan vanderMeer/EHV/CE/PHILIPS@PHILIPS
                cc:        avt@ietf.org
                Subject:        Re: [AVT] Slides for Monday AVT session
                Classification:





  On Tuesday, Nov 11, 2003, at 05:13 US/Eastern,
  jan.vandermeer@philips.com wrote:
  > Thanks for making the slides available. I can read all of them, excep=
t
  > the one on Timed Text, which is corrupted when reading it with my
  > Acrobat Reader 6.0. Acrobat tells it's a font encoding problem. I
  > would appreciate if you could resolve that.

  I updated the zip file - maybe it's better now?

  Colin


  _______________________________________________
  Audio/Video Transport Working Group
  avt@ietf.org
  https://www1.ietf.org/mailman/listinfo/avt






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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Diso-8859-1" =
http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.2919.6307" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN =
class=3D420445016-11112003>Jan,=20
</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D420445016-11112003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN =
class=3D420445016-11112003><FONT=20
face=3D"Courier New" size=3D2>&gt; You can download some clarifying =
slides=20
at<BR>&gt; http://www.pel.panasonic.de/ietf/docs/tt-design.pdf in pdf =
format=20
or<BR>&gt; from http://www.pel.panasonic.de/ietf/docs/tt-design.ppt in=20
MSPowerPoint<BR>&gt; format.</FONT></SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D420445016-11112003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3D"Courier New" size=3D2><SPAN=20
class=3D420445016-11112003>the ppt is also there, as it says=20
above.</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3D"Courier New" size=3D2><SPAN=20
class=3D420445016-11112003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3D"Courier New" size=3D2><SPAN=20
class=3D420445016-11112003>cheers,</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3D"Courier New" size=3D2><SPAN=20
class=3D420445016-11112003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3D"Courier New" size=3D2><SPAN=20
class=3D420445016-11112003>Jos=E9</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3D"Courier New" size=3D2><SPAN=20
class=3D420445016-11112003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3D"Courier New" size=3D2><SPAN=20
class=3D420445016-11112003></SPAN></FONT>&nbsp;</DIV>
<BLOCKQUOTE=20
style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; PADDING-LEFT: =
5px">
  <DIV align=3Dleft class=3DOutlookMessageHeader dir=3Dltr><FONT =
face=3DTahoma=20
  size=3D2>-----Urspr=FCngliche Nachricht-----<BR><B>Von:</B> =
avt-admin@ietf.org=20
  [mailto:avt-admin@ietf.org]<B>Im Auftrag von=20
  </B>jan.vandermeer@philips.com<BR><B>Gesendet:</B> Dienstag, 11. =
November 2003=20
  17:35<BR><B>An:</B> Colin Perkins<BR><B>Cc:</B> avt@ietf.org;=20
  avt-admin@ietf.org<BR><B>Betreff:</B> Re: [AVT] Slides for Monday AVT=20
  session<BR><BR></DIV></FONT><BR><FONT face=3Dsans-serif size=3D2>Hello =

  Colin,</FONT> <BR><BR><FONT face=3Dsans-serif size=3D2>From Jose's =
latest=20
  email:</FONT> <BR><BR><FONT face=3D"Courier New" size=3D2>&gt; You can =
download=20
  some clarifying slides at<BR>&gt;=20
  http://www.pel.panasonic.de/ietf/docs/tt-design.pdf in pdf format =
or<BR>&gt;=20
  from http://www.pel.panasonic.de/ietf/docs/tt-design.ppt in=20
  MSPowerPoint<BR>&gt; format.<BR></FONT><FONT face=3Dsans-serif =
size=3D2><BR>The=20
  pdf above does not work for me either, but the ppt does. Perhaps it is =

  possible to upload the original ppt ?</FONT> <BR><FONT =
face=3Dsans-serif=20
  size=3D2><BR>Best regards,<BR><BR>Jan<BR></FONT><BR><BR><BR>
  <TABLE width=3D"100%">
    <TBODY>
    <TR vAlign=3Dtop>
      <TD>
      <TD><BR><BR><BR><BR><BR><FONT face=3Dsans-serif size=3D1><B>Jan=20
        vanderMeer</B></FONT>=20
        <P><FONT face=3Dsans-serif size=3D1>2003-11-11 05:23 PM</FONT> =
<BR></P>
      <TD><BR><FONT face=3Dsans-serif size=3D1>&nbsp; &nbsp; &nbsp; =
&nbsp; To:=20
        &nbsp; &nbsp; &nbsp; &nbsp;Colin Perkins=20
        &lt;csp@csperkins.org&gt;</FONT> <BR><FONT face=3Dsans-serif =
size=3D1>&nbsp;=20
        &nbsp; &nbsp; &nbsp; cc: &nbsp; &nbsp; &nbsp;=20
        &nbsp;avt@ietf.org<BR>avt-admin@ietf.org</FONT> <BR><FONT=20
        face=3Dsans-serif size=3D1>&nbsp; &nbsp; &nbsp; &nbsp; Subject: =
&nbsp;=20
        &nbsp; &nbsp; &nbsp;Re: [AVT] Slides for Monday AVT =
session</FONT><A=20
        =
href=3D"Notes:///C125672400515F44/38D46BF5E8F08834852564B500129B2C/A19057=
6F657DA492C1256DDB0058E8AE">Link</A>=20

        <P><FONT face=3Dsans-serif size=3D1>&nbsp; &nbsp; &nbsp; &nbsp;=20
        Classification: &nbsp; &nbsp; &nbsp; &nbsp;Unclassified</FONT>=20
    <BR></P></TR></TBODY></TABLE><BR><BR><FONT face=3Dsans-serif=20
  size=3D2>Colin,</FONT> <BR><BR><FONT face=3Dsans-serif size=3D2>Thanks =
for your=20
  effort, but unfortunately the timed-text presentation is still =
corrupted.=20
  Because of the problem I installed the latest version of the Acrobat =
Reader,=20
  but that did not help.</FONT> <BR><FONT face=3Dsans-serif =
size=3D2><BR>Best=20
  regards,<BR><BR>Jan<BR></FONT><BR><BR><BR>
  <TABLE width=3D"100%">
    <TBODY>
    <TR vAlign=3Dtop>
      <TD>
      <TD><BR><BR><BR><BR><BR><FONT face=3Dsans-serif size=3D1><B>Colin =
Perkins=20
        &lt;csp@csperkins.org&gt;</B></FONT>=20
        <P><FONT face=3Dsans-serif size=3D1>Sent by: </FONT><BR><FONT=20
        face=3Dsans-serif size=3D1>avt-admin@ietf.org</FONT>=20
        <P><FONT face=3Dsans-serif size=3D1>2003-11-11 05:11 PM</FONT> =
<BR></P>
      <TD><FONT face=3DArial size=3D1>&nbsp; &nbsp; &nbsp; &nbsp; =
</FONT><BR><FONT=20
        face=3Dsans-serif size=3D1>&nbsp; &nbsp; &nbsp; &nbsp; To: =
&nbsp; &nbsp;=20
        &nbsp; &nbsp;Jan vanderMeer/EHV/CE/PHILIPS@PHILIPS</FONT> =
<BR><FONT=20
        face=3Dsans-serif size=3D1>&nbsp; &nbsp; &nbsp; &nbsp; cc: =
&nbsp; &nbsp;=20
        &nbsp; &nbsp;avt@ietf.org</FONT> <BR><FONT face=3Dsans-serif =
size=3D1>&nbsp;=20
        &nbsp; &nbsp; &nbsp; Subject: &nbsp; &nbsp; &nbsp; &nbsp;Re: =
[AVT]=20
        Slides for Monday AVT session</FONT>=20
        <P><FONT face=3Dsans-serif size=3D1>&nbsp; &nbsp; &nbsp; &nbsp;=20
        Classification: &nbsp; &nbsp; &nbsp; &nbsp;</FONT>=20
  <BR></P></TR></TBODY></TABLE><BR><BR><BR><FONT face=3D"Courier New" =
size=3D2>On=20
  Tuesday, Nov 11, 2003, at 05:13 US/Eastern, =
<BR>jan.vandermeer@philips.com=20
  wrote:<BR>&gt; Thanks for making the slides available. I can read all =
of them,=20
  except <BR>&gt; the one on Timed Text, which is corrupted when reading =
it with=20
  my <BR>&gt; Acrobat Reader 6.0. Acrobat tells it's a font encoding =
problem. I=20
  <BR>&gt; would appreciate if you could resolve that.<BR><BR>I updated =
the zip=20
  file - maybe it's better=20
  =
now?<BR><BR>Colin<BR><BR><BR>____________________________________________=
___<BR>Audio/Video=20
  Transport Working=20
  =
Group<BR>avt@ietf.org<BR>https://www1.ietf.org/mailman/listinfo/avt<BR></=
FONT><BR><BR><BR><BR></BLOCKQUOTE></BODY></HTML>

------=_NextPart_000_0006_01C3A87C.93081F60--


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Thomas Zeng <zeng@PacketVideo.COM>
To: "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>,
        Gamze Seckin
	 <gamze@vidiator.com>
Cc: Alan Tseng <alan@vidiator.com>, IETF AVT WG <avt@ietf.org>
Subject: RE: [AVT] RE: Comments on draft-tseng-rtcp-streaming-extns-00.txt
Date: Tue, 11 Nov 2003 09:09:50 -0800
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Hi Alan and Gamze:

My comments on this draft:

General comment: would it be possible to provide the means for server to
signal to client the interval for Quality Metrics reporting? For instance an
optional parameter in RTSP Play response?

1. Like Magnus, I don't feel "AVP Module" (defined in section 2) is a right
choice of name. I suggest "Transport Handler" might be a better fit in
naming the concept (to differentiate with source codec).

2. Please in the introduction provide some insight of your motivations. If
one implements this draft, would one use it for debugging only, or for
billing also? We'd like to see a generic proposal that will, if possible,
satisfy the desire to do QOS based billing, as well as network monitoring /
user experience monitoring/survey.

3. In section 3.3, it appeared that it is assuming that a RTP packet will
contain only one or partial VOP so that the AVP module could determine the
corruption period from the RTP timestamp.  However, in the multi-VOP RTP
packet, then the corruption period need to contain multi-VOPs.  Just a
suggestion to consider the multi-VOP case.

4. Sometimes player freezes not due to "rebuffering" (underflow of player
buffer).  For example, for live streaming, packets not get sent by the
server due to late arrival from author.  If server drop packets
intentionally, then the player will not detect seq number gap in this case.
There could be other reason that player will freeze not due to rebuffering
(but then jump in play time later).  Will this be included in the
rebuffering matrices?  According to the definition, it should.  So
"rebuffering' in this draft does not always refer to "underflow of player
buffer"?  For instance, we've seen freezing due to overflow (caused by bad
server scheduling) and/or player processor overload.

5.Typo on Page 10, "pts_stop" ...  "beginning" should be "end"

6. On 3.4 last sentence, for rebuffering, then there should not reach end of
playback since the player is not playing and the playtime is stalled.  So we
are  not sure if this sentence applies to rebuffering (at least the
rebuffering due to buffer underflow).  It does have some implications if the
player did not receive any more data and abort due to timeout (i.e., staying
in rebuffering and timeout).  So we think these two cases (normal
rebuffering, or rebuffering with timeout) needs to be considered.




> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Wednesday, November 05, 2003 11:41 PM
> To: Gamze Seckin
> Cc: Alan Tseng; IETF AVT WG
> Subject: Re: [AVT] RE: Comments on 
> draft-tseng-rtcp-streaming-extns-00.txt
> 
> 
> Hi Gamze,
> 
> To my knowledge there is no profile dependencies in RFC 3611. Thus it 
> can be used for any of the four profiles defined or under 
> definition. My 
> comment was to point out this fact and that it seems inappropriate to 
> talk about any special profile in this context.
> 
> cheers
> 
> Magnus
> 
> Gamze Seckin wrote:
> 
> > Hi Magnus,
> > 
> > about your comment
> > "7. Section 2, You talk about AVP, but this extension is 
> not dependent on 
> > any special profile and seems to be compatible with any of the four 
> > available or under development profiles that exist."
> > 
> > Do you also mean AVPF? Since I am confused about the 
> relationship between AVPF and RFC3611.
> > 
> > thx
> > g
> > -----Original Message-----
> > From: Gamze Seckin 
> > Sent: Friday, October 31, 2003 11:08 AM
> > To: Magnus Westerlund; Alan Tseng
> > Cc: IETF AVT WG
> > Subject: [AVT] RE: Comments on 
> draft-tseng-rtcp-streaming-extns-00.txt
> > 
> > 
> > 
> > Hi Magnus, 
> > thanks for the comments.
> > See my response inline.
> > 
> > best regards
> > gamze
> > -----Original Message-----
> > From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> > Sent: Wednesday, October 29, 2003 8:17 AM
> > To: Alan Tseng; Gamze Seckin
> > Cc: IETF AVT WG
> > Subject: Comments on draft-tseng-rtcp-streaming-extns-00.txt
> > 
> > 
> > Hi,
> > 
> > Here is my comments on your draft:
> > 
> > 1. First I think the draft isn't very clear on why any metrics are 
> > needed. In the 3GPP document it only states: "The objective of the 
> > streaming service quality metrics is to offer to the 
> service providers 
> > means to evaluate the end user experience.". This doesn't 
> make it very 
> > clear what the real needs are. I think further developing 
> of this is 
> > needed.
> > 
> > [g]you are right, the draft does not talk about that 
> clearly. We should add that.
> > 
> > 2. Regarding the corruption statistics. For media that uses 
> codec that 
> > have prediction any error normally remains for a rather long time. 
> > However it is very difficult to determine when the effects has been 
> > removed. Therefore this measurement is hard to derive on 
> the receiver 
> > side. Combined with the definition it only provides a good, 
> possibly bad 
> > indication. Also looking at the reasons for corruption, the 
> only real 
> > reason is lost or late packets. A good feeling for the 
> quality can be 
> > determined by the server simply by employing packet loss vector 
> > reporting and knowing when the buffer is empty.
> > 
> > [g]packet loss is definitely a way of estimating 
> corruption, but this gives an algorithm in the decoder (or 
> pre-decoder)a chance to provide better statistics.
> > 
> > 3. The rebuffering measurements makes more sense. However 
> there are at 
> > least one client that does not really perform rebuffering, namely 
> > quicktime. Instead is assumes that if the transmitted bit-rate is 
> > acceptable any short term effects will go away and the 
> client will later 
> >   have a buffer fullness given by the initial buffering delay. The 
> > interesting thing about this measurement is that in 3GPP 
> they already 
> > have a way of letting the server measure when rebuffering 
> takes place, 
> > namely the OBSN reporting specified in TS 26.234 REL 6 
> (only preliminary 
> > versions available).
> > 
> > [g] yes, and I believe those statistics are useful, however 
> they are tight with rate adaptation (only turned on when 
> rate-adaptation is enabled) and tightly coupled with other 
> rtsp messages.
> > 
> > 4. As state privately before this drafts need to be checked 
> against the 
> > ID nits: http://www.ietf.org/ID-nits.html
> > 
> > [g] yes, we should definitely do that thanks.
> > 
> > 5. Section 1. The introduction is basically the same as the 
> > introduction. I would recommend you to write a better more 
> informative 
> > introduction.
> > 
> > [g] agreed.
> > 
> > 6. Section 1. Last sentence: "As all the other report blocks
> > defined in [1], the streaming report blocks are applicable to
> > one-to-one or one-to-many applications that generate
> > RTCP SR or RR reports." This sentence is not really 
> correct. The use of 
> > the RTCP XR reports is possible in any application sending RTCP.
> > 
> > [g] himm,i am confused i thought that the first sentence 
> covers everything - what is left out? 
> > So [1] needs to be updated too?
> > 
> > 7. Section 2, You talk about AVP, but this extension is not 
> dependent on 
> > any special profile and seems to be compatible with any of the four 
> > available or under development profiles that exist.
> > 
> > [g] agreed.
> > 
> > 8. Section 2, Figure 1. This figure shows this report 
> formats potential 
> > need to have reporting from all layers on the receiving 
> side. This might 
> > be a problem for certain application architectures. I am 
> not saying that 
> > it can't be used, its only that some applications may have 
> difficulties 
> > to include your reports.
> > 
> > [g] we believe the statistics reported should be extensible 
> also should be allowed to be ignored. 
> > So if an end side cannot support it should put simply a 
> zero - what do you think?
> > 
> > 9. Section 3.1, first paragraph: "All timestamps reported by the XR 
> > packet SHALL be RTP timestamps. All durations SHALL be in 
> units of the 
> > RTP timebase." This definition is not clear, because in 
> applications 
> > allowing for switching between payload types the rate can change. I 
> > think that the intention of using the RTP timestamp value 
> for indication 
> > over what interval which the report covers is a good one. 
> However some 
> > more consideration over formulation must be done.
> > 
> > [g] valid point - I guess we need to think more about the 
> corner cases.
> > 
> > 10. Section 3.3, PAge 5, first bullet after figure 2:
> > "- at the AVP module: Packet drop and packet errors can sometimes be
> > detected by incorrect or missing sequence number or 
> checksum. " Do you 
> > really need to consider bit-errors, the only allowed 
> possibility for 
> > bit-errors in RTP is if partial checksums are used. 
> Bit-errors surviving 
> > to the transport layer is the meaning to be caught by the 
> UDP, etc., 
> > checksum.
> > 
> > [g] there could be cases where bit errors are propagated 
> (e.g. udp lite).
> > 
> > 11. Section 6. The streaming metrics should probably have a 
> value to 
> > combine with the parameter to indicate how often the report 
> block should 
> > be included. This seem to be a block that is not needed very often.
> > 
> > [g] valid point, note taken.
> > 
> > 12. Section 8. The security section does not look 
> sufficient. I guess 
> > that the consideration for this is rather similar to the 
> VoIP block. 
> > However I think there should be made some effort to 
> indicate the main 
> > security concerns also in this document.
> > 
> > [g] agreed.
> > 
> > 13. Section 12. The references need to be divided into 
> Normative and 
> > Informative ones.
> > 
> > [g] agreed.
> > 
> > Cheers
> > 
> > Magnus Westerlund
> > 
> > Multimedia Technologies, Ericsson Research EAB/TVA/A
> > 
> ----------------------------------------------------------------------
> > Ericsson AB                | Phone +46 8 4048287
> > Torshamsgatan 23           | Fax   +46 8 7575550
> > S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> > 
> > 
> > 
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www1.ietf.org/mailman/listinfo/avt
> > 
> 
> -- 
> 
> Magnus Westerlund
> 
> Multimedia Technologies, Ericsson Research EAB/TVA/A
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> 
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
> 

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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At 17:27 +0100 11/11/03, Jose Rey wrote:
>Hi all,
>
>for those that did not attend the AVT session on Monday, there was a
>number of comments and useful feedback during the timed text
>presentation.
>
>In particular, it was suggested that the draft sticks too much to the
>3gp file format itself and does not look at the particularities of the
>RTP transport (Colin please correct me if the language is wrong).
>Currently the draft just reads the text samples from the 3gp file and
>puts them into RTP packets. If a packet is too big for one RTP packet
>then it is cut and a fragment header is used to be able to interpret the
>information in that fragment. As long as fragmentation happens rarely
>then this could be the best option.
>
>On the other side, there's the option (which I call 'pure ALF approach'
>in the slides) to preprocess the data read from the 3gp file before
>packetising. This preprocessing would not only cut the text samples to
>fix the RTP packet size but also re-arrange the packets so that the
>fragments are not only self-contained but contain all the decorations.
>This approach requires to define extra headers for the cases in which
>the modifier boxes would have to be split into several RTP packets.
>
>You can download some clarifying slides at
>http://www.pel.panasonic.de/ietf/docs/tt-design.pdf in pdf format or
>from http://www.pel.panasonic.de/ietf/docs/tt-design.ppt in MSPowerPoint
>format.
>
>The current approach makes sense if the text samples are regularly small
>and one or several of them fit into an RTP packet. Fragmentation would
>happen seldom. The 'pure alf approach' is, in general, the most correct
>one, and so also the one that requires more header definitions. If
>fragmentation is a rare event, this overhead may be not worth (?).  The
>question to the group is pretty simple: which approach should be
>pursued?

I think we can find a middle ground.  I think=20
that text samples over 512 bytes are quite rare,=20
and over 1400 bytes very rare indeed.  I don't=20
mind going to a format which is less file-format=20
centric, but I don't think we should spend a lot=20
of time optimizing fragmented text access units.

>
>
>We appreciate your feedback,
>
>cheers,
>
>
>Jose
>
>
>
>
>
>>  -----Urspr=FCngliche Nachricht-----
>>  Von: avt-admin@ietf.org [mailto:avt-admin@ietf.org]Im Auftrag von Jos=E9
>>  Rey
>>  Gesendet: Montag, 10. November 2003 21:02
>>  An: jan.vandermeer@philips.com; 'IETF AVT WG'
>>  Cc: Colin Perkins; Magnus Westerlund; 'Dave Singer'; Yoshinori Matsui
>>  Betreff: AW: [AVT] More comments on
>>  draft-rey-avt-rtp-3gpp-timed-text-01.txt
>>
>>
>>
>>  Jan, all,
>>
>>
>>  > -----Urspr=FCngliche Nachricht-----
>>  > Von: Jos=E9 Rey [mailto:rey@panasonic.de]
>>  > Gesendet: Montag, 10. November 2003 18:11
>>  > An: Jose Rey
>>  > Betreff:
>>  >
>>  >
>>  > Dear Jose, all,
>>  >
>>  > I will not be in Minneapolis either but would like to provide
>>  > some input
>>  > on the issue of transport of 3GPP timed text. I know this
>>  > input is very
>>  > late; my appologies for that.
>>  >
>>  > As you know also MPEG is working on timed text; currently
>>  the Committe
>>  > Draft (CD) is ballotted, and MPEG schedules to enter the
>>  next phase in
>>  > about one month and to finish the work in July 2004. The
>>  > end-result will
>>  > be part 17 of MPEG-4, defining MPEG-4 text streams. Through
>>  liaisons,
>>  > MPEG tries to harmonize its work with the work that is
>>  already carried
>>  > out in 3GPP and also with ongoing work in W3C. Most
>>  important in this
>>  > respect is that MPEG focusses at maximum commonality with the
>>  > timed text
>>  > format defined by 3GPP for download. In summary, the current CD of
>>  > MPEG-4 part 17 specifies streaming of the 3GPP timed text format by
>>  > defining a very straightforward way to transcode the 3GPP
>>  format into
>>  > the MPEG-4 structure of a decoder configuration and text
>  > access units.
>>  > Please find the Committee Draft (a public document) attached to this
>>  > email. So far, 3GPP expressed through liaison statements to be very
>  > > pleased with the work done by MPEG.
>>  >
>>  > Once the work is finished, the timed text format can be
>>  streamed using
>>  > "RFC#simple", draft-ietf-avt-mpeg4-simple-08.txt, that is
>>  scheduled to
>>  > be published as RFC very soon. In summary this approach
>>  would work as
>>  > follows:
>>  > - media type: "video",
>>  > - MIME subtype name: mpeg4-generic,
>>  > - streamType (required parameter in RFC#simple): indicates
>>  MPEG-4 text
>>  > stream as defined in ISO 14496-17,
>>  > - profile-level-id (required parameter in RFC#simple): specifies the
>>  > functional subset as defined in the current 3GPP timed text
>>  > specification,
>>  > - config (required parameter in RFC#simple): carries the decoder
>>  > configuration specified for text streams in ISO 14496-17,
>>  > - the RTP payload contains either (a) one fragment of a text
>>  > access unit
>>  > or (b) one or more complete text access units,
>>  > - the RTP payload carries one AU-header for each carried text access
>>  > unit,
>>  > - in the RTP payload the concatenated AU-headers preceed
>>  concatenated
>>  > data of text access units, as defined in RFC#simple,
>>  > - each AU-header contains (at least) the following
>>  information on the
>>  > associated text access unit: (a) the size of the text
>>  access unit, (b)
>>  > the time stamp of the text access unit,
>>  > - if deemed necessary, the AU-header may also carry the index
>>  > of a text
>>  > access unit (fragment) and/or an indication whether the
>>  > associated text
>>  > access unit is a random access point.
>>  > In RFC#simple the fields in the AU-header are configurable, but if
>>  > desired one or two new modes for RFC#simple could be defined
>>  > with fixed
>>  > length fields, for example one using an 8 bit AU-size field
>>  (for most
>>  > common applications) and another using 16 bits AU-size field for
>>  > applications using (very) large text access units.
>>  >
>>  > The above "RFC#simple approach" has a high level of commonality with
>>  > draft-rey-avt-rtp-3gpp-timed-text-01.txt. There are some small
>>  > differences that can be fixed either way, but roughly the
>>  "RFC#simple
>>  > approach" is a simplified version of
>>  > draft-rey-avt-rtp-3gpp-timed-text-01.txt. In particular, the
>>  > "RFC#simple" approach does not have the advanced error resiliency
>>  > measures that are in draft-rey-avt-rtp-3gpp-timed-text-01.txt.
>>  >
>>  > In my view not only in IETF, but also in 3GPP and in MPEG a
>>  number of
>>  > basic issues should be discussed, such as:
>>  > - are the error resiliency features of
>>  > draft-rey-avt-rtp-3gpp-timed-text-01.txt sufficiently valuable to
>>  > justify their complexity (note that a partly corrupted
>>  > subtitle may not
>>  > have much more value to the user than a missing one);
>>
>>  This is a good question which I will also ask during my
>>  presentation on
>>  Monday.  This is a question which certainly needs feedback of
>>  the group.
>>  The complexity can be reduced, we have to agree where to cut
>>  down: do we
>>  need the MTSF headers? or do we want to just keep resiliency for plain
>>  text fragments?, for example.  PT multiplexing will probably be
>>  abandoned for a simpler scheme.
>>
>>  > - is it desirable to define a 3GPP timed text transport
>>  solution based
>>  > on draft-rey-avt-rtp-3gpp-timed-text-01.txt next to the one
>>  > that will be
>>  > available based on RFC#simple;
>>
>>  Allowing both would mean that one does not have to implement
>>  RFC#simple
>>  in order to transport timed text, which I think is desireable.
>>
>>  > - is it desirable that draft-rey-avt-rtp-3gpp-timed-text-01.txt
>>  > specifies a mode or an extension of RFC#simple;
>>
>>  Would this mean that the RFC#simple has to be implemented to use the
>>  draft? I think this is not a good idea.
>>
>>  > - how can a timing conflict (if any) between release 6 of
>>  > 3GPP and part
>>  > 17 of MPEG-4 be resolved (note that such conflict will only exist if
>>  > 3GPP adopts streaming of timed text as a work item and if
>>  3GPP decides
>>  > that this functionality has to be in release 6).
>  > >
>>
>>  We are targetting to include this in Release 6 (03/04). According to
>>  your information above MPEG4 timed text will be ready in
>  > July'04, which
>>  is out of the deadline. Apart from that, current Release 6
>>  does not use
>>  the RFC#simple for streaming... so one would not be able to use timed
>>  text in Release 6.
>>
>>  Assuming timed text comes into the Release 6 agenda, what I would like
>>  to know is what should, in your opinion, the current draft include in
>>  order to consider the MPEG work and the possible extensions it may
>>  incorporate ?. It is now the time to talk about these issues if the
>>  draft shall consider the MPEG work.
>>
>>
>>  > I hope you will have a fruitful discussion on this subject in
>>  > Mineapolis.
>>
>>  You can dowload the session slides and take  a look at the
>>  presentation
>>  which is related to this email.
>>
>>  cheers,
>>
>>  Jose
>>
>>  >But, this is not just an IETF issue. Therefore,
>>  > as both 3GPP
>>  > and MPEG are main "customers" of a timed text streaming solution, I
>>  > suggest that IETF requests 3GPP and MPEG to advice IETF as
>>  to the most
>>  > desirable direction for resolving this issue. Note that
>>  both 3GPP and
>>  > MPEG will have meetings where this issue can be discussed
>>  well before
>>  > the end of this year.
>>  >
>>  > Best regards,
>>  >
>>  > Jan van der Meer
>>  >
>>  >
>>
>>
>>  _______________________________________________
>>  Audio/Video Transport Working Group
>>  avt@ietf.org
>>  https://www1.ietf.org/mailman/listinfo/avt


-- 
David Singer
Apple Computer/QuickTime

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Tue Nov 11 16:05:34 2003
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Date: Tue, 11 Nov 2003 20:57:13 +0000
From: Julian Chesterfield <julian@thechesterfields.org>
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Subject: [AVT] Summary of discussion items for draft-ietf-avt-rtcpssm
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Joerg will be presenting an overview of updates and open issues for 
draft-ietf-avt-rtcpssm at the meeting tomorrow. Here's an outline of 
changes to the draft since last time:

- draft-ietf-avt-rtcpssm current version -05

- Updated refs into normative vs informative.

- Added clarifiying text for each summary type to explain what 
information is provided.

- Clarified bandwidth allocation rules for source in reflection vs 
summarised modes. In summary mode, the sender will use the receiver's 
bandwidth for the feedback channel for summaries.

- 64 bit timestamp used in RSI header.

Open issue:
Security considerations section needs slimming down, it is currently a 
little too detailed. We've tried to present info as a 'cookbook' style 
approach to security components complying to the requirements. We'll 
need to determine what level of information is required to explain the 
motivation for the security requirements, and whether we can cut down 
some of the detail e.g. attack vulnerabilities, types of attack etc.. So 
feedback form the group on this issue is particularly solicited.

Otherwise, we hope to have any final updates/requests into the draft by 
december for a final call version.

Many thanks,

Julian Chesterfield


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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The slides for Wednesday are now available at
   http://www.dcs.gla.ac.uk/~csp/ietf58-avt-wed.zip

Colin


On Monday, Nov 10, 2003, at 10:14 US/Central, Colin Perkins wrote:
> Copies of the presentation materials which will be used in AVT this 
> afternoon can be retrieved as 
> http://www.dcs.gla.ac.uk/~csp/ietf58-avt-mon.zip
>
> These will migrate onto the IETF server after the meeting.
>
> Colin


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Hi,

some comments on the draft.

1. Overall the draft is interesting.

2. I support most of the comments already given by Magnus and Thomas. In =
particular, I would like to stress the fact that the concept of =
corruption is fuzzy. For example a data loss of 1 second does not =
necessarily map to a subjective quality degradation of 1 second. It =
might be shorter or longer, depending on how error concealment is done. =
Especially for video, this is a complicated issue. And yet one aspect =
where the complexity relies, is the difficulty of extracting the needed =
information from the decoder, either because of lack of interfaces or =
because it is computationally intensive to perform this task.=20

Regards,

Igor


=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
Igor D.D. Curcio
Nokia Corporation
Mobile Software, Architecture & Roadmapping
Multimedia streaming and video telephony
P.O. Box 88
33721 Tampere
Finland
=20
Tel. +358 800 8000
Email: igor.curcio@nokia.com
WWW  www.nokia.com


> -----Original Message-----
> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of ext
> Thomas Zeng
> Sent: 11 November, 2003 19:10
> To: 'Magnus Westerlund'; Gamze Seckin
> Cc: Alan Tseng; IETF AVT WG
> Subject: RE: [AVT] RE: Comments on
> draft-tseng-rtcp-streaming-extns-00.txt
>=20
>=20
> Hi Alan and Gamze:
>=20
> My comments on this draft:
>=20
> General comment: would it be possible to provide the means=20
> for server to
> signal to client the interval for Quality Metrics reporting?=20
> For instance an
> optional parameter in RTSP Play response?
>=20
> 1. Like Magnus, I don't feel "AVP Module" (defined in section=20
> 2) is a right
> choice of name. I suggest "Transport Handler" might be a better fit in
> naming the concept (to differentiate with source codec).
>=20
> 2. Please in the introduction provide some insight of your=20
> motivations. If
> one implements this draft, would one use it for debugging only, or for
> billing also? We'd like to see a generic proposal that will,=20
> if possible,
> satisfy the desire to do QOS based billing, as well as=20
> network monitoring /
> user experience monitoring/survey.
>=20
> 3. In section 3.3, it appeared that it is assuming that a RTP=20
> packet will
> contain only one or partial VOP so that the AVP module could=20
> determine the
> corruption period from the RTP timestamp.  However, in the=20
> multi-VOP RTP
> packet, then the corruption period need to contain multi-VOPs.  Just a
> suggestion to consider the multi-VOP case.
>=20
> 4. Sometimes player freezes not due to "rebuffering"=20
> (underflow of player
> buffer).  For example, for live streaming, packets not get sent by the
> server due to late arrival from author.  If server drop packets
> intentionally, then the player will not detect seq number gap=20
> in this case.
> There could be other reason that player will freeze not due=20
> to rebuffering
> (but then jump in play time later).  Will this be included in the
> rebuffering matrices?  According to the definition, it should.  So
> "rebuffering' in this draft does not always refer to=20
> "underflow of player
> buffer"?  For instance, we've seen freezing due to overflow=20
> (caused by bad
> server scheduling) and/or player processor overload.
>=20
> 5.Typo on Page 10, "pts_stop" ...  "beginning" should be "end"
>=20
> 6. On 3.4 last sentence, for rebuffering, then there should=20
> not reach end of
> playback since the player is not playing and the playtime is=20
> stalled.  So we
> are  not sure if this sentence applies to rebuffering (at least the
> rebuffering due to buffer underflow).  It does have some=20
> implications if the
> player did not receive any more data and abort due to timeout=20
> (i.e., staying
> in rebuffering and timeout).  So we think these two cases (normal
> rebuffering, or rebuffering with timeout) needs to be considered.
>=20
>=20
>=20
>=20
> > -----Original Message-----
> > From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> > Sent: Wednesday, November 05, 2003 11:41 PM
> > To: Gamze Seckin
> > Cc: Alan Tseng; IETF AVT WG
> > Subject: Re: [AVT] RE: Comments on=20
> > draft-tseng-rtcp-streaming-extns-00.txt
> >=20
> >=20
> > Hi Gamze,
> >=20
> > To my knowledge there is no profile dependencies in RFC=20
> 3611. Thus it=20
> > can be used for any of the four profiles defined or under=20
> > definition. My=20
> > comment was to point out this fact and that it seems=20
> inappropriate to=20
> > talk about any special profile in this context.
> >=20
> > cheers
> >=20
> > Magnus
> >=20
> > Gamze Seckin wrote:
> >=20
> > > Hi Magnus,
> > >=20
> > > about your comment
> > > "7. Section 2, You talk about AVP, but this extension is=20
> > not dependent on=20
> > > any special profile and seems to be compatible with any=20
> of the four=20
> > > available or under development profiles that exist."
> > >=20
> > > Do you also mean AVPF? Since I am confused about the=20
> > relationship between AVPF and RFC3611.
> > >=20
> > > thx
> > > g
> > > -----Original Message-----
> > > From: Gamze Seckin=20
> > > Sent: Friday, October 31, 2003 11:08 AM
> > > To: Magnus Westerlund; Alan Tseng
> > > Cc: IETF AVT WG
> > > Subject: [AVT] RE: Comments on=20
> > draft-tseng-rtcp-streaming-extns-00.txt
> > >=20
> > >=20
> > >=20
> > > Hi Magnus,=20
> > > thanks for the comments.
> > > See my response inline.
> > >=20
> > > best regards
> > > gamze
> > > -----Original Message-----
> > > From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> > > Sent: Wednesday, October 29, 2003 8:17 AM
> > > To: Alan Tseng; Gamze Seckin
> > > Cc: IETF AVT WG
> > > Subject: Comments on draft-tseng-rtcp-streaming-extns-00.txt
> > >=20
> > >=20
> > > Hi,
> > >=20
> > > Here is my comments on your draft:
> > >=20
> > > 1. First I think the draft isn't very clear on why any=20
> metrics are=20
> > > needed. In the 3GPP document it only states: "The=20
> objective of the=20
> > > streaming service quality metrics is to offer to the=20
> > service providers=20
> > > means to evaluate the end user experience.". This doesn't=20
> > make it very=20
> > > clear what the real needs are. I think further developing=20
> > of this is=20
> > > needed.
> > >=20
> > > [g]you are right, the draft does not talk about that=20
> > clearly. We should add that.
> > >=20
> > > 2. Regarding the corruption statistics. For media that uses=20
> > codec that=20
> > > have prediction any error normally remains for a rather=20
> long time.=20
> > > However it is very difficult to determine when the=20
> effects has been=20
> > > removed. Therefore this measurement is hard to derive on=20
> > the receiver=20
> > > side. Combined with the definition it only provides a good,=20
> > possibly bad=20
> > > indication. Also looking at the reasons for corruption, the=20
> > only real=20
> > > reason is lost or late packets. A good feeling for the=20
> > quality can be=20
> > > determined by the server simply by employing packet loss vector=20
> > > reporting and knowing when the buffer is empty.
> > >=20
> > > [g]packet loss is definitely a way of estimating=20
> > corruption, but this gives an algorithm in the decoder (or=20
> > pre-decoder)a chance to provide better statistics.
> > >=20
> > > 3. The rebuffering measurements makes more sense. However=20
> > there are at=20
> > > least one client that does not really perform rebuffering, namely=20
> > > quicktime. Instead is assumes that if the transmitted bit-rate is=20
> > > acceptable any short term effects will go away and the=20
> > client will later=20
> > >   have a buffer fullness given by the initial buffering=20
> delay. The=20
> > > interesting thing about this measurement is that in 3GPP=20
> > they already=20
> > > have a way of letting the server measure when rebuffering=20
> > takes place,=20
> > > namely the OBSN reporting specified in TS 26.234 REL 6=20
> > (only preliminary=20
> > > versions available).
> > >=20
> > > [g] yes, and I believe those statistics are useful, however=20
> > they are tight with rate adaptation (only turned on when=20
> > rate-adaptation is enabled) and tightly coupled with other=20
> > rtsp messages.
> > >=20
> > > 4. As state privately before this drafts need to be checked=20
> > against the=20
> > > ID nits: http://www.ietf.org/ID-nits.html
> > >=20
> > > [g] yes, we should definitely do that thanks.
> > >=20
> > > 5. Section 1. The introduction is basically the same as the=20
> > > introduction. I would recommend you to write a better more=20
> > informative=20
> > > introduction.
> > >=20
> > > [g] agreed.
> > >=20
> > > 6. Section 1. Last sentence: "As all the other report blocks
> > > defined in [1], the streaming report blocks are applicable to
> > > one-to-one or one-to-many applications that generate
> > > RTCP SR or RR reports." This sentence is not really=20
> > correct. The use of=20
> > > the RTCP XR reports is possible in any application sending RTCP.
> > >=20
> > > [g] himm,i am confused i thought that the first sentence=20
> > covers everything - what is left out?=20
> > > So [1] needs to be updated too?
> > >=20
> > > 7. Section 2, You talk about AVP, but this extension is not=20
> > dependent on=20
> > > any special profile and seems to be compatible with any=20
> of the four=20
> > > available or under development profiles that exist.
> > >=20
> > > [g] agreed.
> > >=20
> > > 8. Section 2, Figure 1. This figure shows this report=20
> > formats potential=20
> > > need to have reporting from all layers on the receiving=20
> > side. This might=20
> > > be a problem for certain application architectures. I am=20
> > not saying that=20
> > > it can't be used, its only that some applications may have=20
> > difficulties=20
> > > to include your reports.
> > >=20
> > > [g] we believe the statistics reported should be extensible=20
> > also should be allowed to be ignored.=20
> > > So if an end side cannot support it should put simply a=20
> > zero - what do you think?
> > >=20
> > > 9. Section 3.1, first paragraph: "All timestamps reported=20
> by the XR=20
> > > packet SHALL be RTP timestamps. All durations SHALL be in=20
> > units of the=20
> > > RTP timebase." This definition is not clear, because in=20
> > applications=20
> > > allowing for switching between payload types the rate can=20
> change. I=20
> > > think that the intention of using the RTP timestamp value=20
> > for indication=20
> > > over what interval which the report covers is a good one.=20
> > However some=20
> > > more consideration over formulation must be done.
> > >=20
> > > [g] valid point - I guess we need to think more about the=20
> > corner cases.
> > >=20
> > > 10. Section 3.3, PAge 5, first bullet after figure 2:
> > > "- at the AVP module: Packet drop and packet errors can=20
> sometimes be
> > > detected by incorrect or missing sequence number or=20
> > checksum. " Do you=20
> > > really need to consider bit-errors, the only allowed=20
> > possibility for=20
> > > bit-errors in RTP is if partial checksums are used.=20
> > Bit-errors surviving=20
> > > to the transport layer is the meaning to be caught by the=20
> > UDP, etc.,=20
> > > checksum.
> > >=20
> > > [g] there could be cases where bit errors are propagated=20
> > (e.g. udp lite).
> > >=20
> > > 11. Section 6. The streaming metrics should probably have a=20
> > value to=20
> > > combine with the parameter to indicate how often the report=20
> > block should=20
> > > be included. This seem to be a block that is not needed=20
> very often.
> > >=20
> > > [g] valid point, note taken.
> > >=20
> > > 12. Section 8. The security section does not look=20
> > sufficient. I guess=20
> > > that the consideration for this is rather similar to the=20
> > VoIP block.=20
> > > However I think there should be made some effort to=20
> > indicate the main=20
> > > security concerns also in this document.
> > >=20
> > > [g] agreed.
> > >=20
> > > 13. Section 12. The references need to be divided into=20
> > Normative and=20
> > > Informative ones.
> > >=20
> > > [g] agreed.
> > >=20
> > > Cheers
> > >=20
> > > Magnus Westerlund
> > >=20
> > > Multimedia Technologies, Ericsson Research EAB/TVA/A
> > >=20
> >=20
> ----------------------------------------------------------------------
> > > Ericsson AB                | Phone +46 8 4048287
> > > Torshamsgatan 23           | Fax   +46 8 7575550
> > > S-164 80 Stockholm, Sweden | mailto:=20
> magnus.westerlund@ericsson.com
> > >=20
> > >=20
> > >=20
> > > _______________________________________________
> > > Audio/Video Transport Working Group
> > > avt@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/avt
> > >=20
> >=20
> > --=20
> >=20
> > Magnus Westerlund
> >=20
> > Multimedia Technologies, Ericsson Research EAB/TVA/A
> >=20
> ----------------------------------------------------------------------
> > Ericsson AB                | Phone +46 8 4048287
> > Torshamsgatan 23           | Fax   +46 8 7575550
> > S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> >=20
> >=20
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www1.ietf.org/mailman/listinfo/avt
> >=20
>=20
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
>=20

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Please allow me to make more clear my points expressed in the final =
minutes of the AVT WG meeting today related to =
http://www.ietf.cnri.reston.va.us/internet-drafts/draft-clark-avt-rtpmibv=
2-00.txt

1. This MIB includes extensions to the RTP MIB (RFC 2959) related to =
RTCP-XR (RFC 3611). At the same time, there seems to be a need to update =
RFC 2959 as result of the evolution of the Real Time Protocol from RFC =
1889 to RFC 3550. I strongly suggest to do this in two separate MIBs =
separate documents:
- a new RTP MIB, updating to RFC 2959
- a supplementary RTCP-XR MIB, with the MIB objects relevant for hosts =
or monitors running RTCP-XR.=20
This more modular and scalable structure would allow the two different =
MIBs to be progressed separately, without creating a tight dependency =
among them, and would allow adding other modules in the future.
2. There is a need for a better coordination, and possibly some =
consolidation between the RTCP-XR MIB and work being done in other WGs =
in the IETF - especially the RMON MIB WG =
(http://www.ietf.cnri.reston.va.us/html.charters/rmonmib-charter.html) =
work on Application Performance Monitoring and Real-Time Application =
Monitoring to create a consistent framework for performance monitoring, =
and avoid duplication. While RTCP-XR covers an important aspect of media =
monitoring when RTP is used, it does not cover the whole application =
space where RTP streams can be used. I would be glad to provide more =
information, or work with the authors of the different proposals in AVT =
to try to reach this goal.=20

Regards,

Dan
=20

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<TITLE>RTP MIB Extensions</TITLE>
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<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">Please allow me to make more =
clear my points expressed in the final minutes of the AVT WG meeting =
today related to <A =
HREF=3D"http://www.ietf.cnri.reston.va.us/internet-drafts/draft-clark-avt=
-rtpmibv2-00.txt">http://www.ietf.cnri.reston.va.us/internet-drafts/draft=
-clark-avt-rtpmibv2-00.txt</A></FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">1. This MIB includes =
extensions to the RTP MIB (RFC 2959) related to RTCP-XR (RFC 3611). At =
the same time, there seems to be a need to update RFC 2959 as result of =
the evolution of the Real Time Protocol from RFC 1889 to RFC 3550. I =
strongly suggest to do this in two separate MIBs separate =
documents:</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">- a new RTP MIB, updating to =
RFC 2959</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">- a supplementary RTCP-XR =
MIB, with the MIB objects relevant for hosts or monitors running =
RTCP-XR. </FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">This more modular and =
scalable structure would allow the two different MIBs to be progressed =
separately, without creating a tight dependency among them, and would =
allow adding other modules in the future.</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">2. There is a need for a =
better coordination, and possibly some consolidation between the RTCP-XR =
MIB and work being done in other WGs in the IETF - especially the RMON =
MIB WG (<A =
HREF=3D"http://www.ietf.cnri.reston.va.us/html.charters/rmonmib-charter.h=
tml">http://www.ietf.cnri.reston.va.us/html.charters/rmonmib-charter.html=
</A>) work on Application Performance Monitoring and Real-Time =
Application Monitoring to create a consistent framework for performance =
monitoring, and avoid duplication. While RTCP-XR covers an important =
aspect of media monitoring when RTP is used, it does not cover the whole =
application space where RTP streams can be used. I would be glad to =
provide more information, or work with the authors of the different =
proposals in AVT to try to reach this goal. </FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">Regards,</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">Dan</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">&nbsp;</FONT></P>

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From: "Lars-Erik Jonsson (LU/EAB)" <lars-erik.jonsson@ericsson.com>
To: avt@ietf.org
Date: Wed, 12 Nov 2003 20:45:12 +0100
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Subject: [AVT] HC over multihop / MPLS
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As I promised in the WG session, I'll try to share my thoughts on
the HC over multihop/MPLS issue.

The AVT charter states the following as a WG task:=20
"to review the applicability of Compressed RTP operation on
  MPLS networks, developing extensions as necessary"

With the following milestones:
Nov 03    Initial draft requirements for ECRTP over MPLS; discuss with =
MPLS WG
Nov 03    Initial draft requirements for ECRTP over MPLS; discuss with =
MPLS WG

However, the current draft refers to "VoIP Header Compression over =
Multi-Hop
Paths", which is not the same thing. The draft clearly shows that =
compression
over multihop paths is one thing, and compression over MPLS another. =
Further,
the draft takes a wider scope, and talks about header compression =
generally,
not only ECRTP as the charter says.

The following observations can be made:
* If work is to done for header compression over multihop and/or MPLS, =
it
  might make sense to not only address ECRTP.
* The only work in this that naturally could belong in the AVT WG would =
be
  to make sure ECRTP would work in these scenarios. In my =
understanding,
  this should already be the case with ECRTP (as well as some ROHC =
profiles),
  which is also indicated by the multihop req draft.
* Header compression over multihop paths [ described by the multihop =
req
  draft as where one wants to avoid decompression/compression steps at
  intermediate routers if the same compression scheme is supported at =
both
  incoming and outgoing interfaces ] is fully possible also without any =
new
  standards, as it would only be an implementation issue for each =
intermediate
  router to bind an incoming compressed stream to its corresponding =
outgoing
  compressed stream. In a path A->B->C, where B is the intermediate =
router
  negotiating use of e.g. CRTP with both A and C, it is fully up to the
  implementation of B to avoid unnecessary decompression/compression =
and
  just forward compressed packets as-is ("CID routing").
* The only significant standardization work I can see is to define how =
to
  do header compression (e.g. IPHC, CRTP/ECRTP, ROHC) over MPLS, which =
is
  probably a work that would require some coordination with AVT and =
ROHC
  individuals, but the core of that work does not naturally belong to
  ROHC or AVT. Would not the MPLS WG be a more proper place?

Just a few thoughts, let's get discussions going!
/L-E=20


-----------------------------------
Lars-Erik Jonsson, M.Sc
Wireless IP Optimizations
AWARE - Advanced Wireless Algorithm Research
Ericsson Research, Corporate Unit
Ericsson AB
Box 920, S-971 28 Lule=E5, Sweden
E-mail: lars-erik.jonsson@ericsson.com
Phone: +46 920 20 21 07
Fax: +46 920 20 20 99
Home: +46 920 999 57

My opinions are my personal opinions and should not be considered
as the opinions of my employer, if not explicitly stated.=20

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Wed Nov 12 15:51:13 2003
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From: "Phelan, Tom" <tphelan@sonusnet.com>
To: "'avt@ietf.org'" <avt@ietf.org>
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Subject: [AVT] DCCP for media applications
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Hi,

Judging from the discussion that just finished, it seems that the AVT group
has a reasonable interest in the issues involved in using DCCP for media
applications.  I encourage the people here to look at
http://www.phelan-4.com/dccp/draft-phelan-dccp-media-00.txt, and to comment
on the issues it describes.  This draft is likely to be the basis for a
forthcoming DCCP User Guide, and application expertise is needed to be sure
that we've covered the issues.

Thanks,
Tom Phelan

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Alan Clark" <alan@telchemy.com>
To: "Romascanu, Dan \(Dan\)" <dromasca@avaya.com>, <avt@ietf.org>
Cc: <mankin@psg.com>, "Bert Wijnen \(E-mail\)" <bwijnen@lucent.com>
Subject: RE: [AVT] RTP MIB Extensions
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RTP MIB ExtensionsDan

I have no issue with the idea of progressing updates to the RTP MIB and
keeping the RTCP XR MIB separate - I'll update the draft accordingly.  There
is also no disagreement that the RTCP XR VoIP metrics only address VoIP
whereas the envisaged scope of RAQMON is broader.

Regards

Alan Clark
Telchemy


  -----Original Message-----
  From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of
Romascanu, Dan (Dan)
  Sent: Wednesday, November 12, 2003 2:40 PM
  To: avt@ietf.org
  Cc: mankin@psg.com; Bert Wijnen (E-mail)
  Subject: [AVT] RTP MIB Extensions


  Please allow me to make more clear my points expressed in the final
minutes of the AVT WG meeting today related to
http://www.ietf.cnri.reston.va.us/internet-drafts/draft-clark-avt-rtpmibv2-0
0.txt

  1. This MIB includes extensions to the RTP MIB (RFC 2959) related to
RTCP-XR (RFC 3611). At the same time, there seems to be a need to update RFC
2959 as result of the evolution of the Real Time Protocol from RFC 1889 to
RFC 3550. I strongly suggest to do this in two separate MIBs separate
documents:

  - a new RTP MIB, updating to RFC 2959

  - a supplementary RTCP-XR MIB, with the MIB objects relevant for hosts or
monitors running RTCP-XR.

  This more modular and scalable structure would allow the two different
MIBs to be progressed separately, without creating a tight dependency among
them, and would allow adding other modules in the future.

  2. There is a need for a better coordination, and possibly some
consolidation between the RTCP-XR MIB and work being done in other WGs in
the IETF - especially the RMON MIB WG
(http://www.ietf.cnri.reston.va.us/html.charters/rmonmib-charter.html) work
on Application Performance Monitoring and Real-Time Application Monitoring
to create a consistent framework for performance monitoring, and avoid
duplication. While RTCP-XR covers an important aspect of media monitoring
when RTP is used, it does not cover the whole application space where RTP
streams can be used. I would be glad to provide more information, or work
with the authors of the different proposals in AVT to try to reach this
goal.

  Regards,

  Dan



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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>RTP MIB Extensions</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dwindows-1255">
<META content=3D"MSHTML 6.00.2800.1264" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D601054016-13112003><FONT face=3DArial color=3D#0000ff =

size=3D2>Dan</FONT></SPAN></DIV>
<DIV><SPAN class=3D601054016-13112003><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D601054016-13112003><FONT face=3DArial color=3D#0000ff =
size=3D2>I have=20
no issue with the idea of progressing updates to the RTP MIB and keeping =
the=20
RTCP XR MIB separate - I'll update the draft accordingly.&nbsp; There is =
also no=20
disagreement that the RTCP XR VoIP metrics only address VoIP whereas the =

envisaged scope of RAQMON is broader.&nbsp; </FONT></SPAN></DIV>
<DIV><SPAN class=3D601054016-13112003><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D601054016-13112003><FONT face=3DArial color=3D#0000ff =

size=3D2>Regards</FONT></SPAN></DIV>
<DIV><SPAN class=3D601054016-13112003><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D601054016-13112003><FONT face=3DArial color=3D#0000ff =
size=3D2>Alan=20
Clark</FONT></SPAN></DIV>
<DIV><SPAN class=3D601054016-13112003><FONT face=3DArial color=3D#0000ff =

size=3D2>Telchemy</FONT></SPAN></DIV>
<DIV><SPAN class=3D601054016-13112003><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D601054016-13112003>&nbsp;</SPAN></DIV>
<BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> avt-admin@ietf.org =

  [mailto:avt-admin@ietf.org]<B>On Behalf Of </B>Romascanu, Dan=20
  (Dan)<BR><B>Sent:</B> Wednesday, November 12, 2003 2:40 =
PM<BR><B>To:</B>=20
  avt@ietf.org<BR><B>Cc:</B> mankin@psg.com; Bert Wijnen=20
  (E-mail)<BR><B>Subject:</B> [AVT] RTP MIB =
Extensions<BR><BR></FONT></DIV><!-- Converted from text/rtf format -->
  <P dir=3Dltr><FONT face=3DArial size=3D2>Please allow me to make more =
clear my=20
  points expressed in the final minutes of the AVT WG meeting today =
related to=20
  <A=20
  =
href=3D"http://www.ietf.cnri.reston.va.us/internet-drafts/draft-clark-avt=
-rtpmibv2-00.txt">http://www.ietf.cnri.reston.va.us/internet-drafts/draft=
-clark-avt-rtpmibv2-00.txt</A></FONT></P>
  <P dir=3Dltr><FONT face=3DArial size=3D2>1. This MIB includes =
extensions to the RTP=20
  MIB (RFC 2959) related to RTCP-XR (RFC 3611). At the same time, there =
seems to=20
  be a need to update RFC 2959 as result of the evolution of the Real =
Time=20
  Protocol from RFC 1889 to RFC 3550. I strongly suggest to do this in =
two=20
  separate MIBs separate documents:</FONT></P>
  <P dir=3Dltr><FONT face=3DArial size=3D2>- a new RTP MIB, updating to =
RFC=20
  2959</FONT></P>
  <P dir=3Dltr><FONT face=3DArial size=3D2>- a supplementary RTCP-XR =
MIB, with the MIB=20
  objects relevant for hosts or monitors running RTCP-XR. </FONT></P>
  <P dir=3Dltr><FONT face=3DArial size=3D2>This more modular and =
scalable structure=20
  would allow the two different MIBs to be progressed separately, =
without=20
  creating a tight dependency among them, and would allow adding other =
modules=20
  in the future.</FONT></P>
  <P dir=3Dltr><FONT face=3DArial size=3D2>2. There is a need for a =
better=20
  coordination, and possibly some consolidation between the RTCP-XR MIB =
and work=20
  being done in other WGs in the IETF - especially the RMON MIB WG (<A=20
  =
href=3D"http://www.ietf.cnri.reston.va.us/html.charters/rmonmib-charter.h=
tml">http://www.ietf.cnri.reston.va.us/html.charters/rmonmib-charter.html=
</A>)=20
  work on Application Performance Monitoring and Real-Time Application=20
  Monitoring to create a consistent framework for performance =
monitoring, and=20
  avoid duplication. While RTCP-XR covers an important aspect of media=20
  monitoring when RTP is used, it does not cover the whole application =
space=20
  where RTP streams can be used. I would be glad to provide more =
information, or=20
  work with the authors of the different proposals in AVT to try to =
reach this=20
  goal. </FONT></P>
  <P dir=3Dltr><FONT face=3DArial size=3D2>Regards,</FONT></P>
  <P dir=3Dltr><FONT face=3DArial size=3D2>Dan</FONT></P>
  <P dir=3Dltr><FONT face=3DArial =
size=3D2></FONT>&nbsp;</P></BLOCKQUOTE></BODY></HTML>

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_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Nov 13 20:14:32 2003
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Subject: [AVT] V.8bis messages in RFC 2833bis
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While rewriting RFC 2833bis I found that the use of V.21 events with 
V.8bis signalling was underspecified.  V.21 simply transmits bits at 300 
bits/s using pure tones of specified frequencies to represent "0"s and 
"1"s.  (V.21 has two channels, but that is irrelevant here.)  A V.8bis 
message is a sequence of octets, where the carried information is 
preceded by HDLC flags and succeeded by a checksum and more HDLC flags. 
  To prevent false identification of HDLC flags within the carried 
information, extra zeros are inserted after sequences of five "1"s and 
stripped out again at the receiving end.

It doesn't seem necessary to carry the HDLC flags, and probably not the 
checksum (assuming redundancy encoding provides a sufficient degree of 
reliability).  The HDLC flags represent a minimum of 24 bits and a 
maximum of 64, each one of which is a separate event.  The checksum adds 
another 16 bits.  Zero stuffing adds some more.

Are there any existing implementations, and do they transmit the flags 
and checksum or not?

On a related topic, how many V.21 bits should be reported per packet, 
given that they are propagated at 33 ms intervals?  Any recommendation?

Tom Taylor


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Thinking about V.21 a bit more, I realized that it isn't clear when an 
event should be rteported.  The choices are to report an event every 
time a bit is sent (i.e., 300 events per second), or else every time a 
transition occurs (which means something less than 150 events per 
second).  Of course, the event rate itself doesn't mean too much without 
also specifying the rate at which event updates are issued.  But if 
updates are issued every 66.7 ms or more, the number of event reports 
would be halved on the average.

So: should we specify one approach or the other, or allow either and 
require implementations to be able to handle events spanning multiple bits?

Tom Taylor


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: "Tom Taylor" <taylor@nortelnetworks.com>, <avt@ietf.org>
Subject: SV: [AVT] More on V.21 events
Date: Thu, 13 Nov 2003 20:11:00 -0600
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Have you decided how you will report V.21 110 bits/second. It is used in
large parts of Europe and South America for text telephony.

( sorry, I should have read what applications are in scope before putting
this question, maybe it is totally out of scope that a textphone call needs
to be decoded and transmitted this way. )

Regards
Gunnar Hellstrom
-------------------------------------------
Gunnar Hellstrom
Omnitor AB
Renathvagen 2
SE 121 37 Johanneshov
SWEDEN
+46 8 556 002 03
Mob: +46 708 204 288
www.omnitor.se
Gunnar.Hellstrom@Omnitor.se
--------------------------------------------


-----Ursprungligt meddelande-----
Fran: avt-admin@ietf.org [mailto:avt-admin@ietf.org]For Tom Taylor
Skickat: den 13 november 2003 19:44
Till: avt@ietf.org
Amne: [AVT] More on V.21 events


Thinking about V.21 a bit more, I realized that it isn't clear when an
event should be rteported.  The choices are to report an event every
time a bit is sent (i.e., 300 events per second), or else every time a
transition occurs (which means something less than 150 events per
second).  Of course, the event rate itself doesn't mean too much without
also specifying the rate at which event updates are issued.  But if
updates are issued every 66.7 ms or more, the number of event reports
would be halved on the average.

So: should we specify one approach or the other, or allow either and
require implementations to be able to handle events spanning multiple bits?

Tom Taylor


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt




_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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I'll adjust the wording to allow for rates lower than 300 bits/s.  My 
original question still stands, but is perhaps less pressing at lower rates.

Actually, I'm in favour of requiring an event to be reported for each 
bit sent.  This increases redundancy in a natural manner, since event 
completions are transmitted redundantly.

Gunnar Hellstrom wrote:

> Have you decided how you will report V.21 110 bits/second. It is used in
> large parts of Europe and South America for text telephony.
> 
> ( sorry, I should have read what applications are in scope before putting
> this question, maybe it is totally out of scope that a textphone call needs
> to be decoded and transmitted this way. )
> 
> Regards
> Gunnar Hellstrom
> -------------------------------------------
> Gunnar Hellstrom
> Omnitor AB
> Renathvagen 2
> SE 121 37 Johanneshov
> SWEDEN
> +46 8 556 002 03
> Mob: +46 708 204 288
> www.omnitor.se
> Gunnar.Hellstrom@Omnitor.se
> --------------------------------------------
> 
> 
> -----Ursprungligt meddelande-----
> Fran: avt-admin@ietf.org [mailto:avt-admin@ietf.org]For Tom Taylor
> Skickat: den 13 november 2003 19:44
> Till: avt@ietf.org
> Amne: [AVT] More on V.21 events
> 
> 
> Thinking about V.21 a bit more, I realized that it isn't clear when an
> event should be rteported.  The choices are to report an event every
> time a bit is sent (i.e., 300 events per second), or else every time a
> transition occurs (which means something less than 150 events per
> second).  Of course, the event rate itself doesn't mean too much without
> also specifying the rate at which event updates are issued.  But if
> updates are issued every 66.7 ms or more, the number of event reports
> would be halved on the average.
> 
> So: should we specify one approach or the other, or allow either and
> require implementations to be able to handle events spanning multiple bits?
> 
> Tom Taylor
> 
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
> 
> 
> 
> 


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: SV: SV: [AVT] More on V.21 events
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OK, fine.

You will have some nasty detection cases. Hope that detection precision is
good enough for that.

E.g. the 300 bps byte FA versus the 110 bit/s byte FF are quite close. ( if
I get bit-order right )

The 300 bps byte FA in V.21 lower channel is 980 Hz lead-in, 10 ms  of 1180
Hz, 29.7 ms of 980 Hz continued in 980 Hz until next character.

The 110 bps byte FF in lower channel V.21 is 980 Hz lead-in, 9.1 ms of 1180
Hz, 81,9 ms of 980 Hz continued in 980 Hz.


So, the difference is that the start bit is 9.1 ms in one case, and the
start bit plus two data bits are 10 ms in the other case. Is this enough to
decide on 110 or 300 bps and time the detector to generate bit events during
the row of bits in steady 980 Hz?

Is there a need for also a detector specification?

Gunnar

-------------------------------------------
Gunnar Hellstrom
Omnitor AB
Renathvagen 2
SE 121 37 Johanneshov
SWEDEN
+46 8 556 002 03
Mob: +46 708 204 288
www.omnitor.se
Gunnar.Hellstrom@Omnitor.se
--------------------------------------------




-----Ursprungligt meddelande-----
Fran: avt-admin@ietf.org [mailto:avt-admin@ietf.org]For Tom Taylor
Skickat: den 13 november 2003 20:28
Till: Gunnar Hellstrom
Kopia: avt@ietf.org
Amne: Re: SV: [AVT] More on V.21 events


I'll adjust the wording to allow for rates lower than 300 bits/s.  My
original question still stands, but is perhaps less pressing at lower rates.

Actually, I'm in favour of requiring an event to be reported for each
bit sent.  This increases redundancy in a natural manner, since event
completions are transmitted redundantly.

Gunnar Hellstrom wrote:

> Have you decided how you will report V.21 110 bits/second. It is used in
> large parts of Europe and South America for text telephony.
>
> ( sorry, I should have read what applications are in scope before putting
> this question, maybe it is totally out of scope that a textphone call
needs
> to be decoded and transmitted this way. )
>
> Regards
> Gunnar Hellstrom
> -------------------------------------------
> Gunnar Hellstrom
> Omnitor AB
> Renathvagen 2
> SE 121 37 Johanneshov
> SWEDEN
> +46 8 556 002 03
> Mob: +46 708 204 288
> www.omnitor.se
> Gunnar.Hellstrom@Omnitor.se
> --------------------------------------------
>
>
> -----Ursprungligt meddelande-----
> Fran: avt-admin@ietf.org [mailto:avt-admin@ietf.org]For Tom Taylor
> Skickat: den 13 november 2003 19:44
> Till: avt@ietf.org
> Amne: [AVT] More on V.21 events
>
>
> Thinking about V.21 a bit more, I realized that it isn't clear when an
> event should be rteported.  The choices are to report an event every
> time a bit is sent (i.e., 300 events per second), or else every time a
> transition occurs (which means something less than 150 events per
> second).  Of course, the event rate itself doesn't mean too much without
> also specifying the rate at which event updates are issued.  But if
> updates are issued every 66.7 ms or more, the number of event reports
> would be halved on the average.
>
> So: should we specify one approach or the other, or allow either and
> require implementations to be able to handle events spanning multiple
bits?
>
> Tom Taylor
>
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
>
>
>
>


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt




_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Nov 13 22:45:28 2003
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From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: "avt IETF" <avt@ietf.org>, <toip@snowshore.com>,
        "Frank Shearar" <frank.shearar@rnid.org.uk>
Subject: [avt] SV: RFC2793bis and byte orders and other details
Date: Thu, 13 Nov 2003 21:44:27 -0600
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In the avt meeting, the RFC2793bis RTP Payload for Text Conversation was
accepted as an avt workgroup item.
I promised to produce draft-avt-rfc2793bis-00 out of
draft-hellstrom-avt-rfc2793bis-02
The current version is found at:
http://www.ietf.org/internet-drafts/draft-hellstrom-avt-rfc2793bis-02.txt

When I do that, I want to include actions on the comments so far.

Therefore, I want to check the reaction on some suggestions.

1. Specify byte-order for the new T140block counter. ( motivation below f=
rom
Frank Shearar )
   I doubt that it is advicable to specify byte order for one little piec=
e
of the packet.
   I assume that byte order issues are enough covered centrally in RTP.
   I would like to get comments on that assumption.

2. Open for other redundancy schemes than RFC2198, at least for the
audio/t140 format
   Especially for the audio intermixed packetization audio/t140, there mi=
ght
be of interest to use any of the
   FEC mechanisms to gain reliability. It may for example be desirable to
use the same FEC mechanism for the
   audio stream as for the interleaved text. The requirement to use the s=
ame
clock frequency for text and audio in the audio/t140 medium may also make=
 it
hard to use RFC2198, because the maximum offset will not be long enough t=
o
cater for transmission only every 300 ms, when the clock frequency of the
audio packetisation is used. ( for text/t140 it is 1000 Hz, and that is a
suitable frequency to allow for redundancy. )
I recommend that I insert wording that for audio/t140 , any suitable FEC
packetisation can be used.

3. IANA considerations
The following IANA registrations shall be included in the draft:

register media types for MIME and SDP usage:

text/t140  - already done
text/red
audio/t140

register the main media type  for sdp usage
text

register the attribute parameter
cps=3D

Register the capability for capability registration.
text


Anything more?

Gunnar Hellstr=F6m
-------------------------------------------
Gunnar Hellstr=F6m
Omnitor AB
Renathv=E4gen 2
SE 121 37 Johanneshov
SWEDEN
+46 8 556 002 03
Mob: +46 708 204 288
www.omnitor.se
Gunnar.Hellstrom@Omnitor.se
--------------------------------------------

-----Ursprungligt meddelande-----
Fr=E5n: owner-toip@snowshore.com [mailto:owner-toip@snowshore.com]F=F6r
Frank Shearar
Skickat: den 12 november 2003 05:07
Till: toip@snowshore.com
=C4mne: RFC2793bis and byte orders


Hi all,

I noticed that the RFC 2793bis draft never speaks of byte ordering. Is it
assumed, since this is usually used in the context of RTP, that the byte
order of the T140block counter is the byte order defined in RTP, i.e.,
network order?

Wouldn't it be slightly better to specifically mention the byte order in =
the
draft, just to avoid any potential confusion?

frank


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From exim@www1.ietf.org  Thu Nov 13 23:25:24 2003
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Subject: Re: SV: SV: [AVT] More on V.21 events
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Where is this stuff specified?  I can't find any sign of it in V.21 itself.

Gunnar Hellstrom wrote:

> OK, fine.
> 
> You will have some nasty detection cases. Hope that detection precision is
> good enough for that.
> 
> E.g. the 300 bps byte FA versus the 110 bit/s byte FF are quite close. ( if
> I get bit-order right )
> 
> The 300 bps byte FA in V.21 lower channel is 980 Hz lead-in, 10 ms  of 1180
> Hz, 29.7 ms of 980 Hz continued in 980 Hz until next character.
> 
> The 110 bps byte FF in lower channel V.21 is 980 Hz lead-in, 9.1 ms of 1180
> Hz, 81,9 ms of 980 Hz continued in 980 Hz.
> 
> 
> So, the difference is that the start bit is 9.1 ms in one case, and the
> start bit plus two data bits are 10 ms in the other case. Is this enough to
> decide on 110 or 300 bps and time the detector to generate bit events during
> the row of bits in steady 980 Hz?
> 
> Is there a need for also a detector specification?
> 
> Gunnar
> 
> -------------------------------------------
> Gunnar Hellstrom
> Omnitor AB
> Renathvagen 2
> SE 121 37 Johanneshov
> SWEDEN
> +46 8 556 002 03
> Mob: +46 708 204 288
> www.omnitor.se
> Gunnar.Hellstrom@Omnitor.se
> --------------------------------------------
> 
> 
> 
> 
> -----Ursprungligt meddelande-----
> Fran: avt-admin@ietf.org [mailto:avt-admin@ietf.org]For Tom Taylor
> Skickat: den 13 november 2003 20:28
> Till: Gunnar Hellstrom
> Kopia: avt@ietf.org
> Amne: Re: SV: [AVT] More on V.21 events
> 
> 
> I'll adjust the wording to allow for rates lower than 300 bits/s.  My
> original question still stands, but is perhaps less pressing at lower rates.
> 
> Actually, I'm in favour of requiring an event to be reported for each
> bit sent.  This increases redundancy in a natural manner, since event
> completions are transmitted redundantly.
> 
> Gunnar Hellstrom wrote:
> 
> 
>>Have you decided how you will report V.21 110 bits/second. It is used in
>>large parts of Europe and South America for text telephony.
>>
>>( sorry, I should have read what applications are in scope before putting
>>this question, maybe it is totally out of scope that a textphone call
> 
> needs
> 
>>to be decoded and transmitted this way. )
>>
>>Regards
>>Gunnar Hellstrom
>>-------------------------------------------
>>Gunnar Hellstrom
>>Omnitor AB
>>Renathvagen 2
>>SE 121 37 Johanneshov
>>SWEDEN
>>+46 8 556 002 03
>>Mob: +46 708 204 288
>>www.omnitor.se
>>Gunnar.Hellstrom@Omnitor.se
>>--------------------------------------------
>>
>>
>>-----Ursprungligt meddelande-----
>>Fran: avt-admin@ietf.org [mailto:avt-admin@ietf.org]For Tom Taylor
>>Skickat: den 13 november 2003 19:44
>>Till: avt@ietf.org
>>Amne: [AVT] More on V.21 events
>>
>>
>>Thinking about V.21 a bit more, I realized that it isn't clear when an
>>event should be rteported.  The choices are to report an event every
>>time a bit is sent (i.e., 300 events per second), or else every time a
>>transition occurs (which means something less than 150 events per
>>second).  Of course, the event rate itself doesn't mean too much without
>>also specifying the rate at which event updates are issued.  But if
>>updates are issued every 66.7 ms or more, the number of event reports
>>would be halved on the average.
>>
>>So: should we specify one approach or the other, or allow either and
>>require implementations to be able to handle events spanning multiple
> 
> bits?
> 
>>Tom Taylor
>>
>>
>>_______________________________________________
>>Audio/Video Transport Working Group
>>avt@ietf.org
>>https://www1.ietf.org/mailman/listinfo/avt
>>
>>
>>
>>
> 
> 
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
> 
> 
> 
> 


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Actually, I'm not sure this is relevant to 2833bis.  2833bis starts with 
the assumption that the sender recognizes the event.  You're saying that 
a sender able to distinguish between 110 bits/s and 300 bits/s V.21 will 
be tough to build.  2833bis isn't specifying a universal gateway, just 
giving gateways a language.

Gunnar Hellstrom wrote:

> OK, fine.
> 
> You will have some nasty detection cases. Hope that detection precision is
> good enough for that.
> 
> E.g. the 300 bps byte FA versus the 110 bit/s byte FF are quite close. ( if
> I get bit-order right )
> 
> The 300 bps byte FA in V.21 lower channel is 980 Hz lead-in, 10 ms  of 1180
> Hz, 29.7 ms of 980 Hz continued in 980 Hz until next character.
> 
> The 110 bps byte FF in lower channel V.21 is 980 Hz lead-in, 9.1 ms of 1180
> Hz, 81,9 ms of 980 Hz continued in 980 Hz.
> 
> 
> So, the difference is that the start bit is 9.1 ms in one case, and the
> start bit plus two data bits are 10 ms in the other case. Is this enough to
> decide on 110 or 300 bps and time the detector to generate bit events during
> the row of bits in steady 980 Hz?
> 
> Is there a need for also a detector specification?
> 
> Gunnar
> 
> -------------------------------------------
> Gunnar Hellstrom
> Omnitor AB
> Renathvagen 2
> SE 121 37 Johanneshov
> SWEDEN
> +46 8 556 002 03
> Mob: +46 708 204 288
> www.omnitor.se
> Gunnar.Hellstrom@Omnitor.se
> --------------------------------------------
> 
> 
> 
> 
> -----Ursprungligt meddelande-----
> Fran: avt-admin@ietf.org [mailto:avt-admin@ietf.org]For Tom Taylor
> Skickat: den 13 november 2003 20:28
> Till: Gunnar Hellstrom
> Kopia: avt@ietf.org
> Amne: Re: SV: [AVT] More on V.21 events
> 
> 
> I'll adjust the wording to allow for rates lower than 300 bits/s.  My
> original question still stands, but is perhaps less pressing at lower rates.
> 
> Actually, I'm in favour of requiring an event to be reported for each
> bit sent.  This increases redundancy in a natural manner, since event
> completions are transmitted redundantly.
> 
> Gunnar Hellstrom wrote:
> 
> 
>>Have you decided how you will report V.21 110 bits/second. It is used in
>>large parts of Europe and South America for text telephony.
>>
>>( sorry, I should have read what applications are in scope before putting
>>this question, maybe it is totally out of scope that a textphone call
> 
> needs
> 
>>to be decoded and transmitted this way. )
>>
>>Regards
>>Gunnar Hellstrom
>>-------------------------------------------
>>Gunnar Hellstrom
>>Omnitor AB
>>Renathvagen 2
>>SE 121 37 Johanneshov
>>SWEDEN
>>+46 8 556 002 03
>>Mob: +46 708 204 288
>>www.omnitor.se
>>Gunnar.Hellstrom@Omnitor.se
>>--------------------------------------------
>>
>>
>>-----Ursprungligt meddelande-----
>>Fran: avt-admin@ietf.org [mailto:avt-admin@ietf.org]For Tom Taylor
>>Skickat: den 13 november 2003 19:44
>>Till: avt@ietf.org
>>Amne: [AVT] More on V.21 events
>>
>>
>>Thinking about V.21 a bit more, I realized that it isn't clear when an
>>event should be rteported.  The choices are to report an event every
>>time a bit is sent (i.e., 300 events per second), or else every time a
>>transition occurs (which means something less than 150 events per
>>second).  Of course, the event rate itself doesn't mean too much without
>>also specifying the rate at which event updates are issued.  But if
>>updates are issued every 66.7 ms or more, the number of event reports
>>would be halved on the average.
>>
>>So: should we specify one approach or the other, or allow either and
>>require implementations to be able to handle events spanning multiple
> 
> bits?
> 
>>Tom Taylor
>>
>>
>>_______________________________________________
>>Audio/Video Transport Working Group
>>avt@ietf.org
>>https://www1.ietf.org/mailman/listinfo/avt
>>
>>
>>
>>
> 
> 
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
> 
> 
> 
> 


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Nov 13 23:50:26 2003
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From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: "Tom Taylor" <taylor@nortelnetworks.com>
Cc: <avt@ietf.org>
Subject: SV: SV: SV: [AVT] More on V.21 events
Date: Thu, 13 Nov 2003 22:49:00 -0600
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The trick is that usually V.21 users have agreed before the call, what
bitrate they will use.
If a Swiss textphone user calls another, they both have V.21s running at 110
bps.

So, then the modems trig on the start bit edge, and clock out its bits and
sample the frequency in the middle of the bits.

I guess that there are some words about that in V.21. Something about
pre-agreed bitrate or so.


But if you are designing a more general detector that has the ambition to
detect and transmit all tone signals reliably, then you end up in the kind
of tricky determination situations I mentioned.

We already have it in the text telephone auto moding modem V.18, that also
need to detect and discriminate between these quite close alternatives.

Gunnar
-------------------------------------------
Gunnar Hellstrom
Omnitor AB
Renathvagen 2
SE 121 37 Johanneshov
SWEDEN
+46 8 556 002 03
Mob: +46 708 204 288
www.omnitor.se
Gunnar.Hellstrom@Omnitor.se
--------------------------------------------





-----Ursprungligt meddelande-----
Fran: Tom Taylor [mailto:taylor@nortelnetworks.com]
Skickat: den 13 november 2003 22:24
Till: Gunnar Hellstrom
Kopia: avt@ietf.org
Amne: Re: SV: SV: [AVT] More on V.21 events


Where is this stuff specified?  I can't find any sign of it in V.21 itself.

Gunnar Hellstrom wrote:

> OK, fine.
>
> You will have some nasty detection cases. Hope that detection precision is
> good enough for that.
>
> E.g. the 300 bps byte FA versus the 110 bit/s byte FF are quite close. (
if
> I get bit-order right )
>
> The 300 bps byte FA in V.21 lower channel is 980 Hz lead-in, 10 ms  of
1180
> Hz, 29.7 ms of 980 Hz continued in 980 Hz until next character.
>
> The 110 bps byte FF in lower channel V.21 is 980 Hz lead-in, 9.1 ms of
1180
> Hz, 81,9 ms of 980 Hz continued in 980 Hz.
>
>
> So, the difference is that the start bit is 9.1 ms in one case, and the
> start bit plus two data bits are 10 ms in the other case. Is this enough
to
> decide on 110 or 300 bps and time the detector to generate bit events
during
> the row of bits in steady 980 Hz?
>
> Is there a need for also a detector specification?
>
> Gunnar
>
> -------------------------------------------
> Gunnar Hellstrom
> Omnitor AB
> Renathvagen 2
> SE 121 37 Johanneshov
> SWEDEN
> +46 8 556 002 03
> Mob: +46 708 204 288
> www.omnitor.se
> Gunnar.Hellstrom@Omnitor.se
> --------------------------------------------
>
>
>
>
> -----Ursprungligt meddelande-----
> Fran: avt-admin@ietf.org [mailto:avt-admin@ietf.org]For Tom Taylor
> Skickat: den 13 november 2003 20:28
> Till: Gunnar Hellstrom
> Kopia: avt@ietf.org
> Amne: Re: SV: [AVT] More on V.21 events
>
>
> I'll adjust the wording to allow for rates lower than 300 bits/s.  My
> original question still stands, but is perhaps less pressing at lower
rates.
>
> Actually, I'm in favour of requiring an event to be reported for each
> bit sent.  This increases redundancy in a natural manner, since event
> completions are transmitted redundantly.
>
> Gunnar Hellstrom wrote:
>
>
>>Have you decided how you will report V.21 110 bits/second. It is used in
>>large parts of Europe and South America for text telephony.
>>
>>( sorry, I should have read what applications are in scope before putting
>>this question, maybe it is totally out of scope that a textphone call
>
> needs
>
>>to be decoded and transmitted this way. )
>>
>>Regards
>>Gunnar Hellstrom
>>-------------------------------------------
>>Gunnar Hellstrom
>>Omnitor AB
>>Renathvagen 2
>>SE 121 37 Johanneshov
>>SWEDEN
>>+46 8 556 002 03
>>Mob: +46 708 204 288
>>www.omnitor.se
>>Gunnar.Hellstrom@Omnitor.se
>>--------------------------------------------
>>
>>
>>-----Ursprungligt meddelande-----
>>Fran: avt-admin@ietf.org [mailto:avt-admin@ietf.org]For Tom Taylor
>>Skickat: den 13 november 2003 19:44
>>Till: avt@ietf.org
>>Amne: [AVT] More on V.21 events
>>
>>
>>Thinking about V.21 a bit more, I realized that it isn't clear when an
>>event should be rteported.  The choices are to report an event every
>>time a bit is sent (i.e., 300 events per second), or else every time a
>>transition occurs (which means something less than 150 events per
>>second).  Of course, the event rate itself doesn't mean too much without
>>also specifying the rate at which event updates are issued.  But if
>>updates are issued every 66.7 ms or more, the number of event reports
>>would be halved on the average.
>>
>>So: should we specify one approach or the other, or allow either and
>>require implementations to be able to handle events spanning multiple
>
> bits?
>
>>Tom Taylor
>>
>>
>>_______________________________________________
>>Audio/Video Transport Working Group
>>avt@ietf.org
>>https://www1.ietf.org/mailman/listinfo/avt
>>
>>
>>
>>
>
>
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
>
>
>
>





_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Paul E. Jones" <paulej@packetizer.com>
To: "Mark Baugher" <mbaugher@cisco.com>,
        "Jean-Francois Mule" <jf.mule@cablelabs.com>
Cc: <avt@ietf.org>,
        =?iso-8859-1?Q?Mats__N=E4slund?= <mats.naslund@ericsson.com>,
        <karl.norrman@ericsson.com>, <mcgrew@cisco.com>,
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Subject: Re: [AVT] Moving draft-jones-avt-audio-t38-01.txt forward
Date: Fri, 14 Nov 2003 17:00:24 -0500
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Mark, Jean-Francois,

Taking your input and that of others, I've drafted the following text for
the "security considerations" section of this draft.  Please tell me if y=
ou
like this or suggest specific changes:

  T38 is vulnerable to attacks that are common to other types
  of RTP and SRTP payloads.  However, unlike audio, T.38 data
  may be manipulated in ways that are more obtrusive than
  audio.  As examples, rogue packets may cause
  transmission failure and manipulated packets may alter
  terminal identity.

  The security considerations discussed in the RTP [3] and
  SRTP [5] specification and any applicable RTP profile (for
  example, [9]) are applicable to T.38.  Regarding SRTP
  configuration, fax payloads SHOULD NOT use an HMAC-SHA1
  authentication tag that is shorter than 80 bits

Looking forward to your comments.

Paul

----- Original Message -----=20
From: "Mark Baugher" <mbaugher@cisco.com>
To: "Paul E. Jones" <paulej@packetizer.com>
Cc: "Jean-Francois Mule" <jf.mule@cablelabs.com>; <avt@ietf.org>; "Mats
N=E4slund" <mats.naslund@ericsson.com>; <karl.norrman@ericsson.com>;
<mcgrew@cisco.com>; "Elisabetta Carrara (KI/EAB)"
<elisabetta.carrara@ericsson.com>
Sent: Thursday, November 06, 2003 7:14 PM
Subject: Re: [AVT] Moving draft-jones-avt-audio-t38-01.txt forward


> Paul
>    I think the approach that you mentioned in the security consideratio=
n
is
> good since it relies on the security consideration and security mechani=
sms
> found in RTP and SRTP.  What is needed is to specify what might be
specific
> to the fax payload type with respect to SRTP.  I'd say that a short 32-=
bit
> tag is a SHOULD NOT for fax and not a MAY as it is with other applicati=
ons
> owing to the fact that the effects of fraud are so much greater form
> certain fax applications than is generally true for voice.  I'd offer
> something like this.
> "T38 is vulnerable to attacks that are common on other types of RTP
> payloads.  The security considerations and mechanisms of RTP and SRTP
apply
> to T38.  Regarding SRTP configuration, fax payloads SHOULD NOT use an
> HMAC-SHA1 authentication tag that is shorter than 80 bits."
>
> Mark
> At 01:40 PM 11/6/2003, Paul E. Jones wrote:
> >Mark,
> >
> >Your comments are valid.  In fact, it's SRTP that is one driver (but n=
ot
the
> >only driver) for wanting to move to RTP.  But, I hesitate to mention S=
RTP
> >without discussing the use of SRTP.
> >
> >Is there specific text that you'd like to propose.  If you propose it =
and
> >folks agree, I definitely have no objections to incorporating it.
> >
> >Paul
> >
> >----- Original Message -----
> >From: "Mark Baugher" <mbaugher@cisco.com>
> >To: "Jean-Francois Mule" <jf.mule@cablelabs.com>
> >Cc: "Paul E. Jones" <paulej@packetizer.com>; <avt@ietf.org>
> >Sent: Thursday, November 06, 2003 2:27 PM
> >Subject: RE: [AVT] Moving draft-jones-avt-audio-t38-01.txt forward
> >
> >
> > > At 10:51 AM 11/6/2003, Jean-Francois Mule wrote:
> > > >Paul wrote:
> > > > > I purposely did not address SRTP, because this brings in a
> > > > > different RTP profile (SAVP) and I've not fully studied what
> > > > > (if anything) needs to be done to use that profile.  Further,
> > > > > SRTP is still in draft form.  I really don't want to have
> > > > > interdependencies.
> > > >I'll let other folks comment on whether it is appropriate to not
address
> > > >the draft's connections with SRTP.
> > >
> > > I think that there is _an_ SRTP connection (or is it connexion :?)
that
> > > should be referred to in the security considerations.  The text tha=
t
is in
> > > the current draft could be extended slightly to say that t38 is
vulnerable
> > > to attacks that are common on other types of RTP payloads and that =
the
> > > security considerations and mechanisms of RTP and SRTP apply to t38.
> > >
> > > Mark
> > >
> > >
> > > > > > --
> > > > > > A number of issues were raised with respect to the usage of
> > > > > UDPTL for
> > > > > >    the long-term, though.  Specifically, there were
> > > > > concerns over the
> > > > > >    fact that UDPTL does not provide the same kind of statisti=
cs
> > > > > >    reporting as RTCP.  Further, there are no procedures in pl=
ace
for
> > > > > >    encrypting the UDPTL stream.  While the latter could be
> > > > > addressed in
> > > > > >    UDPTL, doing so would require a lot of effort and would
> > > > > largely be a
> > > > > >    duplication of the security work already on-going within
> > > > > the IETF.
> > > > > > --
> > > > >
> > > > > What was the specifc change you want made above?  Just
> > > > > calling out the "an RTP"?
> > > >
> > > >How about something like:
> > > >s/encrypting the UDPTL stream/authenticating or encrypting the UDP=
TL
> >stream
> > > >
> > > >
> > > > > > --- technical comment
> > > > > > The section 6 (security considerations) is rather silent on t=
he
> > > > > > security threats. Looking at the IFP packet format, I would
> > > > > argue that
> > > > > > those packets do expose serious threats to the
> > > > > communication that you
> > > > > > don't see in audio voice. If you insert a bad RTP packet voic=
e
> > > > > > payload, oh well, you may hear some garbage or clipping.
> > > > > With rtp/t38
> > > > > > fax, bad packets may cause real harm:
> > > > > > + denial of service attacks can be mounted by inserting RTP/t=
38
> > > > > > + packets
> > > > > > with hdlc-sig-end data field can simply terminate the fax
control
> > > > > > channel. Same with HDLC-FCS-BAD-Sig-End. And one can easily
create
> > > > > > those packets.
> > > > > > + theft of terminal identity? In some countries, faxes have
> > > > > more legal
> > > > > > value that emails and the access to the rtp packet content co=
uld
> > > > > > change the Called Subscriber ID frame (phone numbers), etc.  =
As
you
> > > > > > well know, this info is usually not coded in RTP but in
signaling
> > > > > > protocols.
> > > > > >
> > > > > > I'm sure the Study Group has thought of more. Could some
> > > > > examples of
> > > > > > security threats be given in the security consideration secti=
on
and
> > > > > > shouldn't this section again reference SRTP?
> > > > >
> > > > > These are both valid points.  I fully support the addition of
> > > > > some text that highlights this fact.  I would not want to
> > > > > call these out explicitly, necessarily, as there may be other
> > > > > places (e.g., modification or injection of NSF data, etc.),
> > > > > so I think it's reasonable to speak to this generically.
> > > > >
> > > > > Would you like to suggest some text?  I'd be happy to incorpora=
te
it.
> > > >I would rather let you wordsmith my comments into something genera=
l
you
> > > >like on this one.
> > > >
> > > >Jean-Fran=E7ois
> > > >
> > > >_______________________________________________
> > > >Audio/Video Transport Working Group
> > > >avt@ietf.org
> > > >https://www1.ietf.org/mailman/listinfo/avt
> > >
> > >
> > >
> >
> >
> >_______________________________________________
> >Audio/Video Transport Working Group
> >avt@ietf.org
> >https://www1.ietf.org/mailman/listinfo/avt
>
>
>


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Mon Nov 17 11:51:33 2003
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From: Tom Taylor <taylor@nortelnetworks.com>
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To: avt@ietf.org
Subject: Re: [AVT] V.8bis messages in RFC 2833bis
References: <3FB42C22.20605@nortelnetworks.com>
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I've swung the other way in my thinking.  It seems to me the most common 
application for 2833 data events will be that of an IP bridge between 
two PSTN terminals.  This means the signalling should be such as to 
require a minimum of application knowledge at either gateway.  So Every 
bit should be passed along faithfully.

There is a problem with V.8bis signals (as opposed to messages).  The 
signals consist of two tone segments: 400 ms of dual-frequency tone that 
indicates only which party is sending, followed by 100 ms of 
signal-specific single tone.  With the present set of V.8bis events, 
that means the sender has to wait something in excess of 400 ms before 
indicating what event is happening.  The receiver starts playout 400+ ms 
late.  I'm not sure if that has harmful effects -- the timing does not 
seem to be a problem in V.8bis itself, but maybe in interoperation with 
older modems ...

Anyway, a V.8bis signal is often followed by a preamble tone and then by 
a V.8bis message (i.e. sequence of V.21 bits, as dicussed in my note 
below).  The implication of the 400+ ms delay is that down the line, the 
receiver is going to have to buffer up to 150 bit events for playout 
when their time comes.  I'm putting text in 2833bis to warn of this, but 
it seems to me matters could be improved if we added one or two events 
to indicate that segment 1 signal playout has begun.  This would mean 
the receiver would only be behind by the amount of time the sender 
needed to recognize the tone and send out the initial message.

Any comments by people who have done some implementation?


Taylor, Tom-PT [CAR:5N00:EXCH] wrote:

> While rewriting RFC 2833bis I found that the use of V.21 events with 
> V.8bis signalling was underspecified.  V.21 simply transmits bits at 300 
> bits/s using pure tones of specified frequencies to represent "0"s and 
> "1"s.  (V.21 has two channels, but that is irrelevant here.)  A V.8bis 
> message is a sequence of octets, where the carried information is 
> preceded by HDLC flags and succeeded by a checksum and more HDLC flags. 
>  To prevent false identification of HDLC flags within the carried 
> information, extra zeros are inserted after sequences of five "1"s and 
> stripped out again at the receiving end.
> 
> It doesn't seem necessary to carry the HDLC flags, and probably not the 
> checksum (assuming redundancy encoding provides a sufficient degree of 
> reliability).  The HDLC flags represent a minimum of 24 bits and a 
> maximum of 64, each one of which is a separate event.  The checksum adds 
> another 16 bits.  Zero stuffing adds some more.
> 
> Are there any existing implementations, and do they transmit the flags 
> and checksum or not?
> 
> On a related topic, how many V.21 bits should be reported per packet, 
> given that they are propagated at 33 ms intervals?  Any recommendation?
> 
> Tom Taylor
> 
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
> 


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Mon Nov 17 13:29:25 2003
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Subject: Re: [AVT] V.8bis messages in RFC 2833bis
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On Mon, 17 Nov 2003, Tom Taylor wrote:

> I've swung the other way in my thinking.  It seems to me the most common
> application for 2833 data events will be that of an IP bridge between
> two PSTN terminals.  This means the signalling should be such as to
> require a minimum of application knowledge at either gateway.  So Every
> bit should be passed along faithfully.

I agree with the penultimate sentence, but would modify the last one
to say ever _tone_ should be passed along faithfully.  As I argued a
few meetings back, 2833 is a means to encode signaling tones more
accurately and with fewer bits than low-data rate voice encodings can
do.  It is not a data transport.

                                                        -- Steve

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Tue, 18 Nov 2003 00:35:56 +0000
Subject: Re: AW: [AVT] More comments on draft-rey-avt-rtp-3gpp-timed-text-01.txt
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Cc: Jose Rey <rey@panasonic.de>, jan.vandermeer@philips.com,
        "'IETF AVT WG'" <avt@ietf.org>,
        Magnus Westerlund <magnus.westerlund@ericsson.com>,
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To: Dave Singer <singer@apple.com>
From: Colin Perkins <csp@csperkins.org>
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On Tuesday, Nov 11, 2003, at 18:43 Europe/London, Dave Singer wrote:
> At 17:27 +0100 11/11/03, Jose Rey wrote:
>> Hi all,
>>
>> for those that did not attend the AVT session on Monday, there was a
>> number of comments and useful feedback during the timed text
>> presentation.
...
>> The current approach makes sense if the text samples are regularly 
>> small
>> and one or several of them fit into an RTP packet. Fragmentation would
>> happen seldom. The 'pure alf approach' is, in general, the most 
>> correct
>> one, and so also the one that requires more header definitions. If
>> fragmentation is a rare event, this overhead may be not worth (?).  
>> The
>> question to the group is pretty simple: which approach should be
>> pursued?
>
> I think we can find a middle ground.  I think that text samples over 
> 512 bytes are quite rare, and over 1400 bytes very rare indeed.  I 
> don't mind going to a format which is less file-format centric, but I 
> don't think we should spend a lot of time optimizing fragmented text 
> access units.

I think my question was more fundamental: do we need to support 
fragments at all? Or is it possible, assuming the text generator is 
aware of the network, that timed text samples can always be made it fit 
within the MTU?

Are the large samples an innate property of text, or an artefact of the 
process which can be avoided by careful coding?

-- 
Colin Perkins
http://csperkins.org/


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Tue Nov 18 01:32:38 2003
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From: "Rey Jose" <Rey@panasonic.de>
To: "Colin Perkins" <csp@csperkins.org>, "Dave Singer" <singer@apple.com>
Cc: <jan.vandermeer@philips.com>, "IETF AVT WG" <avt@ietf.org>,
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> -----Original Message-----
> From: Colin Perkins [mailto:csp@csperkins.org]
> Sent: Tuesday, November 18, 2003 1:36 AM
> To: Dave Singer
> Cc: Rey Jose; jan.vandermeer@philips.com; 'IETF AVT WG'; Magnus
> Westerlund; Yoshinori Matsui
> Subject: Re: AW: [AVT] More comments on
> draft-rey-avt-rtp-3gpp-timed-text-01.txt
>=20
>=20
> On Tuesday, Nov 11, 2003, at 18:43 Europe/London, Dave Singer wrote:
> > At 17:27 +0100 11/11/03, Jose Rey wrote:
> >> Hi all,
> >>
> >> for those that did not attend the AVT session on Monday,=20
> there was a
> >> number of comments and useful feedback during the timed text
> >> presentation.
> ...
> >> The current approach makes sense if the text samples are regularly=20
> >> small
> >> and one or several of them fit into an RTP packet.=20
> Fragmentation would
> >> happen seldom. The 'pure alf approach' is, in general, the most=20
> >> correct
> >> one, and so also the one that requires more header definitions. If
> >> fragmentation is a rare event, this overhead may be not=20
> worth (?). =20
> >> The
> >> question to the group is pretty simple: which approach should be
> >> pursued?
> >
> > I think we can find a middle ground.  I think that text=20
> samples over=20
> > 512 bytes are quite rare, and over 1400 bytes very rare indeed.  I=20
> > don't mind going to a format which is less file-format=20
> centric, but I=20
> > don't think we should spend a lot of time optimizing=20
> fragmented text=20
> > access units.
>=20
> I think my question was more fundamental: do we need to support=20
> fragments at all? Or is it possible, assuming the text generator is=20
> aware of the network, that timed text samples can always be=20
> made it fit=20
> within the MTU?
>=20
> Are the large samples an innate property of text, or an=20
> artefact of the=20
> process which can be avoided by careful coding?
>=20

for live streaming, on-the-fly MTU adaptation sounds reasonable; however =
for the normal  streaming scenario should every 3gp text file be =
available as a streameable file in every possible MTU size?=20


Jose


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Thread-Topic: BYE RTCP packet schedule.
Thread-Index: AcOtxQatXlUyERj0EdiSgwDQt7+w2w==
From: "Oren Peleg" <OrenP@audiocodes.com>
To: <schulzrinne@cs.columbia.edu>, <casner@acm.org>
Cc: <avt@ietf.org>, "Sigal Cohen" <SigalC@audiocodes.com>
Subject: [AVT] BYE RTCP packet schedule.
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According to RFC3550, the BYE packet should be sent no less then a =
minimum interval from former RTCP packet. This can cause a resource =
re-usage problems. Shouldn't it be more relaxed (at least for unicast =
sessions) to enable sending an immediate BYE upon RTP session closing?

Thanks,

Oren Peleg

Work phone No. : 03 - 5394167
Home phone No. : 03 - 5335913
Cell phone No. : 055 - 669613
Email : Oren.Peleg@audiocodes.com


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<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">According to RFC3550, the BYE =
packet should be sent no less then a minimum interval from former RTCP =
packet. This can cause a resource re-usage problems. Shouldn't it be =
more relaxed (at least for unicast sessions) to enable sending an =
immediate BYE upon RTP session closing?</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">Thanks,</FONT></P>

<P DIR=3DLTR><FONT COLOR=3D"#000080" SIZE=3D4 FACE=3D"Monotype =
Corsiva">Oren Peleg</FONT></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT COLOR=3D"#000000" SIZE=3D2 =
FACE=3D"Book Antiqua">Work phone No. : 03 - 5394167</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT COLOR=3D"#000000" SIZE=3D2 =
FACE=3D"Book Antiqua">Home phone No. : 03 - 5335913</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT COLOR=3D"#000000" SIZE=3D2 =
FACE=3D"Book Antiqua">Cell phone No. : 055 - 669613</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Book =
Antiqua">Email :<U> </U></FONT><U><FONT COLOR=3D"#0000FF" SIZE=3D2 =
FACE=3D"Book =
Antiqua">Oren.Peleg@audiocodes.com</FONT></U></SPAN><U><SPAN =
LANG=3D"he"></SPAN></U><SPAN LANG=3D"he"></SPAN></P>

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_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Greg Herlein <gregh@arkstream.com>
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To: david.liu@st.com
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Subject: Re: [AVT] Ask open source information on RTP/RTSP
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>     Now I want to find a free RTP/RTSP streaming server to test
> ip set top box which does as a RTP/RTSP client.

RTSP Implementation List at:
http://www.rtsp.org/2001/implementations.html

Of these, MPEG4IP seems very solid and is commercial friendly 
open source, for the most part.
http://mpeg4ip.sourceforge.net/

Apple, Helix, Live.com - all have footprints in the open source 
world.  Helix is from Real and may be used non-commercially, or 
licensed on a per server basis for commercial applications.  
Vovida even has some stack elements.

FFMPEG has ffserver, though it's perhaps not all that stable yet 
for production use.  Good codec library too.

http://ffmpeg.sourceforge.net/

I'm very intrigued by the idea of a set top box that is a 
RTSP/RTP client.  Can you share more (publicly or privately) 
about your new product?  I'd be interested in doing some testing 
on such a product, and even collaborating/assisting with your 
RTSP server needs.

Greg Herlein
VP Engineering
Arkstream, Inc.



_______________________________________________
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avt@ietf.org
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To: Gunnar Hellstrom <gunnar.hellstrom@omnitor.se>
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Subject: Re: SV: SV: [AVT] More on V.21 events
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I've had a couple of after-thoughts on your note.

First: as far as I know, RFC 2833 does not pretend to cover V.21s, as 
opposed to V.21.  As I understand it, the scope of 2833 is limited to 
session setup, not content once a session is started.  For that you use 
G.711 or V.150.1 modem relay or perhaps CLEARMODE.

Secondly, aren't your timings out by an order of magnitude?  At 300 
bits/s, surely each bit event can't take more than 3.3 ms.

Gunnar Hellstrom wrote:

> OK, fine.
> 
> You will have some nasty detection cases. Hope that detection precision is
> good enough for that.
> 
> E.g. the 300 bps byte FA versus the 110 bit/s byte FF are quite close. ( if
> I get bit-order right )
> 
> The 300 bps byte FA in V.21 lower channel is 980 Hz lead-in, 10 ms  of 1180
> Hz, 29.7 ms of 980 Hz continued in 980 Hz until next character.
> 
> The 110 bps byte FF in lower channel V.21 is 980 Hz lead-in, 9.1 ms of 1180
> Hz, 81,9 ms of 980 Hz continued in 980 Hz.
> 
> 
> So, the difference is that the start bit is 9.1 ms in one case, and the
> start bit plus two data bits are 10 ms in the other case. Is this enough to
> decide on 110 or 300 bps and time the detector to generate bit events during
> the row of bits in steady 980 Hz?
> 
> Is there a need for also a detector specification?
> 
> Gunnar
> 
> -------------------------------------------
> Gunnar Hellstrom
> Omnitor AB
> Renathvagen 2
> SE 121 37 Johanneshov
> SWEDEN
> +46 8 556 002 03
> Mob: +46 708 204 288
> www.omnitor.se
> Gunnar.Hellstrom@Omnitor.se
> --------------------------------------------
> 
> 
> 
> 
> -----Ursprungligt meddelande-----
> Fran: avt-admin@ietf.org [mailto:avt-admin@ietf.org]For Tom Taylor
> Skickat: den 13 november 2003 20:28
> Till: Gunnar Hellstrom
> Kopia: avt@ietf.org
> Amne: Re: SV: [AVT] More on V.21 events
> 
> 
> I'll adjust the wording to allow for rates lower than 300 bits/s.  My
> original question still stands, but is perhaps less pressing at lower rates.
> 
> Actually, I'm in favour of requiring an event to be reported for each
> bit sent.  This increases redundancy in a natural manner, since event
> completions are transmitted redundantly.
> 
> Gunnar Hellstrom wrote:
> 
> 
>>Have you decided how you will report V.21 110 bits/second. It is used in
>>large parts of Europe and South America for text telephony.
>>
>>( sorry, I should have read what applications are in scope before putting
>>this question, maybe it is totally out of scope that a textphone call
> 
> needs
> 
>>to be decoded and transmitted this way. )
>>
>>Regards
>>Gunnar Hellstrom
>>-------------------------------------------
>>Gunnar Hellstrom
>>Omnitor AB
>>Renathvagen 2
>>SE 121 37 Johanneshov
>>SWEDEN
>>+46 8 556 002 03
>>Mob: +46 708 204 288
>>www.omnitor.se
>>Gunnar.Hellstrom@Omnitor.se
>>--------------------------------------------
>>
>>
>>-----Ursprungligt meddelande-----
>>Fran: avt-admin@ietf.org [mailto:avt-admin@ietf.org]For Tom Taylor
>>Skickat: den 13 november 2003 19:44
>>Till: avt@ietf.org
>>Amne: [AVT] More on V.21 events
>>
>>
>>Thinking about V.21 a bit more, I realized that it isn't clear when an
>>event should be rteported.  The choices are to report an event every
>>time a bit is sent (i.e., 300 events per second), or else every time a
>>transition occurs (which means something less than 150 events per
>>second).  Of course, the event rate itself doesn't mean too much without
>>also specifying the rate at which event updates are issued.  But if
>>updates are issued every 66.7 ms or more, the number of event reports
>>would be halved on the average.
>>
>>So: should we specify one approach or the other, or allow either and
>>require implementations to be able to handle events spanning multiple
> 
> bits?
> 
>>Tom Taylor
>>
>>
>>_______________________________________________
>>Audio/Video Transport Working Group
>>avt@ietf.org
>>https://www1.ietf.org/mailman/listinfo/avt
>>
>>
>>
>>
> 
> 
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
> 
> 
> 
> 


_______________________________________________
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From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: "Tom Taylor" <taylor@nortelnetworks.com>
Cc: "Paul Jones" <paulej@packetizer.com>, <avt@ietf.org>
Subject: RE: SV: SV: [AVT] More on V.21 events
Date: Tue, 18 Nov 2003 10:17:23 -0600
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Tom,
You are probably right, it would be a waste of bandwidth to send V.21
session data as RFC2833 signals.
For V.21 session data in the text telephone case that I am thinking of,
RFC2793 is to prefer.

However, for DTMF, I think you should have detectors for RFC2833 on during
the whole voice call, not just during session setup.

What is V.21s?

The timings are right.
At 300 bits/s the bits are 3.3 ms. At 110 ms it is 9.1 ms.
The tricky example was to distinguish between a character at 300 bits/s
starting with start bit and two 0-bits giving a 9.9 ms long single tone,
then followed by 6 1-bits, a stop bit and then line fill with steady 1.
That will give the pattern of 9.9 ms of 1180 HZ followed by endless 980 Hz.
The comparison was with a byte of FF in 110 bits/s V.21 that is 9.1 ms of
1180 Hz, followed by endless 980 Hz.

If we follow the original idea, not to use RFC2833 for text telephony text,
it may be OK to limit the support to 300 bits/s V.21. All the session
startup examples of use of v.21 are using 300 bits/s.


Gunnar
-------------------------------------------
Gunnar Hellstrom
Omnitor AB
Renathvagen 2
SE 121 37 Johanneshov
SWEDEN
+46 8 556 002 03
Mob: +46 708 204 288
www.omnitor.se
Gunnar.Hellstrom@Omnitor.se
--------------------------------------------


-----Original Message-----
From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of Tom
Taylor
Sent: Tuesday, November 18, 2003 8:51 AM
To: Gunnar Hellstrom
Cc: avt@ietf.org
Subject: Re: SV: SV: [AVT] More on V.21 events


I've had a couple of after-thoughts on your note.

First: as far as I know, RFC 2833 does not pretend to cover V.21s, as
opposed to V.21.  As I understand it, the scope of 2833 is limited to
session setup, not content once a session is started.  For that you use
G.711 or V.150.1 modem relay or perhaps CLEARMODE.

Secondly, aren't your timings out by an order of magnitude?  At 300
bits/s, surely each bit event can't take more than 3.3 ms.

Gunnar Hellstrom wrote:

> OK, fine.
>
> You will have some nasty detection cases. Hope that detection precision is
> good enough for that.
>
> E.g. the 300 bps byte FA versus the 110 bit/s byte FF are quite close. (
if
> I get bit-order right )
>
> The 300 bps byte FA in V.21 lower channel is 980 Hz lead-in, 10 ms  of
1180
> Hz, 29.7 ms of 980 Hz continued in 980 Hz until next character.
>
> The 110 bps byte FF in lower channel V.21 is 980 Hz lead-in, 9.1 ms of
1180
> Hz, 81,9 ms of 980 Hz continued in 980 Hz.
>
>
> So, the difference is that the start bit is 9.1 ms in one case, and the
> start bit plus two data bits are 10 ms in the other case. Is this enough
to
> decide on 110 or 300 bps and time the detector to generate bit events
during
> the row of bits in steady 980 Hz?
>
> Is there a need for also a detector specification?
>
> Gunnar
>
> -------------------------------------------
> Gunnar Hellstrom
> Omnitor AB
> Renathvagen 2
> SE 121 37 Johanneshov
> SWEDEN
> +46 8 556 002 03
> Mob: +46 708 204 288
> www.omnitor.se
> Gunnar.Hellstrom@Omnitor.se
> --------------------------------------------
>
>
>
>
> -----Ursprungligt meddelande-----
> Fran: avt-admin@ietf.org [mailto:avt-admin@ietf.org]For Tom Taylor
> Skickat: den 13 november 2003 20:28
> Till: Gunnar Hellstrom
> Kopia: avt@ietf.org
> Amne: Re: SV: [AVT] More on V.21 events
>
>
> I'll adjust the wording to allow for rates lower than 300 bits/s.  My
> original question still stands, but is perhaps less pressing at lower
rates.
>
> Actually, I'm in favour of requiring an event to be reported for each
> bit sent.  This increases redundancy in a natural manner, since event
> completions are transmitted redundantly.
>
> Gunnar Hellstrom wrote:
>
>
>>Have you decided how you will report V.21 110 bits/second. It is used in
>>large parts of Europe and South America for text telephony.
>>
>>( sorry, I should have read what applications are in scope before putting
>>this question, maybe it is totally out of scope that a textphone call
>
> needs
>
>>to be decoded and transmitted this way. )
>>
>>Regards
>>Gunnar Hellstrom
>>-------------------------------------------
>>Gunnar Hellstrom
>>Omnitor AB
>>Renathvagen 2
>>SE 121 37 Johanneshov
>>SWEDEN
>>+46 8 556 002 03
>>Mob: +46 708 204 288
>>www.omnitor.se
>>Gunnar.Hellstrom@Omnitor.se
>>--------------------------------------------
>>
>>
>>-----Ursprungligt meddelande-----
>>Fran: avt-admin@ietf.org [mailto:avt-admin@ietf.org]For Tom Taylor
>>Skickat: den 13 november 2003 19:44
>>Till: avt@ietf.org
>>Amne: [AVT] More on V.21 events
>>
>>
>>Thinking about V.21 a bit more, I realized that it isn't clear when an
>>event should be rteported.  The choices are to report an event every
>>time a bit is sent (i.e., 300 events per second), or else every time a
>>transition occurs (which means something less than 150 events per
>>second).  Of course, the event rate itself doesn't mean too much without
>>also specifying the rate at which event updates are issued.  But if
>>updates are issued every 66.7 ms or more, the number of event reports
>>would be halved on the average.
>>
>>So: should we specify one approach or the other, or allow either and
>>require implementations to be able to handle events spanning multiple
>
> bits?
>
>>Tom Taylor
>>
>>
>>_______________________________________________
>>Audio/Video Transport Working Group
>>avt@ietf.org
>>https://www1.ietf.org/mailman/listinfo/avt
>>
>>
>>
>>
>
>
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
>
>
>
>


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Tue, 18 Nov 2003 11:32:32 -0500
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To: Gunnar Hellstrom <gunnar.hellstrom@omnitor.se>
CC: Paul Jones <paulej@packetizer.com>, avt@ietf.org
Subject: Re: SV: SV: [AVT] More on V.21 events
References: <BHEHLFPKIPMLPFNFAHJKAEANDCAA.gunnar.hellstrom@omnitor.se>
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Ah, now I understand -- you were talking bytes where I was thinking 
bits.  It happens that V.21 uses F-subscript-A to designate the higher 
characteristic frequency (binary 0) of a channel.  So I thought you were 
talking bits.

Gunnar Hellstrom wrote:

> Tom,
> You are probably right, it would be a waste of bandwidth to send V.21
> session data as RFC2833 signals.
> For V.21 session data in the text telephone case that I am thinking of,
> RFC2793 is to prefer.
> 
> However, for DTMF, I think you should have detectors for RFC2833 on during
> the whole voice call, not just during session setup.
> 
> What is V.21s?
> 
> The timings are right.
> At 300 bits/s the bits are 3.3 ms. At 110 ms it is 9.1 ms.
> The tricky example was to distinguish between a character at 300 bits/s
> starting with start bit and two 0-bits giving a 9.9 ms long single tone,
> then followed by 6 1-bits, a stop bit and then line fill with steady 1.
> That will give the pattern of 9.9 ms of 1180 HZ followed by endless 980 Hz.
> The comparison was with a byte of FF in 110 bits/s V.21 that is 9.1 ms of
> 1180 Hz, followed by endless 980 Hz.
> 
> If we follow the original idea, not to use RFC2833 for text telephony text,
> it may be OK to limit the support to 300 bits/s V.21. All the session
> startup examples of use of v.21 are using 300 bits/s.
> 
> 
> Gunnar
> -------------------------------------------
> Gunnar Hellstrom
> Omnitor AB
> Renathvagen 2
> SE 121 37 Johanneshov
> SWEDEN
> +46 8 556 002 03
> Mob: +46 708 204 288
> www.omnitor.se
> Gunnar.Hellstrom@Omnitor.se
> --------------------------------------------
> 
> 
> -----Original Message-----
> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of Tom
> Taylor
> Sent: Tuesday, November 18, 2003 8:51 AM
> To: Gunnar Hellstrom
> Cc: avt@ietf.org
> Subject: Re: SV: SV: [AVT] More on V.21 events
> 
> 
> I've had a couple of after-thoughts on your note.
> 
> First: as far as I know, RFC 2833 does not pretend to cover V.21s, as
> opposed to V.21.  As I understand it, the scope of 2833 is limited to
> session setup, not content once a session is started.  For that you use
> G.711 or V.150.1 modem relay or perhaps CLEARMODE.
> 
> Secondly, aren't your timings out by an order of magnitude?  At 300
> bits/s, surely each bit event can't take more than 3.3 ms.
> 
> Gunnar Hellstrom wrote:
> 
> 
>>OK, fine.
>>
>>You will have some nasty detection cases. Hope that detection precision is
>>good enough for that.
>>
>>E.g. the 300 bps byte FA versus the 110 bit/s byte FF are quite close. (
> 
> if
> 
>>I get bit-order right )
>>
>>The 300 bps byte FA in V.21 lower channel is 980 Hz lead-in, 10 ms  of
> 
> 1180
> 
>>Hz, 29.7 ms of 980 Hz continued in 980 Hz until next character.
>>
>>The 110 bps byte FF in lower channel V.21 is 980 Hz lead-in, 9.1 ms of
> 
> 1180
> 
>>Hz, 81,9 ms of 980 Hz continued in 980 Hz.
>>
>>
>>So, the difference is that the start bit is 9.1 ms in one case, and the
>>start bit plus two data bits are 10 ms in the other case. Is this enough
> 
> to
> 
>>decide on 110 or 300 bps and time the detector to generate bit events
> 
> during
> 
>>the row of bits in steady 980 Hz?
>>
>>Is there a need for also a detector specification?
>>
>>Gunnar
>>
>>-------------------------------------------
>>Gunnar Hellstrom
>>Omnitor AB
>>Renathvagen 2
>>SE 121 37 Johanneshov
>>SWEDEN
>>+46 8 556 002 03
>>Mob: +46 708 204 288
>>www.omnitor.se
>>Gunnar.Hellstrom@Omnitor.se
>>--------------------------------------------
>>
>>
>>
>>
>>-----Ursprungligt meddelande-----
>>Fran: avt-admin@ietf.org [mailto:avt-admin@ietf.org]For Tom Taylor
>>Skickat: den 13 november 2003 20:28
>>Till: Gunnar Hellstrom
>>Kopia: avt@ietf.org
>>Amne: Re: SV: [AVT] More on V.21 events
>>
>>
>>I'll adjust the wording to allow for rates lower than 300 bits/s.  My
>>original question still stands, but is perhaps less pressing at lower
> 
> rates.
> 
>>Actually, I'm in favour of requiring an event to be reported for each
>>bit sent.  This increases redundancy in a natural manner, since event
>>completions are transmitted redundantly.
>>
>>Gunnar Hellstrom wrote:
>>
>>
>>
>>>Have you decided how you will report V.21 110 bits/second. It is used in
>>>large parts of Europe and South America for text telephony.
>>>
>>>( sorry, I should have read what applications are in scope before putting
>>>this question, maybe it is totally out of scope that a textphone call
>>
>>needs
>>
>>
>>>to be decoded and transmitted this way. )
>>>
>>>Regards
>>>Gunnar Hellstrom
>>>-------------------------------------------
>>>Gunnar Hellstrom
>>>Omnitor AB
>>>Renathvagen 2
>>>SE 121 37 Johanneshov
>>>SWEDEN
>>>+46 8 556 002 03
>>>Mob: +46 708 204 288
>>>www.omnitor.se
>>>Gunnar.Hellstrom@Omnitor.se
>>>--------------------------------------------
>>>
>>>
>>>-----Ursprungligt meddelande-----
>>>Fran: avt-admin@ietf.org [mailto:avt-admin@ietf.org]For Tom Taylor
>>>Skickat: den 13 november 2003 19:44
>>>Till: avt@ietf.org
>>>Amne: [AVT] More on V.21 events
>>>
>>>
>>>Thinking about V.21 a bit more, I realized that it isn't clear when an
>>>event should be rteported.  The choices are to report an event every
>>>time a bit is sent (i.e., 300 events per second), or else every time a
>>>transition occurs (which means something less than 150 events per
>>>second).  Of course, the event rate itself doesn't mean too much without
>>>also specifying the rate at which event updates are issued.  But if
>>>updates are issued every 66.7 ms or more, the number of event reports
>>>would be halved on the average.
>>>
>>>So: should we specify one approach or the other, or allow either and
>>>require implementations to be able to handle events spanning multiple
>>
>>bits?
>>
>>
>>>Tom Taylor
>>>
>>>
>>>_______________________________________________
>>>Audio/Video Transport Working Group
>>>avt@ietf.org
>>>https://www1.ietf.org/mailman/listinfo/avt
>>>
>>>
>>>
>>>
>>
>>
>>
>>_______________________________________________
>>Audio/Video Transport Working Group
>>avt@ietf.org
>>https://www1.ietf.org/mailman/listinfo/avt
>>
>>
>>
>>
> 
> 
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
> 
> 
> 


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Tue, 18 Nov 2003 11:52:01 -0500
Subject: Re: [AVT] Ask open source information on RTP/RTSP
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Darwin streaming server does this.

Note that rtsp is not generally done with multicast.


On Tuesday, November 18, 2003, at 02:12 AM, david.liu@st.com wrote:

> Dear experts,
>
>     Now I want to find a free RTP/RTSP streaming server to test ip set 
> top box
> which does as a RTP/RTSP client.
>
>     I am not very familiar with existed RTP/RTSP open source code. So 
> would you
> like to give me some suggestions.
>
>     If the server can support VOD and control stream bit rate, it is 
> better.
>
> best regards
> david.liu
>
> ST Microelectronics Inc.
> email: david.liu@st.com
>
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
>
                                  Regards
                                  Marshall Eubanks

T.M. Eubanks
e-mail : marshall.eubanks@telesuite.com
http://www.telesuite.com


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Tue, 18 Nov 2003 09:46:08 -0800
To: Marshall Eubanks <tme@multicasttech.com>
From: Ross Finlayson <finlayson@live.com>
Subject: Re: [AVT] Ask open source information on RTP/RTSP
Cc: david.liu@st.com, avt@ietf.org
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>Note that rtsp is not generally done with multicast.

This, for some odd reason, is a common misconception about RTSP.  But it's 
incorrect.

RTSP can also be used to play multicast streams.  It is an excellent way 
for a player client to get the SDP information for an ongoing multicast 
stream - as an alternative to having the player client read a local ".sdp" 
file (for example).

(See, for example, section 14.4 in RFC 2326.)

People may be confused by the fact that (i) RTSP is not *necessary* for 
playing multicast streams (because a client could, instead, just read a 
local ".sdp" file), and (ii) Apple's "Darwin Streaming Server" does not 
serve multicast streams via RTSP.  (OTOH, Apple's QuickTime Player can play 
multicast RTSP/RTP streams just fine.)

         Ross (who has developed several multicast applications that use RTSP).



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Message-ID: <087d01c3ae00$87f72560$8f67d33d@viola>
From: "Y. Matsui" <matsui.yoshinori@jp.panasonic.com>
To: "Rey Jose" <Rey@panasonic.de>, "Colin Perkins" <csp@csperkins.org>,
        "Dave Singer" <singer@apple.com>
Cc: <jan.vandermeer@philips.com>, "IETF AVT WG" <avt@ietf.org>,
        "Magnus Westerlund" <magnus.westerlund@ericsson.com>
References: <8C7C642E667D8E4389197633C7D0E98301EFC1@lan-ex-01.panasonic.de>
Subject: Re: AW: [AVT] More comments on draft-rey-avt-rtp-3gpp-timed-text-01.txt
Date: Wed, 19 Nov 2003 03:19:02 +0900
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Hi Jose, Colin,

> > On Tuesday, Nov 11, 2003, at 18:43 Europe/London, Dave Singer wrote:
> > > At 17:27 +0100 11/11/03, Jose Rey wrote:
> > >> Hi all,
> > >>
> > >> for those that did not attend the AVT session on Monday, there was a
> > >> number of comments and useful feedback during the timed text
> > >> presentation.
> > ...
> > >> The current approach makes sense if the text samples are regularly small
> > >> and one or several of them fit into an RTP packet. Fragmentation would
> > >> happen seldom. The 'pure alf approach' is, in general, the most correct
> > >> one, and so also the one that requires more header definitions. If
> > >> fragmentation is a rare event, this overhead may be not worth (?). The
> > >> question to the group is pretty simple: which approach should be pursued?
> > >
> > > I think we can find a middle ground.  I think that text samples over
> > > 512 bytes are quite rare, and over 1400 bytes very rare indeed.  I
> > > don't mind going to a format which is less file-format centric, but I
> > > don't think we should spend a lot of time optimizing fragmented text
> > > access units.
> >
> > I think my question was more fundamental: do we need to support
> > fragments at all?

I think fragmentation should be supported, though it must be very rare.

> > Or is it possible, assuming the text generator is
> > aware of the network, that timed text samples can always be
> > made it fit within the MTU?

This could be a recommendation or guideline for text sample creators.

I think it may not be needed to define heavy fragment header for only
very rare case.  However, the payload format for timed text should
allow at least the simple fragment of text sample, in order to
support the timed text specification without any restrictions.

> > Are the large samples an innate property of text, or an artefact of the
> > process which can be avoided by careful coding?

A possible example of the large sample would be the ending roll of the movie.

> for live streaming, on-the-fly MTU adaptation sounds reasonable; however
> for the normal  streaming scenario should every 3gp text file be
> available as a streameable file in every possible MTU size?

Best regards,
Yoshinori Matsui
Panasonic/Matsushita

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Tue Nov 18 13:29:21 2003
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Date: Tue, 18 Nov 2003 10:25:05 -0800
To: "Y. Matsui" <matsui.yoshinori@jp.panasonic.com>,
        Rey Jose <Rey@panasonic.de>, Colin Perkins <csp@csperkins.org>
From: Dave Singer <singer@apple.com>
Subject: Re: AW: [AVT] More comments on
 draft-rey-avt-rtp-3gpp-timed-text-01.txt
Cc: jan.vandermeer@philips.com, IETF AVT WG <avt@ietf.org>,
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I agree with Matsui-san.  Large samples (e.g. a "ticker tape" which 
scrolls a long time, or the ending credits of a movie, likewise) 
might happen, but are rare.  The most we should do is allow you to 
find the string-length (and hence the beginning of the modifier 
boxes).  This might be as naive as putting the beginning of the 
modifier boxes at a packet boundary -- which also means you might 
send them twice, or whatever (very half-formed thought, you 
understand).



At 3:19 AM +0900 11/19/03, Y. Matsui wrote:
>Hi Jose, Colin,
>
>>  > On Tuesday, Nov 11, 2003, at 18:43 Europe/London, Dave Singer wrote:
>>  > > At 17:27 +0100 11/11/03, Jose Rey wrote:
>>  > >> Hi all,
>>  > >>
>>  > >> for those that did not attend the AVT session on Monday, there was a
>>  > >> number of comments and useful feedback during the timed text
>>  > >> presentation.
>>  > ...
>>  > >> The current approach makes sense if the text samples are 
>>regularly small
>>  > >> and one or several of them fit into an RTP packet. Fragmentation would
>>  > >> happen seldom. The 'pure alf approach' is, in general, the most correct
>>  > >> one, and so also the one that requires more header definitions. If
>>  > >> fragmentation is a rare event, this overhead may be not worth (?). The
>>  > >> question to the group is pretty simple: which approach should 
>>be pursued?
>>  > >
>>  > > I think we can find a middle ground.  I think that text samples over
>>  > > 512 bytes are quite rare, and over 1400 bytes very rare indeed.  I
>>  > > don't mind going to a format which is less file-format centric, but I
>>  > > don't think we should spend a lot of time optimizing fragmented text
>>  > > access units.
>>  >
>>  > I think my question was more fundamental: do we need to support
>>  > fragments at all?
>
>I think fragmentation should be supported, though it must be very rare.
>
>>  > Or is it possible, assuming the text generator is
>>  > aware of the network, that timed text samples can always be
>>  > made it fit within the MTU?
>
>This could be a recommendation or guideline for text sample creators.
>
>I think it may not be needed to define heavy fragment header for only
>very rare case.  However, the payload format for timed text should
>allow at least the simple fragment of text sample, in order to
>support the timed text specification without any restrictions.
>
>>  > Are the large samples an innate property of text, or an artefact of the
>>  > process which can be avoided by careful coding?
>
>A possible example of the large sample would be the ending roll of the movie.
>
>>  for live streaming, on-the-fly MTU adaptation sounds reasonable; however
>>  for the normal  streaming scenario should every 3gp text file be
>>  available as a streameable file in every possible MTU size?
>
>Best regards,
>Yoshinori Matsui
>Panasonic/Matsushita


-- 
David Singer
Apple Computer/QuickTime

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Tue Nov 18 14:19:26 2003
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Date: Tue, 18 Nov 2003 14:06:05 -0500
Subject: [AVT] Protocol Action: 'The Secure Real-time Transport Protocol'
 to Proposed Standard
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The IESG has approved following document:

- 'The Secure Real-time Transport Protocol '
   <draft-ietf-avt-srtp-09.txt> as a Proposed Standard

This document is the product of the Audio/Video Transport Working Group. 

The IESG contact persons are Allison Mankin and Jon Peterson.

Technical Summary

This specification defines a profile for the Real-time Transport Protocol
(RTP) and Real-time Transport Control Protocol (RTCP) called the Secure
Real-time Transport Protocol (SRTP).

The security goals for SRTP are to ensure:
       
* the confidentiality of the RTP and RTCP payloads, and
       
* the integrity of the entire RTP and RTCP packets, together with
    protection against replayed packets.
       
These security services are optional and independent from each
other, except that SRTCP integrity protection is mandatory
(malicious or erroneous alteration of RTCP messages could disrupt
the processing of the RTP stream).
       
Other, functional, goals for the protocol are:
       
* a framework that permits upgrading with new cryptographic
    transforms,
       
* low bandwidth cost, i.e., a framework preserving RTP header
    compression efficiency,
       
The provision of integrity is strongly recommended for most applications
of SRTP. The mandatory to implement transform for this profile is AES
counter mode, and there are risks associated with not using cryptographic
integrity with it (see Section 9.5).

Working Group Summary

The initial drafts had a default in which integrity services were not
the norm and in which SRTCP did not have mandatory integrity protection.
There was a lengthy security review to ensure that the authentication tag
is recommended to most RTP recommendations.

Protocol Quality

The specification was reviewed for the IESG by Eric Rescorla and Allison
Mankin. Implementations that tested the specification were discussed by
the working group.

RFC Editor Notes:

Add a reference to RFC 1750 to the end of paragraph 1 of Section 1.1.

Section 3.2.3 -

Old:

   If no valid context can be found for a packet corresponding to a
   certain context identifier, that packet MUST be discarded from
   further processing.


New:

   If no valid context can be found for a packet corresponding to a
   certain context identifier, that packet MUST be discarded.


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: Re: [AVT] Ask open source information on RTP/RTSP
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From: Marshall Eubanks <tme@multicasttech.com>
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On Tuesday, November 18, 2003, at 12:46 PM, Ross Finlayson wrote:

>
>> Note that rtsp is not generally done with multicast.
>
> This, for some odd reason, is a common misconception about RTSP.  But 
> it's incorrect.
>
> RTSP can also be used to play multicast streams.  It is an excellent 
> way for a player client to get the SDP information for an ongoing 
> multicast stream - as an alternative to having the player client read 
> a local ".sdp" file (for example).
>
> (See, for example, section 14.4 in RFC 2326.)
>
> People may be confused by the fact that (i) RTSP is not *necessary* 
> for playing multicast streams (because a client could, instead, just 
> read a local ".sdp" file), and (ii) Apple's "Darwin Streaming Server" 
> does not serve multicast streams via RTSP.  (OTOH, Apple's QuickTime 
> Player can play multicast RTSP/RTP streams just fine.)
>

I agree with you - I believe that RTSP was developed for multicast.
However, neither Darwin nor Helix will serve this (at least, the last 
time
I checked this) - and RealOne won't play them.


>         Ross (who has developed several multicast applications that 
> use RTSP).
>
>
>
                                  Regards
                                  Marshall Eubanks

T.M. Eubanks
e-mail : marshall.eubanks@telesuite.com
http://www.telesuite.com


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Tue Nov 18 16:08:32 2003
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Subject: RE: [AVT] Ask open source information on RTP/RTSP
Date: Tue, 18 Nov 2003 16:06:38 -0500
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Thread-Topic: [AVT] Ask open source information on RTP/RTSP
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From: "Matthew Heaney" <mheaney@on2.com>
To: "Marshall Eubanks" <tme@multicasttech.com>,
        "Ross Finlayson" <finlayson@live.com>
Cc: <david.liu@st.com>, <avt@ietf.org>
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TrueCast uses RTSP to SETUP and PLAY multicast sessions.  We're in the
process of adding SAP support, but right now a client gets SDP info via
the DESCRIBE response.


> -----Original Message-----
> From: Marshall Eubanks [mailto:tme@multicasttech.com]=20
> Sent: Tuesday, November 18, 2003 3:14 PM
> To: Ross Finlayson
> Cc: david.liu@st.com; avt@ietf.org
> Subject: Re: [AVT] Ask open source information on RTP/RTSP
>=20
>=20
>=20
> On Tuesday, November 18, 2003, at 12:46 PM, Ross Finlayson wrote:
>=20
> >
> >> Note that rtsp is not generally done with multicast.
> >
> > This, for some odd reason, is a common misconception about=20
> RTSP.  But
> > it's incorrect.
> >
> > RTSP can also be used to play multicast streams.  It is an excellent
> > way for a player client to get the SDP information for an ongoing=20
> > multicast stream - as an alternative to having the player=20
> client read=20
> > a local ".sdp" file (for example).
> >
> > (See, for example, section 14.4 in RFC 2326.)
> >
> > People may be confused by the fact that (i) RTSP is not *necessary*
> > for playing multicast streams (because a client could,=20
> instead, just=20
> > read a local ".sdp" file), and (ii) Apple's "Darwin=20
> Streaming Server"=20
> > does not serve multicast streams via RTSP.  (OTOH, Apple's=20
> QuickTime=20
> > Player can play multicast RTSP/RTP streams just fine.)
> >
>=20
> I agree with you - I believe that RTSP was developed for multicast.
> However, neither Darwin nor Helix will serve this (at least, the last=20
> time
> I checked this) - and RealOne won't play them.
>=20
>=20
> >         Ross (who has developed several multicast applications that=20
> > use RTSP).
> >
> >
> >
>                                   Regards
>                                   Marshall Eubanks
>=20
> T.M. Eubanks
> e-mail : marshall.eubanks@telesuite.com
> http://www.telesuite.com
>=20
>=20
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
>=20

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: [AVT] Questions on RTP/RTSP server 2
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Dear experts,

    Question1:
    ~~~~~~~~~~
    Whether Networkplay Inc's Helix Universal server and Microsoft Multimedia 
Server can support RTP/RTSP? 

    I only know Apple Inc's Quicktime Streaming server can from Apple website.

    Question2:
    ~~~~~~~~~~
    Whether Networkplay Inc's Helix Universal server only support raw MPEG2 TS 
steam? I know it supports Real Media, Windows Media, QuickTime, MPEG 4, MP3, and
more from realnetwork web site.    

best regards
david.liu
    


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Wed Nov 19 02:51:32 2003
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Subject: [AVT] Re: Questions on RTP/RTSP server 2
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Dear Experts,
    Sorry I send latest letter by mistake. Please ignore it and help me to see 
my questions in this mail.

    Question1:
    ~~~~~~~~~~
    Whether Networkplay Inc's Helix Universal server and Microsoft Multimedia 
Server can support RTP/RTSP? 

    I only know Apple Inc's Quicktime Streaming server can do it from Apple 
website.

    Question2:
    ~~~~~~~~~~
    Whether Networkplay Inc's Helix Universal server supports raw MPEG2 TS 
stream? I know it supports Real Media, Windows Media, QuickTime, MPEG 4, MP3, 
and more from realnetwork web site. But I only want to test RTP/RTSP client, so 
specific stream file format is unnecessary.

    Whether Apple Quicktime server and Darwin server can support raw MPEG2 TS 
stream?

best regards

david.liu

______________________________ Reply Separator _________________________________
Subject: Questions on RTP/RTSP server 2
Author:  david liu at china,shenzhenmktg,sh
Date:    2003-11-19 12:01


Dear experts,

Question1:
~~~~~~~~~~
Whether Networkplay Inc's Helix Universal server and Microsoft Multimedia 
Server can support RTP/RTSP? 

I only know Apple Inc's Quicktime Streaming server can from Apple website.

    Question2:
    ~~~~~~~~~~
    Whether Networkplay Inc's Helix Universal server only support raw MPEG2 TS 
steam? I know it supports Real Media, Windows Media, QuickTime, MPEG 4, MP3, and
more from realnetwork web site.    

best regards
david.liu



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Wed Nov 19 13:03:34 2003
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From: "Phelan, Tom" <tphelan@sonusnet.com>
To: "'dccp@ietf.org'" <dccp@ietf.org>
Cc: "'avt@ietf.org'" <avt@ietf.org>, "'mmusic@ietf.org'" <mmusic@ietf.org>
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Subject: [AVT] DCCP Media Guide
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Hi,

I've sent "Strategies for Media Applications Using DCCP" to the
Internet-Drafts archive.  While we're waiting for it to be added, it's also
available at http://www.phelan-4.com/dccp/draft-phelan-dccp-media-00.txt.
The plan is that this will soon evolve into a general DCCP User Guide.

Abstract:
   This document discusses strategies for using DCCP as the transport
   protocol for streaming media applications.  Of particular interest is
   how media streams can be adapted to the smoothly varying transmit
   rate requirements of CCID3, or TCP-Friendly Rate Control (TFRC).
   Also explored is the resulting network behavior of streams using
   these strategies and the fairness between these streams and TCP-based
   streams.

Comments will be appreciated.  In particular, I make the suggestion that
media apps should transmit at up to twice their nominal maximum rate, when
DCCP allows it, in order to provide protection from competing TCP
applications.  I know that the first time I heard someone make a similar
suggestion I was horrified, but as I developed the issues it seemed to be
necessary.  Reactions to this will be appreciated.

Tom P.

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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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		: Framing RTP and RTCP Packets over Connection-Oriented Transport
	Author(s)	: J. Lazzaro
	Filename	: draft-ietf-avt-rtp-framing-contrans-00.txt
	Pages		: 8
	Date		: 2003-11-19
	
This memo defines a method for framing Real Time Protocol (RTP) and
Real Time Control Protocol (RTCP) packets onto connection-oriented
transport (such as TCP and TLS).  The memo also defines how to
specify the framing method in a session description.

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	Title		: Registration of the text/red MIME Sub-Type
	Author(s)	: P. Jones
	Filename	: draft-jones-avt-text-red-00.txt
	Pages		: 5
	Date		: 2003-11-19
	
This document defines the text/red MIME sub-type.  The packetization
and usage of this MIME type, which is intended for use within SDP, is
specified within RFC 2198 and RFC 2793, respectively.

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From exim@www1.ietf.org  Wed Nov 19 16:17:55 2003
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A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Real-Time Facsimile (T.38) - audio/t38 MIME 
			  Sub-type Registration
	Author(s)	: P. Jones
	Filename	: draft-jones-avt-audio-t38-02.txt
	Pages		: 6
	Date		: 2003-11-19
	
This document defines the MIME sub-type audio/t38.  The packetization
and usage of this MIME type, which is intended for use within SDP, is
specified within ITU-T Recommendation T.38.

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_______________________________________________
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https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Nov 20 02:34:06 2003
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Tom,

I don't think this solution will be acceptable to many of the network 
providers of interactive voip services. Many of them are insistent on 
using low rate codecs, like g.723, in order to reduce the costs of 
capacity in their networks. Doubling the rate would defeat the purpose 
of using these low rate codecs in the first place.

-Jonathan R.

Phelan, Tom wrote:


> Comments will be appreciated.  In particular, I make the suggestion that
> media apps should transmit at up to twice their nominal maximum rate, when
> DCCP allows it, in order to provide protection from competing TCP
> applications.  I know that the first time I heard someone make a similar
> suggestion I was horrified, but as I developed the issues it seemed to be
> necessary.  Reactions to this will be appreciated.
> 
> Tom P.
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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To: Oren Peleg <OrenP@audiocodes.com>
CC: schulzrinne@cs.columbia.edu, casner@acm.org, avt@ietf.org,
        Sigal Cohen <SigalC@audiocodes.com>
Subject: Re: [AVT] BYE RTCP packet schedule.
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Hi Oren,

I think you should go re-read section 6.3.7 of RFC 3550. The relaxation 
you ask for exist. The described algorithm needs only to be used if 
there are more than 50 participants in the session. Most unicast 
sessions does not use more than 50 SSRCs in total. Secondly if on has a 
problem to wait the time the derivation algorithm results in then on MAY 
leave without sending a BYE packet.

Quoted from section 6.3.7 of RFC 3550:

    o  When the participant decides to leave the system, tp is reset to
       tc, the current time, members and pmembers are initialized to 1,
       initial is set to 1, we_sent is set to false, senders is set to 0,
       and avg_rtcp_size is set to the size of the compound BYE packet.
       The calculated interval T is computed.  The BYE packet is then
       scheduled for time tn = tc + T.

    o  Every time a BYE packet from another participant is received,
       members is incremented by 1 regardless of whether that participant
       exists in the member table or not, and when SSRC sampling is in
       use, regardless of whether or not the BYE SSRC would be included
       in the sample.  members is NOT incremented when other RTCP packets
       or RTP packets are received, but only for BYE packets.  Similarly,
       avg_rtcp_size is updated only for received BYE packets.  senders
       is NOT updated when RTP packets arrive; it remains 0.

    o  Transmission of the BYE packet then follows the rules for
       transmitting a regular RTCP packet, as above.

Please look at the first bullet. In this you reset a lot of variables. 
This will result in that if no BYE packet you will based on bandwidth be 
able to send pretty quickly. If a min_rtcp_intreval is set then that 
will be halved due to the initial variable being set to true.

Thus the restriction on sending BYE is only required to be used for 
group size over 50 participants. And even then the sending of a BYE is 
prioritized while still being limited to avoid feedback explosion at 
session endings.

I hope this clarifies things for you.

Best Regards

Magnus


Oren Peleg wrote:

> According to RFC3550, the BYE packet should be sent no less then a 
> minimum interval from former RTCP packet. This can cause a resource 
> re-usage problems. Shouldn't it be more relaxed (at least for unicast 
> sessions) to enable sending an immediate BYE upon RTP session closing?
> 
> Thanks,
> 
> Oren Peleg
> 
> Work phone No. : 03 - 5394167
> 
> Home phone No. : 03 - 5335913
> 
> Cell phone No. : 055 - 669613
> 
> Email : Oren.Peleg@audiocodes.com
> 

-- 

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Nov 20 09:35:27 2003
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From: Tom Taylor <taylor@nortelnetworks.com>
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Subject: [AVT] VBD vs. RFC 2833 vs. V.150.1 for V.21
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I don't know if this is the right place to ask this question, or whether 
I should go to Study Group 16.  During my rewrite of RFC 2833 I'm trying 
to add a few words of advice about the encoding of V.21 used to carry 
the conversation, as opposed to control signalling.  This comes up with 
respect to text telephones.

V.21 is a frequency shift modem operating at up to 300 bits/s, one pure 
tone for "0", another for "1".  By my rough calculations, using G.711 
for carrying the tones takes more bandwidth than RFC 2833, assuming 
similar packetization intervals.  I see it as:

G.711 = 64000 bits/s.

RFC 2833 events = 300 bit events/s x 16 bits/event report x 4 repetition 
factor = 19200 bits/s.  There's more for redundancy headers, but this is 
the right order of magnitude.

V.150.1 modem relay would be more compact still, since it reports octets 
rather than bits per sample.

So the advice would be to carry V.21 as RFC 2833 events or use V.150.1 
modem relay in preference to G.711.  Do I have it right, or was I too 
careless with my calculations?

Tom Taylor


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Nov 20 10:37:27 2003
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From: "Phelan, Tom" <tphelan@sonusnet.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Cc: "'dccp@ietf.org'" <dccp@ietf.org>, "'avt@ietf.org'" <avt@ietf.org>,
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Subject: [AVT] RE: [MMUSIC] DCCP Media Guide
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Hi Jonathan,

Thanks for the comments.  The actual proposal is to send up to double the
nominal transmit rate when there's available bandwidth, and to reduce the
rate as necessary if there isn't enough bandwidth.  This doubling is to
allow fair sharing of the chokepoint bandwidth (between DCCP media apps and
TCP apps), and to avoid what appears to me to be the potential for bullying
from TCP applications that want to suck up all the bandwidth they can.  It
doesn't preclude using low-rate codecs, or mean that the access link must
support the double rate for the connection to function well.

This recommendation was meant for best-effort environments, where the voice
traffic receives no special treatment, and must compete with TCP apps for
network resources.  I can see how this would require a network that gave
special treatment to voice packets to provision double the capacity, though.

This fairness problem is difficult.  All of the research I'm familiar with
approaches it from the direction of other apps being fair to TCP.  I'm not
aware of any other methods for ensuring that non-TCP apps get their fair
share when competing with TCP.  If you (or anyone else) knows of something
I'd appreciate the pointer.

Tom P.

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Thursday, November 20, 2003 2:32 AM
> To: Phelan, Tom
> Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> Subject: Re: [MMUSIC] DCCP Media Guide
> 
> 
> Tom,
> 
> I don't think this solution will be acceptable to many of the network 
> providers of interactive voip services. Many of them are insistent on 
> using low rate codecs, like g.723, in order to reduce the costs of 
> capacity in their networks. Doubling the rate would defeat 
> the purpose 
> of using these low rate codecs in the first place.
> 
> -Jonathan R.
> 
> Phelan, Tom wrote:
> 
> 
> > Comments will be appreciated.  In particular, I make the 
> suggestion that
> > media apps should transmit at up to twice their nominal 
> maximum rate, when
> > DCCP allows it, in order to provide protection from competing TCP
> > applications.  I know that the first time I heard someone 
> make a similar
> > suggestion I was horrified, but as I developed the issues 
> it seemed to be
> > necessary.  Reactions to this will be appreciated.
> > 
> > Tom P.
> > 
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www1.ietf.org/mailman/listinfo/mmusic
> > 
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Technology Officer                    Parsippany, NJ 07054-2711
> dynamicsoft
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
> 

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Nov 20 11:06:25 2003
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Phelan, Tom'" <tphelan@sonusnet.com>,
        "'Jonathan Rosenberg'"
	 <jdrosen@dynamicsoft.com>
Cc: "'dccp@ietf.org'" <dccp@ietf.org>, "'avt@ietf.org'" <avt@ietf.org>,
        "'mmusic@ietf.org'" <mmusic@ietf.org>
Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
Date: Thu, 20 Nov 2003 11:04:49 -0500
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To me, this is just nonsense.
I can't imagine anyone deploying such a solution.

This is forcing the problem to fit the solution, and it's not
the right way to go.  If you want it to work, the protocol
has to take the characteristics of the flows as they are, and not
as you would like them to be.  Real time media, streaming or realtime,
is either constant bit rate, or pretty close to it.  That's how it 
works.  Our challenge is to make the protocols work with this data,
not change the data, especially silly ideas like doubling the bandwidth.

We make video telephones.  They nominally send 4 megabits.  Would
you like us to double that, under any circumstances?  

I thought not. 

Brian

> -----Original Message-----
> From: Phelan, Tom [mailto:tphelan@sonusnet.com]
> Sent: Thursday, November 20, 2003 10:36 AM
> To: 'Jonathan Rosenberg'
> Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> Subject: [AVT] RE: [MMUSIC] DCCP Media Guide
> 
> 
> Hi Jonathan,
> 
> Thanks for the comments.  The actual proposal is to send up 
> to double the
> nominal transmit rate when there's available bandwidth, and 
> to reduce the
> rate as necessary if there isn't enough bandwidth.  This 
> doubling is to
> allow fair sharing of the chokepoint bandwidth (between DCCP 
> media apps and
> TCP apps), and to avoid what appears to me to be the 
> potential for bullying
> from TCP applications that want to suck up all the bandwidth 
> they can.  It
> doesn't preclude using low-rate codecs, or mean that the 
> access link must
> support the double rate for the connection to function well.
> 
> This recommendation was meant for best-effort environments, 
> where the voice
> traffic receives no special treatment, and must compete with 
> TCP apps for
> network resources.  I can see how this would require a 
> network that gave
> special treatment to voice packets to provision double the 
> capacity, though.
> 
> This fairness problem is difficult.  All of the research I'm 
> familiar with
> approaches it from the direction of other apps being fair to 
> TCP.  I'm not
> aware of any other methods for ensuring that non-TCP apps get 
> their fair
> share when competing with TCP.  If you (or anyone else) knows 
> of something
> I'd appreciate the pointer.
> 
> Tom P.
> 
> > -----Original Message-----
> > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > Sent: Thursday, November 20, 2003 2:32 AM
> > To: Phelan, Tom
> > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > Subject: Re: [MMUSIC] DCCP Media Guide
> > 
> > 
> > Tom,
> > 
> > I don't think this solution will be acceptable to many of 
> the network 
> > providers of interactive voip services. Many of them are 
> insistent on 
> > using low rate codecs, like g.723, in order to reduce the costs of 
> > capacity in their networks. Doubling the rate would defeat 
> > the purpose 
> > of using these low rate codecs in the first place.
> > 
> > -Jonathan R.
> > 
> > Phelan, Tom wrote:
> > 
> > 
> > > Comments will be appreciated.  In particular, I make the 
> > suggestion that
> > > media apps should transmit at up to twice their nominal 
> > maximum rate, when
> > > DCCP allows it, in order to provide protection from competing TCP
> > > applications.  I know that the first time I heard someone 
> > make a similar
> > > suggestion I was horrified, but as I developed the issues 
> > it seemed to be
> > > necessary.  Reactions to this will be appreciated.
> > > 
> > > Tom P.
> > > 
> > > _______________________________________________
> > > mmusic mailing list
> > > mmusic@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/mmusic
> > > 
> > 
> > -- 
> > Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> > Chief Technology Officer                    Parsippany, NJ 
> 07054-2711
> > dynamicsoft
> > jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> > http://www.jdrosen.net                      PHONE: (973) 952-5000
> > http://www.dynamicsoft.com
> > 
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
> 

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Nov 20 11:49:25 2003
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From: "Phelan, Tom" <tphelan@sonusnet.com>
To: "'Rosen, Brian'" <Brian.Rosen@marconi.com>
Cc: "'dccp@ietf.org'" <dccp@ietf.org>, "'avt@ietf.org'" <avt@ietf.org>,
        "'mmusic@ietf.org'" <mmusic@ietf.org>
Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
Date: Thu, 20 Nov 2003 11:47:47 -0500
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Hi Brian,

Thanks for the comments.  See inline ...

Tom P.

> -----Original Message-----
> From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> Sent: Thursday, November 20, 2003 11:05 AM
> To: 'Phelan, Tom'; 'Jonathan Rosenberg'
> Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> 
> 
> To me, this is just nonsense.
> I can't imagine anyone deploying such a solution.
> 
> This is forcing the problem to fit the solution, and it's not
> the right way to go.  If you want it to work, the protocol
> has to take the characteristics of the flows as they are, and not
> as you would like them to be.  Real time media, streaming or realtime,
> is either constant bit rate, or pretty close to it.  That's how it 
> works.  Our challenge is to make the protocols work with this data,
> not change the data, especially silly ideas like doubling the 
> bandwidth.
> 
> We make video telephones.  They nominally send 4 megabits.  Would
> you like us to double that, under any circumstances?  
> 
> I thought not. 

Would you prefer to be forced unnecessarily into packet loss and a lower
encoding rate by greedy TCP applications?  Or do you have another solution?

I don't think I'm forcing the problem to fit the solution, just perhaps that
the problem I'm trying to solve is different than you perceive.  The problem
that I'm trying to solve is how can an application that is content with a
maximum bandwidth compete with TCP applications that are insatiably greedy,
in a way that satisfies some concept of fairness in all directions?

The solution that I see is to provide a buffer against TCP probing.  This
would involve sending more data than is nominally necessary.  If the network
has capacity for more than 4M bps, then why shouldn't a media app make use
of it?  A TCP app would.

I must admit that my initial reactions to suggestions like this were very
similar to yours.  I encourage you to look at the draft for more detail on
the issues that convinced me otherwise.

> 
> Brian
> 
> > -----Original Message-----
> > From: Phelan, Tom [mailto:tphelan@sonusnet.com]
> > Sent: Thursday, November 20, 2003 10:36 AM
> > To: 'Jonathan Rosenberg'
> > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > Subject: [AVT] RE: [MMUSIC] DCCP Media Guide
> > 
> > 
> > Hi Jonathan,
> > 
> > Thanks for the comments.  The actual proposal is to send up 
> > to double the
> > nominal transmit rate when there's available bandwidth, and 
> > to reduce the
> > rate as necessary if there isn't enough bandwidth.  This 
> > doubling is to
> > allow fair sharing of the chokepoint bandwidth (between DCCP 
> > media apps and
> > TCP apps), and to avoid what appears to me to be the 
> > potential for bullying
> > from TCP applications that want to suck up all the bandwidth 
> > they can.  It
> > doesn't preclude using low-rate codecs, or mean that the 
> > access link must
> > support the double rate for the connection to function well.
> > 
> > This recommendation was meant for best-effort environments, 
> > where the voice
> > traffic receives no special treatment, and must compete with 
> > TCP apps for
> > network resources.  I can see how this would require a 
> > network that gave
> > special treatment to voice packets to provision double the 
> > capacity, though.
> > 
> > This fairness problem is difficult.  All of the research I'm 
> > familiar with
> > approaches it from the direction of other apps being fair to 
> > TCP.  I'm not
> > aware of any other methods for ensuring that non-TCP apps get 
> > their fair
> > share when competing with TCP.  If you (or anyone else) knows 
> > of something
> > I'd appreciate the pointer.
> > 
> > Tom P.
> > 
> > > -----Original Message-----
> > > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > > Sent: Thursday, November 20, 2003 2:32 AM
> > > To: Phelan, Tom
> > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > Subject: Re: [MMUSIC] DCCP Media Guide
> > > 
> > > 
> > > Tom,
> > > 
> > > I don't think this solution will be acceptable to many of 
> > the network 
> > > providers of interactive voip services. Many of them are 
> > insistent on 
> > > using low rate codecs, like g.723, in order to reduce the 
> costs of 
> > > capacity in their networks. Doubling the rate would defeat 
> > > the purpose 
> > > of using these low rate codecs in the first place.
> > > 
> > > -Jonathan R.
> > > 
> > > Phelan, Tom wrote:
> > > 
> > > 
> > > > Comments will be appreciated.  In particular, I make the 
> > > suggestion that
> > > > media apps should transmit at up to twice their nominal 
> > > maximum rate, when
> > > > DCCP allows it, in order to provide protection from 
> competing TCP
> > > > applications.  I know that the first time I heard someone 
> > > make a similar
> > > > suggestion I was horrified, but as I developed the issues 
> > > it seemed to be
> > > > necessary.  Reactions to this will be appreciated.
> > > > 
> > > > Tom P.
> > > > 
> > > > _______________________________________________
> > > > mmusic mailing list
> > > > mmusic@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/mmusic
> > > > 
> > > 
> > > -- 
> > > Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> > > Chief Technology Officer                    Parsippany, NJ 
> > 07054-2711
> > > dynamicsoft
> > > jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> > > http://www.jdrosen.net                      PHONE: (973) 952-5000
> > > http://www.dynamicsoft.com
> > > 
> > 
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www1.ietf.org/mailman/listinfo/avt
> > 
> 

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: philippe.gentric@philips.com
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Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
To: "Phelan, Tom" <tphelan@sonusnet.com>,
        "'Rosen, Brian'" <Brian.Rosen@marconi.com>
Cc: "'avt@ietf.org'" <avt@ietf.org>, "'dccp@ietf.org'" <dccp@ietf.org>,
        "'mmusic@ietf.org'" <mmusic@ietf.org>
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Brian, all,

It may provide some help in understanding the situation
to describe your exemple as follows:

* you send video at 4 Mb/s, good.

* it happens that you are competing with (a lot of ) TCP

AND that you want to be fair to it:

* so when that happens you decrease your rate to 2 Mb/s for "a while"
because otherwise you are not fair.

* but at some moment you will want to come back up to 4 Mb/s right ?

=> the point is that the **only way**  to know if you "can" do that, is to try it....

regards,


Philippe Gentric
Chief Architect
Philips Software
philippe.gentric@philips.com
http://www.software.philips.com


                                                                                                                                       
                                                                                                                                       
                                                   To:   "'Rosen, Brian'" <Brian.Rosen@marconi.com>                                    
                                                   cc:   "'dccp@ietf.org'" <dccp@ietf.org>                                             
                                                    "'avt@ietf.org'" <avt@ietf.org>                                                    
                                                    "'mmusic@ietf.org'" <mmusic@ietf.org>                                              
               "Phelan, Tom"                        (bcc: Philippe Gentric/MP4-SUR/CE/PHILIPS)                                         
               <tphelan@sonusnet.com>              Subject:    RE: [AVT] RE: [MMUSIC] DCCP Media Guide                                 
                                                                                                                                       
               Sent by:                            Classification:                                                                     
               mmusic-admin@ietf.org                                                                                                   
                                                                                                                                       
               20/11/2003 17:47                                                                                                        
                                                                                                                                       
                                                                                                                                       




Hi Brian,

Thanks for the comments.  See inline ...

Tom P.

> -----Original Message-----
> From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> Sent: Thursday, November 20, 2003 11:05 AM
> To: 'Phelan, Tom'; 'Jonathan Rosenberg'
> Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
>
>
> To me, this is just nonsense.
> I can't imagine anyone deploying such a solution.
>
> This is forcing the problem to fit the solution, and it's not
> the right way to go.  If you want it to work, the protocol
> has to take the characteristics of the flows as they are, and not
> as you would like them to be.  Real time media, streaming or realtime,
> is either constant bit rate, or pretty close to it.  That's how it
> works.  Our challenge is to make the protocols work with this data,
> not change the data, especially silly ideas like doubling the
> bandwidth.
>
> We make video telephones.  They nominally send 4 megabits.  Would
> you like us to double that, under any circumstances?
>
> I thought not.

Would you prefer to be forced unnecessarily into packet loss and a lower
encoding rate by greedy TCP applications?  Or do you have another solution?

I don't think I'm forcing the problem to fit the solution, just perhaps that
the problem I'm trying to solve is different than you perceive.  The problem
that I'm trying to solve is how can an application that is content with a
maximum bandwidth compete with TCP applications that are insatiably greedy,
in a way that satisfies some concept of fairness in all directions?

The solution that I see is to provide a buffer against TCP probing.  This
would involve sending more data than is nominally necessary.  If the network
has capacity for more than 4M bps, then why shouldn't a media app make use
of it?  A TCP app would.

I must admit that my initial reactions to suggestions like this were very
similar to yours.  I encourage you to look at the draft for more detail on
the issues that convinced me otherwise.

>
> Brian
>
> > -----Original Message-----
> > From: Phelan, Tom [mailto:tphelan@sonusnet.com]
> > Sent: Thursday, November 20, 2003 10:36 AM
> > To: 'Jonathan Rosenberg'
> > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > Subject: [AVT] RE: [MMUSIC] DCCP Media Guide
> >
> >
> > Hi Jonathan,
> >
> > Thanks for the comments.  The actual proposal is to send up
> > to double the
> > nominal transmit rate when there's available bandwidth, and
> > to reduce the
> > rate as necessary if there isn't enough bandwidth.  This
> > doubling is to
> > allow fair sharing of the chokepoint bandwidth (between DCCP
> > media apps and
> > TCP apps), and to avoid what appears to me to be the
> > potential for bullying
> > from TCP applications that want to suck up all the bandwidth
> > they can.  It
> > doesn't preclude using low-rate codecs, or mean that the
> > access link must
> > support the double rate for the connection to function well.
> >
> > This recommendation was meant for best-effort environments,
> > where the voice
> > traffic receives no special treatment, and must compete with
> > TCP apps for
> > network resources.  I can see how this would require a
> > network that gave
> > special treatment to voice packets to provision double the
> > capacity, though.
> >
> > This fairness problem is difficult.  All of the research I'm
> > familiar with
> > approaches it from the direction of other apps being fair to
> > TCP.  I'm not
> > aware of any other methods for ensuring that non-TCP apps get
> > their fair
> > share when competing with TCP.  If you (or anyone else) knows
> > of something
> > I'd appreciate the pointer.
> >
> > Tom P.
> >
> > > -----Original Message-----
> > > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > > Sent: Thursday, November 20, 2003 2:32 AM
> > > To: Phelan, Tom
> > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > Subject: Re: [MMUSIC] DCCP Media Guide
> > >
> > >
> > > Tom,
> > >
> > > I don't think this solution will be acceptable to many of
> > the network
> > > providers of interactive voip services. Many of them are
> > insistent on
> > > using low rate codecs, like g.723, in order to reduce the
> costs of
> > > capacity in their networks. Doubling the rate would defeat
> > > the purpose
> > > of using these low rate codecs in the first place.
> > >
> > > -Jonathan R.
> > >
> > > Phelan, Tom wrote:
> > >
> > >
> > > > Comments will be appreciated.  In particular, I make the
> > > suggestion that
> > > > media apps should transmit at up to twice their nominal
> > > maximum rate, when
> > > > DCCP allows it, in order to provide protection from
> competing TCP
> > > > applications.  I know that the first time I heard someone
> > > make a similar
> > > > suggestion I was horrified, but as I developed the issues
> > > it seemed to be
> > > > necessary.  Reactions to this will be appreciated.
> > > >
> > > > Tom P.
> > > >
> > > > _______________________________________________
> > > > mmusic mailing list
> > > > mmusic@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/mmusic
> > > >
> > >
> > > --
> > > Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> > > Chief Technology Officer                    Parsippany, NJ
> > 07054-2711
> > > dynamicsoft
> > > jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> > > http://www.jdrosen.net                      PHONE: (973) 952-5000
> > > http://www.dynamicsoft.com
> > >
> >
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www1.ietf.org/mailman/listinfo/avt
> >
>

_______________________________________________
mmusic mailing list
mmusic@ietf.org
https://www1.ietf.org/mailman/listinfo/mmusic






_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Phelan, Tom wrote:

> Hi Jonathan,
> 
> Thanks for the comments.  The actual proposal is to send up to double the
> nominal transmit rate when there's available bandwidth, and to reduce the
> rate as necessary if there isn't enough bandwidth.  This doubling is to
> allow fair sharing of the chokepoint bandwidth (between DCCP media apps and
> TCP apps), and to avoid what appears to me to be the potential for bullying
> from TCP applications that want to suck up all the bandwidth they can.  It
> doesn't preclude using low-rate codecs, or mean that the access link must
> support the double rate for the connection to function well.
> 
> This recommendation was meant for best-effort environments, where the voice
> traffic receives no special treatment, and must compete with TCP apps for
> network resources.  I can see how this would require a network that gave
> special treatment to voice packets to provision double the capacity, though.

What about networks where there is also a lot of UDP traffic? For 
example, these same networks providing voip traffic today. If I 
incrementally add dccp support, it seems like I will need to increase 
the provisioned capacity so I dont push out the UDP, no?

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Phelan, Tom" <tphelan@sonusnet.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
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Hi Jonathan,

See inline ...

Tom P.

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Thursday, November 20, 2003 12:35 PM
> To: Phelan, Tom
> Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> Subject: Re: [MMUSIC] DCCP Media Guide
> 
> 
> 
> 
> Phelan, Tom wrote:
> 
> > Hi Jonathan,
> > 
> > Thanks for the comments.  The actual proposal is to send up 
> to double the
> > nominal transmit rate when there's available bandwidth, and 
> to reduce the
> > rate as necessary if there isn't enough bandwidth.  This 
> doubling is to
> > allow fair sharing of the chokepoint bandwidth (between 
> DCCP media apps and
> > TCP apps), and to avoid what appears to me to be the 
> potential for bullying
> > from TCP applications that want to suck up all the 
> bandwidth they can.  It
> > doesn't preclude using low-rate codecs, or mean that the 
> access link must
> > support the double rate for the connection to function well.
> > 
> > This recommendation was meant for best-effort environments, 
> where the voice
> > traffic receives no special treatment, and must compete 
> with TCP apps for
> > network resources.  I can see how this would require a 
> network that gave
> > special treatment to voice packets to provision double the 
> capacity, though.
> 
> What about networks where there is also a lot of UDP traffic? For 
> example, these same networks providing voip traffic today. If I 
> incrementally add dccp support, it seems like I will need to increase 
> the provisioned capacity so I dont push out the UDP, no?

Yes, DCCP will behave towards UDP (and vice versa) pretty much the same as
TCP behaves toward UDP.  For streaming media apps following the guidelines
in the DCCP Media Guide, it actually can be a little better than TCP versus
UDP, because there is a maximum transmit rate.

> 
> -Jonathan R.
> -- 
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Technology Officer                    Parsippany, NJ 07054-2711
> dynamicsoft
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
> 

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Phelan, Tom" <tphelan@sonusnet.com>
To: "'philippe.gentric@philips.com'" <philippe.gentric@philips.com>,
        "Phelan, Tom" <tphelan@sonusnet.com>,
        "'Rosen, Brian'"
	 <Brian.Rosen@marconi.com>
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Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
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Hi Phillipe,

The situation I'm worried about is a little different than you describe.
Say you have a 10M bps link, over which there's a 4M bps video stream and a
long-lived TCP app.  The TCP app will take the remaining 6M bps, and then
push for more, which might cause the video stream to lose packets.  The
video stream then drops to 2M bps, and the TCP app expands to 8M bps.  This
hardly seems fair.

The video app can protect itself from this by transmitting enough more than
4M bps to buffer itself against the TCP probes.  In this case we'd come to a
relatively steady state where both apps are sending roughly 5M bps.  This
seems reasonably fair to me, since the TCP app gets what it would get if it
were competing with other TCP apps, and the video app gets to send at its
full rate.

Tom P.

> -----Original Message-----
> From: philippe.gentric@philips.com 
> [mailto:philippe.gentric@philips.com]
> Sent: Thursday, November 20, 2003 12:23 PM
> To: Phelan, Tom; 'Rosen, Brian'
> Cc: 'avt@ietf.org'; 'dccp@ietf.org'; 'mmusic@ietf.org'
> Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> 
> 
> Brian, all,
> 
> It may provide some help in understanding the situation
> to describe your exemple as follows:
> 
> * you send video at 4 Mb/s, good.
> 
> * it happens that you are competing with (a lot of ) TCP
> 
> AND that you want to be fair to it:
> 
> * so when that happens you decrease your rate to 2 Mb/s for "a while"
> because otherwise you are not fair.
> 
> * but at some moment you will want to come back up to 4 Mb/s right ?
> 
> => the point is that the **only way**  to know if you "can" 
> do that, is to try it....
> 
> regards,
> 
> 
> Philippe Gentric
> Chief Architect
> Philips Software
> philippe.gentric@philips.com
> http://www.software.philips.com
> 
> 
>                                                               
>                                                               
>            
>                                                               
>                                                               
>            
>                                                    To:   
> "'Rosen, Brian'" <Brian.Rosen@marconi.com>                    
>                 
>                                                    cc:   
> "'dccp@ietf.org'" <dccp@ietf.org>                             
>                 
>                                                     
> "'avt@ietf.org'" <avt@ietf.org>                               
>                      
>                                                     
> "'mmusic@ietf.org'" <mmusic@ietf.org>                         
>                      
>                "Phelan, Tom"                        (bcc: 
> Philippe Gentric/MP4-SUR/CE/PHILIPS)                          
>                
>                <tphelan@sonusnet.com>              Subject:   
>  RE: [AVT] RE: [MMUSIC] DCCP Media Guide                      
>            
>                                                               
>                                                               
>            
>                Sent by:                            
> Classification:                                               
>                       
>                mmusic-admin@ietf.org                          
>                                                               
>            
>                                                               
>                                                               
>            
>                20/11/2003 17:47                               
>                                                               
>            
>                                                               
>                                                               
>            
>                                                               
>                                                               
>            
> 
> 
> 
> 
> Hi Brian,
> 
> Thanks for the comments.  See inline ...
> 
> Tom P.
> 
> > -----Original Message-----
> > From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> > Sent: Thursday, November 20, 2003 11:05 AM
> > To: 'Phelan, Tom'; 'Jonathan Rosenberg'
> > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> >
> >
> > To me, this is just nonsense.
> > I can't imagine anyone deploying such a solution.
> >
> > This is forcing the problem to fit the solution, and it's not
> > the right way to go.  If you want it to work, the protocol
> > has to take the characteristics of the flows as they are, and not
> > as you would like them to be.  Real time media, streaming 
> or realtime,
> > is either constant bit rate, or pretty close to it.  That's how it
> > works.  Our challenge is to make the protocols work with this data,
> > not change the data, especially silly ideas like doubling the
> > bandwidth.
> >
> > We make video telephones.  They nominally send 4 megabits.  Would
> > you like us to double that, under any circumstances?
> >
> > I thought not.
> 
> Would you prefer to be forced unnecessarily into packet loss 
> and a lower
> encoding rate by greedy TCP applications?  Or do you have 
> another solution?
> 
> I don't think I'm forcing the problem to fit the solution, 
> just perhaps that
> the problem I'm trying to solve is different than you 
> perceive.  The problem
> that I'm trying to solve is how can an application that is 
> content with a
> maximum bandwidth compete with TCP applications that are 
> insatiably greedy,
> in a way that satisfies some concept of fairness in all directions?
> 
> The solution that I see is to provide a buffer against TCP 
> probing.  This
> would involve sending more data than is nominally necessary.  
> If the network
> has capacity for more than 4M bps, then why shouldn't a media 
> app make use
> of it?  A TCP app would.
> 
> I must admit that my initial reactions to suggestions like 
> this were very
> similar to yours.  I encourage you to look at the draft for 
> more detail on
> the issues that convinced me otherwise.
> 
> >
> > Brian
> >
> > > -----Original Message-----
> > > From: Phelan, Tom [mailto:tphelan@sonusnet.com]
> > > Sent: Thursday, November 20, 2003 10:36 AM
> > > To: 'Jonathan Rosenberg'
> > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > Subject: [AVT] RE: [MMUSIC] DCCP Media Guide
> > >
> > >
> > > Hi Jonathan,
> > >
> > > Thanks for the comments.  The actual proposal is to send up
> > > to double the
> > > nominal transmit rate when there's available bandwidth, and
> > > to reduce the
> > > rate as necessary if there isn't enough bandwidth.  This
> > > doubling is to
> > > allow fair sharing of the chokepoint bandwidth (between DCCP
> > > media apps and
> > > TCP apps), and to avoid what appears to me to be the
> > > potential for bullying
> > > from TCP applications that want to suck up all the bandwidth
> > > they can.  It
> > > doesn't preclude using low-rate codecs, or mean that the
> > > access link must
> > > support the double rate for the connection to function well.
> > >
> > > This recommendation was meant for best-effort environments,
> > > where the voice
> > > traffic receives no special treatment, and must compete with
> > > TCP apps for
> > > network resources.  I can see how this would require a
> > > network that gave
> > > special treatment to voice packets to provision double the
> > > capacity, though.
> > >
> > > This fairness problem is difficult.  All of the research I'm
> > > familiar with
> > > approaches it from the direction of other apps being fair to
> > > TCP.  I'm not
> > > aware of any other methods for ensuring that non-TCP apps get
> > > their fair
> > > share when competing with TCP.  If you (or anyone else) knows
> > > of something
> > > I'd appreciate the pointer.
> > >
> > > Tom P.
> > >
> > > > -----Original Message-----
> > > > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > > > Sent: Thursday, November 20, 2003 2:32 AM
> > > > To: Phelan, Tom
> > > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > > Subject: Re: [MMUSIC] DCCP Media Guide
> > > >
> > > >
> > > > Tom,
> > > >
> > > > I don't think this solution will be acceptable to many of
> > > the network
> > > > providers of interactive voip services. Many of them are
> > > insistent on
> > > > using low rate codecs, like g.723, in order to reduce the
> > costs of
> > > > capacity in their networks. Doubling the rate would defeat
> > > > the purpose
> > > > of using these low rate codecs in the first place.
> > > >
> > > > -Jonathan R.
> > > >
> > > > Phelan, Tom wrote:
> > > >
> > > >
> > > > > Comments will be appreciated.  In particular, I make the
> > > > suggestion that
> > > > > media apps should transmit at up to twice their nominal
> > > > maximum rate, when
> > > > > DCCP allows it, in order to provide protection from
> > competing TCP
> > > > > applications.  I know that the first time I heard someone
> > > > make a similar
> > > > > suggestion I was horrified, but as I developed the issues
> > > > it seemed to be
> > > > > necessary.  Reactions to this will be appreciated.
> > > > >
> > > > > Tom P.
> > > > >
> > > > > _______________________________________________
> > > > > mmusic mailing list
> > > > > mmusic@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/mmusic
> > > > >
> > > >
> > > > --
> > > > Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> > > > Chief Technology Officer                    Parsippany, NJ
> > > 07054-2711
> > > > dynamicsoft
> > > > jdrosen@dynamicsoft.com                     FAX:   
> (973) 952-5050
> > > > http://www.jdrosen.net                      PHONE: 
> (973) 952-5000
> > > > http://www.dynamicsoft.com
> > > >
> > >
> > > _______________________________________________
> > > Audio/Video Transport Working Group
> > > avt@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/avt
> > >
> >
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 
> 
> 
> 
> 

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Phelan, Tom'" <tphelan@sonusnet.com>
Cc: "'dccp@ietf.org'" <dccp@ietf.org>, "'avt@ietf.org'" <avt@ietf.org>
Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
Date: Thu, 20 Nov 2003 13:23:39 -0500
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I have looked carefully at the drafts.  I think I understand what 
the proposal is, and how it works.  I'll even agree that it might 
do what you think it will do, but I suspect that my device will 
cause a heck of a lot more congestion to the network if I run the 
video streams on DCCP than if I leave it on UDP.  I think the 
solution will be much worse than the problem.

The solution is just plain broken, in my opinion.  The real, hard, 
problem is that there probably is no good solution, short of fixing 
TCP, or treating real time data differently (ie QoS) in the router.  
Like it or not, RTP is not appropriate for what we usually call 
"best effort", but the problem with that statement is that it's not 
really true that we need something more than for the network to use 
its' best effort to deliver the traffic.  What is true is that other 
streams have to deal better with CBR data than they do.  It is TCP 
that is broken. I don't think there is any way around that.  
TCP makes an assumption that is invalid when real time media is part 
of the traffic mix.

There are so many problems with DCCP applied to real time media (this 
one, slow start,...), that I think we should just stop believing that 
DCCP is appropriate for RTP.

A specific comment to your message below is that if we send more data
than 4Mbit, it is just filler.  We cannot "make use of it".  We can
just congest the network more than we do.  As you might realize,
we already have problems putting 4M into the network.  More is not
better in this case.

Brian

> -----Original Message-----
> From: Phelan, Tom [mailto:tphelan@sonusnet.com]
> Sent: Thursday, November 20, 2003 11:48 AM
> To: 'Rosen, Brian'
> Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> 
> 
> Hi Brian,
> 
> Thanks for the comments.  See inline ...
> 
> Tom P.
> 
> > -----Original Message-----
> > From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> > Sent: Thursday, November 20, 2003 11:05 AM
> > To: 'Phelan, Tom'; 'Jonathan Rosenberg'
> > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> > 
> > 
> > To me, this is just nonsense.
> > I can't imagine anyone deploying such a solution.
> > 
> > This is forcing the problem to fit the solution, and it's not
> > the right way to go.  If you want it to work, the protocol
> > has to take the characteristics of the flows as they are, and not
> > as you would like them to be.  Real time media, streaming 
> or realtime,
> > is either constant bit rate, or pretty close to it.  That's how it 
> > works.  Our challenge is to make the protocols work with this data,
> > not change the data, especially silly ideas like doubling the 
> > bandwidth.
> > 
> > We make video telephones.  They nominally send 4 megabits.  Would
> > you like us to double that, under any circumstances?  
> > 
> > I thought not. 
> 
> Would you prefer to be forced unnecessarily into packet loss 
> and a lower
> encoding rate by greedy TCP applications?  Or do you have 
> another solution?
> 
> I don't think I'm forcing the problem to fit the solution, 
> just perhaps that
> the problem I'm trying to solve is different than you 
> perceive.  The problem
> that I'm trying to solve is how can an application that is 
> content with a
> maximum bandwidth compete with TCP applications that are 
> insatiably greedy,
> in a way that satisfies some concept of fairness in all directions?
> 
> The solution that I see is to provide a buffer against TCP 
> probing.  This
> would involve sending more data than is nominally necessary.  
> If the network
> has capacity for more than 4M bps, then why shouldn't a media 
> app make use
> of it?  A TCP app would.
> 
> I must admit that my initial reactions to suggestions like 
> this were very
> similar to yours.  I encourage you to look at the draft for 
> more detail on
> the issues that convinced me otherwise.
> 
> > 
> > Brian
> > 
> > > -----Original Message-----
> > > From: Phelan, Tom [mailto:tphelan@sonusnet.com]
> > > Sent: Thursday, November 20, 2003 10:36 AM
> > > To: 'Jonathan Rosenberg'
> > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > Subject: [AVT] RE: [MMUSIC] DCCP Media Guide
> > > 
> > > 
> > > Hi Jonathan,
> > > 
> > > Thanks for the comments.  The actual proposal is to send up 
> > > to double the
> > > nominal transmit rate when there's available bandwidth, and 
> > > to reduce the
> > > rate as necessary if there isn't enough bandwidth.  This 
> > > doubling is to
> > > allow fair sharing of the chokepoint bandwidth (between DCCP 
> > > media apps and
> > > TCP apps), and to avoid what appears to me to be the 
> > > potential for bullying
> > > from TCP applications that want to suck up all the bandwidth 
> > > they can.  It
> > > doesn't preclude using low-rate codecs, or mean that the 
> > > access link must
> > > support the double rate for the connection to function well.
> > > 
> > > This recommendation was meant for best-effort environments, 
> > > where the voice
> > > traffic receives no special treatment, and must compete with 
> > > TCP apps for
> > > network resources.  I can see how this would require a 
> > > network that gave
> > > special treatment to voice packets to provision double the 
> > > capacity, though.
> > > 
> > > This fairness problem is difficult.  All of the research I'm 
> > > familiar with
> > > approaches it from the direction of other apps being fair to 
> > > TCP.  I'm not
> > > aware of any other methods for ensuring that non-TCP apps get 
> > > their fair
> > > share when competing with TCP.  If you (or anyone else) knows 
> > > of something
> > > I'd appreciate the pointer.
> > > 
> > > Tom P.
> > > 
> > > > -----Original Message-----
> > > > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > > > Sent: Thursday, November 20, 2003 2:32 AM
> > > > To: Phelan, Tom
> > > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > > Subject: Re: [MMUSIC] DCCP Media Guide
> > > > 
> > > > 
> > > > Tom,
> > > > 
> > > > I don't think this solution will be acceptable to many of 
> > > the network 
> > > > providers of interactive voip services. Many of them are 
> > > insistent on 
> > > > using low rate codecs, like g.723, in order to reduce the 
> > costs of 
> > > > capacity in their networks. Doubling the rate would defeat 
> > > > the purpose 
> > > > of using these low rate codecs in the first place.
> > > > 
> > > > -Jonathan R.
> > > > 
> > > > Phelan, Tom wrote:
> > > > 
> > > > 
> > > > > Comments will be appreciated.  In particular, I make the 
> > > > suggestion that
> > > > > media apps should transmit at up to twice their nominal 
> > > > maximum rate, when
> > > > > DCCP allows it, in order to provide protection from 
> > competing TCP
> > > > > applications.  I know that the first time I heard someone 
> > > > make a similar
> > > > > suggestion I was horrified, but as I developed the issues 
> > > > it seemed to be
> > > > > necessary.  Reactions to this will be appreciated.
> > > > > 
> > > > > Tom P.
> > > > > 
> > > > > _______________________________________________
> > > > > mmusic mailing list
> > > > > mmusic@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/mmusic
> > > > > 
> > > > 
> > > > -- 
> > > > Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> > > > Chief Technology Officer                    Parsippany, NJ 
> > > 07054-2711
> > > > dynamicsoft
> > > > jdrosen@dynamicsoft.com                     FAX:   
> (973) 952-5050
> > > > http://www.jdrosen.net                      PHONE: 
> (973) 952-5000
> > > > http://www.dynamicsoft.com
> > > > 
> > > 
> > > _______________________________________________
> > > Audio/Video Transport Working Group
> > > avt@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/avt
> > > 
> > 
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
> 

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Phelan, Tom" <tphelan@sonusnet.com>
To: "'Rosen, Brian'" <Brian.Rosen@marconi.com>,
        "Phelan, Tom"
	 <tphelan@sonusnet.com>
Cc: "'dccp@ietf.org'" <dccp@ietf.org>, "'avt@ietf.org'" <avt@ietf.org>
Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
Date: Thu, 20 Nov 2003 13:44:47 -0500
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Hi Brian,

I have sympathy for your opinion.  I despair about the possibilities of
changing TCP or getting widespread deployment of suitable QoS mechanisms,
though, so I'm not quite ready to give up on DCCP yet ...

Tom P.

> -----Original Message-----
> From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> Sent: Thursday, November 20, 2003 1:24 PM
> To: 'Phelan, Tom'
> Cc: 'dccp@ietf.org'; 'avt@ietf.org'
> Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> 
> 
> I have looked carefully at the drafts.  I think I understand what 
> the proposal is, and how it works.  I'll even agree that it might 
> do what you think it will do, but I suspect that my device will 
> cause a heck of a lot more congestion to the network if I run the 
> video streams on DCCP than if I leave it on UDP.  I think the 
> solution will be much worse than the problem.
> 
> The solution is just plain broken, in my opinion.  The real, hard, 
> problem is that there probably is no good solution, short of fixing 
> TCP, or treating real time data differently (ie QoS) in the router.  
> Like it or not, RTP is not appropriate for what we usually call 
> "best effort", but the problem with that statement is that it's not 
> really true that we need something more than for the network to use 
> its' best effort to deliver the traffic.  What is true is that other 
> streams have to deal better with CBR data than they do.  It is TCP 
> that is broken. I don't think there is any way around that.  
> TCP makes an assumption that is invalid when real time media is part 
> of the traffic mix.
> 
> There are so many problems with DCCP applied to real time media (this 
> one, slow start,...), that I think we should just stop believing that 
> DCCP is appropriate for RTP.
> 
> A specific comment to your message below is that if we send more data
> than 4Mbit, it is just filler.  We cannot "make use of it".  We can
> just congest the network more than we do.  As you might realize,
> we already have problems putting 4M into the network.  More is not
> better in this case.
> 
> Brian
> 
> > -----Original Message-----
> > From: Phelan, Tom [mailto:tphelan@sonusnet.com]
> > Sent: Thursday, November 20, 2003 11:48 AM
> > To: 'Rosen, Brian'
> > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> > 
> > 
> > Hi Brian,
> > 
> > Thanks for the comments.  See inline ...
> > 
> > Tom P.
> > 
> > > -----Original Message-----
> > > From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> > > Sent: Thursday, November 20, 2003 11:05 AM
> > > To: 'Phelan, Tom'; 'Jonathan Rosenberg'
> > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> > > 
> > > 
> > > To me, this is just nonsense.
> > > I can't imagine anyone deploying such a solution.
> > > 
> > > This is forcing the problem to fit the solution, and it's not
> > > the right way to go.  If you want it to work, the protocol
> > > has to take the characteristics of the flows as they are, and not
> > > as you would like them to be.  Real time media, streaming 
> > or realtime,
> > > is either constant bit rate, or pretty close to it.  
> That's how it 
> > > works.  Our challenge is to make the protocols work with 
> this data,
> > > not change the data, especially silly ideas like doubling the 
> > > bandwidth.
> > > 
> > > We make video telephones.  They nominally send 4 megabits.  Would
> > > you like us to double that, under any circumstances?  
> > > 
> > > I thought not. 
> > 
> > Would you prefer to be forced unnecessarily into packet loss 
> > and a lower
> > encoding rate by greedy TCP applications?  Or do you have 
> > another solution?
> > 
> > I don't think I'm forcing the problem to fit the solution, 
> > just perhaps that
> > the problem I'm trying to solve is different than you 
> > perceive.  The problem
> > that I'm trying to solve is how can an application that is 
> > content with a
> > maximum bandwidth compete with TCP applications that are 
> > insatiably greedy,
> > in a way that satisfies some concept of fairness in all directions?
> > 
> > The solution that I see is to provide a buffer against TCP 
> > probing.  This
> > would involve sending more data than is nominally necessary.  
> > If the network
> > has capacity for more than 4M bps, then why shouldn't a media 
> > app make use
> > of it?  A TCP app would.
> > 
> > I must admit that my initial reactions to suggestions like 
> > this were very
> > similar to yours.  I encourage you to look at the draft for 
> > more detail on
> > the issues that convinced me otherwise.
> > 
> > > 
> > > Brian
> > > 
> > > > -----Original Message-----
> > > > From: Phelan, Tom [mailto:tphelan@sonusnet.com]
> > > > Sent: Thursday, November 20, 2003 10:36 AM
> > > > To: 'Jonathan Rosenberg'
> > > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > > Subject: [AVT] RE: [MMUSIC] DCCP Media Guide
> > > > 
> > > > 
> > > > Hi Jonathan,
> > > > 
> > > > Thanks for the comments.  The actual proposal is to send up 
> > > > to double the
> > > > nominal transmit rate when there's available bandwidth, and 
> > > > to reduce the
> > > > rate as necessary if there isn't enough bandwidth.  This 
> > > > doubling is to
> > > > allow fair sharing of the chokepoint bandwidth (between DCCP 
> > > > media apps and
> > > > TCP apps), and to avoid what appears to me to be the 
> > > > potential for bullying
> > > > from TCP applications that want to suck up all the bandwidth 
> > > > they can.  It
> > > > doesn't preclude using low-rate codecs, or mean that the 
> > > > access link must
> > > > support the double rate for the connection to function well.
> > > > 
> > > > This recommendation was meant for best-effort environments, 
> > > > where the voice
> > > > traffic receives no special treatment, and must compete with 
> > > > TCP apps for
> > > > network resources.  I can see how this would require a 
> > > > network that gave
> > > > special treatment to voice packets to provision double the 
> > > > capacity, though.
> > > > 
> > > > This fairness problem is difficult.  All of the research I'm 
> > > > familiar with
> > > > approaches it from the direction of other apps being fair to 
> > > > TCP.  I'm not
> > > > aware of any other methods for ensuring that non-TCP apps get 
> > > > their fair
> > > > share when competing with TCP.  If you (or anyone else) knows 
> > > > of something
> > > > I'd appreciate the pointer.
> > > > 
> > > > Tom P.
> > > > 
> > > > > -----Original Message-----
> > > > > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > > > > Sent: Thursday, November 20, 2003 2:32 AM
> > > > > To: Phelan, Tom
> > > > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > > > Subject: Re: [MMUSIC] DCCP Media Guide
> > > > > 
> > > > > 
> > > > > Tom,
> > > > > 
> > > > > I don't think this solution will be acceptable to many of 
> > > > the network 
> > > > > providers of interactive voip services. Many of them are 
> > > > insistent on 
> > > > > using low rate codecs, like g.723, in order to reduce the 
> > > costs of 
> > > > > capacity in their networks. Doubling the rate would defeat 
> > > > > the purpose 
> > > > > of using these low rate codecs in the first place.
> > > > > 
> > > > > -Jonathan R.
> > > > > 
> > > > > Phelan, Tom wrote:
> > > > > 
> > > > > 
> > > > > > Comments will be appreciated.  In particular, I make the 
> > > > > suggestion that
> > > > > > media apps should transmit at up to twice their nominal 
> > > > > maximum rate, when
> > > > > > DCCP allows it, in order to provide protection from 
> > > competing TCP
> > > > > > applications.  I know that the first time I heard someone 
> > > > > make a similar
> > > > > > suggestion I was horrified, but as I developed the issues 
> > > > > it seemed to be
> > > > > > necessary.  Reactions to this will be appreciated.
> > > > > > 
> > > > > > Tom P.
> > > > > > 
> > > > > > _______________________________________________
> > > > > > mmusic mailing list
> > > > > > mmusic@ietf.org
> > > > > > https://www1.ietf.org/mailman/listinfo/mmusic
> > > > > > 
> > > > > 
> > > > > -- 
> > > > > Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> > > > > Chief Technology Officer                    Parsippany, NJ 
> > > > 07054-2711
> > > > > dynamicsoft
> > > > > jdrosen@dynamicsoft.com                     FAX:   
> > (973) 952-5050
> > > > > http://www.jdrosen.net                      PHONE: 
> > (973) 952-5000
> > > > > http://www.dynamicsoft.com
> > > > > 
> > > > 
> > > > _______________________________________________
> > > > Audio/Video Transport Working Group
> > > > avt@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/avt
> > > > 
> > > 
> > 
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www1.ietf.org/mailman/listinfo/avt
> > 
> 

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Thu, 20 Nov 2003 13:26:16 -0600
From: Qiaobing Xie <Qiaobing.Xie@motorola.com>
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Subject: Re: [dccp] RE: [AVT] RE: [MMUSIC] DCCP Media Guide
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Hi, Tom,

It seems to me that you are basically: 1) questioning the fairness of TCP itself
and, 2) you want to do something in the app level to outmaneuver TCP so as to
achieve your term of fairness.

I am not convinced this is the right thing to do with DCCP.

I remember similar issue was discussed on tsvwg list (I am cc'ing it). Maybe
people in tsvwg can share their thoughts on this as well.

regards,
-Qiaobing

"Phelan, Tom" wrote:

> Hi Phillipe,
>
> The situation I'm worried about is a little different than you describe.
> Say you have a 10M bps link, over which there's a 4M bps video stream and a
> long-lived TCP app.  The TCP app will take the remaining 6M bps, and then
> push for more, which might cause the video stream to lose packets.  The
> video stream then drops to 2M bps, and the TCP app expands to 8M bps.  This
> hardly seems fair.
>
> The video app can protect itself from this by transmitting enough more than
> 4M bps to buffer itself against the TCP probes.  In this case we'd come to a
> relatively steady state where both apps are sending roughly 5M bps.  This
> seems reasonably fair to me, since the TCP app gets what it would get if it
> were competing with other TCP apps, and the video app gets to send at its
> full rate.
>
> Tom P.
>
> > -----Original Message-----
> > From: philippe.gentric@philips.com
> > [mailto:philippe.gentric@philips.com]
> > Sent: Thursday, November 20, 2003 12:23 PM
> > To: Phelan, Tom; 'Rosen, Brian'
> > Cc: 'avt@ietf.org'; 'dccp@ietf.org'; 'mmusic@ietf.org'
> > Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> >
> >
> > Brian, all,
> >
> > It may provide some help in understanding the situation
> > to describe your exemple as follows:
> >
> > * you send video at 4 Mb/s, good.
> >
> > * it happens that you are competing with (a lot of ) TCP
> >
> > AND that you want to be fair to it:
> >
> > * so when that happens you decrease your rate to 2 Mb/s for "a while"
> > because otherwise you are not fair.
> >
> > * but at some moment you will want to come back up to 4 Mb/s right ?
> >
> > => the point is that the **only way**  to know if you "can"
> > do that, is to try it....
> >
> > regards,
> >
> >
> > Philippe Gentric
> > Chief Architect
> > Philips Software
> > philippe.gentric@philips.com
> > http://www.software.philips.com
> >
> >
> >
> >
> >
> >
> >
> >
> >                                                    To:
> > "'Rosen, Brian'" <Brian.Rosen@marconi.com>
> >
> >                                                    cc:
> > "'dccp@ietf.org'" <dccp@ietf.org>
> >
> >
> > "'avt@ietf.org'" <avt@ietf.org>
> >
> >
> > "'mmusic@ietf.org'" <mmusic@ietf.org>
> >
> >                "Phelan, Tom"                        (bcc:
> > Philippe Gentric/MP4-SUR/CE/PHILIPS)
> >
> >                <tphelan@sonusnet.com>              Subject:
> >  RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> >
> >
> >
> >
> >                Sent by:
> > Classification:
> >
> >                mmusic-admin@ietf.org
> >
> >
> >
> >
> >
> >                20/11/2003 17:47
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > Hi Brian,
> >
> > Thanks for the comments.  See inline ...
> >
> > Tom P.
> >
> > > -----Original Message-----
> > > From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> > > Sent: Thursday, November 20, 2003 11:05 AM
> > > To: 'Phelan, Tom'; 'Jonathan Rosenberg'
> > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> > >
> > >
> > > To me, this is just nonsense.
> > > I can't imagine anyone deploying such a solution.
> > >
> > > This is forcing the problem to fit the solution, and it's not
> > > the right way to go.  If you want it to work, the protocol
> > > has to take the characteristics of the flows as they are, and not
> > > as you would like them to be.  Real time media, streaming
> > or realtime,
> > > is either constant bit rate, or pretty close to it.  That's how it
> > > works.  Our challenge is to make the protocols work with this data,
> > > not change the data, especially silly ideas like doubling the
> > > bandwidth.
> > >
> > > We make video telephones.  They nominally send 4 megabits.  Would
> > > you like us to double that, under any circumstances?
> > >
> > > I thought not.
> >
> > Would you prefer to be forced unnecessarily into packet loss
> > and a lower
> > encoding rate by greedy TCP applications?  Or do you have
> > another solution?
> >
> > I don't think I'm forcing the problem to fit the solution,
> > just perhaps that
> > the problem I'm trying to solve is different than you
> > perceive.  The problem
> > that I'm trying to solve is how can an application that is
> > content with a
> > maximum bandwidth compete with TCP applications that are
> > insatiably greedy,
> > in a way that satisfies some concept of fairness in all directions?
> >
> > The solution that I see is to provide a buffer against TCP
> > probing.  This
> > would involve sending more data than is nominally necessary.
> > If the network
> > has capacity for more than 4M bps, then why shouldn't a media
> > app make use
> > of it?  A TCP app would.
> >
> > I must admit that my initial reactions to suggestions like
> > this were very
> > similar to yours.  I encourage you to look at the draft for
> > more detail on
> > the issues that convinced me otherwise.
> >
> > >
> > > Brian
> > >
> > > > -----Original Message-----
> > > > From: Phelan, Tom [mailto:tphelan@sonusnet.com]
> > > > Sent: Thursday, November 20, 2003 10:36 AM
> > > > To: 'Jonathan Rosenberg'
> > > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > > Subject: [AVT] RE: [MMUSIC] DCCP Media Guide
> > > >
> > > >
> > > > Hi Jonathan,
> > > >
> > > > Thanks for the comments.  The actual proposal is to send up
> > > > to double the
> > > > nominal transmit rate when there's available bandwidth, and
> > > > to reduce the
> > > > rate as necessary if there isn't enough bandwidth.  This
> > > > doubling is to
> > > > allow fair sharing of the chokepoint bandwidth (between DCCP
> > > > media apps and
> > > > TCP apps), and to avoid what appears to me to be the
> > > > potential for bullying
> > > > from TCP applications that want to suck up all the bandwidth
> > > > they can.  It
> > > > doesn't preclude using low-rate codecs, or mean that the
> > > > access link must
> > > > support the double rate for the connection to function well.
> > > >
> > > > This recommendation was meant for best-effort environments,
> > > > where the voice
> > > > traffic receives no special treatment, and must compete with
> > > > TCP apps for
> > > > network resources.  I can see how this would require a
> > > > network that gave
> > > > special treatment to voice packets to provision double the
> > > > capacity, though.
> > > >
> > > > This fairness problem is difficult.  All of the research I'm
> > > > familiar with
> > > > approaches it from the direction of other apps being fair to
> > > > TCP.  I'm not
> > > > aware of any other methods for ensuring that non-TCP apps get
> > > > their fair
> > > > share when competing with TCP.  If you (or anyone else) knows
> > > > of something
> > > > I'd appreciate the pointer.
> > > >
> > > > Tom P.
> > > >
> > > > > -----Original Message-----
> > > > > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > > > > Sent: Thursday, November 20, 2003 2:32 AM
> > > > > To: Phelan, Tom
> > > > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > > > Subject: Re: [MMUSIC] DCCP Media Guide
> > > > >
> > > > >
> > > > > Tom,
> > > > >
> > > > > I don't think this solution will be acceptable to many of
> > > > the network
> > > > > providers of interactive voip services. Many of them are
> > > > insistent on
> > > > > using low rate codecs, like g.723, in order to reduce the
> > > costs of
> > > > > capacity in their networks. Doubling the rate would defeat
> > > > > the purpose
> > > > > of using these low rate codecs in the first place.
> > > > >
> > > > > -Jonathan R.
> > > > >
> > > > > Phelan, Tom wrote:
> > > > >
> > > > >
> > > > > > Comments will be appreciated.  In particular, I make the
> > > > > suggestion that
> > > > > > media apps should transmit at up to twice their nominal
> > > > > maximum rate, when
> > > > > > DCCP allows it, in order to provide protection from
> > > competing TCP
> > > > > > applications.  I know that the first time I heard someone
> > > > > make a similar
> > > > > > suggestion I was horrified, but as I developed the issues
> > > > > it seemed to be
> > > > > > necessary.  Reactions to this will be appreciated.
> > > > > >
> > > > > > Tom P.
> > > > > >
> > > > > > _______________________________________________
> > > > > > mmusic mailing list
> > > > > > mmusic@ietf.org
> > > > > > https://www1.ietf.org/mailman/listinfo/mmusic
> > > > > >
> > > > >
> > > > > --
> > > > > Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> > > > > Chief Technology Officer                    Parsippany, NJ
> > > > 07054-2711
> > > > > dynamicsoft
> > > > > jdrosen@dynamicsoft.com                     FAX:
> > (973) 952-5050
> > > > > http://www.jdrosen.net                      PHONE:
> > (973) 952-5000
> > > > > http://www.dynamicsoft.com
> > > > >
> > > >
> > > > _______________________________________________
> > > > Audio/Video Transport Working Group
> > > > avt@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/avt
> > > >
> > >
> >
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www1.ietf.org/mailman/listinfo/mmusic
> >
> >
> >
> >
> >
>
> _______________________________________________
> dccp IETF mailing list: dccp@ietf.org
> list info:  https://www1.ietf.org/mailman/listinfo/dccp
> wg charter: http://www.ietf.org/html.charters/dccp-charter.html


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Nov 20 15:28:28 2003
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'philippe.gentric@philips.com'" <philippe.gentric@philips.com>,
        "Phelan, Tom" <tphelan@sonusnet.com>
Cc: "'avt@ietf.org'" <avt@ietf.org>, "'dccp@ietf.org'" <dccp@ietf.org>
Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
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Phillipe

You have made invalid assumptions:
	1. That audio/video media data rates will be changed dynamically
		as network traffic changes.  This is very unlikely.  It may
		be that you get so much loss that you drop to a lower data
		rate, although I suspect that will be rare.  When you do,
		you will never revert to the higher rate.  This is because
		humans hate to see the change, and minimizing changes is 
		more desirable than keeping average quality higher.

	2. Real Time apps have no interest in being "fair".  They are all
		or nothing. If you admit them to the network, you expect
		the streams to be maintained.  If you can't maintain them,
		don't start them.  They don't want to be fair to TCP, and
they
		don't want to be fair to other RTP.  They just want to send
		data at the rate they were advised to do by signalling.
		They can tolerate some loss.  Misordering=loss.  Duplicates=
		wasted bandwith.  That's a whole lot different from TCP
		traffic.  

I can imagine working on protocols which allowed CBR and "best effort"
traffic to better coexist, but I think it's only possible if the best
effort traffic recognized the presence of the CBR traffic and reacted
appropriately.  Otherwise, you need to use QoS mechanisms to separate
the traffic.

Brian




> -----Original Message-----
> From: philippe.gentric@philips.com 
> [mailto:philippe.gentric@philips.com]
> Sent: Thursday, November 20, 2003 12:23 PM
> To: Phelan, Tom; 'Rosen, Brian'
> Cc: 'avt@ietf.org'; 'dccp@ietf.org'; 'mmusic@ietf.org'
> Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> 
> 
> Brian, all,
> 
> It may provide some help in understanding the situation
> to describe your exemple as follows:
> 
> * you send video at 4 Mb/s, good.
> 
> * it happens that you are competing with (a lot of ) TCP
> 
> AND that you want to be fair to it:
> 
> * so when that happens you decrease your rate to 2 Mb/s for "a while"
> because otherwise you are not fair.
> 
> * but at some moment you will want to come back up to 4 Mb/s right ?
> 
> => the point is that the **only way**  to know if you "can" 
> do that, is to try it....
> 
> regards,
> 
> 
> Philippe Gentric
> Chief Architect
> Philips Software
> philippe.gentric@philips.com
> http://www.software.philips.com
> 
> 
>                                                               
>                                                               
>            
>                                                               
>                                                               
>            
>                                                    To:   
> "'Rosen, Brian'" <Brian.Rosen@marconi.com>                    
>                 
>                                                    cc:   
> "'dccp@ietf.org'" <dccp@ietf.org>                             
>                 
>                                                     
> "'avt@ietf.org'" <avt@ietf.org>                               
>                      
>                                                     
> "'mmusic@ietf.org'" <mmusic@ietf.org>                         
>                      
>                "Phelan, Tom"                        (bcc: 
> Philippe Gentric/MP4-SUR/CE/PHILIPS)                          
>                
>                <tphelan@sonusnet.com>              Subject:   
>  RE: [AVT] RE: [MMUSIC] DCCP Media Guide                      
>            
>                                                               
>                                                               
>            
>                Sent by:                            
> Classification:                                               
>                       
>                mmusic-admin@ietf.org                          
>                                                               
>            
>                                                               
>                                                               
>            
>                20/11/2003 17:47                               
>                                                               
>            
>                                                               
>                                                               
>            
>                                                               
>                                                               
>            
> 
> 
> 
> 
> Hi Brian,
> 
> Thanks for the comments.  See inline ...
> 
> Tom P.
> 
> > -----Original Message-----
> > From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> > Sent: Thursday, November 20, 2003 11:05 AM
> > To: 'Phelan, Tom'; 'Jonathan Rosenberg'
> > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> >
> >
> > To me, this is just nonsense.
> > I can't imagine anyone deploying such a solution.
> >
> > This is forcing the problem to fit the solution, and it's not
> > the right way to go.  If you want it to work, the protocol
> > has to take the characteristics of the flows as they are, and not
> > as you would like them to be.  Real time media, streaming 
> or realtime,
> > is either constant bit rate, or pretty close to it.  That's how it
> > works.  Our challenge is to make the protocols work with this data,
> > not change the data, especially silly ideas like doubling the
> > bandwidth.
> >
> > We make video telephones.  They nominally send 4 megabits.  Would
> > you like us to double that, under any circumstances?
> >
> > I thought not.
> 
> Would you prefer to be forced unnecessarily into packet loss 
> and a lower
> encoding rate by greedy TCP applications?  Or do you have 
> another solution?
> 
> I don't think I'm forcing the problem to fit the solution, 
> just perhaps that
> the problem I'm trying to solve is different than you 
> perceive.  The problem
> that I'm trying to solve is how can an application that is 
> content with a
> maximum bandwidth compete with TCP applications that are 
> insatiably greedy,
> in a way that satisfies some concept of fairness in all directions?
> 
> The solution that I see is to provide a buffer against TCP 
> probing.  This
> would involve sending more data than is nominally necessary.  
> If the network
> has capacity for more than 4M bps, then why shouldn't a media 
> app make use
> of it?  A TCP app would.
> 
> I must admit that my initial reactions to suggestions like 
> this were very
> similar to yours.  I encourage you to look at the draft for 
> more detail on
> the issues that convinced me otherwise.
> 
> >
> > Brian
> >
> > > -----Original Message-----
> > > From: Phelan, Tom [mailto:tphelan@sonusnet.com]
> > > Sent: Thursday, November 20, 2003 10:36 AM
> > > To: 'Jonathan Rosenberg'
> > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > Subject: [AVT] RE: [MMUSIC] DCCP Media Guide
> > >
> > >
> > > Hi Jonathan,
> > >
> > > Thanks for the comments.  The actual proposal is to send up
> > > to double the
> > > nominal transmit rate when there's available bandwidth, and
> > > to reduce the
> > > rate as necessary if there isn't enough bandwidth.  This
> > > doubling is to
> > > allow fair sharing of the chokepoint bandwidth (between DCCP
> > > media apps and
> > > TCP apps), and to avoid what appears to me to be the
> > > potential for bullying
> > > from TCP applications that want to suck up all the bandwidth
> > > they can.  It
> > > doesn't preclude using low-rate codecs, or mean that the
> > > access link must
> > > support the double rate for the connection to function well.
> > >
> > > This recommendation was meant for best-effort environments,
> > > where the voice
> > > traffic receives no special treatment, and must compete with
> > > TCP apps for
> > > network resources.  I can see how this would require a
> > > network that gave
> > > special treatment to voice packets to provision double the
> > > capacity, though.
> > >
> > > This fairness problem is difficult.  All of the research I'm
> > > familiar with
> > > approaches it from the direction of other apps being fair to
> > > TCP.  I'm not
> > > aware of any other methods for ensuring that non-TCP apps get
> > > their fair
> > > share when competing with TCP.  If you (or anyone else) knows
> > > of something
> > > I'd appreciate the pointer.
> > >
> > > Tom P.
> > >
> > > > -----Original Message-----
> > > > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > > > Sent: Thursday, November 20, 2003 2:32 AM
> > > > To: Phelan, Tom
> > > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > > Subject: Re: [MMUSIC] DCCP Media Guide
> > > >
> > > >
> > > > Tom,
> > > >
> > > > I don't think this solution will be acceptable to many of
> > > the network
> > > > providers of interactive voip services. Many of them are
> > > insistent on
> > > > using low rate codecs, like g.723, in order to reduce the
> > costs of
> > > > capacity in their networks. Doubling the rate would defeat
> > > > the purpose
> > > > of using these low rate codecs in the first place.
> > > >
> > > > -Jonathan R.
> > > >
> > > > Phelan, Tom wrote:
> > > >
> > > >
> > > > > Comments will be appreciated.  In particular, I make the
> > > > suggestion that
> > > > > media apps should transmit at up to twice their nominal
> > > > maximum rate, when
> > > > > DCCP allows it, in order to provide protection from
> > competing TCP
> > > > > applications.  I know that the first time I heard someone
> > > > make a similar
> > > > > suggestion I was horrified, but as I developed the issues
> > > > it seemed to be
> > > > > necessary.  Reactions to this will be appreciated.
> > > > >
> > > > > Tom P.
> > > > >
> > > > > _______________________________________________
> > > > > mmusic mailing list
> > > > > mmusic@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/mmusic
> > > > >
> > > >
> > > > --
> > > > Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> > > > Chief Technology Officer                    Parsippany, NJ
> > > 07054-2711
> > > > dynamicsoft
> > > > jdrosen@dynamicsoft.com                     FAX:   
> (973) 952-5050
> > > > http://www.jdrosen.net                      PHONE: 
> (973) 952-5000
> > > > http://www.dynamicsoft.com
> > > >
> > >
> > > _______________________________________________
> > > Audio/Video Transport Working Group
> > > avt@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/avt
> > >
> >
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 
> 
> 
> 
> 

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Nov 20 17:16:32 2003
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Caitlin Bestler'" <cait@asomi.com>,
        Qiaobing Xie
	 <Qiaobing.Xie@motorola.com>
Cc: mmusic@ietf.org, TSV WG list <tsvwg@ietf.org>,
        "Phelan, Tom"
	 <tphelan@sonusnet.com>, avt@ietf.org, dccp@ietf.org,
        philippe.gentric@philips.com
Subject: RE: [Tsvwg] Re: [dccp] RE: [AVT] RE: [MMUSIC] DCCP Media Guide
Date: Thu, 20 Nov 2003 17:14:23 -0500
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The reason to "avoid" TCP friendly rate control is, at least so far,
because there is no way yet proposed to do it without destroying the 
usability of the RTP flow.

Part of the problem is that the only "rate control" that works is to
negotiate an acceptable rate in the signalling before the stream starts.
There really isn't any other way to change the rate, at least for
interactive streams.

For non-interactive streams, you can always buffer more, which allows
you some short term variation.  I don't think that any of the proposed
mechanisms would actually work even in that case, but maybe there is
a way to do something.  Non interactive streams need CBR averaged over
the buffer time.  That can be up to seconds in some cases, hundreds
of milliseconds in other cases, but over that interval, your goodput
must equate to the transmit offer rate times the interval (some small
loss rate acceptable).

It would be nice if there was a way.  I haven't seen one yet.

Brian

> -----Original Message-----
> From: Caitlin Bestler [mailto:cait@asomi.com]
> Sent: Thursday, November 20, 2003 5:02 PM
> To: Qiaobing Xie
> Cc: mmusic@ietf.org; 'Rosen, Brian'; TSV WG list; Phelan, Tom;
> avt@ietf.org; dccp@ietf.org; philippe.gentric@philips.com
> Subject: Re: [Tsvwg] Re: [dccp] RE: [AVT] RE: [MMUSIC] DCCP 
> Media Guide
> 
> 
> 
> On Nov 20, 2003, at 1:26 PM, Qiaobing Xie wrote:
> 
> > Hi, Tom,
> >
> > It seems to me that you are basically: 1) questioning the 
> fairness of 
> > TCP itself
> > and, 2) you want to do something in the app level to 
> outmaneuver TCP 
> > so as to
> > achieve your term of fairness.
> >
> > I am not convinced this is the right thing to do with DCCP.
> >
> > I remember similar issue was discussed on tsvwg list (I am 
> cc'ing it). 
> > Maybe
> > people in tsvwg can share their thoughts on this as well.
> 
> 
> The fairness of TCP itself could use some questioning -- but I do not
> believe that is a reason to evade TCP friendly rate control.
> 
> Eventually, there should be a solution for proper support of 
> predictably
> variable rate flows that have fixed delivery requirements. But that 
> solution
> should be developed considering TCP, SCTP and DCCP as a whole.
> 
> 
> _______________________________________________
> tsvwg mailing list
> tsvwg@ietf.org
> https://www1.ietf.org/mailman/listinfo/tsvwg
> 

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Nov 20 17:49:21 2003
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From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: <avt@ietf.org>, "Tom Taylor" <taylor@nortelnetworks.com>
Cc: "Toip list" <toip@snowshore.com>
Subject: RE: [AVT] VBD vs. RFC 2833 vs. V.150.1 for V.21
Date: Thu, 20 Nov 2003 23:48:48 +0100
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Tom,
If you know that the V.21 is for text telephony, you would be even better
off using RFC2793 for its transport, because it is textphone type
independant, and can be converted to IP text telephony or any other kind of
text telephony.

But if you are in a big general borderline gateway between PSTN and IP, it
may be hard to tell if V.21 signals are for text telephony or some kind of
data communication. If the origin or destination is a SIP terminal UA, you
may assume text telephony, and use RFC2793.

But for gateway to gateway traffic it may be impossible to distinguish
between the applications.
For that case, the digital transports you mention may be used, and it seems
that your conclusions are right.

If there is a slightest risk ( >0.5%) of loosing packets when transmitting
audio coded modem signals in G.711, redundancy must be used on G.711, and
that is even more bandwidth consuming.

There are textphones based on V.23 as well. They transmit at 1200 bits/s. I
think that including V.23 bit events in RFC2833 would not be good
bit-economy.

Issues like this are studied in ITU-T Q.11/16 and Q.H/16. So, I think the
question should be treated there, to find out if there is any benefit of
RFC2833-code text telephony. The idea was touched last spring, but abandoned
because of bad bit-economy and too tight coupling between the ends.

If any such coding would be of interest, there could be a reason to study
the carrierless modes separately: Baudot(1400/1800Hz FSK), EDT (=V.21 110
bits/s half duplex ) and DTMF.
The carrier based V.21 (full duplex 300 bits/s), V.23, Bell 103, and Native
V.18( based on V.21 full duplex  ) may better be transported by some byte
coded mechanism, like RFC2793 or some other relay coding.

It is the intention of RFC2793bis to offer a transport for all these modes.
So, I think that, after considering the alternatives, we would like to stick
with that code for text transport.

Agree?

Regards

Gunnar
-------------------------------------------
Gunnar Hellstrom
Omnitor AB
Renathvagen 2
SE 121 37 Johanneshov
SWEDEN
+46 8 556 002 03
Mob: +46 708 204 288
www.omnitor.se
Gunnar.Hellstrom@Omnitor.se
--------------------------------------------


-----Original Message-----
From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of Tom
Taylor
Sent: Thursday, November 20, 2003 3:34 PM
To: avt@ietf.org
Subject: [AVT] VBD vs. RFC 2833 vs. V.150.1 for V.21


I don't know if this is the right place to ask this question, or whether
I should go to Study Group 16.  During my rewrite of RFC 2833 I'm trying
to add a few words of advice about the encoding of V.21 used to carry
the conversation, as opposed to control signalling.  This comes up with
respect to text telephones.

V.21 is a frequency shift modem operating at up to 300 bits/s, one pure
tone for "0", another for "1".  By my rough calculations, using G.711
for carrying the tones takes more bandwidth than RFC 2833, assuming
similar packetization intervals.  I see it as:

G.711 = 64000 bits/s.

RFC 2833 events = 300 bit events/s x 16 bits/event report x 4 repetition
factor = 19200 bits/s.  There's more for redundancy headers, but this is
the right order of magnitude.

V.150.1 modem relay would be more compact still, since it reports octets
rather than bits per sample.

So the advice would be to carry V.21 as RFC 2833 events or use V.150.1
modem relay in preference to G.711.  Do I have it right, or was I too
careless with my calculations?

Tom Taylor


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt




_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Cc: mmusic@ietf.org, "'Rosen, Brian'" <Brian.Rosen@marconi.com>,
        TSV WG list <tsvwg@ietf.org>, "Phelan, Tom" <tphelan@sonusnet.com>,
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From: Caitlin Bestler <cait@asomi.com>
Subject: Re: [Tsvwg] Re: [dccp] RE: [AVT] RE: [MMUSIC] DCCP Media Guide
Date: Thu, 20 Nov 2003 16:01:47 -0600
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On Nov 20, 2003, at 1:26 PM, Qiaobing Xie wrote:

> Hi, Tom,
>
> It seems to me that you are basically: 1) questioning the fairness of 
> TCP itself
> and, 2) you want to do something in the app level to outmaneuver TCP 
> so as to
> achieve your term of fairness.
>
> I am not convinced this is the right thing to do with DCCP.
>
> I remember similar issue was discussed on tsvwg list (I am cc'ing it). 
> Maybe
> people in tsvwg can share their thoughts on this as well.


The fairness of TCP itself could use some questioning -- but I do not
believe that is a reason to evade TCP friendly rate control.

Eventually, there should be a solution for proper support of predictably
variable rate flows that have fixed delivery requirements. But that 
solution
should be developed considering TCP, SCTP and DCCP as a whole.


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Nov 20 19:07:23 2003
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To: Gunnar Hellstrom <gunnar.hellstrom@omnitor.se>
CC: avt@ietf.org, Toip list <toip@snowshore.com>
Subject: Re: [AVT] VBD vs. RFC 2833 vs. V.150.1 for V.21
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I forgot about RFC2793.  I'll have a look, then add it to my words of 
advice.

I'll be sending my draft out shortly to selected readers, once Henning 
gives the basic OK.  I'll make sure you're on the list -- there seemed 
to be a few oversights in the definition of V.18-related events, and 
perhaps with my new words you'll spot the gaps or else suggest changes.

Gunnar Hellstrom wrote:

> Tom,
> If you know that the V.21 is for text telephony, you would be even better
> off using RFC2793 for its transport, because it is textphone type
> independant, and can be converted to IP text telephony or any other kind of
> text telephony.
> 
> But if you are in a big general borderline gateway between PSTN and IP, it
> may be hard to tell if V.21 signals are for text telephony or some kind of
> data communication. If the origin or destination is a SIP terminal UA, you
> may assume text telephony, and use RFC2793.
> 
> But for gateway to gateway traffic it may be impossible to distinguish
> between the applications.
> For that case, the digital transports you mention may be used, and it seems
> that your conclusions are right.
> 
> If there is a slightest risk ( >0.5%) of loosing packets when transmitting
> audio coded modem signals in G.711, redundancy must be used on G.711, and
> that is even more bandwidth consuming.
> 
> There are textphones based on V.23 as well. They transmit at 1200 bits/s. I
> think that including V.23 bit events in RFC2833 would not be good
> bit-economy.
> 
> Issues like this are studied in ITU-T Q.11/16 and Q.H/16. So, I think the
> question should be treated there, to find out if there is any benefit of
> RFC2833-code text telephony. The idea was touched last spring, but abandoned
> because of bad bit-economy and too tight coupling between the ends.
> 
> If any such coding would be of interest, there could be a reason to study
> the carrierless modes separately: Baudot(1400/1800Hz FSK), EDT (=V.21 110
> bits/s half duplex ) and DTMF.
> The carrier based V.21 (full duplex 300 bits/s), V.23, Bell 103, and Native
> V.18( based on V.21 full duplex  ) may better be transported by some byte
> coded mechanism, like RFC2793 or some other relay coding.
> 
> It is the intention of RFC2793bis to offer a transport for all these modes.
> So, I think that, after considering the alternatives, we would like to stick
> with that code for text transport.
> 
> Agree?
> 
> Regards
> 
> Gunnar
> -------------------------------------------
> Gunnar Hellstrom
> Omnitor AB
> Renathvagen 2
> SE 121 37 Johanneshov
> SWEDEN
> +46 8 556 002 03
> Mob: +46 708 204 288
> www.omnitor.se
> Gunnar.Hellstrom@Omnitor.se
> --------------------------------------------
> 
> 
> -----Original Message-----
> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of Tom
> Taylor
> Sent: Thursday, November 20, 2003 3:34 PM
> To: avt@ietf.org
> Subject: [AVT] VBD vs. RFC 2833 vs. V.150.1 for V.21
> 
> 
> I don't know if this is the right place to ask this question, or whether
> I should go to Study Group 16.  During my rewrite of RFC 2833 I'm trying
> to add a few words of advice about the encoding of V.21 used to carry
> the conversation, as opposed to control signalling.  This comes up with
> respect to text telephones.
> 
> V.21 is a frequency shift modem operating at up to 300 bits/s, one pure
> tone for "0", another for "1".  By my rough calculations, using G.711
> for carrying the tones takes more bandwidth than RFC 2833, assuming
> similar packetization intervals.  I see it as:
> 
> G.711 = 64000 bits/s.
> 
> RFC 2833 events = 300 bit events/s x 16 bits/event report x 4 repetition
> factor = 19200 bits/s.  There's more for redundancy headers, but this is
> the right order of magnitude.
> 
> V.150.1 modem relay would be more compact still, since it reports octets
> rather than bits per sample.
> 
> So the advice would be to carry V.21 as RFC 2833 events or use V.150.1
> modem relay in preference to G.711.  Do I have it right, or was I too
> careless with my calculations?
> 
> Tom Taylor
> 
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
> 
> 
> 
> 


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: [AVT] Re: I-D ACTION:draft-jones-avt-text-red-00.txt
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Folks,

I just wanted to bring this particular document to your attention.  During
the review of RFC 2793bis, it was proposed by several folks that we extract
the text/red MIME sub-type registration from that document and publish it as
a separate document.  Thus, I've put for this individual contribution honor
those requests.

You'll note that I made reference to RFC 2793.  The rationale is that this
registration should have already been done and we do not necessarily want
the publication of the revisions RFC 2793 to hinder the registration of this
MIME sub-type.  In any case, I'm open to changing the references, especially
if they're both published at about the same time.

Please review this document and provide any comments.

Thanks,
Paul

----- Original Message ----- 
From: <Internet-Drafts@ietf.org>
To: <IETF-Announce:>
Cc: <avt@ietf.org>
Sent: Wednesday, November 19, 2003 3:26 PM
Subject: I-D ACTION:draft-jones-avt-text-red-00.txt


> A New Internet-Draft is available from the on-line Internet-Drafts
directories.
>
>
> Title : Registration of the text/red MIME Sub-Type
> Author(s) : P. Jones
> Filename : draft-jones-avt-text-red-00.txt
> Pages : 5
> Date : 2003-11-19
>
> This document defines the text/red MIME sub-type.  The packetization
> and usage of this MIME type, which is intended for use within SDP, is
> specified within RFC 2198 and RFC 2793, respectively.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-jones-avt-text-red-00.txt
>
> To remove yourself from the IETF Announcement list, send a message to
> ietf-announce-request with the word unsubscribe in the body of the
message.
>
> 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-jones-avt-text-red-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-jones-avt-text-red-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.
>


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Fri Nov 21 05:27:26 2003
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Subject: Re: [avt] SV: RFC2793bis and byte orders and other details
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On Friday, Nov 14, 2003, at 03:44 Europe/London, Gunnar Hellstrom wrote:
> In the avt meeting, the RFC2793bis RTP Payload for Text Conversation  
> was
> accepted as an avt workgroup item.
> I promised to produce draft-avt-rfc2793bis-00 out of
> draft-hellstrom-avt-rfc2793bis-02
> The current version is found at:
> http://www.ietf.org/internet-drafts/draft-hellstrom-avt-rfc2793bis- 
> 02.txt
>
> When I do that, I want to include actions on the comments so far.
>
> Therefore, I want to check the reaction on some suggestions.
>
> 1. Specify byte-order for the new T140block counter. ( motivation  
> below from
> Frank Shearar )
>    I doubt that it is advicable to specify byte order for one little  
> piece
> of the packet.
>    I assume that byte order issues are enough covered centrally in RTP.
>    I would like to get comments on that assumption.

You need to specify the byte order for the payload header, the RTP spec  
only defines the byte order for the standard RTP header.

> 2. Open for other redundancy schemes than RFC2198, at least for the
> audio/t140 format
>    Especially for the audio intermixed packetization audio/t140, there  
> might
> be of interest to use any of the
>    FEC mechanisms to gain reliability. It may for example be desirable  
> to
> use the same FEC mechanism for the
>    audio stream as for the interleaved text. The requirement to use  
> the same
> clock frequency for text and audio in the audio/t140 medium may also  
> make it
> hard to use RFC2198, because the maximum offset will not be long  
> enough to
> cater for transmission only every 300 ms, when the clock frequency of  
> the
> audio packetisation is used. ( for text/t140 it is 1000 Hz, and that  
> is a
> suitable frequency to allow for redundancy. )
> I recommend that I insert wording that for audio/t140 , any suitable  
> FEC
> packetisation can be used.

Yes.

> 3. IANA considerations
> The following IANA registrations shall be included in the draft:
>
> register media types for MIME and SDP usage:
>
> text/t140  - already done
> text/red
> audio/t140
>
> register the main media type  for sdp usage
> text
>
> register the attribute parameter
> cps=
>
> Register the capability for capability registration.
> text
>
> Anything more?

The "text/red" registration is now a separate draft, otherwise these  
look appropriate.

-- 
Colin Perkins
http://csperkins.org/


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Colin Perkins <csp@csperkins.org>
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On Wednesday, Nov 12, 2003, at 19:45 Europe/London, Lars-Erik Jonsson 
(LU/EAB) wrote:
> As I promised in the WG session, I'll try to share my thoughts on
> the HC over multihop/MPLS issue.
...
> However, the current draft refers to "VoIP Header Compression over 
> Multi-Hop
> Paths", which is not the same thing. The draft clearly shows that 
> compression
> over multihop paths is one thing, and compression over MPLS another. 
> Further,
> the draft takes a wider scope, and talks about header compression 
> generally,
> not only ECRTP as the charter says.
>
> The following observations can be made:
> * If work is to done for header compression over multihop and/or MPLS, 
> it
>   might make sense to not only address ECRTP.

Certainly a header compression draft for MPLS should allow the 
negotiation of ROHC as an alternative to ECRTP.

> * The only work in this that naturally could belong in the AVT WG 
> would be
>   to make sure ECRTP would work in these scenarios. In my 
> understanding,
>   this should already be the case with ECRTP (as well as some ROHC 
> profiles),
>   which is also indicated by the multihop req draft.

I believe ECRTP would work, but one of the main purposes of the 
requirements capture process is to confirm this by getting MPLS and 
ECRTP folks in the same room and comparing the environments and 
assumptions.

> * Header compression over multihop paths [ described by the multihop 
> req
>   draft as where one wants to avoid decompression/compression steps at
>   intermediate routers if the same compression scheme is supported at 
> both
>   incoming and outgoing interfaces ] is fully possible also without 
> any new
>   standards, as it would only be an implementation issue for each 
> intermediate
>   router to bind an incoming compressed stream to its corresponding 
> outgoing
>   compressed stream. In a path A->B->C, where B is the intermediate 
> router
>   negotiating use of e.g. CRTP with both A and C, it is fully up to the
>   implementation of B to avoid unnecessary decompression/compression 
> and
>   just forward compressed packets as-is ("CID routing").

Agreed, but it may be worthwhile writing a brief draft explaining the 
process, perhaps targeted at BCP or Informational?

> * The only significant standardization work I can see is to define how 
> to
>   do header compression (e.g. IPHC, CRTP/ECRTP, ROHC) over MPLS, which 
> is
>   probably a work that would require some coordination with AVT and 
> ROHC
>   individuals, but the core of that work does not naturally belong to
>   ROHC or AVT. Would not the MPLS WG be a more proper place?

Probably. The purpose of the requirements work is to understand the 
issues, and to decide if this is the right approach, or if we need to 
do something more.

Cheers,
-- 
Colin Perkins
http://csperkins.org/


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: Re: [AVT] RTP MIB Extensions
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To: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
From: Colin Perkins <csp@csperkins.org>
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On Wednesday, Nov 12, 2003, at 19:39 Europe/London, Romascanu, Dan  
(Dan) wrote:
> Please allow me to make more clear my points expressed in the final  
> minutes of the AVT WG meeting today related to  
> http://www.ietf.cnri.reston.va.us/internet-drafts/draft-clark-avt- 
> rtpmibv2-00.txt
>
> 1. This MIB includes extensions to the RTP MIB (RFC 2959) related to  
> RTCP-XR (RFC 3611). At the same time, there seems to be a need to  
> update RFC 2959 as result of the evolution of the Real Time Protocol  
> from RFC 1889 to RFC 3550. I strongly suggest to do this in two  
> separate MIBs separate documents:
>
> - a new RTP MIB, updating to RFC 2959
>
> - a supplementary RTCP-XR MIB, with the MIB objects relevant for hosts  
> or monitors running RTCP-XR.
>
> This more modular and scalable structure would allow the two different  
> MIBs to be progressed separately, without creating a tight dependency  
> among them, and would allow adding other modules in the future.

Yes, I think this is a good idea.

> 2. There is a need for a better coordination, and possibly some  
> consolidation between the RTCP-XR MIB and work being done in other WGs  
> in the IETF - especially the RMON MIB WG  
> (http://www.ietf.cnri.reston.va.us/html.charters/rmonmib-charter.html)  
> work on Application Performance Monitoring and Real-Time Application  
> Monitoring to create a consistent framework for performance  
> monitoring, and avoid duplication. While RTCP-XR covers an important  
> aspect of media monitoring when RTP is used, it does not cover the  
> whole application space where RTP streams can be used. I would be glad  
> to provide more information, or work with the authors of the different  
> proposals in AVT to try to reach this goal.

Thanks for the information - I'll look into it. If you would be willing  
to work with Alan Clark as the RTCP-XR MIB is developed, that would be  
helpful.

Thanks!
-- 
Colin Perkins
http://csperkins.org/


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: RE: [Tsvwg] Re: [dccp] RE: [AVT] RE: [MMUSIC] DCCP Media Guide
To: "Rosen, Brian" <Brian.Rosen@marconi.com>
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Brian, you wrote:

>The reason to "avoid" TCP friendly rate control is, at least so far,
>because there is no way yet proposed to do it without destroying the
>usability of the RTP flow.

I aggree this is a chalenge and especially I think that the availability
of codecs for which the rate can be rapidly changed over wide ranges is key.

>Part of the problem is that the only "rate control" that works is to
>negotiate an acceptable rate in the signalling before the stream starts.
>There really isn't any other way to change the rate, at least for
>interactive streams.

This is where progress must be done: there is a need for signaling
that the rate must change, during the streaming.

>For non-interactive streams, you can always buffer more, which allows
>you some short term variation.  I don't think that any of the proposed
>mechanisms would actually work even in that case, but maybe there is
>a way to do something.

Stream switching is a solution.

regards,

Philippe Gentric
Chief Architect
Philips Software
philippe.gentric@philips.com
http://www.software.philips.com



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: RE: [AVT] RTP MIB Extensions
Date: Fri, 21 Nov 2003 15:19:51 +0200
Message-ID: <AAB4B3D3CF0F454F98272CBE187FDE2F0425974D@is0004avexu1.global.avaya.com>
Thread-Topic: [AVT] RTP MIB Extensions
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From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
To: "Colin Perkins" <csp@csperkins.org>
Cc: <avt@ietf.org>, <mankin@psg.com>,
        "Bert Wijnen (E-mail)" <bwijnen@lucent.com>
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1. From the point of view of the AVT WG, it looks to me that updating =
the RTP MIB to reflect changes in RTC from RFC 1889 to RFC 3550 has =
higher priority.
2. I met with Alan in Minneapolis, and made quite good progress. One =
possible way to formalize the progress we made in the hallway is to work =
together on a framework document for VoIP performance monitoring. =
Hopefully we will have the time and resources to make this happen.

Regards,

Dan


> -----Original Message-----
> From: Colin Perkins [mailto:csp@csperkins.org]
> Sent: 21 November, 2003 12:47 PM
> To: Romascanu, Dan (Dan)
> Cc: avt@ietf.org; mankin@psg.com; Bert Wijnen (E-mail)
> Subject: Re: [AVT] RTP MIB Extensions
>=20
>=20
> On Wednesday, Nov 12, 2003, at 19:39 Europe/London, Romascanu, Dan =20
> (Dan) wrote:
> > Please allow me to make more clear my points expressed in=20
> the final =20
> > minutes of the AVT WG meeting today related to =20
> > http://www.ietf.cnri.reston.va.us/internet-drafts/draft-clark-avt-=20
> > rtpmibv2-00.txt
> >
> > 1. This MIB includes extensions to the RTP MIB (RFC 2959)=20
> related to =20
> > RTCP-XR (RFC 3611). At the same time, there seems to be a need to =20
> > update RFC 2959 as result of the evolution of the Real Time=20
> Protocol =20
> > from RFC 1889 to RFC 3550. I strongly suggest to do this in two =20
> > separate MIBs separate documents:
> >
> > - a new RTP MIB, updating to RFC 2959
> >
> > - a supplementary RTCP-XR MIB, with the MIB objects=20
> relevant for hosts =20
> > or monitors running RTCP-XR.
> >
> > This more modular and scalable structure would allow the=20
> two different =20
> > MIBs to be progressed separately, without creating a tight=20
> dependency =20
> > among them, and would allow adding other modules in the future.
>=20
> Yes, I think this is a good idea.
>=20
> > 2. There is a need for a better coordination, and possibly some =20
> > consolidation between the RTCP-XR MIB and work being done=20
> in other WGs =20
> > in the IETF - especially the RMON MIB WG =20
> >=20
> (http://www.ietf.cnri.reston.va.us/html.charters/rmonmib-chart
er.html) =20
> work on Application Performance Monitoring and Real-Time Application =20
> Monitoring to create a consistent framework for performance =20
> monitoring, and avoid duplication. While RTCP-XR covers an important =20
> aspect of media monitoring when RTP is used, it does not cover the =20
> whole application space where RTP streams can be used. I would be glad =
=20
> to provide more information, or work with the authors of the different =
=20
> proposals in AVT to try to reach this goal.

Thanks for the information - I'll look into it. If you would be willing  =

to work with Alan Clark as the RTCP-XR MIB is developed, that would be =20
helpful.

Thanks!
--=20
Colin Perkins
http://csperkins.org/


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: Re: [AVT] RTP MIB Extensions
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On Friday, Nov 21, 2003, at 13:19 Europe/London, Romascanu, Dan (Dan) 
wrote:
> 1. From the point of view of the AVT WG, it looks to me that updating 
> the RTP MIB to reflect changes in RTC from RFC 1889 to RFC 3550 has 
> higher priority.

If the MIB is used, then this certainly needs to be done. Maybe anyone 
on the list who uses the MIB would like to comment?

> 2. I met with Alan in Minneapolis, and made quite good progress. One 
> possible way to formalize the progress we made in the hallway is to 
> work together on a framework document for VoIP performance monitoring. 
> Hopefully we will have the time and resources to make this happen.

Good to hear, I look forward to seeing the result.

Colin


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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To: "Phelan, Tom" <tphelan@sonusnet.com>
From: "Michael A. Ramalho" <mramalho@cisco.com>
Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
Cc: "'Rosen, Brian'" <Brian.Rosen@marconi.com>,
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At 11:47 AM 11/20/2003 -0500, Phelan, Tom wrote:

<snip>

>I don't think I'm forcing the problem to fit the solution, just perhaps that
>the problem I'm trying to solve is different than you perceive.  The problem
>that I'm trying to solve is how can an application that is content with a
>maximum bandwidth compete with TCP applications that are insatiably greedy,
>in a way that satisfies some concept of fairness in all directions?
>
>The solution that I see is to provide a buffer against TCP probing.  This
>would involve sending more data than is nominally necessary.  If the network
>has capacity for more than 4M bps, then why shouldn't a media app make use
>of it?  A TCP app would.

<snip>

I think this snippet captures the essence of the argument.

An "insatiably greedy" TCP application has an infinite amount of data to
send. And "fairness" in "TCP-land" is ONLY FAIRNESS IN INSTANTANEOUS
BANDWIDTH ALLOCATION (across TCP/DCCP/SCTP connections/associations).

This is historically how TCP has been viewed. But wait ... if you delay
the "infinite amount of data" in the TCP socket .... what bad happens?
Nothing, it just takes longer (from a "work conservation viewpoint") to send
the data. Did TCP suffer? Well ... it got less work per unit time ... but
nothing really bad occurred for the "insatiable TCP" case.

Most real-time data is DELAY SENSITIVE. To real-time data ... it would be
"fair" to delay some of the TCP data during it's bursts (like some video codecs
have when a high-entropy scene change occurs). That is because TCP
can afford to "wait a little longer" to get it's data across. The bursts 
are usually
bounded in time (a talkspurt, a scene change, etc.) ... and TCP can usually
afford to wait through these bursts. From the real-time application's 
perspective
they are being "good citizens" by not using bandwidth when they don't
need to (during VAD/silence or no video scene changes). Can't the TCP apps
step aside for a moment and acknowledge their good citizenship?

To be fair ... we really need to re-evaluate the issue of "fairness" 
capturing more
than just an INSTANTANEOUS BANDWIDTH ALLOCATION dimension. Is it "fair"
to slow down TCP more than present mechanisms allow? For a "insatiably greedy"
TCP connection the answer may be yes. For a TCP session with a reasonably
bounded amount of data the answer maybe no - because there is a linear
relationship with how long the data will take to get to the other side and
the slowdown rate.

This is a really tough problem.

Why this post? Simply because I'd desire each camp to widen their vision
of what is "fair/right/best" from their own myopic point of view. TCP'ers 
should
try to capture the spirit of Brian's comment ... that "forcing the problem
to fit [TCP's view of] a solution" is not appropriate. And that is, in 
fact, what
I perceive all the well-intentioned "TCP rate control" crowd to be doing here.

Please widen your perspective of the problem. Then comment. Don't expect
people like Brian to be happy with the "you must fit in to TCPs view of the
world" when most of their real-time applications are already being good
citizens (by not sending "all they can" when they don't have to). Do you
like being further punished because you weren't greedy when you could
have been?

Michael Ramalho


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: Re: [AVT] RE: [MMUSIC] DCCP Media Guide
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Hi Brian,

I don't think that the people working with stability of the network are 
sharing you view due to the implications that your claims results in.

First, if a multi-media application transmitting data over a best-effort 
IP network, i.e. without any QoS guarantees across the whole net that 
its traffic will fit, it can definitely cause congestion collapse if the 
mix becomes most, media, little TCP. Please read 
draft-iab-congestion-01.txt for some discussion on this.

Some comments to your assertions.

Rosen, Brian wrote:
> Phillipe
> 
> You have made invalid assumptions:
> 	1. That audio/video media data rates will be changed dynamically
> 		as network traffic changes.  This is very unlikely.  It may
> 		be that you get so much loss that you drop to a lower data
> 		rate, although I suspect that will be rare.  When you do,
> 		you will never revert to the higher rate.  This is because
> 		humans hate to see the change, and minimizing changes is 
> 		more desirable than keeping average quality higher.

First you agree that it is possible to change the rate. Secondly I 
understand that the there are implications on switching rate, i.e. 
quality to often, however there are reasons for switching up when the 
quality gain of a switch if being possible to keep the new rate is clear.

> 
> 	2. Real Time apps have no interest in being "fair".  They are all
> 		or nothing. If you admit them to the network, you expect
> 		the streams to be maintained.  If you can't maintain them,
> 		don't start them.  They don't want to be fair to TCP, and
> they
> 		don't want to be fair to other RTP.  They just want to send
> 		data at the rate they were advised to do by signalling.
> 		They can tolerate some loss.  Misordering=loss.  Duplicates=
> 		wasted bandwith.  That's a whole lot different from TCP
> 		traffic.  
> 

Yes, I agree that the application has no interest in being fair. However 
using best effort networks requires one to being reasonably fair. If one 
are not, the combination of several of media application can totally 
destroy things for them self.

> I can imagine working on protocols which allowed CBR and "best effort"
> traffic to better coexist, but I think it's only possible if the best
> effort traffic recognized the presence of the CBR traffic and reacted
> appropriately.  Otherwise, you need to use QoS mechanisms to separate
> the traffic.

What says that the media traffic is more important that the traffic 
going over TCP?


Cheers

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: [AVT] DCCP Media Guide - STOP crossposting
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Hi,

Please don't crosspost to all these lists. It creates problems for the 
list managers and will fragment the discussion.

As long as we are discussing issues related to media and DCCP please 
keep these topics on the AVT mailing list.

Other DCCP related issues goes on the DCCP list.

Best Regards

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Cc: <avt@ietf.org>, <mankin@psg.com>,
        "'Bert Wijnen \(E-mail\)'" <bwijnen@lucent.com>
Subject: RE: [AVT] RTP MIB Extensions
Date: Fri, 21 Nov 2003 11:34:38 -0500
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Colin

It looks as if there are three action items:-

(i) General RTP MIB update - I'd be happy to contribute to this work but may
not be the best one to lead it.

(ii) RTCP XR MIB - I will trim down the draft submitted to the Minneapolis
meeting and resubmit as an RTCP XR MIB draft within the next two weeks.  Any
offers of help/ input would be very welcome.

(iii) VoIP fault/performance monitoring framework - as Dan mentioned, we did
have some offline discussion on this and agreed to look at this jointly.
I've already submitted a VoIP fault/performance management draft to TIA, we
could start by refining that and then see if/ where in IETF would be the
most appropriate place to introduce it.  The document currently addresses
the requirements for VoIP endpoints and sets out the range of possible
functionality and QoS reporting mechanisms - basically proposing that VoIP
endpoints should provide this type of functionality but not mandating any
particular protocol.  It could certainly be an informational draft to AVT.

>From Dan's comments
> 1. From the point of view of the AVT WG, it looks to me that updating
> the RTP MIB to reflect changes in RTC from RFC 1889 to RFC 3550 has
> higher priority.

With regard to the relative priorities of (i) and (ii), I recall that there
seemed to be no disagreement that the RTP MIB update was important but there
was no suggestion that this was more or less important than progressing an
RTCP XR based draft.  There is considerable interest in using RTCP XR for
VoIP fault/performance management and some definite expressed interest in
seeing a MIB based on the RTCP XR metrics.

Alan Clark







-----Original Message-----
From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of Colin
Perkins
Sent: Friday, November 21, 2003 8:43 AM
To: Romascanu, Dan (Dan)
Cc: avt@ietf.org; mankin@psg.com; Bert Wijnen (E-mail)
Subject: Re: [AVT] RTP MIB Extensions


On Friday, Nov 21, 2003, at 13:19 Europe/London, Romascanu, Dan (Dan)
wrote:
> 1. From the point of view of the AVT WG, it looks to me that updating
> the RTP MIB to reflect changes in RTC from RFC 1889 to RFC 3550 has
> higher priority.

If the MIB is used, then this certainly needs to be done. Maybe anyone
on the list who uses the MIB would like to comment?

> 2. I met with Alan in Minneapolis, and made quite good progress. One
> possible way to formalize the progress we made in the hallway is to
> work together on a framework document for VoIP performance monitoring.
> Hopefully we will have the time and resources to make this happen.

Good to hear, I look forward to seeing the result.

Colin


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: RE: [AVT] HC over multihop / MPLS
Date: Fri, 21 Nov 2003 10:51:10 -0600
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From: "Ash, Gerald R (Jerry), ALABS" <gash@att.com>
To: "Lars-Erik Jonsson (LU/EAB)" <lars-erik.jonsson@ericsson.com>,
        <avt@ietf.org>
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Lars-Erik,

I very much appreciate your taking the time to read the draft and =
comment.

A lot of the questions raised at the AVT meeting are addressed in the =
draft, so it helps when people do read the draft.

Allison, Colin, Magnus, and I met after the meeting and agreed on the =
following approach for next steps:

1. split the current document =
http://ietf.org/internet-drafts/draft-ash-e2e-voip-hdr-comp-rqmts-01.txt =
into 2 requirements documents:
a. 'requirements for hop-by-hop ECRTP over MPLS' (RSVP-TE protocol =
extension oriented)
b. 'requirements for hop-by-hop ECRTP over IP' (SCID switching oriented)
(note that Colin suggests 'hop-by-hop' versus 'multi-hop' terminology; =
'multi-hop was meant to clarify 'end-to-end' in the OO version, but =
still may have led to some confusion).
2. eliminate MPLS jargon such as CE, PE, etc., just talk about routers =
R1, R2, etc.
3. eliminate absolute performance objectives and simply state goal to =
'minimize delta degradation' (for delay, jitter, pkt loss, etc.).  This =
includes performance degradation due to too much resync.
4. eliminate absolute scalability requirement for # of flows, and simply =
state goal to improve scalability of the number of compressed flows that =
can be handled.
5. use already standard protocols for compression, signaling, etc.

It was agreed that requirements 1a should lead to an RSVP-TE protocol =
extension effort and be worked in conjunction with the MPLS SG, and that =
requirements 1b could lead to an Informational RFC out of AVT.

Please let me know of any comments or suggestions on this plan.

I've made a few other brief comments below on your email.=20

Thanks,
Jerry

> The AVT charter states the following as a WG task:=20
> "to review the applicability of Compressed RTP operation on
> MPLS networks, developing extensions as necessary"
>=20
> With the following milestones:
> Nov 03    Initial draft requirements for ECRTP over MPLS;=20
> discuss with MPLS WG
> Nov 03    Initial draft requirements for ECRTP over MPLS;=20
> discuss with MPLS WG
>=20
> However, the current draft refers to "VoIP Header Compression=20
> over Multi-Hop Paths", which is not the same thing. The draft
> clearly shows that compression over multihop paths is one thing,=20
> and compression over MPLS another. Further, the draft takes a=20
> wider scope, and talks about header compression generally,
> not only ECRTP as the charter says.
>=20
> The following observations can be made:
> * If work is to done for header compression over multihop=20
> and/or MPLS, it might make sense to not only address ECRTP.

Agreed.  The plan to split the requirements draft into 2 separate =
requirements drafts (ECRTP over MPLS, ECRTP over IP) addresses this, as =
outlined above.

> * The only work in this that naturally could belong in the=20
> AVT WG would be to make sure ECRTP would work in these=20
> scenarios.  In my understanding, this should already be=20
> the case with ECRTP (as well as some ROHC profiles), which=20
> is also indicated by the multihop req draft.

Agreed.

> * Header compression over multihop paths [ described by the=20
> multihop req draft as where one wants to avoid=20
> decompression/compression steps at intermediate routers if=20
> the same compression scheme is supported at both incoming=20
> and outgoing interfaces ] is fully possible also without any=20
> new standards, as it would only be an implementation issue for=20
> each intermediate router to bind an incoming compressed stream
> to its corresponding outgoing compressed stream. In a path=20
> A->B->C, where B is the intermediate router
> negotiating use of e.g. CRTP with both A and C, it is fully=20
> up to the implementation of B to avoid unnecessary=20
> decompression/compression and just forward compressed packets
> as-is ("CID routing").

Agreed (requirements draft 1b, as outlined above, leading to an =
Informational RFC).

> * The only significant standardization work I can see is to=20
> define how to do header compression (e.g. IPHC, CRTP/ECRTP,=20
> ROHC) over MPLS, which is  probably a work that would require=20
> some coordination with AVT and ROHC individuals, but the core
> of that work does not naturally belong to ROHC or AVT. Would
> not the MPLS WG be a more proper place?

I think the plan is to complete the requirements 1a and the follow-on =
protocol work in conjunction with the MPLS WG.


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: RE: [AVT] RTP MIB Extensions
Date: Fri, 21 Nov 2003 19:45:24 +0200
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Thread-Topic: [AVT] RTP MIB Extensions
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From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
To: <alan@telchemy.com>, "Colin Perkins" <csp@csperkins.org>
Cc: <avt@ietf.org>, <mankin@psg.com>,
        "Bert Wijnen (E-mail)" <bwijnen@lucent.com>
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>=20
> With regard to the relative priorities of (i) and (ii), I=20
> recall that there
> seemed to be no disagreement that the RTP MIB update was=20
> important but there
> was no suggestion that this was more or less important than=20
> progressing an
> RTCP XR based draft.  There is considerable interest in using=20
> RTCP XR for
> VoIP fault/performance management and some definite expressed=20
> interest in
> seeing a MIB based on the RTCP XR metrics.
>=20
> Alan Clark
>=20
>=20
>=20

Alan,

RTCP XR is an extension of RTP/RTCP. Subsequently, the RTCP XR MIB is an =
extension of the RTP MIB. At this point in the evolution of RTP, as RFC =
3550 obsoletes RFC 1889, it looks imperative that the necessary changes =
are being made to the RTP MIB first, so that any extension work (like =
the proposed work on RTCP XR MIB) is based on an updated version of the =
MIB and standard.=20

Regards,

Dan

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Alan Clark" <alan@telchemy.com>
To: "'Romascanu, Dan \(Dan\)'" <dromasca@avaya.com>,
        "'Colin Perkins'" <csp@csperkins.org>
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Subject: RE: [AVT] RTP MIB Extensions
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[Dan R] RTCP XR is an extension of RTP/RTCP. Subsequently, the RTCP XR MIB
is an extension of the RTP MIB. At this point in the evolution of RTP, as
RFC 3550 obsoletes RFC 1889, it looks imperative that the necessary changes
are being made to the RTP MIB first, so that any extension work (like the
proposed work on RTCP XR MIB) is based on an updated version of the MIB and
standard. 

Dan

I have to disagree with you.  There is minimal overlap between the two other
than a need to put the RTCP XR MIB/Group under the RTP MIB OID and hence
they can be progressed in parallel. 

At this moment there are no proposed specific changes to the RTP MIB other
than the draft discussed in Minneapolis and hence making these serial
activities would delay the progress of the RTCP XR MIB work for an
indefinite period, which I'm sure you would not want.  There is strong
interest in the use of RTCP XR for VoIP fault/ performance management - I am
prepared to do the work needed to progress the draft and would welcome your
input and comments.

Regards

Alan


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------=_NextPart_000_0065_01C3B030.7E34BC90--


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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To: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
CC: alan@telchemy.com, Colin Perkins <csp@csperkins.org>, avt@ietf.org,
        mankin@psg.com, "Bert Wijnen (E-mail)" <bwijnen@lucent.com>
Subject: Re: [AVT] RTP MIB Extensions
References: <AAB4B3D3CF0F454F98272CBE187FDE2F04259752@is0004avexu1.global.avaya.com>
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I see no reason why the two can't be developed in parallel.  Yes, the 
RTCP XR MIB will be an extension, but I would think the updates to the 
RTP MIB would stabilize pretty quickly.

Romascanu, Dan (Dan) wrote:

>>With regard to the relative priorities of (i) and (ii), I 
>>recall that there
>>seemed to be no disagreement that the RTP MIB update was 
>>important but there
>>was no suggestion that this was more or less important than 
>>progressing an
>>RTCP XR based draft.  There is considerable interest in using 
>>RTCP XR for
>>VoIP fault/performance management and some definite expressed 
>>interest in
>>seeing a MIB based on the RTCP XR metrics.
>>
>>Alan Clark
>>
>>
>>
> 
> 
> Alan,
> 
> RTCP XR is an extension of RTP/RTCP. Subsequently, the RTCP XR MIB is an extension of the RTP MIB. At this point in the evolution of RTP, as RFC 3550 obsoletes RFC 1889, it looks imperative that the necessary changes are being made to the RTP MIB first, so that any extension work (like the proposed work on RTCP XR MIB) is based on an updated version of the MIB and standard. 
> 
> Regards,
> 
> Dan
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
> 


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: RE: [AVT] RTP MIB Extensions
Date: Fri, 21 Nov 2003 20:21:04 +0200
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From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
To: <alan@telchemy.com>, "Colin Perkins" <csp@csperkins.org>
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> -----Original Message-----
> From: Alan Clark [mailto:alan@telchemy.com]
> Sent: 21 November, 2003 8:08 PM
> To: Romascanu, Dan (Dan); 'Colin Perkins'
> Cc: avt@ietf.org; mankin@psg.com; 'Bert Wijnen (E-mail)'
> Subject: RE: [AVT] RTP MIB Extensions
>=20
>=20
>=20
> [Dan R] RTCP XR is an extension of RTP/RTCP. Subsequently,=20
> the RTCP XR MIB is an extension of the RTP MIB. At this point=20
> in the evolution of RTP, as RFC 3550 obsoletes RFC 1889, it=20
> looks imperative that the necessary changes are being made to=20
> the RTP MIB first, so that any extension work (like the=20
> proposed work on RTCP XR MIB) is based on an updated version=20
> of the MIB and standard.=20
>=20
> Dan
>=20
> I have to disagree with you.  There is minimal overlap=20
> between the two other than a need to put the RTCP XR=20
> MIB/Group under the RTP MIB OID and hence they can be=20
> progressed in parallel.=20
>=20
> At this moment there are no proposed specific changes to the=20
> RTP MIB other than the draft discussed in Minneapolis and=20
> hence making these serial activities would delay the progress=20
> of the RTCP XR MIB work for an indefinite period, which I'm=20
> sure you would not want.  There is strong interest in the use=20
> of RTCP XR for VoIP fault/ performance management - I am=20
> prepared to do the work needed to progress the draft and=20
> would welcome your input and comments.
>=20
> Regards
>=20
> Alan
>=20

Alan,

We need to agree to disagree:-)=20

As a user of the technology, and as a MIB advisor I think that the =
proper way is to have the core RTP MIB updated first. I understand your =
wish to make progress on your specific work on RTCP XR, but the RTP MIB =
is the main item for this WG.  Everybody implementing RTCP-XR need to =
implement RTP as well, and better do its management based on an updated =
MIB. I hope that the RTP MIB authors will do the necessary changes soon. =
Maybe you can do them. The work needs not be serialized, but I think =
that an RTCP XR MIB should not be approved on the standards track before =
the RTP MIB is updated for eventual changes caused by RFC 1889 being =
obsolete by RFC 3550.

Regards,

Dan


=20
>=20

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: RE: [AVT] RTP MIB Extensions
Date: Fri, 21 Nov 2003 20:23:13 +0200
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From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
To: "Tom Taylor" <taylor@nortelnetworks.com>
Cc: <alan@telchemy.com>, "Colin Perkins" <csp@csperkins.org>, <avt@ietf.org>,
        <mankin@psg.com>, "Bert Wijnen (E-mail)" <bwijnen@lucent.com>
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Agreed. There is no for serialization. We just need to make sure that by =
the time extensions MIB modules are to enter the standards track, an =
updated RTP MIB is already available. Dan


> -----Original Message-----
> From: Tom Taylor [mailto:taylor@nortelnetworks.com]
> Sent: 21 November, 2003 8:15 PM
> To: Romascanu, Dan (Dan)
> Cc: alan@telchemy.com; Colin Perkins; avt@ietf.org;=20
> mankin@psg.com; Bert Wijnen (E-mail)
> Subject: Re: [AVT] RTP MIB Extensions
>=20
>=20
> I see no reason why the two can't be developed in parallel.  Yes, the=20
> RTCP XR MIB will be an extension, but I would think the=20
> updates to the=20
> RTP MIB would stabilize pretty quickly.
>=20
> Romascanu, Dan (Dan) wrote:
>=20
> >>With regard to the relative priorities of (i) and (ii), I=20
> >>recall that there
> >>seemed to be no disagreement that the RTP MIB update was=20
> >>important but there
> >>was no suggestion that this was more or less important than=20
> >>progressing an
> >>RTCP XR based draft.  There is considerable interest in using=20
> >>RTCP XR for
> >>VoIP fault/performance management and some definite expressed=20
> >>interest in
> >>seeing a MIB based on the RTCP XR metrics.
> >>
> >>Alan Clark
> >>
> >>
> >>
> >=20
> >=20
> > Alan,
> >=20
> > RTCP XR is an extension of RTP/RTCP. Subsequently, the RTCP=20
> XR MIB is an extension of the RTP MIB. At this point in the=20
> evolution of RTP, as RFC 3550 obsoletes RFC 1889, it looks=20
> imperative that the necessary changes are being made to the=20
> RTP MIB first, so that any extension work (like the proposed=20
> work on RTCP XR MIB) is based on an updated version of the=20
> MIB and standard.=20
> >=20
> > Regards,
> >=20
> > Dan
> >=20
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www1.ietf.org/mailman/listinfo/avt
> >=20
>=20
>=20

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
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From exim@www1.ietf.org  Fri Nov 21 19:10:23 2003
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Date: Fri, 21 Nov 2003 16:09:06 -0800
To: Colin Perkins <csp@csperkins.org>
From: Mark Baugher <mbaugher@cisco.com>
Subject: Re: [AVT] RTP MIB Extensions
Cc: "Romascanu, Dan (Dan)" <dromasca@avaya.com>, <avt@ietf.org>,
        <mankin@psg.com>, "Bert Wijnen (E-mail)" <bwijnen@lucent.com>
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At 05:43 AM 11/21/2003, Colin Perkins wrote:
>On Friday, Nov 21, 2003, at 13:19 Europe/London, Romascanu, Dan (Dan) wrote:
>>1. From the point of view of the AVT WG, it looks to me that updating the 
>>RTP MIB to reflect changes in RTC from RFC 1889 to RFC 3550 has higher 
>>priority.
>
>If the MIB is used, then this certainly needs to be done. Maybe anyone on 
>the list who uses the MIB would like to comment?

The last I heard is that it was deployed in the H.323 MIB and that was the 
end of it.  All the folks I worked on it with seem to be no longer involved 
with H.323 products.  I don't know of any companies shipping the RTP MIB.

That being said, I don't understand why anyone would go to the trouble of 
implementing a piece of the RTP MIB.  Perhaps it would make sense to do 
only the multicast objects or only the unicast objects.  I can't fathom why 
anyone would go to the trouble of developing only a piece of the MIB 
without implementing the session objects, etc., i.e. for little incremental 
effort, the entire RTP system could be monitored and/or configured.

Mark


>>2. I met with Alan in Minneapolis, and made quite good progress. One 
>>possible way to formalize the progress we made in the hallway is to work 
>>together on a framework document for VoIP performance monitoring. 
>>Hopefully we will have the time and resources to make this happen.
>
>Good to hear, I look forward to seeing the result.
>
>Colin
>
>
>_______________________________________________
>Audio/Video Transport Working Group
>avt@ietf.org
>https://www1.ietf.org/mailman/listinfo/avt



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Fri Nov 21 19:52:24 2003
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From: "Raymond Chen" <rchen@broadcom.com>
To: avt@ietf.org
Date: Fri, 21 Nov 2003 16:50:59 -0800
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Subject: [AVT] FW: I-D ACTION:draft-chen-rtp-bv-02.txt
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Dear AVT members,

Your feedback on the following Internet Draft draft-chen-rtp-bv-02.txt
would be appreciated.  I believe this version has addressed all
outstanding issues the AVT chairs had on this Internet Draft.

Your support in accepting this draft as an official AVT work item would
also be greatly appreciated.

Thanks.

Raymond

-----Original Message-----
From: owner-ietf-announce@ietf.org [mailto:owner-ietf-announce@ietf.org]
On Behalf Of Internet-Drafts@ietf.org
Sent: Friday, November 21, 2003 1:04 PM
To: IETF-Announce:
Subject: I-D ACTION:draft-chen-rtp-bv-02.txt

A New Internet-Draft is available from the on-line Internet-Drafts
directories.


	Title		: RTP Payload Format for BroadVoice Speech
Codecs
	Author(s)	: J. Chen
	Filename	: draft-chen-rtp-bv-02.txt
	Pages		: 13
	Date		: 2003-11-21
	
This document describes the RTP payload format for the 
BroadVoice(TM) narrowband and wideband speech codecs developed by 
Broadcom Corporation.  The document also provides specifications for 
the use of BroadVoice with MIME and SDP.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-chen-rtp-bv-02.txt

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_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Sat Nov 22 03:29:40 2003
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From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: "avt IETF" <avt@ietf.org>, "AG 9 ITS" <ag9@its.se>,
        "Wisdom-Cds" <wisdom-cds@bristol.ac.uk>,
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Subject: [avt] Refined specification for text over IP     FW: I-D ACTION:draft-ietf-avt-rfc2793bis-00.txt
Date: Sat, 22 Nov 2003 09:28:23 +0100
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This is a multi-part message in MIME format.

------=_NextPart_000_0004_01C3B0DA.F9B71430
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	charset="iso-8859-1"
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A new IETF draft is available related to real time interactive text
conversation.

An RTP transport for real time character by character transmission of tex=
t
was defined and approved a couple of years ago in IETF RFC2793.
It is now successfully used in conversational applications, usually combi=
ned
with voice and/or video to enable the participants in the session to
communicate in any preferred medium. One important user group is deaf and
hard-of-hearing people, but the feature is of course of value for all.

In the past, most multimedia implementors have been eager to create
interoperability in video and audio, while the text part has been left
behind and implemented as manufacturer specific message-oriented
"chat"-additions. It is time for a change, to raise all the three media o=
f
video, text and audio to the same status of interoperability and deployme=
nt.
That will give multimedia based conversational services a real benefit.

The text transport according to the new draft "RFC2793 bis" is compatible
with the current RFC2793.
It clarifies use of SDP for setting up sessions including text, and
introduces a new format
specifically intended for text transit between gateways that can be merge=
d
into the audio stream.

We(the authors) hope for rapid handling of this draft, so that it can be
used in further work in establishing
interoperable accessible multimedia implementations with all media needed=
 in
human conversation, and in the ongoing work with text-aware gateways,
text-over-ip and transcoding services in various groups.

Please direct comments to the IETF avt list.
Regards

Gunnar Hellstr=F6m

-----Original Message-----
From: owner-ietf-announce@ietf.org
[mailto:owner-ietf-announce@ietf.org]On Behalf Of
Internet-Drafts@ietf.org
Sent: Friday, November 21, 2003 10:04 PM
To: IETF-Announce:
Subject: I-D ACTION:draft-ietf-avt-rfc2793bis-00.txt


A New Internet-Draft is available from the on-line Internet-Drafts
directories.


	Title		: RTP Payload for Text Conversation
	Author(s)	: G. Hellstrom
	Filename	: draft-ietf-avt-rfc2793bis-00.txt
	Pages		: 19
	Date		: 2003-11-21

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

Text conversation is used alone or in connection to other
conversational facilities such as video and voice, to form multimedia
conversation services.

This RTP payload description contains an optional possibility to
include redundant text from already transmitted packets in order to
reduce the risk of text loss caused by packet loss. The redundancy
coding follows RFC 2198.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avt-rfc2793bis-00.txt

To remove yourself from the IETF Announcement list, send a message to
ietf-announce-request with the word unsubscribe in the body of the messag=
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Internet-Drafts are also available by anonymous FTP. Login with the usern=
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_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Thread-Topic: [AVT] RE: [MMUSIC] DCCP Media Guide
Thread-Index: AcOvk9QY7fErcwuYRI2/oFuzWzxiXQAdmRLw
From: =?iso-8859-1?Q?BABONNEAU_G=E9rard_FTRD/DMI/REN?= <gerard.babonneau@francetelecom.com>
To: "Rosen, Brian" <Brian.Rosen@marconi.com>,
        "Phelan, Tom" <tphelan@sonusnet.com>
Cc: <dccp@ietf.org>, <avt@ietf.org>
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I had a carefull look at the draft.
In my opinion, coding with two or more rates is only TCP compliant, and =
it does'nt improve the video quality because the network losses have =
disastrous effects (30% pictures loss for 3% packet loss). We (France =
Telecom R&D) have developped a test bed with MPEG4/RTP/UDP hierarchical =
codec with 2 or more layers for the video. A DSCP marking allows =
diffserv routers to choose packets to be lost when a congestion occurs. =
We have a better quality than other codecs in a lossy environment and it =
also works in multicast sessions. Diffserv routers can also share the =
bandwidth between UDP and TCP flows, to guarantee a good throughput for =
all the flows on the network.=20

G=E9rard BABONNEAU

-----Message d'origine-----
De : Rosen, Brian [mailto:Brian.Rosen@marconi.com]=20
Envoy=E9 : jeudi 20 novembre 2003 19:24
=C0 : 'Phelan, Tom'
Cc : 'dccp@ietf.org'; 'avt@ietf.org'
Objet : RE: [AVT] RE: [MMUSIC] DCCP Media Guide


I have looked carefully at the drafts.  I think I understand what=20
the proposal is, and how it works.  I'll even agree that it might=20
do what you think it will do, but I suspect that my device will=20
cause a heck of a lot more congestion to the network if I run the=20
video streams on DCCP than if I leave it on UDP.  I think the=20
solution will be much worse than the problem.

The solution is just plain broken, in my opinion.  The real, hard,=20
problem is that there probably is no good solution, short of fixing=20
TCP, or treating real time data differently (ie QoS) in the router. =20
Like it or not, RTP is not appropriate for what we usually call=20
"best effort", but the problem with that statement is that it's not=20
really true that we need something more than for the network to use=20
its' best effort to deliver the traffic.  What is true is that other=20
streams have to deal better with CBR data than they do.  It is TCP=20
that is broken. I don't think there is any way around that. =20
TCP makes an assumption that is invalid when real time media is part=20
of the traffic mix.

There are so many problems with DCCP applied to real time media (this=20
one, slow start,...), that I think we should just stop believing that=20
DCCP is appropriate for RTP.

A specific comment to your message below is that if we send more data =
than 4Mbit, it is just filler.  We cannot "make use of it".  We can just =
congest the network more than we do.  As you might realize, we already =
have problems putting 4M into the network.  More is not better in this =
case.

Brian

> -----Original Message-----
> From: Phelan, Tom [mailto:tphelan@sonusnet.com]
> Sent: Thursday, November 20, 2003 11:48 AM
> To: 'Rosen, Brian'
> Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
>=20
>=20
> Hi Brian,
>=20
> Thanks for the comments.  See inline ...
>=20
> Tom P.
>=20
> > -----Original Message-----
> > From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> > Sent: Thursday, November 20, 2003 11:05 AM
> > To: 'Phelan, Tom'; 'Jonathan Rosenberg'
> > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> >=20
> >=20
> > To me, this is just nonsense.
> > I can't imagine anyone deploying such a solution.
> >=20
> > This is forcing the problem to fit the solution, and it's not the
> > right way to go.  If you want it to work, the protocol has to take=20
> > the characteristics of the flows as they are, and not as you would=20
> > like them to be.  Real time media, streaming
> or realtime,
> > is either constant bit rate, or pretty close to it.  That's how it=20
> > works.  Our challenge is to make the protocols work with this data,=20
> > not change the data, especially silly ideas like doubling the=20
> > bandwidth.
> >=20
> > We make video telephones.  They nominally send 4 megabits.  Would
> > you like us to double that, under any circumstances?
> >=20
> > I thought not.
>=20
> Would you prefer to be forced unnecessarily into packet loss and a=20
> lower encoding rate by greedy TCP applications?  Or do you have
> another solution?
>=20
> I don't think I'm forcing the problem to fit the solution, just=20
> perhaps that the problem I'm trying to solve is different than you
> perceive.  The problem
> that I'm trying to solve is how can an application that is=20
> content with a
> maximum bandwidth compete with TCP applications that are=20
> insatiably greedy,
> in a way that satisfies some concept of fairness in all directions?
>=20
> The solution that I see is to provide a buffer against TCP probing. =20
> This would involve sending more data than is nominally necessary.
> If the network
> has capacity for more than 4M bps, then why shouldn't a media=20
> app make use
> of it?  A TCP app would.
>=20
> I must admit that my initial reactions to suggestions like this were=20
> very similar to yours.  I encourage you to look at the draft for
> more detail on
> the issues that convinced me otherwise.
>=20
> >=20
> > Brian
> >=20
> > > -----Original Message-----
> > > From: Phelan, Tom [mailto:tphelan@sonusnet.com]
> > > Sent: Thursday, November 20, 2003 10:36 AM
> > > To: 'Jonathan Rosenberg'
> > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > Subject: [AVT] RE: [MMUSIC] DCCP Media Guide
> > >=20
> > >=20
> > > Hi Jonathan,
> > >=20
> > > Thanks for the comments.  The actual proposal is to send up to=20
> > > double the nominal transmit rate when there's available bandwidth, =

> > > and to reduce the
> > > rate as necessary if there isn't enough bandwidth.  This=20
> > > doubling is to
> > > allow fair sharing of the chokepoint bandwidth (between DCCP=20
> > > media apps and
> > > TCP apps), and to avoid what appears to me to be the=20
> > > potential for bullying
> > > from TCP applications that want to suck up all the bandwidth=20
> > > they can.  It
> > > doesn't preclude using low-rate codecs, or mean that the=20
> > > access link must
> > > support the double rate for the connection to function well.
> > >=20
> > > This recommendation was meant for best-effort environments, where=20
> > > the voice traffic receives no special treatment, and must compete=20
> > > with TCP apps for
> > > network resources.  I can see how this would require a=20
> > > network that gave
> > > special treatment to voice packets to provision double the=20
> > > capacity, though.
> > >=20
> > > This fairness problem is difficult.  All of the research I'm=20
> > > familiar with approaches it from the direction of other apps being =

> > > fair to TCP.  I'm not
> > > aware of any other methods for ensuring that non-TCP apps get=20
> > > their fair
> > > share when competing with TCP.  If you (or anyone else) knows=20
> > > of something
> > > I'd appreciate the pointer.
> > >=20
> > > Tom P.
> > >=20
> > > > -----Original Message-----
> > > > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > > > Sent: Thursday, November 20, 2003 2:32 AM
> > > > To: Phelan, Tom
> > > > Cc: 'dccp@ietf.org'; 'avt@ietf.org'; 'mmusic@ietf.org'
> > > > Subject: Re: [MMUSIC] DCCP Media Guide
> > > >=20
> > > >=20
> > > > Tom,
> > > >=20
> > > > I don't think this solution will be acceptable to many of
> > > the network
> > > > providers of interactive voip services. Many of them are
> > > insistent on
> > > > using low rate codecs, like g.723, in order to reduce the
> > costs of
> > > > capacity in their networks. Doubling the rate would defeat the=20
> > > > purpose of using these low rate codecs in the first place.
> > > >=20
> > > > -Jonathan R.
> > > >=20
> > > > Phelan, Tom wrote:
> > > >=20
> > > >=20
> > > > > Comments will be appreciated.  In particular, I make the
> > > > suggestion that
> > > > > media apps should transmit at up to twice their nominal
> > > > maximum rate, when
> > > > > DCCP allows it, in order to provide protection from
> > competing TCP
> > > > > applications.  I know that the first time I heard someone
> > > > make a similar
> > > > > suggestion I was horrified, but as I developed the issues
> > > > it seemed to be
> > > > > necessary.  Reactions to this will be appreciated.
> > > > >=20
> > > > > Tom P.
> > > > >=20
> > > > > _______________________________________________
> > > > > mmusic mailing list
> > > > > mmusic@ietf.org https://www1.ietf.org/mailman/listinfo/mmusic
> > > > >=20
> > > >=20
> > > > --=20
> > > > Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> > > > Chief Technology Officer                    Parsippany, NJ=20
> > > 07054-2711
> > > > dynamicsoft
> > > > jdrosen@dynamicsoft.com                     FAX:  =20
> (973) 952-5050
> > > > http://www.jdrosen.net                      PHONE:=20
> (973) 952-5000
> > > > http://www.dynamicsoft.com
> > > >=20
> > >=20
> > > _______________________________________________
> > > Audio/Video Transport Working Group
> > > avt@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/avt
> > >=20
> >=20
>=20
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
>=20

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Sun Nov 23 09:10:30 2003
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From: "Vinay Sathe" <vsathe@san.rr.com>
To: "=?iso-8859-1?Q?'BABONNEAU_G=E9rard_FTRD/DMI/REN'?=" <gerard.babonneau@francetelecom.com>
Cc: <avt@ietf.org>
Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
Date: Sun, 23 Nov 2003 06:11:50 -0800
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Hello Gerard
Your results sound very interesting. Is there a reference/ web site you =
can
quote to read more about your work?=20
Thank you,
Vinay

-----Original Message-----
From: avt-admin@ietf.org [mailto:avt-admin@ietf.org] On Behalf Of =
BABONNEAU
G=E9rard FTRD/DMI/REN
Sent: Friday, November 21, 2003 7:54 AM
To: Rosen, Brian; Phelan, Tom
Cc: dccp@ietf.org; avt@ietf.org
Subject: RE : [AVT] RE: [MMUSIC] DCCP Media Guide

I had a carefull look at the draft.
In my opinion, coding with two or more rates is only TCP compliant, and =
it
does'nt improve the video quality because the network losses have =
disastrous
effects (30% pictures loss for 3% packet loss). We (France Telecom R&D) =
have
developped a test bed with MPEG4/RTP/UDP hierarchical codec with 2 or =
more
layers for the video. A DSCP marking allows diffserv routers to choose
packets to be lost when a congestion occurs. We have a better quality =
than
other codecs in a lossy environment and it also works in multicast =
sessions.
Diffserv routers can also share the bandwidth between UDP and TCP flows, =
to
guarantee a good throughput for all the flows on the network.=20

G=E9rard BABONNEAU





_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: Re: [AVT] RE: [MMUSIC] DCCP Media Guide
To: Magnus Westerlund <magnus.westerlund@ericsson.com>,
        "Rosen, Brian" 
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On Fri, 21 Nov 2003 16:40:06 +0100
 Magnus Westerlund <magnus.westerlund@ericsson.com> wrote:
> Hi Brian,
> 
> I don't think that the people working with stability of the network are 
> sharing you view due to the implications that your claims results in.
> 
> First, if a multi-media application transmitting data over a best-effort 
> IP network, i.e. without any QoS guarantees across the whole net that 
> its traffic will fit, it can definitely cause congestion collapse if the 
> mix becomes most, media, little TCP. Please read 
> draft-iab-congestion-01.txt for some discussion on this.

Dear Magnus;

Yes, I have read this. I also operate networks that are mostly UDP - typically 90%+ of the traffic.
I have never seen true congestive collapse on these networks. Never. 

The typical paper that I have seen that deals with post 1987 congestion collapse
hypotheses a UDP flow that is too large to fit in a end to end "pipe" that is subject to several
bottlenecks, with the second being smaller than the first. The UDP packets clog 
up the first bottleneck,
and are then dropped on the second. Since the packets dropped on the second bottleneck are wasted,
this lowers the goodput of the first bottleneck, in principle to as near zero as you want.

Note that this is very different from the 1987 congestive collapse where congestion causes
the number of retrials and control packets to increase, until they dominate the available 
bandwidth - and this "auto-congestion" causes the link goodput to vanish. This UDP congestion
collapse is _not_ a run-away situation like the original case.

From RFC896 :
-------
Even with the
better adaptive host retransmission algorithms, though, a  sudden
load on the net can cause the round-trip time to rise faster than
the sending hosts measurements of round-trip time can be updated.
Such  a  load  occurs  when  a  new  bulk  transfer,  such a file
transfer, begins and starts filling a large window.   Should  the
round-trip  time  exceed  the maximum retransmission interval for
any host, that host will begin to introduce more and more  copies
of  the same datagrams into the net.  The network is now in seri-
ous trouble.  Eventually all available buffers in  the  switching
nodes  will  be full and packets must be dropped.  The round-trip
time for packets that are delivered is now at its maximum.  Hosts
are  sending  each packet several times, and eventually some copy
of each packet arrives at its destination.   This  is  congestion
collapse.

This condition is stable.  Once the  saturation  point  has  been
reached,  if the algorithm for selecting packets to be dropped is
fair, the network will continue to operate in a  degraded  condi-
tion.   In  this  condition  every  packet  is  being transmitted
several times and throughput is reduced to a  small  fraction  of
normal.   We  have pushed our network into this condition experi-
mentally and observed its stability.  It is possible  for  round-
trip  time to become so large that connections are broken because
the hosts involved time out.
-------

This run-away collapse was what TCP Reno was introduced to fix. 

I have never seen a claim that excess UDP could cause this type of behavior in
the TCP + UDP system using modern TCP algorithms, and frankly regard
use of the term  congestion collapse in the present context as being misleading.

Again, I have never seen anything like this on a network I operate or monitor.
There can, of course, be "congestion events", where the TCP + UDP is overfilling the
available bandwidth. These stop immediately when the offending flows do.

Why is this not a problem in practice ? 

1.) For sources :
 
Because the throughput / goodput definition is arbitrary - it is a global metric, when
a local one would be more appropriate.

Suppose I am running a source network. I can easily arrange for the sum of all of my sources
to be < my link bandwidth. So, my network is always happy, at least outbound.

Let's say that I am sourcing high bit rate video. Let's say that for some reason every viewer
is trying to get these video's through a dial up modem. Now, they will not be happy, and the
vast majority of the outbound packets will be dropped at the user side.  

By definitions such as 

http://www.ietf.org/internet-drafts/draft-iab-congestion-01.txt
Section 3.1

my network is experiencing congestion collapse (most packets going out my links are wasted), even
though locally I can see no signs of it. (My clients may not be happy, and I may want to give
preference to non-dial-up users, but that is a different matter and I would argue out of scope.)

2.) For Sinks :

Because UDP flows (except for DOS attacks) are self regulating. If I try and put 10 Mbps of UDP
through a 30 kbps link, I will stop quickly.

I in fact see this all of the time - and I bet many people on this list have tried to 
watch a video through a connection at home or over a T1 only to realize that the 
packet losses are just too high and stopped the
process. This is clearly not a run-away situation and I would again argue that the use of the term
congestion collapse to describe it is inappropriate.

I of course, fully support the recommendations of draft-iab-congestion-01.txt
and elsewhere to drop UDP sessions or layers in the case of excessive loss, which tends to be set
very high (30% packet loss in the case
of draft-iab-congestion-01.txt, which is long past the point where most media players
will time out and users will leave in my experience).

Since we are not talking about a run-away collapse, then any problems that are not self
regulating can be dealt with administratively - which is common practice, especially
on college campus networks.

So we have

- Something that is not a run-away situation, despite the re-use of terms that may give that
impression and

- Something that exists on real networks and is dealt with routinely already without causing undue
problems.


> 
> Some comments to your assertions.
> 
> Rosen, Brian wrote:
> > Phillipe
> > 
> > You have made invalid assumptions:
> > 	1. That audio/video media data rates will be changed dynamically
> > 		as network traffic changes.  This is very unlikely.  It may
> > 		be that you get so much loss that you drop to a lower data
> > 		rate, although I suspect that will be rare.  When you do,
> > 		you will never revert to the higher rate.  This is because
> > 		humans hate to see the change, and minimizing changes is 
> > 		more desirable than keeping average quality higher.
> 
> First you agree that it is possible to change the rate. Secondly I 

May be possible. There can be real costs in doing this, and it may not
be practical in many cases.

> understand that the there are implications on switching rate, i.e. 
> quality to often, however there are reasons for switching up when the 
> quality gain of a switch if being possible to keep the new rate is clear.
> 
> > 
> > 	2. Real Time apps have no interest in being "fair".  They are all
> > 		or nothing. If you admit them to the network, you expect
> > 		the streams to be maintained.  If you can't maintain them,
> > 		don't start them.  They don't want to be fair to TCP, and
> > they
> > 		don't want to be fair to other RTP.  They just want to send
> > 		data at the rate they were advised to do by signalling.
> > 		They can tolerate some loss.  Misordering=loss.  Duplicates=
> > 		wasted bandwith.  That's a whole lot different from TCP
> > 		traffic.  
> > 
> 
> Yes, I agree that the application has no interest in being fair. However 
> using best effort networks requires one to being reasonably fair. If one 
> are not, the combination of several of media application can totally 
> destroy things for them self.
> 

This is commonly said, but I would argue is routinely shown to be not true in practice.


> > I can imagine working on protocols which allowed CBR and "best effort"
> > traffic to better coexist, but I think it's only possible if the best
> > effort traffic recognized the presence of the CBR traffic and reacted
> > appropriately.  Otherwise, you need to use QoS mechanisms to separate
> > the traffic.
> 
> What says that the media traffic is more important that the traffic 
> going over TCP?
> 

The network administrator decides this yeah or nay. Since there is no danger of
a run-away collapse, this is the appropriate level for this decision.

Even for TCP flows, there are all sorts of ways that network admins can and do show
preference, so fairness is something of an illusion there as well. 

It may well be that there are networks (say, mobile IP), where excessive use of UDP could
cause a true congestive collapse (i.e., a run-away of useless traffic). If so, then I would
argue that should be demonstrated clearly, and then be addressed for those networks
specifically. In the absence of this I would be very careful to impose apparently 
innocent conditions which, if actually followed, would kill most real time media distribution over
the Internet.

As I hate to complain without offering alternatives, here is my definition of UDP fairness and how
to achieve it :

UDP flows should not significantly increase the packet loss rate on any link that they transverse.
A UDP flow or flows from an application that 
do significantly increase the packet loss rate on a link MUST either be
dropped or reduced in bandwidth  on that link until the excess packet loss rate on the link is
reduced to an acceptable level.

Of course, averaging times and acceptable loss changes have to be defined - I would suggest
5 RTT's and 1%.

> 
> Cheers
> 
> Magnus Westerlund

Regards
Marshall Eubanks

> 
> Multimedia Technologies, Ericsson Research EAB/TVA/A
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> 
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Mon, 24 Nov 2003 00:36:58 +0000
From: Colin Perkins <csp@csperkins.org>
To: avt@ietf.org
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Subject: [AVT] Last call for review: draft-jones-avt-audio-t38-02.txt
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This is to announce a last call for comments on the audio/t38 MIME type
registration.  Please send comments to the mailing list by Friday 28th
November 2003. If no substantive comments are received by this time we
shall recommend the IESG publish this draft as a Proposed standard RFC,
conditional on the publication of the amended version of T.38 by the ITU.

Colin



------- Forwarded message:
>Date: Wed, 19 Nov 2003 15:25:59 -0500
>From: Internet-Drafts@ietf.org
>To: IETF-Announce:
>Cc: avt@ietf.org
>Subject: [AVT] I-D ACTION:draft-jones-avt-audio-t38-02.txt
>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> 
> 
> 	Title		: Real-Time Facsimile (T.38) - audio/t38 MIME 
> 			  Sub-type Registration
> 	Author(s)	: P. Jones
> 	Filename	: draft-jones-avt-audio-t38-02.txt
> 	Pages		: 6
> 	Date		: 2003-11-19
> 	
> This document defines the MIME sub-type audio/t38.  The packetization
> and usage of this MIME type, which is intended for use within SDP, is
> specified within ITU-T Recommendation T.38.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-jones-avt-audio-t38-02.txt
...
------- End of Forwarded Message

-- 
Colin Perkins
csp@csperkins.org

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Colin Perkins <csp@csperkins.org>
To: "Paul E. Jones" <paulej@packetizer.com>
Cc: avt@ietf.org, magnus.westerlund@ericsson.com
Subject: Re: [AVT] Re: I-D ACTION:draft-jones-avt-text-red-00.txt
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Paul, all,

Looks good -- I noticed two minor nits: reference [2] is not actually
referenced anywhere in the draft, and there is a Microsoft smart quote 
in Reference 1.

I suggest we take this as an AVT work item, and progress it quickly. 
If any group member objects to this becoming a work item, please send
your comment to this list. If there are no objections, I suggest this 
is submitted as draft-ietf-avt-text-red-00.txt at the end of the week,
with my comments above corrected.

Colin




--> "Paul E. Jones" <paulej@packetizer.com> writes:
> Folks,
> 
> I just wanted to bring this particular document to your attention. 
> During the review of RFC 2793bis, it was proposed by several folks that
> we extract the text/red MIME sub-type registration from that document and
> publish it as a separate document.  Thus, I've put for this individual
> contribution honor those requests.
> 
> You'll note that I made reference to RFC 2793.  The rationale is that
> this registration should have already been done and we do not necessarily
> want the publication of the revisions RFC 2793 to hinder the registration
> of this MIME sub-type.  In any case, I'm open to changing the references,
> especially if they're both published at about the same time.
> 
> Please review this document and provide any comments.
> 
> Thanks,
> Paul
> 
> ----- Original Message ----- 
> From: <Internet-Drafts@ietf.org>
> To: <IETF-Announce:>
> Cc: <avt@ietf.org>
> Sent: Wednesday, November 19, 2003 3:26 PM
> Subject: I-D ACTION:draft-jones-avt-text-red-00.txt
> 
> 
> > A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> >
> >
> > Title : Registration of the text/red MIME Sub-Type
> > Author(s) : P. Jones
> > Filename : draft-jones-avt-text-red-00.txt
> > Pages : 5
> > Date : 2003-11-19
> >
> > This document defines the text/red MIME sub-type.  The packetization
> > and usage of this MIME type, which is intended for use within SDP, is
> > specified within RFC 2198 and RFC 2793, respectively.
> >
> > A URL for this Internet-Draft is:
> > http://www.ietf.org/internet-drafts/draft-jones-avt-text-red-00.txt
> >
> > To remove yourself from the IETF Announcement list, send a message to
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> > "get draft-jones-avt-text-red-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-jones-avt-text-red-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
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> >
> >
> > Below is the data which will enable a MIME compliant mail reader
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> > Internet-Draft.
> >
> 
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt


-- 
Colin Perkins
csp@csperkins.org

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>
Cc: "Phelan, Tom" <tphelan@sonusnet.com>, "'avt@ietf.org'" <avt@ietf.org>
Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
Date: Mon, 24 Nov 2003 09:49:03 -0500
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Magnus

I don't think the people working on the stability of the network
understand what they are proposing, or they wouldn't be proposing it,
except for Tom, who probably understands what the effect of 2X
bandwidth (when congestion was not occurring), and figures if that's
the only knob he's got, then he turns it far enough to make his app work,
and since he doesn't violate any rules, all is well.

I don't agree.  I think that, as Marshall observes, RTP media flows
tend to be self policing, and in any event, following the IAB
recommendation of dropping the flow if excessive packet loss is observed
will have the same effect - congestion collapse will not occur.
I think that we should stop thinking that we need anything else to
prevent congestion collapse.  

I think the real question this thread is addressing is whether DCCP
is at all suitable for real time flows.  I think the answer is no.
I think we should try see if we can reach consensus on this.
Given what has been said on this thread, is there anyone left who
thinks DCCP is suitable for RTP flows?  

This leaves us in a quandary, which, I think, is what Tom was trying to
avoid.  If DCCP is not suitable, is there any way to hold off the greedy
TCP?  This is the question Mike raises.  It's not an issue of congestion
collapse, it is the issue of "fairness".

My own feeling is that fairness to RTP is let the flow go through or don't.
There exists a variety of ways to vary the rate, but none of them are
useful.
Have you ever actually listened (or watched) a media stream with varying
rate?  I have, several times, several different algorithms.  It sucks, to
put it mildly.  Unless you have some evidence that there exists an
algorithm which does not annoy a user when the rate changes, I suspect 
our experience will be the common one - you will go down, once, and 
never go back up.  I expect even this will be very rare.   Much more common
is that you will try one rate, and if you don't get good results, the
call/stream will be dropped and tried again.  There are lots of algorithms
to vary rates.  There are no ways I know to vary quality, how ever it is
done,
and not be very distracting to a user.  Unless there is some kind of
concealment mechanism I am not aware of, it just won't be used, regardless
of how clever the algorithm is, and how bandwidth efficient it is.

So, to me, the idea that RTP reacts to whatever else is happening in any
way other than dropping the entire stream is not credible.  That means
there is no real rate control for RTP, TCP friendly or otherwise.
(I'll just use RTP vs TCP, even though we all understand that RTP is
carried on some other transport at the same level as TCP).

So what does it mean to be "fair"?
	For RTP, being fair to TCP (or other best effort traffic) is to
		not get started, or get dropped if congestion is excessive
	For TCP, being fair to RTP (or other constant rate traffic) is
		to not be greedy.

This is why QoS mechanisms work.  If you put RTP ahead of TCP,
and use admission control to limit the amount of RTP, everything works.
If you don't have a QoS mechanism, you can still perform admission
control to the RTP flows to enforce RTP fairness on TCP, but you need
to put some kind of mechanism to allow TCP to be fair to RTP.  I'm happy
to work on mechanisms that would afford effective admission control
in reaction to best effort traffic, but only if there is a commitment
to fix TCP.

Brian



> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Friday, November 21, 2003 10:40 AM
> To: Rosen, Brian
> Cc: Phelan, Tom; 'avt@ietf.org'
> Subject: Re: [AVT] RE: [MMUSIC] DCCP Media Guide
> 
> 
> Hi Brian,
> 
> I don't think that the people working with stability of the 
> network are 
> sharing you view due to the implications that your claims results in.
> 
> First, if a multi-media application transmitting data over a 
> best-effort 
> IP network, i.e. without any QoS guarantees across the whole net that 
> its traffic will fit, it can definitely cause congestion 
> collapse if the 
> mix becomes most, media, little TCP. Please read 
> draft-iab-congestion-01.txt for some discussion on this.
> 
> Some comments to your assertions.
> 
> Rosen, Brian wrote:
> > Phillipe
> > 
> > You have made invalid assumptions:
> > 	1. That audio/video media data rates will be changed dynamically
> > 		as network traffic changes.  This is very 
> unlikely.  It may
> > 		be that you get so much loss that you drop to a 
> lower data
> > 		rate, although I suspect that will be rare.  
> When you do,
> > 		you will never revert to the higher rate.  This 
> is because
> > 		humans hate to see the change, and minimizing 
> changes is 
> > 		more desirable than keeping average quality higher.
> 
> First you agree that it is possible to change the rate. Secondly I 
> understand that the there are implications on switching rate, i.e. 
> quality to often, however there are reasons for switching up when the 
> quality gain of a switch if being possible to keep the new 
> rate is clear.
> 
> > 
> > 	2. Real Time apps have no interest in being "fair".  
> They are all
> > 		or nothing. If you admit them to the network, you expect
> > 		the streams to be maintained.  If you can't 
> maintain them,
> > 		don't start them.  They don't want to be fair 
> to TCP, and
> > they
> > 		don't want to be fair to other RTP.  They just 
> want to send
> > 		data at the rate they were advised to do by signalling.
> > 		They can tolerate some loss.  Misordering=loss. 
>  Duplicates=
> > 		wasted bandwith.  That's a whole lot different from TCP
> > 		traffic.  
> > 
> 
> Yes, I agree that the application has no interest in being 
> fair. However 
> using best effort networks requires one to being reasonably 
> fair. If one 
> are not, the combination of several of media application can totally 
> destroy things for them self.
> 
> > I can imagine working on protocols which allowed CBR and 
> "best effort"
> > traffic to better coexist, but I think it's only possible 
> if the best
> > effort traffic recognized the presence of the CBR traffic 
> and reacted
> > appropriately.  Otherwise, you need to use QoS mechanisms 
> to separate
> > the traffic.
> 
> What says that the media traffic is more important that the traffic 
> going over TCP?
> 
> 
> Cheers
> 
> Magnus Westerlund
> 
> Multimedia Technologies, Ericsson Research EAB/TVA/A
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> 

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Mon, 24 Nov 2003 10:34:01 -0500
From: "David R. Oran" <oran@cisco.com>
To: "Rosen, Brian" <Brian.Rosen@marconi.com>,
        "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>
cc: "Phelan, Tom" <tphelan@sonusnet.com>, "'avt@ietf.org'" <avt@ietf.org>
Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
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Been watching this thread. Throwing caution to the wind and jumping in...

--On Monday, November 24, 2003 9:49 AM -0500 "Rosen, Brian"=20
<Brian.Rosen@marconi.com> wrote:

> Magnus
>
> I don't think the people working on the stability of the network
> understand what they are proposing, or they wouldn't be proposing it,
> except for Tom, who probably understands what the effect of 2X
> bandwidth (when congestion was not occurring), and figures if that's
> the only knob he's got, then he turns it far enough to make his app work,
> and since he doesn't violate any rules, all is well.
>
A factor of two of breathing room for a single low-rate audio session is=20
one thing. A factor of two on a multi-megabit video flow is quite another.=20
I think we need to take some factors into account:

a) nearly all the deployed coders, both audio and video, are CBR. More=20
importantly, they have extremely limited dynamic range on packetization=20
period, and hence on PPS generated by a transmitter. Schemes which switch=20
coders based on feedback are possible, but don't help much with the PPS=20
part and run into the human factors problems Brian mentioned in an earlier=20
post. I'm not enough of an expert in this aspect to have a useful opinion,=20
except to say that if the results of switching coders is less annoying than =

dropping lots of packets or dropping the whole flow, then it may still be=20
worthwhile.

b) VBR coders are a possibility, but it is not clear that they have enough=20
dynamic range (2x for audio, maybe 3x for video) to make a significant dent =

in congestion events. There is also the problem of whether they can adapt=20
quickly enough (i.e. 1 RTT) to play nice in a world with greedy TCPs that=20
go over the edge rather than throttle back gracefully using ECN.

c) I suspect Tom's 2x bandwidth scheme would not be "necessary" if all the=20
TCPs did ECN. True or false?

> I don't agree.  I think that, as Marshall observes, RTP media flows
> tend to be self policing, and in any event, following the IAB
> recommendation of dropping the flow if excessive packet loss is observed
> will have the same effect - congestion collapse will not occur.
> I think that we should stop thinking that we need anything else to
> prevent congestion collapse.
>
I agree with this, and this seems to cut to the fundamental architectural=20
fact-of-life - realtime media streams have very limited ability to adapt to =

congestion.

> I think the real question this thread is addressing is whether DCCP
> is at all suitable for real time flows.  I think the answer is no.
> I think we should try see if we can reach consensus on this.
> Given what has been said on this thread, is there anyone left who
> thinks DCCP is suitable for RTP flows?
>
I suspect that DCCP will be of very limited utility for *interactive*=20
realtime flows, where end-to-end delay has to fit within human interaction=20
timescales (1/4 second or less). However, there's a baby in the bath water, =

which is realtime media streaming/distribution, where DCCP may be a big win =

compared to either UDP or TCP encapsulation for RTP.

> This leaves us in a quandary, which, I think, is what Tom was trying to
> avoid.  If DCCP is not suitable, is there any way to hold off the greedy
> TCP?  This is the question Mike raises.  It's not an issue of congestion
> collapse, it is the issue of "fairness".
>
> My own feeling is that fairness to RTP is let the flow go through or
> don't. There exists a variety of ways to vary the rate, but none of them
> are useful.
I mostly agree with this.

> Have you ever actually listened (or watched) a media stream with varying
> rate?  I have, several times, several different algorithms.  It sucks, to
> put it mildly.
The question is whether it sucks worse than persistent x% loss or sucks=20
worse than denial of service. At some loss/delay level it is in fact better =

to drop the flow than try to adapt. I think we need to think in terms of=20
which impairment is worse rather than trying to convince people that=20
adaptation will be good enough so people don't notice.

> Unless you have some evidence that there exists an
> algorithm which does not annoy a user when the rate changes, I suspect
> our experience will be the common one - you will go down, once, and
> never go back up.  I expect even this will be very rare.   Much more
> common is that you will try one rate, and if you don't get good results,
> the call/stream will be dropped and tried again.  There are lots of
> algorithms to vary rates.  There are no ways I know to vary quality, how
> ever it is done,
> and not be very distracting to a user.  Unless there is some kind of
> concealment mechanism I am not aware of, it just won't be used, =
regardless
> of how clever the algorithm is, and how bandwidth efficient it is.
>
> So, to me, the idea that RTP reacts to whatever else is happening in any
> way other than dropping the entire stream is not credible.  That means
> there is no real rate control for RTP, TCP friendly or otherwise.
> (I'll just use RTP vs TCP, even though we all understand that RTP is
> carried on some other transport at the same level as TCP).
>
I'm not quite as pessimistic as Brian but I believe the onus is on those=20
who propose congestion adaptation schemes to show that they work from a=20
human-factors perspective rather than on the media folks to put up with=20
whatever loss and delay behavior congestion adaptation forces upon them.

> So what does it mean to be "fair"?
> 	For RTP, being fair to TCP (or other best effort traffic) is to
> 		not get started, or get dropped if congestion is excessive
> 	For TCP, being fair to RTP (or other constant rate traffic) is
> 		to not be greedy.
>
> This is why QoS mechanisms work.  If you put RTP ahead of TCP,
> and use admission control to limit the amount of RTP, everything works.
Yes...but...we'd like to have less state-heavy admission control schemes=20
which do measurement-based admission control, which puts us to some extent=20
right back  where we started.

> If you don't have a QoS mechanism, you can still perform admission
> control to the RTP flows to enforce RTP fairness on TCP, but you need
> to put some kind of mechanism to allow TCP to be fair to RTP.  I'm happy
> to work on mechanisms that would afford effective admission control
> in reaction to best effort traffic, but only if there is a commitment
> to fix TCP.
>
I'd like to understand why ECN won't bring the TCPs down enough in=20
greediness to avoid badly increasing the delay/loss behavior of the RTP=20
flows.

Dave.

> Brian
>
>
>
>> -----Original Message-----
>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
>> Sent: Friday, November 21, 2003 10:40 AM
>> To: Rosen, Brian
>> Cc: Phelan, Tom; 'avt@ietf.org'
>> Subject: Re: [AVT] RE: [MMUSIC] DCCP Media Guide
>>
>>
>> Hi Brian,
>>
>> I don't think that the people working with stability of the
>> network are
>> sharing you view due to the implications that your claims results in.
>>
>> First, if a multi-media application transmitting data over a
>> best-effort
>> IP network, i.e. without any QoS guarantees across the whole net that
>> its traffic will fit, it can definitely cause congestion
>> collapse if the
>> mix becomes most, media, little TCP. Please read
>> draft-iab-congestion-01.txt for some discussion on this.
>>
>> Some comments to your assertions.
>>
>> Rosen, Brian wrote:
>> > Phillipe
>> >
>> > You have made invalid assumptions:
>> > 	1. That audio/video media data rates will be changed dynamically
>> > 		as network traffic changes.  This is very
>> unlikely.  It may
>> > 		be that you get so much loss that you drop to a
>> lower data
>> > 		rate, although I suspect that will be rare.
>> When you do,
>> > 		you will never revert to the higher rate.  This
>> is because
>> > 		humans hate to see the change, and minimizing
>> changes is
>> > 		more desirable than keeping average quality higher.
>>
>> First you agree that it is possible to change the rate. Secondly I
>> understand that the there are implications on switching rate, i.e.
>> quality to often, however there are reasons for switching up when the
>> quality gain of a switch if being possible to keep the new
>> rate is clear.
>>
>> >
>> > 	2. Real Time apps have no interest in being "fair".
>> They are all
>> > 		or nothing. If you admit them to the network, you expect
>> > 		the streams to be maintained.  If you can't
>> maintain them,
>> > 		don't start them.  They don't want to be fair
>> to TCP, and
>> > they
>> > 		don't want to be fair to other RTP.  They just
>> want to send
>> > 		data at the rate they were advised to do by signalling.
>> > 		They can tolerate some loss.  Misordering=3Dloss.
>>  Duplicates=3D
>> > 		wasted bandwith.  That's a whole lot different from TCP
>> > 		traffic.
>> >
>>
>> Yes, I agree that the application has no interest in being
>> fair. However
>> using best effort networks requires one to being reasonably
>> fair. If one
>> are not, the combination of several of media application can totally
>> destroy things for them self.
>>
>> > I can imagine working on protocols which allowed CBR and
>> "best effort"
>> > traffic to better coexist, but I think it's only possible
>> if the best
>> > effort traffic recognized the presence of the CBR traffic
>> and reacted
>> > appropriately.  Otherwise, you need to use QoS mechanisms
>> to separate
>> > the traffic.
>>
>> What says that the media traffic is more important that the traffic
>> going over TCP?
>>
>>
>> Cheers
>>
>> Magnus Westerlund
>>
>> Multimedia Technologies, Ericsson Research EAB/TVA/A
>> ----------------------------------------------------------------------
>> Ericsson AB                | Phone +46 8 4048287
>> Torshamsgatan 23           | Fax   +46 8 7575550
>> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>>
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt



------------------------
David R. Oran
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720
Office: +1 978 264 2048
VoIP: +1 408 571 4576
Email: oran@cisco.com
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_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Mon Nov 24 11:02:21 2003
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Hi Alan,

Based on the discussion in this thread, and the dependency that the XR 
MIB extension will have on the core (as I see it), it seems necessary to 
make the extension dependent on a updated core. This means that the work 
can be performed in parallel, but the XR MIB extension can't go forward 
until the updated core is also available.

Cheers

Magnus Westerlund




_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Hi Paul,

I have a few comments and question.

1. In a WG document version you need to include the short copyright 
statement on the first page.

2. The abstract and first chapter: Is really usage of this mime type 
specified in RFC 2793? I thought this was only one possible usage of 
this MIME type. Technically this mime type could be used with any other 
format using the text and fulfilling any other requirement.

3. Shouldn't there be a chapter on mapping to SDP? As there are some 
tricks in relation to the defined MIME parameter "pt" and SDP. I think 
this is in line with the current explicit declaration of SDP usage.

Cheers

Magnus





_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'David R. Oran'" <oran@cisco.com>,
        "'Magnus Westerlund'"
	 <magnus.westerlund@ericsson.com>
Cc: "Phelan, Tom" <tphelan@sonusnet.com>, "'avt@ietf.org'" <avt@ietf.org>
Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
Date: Mon, 24 Nov 2003 12:06:45 -0500
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My experience is that changing rate more than once is much more annoying
than
changing it several times.  

In our system, a percentage of errors is less annoying than a single rate
change.  Go over that percentage (which of course is somewhat variable
from human to human), and you would prefer to drop the rate.  YCMV
(C=codec).

Neither is "acceptable", in the sense of what you want to see, and what
you expect to see.  If that's the expectation we create for VoIP or
VideoOverIP, then we had better be prepared for disappointing usage
except in the case where free trumps quality.

I don't see why DCCP is of significant value to a real time, non interactive
stream.  Why is it better than UDP or TCP?  I think that TRFC
is still a killer that extra buffering won't fix.  

Brian

> -----Original Message-----
> From: David R. Oran [mailto:oran@cisco.com]
> Sent: Monday, November 24, 2003 10:34 AM
> To: Rosen, Brian; 'Magnus Westerlund'
> Cc: Phelan, Tom; 'avt@ietf.org'
> Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> 
> 
> Been watching this thread. Throwing caution to the wind and 
> jumping in...
> 
> --On Monday, November 24, 2003 9:49 AM -0500 "Rosen, Brian" 
> <Brian.Rosen@marconi.com> wrote:
> 
> > Magnus
> >
> > I don't think the people working on the stability of the network
> > understand what they are proposing, or they wouldn't be 
> proposing it,
> > except for Tom, who probably understands what the effect of 2X
> > bandwidth (when congestion was not occurring), and figures if that's
> > the only knob he's got, then he turns it far enough to make 
> his app work,
> > and since he doesn't violate any rules, all is well.
> >
> A factor of two of breathing room for a single low-rate audio 
> session is 
> one thing. A factor of two on a multi-megabit video flow is 
> quite another. 
> I think we need to take some factors into account:
> 
> a) nearly all the deployed coders, both audio and video, are 
> CBR. More 
> importantly, they have extremely limited dynamic range on 
> packetization 
> period, and hence on PPS generated by a transmitter. Schemes 
> which switch 
> coders based on feedback are possible, but don't help much 
> with the PPS 
> part and run into the human factors problems Brian mentioned 
> in an earlier 
> post. I'm not enough of an expert in this aspect to have a 
> useful opinion, 
> except to say that if the results of switching coders is less 
> annoying than 
> dropping lots of packets or dropping the whole flow, then it 
> may still be 
> worthwhile.
> 
> b) VBR coders are a possibility, but it is not clear that 
> they have enough 
> dynamic range (2x for audio, maybe 3x for video) to make a 
> significant dent 
> in congestion events. There is also the problem of whether 
> they can adapt 
> quickly enough (i.e. 1 RTT) to play nice in a world with 
> greedy TCPs that 
> go over the edge rather than throttle back gracefully using ECN.
> 
> c) I suspect Tom's 2x bandwidth scheme would not be 
> "necessary" if all the 
> TCPs did ECN. True or false?
> 
> > I don't agree.  I think that, as Marshall observes, RTP media flows
> > tend to be self policing, and in any event, following the IAB
> > recommendation of dropping the flow if excessive packet 
> loss is observed
> > will have the same effect - congestion collapse will not occur.
> > I think that we should stop thinking that we need anything else to
> > prevent congestion collapse.
> >
> I agree with this, and this seems to cut to the fundamental 
> architectural 
> fact-of-life - realtime media streams have very limited 
> ability to adapt to 
> congestion.
> 
> > I think the real question this thread is addressing is whether DCCP
> > is at all suitable for real time flows.  I think the answer is no.
> > I think we should try see if we can reach consensus on this.
> > Given what has been said on this thread, is there anyone left who
> > thinks DCCP is suitable for RTP flows?
> >
> I suspect that DCCP will be of very limited utility for *interactive* 
> realtime flows, where end-to-end delay has to fit within 
> human interaction 
> timescales (1/4 second or less). However, there's a baby in 
> the bath water, 
> which is realtime media streaming/distribution, where DCCP 
> may be a big win 
> compared to either UDP or TCP encapsulation for RTP.
> 
> > This leaves us in a quandary, which, I think, is what Tom 
> was trying to
> > avoid.  If DCCP is not suitable, is there any way to hold 
> off the greedy
> > TCP?  This is the question Mike raises.  It's not an issue 
> of congestion
> > collapse, it is the issue of "fairness".
> >
> > My own feeling is that fairness to RTP is let the flow go through or
> > don't. There exists a variety of ways to vary the rate, but 
> none of them
> > are useful.
> I mostly agree with this.
> 
> > Have you ever actually listened (or watched) a media stream 
> with varying
> > rate?  I have, several times, several different algorithms. 
>  It sucks, to
> > put it mildly.
> The question is whether it sucks worse than persistent x% 
> loss or sucks 
> worse than denial of service. At some loss/delay level it is 
> in fact better 
> to drop the flow than try to adapt. I think we need to think 
> in terms of 
> which impairment is worse rather than trying to convince people that 
> adaptation will be good enough so people don't notice.
> 
> > Unless you have some evidence that there exists an
> > algorithm which does not annoy a user when the rate 
> changes, I suspect
> > our experience will be the common one - you will go down, once, and
> > never go back up.  I expect even this will be very rare.   Much more
> > common is that you will try one rate, and if you don't get 
> good results,
> > the call/stream will be dropped and tried again.  There are lots of
> > algorithms to vary rates.  There are no ways I know to vary 
> quality, how
> > ever it is done,
> > and not be very distracting to a user.  Unless there is some kind of
> > concealment mechanism I am not aware of, it just won't be 
> used, regardless
> > of how clever the algorithm is, and how bandwidth efficient it is.
> >
> > So, to me, the idea that RTP reacts to whatever else is 
> happening in any
> > way other than dropping the entire stream is not credible.  
> That means
> > there is no real rate control for RTP, TCP friendly or otherwise.
> > (I'll just use RTP vs TCP, even though we all understand that RTP is
> > carried on some other transport at the same level as TCP).
> >
> I'm not quite as pessimistic as Brian but I believe the onus 
> is on those 
> who propose congestion adaptation schemes to show that they 
> work from a 
> human-factors perspective rather than on the media folks to 
> put up with 
> whatever loss and delay behavior congestion adaptation forces 
> upon them.
> 
> > So what does it mean to be "fair"?
> > 	For RTP, being fair to TCP (or other best effort traffic) is to
> > 		not get started, or get dropped if congestion 
> is excessive
> > 	For TCP, being fair to RTP (or other constant rate traffic) is
> > 		to not be greedy.
> >
> > This is why QoS mechanisms work.  If you put RTP ahead of TCP,
> > and use admission control to limit the amount of RTP, 
> everything works.
> Yes...but...we'd like to have less state-heavy admission 
> control schemes 
> which do measurement-based admission control, which puts us 
> to some extent 
> right back  where we started.
> 
> > If you don't have a QoS mechanism, you can still perform admission
> > control to the RTP flows to enforce RTP fairness on TCP, 
> but you need
> > to put some kind of mechanism to allow TCP to be fair to 
> RTP.  I'm happy
> > to work on mechanisms that would afford effective admission control
> > in reaction to best effort traffic, but only if there is a 
> commitment
> > to fix TCP.
> >
> I'd like to understand why ECN won't bring the TCPs down enough in 
> greediness to avoid badly increasing the delay/loss behavior 
> of the RTP 
> flows.
> 
> Dave.
> 
> > Brian
> >
> >
> >
> >> -----Original Message-----
> >> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> >> Sent: Friday, November 21, 2003 10:40 AM
> >> To: Rosen, Brian
> >> Cc: Phelan, Tom; 'avt@ietf.org'
> >> Subject: Re: [AVT] RE: [MMUSIC] DCCP Media Guide
> >>
> >>
> >> Hi Brian,
> >>
> >> I don't think that the people working with stability of the
> >> network are
> >> sharing you view due to the implications that your claims 
> results in.
> >>
> >> First, if a multi-media application transmitting data over a
> >> best-effort
> >> IP network, i.e. without any QoS guarantees across the 
> whole net that
> >> its traffic will fit, it can definitely cause congestion
> >> collapse if the
> >> mix becomes most, media, little TCP. Please read
> >> draft-iab-congestion-01.txt for some discussion on this.
> >>
> >> Some comments to your assertions.
> >>
> >> Rosen, Brian wrote:
> >> > Phillipe
> >> >
> >> > You have made invalid assumptions:
> >> > 	1. That audio/video media data rates will be changed dynamically
> >> > 		as network traffic changes.  This is very
> >> unlikely.  It may
> >> > 		be that you get so much loss that you drop to a
> >> lower data
> >> > 		rate, although I suspect that will be rare.
> >> When you do,
> >> > 		you will never revert to the higher rate.  This
> >> is because
> >> > 		humans hate to see the change, and minimizing
> >> changes is
> >> > 		more desirable than keeping average quality higher.
> >>
> >> First you agree that it is possible to change the rate. Secondly I
> >> understand that the there are implications on switching rate, i.e.
> >> quality to often, however there are reasons for switching 
> up when the
> >> quality gain of a switch if being possible to keep the new
> >> rate is clear.
> >>
> >> >
> >> > 	2. Real Time apps have no interest in being "fair".
> >> They are all
> >> > 		or nothing. If you admit them to the network, you expect
> >> > 		the streams to be maintained.  If you can't
> >> maintain them,
> >> > 		don't start them.  They don't want to be fair
> >> to TCP, and
> >> > they
> >> > 		don't want to be fair to other RTP.  They just
> >> want to send
> >> > 		data at the rate they were advised to do by signalling.
> >> > 		They can tolerate some loss.  Misordering=loss.
> >>  Duplicates=
> >> > 		wasted bandwith.  That's a whole lot different from TCP
> >> > 		traffic.
> >> >
> >>
> >> Yes, I agree that the application has no interest in being
> >> fair. However
> >> using best effort networks requires one to being reasonably
> >> fair. If one
> >> are not, the combination of several of media application 
> can totally
> >> destroy things for them self.
> >>
> >> > I can imagine working on protocols which allowed CBR and
> >> "best effort"
> >> > traffic to better coexist, but I think it's only possible
> >> if the best
> >> > effort traffic recognized the presence of the CBR traffic
> >> and reacted
> >> > appropriately.  Otherwise, you need to use QoS mechanisms
> >> to separate
> >> > the traffic.
> >>
> >> What says that the media traffic is more important that the traffic
> >> going over TCP?
> >>
> >>
> >> Cheers
> >>
> >> Magnus Westerlund
> >>
> >> Multimedia Technologies, Ericsson Research EAB/TVA/A
> >> 
> ----------------------------------------------------------------------
> >> Ericsson AB                | Phone +46 8 4048287
> >> Torshamsgatan 23           | Fax   +46 8 7575550
> >> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> >>
> >
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www1.ietf.org/mailman/listinfo/avt
> 
> 
> 
> ------------------------
> David R. Oran
> Cisco Systems
> 7 Ladyslipper Lane
> Acton, MA 01720
> Office: +1 978 264 2048
> VoIP: +1 408 571 4576
> Email: oran@cisco.com
> 

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Mon Nov 24 12:29:16 2003
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	(envelope-from casner@acm.org)
Date: Mon, 24 Nov 2003 09:27:25 -0800 (PST)
From: Stephen Casner <casner@acm.org>
To: "Phelan, Tom" <tphelan@sonusnet.com>
cc: AVT WG <avt@ietf.org>
Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
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I think Tom's description of the problem as enabling DCCP to get its
fair share vs. TCP has misdirected the thinking.  The problem he
identified was that TCP's bandwidth probing will cause some loss in
the DCCP stream as well.  Here is the key: that probing will cause
exactly the same loss in an RTP/UDP stream.  It is only DCCP's
reaction to the loss that is a design question.

For interactive (delay-sensitive) media, FEC may be an option for
protecting against the isolated packet losses induced by TCP probing.
This can be achieved with less than 2X bandwidth.  But other than
removing the FEC, and possibly shifting encodings, there is not much
room for the media to accommodate any direction from DCCP to back
off.

As I've said more than once in DCCP WG meetings, these applications
don't want to be congestion controlled.  Unless there is policing in
the network that causes performance with UDP to be worse than with
DCCP, there is no motivation to change protocols.

On the other hand, for streaming media to clients with significant
buffer capacity, then the approach of sending at 2X rate (if the
bottleneck allows) until the buffer is filled is a fine idea.  During
the 2X period, if DCCP says to back off by half, the application is
still getting its necessary throughput.  After the buffer is full, the
rate will have to be 1X, but if DCCP says to back off from that as a
result of TCP probing, the buffer will cover the period until the rate
ramps back up.  The rate can go over 1x again to refill the buffer,
then back to 1X.

This isn't a new idea.  DCCP would just legitimize and standardize
it.

                                                        -- Steve

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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To: "Colin Perkins" <csp@csperkins.org>
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Subject: Re: [AVT] Re: I-D ACTION:draft-jones-avt-text-red-00.txt
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Colin,

Thanks.  I've corrected both of those.  I'll wait for advise on whether this
should be re-published as an AVT working group item.

Paul

----- Original Message ----- 
From: "Colin Perkins" <csp@csperkins.org>
To: "Paul E. Jones" <paulej@packetizer.com>
Cc: <avt@ietf.org>; <magnus.westerlund@ericsson.com>
Sent: Sunday, November 23, 2003 8:01 PM
Subject: Re: [AVT] Re: I-D ACTION:draft-jones-avt-text-red-00.txt


> Paul, all,
>
> Looks good -- I noticed two minor nits: reference [2] is not actually
> referenced anywhere in the draft, and there is a Microsoft smart quote
> in Reference 1.
>
> I suggest we take this as an AVT work item, and progress it quickly.
> If any group member objects to this becoming a work item, please send
> your comment to this list. If there are no objections, I suggest this
> is submitted as draft-ietf-avt-text-red-00.txt at the end of the week,
> with my comments above corrected.
>
> Colin
>
>
>
>
> --> "Paul E. Jones" <paulej@packetizer.com> writes:
> > Folks,
> >
> > I just wanted to bring this particular document to your attention.
> > During the review of RFC 2793bis, it was proposed by several folks that
> > we extract the text/red MIME sub-type registration from that document
and
> > publish it as a separate document.  Thus, I've put for this individual
> > contribution honor those requests.
> >
> > You'll note that I made reference to RFC 2793.  The rationale is that
> > this registration should have already been done and we do not
necessarily
> > want the publication of the revisions RFC 2793 to hinder the
registration
> > of this MIME sub-type.  In any case, I'm open to changing the
references,
> > especially if they're both published at about the same time.
> >
> > Please review this document and provide any comments.
> >
> > Thanks,
> > Paul
> >
> > ----- Original Message ----- 
> > From: <Internet-Drafts@ietf.org>
> > To: <IETF-Announce:>
> > Cc: <avt@ietf.org>
> > Sent: Wednesday, November 19, 2003 3:26 PM
> > Subject: I-D ACTION:draft-jones-avt-text-red-00.txt
> >
> >
> > > A New Internet-Draft is available from the on-line Internet-Drafts
> > directories.
> > >
> > >
> > > Title : Registration of the text/red MIME Sub-Type
> > > Author(s) : P. Jones
> > > Filename : draft-jones-avt-text-red-00.txt
> > > Pages : 5
> > > Date : 2003-11-19
> > >
> > > This document defines the text/red MIME sub-type.  The packetization
> > > and usage of this MIME type, which is intended for use within SDP, is
> > > specified within RFC 2198 and RFC 2793, respectively.
> > >
> > > A URL for this Internet-Draft is:
> > > http://www.ietf.org/internet-drafts/draft-jones-avt-text-red-00.txt
> > >
> > > To remove yourself from the IETF Announcement list, send a message to
> > > ietf-announce-request with the word unsubscribe in the body of the
> > message.
> > >
> > > 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-jones-avt-text-red-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-jones-avt-text-red-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.
> > >
> >
> >
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www1.ietf.org/mailman/listinfo/avt
>
>
> -- 
> Colin Perkins
> csp@csperkins.org
>


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Paul E. Jones" <paulej@packetizer.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>
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Magnus,

> 1. In a WG document version you need to include the short copyright
> statement on the first page.

Can you point to a sample document and section?  I tried to copy the text I
found in other WG documents, though it would surprise me that I left
something out.

> 2. The abstract and first chapter: Is really usage of this mime type
> specified in RFC 2793? I thought this was only one possible usage of
> this MIME type. Technically this mime type could be used with any other
> format using the text and fulfilling any other requirement.

Your point is valid, as RFC 2793 might be just one of several documents that
might specify the use of this.  However, if I think of audio/red, I think of
RFC 2198.  I think what you're saying is that for use with SDP, 2198/2793
are apropos, but necessarily other applications that might use this mime
sub-type.  Since I have no other example, should I just say something along
those lines?

> 3. Shouldn't there be a chapter on mapping to SDP? As there are some
> tricks in relation to the defined MIME parameter "pt" and SDP. I think
> this is in line with the current explicit declaration of SDP usage.

Yes, you're right.  Essentially, it would follow chapter 3 of RFC 3555.  I
will create a chapter and simply refer to that text, if that's acceptable.

Paul


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Hi,

Sorry for being so late with my comments. I think the concept is good 
enough however there is quite a lot of work left until this is a good 
protocol specification.

1. I think of a lot of the background material on how Reed Salomon codes 
work should be moved to the introduction. In other word section 3,4,5 
can be put into the a common chapter, possibly with subsections, however 
I think one should be clearer about they being introductionary.

2. Then there is need to start writing some real payload specification 
chapter. Of the present chapter I think the largest hole is how one goes 
from a unprotected stream into a protected one. As I see it one will 
take a series of complete RTP packets and then put them into payloads of 
new RTP packets.

3. There is need to be much clearer on how one fills in the protection 
matrix and then can on the receiver side understand how one has written 
the bytes from different packets into the matrix to correctly recover it.

4. I think you authors should consider to include some more blank lines, 
or in certain cases do a page break between section to improve the 
readability of the text. For example the TOC should be moved into a 
separate page. Please do also put in a blank line between each paragraph.

5. You are missing the short copyright statement after status of this 
memo and the abstract. See ID-nits http://www.ietf.org/ID-nits.html

6. Chapter 2 is very hard to read. I would suggest that instead of using 
numbered bullets, you use the definition being left justified and then 
put the definition more right justified. You may also consider of all 
these definitions really make sense to be defined in conventions or if 
they more belong in there respective chapter.

7. The H.26L is now H.264,

8. In section 4. The following sentence exist:
"Furthermore, if channel
    conditions get worse during transmission, only more and more
    enhancement layers are lost, i.e. a graceful degradation in
    application quality at the receiver is achieved [6]."

This sentence can be rather dangerous in IETF. The general channel you 
are talking about is the IP channel which normally only losses packets 
due to congestion. Therefore this sentence can be interpreted as by 
simply adding more protection one can avoid most effects of congestion. 
So if you are talking about the very special case for links that loses 
packet of other reasons I think you need to be very explicit.

9. Figure 3 should be moved into the introduction when you discuss Reed 
Salomon codes.

10.  Page 11, bullet 2. How does on determine the expected number of TSB 
rows?

11. What RTP timestamp rate should be used for this format? Does it 
really need one? or could one simply transmission time stamp the packets?

12. Please use more normative wording when defining how the marker bit 
should be set.

13. Section 6.1: "All other fields in the RTP header are set to those 
values
    proposed for regular multimedia transmission using the RTP-format
    of the media stream which is protected by UXP, e.g. for MPEG-4
    Visual as specified in RFC 3016."
Normal there are no more fields then RTP TS and Marker bit that the 
payload format defines, the rest is set in accordance with the RTP 
specification and the profile in use.

14. What is the gain of having the protected media's PT available in the 
UXP header? As I understand it you will need to decode the minimal 
number packets possible to reconstruct the stream to use it at all.

15. How does one treat packets where one is able to recover parts of the 
packets?

16. The proposed RTP Timestamp assignment mechanism will make the RTCP 
jitter unusable, if this is deemed acceptable, then at least it should 
be noted.

17. I would recommend that the description of the processing becomes 
much less media oriented and much more common RTP and its payload specific.

18. I would also like that the specification is more explicit how can 
give different protection levels to different packets or parts of a 
packet. There need to be more information about this topic making it 
possible for a implementor of UXP for a certain purpose being able to 
determine how they should behave.

19. This payload format need to register a MIME subtype for all the main 
media types that uses RTP at all.

20. Then one can write a chapter on how to map this to SDP.

21. In the security section the following text is present:
" The overall ratio between parity and info octets MUST NOT be
    higher than 1:1, i.e. the absolute bitrate spent for redundancy
    must not be larger than the bitrate required for transmission of
    multimedia data itself."
   
This statement I understand is intended to address congestion control. 
This isn't very useful, as the transmission rate in the face of loss in 
a best effort network shall not be higher than what a TCP flow would 
achieve over the same path. However the mix between protection and real 
data can be any and should be controlled only in regards to the paths 
error characteristics. This results in a the trade-off between 
protection and media.

22. You are missing the IETF standard IPR statement, see ID-nits for 
references to the sections from RFC 2026 you need.

23. I think you can write an separate address for each of the authors.

I hope this will provide you with input in writing the next version.

Cheers

Magnus Westerlund




   



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Hi Paul,



Paul E. Jones wrote:

>Magnus,
>
>  
>
>>1. In a WG document version you need to include the short copyright
>>statement on the first page.
>>    
>>
>
>Can you point to a sample document and section?  I tried to copy the text I
>found in other WG documents, though it would surprise me that I left
>something out.
>  
>

The copyright statement people has been using is the first line of the 
section 10.3. (C) of RFC 2026. However it seems that the ID-nits are no 
longer required that it is included. So I withdraw my comment.

>  
>
>>2. The abstract and first chapter: Is really usage of this mime type
>>specified in RFC 2793? I thought this was only one possible usage of
>>this MIME type. Technically this mime type could be used with any other
>>format using the text and fulfilling any other requirement.
>>    
>>
>
>Your point is valid, as RFC 2793 might be just one of several documents that
>might specify the use of this.  However, if I think of audio/red, I think of
>RFC 2198.  I think what you're saying is that for use with SDP, 2198/2793
>are apropos, but necessarily other applications that might use this mime
>sub-type.  Since I have no other example, should I just say something along
>those lines?
>

Yes, the MIME type can be used by any current or future text payload 
format that fulfil the requirement imposed by RFC 2198.

>
>  
>
>>3. Shouldn't there be a chapter on mapping to SDP? As there are some
>>tricks in relation to the defined MIME parameter "pt" and SDP. I think
>>this is in line with the current explicit declaration of SDP usage.
>>    
>>
>
>Yes, you're right.  Essentially, it would follow chapter 3 of RFC 3555.  I
>will create a chapter and simply refer to that text, if that's acceptable.
>  
>

It has been usually to actually include a copy and provide the details 
of how things are included. Especially you need to make it clear how the 
mapping of PT is done to the FMTP line.

Cheers

Magnus



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Lars-Erik Jonsson (LU/EAB)" <lars-erik.jonsson@ericsson.com>
To: "'Ash, Gerald R (Jerry), ALABS'" <gash@att.com>, avt@ietf.org
Subject: RE: [AVT] HC over multihop / MPLS
Date: Tue, 25 Nov 2003 12:39:10 +0100
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Jerry,

This seems like a reasonable work plan!

Rgds,
/L-E


> -----Original Message-----
> From: Ash, Gerald R (Jerry), ALABS [mailto:gash@att.com]
> Sent: den 21 november 2003 17:51
> To: Lars-Erik Jonsson (LU/EAB); avt@ietf.org
> Cc: Ash, Gerald R (Jerry), ALABS
> Subject: RE: [AVT] HC over multihop / MPLS
> 
> 
> Lars-Erik,
> 
> I very much appreciate your taking the time to read the draft 
> and comment.
> 
> A lot of the questions raised at the AVT meeting are 
> addressed in the draft, so it helps when people do read the draft.
> 
> Allison, Colin, Magnus, and I met after the meeting and 
> agreed on the following approach for next steps:
> 
> 1. split the current document 
> http://ietf.org/internet-drafts/draft-ash-e2e-voip-hdr-comp-rq
> mts-01.txt into 2 requirements documents:
> a. 'requirements for hop-by-hop ECRTP over MPLS' (RSVP-TE 
> protocol extension oriented)
> b. 'requirements for hop-by-hop ECRTP over IP' (SCID 
> switching oriented)
> (note that Colin suggests 'hop-by-hop' versus 'multi-hop' 
> terminology; 'multi-hop was meant to clarify 'end-to-end' in 
> the OO version, but still may have led to some confusion).
> 2. eliminate MPLS jargon such as CE, PE, etc., just talk 
> about routers R1, R2, etc.
> 3. eliminate absolute performance objectives and simply state 
> goal to 'minimize delta degradation' (for delay, jitter, pkt 
> loss, etc.).  This includes performance degradation due to 
> too much resync.
> 4. eliminate absolute scalability requirement for # of flows, 
> and simply state goal to improve scalability of the number of 
> compressed flows that can be handled.
> 5. use already standard protocols for compression, signaling, etc.
> 
> It was agreed that requirements 1a should lead to an RSVP-TE 
> protocol extension effort and be worked in conjunction with 
> the MPLS SG, and that requirements 1b could lead to an 
> Informational RFC out of AVT.
> 
> Please let me know of any comments or suggestions on this plan.
> 
> I've made a few other brief comments below on your email. 
> 
> Thanks,
> Jerry
> 
> > The AVT charter states the following as a WG task: 
> > "to review the applicability of Compressed RTP operation on
> > MPLS networks, developing extensions as necessary"
> > 
> > With the following milestones:
> > Nov 03    Initial draft requirements for ECRTP over MPLS; 
> > discuss with MPLS WG
> > Nov 03    Initial draft requirements for ECRTP over MPLS; 
> > discuss with MPLS WG
> > 
> > However, the current draft refers to "VoIP Header Compression 
> > over Multi-Hop Paths", which is not the same thing. The draft
> > clearly shows that compression over multihop paths is one thing, 
> > and compression over MPLS another. Further, the draft takes a 
> > wider scope, and talks about header compression generally,
> > not only ECRTP as the charter says.
> > 
> > The following observations can be made:
> > * If work is to done for header compression over multihop 
> > and/or MPLS, it might make sense to not only address ECRTP.
> 
> Agreed.  The plan to split the requirements draft into 2 
> separate requirements drafts (ECRTP over MPLS, ECRTP over IP) 
> addresses this, as outlined above.
> 
> > * The only work in this that naturally could belong in the 
> > AVT WG would be to make sure ECRTP would work in these 
> > scenarios.  In my understanding, this should already be 
> > the case with ECRTP (as well as some ROHC profiles), which 
> > is also indicated by the multihop req draft.
> 
> Agreed.
> 
> > * Header compression over multihop paths [ described by the 
> > multihop req draft as where one wants to avoid 
> > decompression/compression steps at intermediate routers if 
> > the same compression scheme is supported at both incoming 
> > and outgoing interfaces ] is fully possible also without any 
> > new standards, as it would only be an implementation issue for 
> > each intermediate router to bind an incoming compressed stream
> > to its corresponding outgoing compressed stream. In a path 
> > A->B->C, where B is the intermediate router
> > negotiating use of e.g. CRTP with both A and C, it is fully 
> > up to the implementation of B to avoid unnecessary 
> > decompression/compression and just forward compressed packets
> > as-is ("CID routing").
> 
> Agreed (requirements draft 1b, as outlined above, leading to 
> an Informational RFC).
> 
> > * The only significant standardization work I can see is to 
> > define how to do header compression (e.g. IPHC, CRTP/ECRTP, 
> > ROHC) over MPLS, which is  probably a work that would require 
> > some coordination with AVT and ROHC individuals, but the core
> > of that work does not naturally belong to ROHC or AVT. Would
> > not the MPLS WG be a more proper place?
> 
> I think the plan is to complete the requirements 1a and the 
> follow-on protocol work in conjunction with the MPLS WG.
> 

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Lars-Erik Jonsson (LU/EAB)" <lars-erik.jonsson@ericsson.com>
To: "'Phelan, Tom'" <tphelan@sonusnet.com>
Cc: "'avt@ietf.org'" <avt@ietf.org>, "'dccp@ietf.org'" <dccp@ietf.org>
Subject: RE: [dccp] RE: [AVT] RE: [MMUSIC] DCCP Media Guide
Date: Tue, 25 Nov 2003 12:52:13 +0100
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Tom,

We do not just have to find ways to get some kind of "fairness"
between applications, but we actually have two main goals:
1) Avoid congestion
2) Improve network utilization

By probing the network, TCP strive for 2), and that is a good
think. When there is available bandwidth, it is a good thing
that this resource gets used for actual application data. 
However, sending data just to keep network resources degrades
goal 2). To me, this seems to be a tricky problem as TCP has
not at all been designed to interact with real-time streams.

/L-E

> -----Original Message-----
> From: dccp-admin@ietf.org [mailto:dccp-admin@ietf.org]On Behalf Of
> Phelan, Tom
> Sent: den 20 november 2003 19:04
> To: 'philippe.gentric@philips.com'; Phelan, Tom; 'Rosen, Brian'
> Cc: 'avt@ietf.org'; 'dccp@ietf.org'; 'mmusic@ietf.org'
> Subject: [dccp] RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> 
> 
> Hi Phillipe,
> 
> The situation I'm worried about is a little different than 
> you describe. Say you have a 10M bps link, over which there's
> a 4M bps video stream and a long-lived TCP app.  The TCP app
> will take the remaining 6M bps, and then push for more,
> which might cause the video stream to lose packets.  The
> video stream then drops to 2M bps, and the TCP app expands to 
> 8M bps.  This hardly seems fair.
> 
> The video app can protect itself from this by transmitting 
> enough more than 4M bps to buffer itself against the TCP
> probes.  In this case we'd come to a relatively steady state
> where both apps are sending roughly 5M bps.  This seems
> reasonably fair to me, since the TCP app gets what it would
> get if it were competing with other TCP apps, and the video
> app gets to send at its full rate.
> 
> Tom P.

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Lars-Erik Jonsson (LU/EAB)" <lars-erik.jonsson@ericsson.com>
To: "'Rosen, Brian'" <Brian.Rosen@marconi.com>
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Subject: RE: [dccp] RE: [AVT] RE: [MMUSIC] DCCP Media Guide
Date: Tue, 25 Nov 2003 12:53:01 +0100
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> I have looked carefully at the drafts.  I think I understand what 
> the proposal is, and how it works.  I'll even agree that it might 
> do what you think it will do, but I suspect that my device will 
> cause a heck of a lot more congestion to the network if I run the 
> video streams on DCCP than if I leave it on UDP.  I think the 
> solution will be much worse than the problem.
> 
> The solution is just plain broken, in my opinion.  The real, hard, 
> problem is that there probably is no good solution, short of fixing 
> TCP, or treating real time data differently (ie QoS) in the router.  
> Like it or not, RTP is not appropriate for what we usually call 
> "best effort", but the problem with that statement is that it's not 
> really true that we need something more than for the network to use 
> its' best effort to deliver the traffic.  What is true is that other 
> streams have to deal better with CBR data than they do.  It is TCP 
> that is broken. I don't think there is any way around that.  
> TCP makes an assumption that is invalid when real time media is part 
> of the traffic mix.
> 
> There are so many problems with DCCP applied to real time media (this 
> one, slow start,...), that I think we should just stop believing that 
> DCCP is appropriate for RTP.

Unfortunately, I think this analysis is rather correct. If so, I 
wonder if it really makes sense to continue with the DCCP work,
if it can not solve the problem it is supposed to address. At
least we should slow down a bit and give it more time, hopefully
we can activate more people and get them involved in the work.

More specifically, I think it is totally wrong to postpone the
actual work as "future CC work" if DCCP with current CC's do not
solve the problems it was chartered to solve and claims to address.

/L-E

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
To: Stephen Casner <casner@acm.org>
Cc: AVT WG <avt@ietf.org>
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Stephen,

Unfortunately filling the client buffer at a rate faster than the media rate
and "when you can" which introduces substancial amounts of time
(many seconds) of delay is a technique for media distribution
for which DCCP is not needed:

It is called "progressive" download, it uses HTTP and it is deployed ...

regards,


Philippe Gentric
Chief Architect
Philips Software
philippe.gentric@philips.com
http://www.software.philips.com


                                                                                                                                       
                                                                                                                                       
                                                   To:   "Phelan, Tom" <tphelan@sonusnet.com>                                          
                                                   cc:   AVT WG <avt@ietf.org>                                                         
                                                    (bcc: Philippe Gentric/MP4-SUR/CE/PHILIPS)                                         
                                                   Subject:    RE: [AVT] RE: [MMUSIC] DCCP Media Guide                                 
               Stephen Casner                                                                                                          
               <casner@acm.org>                    Classification:                                                                     
                                                                                                                                       
               Sent by:                                                                                                                
               avt-admin@ietf.org                                                                                                      
                                                                                                                                       
               24/11/2003 18:27                                                                                                        
                                                                                                                                       
                                                                                                                                       




I think Tom's description of the problem as enabling DCCP to get its
fair share vs. TCP has misdirected the thinking.  The problem he
identified was that TCP's bandwidth probing will cause some loss in
the DCCP stream as well.  Here is the key: that probing will cause
exactly the same loss in an RTP/UDP stream.  It is only DCCP's
reaction to the loss that is a design question.

For interactive (delay-sensitive) media, FEC may be an option for
protecting against the isolated packet losses induced by TCP probing.
This can be achieved with less than 2X bandwidth.  But other than
removing the FEC, and possibly shifting encodings, there is not much
room for the media to accommodate any direction from DCCP to back
off.

As I've said more than once in DCCP WG meetings, these applications
don't want to be congestion controlled.  Unless there is policing in
the network that causes performance with UDP to be worse than with
DCCP, there is no motivation to change protocols.

On the other hand, for streaming media to clients with significant
buffer capacity, then the approach of sending at 2X rate (if the
bottleneck allows) until the buffer is filled is a fine idea.  During
the 2X period, if DCCP says to back off by half, the application is
still getting its necessary throughput.  After the buffer is full, the
rate will have to be 1X, but if DCCP says to back off from that as a
result of TCP probing, the buffer will cover the period until the rate
ramps back up.  The rate can go over 1x again to refill the buffer,
then back to 1X.

This isn't a new idea.  DCCP would just legitimize and standardize
it.

                                                        -- Steve

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt






_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: RE: [dccp] RE: [AVT] RE: [MMUSIC] DCCP Media Guide
To: "Lars-Erik Jonsson (LU/EAB)" <lars-erik.jonsson@ericsson.com>
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Lars-Erik,

I share your concern about "sending garbage" to "keep the pipe open".

I understand now that "probing" is required to achieve (2) in fairness to TCP
i.e. a traffic that is competing with TCP has to use the same "weapon".

My idea would be that if the TCP traffic is allowed to push the DCCP-born media traffic down in rate
I expect "probing" to be performed in order for the media traffic to discover that it can increase its rate,
**after a decrease**, if/when it can, i.e. when some/all of the TCP traffic is finished (which would be fair).

In short for media traffic via DCCP "probing" would be a continuous activity in order to push against
the TCP traffic,(the amount of pressure/resistance being tunable ?)

I am expecting a TCP-well-behaved media application to lower its rate (but not systematically by half ?)
if more TCP traffic happens and I also assume that all media applications should be
prepared/defined to either stop reducing their rate AND/OR cut off below a certain rate,
i.e. this should be tunable by marketing/policy/management.

I agree with Stephen that in all case probing means losses on both sides but
I expect error resiliency to cover that.

 I think that the sensible way to "probe" is to send "more media/packets" periodically (in bursts)
(instead of sending garbage as a shield of some sort).

In all cases I beleive the key is to understand the minimum rate of change of the data/packet rates
and minimum amplitude of the rate changes, the periodicty of the "probes" etc
that are required for maintaining a tunable level of "resistance" against TCP traffic.

So far it seems that only the "factor of 2" has been considered,
this may seem quite large (too large for a lot of apps and codecs)
and could make the future of DCCP pretty bleak.

Let me express my optimistic view as a metaphor:

To me it looks like the discussion has been a bit like discussing the usage of tires on cars and there would be 2 sides,
one that is against it and defends that raw steel is good enough (with FEC cushions on the seats !)
and the other extreme opinion that 2 meters thick balloon tires are what is required for good confort.

Could the truth be somewhere in the middle ?-)


Regards,


Philippe Gentric
Chief Architect
Philips Software
philippe.gentric@philips.com
http://www.software.philips.com


                                                                                                                                           
                                                                                                                                           
                                                       To:   "'Phelan, Tom'" <tphelan@sonusnet.com>                                        
                                                       cc:   "'avt@ietf.org'" <avt@ietf.org>                                               
                                                        "'dccp@ietf.org'" <dccp@ietf.org>                                                  
                                                        (bcc: Philippe Gentric/MP4-SUR/CE/PHILIPS)                                         
               "Lars-Erik Jonsson (LU/EAB)"            Subject:    RE: [dccp] RE: [AVT] RE: [MMUSIC] DCCP Media Guide                      
               <lars-erik.jonsson@ericsson.                                                                                                
               com>                                    Classification:                                                                     
                                                                                                                                           
               Sent by:                                                                                                                    
               avt-admin@ietf.org                                                                                                          
                                                                                                                                           
               25/11/2003 12:52                                                                                                            
                                                                                                                                           
                                                                                                                                           




Tom,

We do not just have to find ways to get some kind of "fairness"
between applications, but we actually have two main goals:
1) Avoid congestion
2) Improve network utilization

By probing the network, TCP strive for 2), and that is a good
think. When there is available bandwidth, it is a good thing
that this resource gets used for actual application data.
However, sending data just to keep network resources degrades
goal 2). To me, this seems to be a tricky problem as TCP has
not at all been designed to interact with real-time streams.

/L-E

> -----Original Message-----
> From: dccp-admin@ietf.org [mailto:dccp-admin@ietf.org]On Behalf Of
> Phelan, Tom
> Sent: den 20 november 2003 19:04
> To: 'philippe.gentric@philips.com'; Phelan, Tom; 'Rosen, Brian'
> Cc: 'avt@ietf.org'; 'dccp@ietf.org'; 'mmusic@ietf.org'
> Subject: [dccp] RE: [AVT] RE: [MMUSIC] DCCP Media Guide
>
>
> Hi Phillipe,
>
> The situation I'm worried about is a little different than
> you describe. Say you have a 10M bps link, over which there's
> a 4M bps video stream and a long-lived TCP app.  The TCP app
> will take the remaining 6M bps, and then push for more,
> which might cause the video stream to lose packets.  The
> video stream then drops to 2M bps, and the TCP app expands to
> 8M bps.  This hardly seems fair.
>
> The video app can protect itself from this by transmitting
> enough more than 4M bps to buffer itself against the TCP
> probes.  In this case we'd come to a relatively steady state
> where both apps are sending roughly 5M bps.  This seems
> reasonably fair to me, since the TCP app gets what it would
> get if it were competing with other TCP apps, and the video
> app gets to send at its full rate.
>
> Tom P.

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt






_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Tue Nov 25 11:38:24 2003
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From: Greg Herlein <gregh@arkstream.com>
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Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
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> Unfortunately filling the client buffer at a rate faster than
> the media rate and "when you can" which introduces substancial
> amounts of time (many seconds) of delay is a technique for
> media distribution for which DCCP is not needed:
> 
> It is called "progressive" download, it uses HTTP and it is deployed ...

That's somewhat simplistic, really.  Progressive download has a 
similar effect but it's not controlled, and it's TCP only of 
course.

For unicast scenarios this kind of thing makes total sense if the
source is non-real time (ie, streaming from disk).  It's not much
more than an enhanced jitter buffer, really - you are just adding
some extra buffer for times when the normal real-time flow is
impaired, and a mechanism to allow the client to tell the source
to "fill me up"  and "back it off."  There need not be any extra
delay at startup, though not doing so creates a window of
vulnerability.

I'd very much like to see some standardization effort on a
protocol to support this kind of thing - perhaps an RTCP
extension?  Perhaps soemthing that can apply to TCP as well?

I'm willing to help on such an effort.

Greg Herlein
VP Engineering
Arkstream, Inc.


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: RE: [dccp] RE: [AVT] RE: [MMUSIC] DCCP Media Guide
Date: Tue, 25 Nov 2003 14:06:42 -0500
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This would seem to only apply to a non-interactive stream, where
several seconds of buffering is acceptable, correct?
You also, in this case, would not actually change the encoding
rate, but merely change the instantaneous transmit rate.

I'm not clear on how this is better than just using TCP.

Brian

> -----Original Message-----
> From: philippe.gentric@philips.com 
> [mailto:philippe.gentric@philips.com]
> Sent: Tuesday, November 25, 2003 10:40 AM
> To: Lars-Erik Jonsson (LU/EAB)
> Cc: 'avt@ietf.org'; 'dccp@ietf.org'
> Subject: RE: [dccp] RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> 
> 
> Lars-Erik,
> 
> I share your concern about "sending garbage" to "keep the pipe open".
> 
> I understand now that "probing" is required to achieve (2) in 
> fairness to TCP
> i.e. a traffic that is competing with TCP has to use the same 
> "weapon".
> 
> My idea would be that if the TCP traffic is allowed to push 
> the DCCP-born media traffic down in rate
> I expect "probing" to be performed in order for the media 
> traffic to discover that it can increase its rate,
> **after a decrease**, if/when it can, i.e. when some/all of 
> the TCP traffic is finished (which would be fair).
> 
> In short for media traffic via DCCP "probing" would be a 
> continuous activity in order to push against
> the TCP traffic,(the amount of pressure/resistance being tunable ?)
> 
> I am expecting a TCP-well-behaved media application to lower 
> its rate (but not systematically by half ?)
> if more TCP traffic happens and I also assume that all media 
> applications should be
> prepared/defined to either stop reducing their rate AND/OR 
> cut off below a certain rate,
> i.e. this should be tunable by marketing/policy/management.
> 
> I agree with Stephen that in all case probing means losses on 
> both sides but
> I expect error resiliency to cover that.
> 
>  I think that the sensible way to "probe" is to send "more 
> media/packets" periodically (in bursts)
> (instead of sending garbage as a shield of some sort).
> 
> In all cases I beleive the key is to understand the minimum 
> rate of change of the data/packet rates
> and minimum amplitude of the rate changes, the periodicty of 
> the "probes" etc
> that are required for maintaining a tunable level of 
> "resistance" against TCP traffic.
> 
> So far it seems that only the "factor of 2" has been considered,
> this may seem quite large (too large for a lot of apps and codecs)
> and could make the future of DCCP pretty bleak.
> 
> Let me express my optimistic view as a metaphor:
> 
> To me it looks like the discussion has been a bit like 
> discussing the usage of tires on cars and there would be 2 sides,
> one that is against it and defends that raw steel is good 
> enough (with FEC cushions on the seats !)
> and the other extreme opinion that 2 meters thick balloon 
> tires are what is required for good confort.
> 
> Could the truth be somewhere in the middle ?-)
> 
> 
> Regards,
> 
> 
> Philippe Gentric
> Chief Architect
> Philips Software
> philippe.gentric@philips.com
> http://www.software.philips.com
> 
> 
>                                                               
>                                                               
>                
>                                                               
>                                                               
>                
>                                                        To:   
> "'Phelan, Tom'" <tphelan@sonusnet.com>                        
>                 
>                                                        cc:   
> "'avt@ietf.org'" <avt@ietf.org>                               
>                 
>                                                         
> "'dccp@ietf.org'" <dccp@ietf.org>                             
>                      
>                                                         (bcc: 
> Philippe Gentric/MP4-SUR/CE/PHILIPS)                          
>                
>                "Lars-Erik Jonsson (LU/EAB)"            
> Subject:    RE: [dccp] RE: [AVT] RE: [MMUSIC] DCCP Media 
> Guide                      
>                <lars-erik.jonsson@ericsson.                   
>                                                               
>                
>                com>                                    
> Classification:                                               
>                       
>                                                               
>                                                               
>                
>                Sent by:                                       
>                                                               
>                
>                avt-admin@ietf.org                             
>                                                               
>                
>                                                               
>                                                               
>                
>                25/11/2003 12:52                               
>                                                               
>                
>                                                               
>                                                               
>                
>                                                               
>                                                               
>                
> 
> 
> 
> 
> Tom,
> 
> We do not just have to find ways to get some kind of "fairness"
> between applications, but we actually have two main goals:
> 1) Avoid congestion
> 2) Improve network utilization
> 
> By probing the network, TCP strive for 2), and that is a good
> think. When there is available bandwidth, it is a good thing
> that this resource gets used for actual application data.
> However, sending data just to keep network resources degrades
> goal 2). To me, this seems to be a tricky problem as TCP has
> not at all been designed to interact with real-time streams.
> 
> /L-E
> 
> > -----Original Message-----
> > From: dccp-admin@ietf.org [mailto:dccp-admin@ietf.org]On Behalf Of
> > Phelan, Tom
> > Sent: den 20 november 2003 19:04
> > To: 'philippe.gentric@philips.com'; Phelan, Tom; 'Rosen, Brian'
> > Cc: 'avt@ietf.org'; 'dccp@ietf.org'; 'mmusic@ietf.org'
> > Subject: [dccp] RE: [AVT] RE: [MMUSIC] DCCP Media Guide
> >
> >
> > Hi Phillipe,
> >
> > The situation I'm worried about is a little different than
> > you describe. Say you have a 10M bps link, over which there's
> > a 4M bps video stream and a long-lived TCP app.  The TCP app
> > will take the remaining 6M bps, and then push for more,
> > which might cause the video stream to lose packets.  The
> > video stream then drops to 2M bps, and the TCP app expands to
> > 8M bps.  This hardly seems fair.
> >
> > The video app can protect itself from this by transmitting
> > enough more than 4M bps to buffer itself against the TCP
> > probes.  In this case we'd come to a relatively steady state
> > where both apps are sending roughly 5M bps.  This seems
> > reasonably fair to me, since the TCP app gets what it would
> > get if it were competing with other TCP apps, and the video
> > app gets to send at its full rate.
> >
> > Tom P.
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
> 
> 
> 
> 
> 
> 
> _______________________________________________
> dccp IETF mailing list: dccp@ietf.org
> list info:  https://www1.ietf.org/mailman/listinfo/dccp
> wg charter: http://www.ietf.org/html.charters/dccp-charter.html
> 

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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        "Rosen, Brian" <Brian.Rosen@marconi.com>
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Replies to a couple of related messages, edited:

On Tue, 25 Nov 2003, Greg Herlein wrote:

> > Unfortunately filling the client buffer at a rate faster than
> > the media rate and "when you can" which introduces substancial
> > amounts of time (many seconds) of delay is a technique for
> > media distribution for which DCCP is not needed:
> >
> > It is called "progressive" download, it uses HTTP and it is deployed ...
>
> That's somewhat simplistic, really.  Progressive download has a
> similar effect but it's not controlled, and it's TCP only of
> course.

Yes.  More on the TCP point below.

> For unicast scenarios this kind of thing makes total sense if the
> source is non-real time (ie, streaming from disk).  It's not much
> more than an enhanced jitter buffer, really - you are just adding
> some extra buffer for times when the normal real-time flow is
> impaired, and a mechanism to allow the client to tell the source
> to "fill me up"  and "back it off."  There need not be any extra
> delay at startup, though not doing so creates a window of
> vulnerability.

Exactly right.  Thanks for saving me having to write this.

The point about no additional delay is important: the vulnerability at
startup is no worse than and in fact increasingly less than the
vulnerability for a stream sent constantly at rate 1.

> I'd very much like to see some standardization effort on a
> protocol to support this kind of thing - perhaps an RTCP
> extension?  Perhaps soemthing that can apply to TCP as well?

We talked about doing something like this in RTP itself a few times,
but deferred to congestion control work in other groups.  So DCCP is
now the proposed standardization effort.


On Tue, 25 Nov 2003, Rosen, Brian wrote:

> This would seem to only apply to a non-interactive stream, where
> several seconds of buffering is acceptable, correct?

Yes.

> You also, in this case, would not actually change the encoding
> rate, but merely change the instantaneous transmit rate.

Yes.  As I said earlier, you might use FEC to protect against the
probing initially here, same as you could use it to protect an RTP/UDP
stream, but once some buffer is built up for the non-interactive case,
then retransmission is more efficient.

> I'm not clear on how this is better than just using TCP.

The key difference is that TCP insists on 100% reliability.  It must
retransmit a lost packet before any subsequent data can be delivered.
For these media applications, retransmission may still be useful, but
you want the application to have control over the decision to
retransmit or to give up and continue on.

                                                        -- Steve

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Replies to a couple of related messages, edited:

On Tue, 25 Nov 2003, Greg Herlein wrote:

> > Unfortunately filling the client buffer at a rate faster than
> > the media rate and "when you can" which introduces substancial
> > amounts of time (many seconds) of delay is a technique for
> > media distribution for which DCCP is not needed:
> >
> > It is called "progressive" download, it uses HTTP and it is deployed ...
>
> That's somewhat simplistic, really.  Progressive download has a
> similar effect but it's not controlled, and it's TCP only of
> course.

Yes.  More on the TCP point below.

> For unicast scenarios this kind of thing makes total sense if the
> source is non-real time (ie, streaming from disk).  It's not much
> more than an enhanced jitter buffer, really - you are just adding
> some extra buffer for times when the normal real-time flow is
> impaired, and a mechanism to allow the client to tell the source
> to "fill me up"  and "back it off."  There need not be any extra
> delay at startup, though not doing so creates a window of
> vulnerability.

Exactly right.  Thanks for saving me having to write this.

The point about no additional delay is important: the vulnerability at
startup is no worse than and in fact increasingly less than the
vulnerability for a stream sent constantly at rate 1.

> I'd very much like to see some standardization effort on a
> protocol to support this kind of thing - perhaps an RTCP
> extension?  Perhaps soemthing that can apply to TCP as well?

We talked about doing something like this in RTP itself a few times,
but deferred to congestion control work in other groups.  So DCCP is
now the proposed standardization effort.


On Tue, 25 Nov 2003, Rosen, Brian wrote:

> This would seem to only apply to a non-interactive stream, where
> several seconds of buffering is acceptable, correct?

Yes.

> You also, in this case, would not actually change the encoding
> rate, but merely change the instantaneous transmit rate.

Yes.  As I said earlier, you might use FEC to protect against the
probing initially here, same as you could use it to protect an RTP/UDP
stream, but once some buffer is built up for the non-interactive case,
then retransmission is more efficient.

> I'm not clear on how this is better than just using TCP.

The key difference is that TCP insists on 100% reliability.  It must
retransmit a lost packet before any subsequent data can be delivered.
For these media applications, retransmission may still be useful, but
you want the application to have control over the decision to
retransmit or to give up and continue on.

                                                        -- Steve

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Stephen,

you wrote:

>> > Unfortunately filling the client buffer at a rate faster than
>> > the media rate and "when you can" which introduces substancial
>> > amounts of time (many seconds) of delay is a technique for
>> > media distribution for which DCCP is not needed:
>> >
>> > It is called "progressive" download, it uses HTTP and it is deployed ...
>>
>> That's somewhat simplistic, really.  Progressive download has a
>> similar effect but it's not controlled, and it's TCP only of
>> course.
>
>Yes.
>

I am concerned that since the barrier for DCCP is very high
(implementing a new protocol at IP level is not something the industry
will go for without a strong incentive) the added value must be equally high.

My point was that Progressive download (although different of course ;-)
has pretty similar performances compared to the solution/application you describe,
i.e. media delivery with large buffers/delays....

But progressive download has *zero* hurdle(e.g. NAT/FW traversal ;-)...
so if we conclude that DCCP is good only for that application
there is the clear risk that this will not help promote DCCP...

>The key difference is that TCP insists on 100% reliability.  It must
>retransmit a lost packet before any subsequent data can be delivered.
>For these media applications, retransmission may still be useful, but
>you want the application to have control over the decision to
>retransmit or to give up and continue on.

I fully agree this is the fundamental reason why another protocol than TCP
is a good idea: media does not need 100% reliability.

The other idea I have been trying to advocate for a while because I believe it is another key to the problem
is that media does not need to be constant bit rate, nor constant packet rate, nor constant packet size.

These 3 things (delay, reliability, variable rate/packaging) should give some slack
for the design of a solution with added value no?

regards,


Philippe Gentric
Chief Architect
Philips Software
philippe.gentric@philips.com
http://www.software.philips.com



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
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From: "Phelan, Tom" <tphelan@sonusnet.com>
To: "'philippe.gentric@philips.com'" <philippe.gentric@philips.com>
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Hi Phillipe,

[snipped]
> The other idea I have been trying to advocate for a while 
> because I believe it is another key to the problem
> is that media does not need to be constant bit rate, nor 
> constant packet rate, nor constant packet size.

I'm glad you brought this up.  There have been a lot of comments on this
thread about the difficulties in the human perception of rate-shifting, and,
in general, I'd have to say the comments are pretty accurate, especially
given the current state of the art in codec design.  However, I can think of
at least two instances where rate shifting is either workable or better than
the current practice.

One case involves audio codecs such as G.711 and changing the packetization
interval.  G.711 with 10ms packetization is about 96K IP bps on the wire.
40ms packetization gives about 72K bps.  I think that changing between these
two rates is often close to imperceptible.  A similar thing can be done with
G.729A (although I'm not sure the standard allows it), giving more variation
in the bit rate - 10ms packetization is about 38.4K bps and 40ms is 14.4K.

Another case involves video transmission.  As you change between a high and
low encoding rate the image the user sees is likely to change back and forth
between clear and grainy, and this is likely to be annoying.  But I imagine
it's less annoying than the current practice of freezing the video (and
continuing the audio) when there's insufficient bandwidth for the high-res
video.  I imagine that layered encoding techniques like Gerard Babonneua
mentioned can fit this mode of operation nicely.

Given that there are these low-hanging-fruit stream shifting methods, I
wonder what others are available if we put our minds to it?

Tom Phelan

> These 3 things (delay, reliability, variable rate/packaging) 
> should give some slack
> for the design of a solution with added value no?
> 
> regards,
> 
> 
> Philippe Gentric
> Chief Architect
> Philips Software
> philippe.gentric@philips.com
> http://www.software.philips.com
> 
> 
> 
> _______________________________________________
> dccp IETF mailing list: dccp@ietf.org
> list info:  https://www1.ietf.org/mailman/listinfo/dccp
> wg charter: http://www.ietf.org/html.charters/dccp-charter.html
> 

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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> I fully agree this is the fundamental reason why another protocol than TCP
> is a good idea: media does not need 100% reliability.

The only reason not to get 100% reliability is that relaxing reliability 
constraints reduces delay, as even modest loss rates, if unfortuitously 
placed, can wreak havoc with user experience. For the higher-latency 
streaming application contemplated here, it is unclear that removing 
reliability adds any significant value, particularly given that you end 
up having to add some protection (FEC) to deal with losses, thus 
increasing the overall average bit rate or reducing the quality. 
(Obviously, FEC doesn't deal well with loss spurts, so quality will be 
rather poor if losses are indeed bursty, even with aggressive FEC.) 
Typical FEC overhead for useful error resiliency is usually in excess of 
20%, so the bandwidth penalty is substantial.

Thus, before accepting that DCCP is the right thing for streaming media, 
a bit more human-factors and measurement evidence seems called for.

_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Wed Nov 26 09:19:20 2003
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From: Greg Herlein <gregh@arkstream.com>
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> I fully agree this is the fundamental reason why another
> protocol than TCP is a good idea: media does not need 100%
> reliability.

I don't agree.  *Some* media does not need 100% reliability.  
But many video codecs have pathetic performance in the face of
packet loss - and FEC is not always a good solution (impact on
bandwidth is one reason - complexity is another).  Audio is often 
a lot more tolerant (at least in VoIP) - a loss spurt in video is 
radically painful for the viewer in most cases.

While TCP does have it's own problems for video delivery (and
some advantages too) it's not the ideal solution, clearly.

I most wholeheartedly agree that delay, reliability, variable
rate/packaging are the three elements that *can* be adjusted to 
modify the performance of a video stream.  To that I would add 
dynamic buffering at the endpoints (goes with rate, I guess).  
I'd like to see a protocol (or extension) that can implement this 
into apps that can use it to good measure, as opposed to 
generating YAP (yet another protocol) that addresses only one 
aspect of the problem (and in only one way).

Greg Herlein
VP Engineering
Arkstream, Inc.



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Wed Nov 26 09:27:20 2003
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Phelan, Tom'" <tphelan@sonusnet.com>,
        "'philippe.gentric@philips.com'" <philippe.gentric@philips.com>
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Subject: [AVT] RE: [dccp] Re: DCCP Media Guide
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Again, I think the most acceptable response to congestion for 
video is to rate shift down once, or if necessary, twice to
get to an "acceptable" level of errors.  So, it's not freeze 
frame vs rapid rate shift, it's rate shift down vs rapid rate shift.

We have a different error result than freeze, and it seems to work
better (less objectionable), but it doesn't change the equation.
If you get too many errors, you lower the rate until you don't
get "too many".

It would be interesting to look at the voice response to changing
packetization period.  Clearly, the effect of the loss of one large
packet is worse than the effect of a small packet.  Small packets have
more bit rate, so, you would presumably be sending larger packets
when you have congestion.  What does that do to the perceived quality?
It's going to get worse as congestion gets worse, which in some sense
is "good", but is it actually worse than a higher packet loss rate,
but less problem for each loss?  Probably depends on the loss pattern.
If it were random, than the smaller problem, larger percentage is
probably better, but it's not random.

There are people who know a lot about such things.  I'm not one of them.
Do we have some of that expertise around on the list?  Has that idea
been tried?

I do think we need to split this thread, or at least mark replies with
"Applies to interactive real time" vs "applies to non-interactive real
time".
I appreciate several comments on the thread that DCCP may have more
applicability to non-interactive streams.  While many of the effects
are similar (rate shift to non-interactive is just as objectionable
to non-interactive as it is to interactive), some things are clearly
different (slow start, for example).

Brian


> -----Original Message-----
> From: Phelan, Tom [mailto:tphelan@sonusnet.com]
> Sent: Wednesday, November 26, 2003 8:25 AM
> To: 'philippe.gentric@philips.com'
> Cc: avt@ietf.org; dccp@ietf.org
> Subject: RE: [dccp] Re: DCCP Media Guide
> 
> 
> Hi Phillipe,
> 
> [snipped]
> > The other idea I have been trying to advocate for a while 
> > because I believe it is another key to the problem
> > is that media does not need to be constant bit rate, nor 
> > constant packet rate, nor constant packet size.
> 
> I'm glad you brought this up.  There have been a lot of 
> comments on this
> thread about the difficulties in the human perception of 
> rate-shifting, and,
> in general, I'd have to say the comments are pretty accurate, 
> especially
> given the current state of the art in codec design.  However, 
> I can think of
> at least two instances where rate shifting is either workable 
> or better than
> the current practice.
> 
> One case involves audio codecs such as G.711 and changing the 
> packetization
> interval.  G.711 with 10ms packetization is about 96K IP bps 
> on the wire.
> 40ms packetization gives about 72K bps.  I think that 
> changing between these
> two rates is often close to imperceptible.  A similar thing 
> can be done with
> G.729A (although I'm not sure the standard allows it), giving 
> more variation
> in the bit rate - 10ms packetization is about 38.4K bps and 
> 40ms is 14.4K.
> 
> Another case involves video transmission.  As you change 
> between a high and
> low encoding rate the image the user sees is likely to change 
> back and forth
> between clear and grainy, and this is likely to be annoying.  
> But I imagine
> it's less annoying than the current practice of freezing the 
> video (and
> continuing the audio) when there's insufficient bandwidth for 
> the high-res
> video.  I imagine that layered encoding techniques like 
> Gerard Babonneua
> mentioned can fit this mode of operation nicely.
> 
> Given that there are these low-hanging-fruit stream shifting 
> methods, I
> wonder what others are available if we put our minds to it?
> 
> Tom Phelan
> 
> > These 3 things (delay, reliability, variable rate/packaging) 
> > should give some slack
> > for the design of a solution with added value no?
> > 
> > regards,
> > 
> > 
> > Philippe Gentric
> > Chief Architect
> > Philips Software
> > philippe.gentric@philips.com
> > http://www.software.philips.com
> > 
> > 
> > 
> > _______________________________________________
> > dccp IETF mailing list: dccp@ietf.org
> > list info:  https://www1.ietf.org/mailman/listinfo/dccp
> > wg charter: http://www.ietf.org/html.charters/dccp-charter.html
> > 
> 
> _______________________________________________
> dccp IETF mailing list: dccp@ietf.org
> list info:  https://www1.ietf.org/mailman/listinfo/dccp
> wg charter: http://www.ietf.org/html.charters/dccp-charter.html
> 

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
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From: "Vinay Sathe" <vsathe@san.rr.com>
To: "'Phelan, Tom'" <tphelan@sonusnet.com>, <philippe.gentric@philips.com>
Cc: <avt@ietf.org>, <dccp@ietf.org>
Subject: RE: [AVT] RE: [dccp] Re: DCCP Media Guide
Date: Wed, 26 Nov 2003 06:31:22 -0800
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This comment is not really related to DCCP. However, the issue of rate
variability of video has been brought up during the discussions. I concur
with the comments regarding video rates below. While it may be true that
"traditional" video codec (MPEG-1 MPEG-2) may have undesirable artifacts due
to transport rate changes (that were unknown to the rate control loop when
the video was encoded), several standards-based and proprietary video codec
exist that minimize visual degradation through severe video rate
changes/dropouts. (M-JPEG2000 is one example).

Best regards,
Vinay 

-----Original Message-----
From: avt-admin@ietf.org [mailto:avt-admin@ietf.org] On Behalf Of Phelan,
Tom
Sent: Wednesday, November 26, 2003 5:25 AM
To: 'philippe.gentric@philips.com'
Cc: avt@ietf.org; dccp@ietf.org
Subject: [AVT] RE: [dccp] Re: DCCP Media Guide

Hi Phillipe,

[snipped]
> The other idea I have been trying to advocate for a while 
> because I believe it is another key to the problem
> is that media does not need to be constant bit rate, nor 
> constant packet rate, nor constant packet size.

I'm glad you brought this up.  There have been a lot of comments on this
thread about the difficulties in the human perception of rate-shifting, and,
in general, I'd have to say the comments are pretty accurate, especially
given the current state of the art in codec design.  However, I can think of
at least two instances where rate shifting is either workable or better than
the current practice.

One case involves audio codecs such as G.711 and changing the packetization
interval.  G.711 with 10ms packetization is about 96K IP bps on the wire.
40ms packetization gives about 72K bps.  I think that changing between these
two rates is often close to imperceptible.  A similar thing can be done with
G.729A (although I'm not sure the standard allows it), giving more variation
in the bit rate - 10ms packetization is about 38.4K bps and 40ms is 14.4K.

Another case involves video transmission.  As you change between a high and
low encoding rate the image the user sees is likely to change back and forth
between clear and grainy, and this is likely to be annoying.  But I imagine
it's less annoying than the current practice of freezing the video (and
continuing the audio) when there's insufficient bandwidth for the high-res
video.  I imagine that layered encoding techniques like Gerard Babonneua
mentioned can fit this mode of operation nicely.

Given that there are these low-hanging-fruit stream shifting methods, I
wonder what others are available if we put our minds to it?

Tom Phelan

> These 3 things (delay, reliability, variable rate/packaging) 
> should give some slack
> for the design of a solution with added value no?
> 
> regards,
> 
> 
> Philippe Gentric
> Chief Architect
> Philips Software
> philippe.gentric@philips.com
> http://www.software.philips.com
> 
> 
> 
> _______________________________________________
> dccp IETF mailing list: dccp@ietf.org
> list info:  https://www1.ietf.org/mailman/listinfo/dccp
> wg charter: http://www.ietf.org/html.charters/dccp-charter.html
> 

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: RE: [dccp] RE: [AVT] RE: [MMUSIC] DCCP Media Guide
To: "Rosen, Brian" <Brian.Rosen@marconi.com>
Cc: "'avt@ietf.org'" <avt@ietf.org>, "'dccp@ietf.org'" <dccp@ietf.org>
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Brian, you wrote

>This would seem to only apply to a non-interactive stream, where
>several seconds of buffering is acceptable, correct?

When you play with the transmit rate the larger the buffer/delay the easier ... .

And if you dont have delay then you really need to be able to change the media rate !

>You also, in this case, would not actually change the encoding
>rate, but merely change the instantaneous transmit rate.

in all cases when the temporary bandwidth reduction lasts longer than your buffer
...then you need to change the media rate, there is just no other way.

Actually for interactive (conferencing) case changing the rate
of the codec (media rate) is actually "easier" (in theory):
you just tell the codec(s) to produce less bits (!)
(it is "easier" in the sense that with pre-encoded media delivery you have to do stream switching instead)

>I'm not clear on how this is better than just using TCP.

If you only consider the case where the buffers/delays are really large
then yes, I tend to agree, (see other mail about progressive download)...


regards,


Philippe Gentric
Chief Architect
Philips Software
philippe.gentric@philips.com
http://www.software.philips.com



_______________________________________________
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Date: Wed, 26 Nov 2003 09:09:44 -0500
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Subject: Re: [AVT] RE: [dccp] Re: DCCP Media Guide
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> One case involves audio codecs such as G.711 and changing the packetization
> interval.  G.711 with 10ms packetization is about 96K IP bps on the wire.
> 40ms packetization gives about 72K bps.  I think that changing between these
> two rates is often close to imperceptible.  A similar thing can be done with
> G.729A (although I'm not sure the standard allows it), giving more variation
> in the bit rate - 10ms packetization is about 38.4K bps and 40ms is 14.4K.

While this is numerically accurate, it seems largely pointless. If I'm 
below the perceptible delay threshold (say 150 ms one-way), there is no 
reason to transmit at 10 ms packetization interval since the user 
experience will not improve noticeably. Indeed, very few systems 
transmit at 10 ms packetization today. In addition, at the access links 
where congestion is most likely, IP/UDP/RTP header compression reduces 
the difference in bit rates to much smaller values, further reducing the 
utility of doing variable packetization. (From our experiments, a 
packetization interval of roughly 30-40 ms also appears to work best 
with FEC.)

> 
> Another case involves video transmission.  As you change between a high and
> low encoding rate the image the user sees is likely to change back and forth
> between clear and grainy, and this is likely to be annoying.  But I imagine
> it's less annoying than the current practice of freezing the video (and
> continuing the audio) when there's insufficient bandwidth for the high-res
> video.  I imagine that layered encoding techniques like Gerard Babonneua
> mentioned can fit this mode of operation nicely.

Again, like Brian pointed out, this assumes that you're aiming for a 
low-quality hey-it's-free experience.

> 
> Given that there are these low-hanging-fruit stream shifting methods, I
> wonder what others are available if we put our minds to it?

Unfortunately, the first is unlikely to be productive and the second has 
very limited applicability.


_______________________________________________
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avt@ietf.org
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> It would be interesting to look at the voice response to changing
> packetization period.  Clearly, the effect of the loss of one large
> packet is worse than the effect of a small packet.  Small packets have
> more bit rate, so, you would presumably be sending larger packets
> when you have congestion.  What does that do to the perceived quality?
> It's going to get worse as congestion gets worse, which in some sense
> is "good", but is it actually worse than a higher packet loss rate,
> but less problem for each loss?  Probably depends on the loss pattern.
> If it were random, than the smaller problem, larger percentage is
> probably better, but it's not random.

We have looked at this systematically, varying the packetization 
threshold. See the papers by Wenyu Jiang and myself at 
http://netbib.info and http://www1.cs.columbia.edu/~wenyu/


> 
> There are people who know a lot about such things.  I'm not one of them.
> Do we have some of that expertise around on the list?  Has that idea
> been tried?

See above and references therein.



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Reply-To: <qli@brainmediaco.com>
From: "Qiong Li" <qli@brainmediaco.com>
To: "'Rosen, Brian'" <Brian.Rosen@marconi.com>, <avt@ietf.org>
Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide
Date: Wed, 26 Nov 2003 10:18:04 -0500
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Hi Everyone,

Here is some of my thought related to this topic.

It may be arguable whether layered audio and video are needed for wired
sections of the Internet, However, it seems that scalable coded media
contents may still be necessary for wireless delivery, especially in
mobile environments, or in cases where the end-to-end paths have
wireless link segments. The bandwidth variations on wireless links could
be due to completely different reasons from congestion, such as fading,
interference, mobility, etc.

It seems that when wireless link degradation happens, there are no known
QoS mechanisms that can still provide previously negotiated services. To
deal with this problem, many research works proposed to deploy so-called
soft QoS in which service quality can vary in a range, or to encourage
application repeatedly renegotiate a different QoS level with the
network. Once any of these happens, adaptation on content bit rate seems
unavoidable.

Also from a user perspective (I don't have experiments to support this),
assuming the case where a person who listens to music (may be companied
with low bit rate video). He may from time to time experience the
degradation of sound and picture qualities whenever he walks under a
construction or enter a new cellar domain, etc. this degradation would
be understandable and acceptable to the user. However, if the sound
completely stopped and the picture frozen, he would have no clue whether
this problem is because he just got a broken phone or else.

Therefore, if the digitized media can be delivered as resilient as
analog format, that may be the direction we want to pursue.

Of course, layered media contents are not necessary to be delivered by
DCCP-like protocols.

Thanks,
-qiong
 
-----Original Message-----
From: avt-admin@ietf.org [mailto:avt-admin@ietf.org] On Behalf Of Rosen,
Brian
Sent: Monday, November 24, 2003 9:49 AM
To: 'Magnus Westerlund'
Cc: Phelan, Tom; 'avt@ietf.org'
Subject: RE: [AVT] RE: [MMUSIC] DCCP Media Guide

Magnus

I don't think the people working on the stability of the network
understand what they are proposing, or they wouldn't be proposing it,
except for Tom, who probably understands what the effect of 2X
bandwidth (when congestion was not occurring), and figures if that's
the only knob he's got, then he turns it far enough to make his app
work,
and since he doesn't violate any rules, all is well.

I don't agree.  I think that, as Marshall observes, RTP media flows
tend to be self policing, and in any event, following the IAB
recommendation of dropping the flow if excessive packet loss is observed
will have the same effect - congestion collapse will not occur.
I think that we should stop thinking that we need anything else to
prevent congestion collapse.  

I think the real question this thread is addressing is whether DCCP
is at all suitable for real time flows.  I think the answer is no.
I think we should try see if we can reach consensus on this.
Given what has been said on this thread, is there anyone left who
thinks DCCP is suitable for RTP flows?  

This leaves us in a quandary, which, I think, is what Tom was trying to
avoid.  If DCCP is not suitable, is there any way to hold off the greedy
TCP?  This is the question Mike raises.  It's not an issue of congestion
collapse, it is the issue of "fairness".

My own feeling is that fairness to RTP is let the flow go through or
don't.
There exists a variety of ways to vary the rate, but none of them are
useful.
Have you ever actually listened (or watched) a media stream with varying
rate?  I have, several times, several different algorithms.  It sucks,
to
put it mildly.  Unless you have some evidence that there exists an
algorithm which does not annoy a user when the rate changes, I suspect 
our experience will be the common one - you will go down, once, and 
never go back up.  I expect even this will be very rare.   Much more
common
is that you will try one rate, and if you don't get good results, the
call/stream will be dropped and tried again.  There are lots of
algorithms
to vary rates.  There are no ways I know to vary quality, how ever it is
done,
and not be very distracting to a user.  Unless there is some kind of
concealment mechanism I am not aware of, it just won't be used,
regardless
of how clever the algorithm is, and how bandwidth efficient it is.

So, to me, the idea that RTP reacts to whatever else is happening in any
way other than dropping the entire stream is not credible.  That means
there is no real rate control for RTP, TCP friendly or otherwise.
(I'll just use RTP vs TCP, even though we all understand that RTP is
carried on some other transport at the same level as TCP).

So what does it mean to be "fair"?
	For RTP, being fair to TCP (or other best effort traffic) is to
		not get started, or get dropped if congestion is
excessive
	For TCP, being fair to RTP (or other constant rate traffic) is
		to not be greedy.

This is why QoS mechanisms work.  If you put RTP ahead of TCP,
and use admission control to limit the amount of RTP, everything works.
If you don't have a QoS mechanism, you can still perform admission
control to the RTP flows to enforce RTP fairness on TCP, but you need
to put some kind of mechanism to allow TCP to be fair to RTP.  I'm happy
to work on mechanisms that would afford effective admission control
in reaction to best effort traffic, but only if there is a commitment
to fix TCP.

Brian



> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Friday, November 21, 2003 10:40 AM
> To: Rosen, Brian
> Cc: Phelan, Tom; 'avt@ietf.org'
> Subject: Re: [AVT] RE: [MMUSIC] DCCP Media Guide
> 
> 
> Hi Brian,
> 
> I don't think that the people working with stability of the 
> network are 
> sharing you view due to the implications that your claims results in.
> 
> First, if a multi-media application transmitting data over a 
> best-effort 
> IP network, i.e. without any QoS guarantees across the whole net that 
> its traffic will fit, it can definitely cause congestion 
> collapse if the 
> mix becomes most, media, little TCP. Please read 
> draft-iab-congestion-01.txt for some discussion on this.
> 
> Some comments to your assertions.
> 
> Rosen, Brian wrote:
> > Phillipe
> > 
> > You have made invalid assumptions:
> > 	1. That audio/video media data rates will be changed dynamically
> > 		as network traffic changes.  This is very 
> unlikely.  It may
> > 		be that you get so much loss that you drop to a 
> lower data
> > 		rate, although I suspect that will be rare.  
> When you do,
> > 		you will never revert to the higher rate.  This 
> is because
> > 		humans hate to see the change, and minimizing 
> changes is 
> > 		more desirable than keeping average quality higher.
> 
> First you agree that it is possible to change the rate. Secondly I 
> understand that the there are implications on switching rate, i.e. 
> quality to often, however there are reasons for switching up when the 
> quality gain of a switch if being possible to keep the new 
> rate is clear.
> 
> > 
> > 	2. Real Time apps have no interest in being "fair".  
> They are all
> > 		or nothing. If you admit them to the network, you expect
> > 		the streams to be maintained.  If you can't 
> maintain them,
> > 		don't start them.  They don't want to be fair 
> to TCP, and
> > they
> > 		don't want to be fair to other RTP.  They just 
> want to send
> > 		data at the rate they were advised to do by signalling.
> > 		They can tolerate some loss.  Misordering=loss. 
>  Duplicates=
> > 		wasted bandwith.  That's a whole lot different from TCP
> > 		traffic.  
> > 
> 
> Yes, I agree that the application has no interest in being 
> fair. However 
> using best effort networks requires one to being reasonably 
> fair. If one 
> are not, the combination of several of media application can totally 
> destroy things for them self.
> 
> > I can imagine working on protocols which allowed CBR and 
> "best effort"
> > traffic to better coexist, but I think it's only possible 
> if the best
> > effort traffic recognized the presence of the CBR traffic 
> and reacted
> > appropriately.  Otherwise, you need to use QoS mechanisms 
> to separate
> > the traffic.
> 
> What says that the media traffic is more important that the traffic 
> going over TCP?
> 
> 
> Cheers
> 
> Magnus Westerlund
> 
> Multimedia Technologies, Ericsson Research EAB/TVA/A
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> 

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: Re: [AVT] RE: [dccp] Re: DCCP Media Guide
To: Henning Schulzrinne <hgs@cs.columbia.edu>
Cc: avt@ietf.org, dccp@ietf.org
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Henning, TFRC experts,


>> It would be interesting to look at the voice response to changing
>> packetization period.

...

>
>We have looked at this systematically, varying the packetization
>threshold. See the papers by Wenyu Jiang and myself at
>http://netbib.info and http://www1.cs.columbia.edu/~wenyu/
>

Does anyone have a clue of how changing the packetization
impacts the behavior of TFRC in presence of TCP traffic ?

For example if an application increases its packet rate
(same media rate, higher transmit rate due to packet headers contribution increasing)
is that added "pressure" against competing TCP traffic ?


regards,


Philippe Gentric
Chief Architect
Philips Software
philippe.gentric@philips.com
http://www.software.philips.com



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Phelan, Tom" <tphelan@sonusnet.com>
To: "'philippe.gentric@philips.com'" <philippe.gentric@philips.com>
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Subject: RE: [AVT] RE: [dccp] Re: DCCP Media Guide
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Hi Phillipe,

TFRC is packet rate based, not bit rate.  If TFRC cut your allowed packet
rate in half, it'd be OK (at least as far as TFRC is concerned) if you cut
your packet rate in half, but doubled the packet size.  Of course in many
circumstances this action would be unlikely to alleviate the network
congestion, and some of these new big packets would probably be lost,
causing TFRC to cut the allowed rate even more.

TCP, on the other hand, is bit (or byte) rate based.  Network congestion is
both bit and packet rate based (others things too, I suppose), although I
think the packet rate problems are diminishing (Moore's law helping us
there).

If going to smaller packets gets you a higher bits-on-the-wire rate, then
you'd get some of the buffering I was talking about.  On the other hand, if
the chokepoint was packet-limited rather than bit-limited ...

Tom Phelan

> -----Original Message-----
> From: philippe.gentric@philips.com 
> [mailto:philippe.gentric@philips.com]
> Sent: Wednesday, November 26, 2003 11:31 AM
> To: Henning Schulzrinne
> Cc: avt@ietf.org; dccp@ietf.org
> Subject: Re: [AVT] RE: [dccp] Re: DCCP Media Guide
> 
> 
> 
> Henning, TFRC experts,
> 
> 
> >> It would be interesting to look at the voice response to changing
> >> packetization period.
> 
> ...
> 
> >
> >We have looked at this systematically, varying the packetization
> >threshold. See the papers by Wenyu Jiang and myself at
> >http://netbib.info and http://www1.cs.columbia.edu/~wenyu/
> >
> 
> Does anyone have a clue of how changing the packetization
> impacts the behavior of TFRC in presence of TCP traffic ?
> 
> For example if an application increases its packet rate
> (same media rate, higher transmit rate due to packet headers 
> contribution increasing)
> is that added "pressure" against competing TCP traffic ?
> 
> 
> regards,
> 
> 
> Philippe Gentric
> Chief Architect
> Philips Software
> philippe.gentric@philips.com
> http://www.software.philips.com
> 
> 
> 
> _______________________________________________
> dccp IETF mailing list: dccp@ietf.org
> list info:  https://www1.ietf.org/mailman/listinfo/dccp
> wg charter: http://www.ietf.org/html.charters/dccp-charter.html
> 

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Stephen Casner <casner@acm.org>
To: "Rosen, Brian" <Brian.Rosen@marconi.com>
cc: AVT WG <avt@ietf.org>, dccp@ietf.org
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On Wed, 26 Nov 2003, Rosen, Brian wrote:

> Again, I think the most acceptable response to congestion for
> video is to rate shift down once, or if necessary, twice to
> get to an "acceptable" level of errors.  So, it's not freeze
> frame vs rapid rate shift, it's rate shift down vs rapid rate shift.
>
> We have a different error result than freeze, and it seems to work
> better (less objectionable), but it doesn't change the equation.
> If you get too many errors, you lower the rate until you don't
> get "too many".

One quick point, relating back to the start of the thread:  If you are
competing with one or more TCP streams and the total traffic is
filling the available bottleneck bandwidth, then if you back off the
video rate as a result of the loss induced by TCP rate probing, the
TCP streams will expand to fill the bandwidth you freed and then probe
for more until you stop.

                                                        -- Steve

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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To: "Phelan, Tom" <tphelan@sonusnet.com>
Cc: avt@ietf.org, dccp@ietf.org
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Tom, you wrote:

>There have been a lot of comments on this
>thread about the difficulties in the human perception of rate-shifting, and,
>in general, I'd have to say the comments are pretty accurate, especially
>given the current state of the art in codec design.

Recent/hi-end digital TV (mpeg-2) codecs change their rate *all the time*.
They do that  to assign different bit rates to different channels based on the program/time
(you put more bit rate on channel 12 when the movie or baseball match starts)
and/or based on statistical multiplexing (i.e. the rate change several times per picture).
This is deployed all over the face of the planet.
Of course if you change from 4 Mb/s to 1 Mb/s the users may see the difference,
but less drastic changes are hardly noticed but by experts (did you notice it ?) ...

My group has products that dynamically change the rate of (MPEG-4) video
(even down to very low bit rates e.g. the GPRS range at 20 to 50 kb/s)
We do the same for speech in the 5-24 kb/s range.
(BTW: AMR stands for Adaptive Multi Rate !)

So I am really surprised with all this negative feedback about codecs
and this notion that it is impossible/bad to change rates ?

regards,


Philippe Gentric
Chief Architect
Philips Software
philippe.gentric@philips.com
http://www.software.philips.com



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: Re: [AVT] RE: [dccp] Re: DCCP Media Guide
To: Stephen Casner <casner@acm.org>
Cc: AVT WG <avt@ietf.org>, dccp@ietf.org
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Stephen, you wrote:

>One quick point, relating back to the start of the thread:  If you are
>competing with one or more TCP streams and the total traffic is
>filling the available bottleneck bandwidth, then if you back off the
>video rate as a result of the loss induced by TCP rate probing, the
>TCP streams will expand to fill the bandwidth you freed and then probe
>for more until you stop.

right, so it proves the need for "probing" (in the limit allowed by TFRC of course ;-)
i.e. the media stream must "put some pressure" on the TCP traffic.

Now when you try to find out how to do that you find the following possibilities:

* increase the instantaneous transmit rate = works only within buffers/delay limits

* increase the instantaneous media rate = works always (assuming the right codecs)

regards,

Philippe Gentric
Chief Architect
Philips Software
philippe.gentric@philips.com
http://www.software.philips.com


                                                                                                                                       
                                                                                                                                       
                                                   To:   "Rosen, Brian" <Brian.Rosen@marconi.com>                                      
                                                   cc:   AVT WG <avt@ietf.org>                                                         
                                                    dccp@ietf.org                                                                      
                                                    (bcc: Philippe Gentric/MP4-SUR/CE/PHILIPS)                                         
               Stephen Casner                      Subject:    [AVT] RE: [dccp] Re: DCCP Media Guide                                   
               <casner@acm.org>                                                                                                        
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               Sent by:                                                                                                                
               avt-admin@ietf.org                                                                                                      
                                                                                                                                       
               26/11/2003 18:35                                                                                                        
                                                                                                                                       
                                                                                                                                       




On Wed, 26 Nov 2003, Rosen, Brian wrote:

> Again, I think the most acceptable response to congestion for
> video is to rate shift down once, or if necessary, twice to
> get to an "acceptable" level of errors.  So, it's not freeze
> frame vs rapid rate shift, it's rate shift down vs rapid rate shift.
>
> We have a different error result than freeze, and it seems to work
> better (less objectionable), but it doesn't change the equation.
> If you get too many errors, you lower the rate until you don't
> get "too many".

One quick point, relating back to the start of the thread:  If you are
competing with one or more TCP streams and the total traffic is
filling the available bottleneck bandwidth, then if you back off the
video rate as a result of the loss induced by TCP rate probing, the
TCP streams will expand to fill the bandwidth you freed and then probe
for more until you stop.

                                                        -- Steve

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt






_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: Re: [AVT] Re: [dccp] Re: DCCP Media Guide
To: Greg Herlein <gregh@arkstream.com>
Cc: avt@ietf.org, dccp@ietf.org
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greg you wrote:

>> I fully agree this is the fundamental reason why another
>> protocol than TCP is a good idea: media does not need 100%
>> reliability.
>
>I don't agree.  *Some* media does not need 100% reliability.
>But many video codecs have pathetic performance in the face of
>packet loss - and FEC is not always a good solution

the bad news is probably  that if you have to use such codecs then DCCP is not for you ?

BTW the more recent/advanced codecs have error resilience features designed-in for that reason:
people were horrified when they found out how the lack of resistance to even a small error rate
was generating additional cost in deployments e.g. in DTV  the additional bandwidth cost of the required FEC
plus bill-of-material cost of FEC decoders on the STB is quite substancial...

regards,


Philippe Gentric
Chief Architect
Philips Software
philippe.gentric@philips.com
http://www.software.philips.com



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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> philippe.gentric@philips.com writes:
>
> The other idea I have been trying to advocate for a while because I
> believe it i s another key to the problem is that media does not need
> to be constant bit rate, nor constant packet rate, n or constant
> packet size.

As an existence proof, RTP MIDI is potentially non-constant in all of
these ways.  And, with the appropriate choice of renderer, the RTP
MIDI stream could be just as normative a codec as AAC -- the SDP
machinery in Appendix C.5 of the RTP MIDI format I-D is designed to
support this.

This was Eric's motivation for doing Structured Audio, and my original
motivation for writing an SA compiler (although the emphasis switched
to network musical performance a few years into the project ...).

---------------------------------------------------
John Lazzaro    http://www.cs.berkeley.edu/~lazzaro
     lazzaro [at] cs [dot] berkeley [dot] edu     
---------------------------------------------------








_______________________________________________
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https://www1.ietf.org/mailman/listinfo/avt



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

CALL FOR PAPERS

The 14th International Workshop on Network and Operating Systems Support
for Digital Audio and Video (NOSSDAV 2004)
June 16-18, 2004 --- Kinsale, County Cork, Ireland 

http://www.nossdav.org/2004/


NOSSDAV 2004 will take place in Kinsale on the exceptionally scenic
southern coast of Ireland. Kinsale is approximately a 20 minute drive
outside Cork city, which is within easy reach of major European capitals
and the east coast of the US. The workshop is sponsored by ACM SIG
Multimedia (preliminary approval), Microsoft Research and Science
Foundation Ireland, and is in cooperation with ACM SIGCOMM (preliminary
approval). 

NOSSDAV invites submissions on all areas of multimedia, but strongly
encourages submissions in newly emerging specialized areas such as
networked games and peer-to-peer streaming. NOSSDAV 2004 will be run
very much as a workshop with an emphasis on cutting-edge research and
lively discussion. Papers grounded in high-quality experimental research
based on prototype or real systems, including innovative experimental
commercial designs and prototypes, as well as papers proposing new
research directions for the community or calling into question existing
conventional wisdom are particularly welcome. ***Student participation
is strongly encouraged*** - supported by several student travel grants
and registration fee rebates. 

Submissions should be no longer than 6 pages. The expectation is that
papers accepted at the workshop will eventually lead to full-length
papers at high-quality conferences or journals. The topics of interest
include, but are not limited to:
 
* Networked games 
* Peer-to-peer multimedia 
* Broadband streaming media content distribution 
* 3D multimedia and (networked) immersive environments 
* Wireless and ad-hoc network multimedia systems 
* Embedded multimedia devices (e.g., PVRs) 
* Multimedia security 
* Internet telephony 
* Sensory interactive applications 
* Digital rights management 


Deadlines:

Paper registration deadline - February 23, 2004, (5pm EST) (*firm*)
Paper submission deadline - March 1, 2004, (5pm EST) (*firm*) 


Program Co-Chairs:

Venkat Padmanabhan, Microsoft Research (USA)
Cormac Sreenan, University College Cork (Ireland)
Email: nossdav2004-pcchairs@cs.ucc.ie 


Program Committee:

Kevin Almeroth, UC Santa Barbara (USA)
Ernst Biersack, Institut Eurecom (France)
Phil Chou, Microsoft Research (USA)
Christophe Diot, Intel Research (UK)
Kevin Jeffay, UNC Chapel Hill (USA)
Srinivasan Keshav, University of Waterloo (Canada)
Sue Moon, KAIST (Korea)
Reza Rejaie, University of Oregon (USA)
Henning Schulzrinne, Columbia University (USA)
Srinivasan Seshan, CMU (USA)
Anees Shaikh, IBM Research (USA)
Lars Wolf, TU Braunsch

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



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Hi,

I have some comments on the draft.

1. The abstract should probably be rewritten, I think the second 
paragraph is not relevant enough for this, while the RTP payload format 
for gateways are not mentioned at all.

2. I think the TOC is long enough to have it alone on the second page, 
i.e. I would propose that one start the introduction chapter on page 3.

3. The third last paragraph in chapter 1 seems to be a clear motivation 
for using TCP to transport text. I think there need to be some further 
elaboration on the reasons for using RTP, because the second last 
paragraph is not a obvious result of the previous one.

4. Chapter 1, last paragraph, last sentence "and introduces a means of 
transporting text
   interleaved with voice within the same RTP session.". Is " a means" 
correct english?

5. In section 2, a reference to the complete information for RFC 2119 
would be nice.

6. Section 3:   "When transport of T.140 text session data in RTP is 
desired, the
   payloads as described in this specification SHOULD be used." This 
sentence seems a bit strange. I don't think using normative words about 
using this payload format real has any meaning.

7. Section 3.1, "This format is primarily used when text is transmitted 
on a separate
   RTP session dedicated for the transmission of text and not shared
   with other media, such as audio, DTMF, etc.  IP textphone devices
   most commonly use this format."

I think the restriction this payload format imposes on the media stream 
should be clearly documented in this section.

8. Section 3.2: "This T140block counter may be utilized to detect lost 
characters." I think one should clearly indicate that the counter if for 
detecting not lost characters but lost T140 blocks. I also think the 
comments made in the WG session in Minneapolis that the last block 
before a switch to another audio format can't be detected.

9. Section 3.6, "Note, also, that incoming text from a PSTN circuit 
might be at a higher bit-rate than can be played out on an egress PSTN 
circuit." The scenario and what is meant in this sentence is not clear.

10. Section 3.7; "For profiles that use dynamic payload type number 
assignment, this payload format is identified by the name "T140" (see 
section 8). If redundancy is used per RFC 2198, the Payload Type MUST 
indicate that payload format ("RED")." I think that one should be 
clearer. We are using MIME type according to the definition in chapter 
8, and also RED has its MIME type that can be used to identify the name. 
Therefore I would reformulate the sentences to: "For profiles that use 
dynamic payload type number assignment, this payload format can be 
identified by the MIME types "text/T140" and "audio/T140" (see section 
8). If redundancy is used per RFC 2198, another payload type number 
needs to be provided for the redundancy format. MIME types for 
identifying RFC 2198 are available in RFC 3555 and RFC YYYY." Where RFC 
YYYY is the RFC number for the text/RED registration.

11. Section 3.7, The RTP sequence number: redefining the RTP sequence 
number through normative text is not good. Please provide a reference to 
RFC 3550 and then you can have the informative text.

12. Section 3.7: "Timestamp: The RTP Timestamp encodes the approximate 
instance of entry of the primary text in the packet. A clock frequency 
of 1000
     Hz MUST be used for text/t140.  The clock frequency may be 
specified for audio/t140 and is generally set to be 8000 Hz, as that is 
most common for audio."

In the last sentence above, it says: The clock frequency may be ... This 
is unclear, it needs normative language, making it clear that the rate 
MAY be set to any value, if not specified by out of band mechanism a RTP 
TS rate value of 8000 SHALL be used.

12.5. Section 3.7, why is the marker bit not defined?

13. Section 3.8: What is the following text doing in this chapter?
  "When redundant transmission of the data according to RFC 2198 is
   desired, the RTP header is followed by one or more redundant data
   block headers, one for each redundant data block to be included.
   Each of these headers provides the timestamp offset and length of the
   corresponding data block plus a payload type number indicating this
   payload format ("T140").  Redundant data older than 16383 divided by
   the clock frequency MUST not be transmitted."

It would seem to belong in section describing the usage of RED. I think 
there are more things in regards to usage of RED, like section 3.9,  
that can be collected in a separate chapter clarifying that it concerns 
the usage of RED for t140.

14. Section 4.1 "A maximum buffering time of 500 ms is specified." Where 
is it specified?

15. Section 5: Another comment from the WG meeting in Minneapolis, was 
the need to make it clear that although the MIME parameter CPS  the 
instantaneous reception rate might be higher due to jitter.

16. Section 8.1, as text/t140 in RFC 2793 does not have rate, and it is 
not need (due to fixed rate), I would remove it. The mapping to SDP 
chapter can simply explain that one simply puts in the rate.

17. There should also be an interoperability consideration regarding CPS 
parameter.

Cheers

Magnus



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Nov 27 10:26:21 2003
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From: "Marshall Eubanks" <tme@multicasttech.com>
Subject: Re: [AVT] RE: [dccp] Re: DCCP Media Guide
To: Stephen Casner <casner@acm.org>,
        "Rosen, Brian" 
 <Brian.Rosen@marconi.com>
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On Wed, 26 Nov 2003 09:35:51 -0800 (PST)
 Stephen Casner <casner@acm.org> wrote:
> On Wed, 26 Nov 2003, Rosen, Brian wrote:
> 
> > Again, I think the most acceptable response to congestion for
> > video is to rate shift down once, or if necessary, twice to
> > get to an "acceptable" level of errors.  So, it's not freeze
> > frame vs rapid rate shift, it's rate shift down vs rapid rate shift.
> >
> > We have a different error result than freeze, and it seems to work
> > better (less objectionable), but it doesn't change the equation.
> > If you get too many errors, you lower the rate until you don't
> > get "too many".
> 
> One quick point, relating back to the start of the thread:  If you are
> competing with one or more TCP streams and the total traffic is
> filling the available bottleneck bandwidth, then if you back off the
> video rate as a result of the loss induced by TCP rate probing, the
> TCP streams will expand to fill the bandwidth you freed and then probe
> for more until you stop.

Dear Stephen;

Tue in theory, but maybe not in practice. See, e.g.,
Stanislav Shalunov's work on tcp tuning.

According to 
http://www.internet2.edu/~shalunov/writing/tcp-perf.html

most TCP implementations are badly tuned for broadband 
and don't actually deliver much more than 500 kbps at best.

I notice empirically that UDP flat rate audio and video can co-exist with 
tcp in practice, and wonder if poor tcp performance might be one of the reasons why.

Maybe if the ideas in
http://www.internet2.edu/~shalunov/tcpar/
were adopted it would become much harder to send video over the Internet.

Regards
Marshall

> 
>                                                         -- Steve
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Stephen Casner <casner@acm.org>
To: Marshall Eubanks <tme@multicasttech.com>
cc: AVT WG <avt@ietf.org>, dccp@ietf.org
Subject: Re: [AVT] RE: [dccp] Re: DCCP Media Guide
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Marshall,

> most TCP implementations are badly tuned for broadband
> and don't actually deliver much more than 500 kbps at best.

Banking on that for a media protocol design would be unwise.

> I notice empirically that UDP flat rate audio and video can co-exist
> with tcp in practice, and wonder if poor tcp performance might be
> one of the reasons why.

Sure.  Flat-rate is the key.  If the UDP traffic does not back off in
response to TCP's probing, then TCP stays within the bandwidth that is
left.  The root of this whole discussion is that if you have ever more
flat-rate UDP traffic, it will push all the TCP out of the way until
the network is saturated with UDP traffic, then the UDP streams will
experience loss.  The TCP folks say that UDP traffic is unfair; DCCP
is being developed as a response.  Now in this conversation, we're
discussing how to keep TCP from pushing DCCP out of the way.

                                                        -- Steve

_______________________________________________
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From: "Phelan, Tom" <tphelan@sonusnet.com>
To: "'philippe.gentric@philips.com'" <philippe.gentric@philips.com>
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Subject: RE: [AVT] RE: [dccp] Re: DCCP Media Guide
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Hi Phillipe,

TFRC is packet rate based, not bit rate.  If TFRC cut your allowed packet
rate in half, it'd be OK (at least as far as TFRC is concerned) if you cut
your packet rate in half, but doubled the packet size.  Of course in many
circumstances this action would be unlikely to alleviate the network
congestion, and some of these new big packets would probably be lost,
causing TFRC to cut the allowed rate even more.

TCP, on the other hand, is bit (or byte) rate based.  Network congestion is
both bit and packet rate based (others things too, I suppose), although I
think the packet rate problems are diminishing (Moore's law helping us
there).

If going to smaller packets gets you a higher bits-on-the-wire rate, then
you'd get some of the buffering I was talking about.  On the other hand, if
the chokepoint was packet-limited rather than bit-limited ...

Tom Phelan

> -----Original Message-----
> From: philippe.gentric@philips.com 
> [mailto:philippe.gentric@philips.com]
> Sent: Wednesday, November 26, 2003 11:31 AM
> To: Henning Schulzrinne
> Cc: avt@ietf.org; dccp@ietf.org
> Subject: Re: [AVT] RE: [dccp] Re: DCCP Media Guide
> 
> 
> 
> Henning, TFRC experts,
> 
> 
> >> It would be interesting to look at the voice response to changing
> >> packetization period.
> 
> ...
> 
> >
> >We have looked at this systematically, varying the packetization
> >threshold. See the papers by Wenyu Jiang and myself at
> >http://netbib.info and http://www1.cs.columbia.edu/~wenyu/
> >
> 
> Does anyone have a clue of how changing the packetization
> impacts the behavior of TFRC in presence of TCP traffic ?
> 
> For example if an application increases its packet rate
> (same media rate, higher transmit rate due to packet headers 
> contribution increasing)
> is that added "pressure" against competing TCP traffic ?
> 
> 
> regards,
> 
> 
> Philippe Gentric
> Chief Architect
> Philips Software
> philippe.gentric@philips.com
> http://www.software.philips.com
> 
> 
> 
> _______________________________________________
> dccp IETF mailing list: dccp@ietf.org
> list info:  https://www1.ietf.org/mailman/listinfo/dccp
> wg charter: http://www.ietf.org/html.charters/dccp-charter.html
> 

_______________________________________________
dccp IETF mailing list: dccp@ietf.org
list info:  https://www1.ietf.org/mailman/listinfo/dccp
wg charter: http://www.ietf.org/html.charters/dccp-charter.html

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Phelan, Tom'" <tphelan@sonusnet.com>,
        "'philippe.gentric@philips.com'" <philippe.gentric@philips.com>
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Again, I think the most acceptable response to congestion for 
video is to rate shift down once, or if necessary, twice to
get to an "acceptable" level of errors.  So, it's not freeze 
frame vs rapid rate shift, it's rate shift down vs rapid rate shift.

We have a different error result than freeze, and it seems to work
better (less objectionable), but it doesn't change the equation.
If you get too many errors, you lower the rate until you don't
get "too many".

It would be interesting to look at the voice response to changing
packetization period.  Clearly, the effect of the loss of one large
packet is worse than the effect of a small packet.  Small packets have
more bit rate, so, you would presumably be sending larger packets
when you have congestion.  What does that do to the perceived quality?
It's going to get worse as congestion gets worse, which in some sense
is "good", but is it actually worse than a higher packet loss rate,
but less problem for each loss?  Probably depends on the loss pattern.
If it were random, than the smaller problem, larger percentage is
probably better, but it's not random.

There are people who know a lot about such things.  I'm not one of them.
Do we have some of that expertise around on the list?  Has that idea
been tried?

I do think we need to split this thread, or at least mark replies with
"Applies to interactive real time" vs "applies to non-interactive real
time".
I appreciate several comments on the thread that DCCP may have more
applicability to non-interactive streams.  While many of the effects
are similar (rate shift to non-interactive is just as objectionable
to non-interactive as it is to interactive), some things are clearly
different (slow start, for example).

Brian


> -----Original Message-----
> From: Phelan, Tom [mailto:tphelan@sonusnet.com]
> Sent: Wednesday, November 26, 2003 8:25 AM
> To: 'philippe.gentric@philips.com'
> Cc: avt@ietf.org; dccp@ietf.org
> Subject: RE: [dccp] Re: DCCP Media Guide
> 
> 
> Hi Phillipe,
> 
> [snipped]
> > The other idea I have been trying to advocate for a while 
> > because I believe it is another key to the problem
> > is that media does not need to be constant bit rate, nor 
> > constant packet rate, nor constant packet size.
> 
> I'm glad you brought this up.  There have been a lot of 
> comments on this
> thread about the difficulties in the human perception of 
> rate-shifting, and,
> in general, I'd have to say the comments are pretty accurate, 
> especially
> given the current state of the art in codec design.  However, 
> I can think of
> at least two instances where rate shifting is either workable 
> or better than
> the current practice.
> 
> One case involves audio codecs such as G.711 and changing the 
> packetization
> interval.  G.711 with 10ms packetization is about 96K IP bps 
> on the wire.
> 40ms packetization gives about 72K bps.  I think that 
> changing between these
> two rates is often close to imperceptible.  A similar thing 
> can be done with
> G.729A (although I'm not sure the standard allows it), giving 
> more variation
> in the bit rate - 10ms packetization is about 38.4K bps and 
> 40ms is 14.4K.
> 
> Another case involves video transmission.  As you change 
> between a high and
> low encoding rate the image the user sees is likely to change 
> back and forth
> between clear and grainy, and this is likely to be annoying.  
> But I imagine
> it's less annoying than the current practice of freezing the 
> video (and
> continuing the audio) when there's insufficient bandwidth for 
> the high-res
> video.  I imagine that layered encoding techniques like 
> Gerard Babonneua
> mentioned can fit this mode of operation nicely.
> 
> Given that there are these low-hanging-fruit stream shifting 
> methods, I
> wonder what others are available if we put our minds to it?
> 
> Tom Phelan
> 
> > These 3 things (delay, reliability, variable rate/packaging) 
> > should give some slack
> > for the design of a solution with added value no?
> > 
> > regards,
> > 
> > 
> > Philippe Gentric
> > Chief Architect
> > Philips Software
> > philippe.gentric@philips.com
> > http://www.software.philips.com
> > 
> > 
> > 
> > _______________________________________________
> > dccp IETF mailing list: dccp@ietf.org
> > list info:  https://www1.ietf.org/mailman/listinfo/dccp
> > wg charter: http://www.ietf.org/html.charters/dccp-charter.html
> > 
> 
> _______________________________________________
> dccp IETF mailing list: dccp@ietf.org
> list info:  https://www1.ietf.org/mailman/listinfo/dccp
> wg charter: http://www.ietf.org/html.charters/dccp-charter.html
> 

_______________________________________________
dccp IETF mailing list: dccp@ietf.org
list info:  https://www1.ietf.org/mailman/listinfo/dccp
wg charter: http://www.ietf.org/html.charters/dccp-charter.html

_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Greg Herlein <gregh@arkstream.com>
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> I fully agree this is the fundamental reason why another
> protocol than TCP is a good idea: media does not need 100%
> reliability.

I don't agree.  *Some* media does not need 100% reliability.  
But many video codecs have pathetic performance in the face of
packet loss - and FEC is not always a good solution (impact on
bandwidth is one reason - complexity is another).  Audio is often 
a lot more tolerant (at least in VoIP) - a loss spurt in video is 
radically painful for the viewer in most cases.

While TCP does have it's own problems for video delivery (and
some advantages too) it's not the ideal solution, clearly.

I most wholeheartedly agree that delay, reliability, variable
rate/packaging are the three elements that *can* be adjusted to 
modify the performance of a video stream.  To that I would add 
dynamic buffering at the endpoints (goes with rate, I guess).  
I'd like to see a protocol (or extension) that can implement this 
into apps that can use it to good measure, as opposed to 
generating YAP (yet another protocol) that addresses only one 
aspect of the problem (and in only one way).

Greg Herlein
VP Engineering
Arkstream, Inc.



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Subject: Re: [AVT] RE: [dccp] Re: DCCP Media Guide
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> One case involves audio codecs such as G.711 and changing the packetization
> interval.  G.711 with 10ms packetization is about 96K IP bps on the wire.
> 40ms packetization gives about 72K bps.  I think that changing between these
> two rates is often close to imperceptible.  A similar thing can be done with
> G.729A (although I'm not sure the standard allows it), giving more variation
> in the bit rate - 10ms packetization is about 38.4K bps and 40ms is 14.4K.

While this is numerically accurate, it seems largely pointless. If I'm 
below the perceptible delay threshold (say 150 ms one-way), there is no 
reason to transmit at 10 ms packetization interval since the user 
experience will not improve noticeably. Indeed, very few systems 
transmit at 10 ms packetization today. In addition, at the access links 
where congestion is most likely, IP/UDP/RTP header compression reduces 
the difference in bit rates to much smaller values, further reducing the 
utility of doing variable packetization. (From our experiments, a 
packetization interval of roughly 30-40 ms also appears to work best 
with FEC.)

> 
> Another case involves video transmission.  As you change between a high and
> low encoding rate the image the user sees is likely to change back and forth
> between clear and grainy, and this is likely to be annoying.  But I imagine
> it's less annoying than the current practice of freezing the video (and
> continuing the audio) when there's insufficient bandwidth for the high-res
> video.  I imagine that layered encoding techniques like Gerard Babonneua
> mentioned can fit this mode of operation nicely.

Again, like Brian pointed out, this assumes that you're aiming for a 
low-quality hey-it's-free experience.

> 
> Given that there are these low-hanging-fruit stream shifting methods, I
> wonder what others are available if we put our minds to it?

Unfortunately, the first is unlikely to be productive and the second has 
very limited applicability.


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_______________________________________________
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From exim@www1.ietf.org  Sat Nov 29 17:21:37 2003
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Hello Tom,

I think it is important to highlight the lack of performances of most =
codecs to a few losses in a network. Today, without losses, =
hierarchicals codecs give a little bit less quality than state of the =
art as MPEG4-AVC. Works are in progress to obtain the same quality with =
multilayers codecs than monolayer (ie: MPEG SVC, MPEG-21 Part 13). =
Hierarchical codecs allow an easy differentiation of data in network and =
in transport layer, and they can help in the objective that a minimum of =
data reach the decoder for a minimal quality, instead of nothing or =
something out of tune.

Best regards,

G=E9rard
=20

-----Message d'origine-----
De : Phelan, Tom [mailto:tphelan@sonusnet.com]=20
Envoy=E9 : mercredi 26 novembre 2003 14:25
=C0 : 'philippe.gentric@philips.com'
Cc : avt@ietf.org; dccp@ietf.org
Objet : [AVT] RE: [dccp] Re: DCCP Media Guide


Hi Phillipe,

[snipped]
> The other idea I have been trying to advocate for a while because I=20
> believe it is another key to the problem is that media does not need=20
> to be constant bit rate, nor constant packet rate, nor constant packet =

> size.

I'm glad you brought this up.  There have been a lot of comments on this =
thread about the difficulties in the human perception of rate-shifting, =
and, in general, I'd have to say the comments are pretty accurate, =
especially given the current state of the art in codec design.  However, =
I can think of at least two instances where rate shifting is either =
workable or better than the current practice.

One case involves audio codecs such as G.711 and changing the =
packetization interval.  G.711 with 10ms packetization is about 96K IP =
bps on the wire. 40ms packetization gives about 72K bps.  I think that =
changing between these two rates is often close to imperceptible.  A =
similar thing can be done with G.729A (although I'm not sure the =
standard allows it), giving more variation in the bit rate - 10ms =
packetization is about 38.4K bps and 40ms is 14.4K.

Another case involves video transmission.  As you change between a high =
and low encoding rate the image the user sees is likely to change back =
and forth between clear and grainy, and this is likely to be annoying.  =
But I imagine it's less annoying than the current practice of freezing =
the video (and continuing the audio) when there's insufficient bandwidth =
for the high-res video.  I imagine that layered encoding techniques like =
Gerard Babonneua mentioned can fit this mode of operation nicely.

Given that there are these low-hanging-fruit stream shifting methods, I =
wonder what others are available if we put our minds to it?

Tom Phelan

> These 3 things (delay, reliability, variable rate/packaging) should=20
> give some slack for the design of a solution with added value no?
>=20
> regards,
>=20
>=20
> Philippe Gentric
> Chief Architect
> Philips Software
> philippe.gentric@philips.com
> http://www.software.philips.com
>=20
>=20
>=20
> _______________________________________________
> dccp IETF mailing list: dccp@ietf.org
> list info:  https://www1.ietf.org/mailman/listinfo/dccp
> wg charter: http://www.ietf.org/html.charters/dccp-charter.html
>=20

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt

_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Wagner Marcel <marcel.wagner@siemens.com>
To: "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>,
        liebl@lnt.e-technik.tu-muenchen.de,
        Pandel Juergen
	 <juergen.pandel@siemens.com>,
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Subject: [AVT] AW: Comments on draft-ietf-avt-uxp-06.txt
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Hi Magnus,

thank you very much for your helpful comments. We will consider them in =
the
next revision of the UXP draft.=20

Regards,
Marcel



> -----Urspr=FCngliche Nachricht-----
> Von: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Gesendet: Montag, 24. November 2003 23:10
> An: liebl@lnt.e-technik.tu-muenchen.de; Wagner Marcel; Pandel=20
> Juergen; Weng Wenrong
> Cc: avt@ietf.org
> Betreff: Comments on draft-ietf-avt-uxp-06.txt
>=20
>=20
> Hi,
>=20
> Sorry for being so late with my comments. I think the concept is good
> enough however there is quite a lot of work left until this is a good =

> protocol specification.
>=20
> 1. I think of a lot of the background material on how Reed
> Salomon codes=20
> work should be moved to the introduction. In other word section 3,4,5 =

> can be put into the a common chapter, possibly with=20
> subsections, however=20
> I think one should be clearer about they being introductionary.
>=20
> 2. Then there is need to start writing some real payload
> specification=20
> chapter. Of the present chapter I think the largest hole is=20
> how one goes=20
> from a unprotected stream into a protected one. As I see it one will=20
> take a series of complete RTP packets and then put them into=20
> payloads of=20
> new RTP packets.
>=20
> 3. There is need to be much clearer on how one fills in the
> protection=20
> matrix and then can on the receiver side understand how one=20
> has written=20
> the bytes from different packets into the matrix to correctly=20
> recover it.
>=20
> 4. I think you authors should consider to include some more
> blank lines,=20
> or in certain cases do a page break between section to improve the=20
> readability of the text. For example the TOC should be moved into a=20
> separate page. Please do also put in a blank line between=20
> each paragraph.
>=20
> 5. You are missing the short copyright statement after status of this
> memo and the abstract. See ID-nits http://www.ietf.org/ID-nits.html
>=20
> 6. Chapter 2 is very hard to read. I would suggest that
> instead of using=20
> numbered bullets, you use the definition being left justified=20
> and then=20
> put the definition more right justified. You may also consider of all =

> these definitions really make sense to be defined in=20
> conventions or if=20
> they more belong in there respective chapter.
>=20
> 7. The H.26L is now H.264,
>=20
> 8. In section 4. The following sentence exist:
> "Furthermore, if channel
>     conditions get worse during transmission, only more and more
>     enhancement layers are lost, i.e. a graceful degradation in
>     application quality at the receiver is achieved [6]."
>=20
> This sentence can be rather dangerous in IETF. The general
> channel you=20
> are talking about is the IP channel which normally only=20
> losses packets=20
> due to congestion. Therefore this sentence can be interpreted as by=20
> simply adding more protection one can avoid most effects of=20
> congestion.=20
> So if you are talking about the very special case for links=20
> that loses=20
> packet of other reasons I think you need to be very explicit.
>=20
> 9. Figure 3 should be moved into the introduction when you
> discuss Reed=20
> Salomon codes.
>=20
> 10.  Page 11, bullet 2. How does on determine the expected
> number of TSB=20
> rows?
>=20
> 11. What RTP timestamp rate should be used for this format? Does it
> really need one? or could one simply transmission time stamp=20
> the packets?
>=20
> 12. Please use more normative wording when defining how the
> marker bit=20
> should be set.
>=20
> 13. Section 6.1: "All other fields in the RTP header are set to those
> values
>     proposed for regular multimedia transmission using the RTP-format
>     of the media stream which is protected by UXP, e.g. for MPEG-4
>     Visual as specified in RFC 3016."
> Normal there are no more fields then RTP TS and Marker bit that the=20
> payload format defines, the rest is set in accordance with the RTP=20
> specification and the profile in use.
>=20
> 14. What is the gain of having the protected media's PT
> available in the=20
> UXP header? As I understand it you will need to decode the minimal=20
> number packets possible to reconstruct the stream to use it at all.
>=20
> 15. How does one treat packets where one is able to recover
> parts of the=20
> packets?
>=20
> 16. The proposed RTP Timestamp assignment mechanism will make
> the RTCP=20
> jitter unusable, if this is deemed acceptable, then at least=20
> it should=20
> be noted.
>=20
> 17. I would recommend that the description of the processing becomes
> much less media oriented and much more common RTP and its=20
> payload specific.
>=20
> 18. I would also like that the specification is more explicit how can
> give different protection levels to different packets or parts of a=20
> packet. There need to be more information about this topic making it=20
> possible for a implementor of UXP for a certain purpose being able to =

> determine how they should behave.
>=20
> 19. This payload format need to register a MIME subtype for
> all the main=20
> media types that uses RTP at all.
>=20
> 20. Then one can write a chapter on how to map this to SDP.
>=20
> 21. In the security section the following text is present:
> " The overall ratio between parity and info octets MUST NOT be
>     higher than 1:1, i.e. the absolute bitrate spent for redundancy
>     must not be larger than the bitrate required for transmission of
>     multimedia data itself."
>   =20
> This statement I understand is intended to address congestion
> control.=20
> This isn't very useful, as the transmission rate in the face=20
> of loss in=20
> a best effort network shall not be higher than what a TCP flow would=20
> achieve over the same path. However the mix between=20
> protection and real=20
> data can be any and should be controlled only in regards to the paths =

> error characteristics. This results in a the trade-off between=20
> protection and media.
>=20
> 22. You are missing the IETF standard IPR statement, see ID-nits for
> references to the sections from RFC 2026 you need.
>=20
> 23. I think you can write an separate address for each of the =
authors.
>=20
> I hope this will provide you with input in writing the next version.
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
>=20
>=20
>=20
>   =20
>=20
>=20

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