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Date: Mon, 01 Dec 2003 09:34:56 -0500
To: Stephen Casner <casner@acm.org>
From: "Michael A. Ramalho" <mramalho@cisco.com>
Subject: Re: [AVT] RE: [dccp] Re: DCCP Media Guide
Cc: Marshall Eubanks <tme@multicasttech.com>, AVT WG <avt@ietf.org>,
        dccp@ietf.org
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All,

I think Steve has hit the essence of the issue.

Some comments in-line.

Michael Ramalho

At 11:10 AM 11/27/2003 -0800, Stephen Casner wrote:
>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.

I did some research on this issue for a major RBOC about 5 years
ago when VoIP was the "imminent threat" to the PSTN. The study
was for "best effort" networks (the story changes with QoS provisioned
flows for UDP/VoIP).

The whole issue on whether UDP-based applications would
"crowd out" TCP to oblivion was the MIX of traffic ON A
PARTICULAR service providers IP network.

If VoIP/UDP was a minority traffic of the mix ... TCP backed off a
little ... was never starved to oblivion (because UDP was a minority
traffic type) ... network collapse did not occur ... and life went
on (even though the UDP was "unfair" to TCP in the bandwidth
sharing socialistic sense).

If, however, the dominant traffic was VoIP/UDP (perhaps an
IP network set up as an Internet Telephone Service Provider) the
story was different. First, said service provider needed to size
(that is traffic engineer) their network for: 1) all the VoIP/UDP
traffic expected to be "admitted" and 2) have enough bandwidth
"left over" for the anticipated TCP traffic running on the network
(e.g., VoIP signalling, messaging, etc.). There is no getting
around "traffic engineering" for this case! It becomes very much
like QoS provisioning your whole network, except that you
size the network based on monitoring UDP and TCP trends rather
than admitting on every UDP flow. You also "allocate" some
percentage of your network for TCP traffic (to ensure it's
timely delivery/performance). This is doable for large backbone
networks due to the law of large numbers.

Now forgetting "fairness" for a while and considering only: 1)
network collapse and 2) "best effort networks" ... the question
comes down to insuring that UDP applications do not EVER
become close to being the dominant traffic source. One way
is forecasting - monitor your network to make sure UDP isn't
becoming your dominant traffic source. If UDP applications are
becoming a dominant source you must do something about
it! You must either:
1) specifically admit/police UDP traffic (essentially providing
a QoS mechanism for the flow or aggregate flow) or
2) design something in DCCP (or equivalent transport) to
back it off or shut it down if the required rate changes are
"too large".

[This is a vast simplification. The needed traffic engineering would
also consider the performance of the remaining TCP
applications in the provisioning/sizing of the network design.]

If DCCP only did #2 above .... it would still be "unfair to TCP"
but it's unfairness would not lead to network collapse. Unfortunately
the DCCP flow cannot figure out that the cause of the congestion
is that the dominant traffic type in a given network traversed is
non-elastic (or near in-elastic). This is a hard problem - you need
network help here (use congestion notifications to provide hints
to the transport?).

I see DCCP as trying to go a step further and, out of "bandwidth
fairness" to TCP, it is trying rate control the inherently "near in-elastic"
application. This leads to the present quandary of DCCPs unsuitability
for VoIP/UDP flows in best effort networks.

Sorry for the long post - but this is a hard problem.

>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.

This is hard because any particular flow doesn't know the overall
network condition.


>                                                         -- 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



From exim@www1.ietf.org  Mon Dec  1 11:38:33 2003
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Date: Mon, 01 Dec 2003 11:36:34 -0500
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From: Tom Taylor <taylor@nortelnetworks.com>
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Subject: [AVT] Missing events in RFC 2833
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I'm finding it a little difficult to see how some of the applications 
supported by RFC 2833 work without a few more events.  Here are some 
examples:

1. V.18 text telephones interwork with a number of other text telephone 
types.  The only event defined specifically for V.18 operation seems 
rather arbitrarily chosen: the ANS2225 event for Bell 103 answer tone. 
There don't seem to be any events for Baudot or V.23 or for Bell 103 in 
calling mode.

2. Signalling System No. 5 has two sets of signals: line signals, which 
are tones controlling trunk state, and register signals, which convey 
digits with some supporting information.  There is a timing constraint 
that register signalling must begin within 80 +/- 20 ms of the end of 
the "seize" line signal.  Yet no event is defined for "seize", only for 
the register signals.  It's a little hard to see how the receiving 
gateway can propagate KPx (the initial register signal) on time if it 
doesn't know that "seize" was playing, even if the sending end can 
recognize and report the start of KPx within the 20 ms tolerance.  As I 
see it, the present set of events is useful only to transfer the 
register events from one gateway to another which will report them to 
call control rather than propagating them as tones.

These missing events raise the issue of whether proposals to add events 
to RFC 2833 should be backed up by requirements documents describing 
their use.  I have been adding brief application descriptions in my 
rewrite of 2833bis, as justification for the normative text which I have 
also been adding.  (Basically just obvious stuff, missing the first time 
around and highlighted by Flemming's comments earlier this year.)  I am 
hoping that this additional information prompts a critical review of the 
current set of 2833/2833bis events, with a view to rounding them out as 
necessary to make applications work well.

Tom Taylor


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



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> "Michael A. Ramalho" <mramalho@cisco.com>
>
> If UDP applications are becoming a dominant source you must do
> something about it! You must either:
>
> 1) specifically admit/police UDP traffic (essentially providing
> a QoS mechanism for the flow or aggregate flow)

This is an opportunity, folks -- for QoS deployment, for multicast
deployment, and for the migration of real-time non-standardized UDP
apps like networked games to RTP and SIP (to take advantage of QoS and
multicast-enabled networks).

---------------------------------------------------
John Lazzaro    http://www.cs.berkeley.edu/~lazzaro
     lazzaro [at] cs [dot] berkeley [dot] edu     
---------------------------------------------------

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



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> "Michael A. Ramalho" <mramalho@cisco.com>
>
> If UDP applications are becoming a dominant source you must do
> something about it! You must either:
>
> 1) specifically admit/police UDP traffic (essentially providing
> a QoS mechanism for the flow or aggregate flow)

This is an opportunity, folks -- for QoS deployment, for multicast
deployment, and for the migration of real-time non-standardized UDP
apps like networked games to RTP and SIP (to take advantage of QoS and
multicast-enabled networks).

---------------------------------------------------
John Lazzaro    http://www.cs.berkeley.edu/~lazzaro
     lazzaro [at] cs [dot] berkeley [dot] edu     
---------------------------------------------------

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



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Subject: [AVT] RFC 3640 on RTP Payload Format for Transport of MPEG-4 Elementary Streams
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--NextPart


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


        RFC 3640

        Title:      RTP Payload Format for Transport of MPEG-4
                    Elementary Streams
        Author(s):  J. van der Meer, D. Mackie, V. Swaminathan,
                    D. Singer, P. Gentric
        Status:     Standards Track
        Date:       November 2003
        Mailbox:    jan.vandermeer@philips.com, dmackie@apple.com,
                    viswanathan.swaminathan@sun.com, singer@apple.com,
                    philippe.gentric@philips.com
        Pages:      43
        Characters: 102606
        Updates/Obsoletes/SeeAlso:    None

        I-D Tag:    draft-ietf-avt-mpeg4-simple-08.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3640.txt


The Motion Picture Experts Group (MPEG) Committee (ISO/IEC JTC1/SC29
WG11) is a working group in ISO that produced the MPEG-4
standard.  MPEG defines tools to compress content such as audio-visual
information into elementary streams.  This specification defines a
simple, but generic RTP payload format for transport of any
non-multiplexed MPEG-4 elementary stream.

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.

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



From exim@www1.ietf.org  Mon Dec  1 16:06:22 2003
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Date: Mon, 1 Dec 2003 13:04:58 -0800 (PST)
From: Stephen Casner <casner@acm.org>
To: AVT WG <avt@ietf.org>
Subject: Re: [AVT] RFC 3640 on RTP Payload Format for Transport of MPEG-4
 Elementary Streams
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On Mon, 1 Dec 2003 rfc-editor@rfc-editor.org wrote:

>
> A new Request for Comments is now available in online RFC libraries.
>
>
>         RFC 3640
>
>         Title:      RTP Payload Format for Transport of MPEG-4
>                     Elementary Streams
>         Author(s):  J. van der Meer, D. Mackie, V. Swaminathan,
>                     D. Singer, P. Gentric
>         Status:     Standards Track
>         Date:       November 2003

Yea!

                                                        -- Steve

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From exim@www1.ietf.org  Mon Dec  1 17:21:34 2003
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From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: "avt IETF" <avt@ietf.org>, "Gregg Vanderheiden" <gv@trace.wisc.edu>,
        "Paul Jones" <paulej@packetizer.com>, "Toip list" <toip@snowshore.com>
Subject: FW: [AVT] Missing events in RFC 2833
Date: Mon, 1 Dec 2003 23:20:15 +0100
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Below, Tom Taylor asks if there is any interest to complete the textphone
related tone events in RFC2833 so that all V.18 submodes can be transported
with RFC2833.

We have to verify that we have satisfying detection and transport methods in
RFC2793, and e.g. V.18 or the H.248.2 package descriptions, so that we can
leave RFC2833 half done for text transport.

I would like to do so, but then the market forces must be strong enough to
make RFC2793 and textphone detection commonplace.

Using RFC2833 will place a task similar to V.18 logic on the IP endpoint,
but with less exact information than the native V.18 has. We have no
analysis if that task is realistic, nor if it is likely to be implemented by
IP endpoint developers.

( The missing events for complete text transport are probably:
V.21 data in lo channel at 110 bits/s
Bell 103 hi channel data at 300 bits/s
Bell 103 lo channel data at 300 bits/s
V.23 data at 1200 bits/s
V.23 data at 75 bits/s
Baudot data at 45.45 bits/s
Baudot data at 50 bits/s

(assuming that V.8 signals and V.21 at 300 bits/s in both directions and
DTMF are covered )

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: Monday, December 01, 2003 5:37 PM
To: avt@ietf.org
Subject: [AVT] Missing events in RFC 2833


I'm finding it a little difficult to see how some of the applications
supported by RFC 2833 work without a few more events.  Here are some
examples:

1. V.18 text telephones interwork with a number of other text telephone
types.  The only event defined specifically for V.18 operation seems
rather arbitrarily chosen: the ANS2225 event for Bell 103 answer tone.
There don't seem to be any events for Baudot or V.23 or for Bell 103 in
calling mode.

2. Signalling System No. 5 has two sets of signals: line signals, which
are tones controlling trunk state, and register signals, which convey
digits with some supporting information.  There is a timing constraint
that register signalling must begin within 80 +/- 20 ms of the end of
the "seize" line signal.  Yet no event is defined for "seize", only for
the register signals.  It's a little hard to see how the receiving
gateway can propagate KPx (the initial register signal) on time if it
doesn't know that "seize" was playing, even if the sending end can
recognize and report the start of KPx within the 20 ms tolerance.  As I
see it, the present set of events is useful only to transfer the
register events from one gateway to another which will report them to
call control rather than propagating them as tones.

These missing events raise the issue of whether proposals to add events
to RFC 2833 should be backed up by requirements documents describing
their use.  I have been adding brief application descriptions in my
rewrite of 2833bis, as justification for the normative text which I have
also been adding.  (Basically just obvious stuff, missing the first time
around and highlighted by Flemming's comments earlier this year.)  I am
hoping that this additional information prompts a critical review of the
current set of 2833/2833bis events, with a view to rounding them out as
necessary to make applications work well.

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  Tue Dec  2 11:32:39 2003
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--> Internet-Drafts@ietf.org writes:
> 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

A few comments on this draft:

In Section 2, the draft needs to cite RFC 2119 as a numbered reference

In Section 3, why "SHOULD be used" rather than "MUST be used"?

Regarding Sections 3.4 and 3.5, I think it's important to distinguish
between the text/t140 format - which imposes restrictions on the type of
FEC which can be used - and the audio/t140 format which is largely agnostic
on the FEC/redundancy format used. Section 3.5 can be interpreted to imply
that the only form of FEC that can be used is RFC 2733, but should allow
the use of FEC with RFC 2733 as an example.

Section 3.7 needs to require that the clock frequency for audio/t140 SHOULD
match that of any audio send as part of the same RTP session.

As I mentioned in the meeting, there are corner cases where the T140 block
counter can't detect loss (for example consider, when switching from text
to audio, I can't distinguish between the last packet of text and first
packet of audio being lost, or the first two packets of audio being lost -
until I get the next text packet). Even if this is not considered aa issue,
the draft needs to highlight the scenario, and explain why it's not an
issue.

How is the RTP M bit used? Usual audio conventions?

In Section 5, the draft should note that jitter in the network may cause
packets to arrive faster than the signalled cps for short periods of time.
A receiver has to be aware that this might happen, and buffer to smooth
the arrival rate, even if it signals the cps.

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

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



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        <gv@trace.wisc.edu>
References: <sfcc9778.053@mailserver.viataal.nl>
Subject: Re: Betr.: FW: [AVT] Missing events in RFC 2833
Date: Tue, 2 Dec 2003 11:56:36 -0500
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Arnoud,

RFC 2833 can be used to convey tone information (frequencies and durations),
as well as tone events.  It's not restricted to DTMF.

RFC 2833 could potentially be used as an alternative to RFC 2793.  A gateway
or IP phone, for example, could simply examine the tones and convey those as
"descriptions" (either frequency and duration information or event codes)
over an IP network.  The other end could then receive those descriptions and
reproduce the tones.  They would not have to understand the underlying
textphone protocol.

The problem with RFC 2833, as I see it, is that it stands at odds with the
desire of folks on the ToIP mailing list to try to remove textphone
signaling from the IP network.  There has been a strong desire expressed to
terminate that textphone signaling at the gateway and only use text on the
IP network.  Doing that, arguably, would make interfacing with IP-based IVR
systems, IP-based voice/text-mail systems, or other IP-based systems much
easier.  I say "arguably" because some have suggested to me that it's very
simple to write software that understand the textphone signals received in
RFC 2833 packets... though I personally think it puts an unfortunate burden
on a Pocket PC, for example.

To this point, neither the ITU nor the IETF has considered RFC 2833 as an
option or alternative to RFC 2793.  I think this is probably an important
topic that we should take up immediately, as both bodies are forging ahead
with plans based on RFC 2793 only.

Paul

----- Original Message ----- 
From: "A vWijk" <A.vWijk@viataal.nl>
To: <avt@ietf.org>; <gunnar.hellstrom@omnitor.se>; <paulej@packetizer.com>;
<toip@snowshore.com>; <gv@trace.wisc.edu>
Sent: Tuesday, December 02, 2003 7:45 AM
Subject: Betr.: FW: [AVT] Missing events in RFC 2833


> Gunnar,
> why is this desirable?
>
> We will have T.140 over RTP and in some cases over TCP/IP.
> Why the tones?
>
> Or am I mixing things up again? RFC2833 is for DTMF signalling?
>
> greetz
>
> Arnoud
>
> Drs. Arnoud A. T. van Wijk
> Viataal
> Research & Development
> Afdeling RDS
> Theerestraat 42
> 5271 GD Sint-Michielsgestel
> The Netherlands.
> Mobile: +31651921948
> International text telephone: +31735588408
>
> >>> "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se> 01-12-03 23:20 >>>
> Below, Tom Taylor asks if there is any interest to complete the textphone
> related tone events in RFC2833 so that all V.18 submodes can be
transported
> with RFC2833.
>
> We have to verify that we have satisfying detection and transport methods
in
> RFC2793, and e.g. V.18 or the H.248.2 package descriptions, so that we can
> leave RFC2833 half done for text transport.
>
> I would like to do so, but then the market forces must be strong enough to
> make RFC2793 and textphone detection commonplace.
>
> Using RFC2833 will place a task similar to V.18 logic on the IP endpoint,
> but with less exact information than the native V.18 has. We have no
> analysis if that task is realistic, nor if it is likely to be implemented
by
> IP endpoint developers.
>
> ( The missing events for complete text transport are probably:
> V.21 data in lo channel at 110 bits/s
> Bell 103 hi channel data at 300 bits/s
> Bell 103 lo channel data at 300 bits/s
> V.23 data at 1200 bits/s
> V.23 data at 75 bits/s
> Baudot data at 45.45 bits/s
> Baudot data at 50 bits/s
>
> (assuming that V.8 signals and V.21 at 300 bits/s in both directions and
> DTMF are covered )
>
> 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: Monday, December 01, 2003 5:37 PM
> To: avt@ietf.org
> Subject: [AVT] Missing events in RFC 2833
>
>
> I'm finding it a little difficult to see how some of the applications
> supported by RFC 2833 work without a few more events.  Here are some
> examples:
>
> 1. V.18 text telephones interwork with a number of other text telephone
> types.  The only event defined specifically for V.18 operation seems
> rather arbitrarily chosen: the ANS2225 event for Bell 103 answer tone.
> There don't seem to be any events for Baudot or V.23 or for Bell 103 in
> calling mode.
>
> 2. Signalling System No. 5 has two sets of signals: line signals, which
> are tones controlling trunk state, and register signals, which convey
> digits with some supporting information.  There is a timing constraint
> that register signalling must begin within 80 +/- 20 ms of the end of
> the "seize" line signal.  Yet no event is defined for "seize", only for
> the register signals.  It's a little hard to see how the receiving
> gateway can propagate KPx (the initial register signal) on time if it
> doesn't know that "seize" was playing, even if the sending end can
> recognize and report the start of KPx within the 20 ms tolerance.  As I
> see it, the present set of events is useful only to transfer the
> register events from one gateway to another which will report them to
> call control rather than propagating them as tones.
>
> These missing events raise the issue of whether proposals to add events
> to RFC 2833 should be backed up by requirements documents describing
> their use.  I have been adding brief application descriptions in my
> rewrite of 2833bis, as justification for the normative text which I have
> also been adding.  (Basically just obvious stuff, missing the first time
> around and highlighted by Flemming's comments earlier this year.)  I am
> hoping that this additional information prompts a critical review of the
> current set of 2833/2833bis events, with a view to rounding them out as
> necessary to make applications work well.
>
> Tom Taylor
>
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
>
>
> -
> This list is maintained by Snowshore Networks - http://www.snowshore.com
> All comments on this list are the comments of the message originators and
> Snowshore is not to be held responsible for any actions or comments found
> on this list. The archives for this list can be found at
> http://flyingfox.snowshore.com/toip_archive/maillist.html
>


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



From exim@www1.ietf.org  Tue Dec  2 13:43:26 2003
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Subject: Re: [AVT] RFC 3640 on RTP Payload Format for Transport of MPEG-4 Elementary
 Streams
To: Stephen Casner <casner@acm.org>
Cc: AVT WG <avt@ietf.org>
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stephen you wrote:

>>
>>         RFC 3640
>>
>>         Title:      RTP Payload Format for Transport of MPEG-4
>>                     Elementary Streams
>>         Author(s):  J. van der Meer, D. Mackie, V. Swaminathan,
>>                     D. Singer, P. Gentric
>>         Status:     Standards Track
>>         Date:       November 2003
>
>Yea!
>
>                                                 -- Steve

double yea !

a bit of history:
the subject was initiated at the Munich IETF meeting in August 1997 (39th) !
then at least 4 drafts were generated (?) and it went a bit asleep,
then I started to work on it after the La Baule MPEG meeting in Octobre 2000
because audio frame aggregation was required for efficiency
and made 8+ drafts for MultiSL/SingleSL in 2000 and 2001 (I lost the count !)
Then based on crowd feedback the SL went out of
the picture/story (the format remained the same ;-)
and Jan made the 9 drafts of the "simple" series:
so this adds up to at least 21 drafts ...
almost 19 IETF meetings, plus probably as many MPEG meetings,
going around the planet several times on the way,
plus many different people and quite some work.

So I would say it should go into the book named after a beer
or speaking of beverage with bubbles (we all have our preferences ;-)
I invite every one involved to try a new concept:
the virtual distributed jet-lag-free toast:
whenever it will be the evening for you today,
wherever you are, raise your glass and shout: 3640!-)

Anyway thanks everybody,
for your help if you cared,
for your patience if you did not ;-)


Philippe Gentric
Chief Architect
Philips Software
philippe.gentric@philips.com
http://www.software.philips.com



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



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From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: "A vWijk" <A.vWijk@viataal.nl>, <avt@ietf.org>, <paulej@packetizer.com>,
        <toip@snowshore.com>, <gv@trace.wisc.edu>
Subject: RE: Betr.: FW: [AVT] Missing events in RFC 2833
Date: Tue, 2 Dec 2003 20:47:36 +0100
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Arnoud and all,

I do not think it is desirable to try to patch up RFC2833 to transport all
kinds of textphone data events. In my view it is likely to be complex,
unreliable and against the efforts to create mainstream conversational
services for all with high functionality in the IP network.

But I wanted to expose this question from Tom to the text conversation
interested people to get your view, and give som good advice back to Tom,
who is just now cleaning up RFC2833.

You are rtight that RFC2833 started as a mechanism for DTMF transport and
that is what it is most famous for. But have a look in it, and you will find
that many other signals that are likely to appear in PSTN connections can be
coded with RFC2833.

If we had any wish to recreate the same tones at exit as was input, there
could be a case for patching up RFC2833 to (detect and) transport all
textphone data properly.

Now, the case is the opposite, we want to disconnect the technologies in the
both ends as much as possible, and have a pure text code in between.
Therefore, complete V.18 in the PSTN edge and RFC2793 transport is the
solution I see have a future.

Agree?

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: A vWijk [mailto:A.vWijk@viataal.nl]
Sent: Tuesday, December 02, 2003 1:45 PM
To: avt@ietf.org; gunnar.hellstrom@omnitor.se; paulej@packetizer.com;
toip@snowshore.com; gv@trace.wisc.edu
Subject: Betr.: FW: [AVT] Missing events in RFC 2833


Gunnar,
why is this desirable?

We will have T.140 over RTP and in some cases over TCP/IP.
Why the tones?

Or am I mixing things up again? RFC2833 is for DTMF signalling?

greetz

Arnoud

Drs. Arnoud A. T. van Wijk
Viataal
Research & Development
Afdeling RDS
Theerestraat 42
5271 GD Sint-Michielsgestel
The Netherlands.
Mobile: +31651921948
International text telephone: +31735588408

>>> "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se> 01-12-03 23:20 >>>
Below, Tom Taylor asks if there is any interest to complete the textphone
related tone events in RFC2833 so that all V.18 submodes can be transported
with RFC2833.

We have to verify that we have satisfying detection and transport methods in
RFC2793, and e.g. V.18 or the H.248.2 package descriptions, so that we can
leave RFC2833 half done for text transport.

I would like to do so, but then the market forces must be strong enough to
make RFC2793 and textphone detection commonplace.

Using RFC2833 will place a task similar to V.18 logic on the IP endpoint,
but with less exact information than the native V.18 has. We have no
analysis if that task is realistic, nor if it is likely to be implemented by
IP endpoint developers.

( The missing events for complete text transport are probably:
V.21 data in lo channel at 110 bits/s
Bell 103 hi channel data at 300 bits/s
Bell 103 lo channel data at 300 bits/s
V.23 data at 1200 bits/s
V.23 data at 75 bits/s
Baudot data at 45.45 bits/s
Baudot data at 50 bits/s

(assuming that V.8 signals and V.21 at 300 bits/s in both directions and
DTMF are covered )

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: Monday, December 01, 2003 5:37 PM
To: avt@ietf.org
Subject: [AVT] Missing events in RFC 2833


I'm finding it a little difficult to see how some of the applications
supported by RFC 2833 work without a few more events.  Here are some
examples:

1. V.18 text telephones interwork with a number of other text telephone
types.  The only event defined specifically for V.18 operation seems
rather arbitrarily chosen: the ANS2225 event for Bell 103 answer tone.
There don't seem to be any events for Baudot or V.23 or for Bell 103 in
calling mode.

2. Signalling System No. 5 has two sets of signals: line signals, which
are tones controlling trunk state, and register signals, which convey
digits with some supporting information.  There is a timing constraint
that register signalling must begin within 80 +/- 20 ms of the end of
the "seize" line signal.  Yet no event is defined for "seize", only for
the register signals.  It's a little hard to see how the receiving
gateway can propagate KPx (the initial register signal) on time if it
doesn't know that "seize" was playing, even if the sending end can
recognize and report the start of KPx within the 20 ms tolerance.  As I
see it, the present set of events is useful only to transfer the
register events from one gateway to another which will report them to
call control rather than propagating them as tones.

These missing events raise the issue of whether proposals to add events
to RFC 2833 should be backed up by requirements documents describing
their use.  I have been adding brief application descriptions in my
rewrite of 2833bis, as justification for the normative text which I have
also been adding.  (Basically just obvious stuff, missing the first time
around and highlighted by Flemming's comments earlier this year.)  I am
hoping that this additional information prompts a critical review of the
current set of 2833/2833bis events, with a view to rounding them out as
necessary to make applications work well.

Tom Taylor


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


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--NextPart

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

	Title		: RTP Payload Format for iLBC Speech
	Author(s)	: A. Duric, S. Andersen
	Filename	: draft-ietf-avt-rtp-ilbc-04.txt
	Pages		: 12
	Date		: 2003-12-2
	
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.

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



From exim@www1.ietf.org  Tue Dec  2 16:05:31 2003
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Subject: [AVT] Trunk unavailable event in 2833bis
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What was the reasoning for not defining "trunk unavailable" as a state 
in 2833bis?  It would seem natural to do so, and to say that if the 
duration expires without a refresh, the receiver assumes that the state 
has expired.  (That is, you don't have to define and keep sending "trunk 
available" events.)

Tom Taylor


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



From exim@www1.ietf.org  Tue Dec  2 19:22:25 2003
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From: Eddie Kohler <kohler@icir.org>
Subject: [AVT] Is "congestion collapse" misleading?
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There was a discussion earlier about whether "congestion collapse" is a
fair description of the situation when nonresponsive UDP flows cause TCP
bandwidth to suffer.

Well, the *situation* is a problem, and local administration cannot correct
it.

Say that S is sending a nonresponsive flow to R.  At the same time A is
sending a responsive flow to B.  S-R and A-B share parts of a path.

    S ------+----------+-----> R
            |          |
            |          v
            A          B

S-R experiences a high loss rate, say somewhere between A and R.  Sure, R's
administrator can take steps to prevent other flows from/to R from
degrading.  But the A-B flow is affected too; and depending on loss rates
and bandwidths, the A-B flow might "collapse", since S-R doesn't respond to
loss rates.

The user at R might or might not stop the flow.  Anyway, we can't count on
them.

The Internet architecture should support this kind of topology, which isn't
entirely unusual in practice.

Eddie

_______________________________________________
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From exim@www1.ietf.org  Tue Dec  2 19:36:21 2003
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From: Eddie Kohler <kohler@icir.org>
Subject: [AVT] Would a DCCP flow need to send 2x its preferred rate?
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(To protect itself from "greedy TCP flows" gobbling up available
bandwidth?)

No.

TFRC exists to smooth out responses to congestion.  A single loss (such as
that caused by TCP probing) does not halve the TFRC transmit rate.  In many
cases it will cause only small changes (which, as we've discussed, the
application might be able to ignore).  In many cases it will not change the
rate at all.

If the congested link has high levels of statistical multiplexing, the TCP
flows won't likely gobble up the available bandwidth anyway -- even during
a silent period.

If the congested link has low levels of statmux -- like one DCCP flow and
one TCP flow -- then yes, the TCP would try for the available bandwidth.
And during a silent period, the TFRC flow would have to transmit something
so it wasn't crowded out entirely.  But this amount would be *half* the
nominal rate, or perhaps less.  Not twice.

Eddie

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



From exim@www1.ietf.org  Tue Dec  2 19:46:22 2003
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Thread-Topic: Trunk unavailable event in 2833bis
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From: "Rajesh Kumar" <rkumar@cisco.com>
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Subject: [AVT] RE: Trunk unavailable event in 2833bis
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This was at the recommendation of the AVT folks that on/off state
signaling was out of the scope of RFC 2833.The trunk available acts like
a per-DS0 AIS. In North America, you do not need it because you can use
the signaling path to take individual DS0s out of service. In the ETSI
countries, this is done by sending special patterns in the bearer path.
Now you could use RTP to send those patterns but that would be less
efficient since you would need to send the bit pattern continuously
without using VAD. We concluded that an RFC 2833 event message with
periodic, albeit longer, refresh was more efficient.

Rajesh Kumar

> -----Original Message-----
> From: Tom Taylor [mailto:taylor@nortelnetworks.com]
> Sent: Tuesday, December 02, 2003 1:03 PM
> To: Rajesh Kumar; avt@ietf.org
> Subject: Trunk unavailable event in 2833bis
>=20
> What was the reasoning for not defining "trunk unavailable" as a state
> in 2833bis?  It would seem natural to do so, and to say that if the
> duration expires without a refresh, the receiver assumes that the
state
> has expired.  (That is, you don't have to define and keep sending
"trunk
> available" events.)
>=20
> Tom Taylor


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



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From: Eddie Kohler <kohler@icir.org>
Subject: [AVT] Is dropping flows enough to avoid congestion collapse?
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There have been suggestions that dropping flows is enough to avoid
congestion collapse, and that RTP-over-UDP already does this.

Well, sort of.  Currently, *users* may drop flows when they see high levels
of loss.  On the other hand they may not.  Certainly we can't count on it,
and we can't count on users watching for any particular loss rate.  1%?
30%? 40%?  Depends on the codec and the user.

Carrying through the IAB's suggestion would mean endpoint operating
systems, or perhaps routers or middleboxes, terminating UDP flows, or
treating UDP as less than best effort, when there is congestion.

That would seem a deeply nonsatisfying result, and probably far worse for
user experience than DCCP.  (I'm also not sure DCCP will be that bad --
there are several misunderstandings floating around, as well as some real
issues.)

Of course, UDP-penalization might not ever happen, but some "people working
on the stability of the network" want it to.  (I'm thinking of Sally
Floyd.)

Eddie

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



From exim@www1.ietf.org  Tue Dec  2 19:51:21 2003
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Subject: RE: [AVT] RE: Trunk unavailable event in 2833bis
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From: "Rajesh Kumar" <rkumar@cisco.com>
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Another point: I need to correct an unintentional typo in both your
email and mine. The event is "trunk UN-available" lest I cause
unnecessary heartburn about the need to periodically refresh a trunk
available event!

> -----Original Message-----
> From: Rajesh Kumar
> Sent: Tuesday, December 02, 2003 4:43 PM
> To: Tom Taylor; avt@ietf.org
> Subject: [AVT] RE: Trunk unavailable event in 2833bis
>=20
> This was at the recommendation of the AVT folks that on/off state
> signaling was out of the scope of RFC 2833.The trunk available acts
like
> a per-DS0 AIS. In North America, you do not need it because you can
use
> the signaling path to take individual DS0s out of service. In the ETSI
> countries, this is done by sending special patterns in the bearer
path.
> Now you could use RTP to send those patterns but that would be less
> efficient since you would need to send the bit pattern continuously
> without using VAD. We concluded that an RFC 2833 event message with
> periodic, albeit longer, refresh was more efficient.
>=20
> Rajesh Kumar
>=20
> > -----Original Message-----
> > From: Tom Taylor [mailto:taylor@nortelnetworks.com]
> > Sent: Tuesday, December 02, 2003 1:03 PM
> > To: Rajesh Kumar; avt@ietf.org
> > Subject: Trunk unavailable event in 2833bis
> >
> > What was the reasoning for not defining "trunk unavailable" as a
state
> > in 2833bis?  It would seem natural to do so, and to say that if the
> > duration expires without a refresh, the receiver assumes that the
> state
> > has expired.  (That is, you don't have to define and keep sending
> "trunk
> > available" events.)
> >
> > Tom Taylor
>=20
>=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  Tue Dec  2 20:09:22 2003
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Date: Tue, 02 Dec 2003 17:08:14 -0800
From: Eddie Kohler <kohler@icir.org>
Subject: [AVT] Is TCP broken?
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There was a discussion earlier about whether "it is TCP that is broken",
meaning does TCP behave particularly badly when paired with real-time
media.

TCP is, of course, pretty stupid in many ways, but I didn't quite
understand *how* TCP was broken here...?  Was it this? (the most likely
possibility)

* TCP is broken because it probes for available bandwidth during real-time
  media's quiet periods.

  Guilty as charged, but it might not matter.  If the congested link has a
  large level of statistical multiplexing, or if the TCP flow is
  short-lived, it'll all come out in the wash.  On the other hand, if the
  congested link has a small level of statmux, then the congested router or
  media could just punish TCP a bit to get it back in line.

  Without a bandwidth reservation, it's hard to know how to do better.

  Now, if the real-time media flow was using TFRC, it could preserve its
  bandwidth allocation essentially indefinitely by sending probe traffic at
  *half* its full-speed rate during silent periods (NOT twice).

Other possibilities:

* TCP is broken because it probes for available bandwidth, period.

  I don't think TCP's bandwidth probes will punish TFRC unduly.
  Simulations have been done showing that TCP and TFRC achieve pretty fair
  allocations.

* TCP is broken because it reacts so violently to loss, which constrains
  real-time media unduly.

  Guilty.  TCP does halve its rate in response to each congestion event.
  And yes, that should, up to a point, constrain application designers who
  want to run over the best-effort Internet.  But I'm not yet convinced
  that real-time media won't work.  Many of the constraints don't seem that
  onerous.  (A later mail.)

Eddie

_______________________________________________
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: "'Eddie Kohler'" <kohler@icir.org>, avt@ietf.org
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Subject: [AVT] RE: [dccp] Would a DCCP flow need to send 2x its preferred rate?
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Hi Eddie,

See inline ...

Tom Phelan

> -----Original Message-----
> From: Eddie Kohler [mailto:kohler@icir.org]
> Sent: Tuesday, December 02, 2003 7:36 PM
> To: avt@ietf.org
> Cc: dccp@ietf.org
> Subject: [dccp] Would a DCCP flow need to send 2x its preferred rate?
> 
> 
> (To protect itself from "greedy TCP flows" gobbling up available
> bandwidth?)
> 
> No.
> 
> TFRC exists to smooth out responses to congestion.  A single 
> loss (such as
> that caused by TCP probing) does not halve the TFRC transmit 
> rate.  In many
> cases it will cause only small changes (which, as we've discussed, the
> application might be able to ignore).  In many cases it will 
> not change the
> rate at all.
> 
> If the congested link has high levels of statistical 
> multiplexing, the TCP
> flows won't likely gobble up the available bandwidth anyway 
> -- even during
> a silent period.

Huh?  You seem to be saying that if you have a lot of TCP streams on a link
they won't use all of the available bandwidth.  I thought just the opposite.
With a low level of multiplexing there's more chance of synchronizing losses
and cutbacks, and not fully utilizing the link.  But with a large number of
flows enough of them will be blasting away while others are cut back.

> If the congested link has low levels of statmux -- like one 
> DCCP flow and
> one TCP flow -- then yes, the TCP would try for the available 
> bandwidth.
> And during a silent period, the TFRC flow would have to 
> transmit something
> so it wasn't crowded out entirely.  But this amount would be 
> *half* the
> nominal rate, or perhaps less.  Not twice.

So this relates to our previous conversation?  When you're transmitting at
less than the TFRC allowed rate, your allowed rate could be twice what
you're transmitting.  If a loss event happens, your allowed rate is cut
back, but since it was twice what you were actually transmitting, no effect.
This gives you the effective "protection" of transmitting at twice your
necessary rate, without actually doing it.

Great if it works.  A simulation would be nice :-).

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



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Date: Tue, 02 Dec 2003 17:37:32 -0800
From: Eddie Kohler <kohler@icir.org>
Subject: [AVT] Re: [dccp] Would a DCCP flow need to send 2x its preferred rate?
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> > If the congested link has high levels of statistical multiplexing, the
> > TCP flows won't likely gobble up the available bandwidth anyway -- even
> > during a silent period.
> 
> Huh?  You seem to be saying that if you have a lot of TCP streams on a link
> they won't use all of the available bandwidth.  I thought just the opposite.
> With a low level of multiplexing there's more chance of synchronizing losses
> and cutbacks, and not fully utilizing the link.  But with a large number of
> flows enough of them will be blasting away while others are cut back.

Most links nowadays with high levels of statmux have lots of TCP flows
coming and going all the time.  Yes, long-lived flows would take up all the
available bandwidth, and they might be hard to beat back.  But that's not
a realistic scenario in most parts of the Internet.

> So this relates to our previous conversation?  When you're transmitting at
> less than the TFRC allowed rate, your allowed rate could be twice what
> you're transmitting.  If a loss event happens, your allowed rate is cut
> back, but since it was twice what you were actually transmitting, no effect.
> This gives you the effective "protection" of transmitting at twice your
> necessary rate, without actually doing it.

Yep.  Also note that your allowed rate wouldn't be cut back by half right
away -- that's the power of TFRC.  It would drop slowly if at all.

> A simulation would be nice :-).

Yep. :)

Eddie

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



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Date: Tue, 02 Dec 2003 17:32:38 -0800
From: Eddie Kohler <kohler@icir.org>
Subject: [AVT] Would DCCP kill Internet real-time media distribution?
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By "real-time media", I mean "current more-or-less-CBR real-time media".

To summarize the problems that a best-effort media stream might suffer:
(Please correct me if I'm wrong or have left anything out!)

Axiom 1: Loss sucks.  ("a 3% loss rate means a 30% reduction in quality")
Axiom 2: Rate changes suck.  ("Have you ever actually listened ... stream
         with varying rate? ... It sucks, to put it mildly.")

So, following this through:

* Say there's no loss.

  If there's no loss, DCCP and UDP are equivalent, except that DCCP
  enforces slow-start.  Slow-start will not matter for low-rate flows like
  most VoIP flows.  It will for higher-rate flows like the 4 Mb/s video
  flow Brian Rosen has mentioned.  A TFRC/DCCP flow could probably
  slow-start up to 4 Mb/s in 7 or 8 round-trip times, in the absence of
  loss.

  You might also have to slow-start after a quiet period, but this could be
  addressed by the application by sending some probe traffic.  NOT 2x the
  nominal rate; 1/2 the nominal rate, or perhaps some more clever scheme.

  If the DCCP authors need to solve the quiet-period problem, then we will
  try to solve it.  But the current choice -- either send 1/2 the nominal
  rate during the quiet period, or slow-start after the quiet period --
  doesn't seem like a killer for all real-time media distribution.

* Say there's loss.

  Well, performance already sucks by Axiom 1 :)  UDP would have suffered the
  same loss.  The problem is whether or not DCCP reduces its rate,
  violating Axiom 2.

  TFRC by design reduces its rate far more slowly than TCP would.  A single
  loss will not greatly affect the rate.  TFRC seems well-protected against
  probing TCP flows and/or a low background loss rate.  We will allow
  applications to send at slightly above the TFRC rate as well.  So
  probably the situation is not that bad.

  Despite this, though, TFRC may reduce the allowed sending rate to below
  the application's nominal rate.  The application can react to this
  however it would like.  For instance, it could stop sending, which I
  think is the only other best-effort alternative on the table.  But if the
  application *wanted* to, it could try to do better.

So, I think No, DCCP would not kill Internet real-time media distribution.

If these arguments aren't sound, we'd really like to hear it.

Eddie

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



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Subject: Re: [AVT] Would DCCP kill Internet real-time media
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On Tue, 02 Dec 2003 17:32:38 -0800
 Eddie Kohler <kohler@icir.org> wrote:
> By "real-time media", I mean "current more-or-less-CBR real-time media".
> 
> To summarize the problems that a best-effort media stream might suffer:
> (Please correct me if I'm wrong or have left anything out!)
> 
> Axiom 1: Loss sucks.  ("a 3% loss rate means a 30% reduction in quality")
> Axiom 2: Rate changes suck.  ("Have you ever actually listened ... stream
>          with varying rate? ... It sucks, to put it mildly.")
> 
> So, following this through:
> 
> * Say there's no loss.
> 
>   If there's no loss, DCCP and UDP are equivalent, except that DCCP
>   enforces slow-start.  Slow-start will not matter for low-rate flows like
>   most VoIP flows.  It will for higher-rate flows like the 4 Mb/s video
>   flow Brian Rosen has mentioned.  A TFRC/DCCP flow could probably
>   slow-start up to 4 Mb/s in 7 or 8 round-trip times, in the absence of
>   loss.
> 
>   You might also have to slow-start after a quiet period, but this could be
>   addressed by the application by sending some probe traffic.  NOT 2x the
>   nominal rate; 1/2 the nominal rate, or perhaps some more clever scheme.
> 
>   If the DCCP authors need to solve the quiet-period problem, then we will
>   try to solve it.  But the current choice -- either send 1/2 the nominal
>   rate during the quiet period, or slow-start after the quiet period --
>   doesn't seem like a killer for all real-time media distribution.
> 
> * Say there's loss.
> 
>   Well, performance already sucks by Axiom 1 :)  UDP would have suffered the
>   same loss.  The problem is whether or not DCCP reduces its rate,
>   violating Axiom 2.
> 
>   TFRC by design reduces its rate far more slowly than TCP would.  A single
>   loss will not greatly affect the rate.  TFRC seems well-protected against
>   probing TCP flows and/or a low background loss rate.  We will allow
>   applications to send at slightly above the TFRC rate as well.  So
>   probably the situation is not that bad.
> 
>   Despite this, though, TFRC may reduce the allowed sending rate to below
>   the application's nominal rate.  The application can react to this
>   however it would like.  For instance, it could stop sending, which I
>   think is the only other best-effort alternative on the table.  But if the
>   application *wanted* to, it could try to do better.
> 
> So, I think No, DCCP would not kill Internet real-time media distribution.
> 
> If these arguments aren't sound, we'd really like to hear it.
> 

On the Internet2, I would estimate that a substantial fraction of the loss that I
see in multicast UDP streaming is not due to congestion, in many cases > 50%. How
would DCCP respond to such false congestion signals ? Is there a CETEN like mechanism
to deal with this ?

Regards
Marshall Eubanks


> Eddie
> 
> _______________________________________________
> 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 Dec  2 21:38:20 2003
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Subject: Re: [AVT] Would DCCP kill Internet real-time media distribution? 
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> On the Internet2, I would estimate that a substantial fraction of the
> loss that I see in multicast UDP streaming is not due to congestion, in
> many cases > 50%. How would DCCP respond to such false congestion signals
> ? Is there a CETEN like mechanism to deal with this ?

NB: DCCP is unicast.

But DCCP does in fact have a mechanism, called Data Dropped, for
differentiating congestion losses from non-congestion losses (such as
receive-buffer or corruption losses).  Non-congestion losses aren't treated
as congestion, as long as we _know_ they weren't due to congestion.

Eddie

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



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Date: Tue, 02 Dec 2003 21:03:25 -0600
From: Qiaobing Xie <Qiaobing.Xie@motorola.com>
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Subject: [AVT] Re: [dccp] Would a DCCP flow need to send 2x its preferred rate?
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> ....
> And during a silent period, the TFRC flow would have to transmit something
> so it wasn't crowded out entirely.  But this amount would be *half* the
> nominal rate, or perhaps less.  Not twice.

This may be an encouragement to the codec folks in the future to stop doing
silence detection since it would be better to always have something to send.
But, if people suddenly realize that turning off their encoder silence
detection immediately gives them better voice quality and start to do so in a
large scale, what good it would do to the network? Would IAB have the power to
tell people not to do so?

regards,
-Qiaobing


>
>
> Eddie
>
> _______________________________________________
> 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  Tue Dec  2 22:12:18 2003
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Date: Tue, 02 Dec 2003 19:11:10 -0800
From: Eddie Kohler <kohler@icir.org>
Subject: [AVT] Re: [dccp] Would a DCCP flow need to send 2x its preferred rate?
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> But, if people suddenly realize that turning off their encoder silence
> detection immediately gives them better voice quality and start to do so in a
> large scale, what good it would do to the network? Would IAB have the power to
> tell people not to do so?

TCP-friendliness without silence suppression is probably better for the
best-effort Internet than non-TCP-friendliness with silence suppression.

Eddie

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



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From: John Lazzaro <lazzaro@CS.Berkeley.EDU>
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> Eddie Kohler <kohler@icir.org> writes:
>
> Of course, UDP-penalization might not ever happen, but some "people
> working on the stability of the network" want it to.

I found the November 2003 Communications ACM article on TransLight to
be very illuminating on this point (no pun intended).  So much comes
for free if everyone gets (at least) two lamdbas to their endpoint,
one regulated QoS lamdba for media flows, and one legacy lambda for
TCP and other best-effort uses.  If you peek under Layer 2 to Layer 1,
such separation is often there physically -- the wire-pairs of modern
Ethernet, the channels of cable TV plant, the lambdas of PONS.

Of course, the TransLight folks have it even better -- unrouted,
point-to-point, knowable latency lambdas ...

---------------------------------------------------
John Lazzaro    http://www.cs.berkeley.edu/~lazzaro
     lazzaro [at] cs [dot] berkeley [dot] edu
---------------------------------------------------


_______________________________________________
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To: "Marshall Eubanks" <tme@multicasttech.com>
cc: Eddie Kohler <kohler@icir.org>, avt@ietf.org, dccp@ietf.org
Subject: Re: [dccp] Re: [AVT] Would DCCP kill Internet real-time media distribution? 
In-Reply-To: Message from "Marshall Eubanks" <tme@multicasttech.com> 
   of "Tue, 02 Dec 2003 21:32:00 EST." <web-2064542@multicasttech.com> 
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In missive <web-2064542@multicasttech.com>, "Marshall Eubanks" typed:

 >>On the Internet2, I would estimate that a substantial fraction of the loss that I
 >>see in multicast UDP streaming is not due to congestion, in many cases > 50%. How
 >>would DCCP respond to such false congestion signals ? Is there a CETEN like mechanism
 >>to deal with this ?

1/ do you mean half the time that you see loss, or that you see 50% loss? i'venever seen any loss level like that
except on a severaly broken network, where no amount of dumb CBR streamed UDP or  clever adaptive dccp will be any
use nor will TCP btw

2/ deploy ECN. that will let you cope with non congestive loss through other means - hey, >10% loss is the
Internet's way of just telling you "NO"

[its called CALL BLOCKING in phone nets, but people just dont seem to get that:-)

btw the comment on adaptive rate sucks - its correct - the usertrials folks at UCL did certainly show that there is
little point increasing the quality of video/audio if you are going to have to decrease it , within a session, as
the user experiecne contains a lot of hysteresis - i.e. they habituate, but the user reports the worst rate they
see, and variation (up from that and down to it) degrade the overall experience more (even though the average rate
may be better)....

i think this points the way (sorry to say it) to a model where we do admission control for streaming based on large
aggregate measurements and ECN, and not to rate adaption but i know this is heretical

technically, one can estimate a rate over the multiple RTT time scale via TFRC but the question is what do do about
large scale fluctuation in the load at a bottlenect ...and it seems to me that we dont have a story here yet, even
though we've been muckng about with this stuff for about 15 years...

cheers

j

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



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From: "Arnoud van Wijk" <a.vwijk@viataal.nl>
To: "'Gunnar Hellstrom'" <gunnar.hellstrom@omnitor.se>, <avt@ietf.org>,
        <paulej@packetizer.com>, <toip@snowshore.com>, <gv@trace.wisc.edu>
Subject: RE: Betr.: FW: [AVT] Missing events in RFC 2833
Date: Tue, 2 Dec 2003 23:59:49 +0100
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Hello Gunnar,
Yes. I agree.

I will give RFC 2833 a read to see what has been changed.
And we should make it clear that RFC 2833 will not mention text =
telephony
except: " For text telephony, please read RFC 2793 and the Text over IP
draft (RFC # something in the future)". Or something like that.

Greetz

Arnoud


Drs. Arnoud A. T. van Wijk=20
Viataal=20
Research & Development=20
Afdeling RDS=20
Theerestraat 42=20
5271 GD Sint-Michielsgestel=20
The Netherlands.=20
Mobile: +31651921948=20
International text telephone: +31735588408=20
-----Original Message-----
From: Gunnar Hellstrom [mailto:gunnar.hellstrom@omnitor.se]=20
Sent: dinsdag 2 december 2003 20:48
To: A vWijk; avt@ietf.org; paulej@packetizer.com; toip@snowshore.com;
gv@trace.wisc.edu
Subject: RE: Betr.: FW: [AVT] Missing events in RFC 2833

Arnoud and all,

I do not think it is desirable to try to patch up RFC2833 to transport =
all
kinds of textphone data events. In my view it is likely to be complex,
unreliable and against the efforts to create mainstream conversational
services for all with high functionality in the IP network.

But I wanted to expose this question from Tom to the text conversation
interested people to get your view, and give som good advice back to =
Tom,
who is just now cleaning up RFC2833.

You are rtight that RFC2833 started as a mechanism for DTMF transport =
and
that is what it is most famous for. But have a look in it, and you will =
find
that many other signals that are likely to appear in PSTN connections =
can be
coded with RFC2833.

If we had any wish to recreate the same tones at exit as was input, =
there
could be a case for patching up RFC2833 to (detect and) transport all
textphone data properly.

Now, the case is the opposite, we want to disconnect the technologies in =
the
both ends as much as possible, and have a pure text code in between.
Therefore, complete V.18 in the PSTN edge and RFC2793 transport is the
solution I see have a future.

Agree?

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: A vWijk [mailto:A.vWijk@viataal.nl]
Sent: Tuesday, December 02, 2003 1:45 PM
To: avt@ietf.org; gunnar.hellstrom@omnitor.se; paulej@packetizer.com;
toip@snowshore.com; gv@trace.wisc.edu
Subject: Betr.: FW: [AVT] Missing events in RFC 2833


Gunnar,
why is this desirable?

We will have T.140 over RTP and in some cases over TCP/IP.
Why the tones?

Or am I mixing things up again? RFC2833 is for DTMF signalling?

greetz

Arnoud

Drs. Arnoud A. T. van Wijk
Viataal
Research & Development
Afdeling RDS
Theerestraat 42
5271 GD Sint-Michielsgestel
The Netherlands.
Mobile: +31651921948
International text telephone: +31735588408

>>> "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se> 01-12-03 23:20 >>>
Below, Tom Taylor asks if there is any interest to complete the =
textphone
related tone events in RFC2833 so that all V.18 submodes can be =
transported
with RFC2833.

We have to verify that we have satisfying detection and transport =
methods in
RFC2793, and e.g. V.18 or the H.248.2 package descriptions, so that we =
can
leave RFC2833 half done for text transport.

I would like to do so, but then the market forces must be strong enough =
to
make RFC2793 and textphone detection commonplace.

Using RFC2833 will place a task similar to V.18 logic on the IP =
endpoint,
but with less exact information than the native V.18 has. We have no
analysis if that task is realistic, nor if it is likely to be =
implemented by
IP endpoint developers.

( The missing events for complete text transport are probably:
V.21 data in lo channel at 110 bits/s
Bell 103 hi channel data at 300 bits/s
Bell 103 lo channel data at 300 bits/s
V.23 data at 1200 bits/s
V.23 data at 75 bits/s
Baudot data at 45.45 bits/s
Baudot data at 50 bits/s

(assuming that V.8 signals and V.21 at 300 bits/s in both directions and
DTMF are covered )

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: Monday, December 01, 2003 5:37 PM
To: avt@ietf.org
Subject: [AVT] Missing events in RFC 2833


I'm finding it a little difficult to see how some of the applications
supported by RFC 2833 work without a few more events.  Here are some
examples:

1. V.18 text telephones interwork with a number of other text telephone
types.  The only event defined specifically for V.18 operation seems
rather arbitrarily chosen: the ANS2225 event for Bell 103 answer tone.
There don't seem to be any events for Baudot or V.23 or for Bell 103 in
calling mode.

2. Signalling System No. 5 has two sets of signals: line signals, which
are tones controlling trunk state, and register signals, which convey
digits with some supporting information.  There is a timing constraint
that register signalling must begin within 80 +/- 20 ms of the end of
the "seize" line signal.  Yet no event is defined for "seize", only for
the register signals.  It's a little hard to see how the receiving
gateway can propagate KPx (the initial register signal) on time if it
doesn't know that "seize" was playing, even if the sending end can
recognize and report the start of KPx within the 20 ms tolerance.  As I
see it, the present set of events is useful only to transfer the
register events from one gateway to another which will report them to
call control rather than propagating them as tones.

These missing events raise the issue of whether proposals to add events
to RFC 2833 should be backed up by requirements documents describing
their use.  I have been adding brief application descriptions in my
rewrite of 2833bis, as justification for the normative text which I have
also been adding.  (Basically just obvious stuff, missing the first time
around and highlighted by Flemming's comments earlier this year.)  I am
hoping that this additional information prompts a critical review of the
current set of 2833/2833bis events, with a view to rounding them out as
necessary to make applications work well.

Tom Taylor


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


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Gunnar,
why is this desirable?

We will have T.140 over RTP and in some cases over TCP/IP.
Why the tones?

Or am I mixing things up again? RFC2833 is for DTMF signalling?

greetz

Arnoud

Drs. Arnoud A. T. van Wijk
Viataal
Research & Development
Afdeling RDS
Theerestraat 42
5271 GD Sint-Michielsgestel
The Netherlands.
Mobile: +31651921948
International text telephone: +31735588408

>>> "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se> 01-12-03 23:20 >>>
Below, Tom Taylor asks if there is any interest to complete the textphone
related tone events in RFC2833 so that all V.18 submodes can be transported=

with RFC2833.

We have to verify that we have satisfying detection and transport methods =
in
RFC2793, and e.g. V.18 or the H.248.2 package descriptions, so that we can
leave RFC2833 half done for text transport.

I would like to do so, but then the market forces must be strong enough to
make RFC2793 and textphone detection commonplace.

Using RFC2833 will place a task similar to V.18 logic on the IP endpoint,
but with less exact information than the native V.18 has. We have no
analysis if that task is realistic, nor if it is likely to be implemented =
by
IP endpoint developers.

( The missing events for complete text transport are probably:
V.21 data in lo channel at 110 bits/s
Bell 103 hi channel data at 300 bits/s
Bell 103 lo channel data at 300 bits/s
V.23 data at 1200 bits/s
V.23 data at 75 bits/s
Baudot data at 45.45 bits/s
Baudot data at 50 bits/s

(assuming that V.8 signals and V.21 at 300 bits/s in both directions and
DTMF are covered )

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=20
Gunnar.Hellstrom@Omnitor.se=20
--------------------------------------------


-----Original Message-----
From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of Tom
Taylor
Sent: Monday, December 01, 2003 5:37 PM
To: avt@ietf.org=20
Subject: [AVT] Missing events in RFC 2833


I'm finding it a little difficult to see how some of the applications
supported by RFC 2833 work without a few more events.  Here are some
examples:

1. V.18 text telephones interwork with a number of other text telephone
types.  The only event defined specifically for V.18 operation seems
rather arbitrarily chosen: the ANS2225 event for Bell 103 answer tone.
There don't seem to be any events for Baudot or V.23 or for Bell 103 in
calling mode.

2. Signalling System No. 5 has two sets of signals: line signals, which
are tones controlling trunk state, and register signals, which convey
digits with some supporting information.  There is a timing constraint
that register signalling must begin within 80 +/- 20 ms of the end of
the "seize" line signal.  Yet no event is defined for "seize", only for
the register signals.  It's a little hard to see how the receiving
gateway can propagate KPx (the initial register signal) on time if it
doesn't know that "seize" was playing, even if the sending end can
recognize and report the start of KPx within the 20 ms tolerance.  As I
see it, the present set of events is useful only to transfer the
register events from one gateway to another which will report them to
call control rather than propagating them as tones.

These missing events raise the issue of whether proposals to add events
to RFC 2833 should be backed up by requirements documents describing
their use.  I have been adding brief application descriptions in my
rewrite of 2833bis, as justification for the normative text which I have
also been adding.  (Basically just obvious stuff, missing the first time
around and highlighted by Flemming's comments earlier this year.)  I am
hoping that this additional information prompts a critical review of the
current set of 2833/2833bis events, with a view to rounding them out as
necessary to make applications work well.

Tom Taylor


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org=20
https://www1.ietf.org/mailman/listinfo/avt=20


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To: "'Marshall Eubanks'" <tme@multicasttech.com>,
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Subject: RE: [AVT] Would DCCP kill Internet real-time media distribution?
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Hi Marshall,

> On the Internet2, I would estimate that a substantial fraction of the loss
> that I see in multicast UDP streaming is not due to congestion, in 
> many cases > 50%. How would DCCP respond to such false congestion signals
?
> Is  there a CETEN like mechanism to deal with this ?

What are the causes of these non-congestion losses?

Tom Phelan

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



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To: Arnoud van Wijk <a.vwijk@viataal.nl>
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Does this mean you also deprecate any use of V.150.1 for carrying text 
telephony?

Arnoud van Wijk wrote:
> Hello Gunnar,
> Yes. I agree.
> 
> I will give RFC 2833 a read to see what has been changed.
> And we should make it clear that RFC 2833 will not mention text telephony
> except: " For text telephony, please read RFC 2793 and the Text over IP
> draft (RFC # something in the future)". Or something like that.
> 
> Greetz
> 
> Arnoud
> 
> 
> Drs. Arnoud A. T. van Wijk 
> Viataal 
> Research & Development 
> Afdeling RDS 
> Theerestraat 42 
> 5271 GD Sint-Michielsgestel 
> The Netherlands. 
> Mobile: +31651921948 
> International text telephone: +31735588408 
> -----Original Message-----
> From: Gunnar Hellstrom [mailto:gunnar.hellstrom@omnitor.se] 
> Sent: dinsdag 2 december 2003 20:48
> To: A vWijk; avt@ietf.org; paulej@packetizer.com; toip@snowshore.com;
> gv@trace.wisc.edu
> Subject: RE: Betr.: FW: [AVT] Missing events in RFC 2833
> 
> Arnoud and all,
> 
> I do not think it is desirable to try to patch up RFC2833 to transport all
> kinds of textphone data events. In my view it is likely to be complex,
> unreliable and against the efforts to create mainstream conversational
> services for all with high functionality in the IP network.
> 
> But I wanted to expose this question from Tom to the text conversation
> interested people to get your view, and give som good advice back to Tom,
> who is just now cleaning up RFC2833.
> 
> You are rtight that RFC2833 started as a mechanism for DTMF transport and
> that is what it is most famous for. But have a look in it, and you will find
> that many other signals that are likely to appear in PSTN connections can be
> coded with RFC2833.
> 
> If we had any wish to recreate the same tones at exit as was input, there
> could be a case for patching up RFC2833 to (detect and) transport all
> textphone data properly.
> 
> Now, the case is the opposite, we want to disconnect the technologies in the
> both ends as much as possible, and have a pure text code in between.
> Therefore, complete V.18 in the PSTN edge and RFC2793 transport is the
> solution I see have a future.
> 
> Agree?
> 
> 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: A vWijk [mailto:A.vWijk@viataal.nl]
> Sent: Tuesday, December 02, 2003 1:45 PM
> To: avt@ietf.org; gunnar.hellstrom@omnitor.se; paulej@packetizer.com;
> toip@snowshore.com; gv@trace.wisc.edu
> Subject: Betr.: FW: [AVT] Missing events in RFC 2833
> 
> 
> Gunnar,
> why is this desirable?
> 
> We will have T.140 over RTP and in some cases over TCP/IP.
> Why the tones?
> 
> Or am I mixing things up again? RFC2833 is for DTMF signalling?
> 
> greetz
> 
> Arnoud
> 
> Drs. Arnoud A. T. van Wijk
> Viataal
> Research & Development
> Afdeling RDS
> Theerestraat 42
> 5271 GD Sint-Michielsgestel
> The Netherlands.
> Mobile: +31651921948
> International text telephone: +31735588408
> 
> 
>>>>"Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se> 01-12-03 23:20 >>>
> 
> Below, Tom Taylor asks if there is any interest to complete the textphone
> related tone events in RFC2833 so that all V.18 submodes can be transported
> with RFC2833.
> 
> We have to verify that we have satisfying detection and transport methods in
> RFC2793, and e.g. V.18 or the H.248.2 package descriptions, so that we can
> leave RFC2833 half done for text transport.
> 
> I would like to do so, but then the market forces must be strong enough to
> make RFC2793 and textphone detection commonplace.
> 
> Using RFC2833 will place a task similar to V.18 logic on the IP endpoint,
> but with less exact information than the native V.18 has. We have no
> analysis if that task is realistic, nor if it is likely to be implemented by
> IP endpoint developers.
> 
> ( The missing events for complete text transport are probably:
> V.21 data in lo channel at 110 bits/s
> Bell 103 hi channel data at 300 bits/s
> Bell 103 lo channel data at 300 bits/s
> V.23 data at 1200 bits/s
> V.23 data at 75 bits/s
> Baudot data at 45.45 bits/s
> Baudot data at 50 bits/s
> 
> (assuming that V.8 signals and V.21 at 300 bits/s in both directions and
> DTMF are covered )
> 
> 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: Monday, December 01, 2003 5:37 PM
> To: avt@ietf.org
> Subject: [AVT] Missing events in RFC 2833
> 
> 
> I'm finding it a little difficult to see how some of the applications
> supported by RFC 2833 work without a few more events.  Here are some
> examples:
> 
> 1. V.18 text telephones interwork with a number of other text telephone
> types.  The only event defined specifically for V.18 operation seems
> rather arbitrarily chosen: the ANS2225 event for Bell 103 answer tone.
> There don't seem to be any events for Baudot or V.23 or for Bell 103 in
> calling mode.
> 
> 2. Signalling System No. 5 has two sets of signals: line signals, which
> are tones controlling trunk state, and register signals, which convey
> digits with some supporting information.  There is a timing constraint
> that register signalling must begin within 80 +/- 20 ms of the end of
> the "seize" line signal.  Yet no event is defined for "seize", only for
> the register signals.  It's a little hard to see how the receiving
> gateway can propagate KPx (the initial register signal) on time if it
> doesn't know that "seize" was playing, even if the sending end can
> recognize and report the start of KPx within the 20 ms tolerance.  As I
> see it, the present set of events is useful only to transfer the
> register events from one gateway to another which will report them to
> call control rather than propagating them as tones.
> 
> These missing events raise the issue of whether proposals to add events
> to RFC 2833 should be backed up by requirements documents describing
> their use.  I have been adding brief application descriptions in my
> rewrite of 2833bis, as justification for the normative text which I have
> also been adding.  (Basically just obvious stuff, missing the first time
> around and highlighted by Flemming's comments earlier this year.)  I am
> hoping that this additional information prompts a critical review of the
> current set of 2833/2833bis events, with a view to rounding them out as
> necessary to make applications work well.
> 
> Tom Taylor
> 
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
> 
> 
> -
> This list is maintained by Snowshore Networks - http://www.snowshore.com
> All comments on this list are the comments of the message originators and
> Snowshore is not to be held responsible for any actions or comments found
> on this list. The archives for this list can be found at
> http://flyingfox.snowshore.com/toip_archive/maillist.html
> 
> 
> 
> 
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
> 


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



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Eddie

> * TCP is broken because it probes for available bandwidth 
> during real-time
>   media's quiet periods.
> 
>   Guilty as charged, but it might not matter.  If the 
> congested link has a
>   large level of statistical multiplexing, or if the TCP flow is
>   short-lived, it'll all come out in the wash.  On the other 
> hand, if the
>   congested link has a small level of statmux, then the 
> congested router or
>   media could just punish TCP a bit to get it back in line.
I don't think there exists mechanisms to do this, because you
need immediate response when the stream starts, and nothing
in TCP does immediate - it's designed to react slowly.
> 
>   Without a bandwidth reservation, it's hard to know how to do better.
I don't believe this.  I think it's possible to do some relatively
simple things to TCP that would help media streams quite a bit.
> 
>   Now, if the real-time media flow was using TFRC, it could 
> preserve its
>   bandwidth allocation essentially indefinitely by sending 
> probe traffic at
>   *half* its full-speed rate during silent periods (NOT twice).
This is not tenable.  No application would do this; they would
simply turn off the Voice Activity Detection.  It's also wrong.
It illustrates that indeed TCP is broken.  TCP needs a mechanism
to determine it's "fair" share and use extra bandwidth opportunistically,
returning to "fair" immediately when asked.

> 
> Other possibilities:
> 
> * TCP is broken because it probes for available bandwidth, period.
> 
>   I don't think TCP's bandwidth probes will punish TFRC unduly.
>   Simulations have been done showing that TCP and TFRC 
> achieve pretty fair
>   allocations.
TRFC, as I understand it, does not work for media flows,
which are on/off constant rate.  It is not interesting to look
at steady state; you have to include startup/shutdown. 
All the simulations I've seen either show that TRFC won't start
cleanly for CBR-like flows, or don't show how start/stop works
at all.

The problem here is "rate control"; media streams don't rate control,
except as we have discussed, rather crudely, and only downward.

> 
> * TCP is broken because it reacts so violently to loss, which 
> constrains
>   real-time media unduly.
> 
>   Guilty.  TCP does halve its rate in response to each 
> congestion event.
>   And yes, that should, up to a point, constrain application 
> designers who
>   want to run over the best-effort Internet.  But I'm not yet 
> convinced
>   that real-time media won't work.  Many of the constraints 
> don't seem that
>   onerous.  (A later mail.)
Well, I'd say this was an example where TCP is broken in that
it does not use available bandwidth appropriately, but in this case,
it is TCP that loses.  I don't want to promote solutions that
break TCP, or, as in this case, not use available bandwidth efficiently.

Brian


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



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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Eddie Kohler'" <kohler@icir.org>, avt@ietf.org
Cc: dccp@ietf.org
Subject: RE: [AVT] Would DCCP kill Internet real-time media distribution?
Date: Wed, 3 Dec 2003 09:28:49 -0500 
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> * Say there's no loss.
> 
>   If there's no loss, DCCP and UDP are equivalent, except that DCCP
>   enforces slow-start.  Slow-start will not matter for 
> low-rate flows like
>   most VoIP flows.  It will for higher-rate flows like the 4 
> Mb/s video
>   flow Brian Rosen has mentioned.  A TFRC/DCCP flow could probably
>   slow-start up to 4 Mb/s in 7 or 8 round-trip times, in the 
> absence of
>   loss.
A typical RTT on the internet these days is 50-100 ms.
If you experience 30% loss in the first 3 RTTs of voice, you will clip
the "hello" pretty badly.  If it's 7-8 RTTs, you will definitely have
a very poor experience.

I think you can pretty safely say that slow start is a very big
no-no for interactive media.  For non-interactive media, it only
annoys you a little bit, and while not desirable, isn't fatal.

Are you really proposing that we create a new protocol that only works
for small bit rate media?  I think that's a really poor idea.
Better to work harder and fix the problem for all, or nearly all
media streams now, and in the foreseeable future.  The foreseeable
future will have multimegabit media streams.

> 
>   You might also have to slow-start after a quiet period, but 
> this could be
>   addressed by the application by sending some probe traffic. 
>  NOT 2x the
>   nominal rate; 1/2 the nominal rate, or perhaps some more 
> clever scheme.
Slow start is not acceptable, as above.
A more clever scheme might work.  1/2 nominal is not realistic,
as has been pointed out.  I suspect that "more clever" means
fixing TCP.

> 
>   If the DCCP authors need to solve the quiet-period problem, 
> then we will
>   try to solve it.  But the current choice -- either send 1/2 
> the nominal
>   rate during the quiet period, or slow-start after the quiet 
> period --
>   doesn't seem like a killer for all real-time media distribution.
Maybe not, but probably enough that it isn't interesting to deploy.

> 
> * Say there's loss.
> 
>   Well, performance already sucks by Axiom 1 :)  UDP would 
> have suffered the
>   same loss.  The problem is whether or not DCCP reduces its rate,
>   violating Axiom 2.
> 
>   TFRC by design reduces its rate far more slowly than TCP 
> would.  A single
>   loss will not greatly affect the rate.  TFRC seems 
> well-protected against
>   probing TCP flows and/or a low background loss rate.  We will allow
>   applications to send at slightly above the TFRC rate as well.  So
>   probably the situation is not that bad.
This is gobbledy-gook to me.  We already have tried to point out
that you don't change the rate, except grossly, downward only,
and only once or twice.  You do that when your loss rate exceeds
some threshold.  You don't need any new mechanism to do this.

So "reducing its rate far more slowly" is not possible for the media
streams.  There is no "send at slightly above the TRFC rate".  The
rate is fixed; TRFC can't tell the media stream anything interesting.

> 
>   Despite this, though, TFRC may reduce the allowed sending 
> rate to below
>   the application's nominal rate.  The application can react to this
>   however it would like.  For instance, it could stop sending, which I
>   think is the only other best-effort alternative on the 
> table.  But if the
>   application *wanted* to, it could try to do better.
Same issues.  There is no "nominal rate" there is only the fixed rate.

> 
> So, I think No, DCCP would not kill Internet real-time media 
> distribution.
> 
> If these arguments aren't sound, we'd really like to hear it.
I don't think they are sound:
	Slow start is not acceptable
	TRFC is not useful/acceptable
	There is only very limited rate change, we already have the
		mechanisms to do it

DCCP isn't solving any problem that UDP has for media streams.

Brian 

_______________________________________________
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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>, avt@ietf.org
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Subject: RE: [dccp] RE: [AVT] Would DCCP kill Internet real-time media dis
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Hi Brian,

>> * Say there's no loss.
>> 
>>   If there's no loss, DCCP and UDP are equivalent, except that DCCP
>>   enforces slow-start.  Slow-start will not matter for 
>> low-rate flows like
>>   most VoIP flows.  It will for higher-rate flows like the 4 
>> Mb/s video
>>   flow Brian Rosen has mentioned.  A TFRC/DCCP flow could probably
>>   slow-start up to 4 Mb/s in 7 or 8 round-trip times, in the 
>> absence of
>>   loss.
> A typical RTT on the internet these days is 50-100 ms.
> If you experience 30% loss in the first 3 RTTs of voice, you will clip
> the "hello" pretty badly.  If it's 7-8 RTTs, you will definitely have
> a very poor experience.

Are you saying that an interactive video session can't afford a
pre-connection training period?  It seems to me that all such systems that
I've used have a several second delay at the start of the connection during
which several things are negotiated.

Voice-only connections have a stricter expectation of post-dial delay, but
the four packets per RTT at the beginning of slow start is almost always
enough for telephony-quality voice, so there is in practice no slow start.

Are you also saying that you expect an app that sends at 4M bps to
immediately change from 0 bps to 4M bps?

Tom Phelan

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



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Huh?  You seem to be saying that if you have a lot of TCP streams on a
link
they won't use all of the available bandwidth.  I thought just the
opposite.
With a low level of multiplexing there's more chance of synchronizing
losses
and cutbacks, and not fully utilizing the link.  But with a large number
of
flows enough of them will be blasting away while others are cut back.

> If the congested link has low levels of statmux -- like one 
> DCCP flow and
> one TCP flow -- then yes, the TCP would try for the available 
> bandwidth.
> And during a silent period, the TFRC flow would have to 
> transmit something
> so it wasn't crowded out entirely.  But this amount would be 
> *half* the
> nominal rate, or perhaps less.  Not twice.

So this relates to our previous conversation?  When you're transmitting
at
less than the TFRC allowed rate, your allowed rate could be twice what
you're transmitting.  If a loss event happens, your allowed rate is cut
back, but since it was twice what you were actually transmitting, no
effect.
This gives you the effective "protection" of transmitting at twice your
necessary rate, without actually doing it.

Great if it works.  A simulation would be nice :-).

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



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



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From: "Qiong Li" <qli@brainmediaco.com>
To: <avt@ietf.org>
Cc: <dccp@ietf.org>
Subject: RE: [AVT] RE: [dccp] Would a DCCP flow need to send 2x its preferred rate?
Date: Wed, 3 Dec 2003 10:34:07 -0500
Organization: BrainMedia
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Sorry for the previous message. I was intending to make comments on the
following posting regarding "TFRC allowed rate".

--------------------------------------------------------------
...... When you're transmitting at
less than the TFRC allowed rate, your allowed rate could be twice what
you're transmitting.  If a loss event happens, your allowed rate is cut
back, but since it was twice what you were actually transmitting, no
effect.
This gives you the effective "protection" of transmitting at twice your
necessary rate, without actually doing it.

.......
----------------------------------------------------

Since TRFC allowed rate (R0) is estimated from observed loss rate (L) by
the DCCP flow, and the two are positively correlated, R0 may be
interpreted and used in different ways:

- If the DCCP flow can tolerate a loss rate > L, it can consider to
increase its sending rate. As long as its sending rate R < R0, it will
not be considered to be unfair to other TCP flows. It does not
necessarily means that when R < R0, it is always good to increase R.
Doing that could result higher loss rate when the end-to-end path is
lightly multiplexed with TCP flows.

- If the estimated R0 is reduced, even that it is still higher than R,
keeping the current sending rate for the DCCP flow may imply the flow
will experience higher loss rate.

The point: the adaptation of DCCP flow rate may need to combine the
considerations of both R0 and L, depending on the codec characteristics.
When the decoder can only perform error concealment under certain
pre-designed loss rate, then the L should dominate the adaptation,
assuming scalable media content is used, which may imply that R0 could
become irrelevant under some circumstances.  

Qiong Li



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



From exim@www1.ietf.org  Wed Dec  3 10:56:28 2003
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Phelan, Tom'" <tphelan@sonusnet.com>, avt@ietf.org
Cc: dccp@ietf.org
Subject: RE: [dccp] RE: [AVT] Would DCCP kill Internet real-time media dis
	 tribution?
Date: Wed, 3 Dec 2003 10:53:48 -0500 
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> Are you saying that an interactive video session can't afford a
> pre-connection training period?  It seems to me that all such 
> systems that
> I've used have a several second delay at the start of the 
> connection during
> which several things are negotiated.
There are indeed a couple of RTTs for signalling and codec 
negotiation.  There could be some ms of codec initialization, then 
data starts full steam. In most video codecs, it's actually worse, 
because you usually start with an I frame, that has the least 
compression.  You end up buffering this away often to force the data 
to be more CBR like.  We don't have this problem.

Our video codec has an 8 frame (~250 ms) start time.  That's around 
2-5 RTTs. The data is flying at the CBR rate right away, but we 
don't display the video until we get around 8 frames into the 
sequence.  If you delay the arrival of the 8 frame @ 4Mbit data, 
we'll have to delay the start of video.  It's already annoying, 
because it's slower than the voice.

> 
> Voice-only connections have a stricter expectation of 
> post-dial delay, but
> the four packets per RTT at the beginning of slow start is 
> almost always
> enough for telephony-quality voice, so there is in practice 
> no slow start.
What I said was that if you have 30% loss for 4 RTTs and you 
listen to that, you will clip the "hello".  Normally, you send 
RTP at the negotiated rate, and you play what you get.

You could force more buffering before playback, which only 
increases PDD. Increases in PDD are measured in seconds.  
At 100ms, you need 10 RTTs to get an extra second.  That would 
only be interesting if you don't experience loss during this period.
However, there is a hard upper limit.  You can't delay the voice
more than 150 ms mouth to ear.  The codec packetization period
eats up a lot of that (40 ms), and the network eats up a lot
(25-50ms typ one way, but usually you engineer for more). 
You don't get 4 RTTs of jitter buffering, that's for sure.
If you are filling and emptying a buffer at a fixed rate
(less jitter on the network), you can't build some buffer
into the beginning of the stream and expect to bleed it
off later.  Once you have buffered up bits, you are stuck
with that buffer +/- jitter. 

I don't have a lot of experience using TCP on media streams, 
so I'm not really sure what happens in practice.  In UDP, the 
transmitter starts up sending at the fixed, negotiated rate.  
The receiver starts by filling the jitter buffer.  When the 
buffer is "half" full, it starts playing at the fixed, negotiated 
rate. Filling the buffer is part of PDD, and it can cover some 
ugliness in actual arrival rate, but I don't think it covers what 
slow start actually does.  Again, I really don't have enough 
experience to know for sure.  
> 
> Are you also saying that you expect an app that sends at 4M bps to
> immediately change from 0 bps to 4M bps?
Yep; bang! 0 to 4Mbit in two packets.  Actually, there is some 
buffering at both ends in our system, but much less than you expect.  
We spent a LOT of time getting rid of buffering to make it work well.  
The entire video chain has less than 90 ms delay (flash a led in the 
camera, measure it on the screen) including camera and screen sync,
encode/decode and all buffering.  The video delay determines the 
audio delay in order to get lip sync.  This is a very important 
characteristic it turns out, because it allows you to argue :) 
Not every system works like ours, for sure.  Every video codec
I know either has a burst at the beginning at higher than nominal
rate, or has immediate CBR.

Brian

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



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Subject: RE: [AVT] Would DCCP kill Internet real-time media distribution?
To: "Rosen, Brian" <Brian.Rosen@marconi.com>
Cc: avt@ietf.org, dccp@ietf.org
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Rosen you wrote:

>So "reducing its rate far more slowly" is not possible for the media
>streams.  There is no "send at slightly above the TRFC rate".  The
>rate is fixed; TRFC can't tell the media stream anything interesting.

I though this was already resolved ?

case A: you have recent codecs and you *can* change the rate,
(only of course not drastically due to the quality differences
already mentioned, but again systems with moderate rate changes
have been deployed and running for years in MPEG-2 DTV )

case B: your codec is strict CBR and then you need to tweak something
so that this CBR stream is transmitted with some variable amount of garbage
(this seems like an horrible shame to me but it seems also it would work)

Is that correct ?

regards,

Philippe Gentric
Chief Architect
Philips Software
philippe.gentric@philips.com
http://www.software.philips.com



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



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To: "Paul E. Jones" <paulej@packetizer.com>
CC: A vWijk <A.vWijk@viataal.nl>, avt@ietf.org, gunnar.hellstrom@omnitor.se,
        toip@snowshore.com, gv@trace.wisc.edu
Subject: Re: Betr.: FW: [AVT] Missing events in RFC 2833
References: <sfcc9778.053@mailserver.viataal.nl> <00c801c3b8f5$40ed2ed0$0401a8c0@berlin>
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The existing events in 2833 have to do primarily with startup.  I have 
to ask how the TOIP people see startup negotiation working without 2833 
support.  You still have to deal with V.8 or V.8bis in the more advanced 
terminals.

There is the fact that two V.18 terminals in conversation phase use 
V.21, and V.21 events are included in the 2833 repertoire.  This means 
that telephone events could be used during conversation phase in some 
cases.  However, the basic discussion is over the set of events needed 
to achieve transition to conversation phase successfully.

Turning to the conversational phase, I do not see how the use of RFC 
2793 between gateways in place of the various alternatives (G.711 VBD, 
V.150.1 modem relay, the tones payload type, V.21 or other events) will 
affect the user experience.  I assume the reason this is desired is to 
simplify the transition to packet-mode terminals.

The problem here is that gateways have to handle a wide variety of 
traffic, not just text.  This makes more generic approaches like V.150.1 
for voice-band data simpler to implement, so I see some vendors making 
that choice.  Obviously, if packet terminals require RFC 2793 and 
gateways cannot handle it, deployment of these gateways is limited to 
bridging of PSTN segments.  Given that this application still represents 
a substantial market segment, we should probably add the 2833 events 
needed to handle it properly.

Paul E. Jones wrote:

> Arnoud,
> 
> RFC 2833 can be used to convey tone information (frequencies and durations),
> as well as tone events.  It's not restricted to DTMF.
> 
> RFC 2833 could potentially be used as an alternative to RFC 2793.  A gateway
> or IP phone, for example, could simply examine the tones and convey those as
> "descriptions" (either frequency and duration information or event codes)
> over an IP network.  The other end could then receive those descriptions and
> reproduce the tones.  They would not have to understand the underlying
> textphone protocol.
> 
> The problem with RFC 2833, as I see it, is that it stands at odds with the
> desire of folks on the ToIP mailing list to try to remove textphone
> signaling from the IP network.  There has been a strong desire expressed to
> terminate that textphone signaling at the gateway and only use text on the
> IP network.  Doing that, arguably, would make interfacing with IP-based IVR
> systems, IP-based voice/text-mail systems, or other IP-based systems much
> easier.  I say "arguably" because some have suggested to me that it's very
> simple to write software that understand the textphone signals received in
> RFC 2833 packets... though I personally think it puts an unfortunate burden
> on a Pocket PC, for example.
> 
> To this point, neither the ITU nor the IETF has considered RFC 2833 as an
> option or alternative to RFC 2793.  I think this is probably an important
> topic that we should take up immediately, as both bodies are forging ahead
> with plans based on RFC 2793 only.
> 
> Paul
> 
> ----- Original Message ----- 
> From: "A vWijk" <A.vWijk@viataal.nl>
> To: <avt@ietf.org>; <gunnar.hellstrom@omnitor.se>; <paulej@packetizer.com>;
> <toip@snowshore.com>; <gv@trace.wisc.edu>
> Sent: Tuesday, December 02, 2003 7:45 AM
> Subject: Betr.: FW: [AVT] Missing events in RFC 2833
> 
> 
> 
>>Gunnar,
>>why is this desirable?
>>
>>We will have T.140 over RTP and in some cases over TCP/IP.
>>Why the tones?
>>
>>Or am I mixing things up again? RFC2833 is for DTMF signalling?
>>
>>greetz
>>
>>Arnoud
>>
>>Drs. Arnoud A. T. van Wijk
>>Viataal
>>Research & Development
>>Afdeling RDS
>>Theerestraat 42
>>5271 GD Sint-Michielsgestel
>>The Netherlands.
>>Mobile: +31651921948
>>International text telephone: +31735588408
>>
>>
>>>>>"Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se> 01-12-03 23:20 >>>
>>
>>Below, Tom Taylor asks if there is any interest to complete the textphone
>>related tone events in RFC2833 so that all V.18 submodes can be
> 
> transported
> 
>>with RFC2833.
>>
>>We have to verify that we have satisfying detection and transport methods
> 
> in
> 
>>RFC2793, and e.g. V.18 or the H.248.2 package descriptions, so that we can
>>leave RFC2833 half done for text transport.
>>
>>I would like to do so, but then the market forces must be strong enough to
>>make RFC2793 and textphone detection commonplace.
>>
>>Using RFC2833 will place a task similar to V.18 logic on the IP endpoint,
>>but with less exact information than the native V.18 has. We have no
>>analysis if that task is realistic, nor if it is likely to be implemented
> 
> by
> 
>>IP endpoint developers.
>>
>>( The missing events for complete text transport are probably:
>>V.21 data in lo channel at 110 bits/s
>>Bell 103 hi channel data at 300 bits/s
>>Bell 103 lo channel data at 300 bits/s
>>V.23 data at 1200 bits/s
>>V.23 data at 75 bits/s
>>Baudot data at 45.45 bits/s
>>Baudot data at 50 bits/s
>>
>>(assuming that V.8 signals and V.21 at 300 bits/s in both directions and
>>DTMF are covered )
>>
>>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: Monday, December 01, 2003 5:37 PM
>>To: avt@ietf.org
>>Subject: [AVT] Missing events in RFC 2833
>>
>>
>>I'm finding it a little difficult to see how some of the applications
>>supported by RFC 2833 work without a few more events.  Here are some
>>examples:
>>
>>1. V.18 text telephones interwork with a number of other text telephone
>>types.  The only event defined specifically for V.18 operation seems
>>rather arbitrarily chosen: the ANS2225 event for Bell 103 answer tone.
>>There don't seem to be any events for Baudot or V.23 or for Bell 103 in
>>calling mode.
>>
>>2. Signalling System No. 5 has two sets of signals: line signals, which
>>are tones controlling trunk state, and register signals, which convey
>>digits with some supporting information.  There is a timing constraint
>>that register signalling must begin within 80 +/- 20 ms of the end of
>>the "seize" line signal.  Yet no event is defined for "seize", only for
>>the register signals.  It's a little hard to see how the receiving
>>gateway can propagate KPx (the initial register signal) on time if it
>>doesn't know that "seize" was playing, even if the sending end can
>>recognize and report the start of KPx within the 20 ms tolerance.  As I
>>see it, the present set of events is useful only to transfer the
>>register events from one gateway to another which will report them to
>>call control rather than propagating them as tones.
>>
>>These missing events raise the issue of whether proposals to add events
>>to RFC 2833 should be backed up by requirements documents describing
>>their use.  I have been adding brief application descriptions in my
>>rewrite of 2833bis, as justification for the normative text which I have
>>also been adding.  (Basically just obvious stuff, missing the first time
>>around and highlighted by Flemming's comments earlier this year.)  I am
>>hoping that this additional information prompts a critical review of the
>>current set of 2833/2833bis events, with a view to rounding them out as
>>necessary to make applications work well.
>>
>>Tom Taylor
>>
>>
>>_______________________________________________
>>Audio/Video Transport Working Group
>>avt@ietf.org
>>https://www1.ietf.org/mailman/listinfo/avt
>>
>>
>>-
>>This list is maintained by Snowshore Networks - http://www.snowshore.com
>>All comments on this list are the comments of the message originators and
>>Snowshore is not to be held responsible for any actions or comments found
>>on this list. The archives for this list can be found at
>>http://flyingfox.snowshore.com/toip_archive/maillist.html
>>
> 
> 
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
> 


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



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Subject: Re: [dccp] Re: [AVT] Would DCCP kill Internet real-time media distribution? 
In-Reply-To: Your message of "Wed, 03 Dec 2003 07:46:53 GMT."
             <E1ARRio-0008HU-00@wisbech.cl.cam.ac.uk> 
Date: Wed, 03 Dec 2003 08:18:28 -0800
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> btw the comment on adaptive rate sucks - its correct - the usertrials
> folks at UCL did certainly show that there is little point increasing the
> quality of video/audio if you are going to have to decrease it , within a
> session, as the user experiecne contains a lot of hysteresis - i.e. they
> habituate, but the user reports the worst rate they see, and variation
> (up from that and down to it) degrade the overall experience more (even
> though the average rate may be better)....

This is surely a comment about *one* system/codec, or one family -- other
people on the list have mentioned that some codecs behave better?

E

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



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Subject: Re: [AVT] Is TCP broken? 
In-Reply-To: Message from "Rosen, Brian" <Brian.Rosen@marconi.com> 
   of "Wed, 03 Dec 2003 09:10:17 EST." <313680C9A886D511A06000204840E1CF070B6162@whq-msgusr-02.pit.comms.marconi.com> 
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> >   if the congested link has a small level of statmux, then the
> >   congested router or media could just punish TCP a bit to get it back
> >   in line.
> I don't think there exists mechanisms to do this, because you
> need immediate response when the stream starts, and nothing
> in TCP does immediate - it's designed to react slowly.

? TCP is designed to react *quickly*.  It halves its rate after a single
dropped packet (after an RTT of course).

> >   Without a bandwidth reservation, it's hard to know how to do better.
> I don't believe this.  I think it's possible to do some relatively
> simple things to TCP that would help media streams quite a bit.

Such as?  (1) I don't think the situation is that bad; (2) it seems like
any modification to TCP to cause it to probe slower for excess bandwidth
would cut to TCP's heart.

> >   Now, if the real-time media flow was using TFRC, it could preserve
> >   its bandwidth allocation essentially indefinitely by sending probe
> >   traffic at *half* its full-speed rate during silent periods (NOT
> >   twice).
> This is not tenable.  No application would do this; they would
> simply turn off the Voice Activity Detection.  It's also wrong.
> It illustrates that indeed TCP is broken.  TCP needs a mechanism
> to determine it's "fair" share and use extra bandwidth opportunistically,
> returning to "fair" immediately when asked.

I disagree with each of these points.  Would 'no' application really do
this?  If an application was given the choice of (1) not sending over the
best-effort internet or (2) sending traffic during silence periods, why
wouldn't the application choose (2)?  And why wouldn't an application that
sent at a reduced rate during silence periods have a competitive advantage
(assuming people cared about bandwidth)?

TCP has a mechanism to determine its fair share, namely bandwidth probing.
It has a mechanism to adjust to the introduction of new flows, namely
halving its rate after single loss periods.  TFRC is robust against
bandwidth probing.

You want your media flow to start up from nothing, after a silent period of
seconds or minutes, with no loss.  That is not best effort.  That is a
bandwidth reservation.

I don't understand how this would work even in an environment with only
media streams.  If the sum of the streams' non-silent bandwidth is greater
than the link bandwidth, then you may end up with loss when all the streams
aren't silent.  What should the streams do then?  Suicide?  Live with high
loss rates?  Rely on the users to go away?  If the network has been
engineered so that this situation cannot happen, then it's a QoS network.

> >   I don't think TCP's bandwidth probes will punish TFRC unduly.
> >   Simulations have been done showing that TCP and TFRC achieve pretty
> >   fair allocations.
> TRFC, as I understand it, does not work for media flows,
> which are on/off constant rate.  It is not interesting to look
> at steady state; you have to include startup/shutdown. 
> All the simulations I've seen either show that TRFC won't start
> cleanly for CBR-like flows, or don't show how start/stop works
> at all.

Not all media streams are on/off constant rate as people have brought up,
but yes, slow-start exists and could be problematic for CBR users.  But I
don't understand why this is the complete killer it's being painted to be.
Not for all media flows anyway.

Is delivering media over the current Internet important enough that users
could afford to wait < 1 second for slow start?  (Which, incidentally, is
entirely within the range application startup delay times.)  If the answer
is no, fine; use QoS.

Eddie

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



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Subject: Re: [AVT] Would DCCP kill Internet real-time media distribution? 
In-Reply-To: Message from "Rosen, Brian" <Brian.Rosen@marconi.com> 
   of "Wed, 03 Dec 2003 09:28:49 EST." <313680C9A886D511A06000204840E1CF070B6163@whq-msgusr-02.pit.comms.marconi.com> 
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> A typical RTT on the internet these days is 50-100 ms.
> If you experience 30% loss in the first 3 RTTs of voice, you will clip
> the "hello" pretty badly.  If it's 7-8 RTTs, you will definitely have
> a very poor experience.

Hm.  I think users are pretty used to startup delays of far more than 1
second when making telephone calls.  The application can mask this easily.

Is the sense of the avt group really that slow start is completely
unacceptable for any important media flows?

> This is gobbledy-gook to me.  We already have tried to point out
> that you don't change the rate, except grossly, downward only,
> and only once or twice.  You do that when your loss rate exceeds
> some threshold.  You don't need any new mechanism to do this.
> 
> So "reducing its rate far more slowly" is not possible for the media
> streams.  There is no "send at slightly above the TRFC rate".  The
> rate is fixed; TRFC can't tell the media stream anything interesting.

First, not all applications cannot change the rate, as people have
mentioned.

Second, TFRC provides you with a reference rate.  For fail-stop CBR
applications, this reference rate could be taken as an indication of when
the application must stop.  When I said "the rate changes slowly", it was
simply to point out that TFRC changes its reference rate slowly.  Thus, a
fail-stop CBR application would not be forced to stop after a small number
of losses -- only after the loss rate really changed.

Eddie

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



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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Eddie Kohler'" <kohler@icir.org>
Cc: Jon Crowcroft <Jon.Crowcroft@cl.cam.ac.uk>, avt@ietf.org
Subject: RE: [dccp] Re: [AVT] Would DCCP kill Internet real-time media dis
	tribution? 
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I have never heard/seen a codec that can dynamically, in real time,
vary its rate in response to network conditions in a way that
is not objectionable to the user.

I have seen/heard a lot of codecs, but clearly, not all, or most,
or even a significant percentage on live networks.

Your mileage may vary.

I'm dimly aware of what Philippe is talking about, but I don't
think it does what the first paragraph above describes today.  
It does hint that it might be possible to build such a thing,
but I don't know what the tradeoffs are.  For example, what is
the effect on delay?  It's certainly not in the realm of practical
deployment in the near future for interactive media. I don't
think it's practical for deployment in the near future for
non-interactive media, but I don't know as much about that
world as I do the interactive world.

Brian

> -----Original Message-----
> From: Eddie Kohler [mailto:kohler@icir.org]
> Sent: Wednesday, December 03, 2003 11:18 AM
> To: Jon Crowcroft
> Cc: avt@ietf.org
> Subject: Re: [dccp] Re: [AVT] Would DCCP kill Internet real-time media
> distribution? 
> 
> 
> > btw the comment on adaptive rate sucks - its correct - the 
> usertrials
> > folks at UCL did certainly show that there is little point 
> increasing the
> > quality of video/audio if you are going to have to decrease 
> it , within a
> > session, as the user experiecne contains a lot of 
> hysteresis - i.e. they
> > habituate, but the user reports the worst rate they see, 
> and variation
> > (up from that and down to it) degrade the overall 
> experience more (even
> > though the average rate may be better)....
> 
> This is surely a comment about *one* system/codec, or one 
> family -- other
> people on the list have mentioned that some codecs behave better?
> 
> E
> 
> _______________________________________________
> 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  Wed Dec  3 12:13:19 2003
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To: "Rosen, Brian" <Brian.Rosen@marconi.com>, avt@ietf.org, dccp@ietf.org
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   of "Wed, 03 Dec 2003 08:43:01 PST." <200312031643.hB3Gh1sD016017@coyote.icir.org> 
Date: Wed, 03 Dec 2003 09:12:45 -0800
From: Eddie Kohler <kohler@icir.org>
Subject: [AVT] Re: Would DCCP kill Internet real-time media distribution?
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> Is the sense of the avt group really that slow start is completely
> unacceptable for any important media flows?

Also, for what it's worth, I do understand and sympathize with the problems
that slow start can cause a truly-CBR application.  I don't think
slow-start is all peaches and cream.

The problem is, Best Effort Public Internet == Slow Start.

There are proposals to eliminate the need for slow-start through explicit
feedback from routers (XCP, QuickStart), but they're a ways off.  Maybe
media will be the app that pushes them through Cisco?  Significant TCP
changes are probably even less likely.  (They're certainly less likely
without any proposal on the board.)

In the meantime, we really do have a situation where a media application
should either do slow start or something similar, or it should stay off the
public internet.  Brian, I'm coming to the conclusion that your app should
stay off the public internet -- or you may need to change the app so that
it will work best off the public internet, when it can get a true bandwidth
reservation.  Maybe that's OK for how your app is used in practice?

Eddie

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



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 tribution?
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Multi-party, non-multicast (central server) conferences should not be 
forgotten in this discussion. They complicate life in two ways:

- The penalty of not using silence suppression gets much larger. If you 
have a conference with 100 participants, a modern implementation would 
have one or two active inbound stream and 100 outbound (mixed) streams. 
If applications need to keep sending data to keep their place, suddenly 
inbound data volume will be multiplied by a factor of between 50 
(transmitting at half rate) and 100.

This gets worse for the typical video-follows-audio setup.

- Similarly, for a useful interactive conference, participants have to 
be able to interject and speak up immediately, with the video stream 
following instantaneously. They cannot wait multiple round-trip times as 
they'll never get a word in edgewise. Thus, there is no masking of 
ramp-up delays if you want a reasonable user experience.

(Similar considerations apply to the now very popular push-to-talk 
services. Trying to sell a service where the user has to push, wait and 
then talk does not seem like a winning proposition.)

Masking is also rather tricky for two-party audio. There was a long 
discussion in the SIP working group on this topic a few years ago, 
leading to fairly elaborate hand-offs between reservation and session 
signaling to minimize the post-pickup delay. As soon as I hit the 
'accept call' button, I need to be able to say 'hello' and be heard by 
the other side. The only masking opportunity is to start sending audio 
data 'to warm up the line' while the phone is ringing, adding more 
bandwidth consumption for the large number of calls that don't succeed. 
At $4/MB for mobile voice data, this is unlikely to be popular. (Typical 
ring-to-answer delays are about 20 seconds; average call durations for 
mobile are probably three minutes, so this immediately adds about 10% 
overhead, not counting the 60% or so of calls that fail over to 
voicemail after 30 seconds.)

I believe that there now is a suggested initial TCP windows size > 1. Is 
this correct? (If you figure a maximum acceptable network delay of 100 
ms and a typical audio/video packet spacing of 50 ms, you would be 
sending about 4 packets per RTT.)


Henning


_______________________________________________
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: "'qli@brainmediaco.com'" <qli@brainmediaco.com>, avt@ietf.org
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Subject: RE: [AVT] RE: [dccp] Would a DCCP flow need to send 2x its prefer
	red rate?
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Hi Qiong,

See inline ...

Tom Phelan

> --------------------------------------------------------------
> ...... When you're transmitting at
> less than the TFRC allowed rate, your allowed rate could be twice what
> you're transmitting.  If a loss event happens, your allowed 
> rate is cut
> back, but since it was twice what you were actually transmitting, no
> effect.
> This gives you the effective "protection" of transmitting at 
> twice your
> necessary rate, without actually doing it.
> 
> .......
> ----------------------------------------------------
> 
> Since TRFC allowed rate (R0) is estimated from observed loss 
> rate (L) by
> the DCCP flow, and the two are positively correlated, R0 may be
> interpreted and used in different ways:
> 
> - If the DCCP flow can tolerate a loss rate > L, it can consider to
> increase its sending rate. As long as its sending rate R < R0, it will
> not be considered to be unfair to other TCP flows. It does not
> necessarily means that when R < R0, it is always good to increase R.
> Doing that could result higher loss rate when the end-to-end path is
> lightly multiplexed with TCP flows.
> 
> - If the estimated R0 is reduced, even that it is still higher than R,
> keeping the current sending rate for the DCCP flow may imply the flow
> will experience higher loss rate.

If the loss rate persists, TFRC will continue to reduce R0 until it's below
R.  At that time the app could reduce its rate, or discard packets until the
loss rate is too great.

But this isn't necessarily a persistent loss situation.  If the loss was
induced by other congestion controlled streams probing for more capacity,
then they should back off and the losses should stop.  It should be
persistent only when there are enough competing streams that each stream's
share is less than R.

> 
> The point: the adaptation of DCCP flow rate may need to combine the
> considerations of both R0 and L, depending on the codec 
> characteristics.
> When the decoder can only perform error concealment under certain
> pre-designed loss rate, then the L should dominate the adaptation,
> assuming scalable media content is used, which may imply that R0 could
> become irrelevant under some circumstances.  
> 
> Qiong Li
> 
> 
> 
> _______________________________________________
> 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: Qiaobing Xie <Qiaobing.Xie@motorola.com>
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Eddie Kohler wrote:

> > But, if people suddenly realize that turning off their encoder silence
> > detection immediately gives them better voice quality and start to do so in a
> > large scale, what good it would do to the network? Would IAB have the power to
> > tell people not to do so?
>
> TCP-friendliness without silence suppression is probably better for the
> best-effort Internet than non-TCP-friendliness with silence suppression.
>

Eddie,

I wish I could be as optimistic as you. I wonder if anyone on the list has some
numbers that can show what percentage of additional media data will be injected into
the network if codecs would stop doing silence detection (motion detection for
video).

Fundamentally, I am concerned that this DCCP "app-who-sends-more-will-get-rewarded"
feature will be taken advantage of in the wrong way (the draft that starts this
discussion is the first attempt for this, but it won't be the last).

If this behavior of "sending filler data when no real data to send" becomes a
recommended practice for media-over-DCCP (as you and Tom seem to suggest), what
would stop the same recommendation be made for app over TCP? For an app over a long
lived TCP connection, the sender can also benefit from sending "filler data" to keep
the pipe from collapsing, so as to avoid the penalty of going through another slow
start when it has real data to send next time. But I think this is hardly something
that should be recommended for TCP based apps. Why should it be more acceptable for
DCCP then?

regards,
-Qiaobing


>
> Eddie


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



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To: Arnoud van Wijk <a.vwijk@viataal.nl>
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OK, so in North America the gateway always emulates a Baudot terminal. 
That means it won't be possible to negotiate alternating text and voice 
with a V.18 terminal.  I imagine Gunnar's bitter response to that one is 
that nobody is making V.18 terminals anyway.

Arnoud van Wijk wrote:

> I think the question we need answered is:
> " Do we need to transport real-timing streaming text (aka Interactive Text)
> through PSTNs with RFC 2833 (a beefed up version of it)?"
> 
> For Interactive Texting, gateways it will be necessary to allow backward
> compatibility with analogue textphones. Transcoding from T.140/RTP into the
> local legacy textphone protocol. For America it is then Boudot.
> All that the gateway has to do is simply connect to a PSTN using Boudot over
> the voice channel. Every PSTN connects with an Interactive Text gateway. And
> that IT gateway does the transcoding, so that the PSTN sees it as a voice
> channel as it is done today.
> 
> There is then no RFC 2833 involved for the Interactive Texting.
> 
> Thus when we get 2 PSTN's that are linked together over IP for a voice call,
> then is it much easier to send the text as RFC 2793.
> So, the only requirement is then that every PSTN must have a means to
> transcode RFC 2793 into local analogue text telephony (directly or via an IT
> gateway).
> 
> I want to have clarity on this before starting to complicate matters even
> more. Analogue text telephony is a dying beast. And my goal is to kill it as
> fast as possible without any problems.
> 
> Greetz
> 
> Arnoud
> 
> Drs. Arnoud A. T. van Wijk 
> Viataal 
> Research & Development 
> Afdeling RDS 
> Theerestraat 42 
> 5271 GD Sint-Michielsgestel 
> The Netherlands. 
> Mobile: +31651921948 
> International text telephone: +31735588408 



_______________________________________________
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: "'Qiaobing Xie'" <Qiaobing.Xie@motorola.com>,
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Subject: [AVT] RE: [dccp] Would a DCCP flow need to send 2x its preferred rate?
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Hi Qiaobing,

It appears that it isn't necessary to send extra data to get the
"protection" I was seeking.  I was erroneously assuming that when TFRC
reduced the send rate, it reduced from what you were actually sending, not
from what you were allowed to send.  Given this I think that the
recommendation in my draft was unnecessary, and I'll be modifying it.

If DCCP can do it why can't TCP?  In the particular case I was talking
about, the DCCP app was sending less than it was "allowed" to.  It seemed
reasonable to put some (but not necessarily all) of that excess allowance to
good use.  Most TCP apps always send at the maximum allowed rate, so padding
wouldn't be possible.  If a TCP app was satisfied with less than the allowed
rate, I suppose it could "pad" to preserve that rate.  But it's difficult to
imagine such a restrained app using TCP with its abrupt rate variations and
in-order delivery.  It seems to me that these types of apps are better
served by DCCP/TFRC (or they just ignore congestion issues and use UDP).

Tom Phelan


> -----Original Message-----
> From: Qiaobing Xie [mailto:Qiaobing.Xie@motorola.com]
> Sent: Wednesday, December 03, 2003 12:39 PM
> To: Eddie Kohler
> Cc: avt@ietf.org; dccp@ietf.org
> Subject: Re: [dccp] Would a DCCP flow need to send 2x its preferred
> rate?
> 
> 
> Eddie Kohler wrote:
> 
> > > But, if people suddenly realize that turning off their 
> encoder silence
> > > detection immediately gives them better voice quality and 
> start to do so in a
> > > large scale, what good it would do to the network? Would 
> IAB have the power to
> > > tell people not to do so?
> >
> > TCP-friendliness without silence suppression is probably 
> better for the
> > best-effort Internet than non-TCP-friendliness with silence 
> suppression.
> >
> 
> Eddie,
> 
> I wish I could be as optimistic as you. I wonder if anyone on 
> the list has some
> numbers that can show what percentage of additional media 
> data will be injected into
> the network if codecs would stop doing silence detection 
> (motion detection for
> video).
> 
> Fundamentally, I am concerned that this DCCP 
> "app-who-sends-more-will-get-rewarded"
> feature will be taken advantage of in the wrong way (the 
> draft that starts this
> discussion is the first attempt for this, but it won't be the last).
> 
> If this behavior of "sending filler data when no real data to 
> send" becomes a
> recommended practice for media-over-DCCP (as you and Tom seem 
> to suggest), what
> would stop the same recommendation be made for app over TCP? 
> For an app over a long
> lived TCP connection, the sender can also benefit from 
> sending "filler data" to keep
> the pipe from collapsing, so as to avoid the penalty of going 
> through another slow
> start when it has real data to send next time. But I think 
> this is hardly something
> that should be recommended for TCP based apps. Why should it 
> be more acceptable for
> DCCP then?
> 
> regards,
> -Qiaobing
> 
> 
> >
> > Eddie
> 
> 
> _______________________________________________
> 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>
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> > Is the sense of the avt group really that slow start is completely
> > unacceptable for any important media flows?
> 
> Also, for what it's worth, I do understand and sympathize 
> with the problems
> that slow start can cause a truly-CBR application.  I don't think
> slow-start is all peaches and cream.
> 
> The problem is, Best Effort Public Internet == Slow Start.
Where does it say that?

Right now, we send UDP.  It actually works pretty well, but it
clearly doesn't work if you have a bunch of TCP sources gobbling
up the bandwidth.  DCCP is not going to change the effect of TCP
on the media, it's going to change the effect of the media on
TCP.  We don't need no DCCP.  "Fair" to media is all or nothing,
Philippe's ideas notwithstanding.  Nothing is fair if the alternative 
is garbage. 

If your goal is to prevent what is today mythical congestion
collapse, then we would be much better served by talking
about how signalling can get loss rates so that it can shut
down flows.  That's simple, and it won't screw anything up.

Brian

> 
> There are proposals to eliminate the need for slow-start 
> through explicit
> feedback from routers (XCP, QuickStart), but they're a ways 
> off.  Maybe
> media will be the app that pushes them through Cisco?  Significant TCP
> changes are probably even less likely.  (They're certainly less likely
> without any proposal on the board.)
I'll agree with this, although I'm also willing to work on solutions.

> 
> In the meantime, we really do have a situation where a media 
> application
> should either do slow start or something similar, or it 
> should stay off the
> public internet.  Brian, I'm coming to the conclusion that 
> your app should
> stay off the public internet -- or you may need to change the 
> app so that
> it will work best off the public internet, when it can get a 
> true bandwidth
> reservation.  Maybe that's OK for how your app is used in practice?
Sorry, but as you might imagine, I don't agree. See the note from
Henning.  We use 4M, while other systems might use 768K.  That's no
reason to have any difference in protocols.  Slow start is not
a practical thing for interactive use, sorry.

_______________________________________________
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: "'Henning Schulzrinne'" <hgs@cs.columbia.edu>
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Yes yes.

Another example is Push-To-Talk

Brian

> -----Original Message-----
> From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
> Sent: Wednesday, December 03, 2003 12:19 PM
> To: Rosen, Brian
> Cc: 'Eddie Kohler'; avt@ietf.org
> Subject: Re: [dccp] Re: [AVT] Would DCCP kill Internet real-time media
> dis tribution?
> 
> 
> Multi-party, non-multicast (central server) conferences should not be 
> forgotten in this discussion. They complicate life in two ways:
> 
> - The penalty of not using silence suppression gets much 
> larger. If you 
> have a conference with 100 participants, a modern 
> implementation would 
> have one or two active inbound stream and 100 outbound 
> (mixed) streams. 
> If applications need to keep sending data to keep their 
> place, suddenly 
> inbound data volume will be multiplied by a factor of between 50 
> (transmitting at half rate) and 100.
> 
> This gets worse for the typical video-follows-audio setup.
> 
> - Similarly, for a useful interactive conference, 
> participants have to 
> be able to interject and speak up immediately, with the video stream 
> following instantaneously. They cannot wait multiple 
> round-trip times as 
> they'll never get a word in edgewise. Thus, there is no masking of 
> ramp-up delays if you want a reasonable user experience.
> 
> (Similar considerations apply to the now very popular push-to-talk 
> services. Trying to sell a service where the user has to 
> push, wait and 
> then talk does not seem like a winning proposition.)
> 
> Masking is also rather tricky for two-party audio. There was a long 
> discussion in the SIP working group on this topic a few years ago, 
> leading to fairly elaborate hand-offs between reservation and session 
> signaling to minimize the post-pickup delay. As soon as I hit the 
> 'accept call' button, I need to be able to say 'hello' and be 
> heard by 
> the other side. The only masking opportunity is to start 
> sending audio 
> data 'to warm up the line' while the phone is ringing, adding more 
> bandwidth consumption for the large number of calls that 
> don't succeed. 
> At $4/MB for mobile voice data, this is unlikely to be 
> popular. (Typical 
> ring-to-answer delays are about 20 seconds; average call 
> durations for 
> mobile are probably three minutes, so this immediately adds about 10% 
> overhead, not counting the 60% or so of calls that fail over to 
> voicemail after 30 seconds.)
> 
> I believe that there now is a suggested initial TCP windows 
> size > 1. Is 
> this correct? (If you figure a maximum acceptable network 
> delay of 100 
> ms and a typical audio/video packet spacing of 50 ms, you would be 
> sending about 4 packets per RTT.)
> 
> 
> Henning
> 

_______________________________________________
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From exim@www1.ietf.org  Wed Dec  3 16:03:27 2003
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To: "Tom Taylor" <taylor@nortelnetworks.com>,
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References: <000201c3b9bd$451e7f90$0300a8c0@solstice> <3FCE294A.20008@nortelnetworks.com>
Subject: Re: Betr.: FW: [AVT] Missing events in RFC 2833
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Tom,

I don't think we can say that North American GWs only speak Baudot.  We must
allow a V.21 user to call home to England, for example.  Your point about
the use of RFC 2833 for start-up messages is valid and is likely to occur
between two gateways, as they would have negotiated the use of certain RFC
2833 tones and event messages.  Such description falls into the scope of
ITU-T V.ToIP.

However, if a gateway happened to terminate to a Pocket PC, for example, no
such tones would be negotiated or used.  The gateway could either "eat"
those tones or assume it's a textphone device.  The behavior needs to be
specified somewhere... perhaps also in V.ToIP?

In any case, I think we're in agreement that RFC 2833 should not be used as
a text transport mechanism.  It might be used as part of start-up exchanges
before it can be determined that the calling/called device is a modem, fax,
or TTY device.  GW-GW calls have slightly special requirements that IP
phones do not have.

Paul

----- Original Message ----- 
From: "Tom Taylor" <taylor@nortelnetworks.com>
To: "Arnoud van Wijk" <a.vwijk@viataal.nl>
Cc: "'Paul E. Jones'" <paulej@packetizer.com>; <avt@ietf.org>;
<gunnar.hellstrom@omnitor.se>; <toip@snowshore.com>; <gv@trace.wisc.edu>
Sent: Wednesday, December 03, 2003 1:19 PM
Subject: Re: Betr.: FW: [AVT] Missing events in RFC 2833


> OK, so in North America the gateway always emulates a Baudot terminal.
> That means it won't be possible to negotiate alternating text and voice
> with a V.18 terminal.  I imagine Gunnar's bitter response to that one is
> that nobody is making V.18 terminals anyway.
>
> Arnoud van Wijk wrote:
>
> > I think the question we need answered is:
> > " Do we need to transport real-timing streaming text (aka Interactive
Text)
> > through PSTNs with RFC 2833 (a beefed up version of it)?"
> >
> > For Interactive Texting, gateways it will be necessary to allow backward
> > compatibility with analogue textphones. Transcoding from T.140/RTP into
the
> > local legacy textphone protocol. For America it is then Boudot.
> > All that the gateway has to do is simply connect to a PSTN using Boudot
over
> > the voice channel. Every PSTN connects with an Interactive Text gateway.
And
> > that IT gateway does the transcoding, so that the PSTN sees it as a
voice
> > channel as it is done today.
> >
> > There is then no RFC 2833 involved for the Interactive Texting.
> >
> > Thus when we get 2 PSTN's that are linked together over IP for a voice
call,
> > then is it much easier to send the text as RFC 2793.
> > So, the only requirement is then that every PSTN must have a means to
> > transcode RFC 2793 into local analogue text telephony (directly or via
an IT
> > gateway).
> >
> > I want to have clarity on this before starting to complicate matters
even
> > more. Analogue text telephony is a dying beast. And my goal is to kill
it as
> > fast as possible without any problems.
> >
> > Greetz
> >
> > Arnoud
> >
> > Drs. Arnoud A. T. van Wijk
> > Viataal
> > Research & Development
> > Afdeling RDS
> > Theerestraat 42
> > 5271 GD Sint-Michielsgestel
> > The Netherlands.
> > Mobile: +31651921948
> > International text telephone: +31735588408
>
>
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
>


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



From exim@www1.ietf.org  Thu Dec  4 03:49:26 2003
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Subject: [AVT] Mixers, and echo cancellation
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All,

  Have a query about RTP mixing, and echo
cancellation/prevention in the mixed stream during playback.

Am currently considering a problem of payload
identification/separation in conferences, where multiple
speakers are active at an instant of time.

 To elaborate, the output of a mixer generally does not have any
input stream specific information(apart from the SSRC that it
carries as the CSRC).It appears to me that the only means of
preventing an echo during playback would be: 
a.) The input stream(s) are buffered at the mixer and 
subtracted out from the mixed stream, thereby creating N+1
streams, where N is the number of active speakers. 
b.) The input streams are buffered at the endpoints (or at a
different mixer in the case of cascaded mixers) and are
subtracted out from the mixed stream that comes back.In the
event of using cascaded mixers, we "might" generate lesser
number of streams than option (a).

 However,in case (b), it would be necessary to know which
stream, from the set of mixed streams  that comes back for
playback is to be used for the subtraction. Again it appears to
me that, the only way it could be done, is by tagging a header
extension to the stream, buffering it at the endpoint, create a
mixed stream with the same header extension and then subtracting
it out before playback, by comparing the extensions. Needless to
say, the scheme is complex and has slim chances of working ,
considering network delays, the fact that all endpoints (or
mixers in the case of cascaded mixers )might need to be sync'ed,
and that the header extension would have a high overhead.

Just wanted to know if anybody else has considered this problem,
and if you would you be aware of any other mechanism that would
identify a stream in a mixed stream?
(--- Or is it that I am missing something ??)

Also this brings me to my next question -- is it possible that
in some fashion, a set of cascaded mixers could be synchronized
so that there would be just one timing trigger that could
signal
the start of mixing operation at all mixers?


Would be glad if someone could comment on my queries,

Thanks in advance,
B.


________________________________________________
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From: "Arnoud van Wijk" <a.vwijk@viataal.nl>
To: "'Tom Taylor'" <taylor@nortelnetworks.com>,
        "'Paul E. Jones'" <paulej@packetizer.com>
Cc: <avt@ietf.org>, <gunnar.hellstrom@omnitor.se>, <toip@snowshore.com>,
        <gv@trace.wisc.edu>
Subject: RE: Betr.: FW: [AVT] Missing events in RFC 2833
Date: Wed, 3 Dec 2003 17:48:23 +0100
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I think the question we need answered is:
" Do we need to transport real-timing streaming text (aka Interactive =
Text)
through PSTNs with RFC 2833 (a beefed up version of it)?"

For Interactive Texting, gateways it will be necessary to allow backward
compatibility with analogue textphones. Transcoding from T.140/RTP into =
the
local legacy textphone protocol. For America it is then Boudot.
All that the gateway has to do is simply connect to a PSTN using Boudot =
over
the voice channel. Every PSTN connects with an Interactive Text gateway. =
And
that IT gateway does the transcoding, so that the PSTN sees it as a =
voice
channel as it is done today.

There is then no RFC 2833 involved for the Interactive Texting.

Thus when we get 2 PSTN's that are linked together over IP for a voice =
call,
then is it much easier to send the text as RFC 2793.
So, the only requirement is then that every PSTN must have a means to
transcode RFC 2793 into local analogue text telephony (directly or via =
an IT
gateway).

I want to have clarity on this before starting to complicate matters =
even
more. Analogue text telephony is a dying beast. And my goal is to kill =
it as
fast as possible without any problems.

Greetz

Arnoud

Drs. Arnoud A. T. van Wijk=20
Viataal=20
Research & Development=20
Afdeling RDS=20
Theerestraat 42=20
5271 GD Sint-Michielsgestel=20
The Netherlands.=20
Mobile: +31651921948=20
International text telephone: +31735588408=20
-----Original Message-----
From: Tom Taylor [mailto:taylor@nortelnetworks.com]=20
Sent: woensdag 3 december 2003 16:58
To: Paul E. Jones
Cc: A vWijk; avt@ietf.org; gunnar.hellstrom@omnitor.se; =
toip@snowshore.com;
gv@trace.wisc.edu
Subject: Re: Betr.: FW: [AVT] Missing events in RFC 2833

The existing events in 2833 have to do primarily with startup.  I have=20
to ask how the TOIP people see startup negotiation working without 2833=20
support.  You still have to deal with V.8 or V.8bis in the more advanced =

terminals.

There is the fact that two V.18 terminals in conversation phase use=20
V.21, and V.21 events are included in the 2833 repertoire.  This means=20
that telephone events could be used during conversation phase in some=20
cases.  However, the basic discussion is over the set of events needed=20
to achieve transition to conversation phase successfully.

Turning to the conversational phase, I do not see how the use of RFC=20
2793 between gateways in place of the various alternatives (G.711 VBD,=20
V.150.1 modem relay, the tones payload type, V.21 or other events) will=20
affect the user experience.  I assume the reason this is desired is to=20
simplify the transition to packet-mode terminals.

The problem here is that gateways have to handle a wide variety of=20
traffic, not just text.  This makes more generic approaches like V.150.1 =

for voice-band data simpler to implement, so I see some vendors making=20
that choice.  Obviously, if packet terminals require RFC 2793 and=20
gateways cannot handle it, deployment of these gateways is limited to=20
bridging of PSTN segments.  Given that this application still represents =

a substantial market segment, we should probably add the 2833 events=20
needed to handle it properly.

Paul E. Jones wrote:

> Arnoud,
>=20
> RFC 2833 can be used to convey tone information (frequencies and
durations),
> as well as tone events.  It's not restricted to DTMF.
>=20
> RFC 2833 could potentially be used as an alternative to RFC 2793.  A
gateway
> or IP phone, for example, could simply examine the tones and convey =
those
as
> "descriptions" (either frequency and duration information or event =
codes)
> over an IP network.  The other end could then receive those =
descriptions
and
> reproduce the tones.  They would not have to understand the underlying
> textphone protocol.
>=20
> The problem with RFC 2833, as I see it, is that it stands at odds with =
the
> desire of folks on the ToIP mailing list to try to remove textphone
> signaling from the IP network.  There has been a strong desire =
expressed
to
> terminate that textphone signaling at the gateway and only use text on =
the
> IP network.  Doing that, arguably, would make interfacing with =
IP-based
IVR
> systems, IP-based voice/text-mail systems, or other IP-based systems =
much
> easier.  I say "arguably" because some have suggested to me that it's =
very
> simple to write software that understand the textphone signals =
received in
> RFC 2833 packets... though I personally think it puts an unfortunate
burden
> on a Pocket PC, for example.
>=20
> To this point, neither the ITU nor the IETF has considered RFC 2833 as =
an
> option or alternative to RFC 2793.  I think this is probably an =
important
> topic that we should take up immediately, as both bodies are forging =
ahead
> with plans based on RFC 2793 only.
>=20
> Paul
>=20
> ----- Original Message -----=20
> From: "A vWijk" <A.vWijk@viataal.nl>
> To: <avt@ietf.org>; <gunnar.hellstrom@omnitor.se>;
<paulej@packetizer.com>;
> <toip@snowshore.com>; <gv@trace.wisc.edu>
> Sent: Tuesday, December 02, 2003 7:45 AM
> Subject: Betr.: FW: [AVT] Missing events in RFC 2833
>=20
>=20
>=20
>>Gunnar,
>>why is this desirable?
>>
>>We will have T.140 over RTP and in some cases over TCP/IP.
>>Why the tones?
>>
>>Or am I mixing things up again? RFC2833 is for DTMF signalling?
>>
>>greetz
>>
>>Arnoud
>>
>>Drs. Arnoud A. T. van Wijk
>>Viataal
>>Research & Development
>>Afdeling RDS
>>Theerestraat 42
>>5271 GD Sint-Michielsgestel
>>The Netherlands.
>>Mobile: +31651921948
>>International text telephone: +31735588408
>>
>>
>>>>>"Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se> 01-12-03 23:20 >>>
>>
>>Below, Tom Taylor asks if there is any interest to complete the =
textphone
>>related tone events in RFC2833 so that all V.18 submodes can be
>=20
> transported
>=20
>>with RFC2833.
>>
>>We have to verify that we have satisfying detection and transport =
methods
>=20
> in
>=20
>>RFC2793, and e.g. V.18 or the H.248.2 package descriptions, so that we =
can
>>leave RFC2833 half done for text transport.
>>
>>I would like to do so, but then the market forces must be strong =
enough to
>>make RFC2793 and textphone detection commonplace.
>>
>>Using RFC2833 will place a task similar to V.18 logic on the IP =
endpoint,
>>but with less exact information than the native V.18 has. We have no
>>analysis if that task is realistic, nor if it is likely to be =
implemented
>=20
> by
>=20
>>IP endpoint developers.
>>
>>( The missing events for complete text transport are probably:
>>V.21 data in lo channel at 110 bits/s
>>Bell 103 hi channel data at 300 bits/s
>>Bell 103 lo channel data at 300 bits/s
>>V.23 data at 1200 bits/s
>>V.23 data at 75 bits/s
>>Baudot data at 45.45 bits/s
>>Baudot data at 50 bits/s
>>
>>(assuming that V.8 signals and V.21 at 300 bits/s in both directions =
and
>>DTMF are covered )
>>
>>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: Monday, December 01, 2003 5:37 PM
>>To: avt@ietf.org
>>Subject: [AVT] Missing events in RFC 2833
>>
>>
>>I'm finding it a little difficult to see how some of the applications
>>supported by RFC 2833 work without a few more events.  Here are some
>>examples:
>>
>>1. V.18 text telephones interwork with a number of other text =
telephone
>>types.  The only event defined specifically for V.18 operation seems
>>rather arbitrarily chosen: the ANS2225 event for Bell 103 answer tone.
>>There don't seem to be any events for Baudot or V.23 or for Bell 103 =
in
>>calling mode.
>>
>>2. Signalling System No. 5 has two sets of signals: line signals, =
which
>>are tones controlling trunk state, and register signals, which convey
>>digits with some supporting information.  There is a timing constraint
>>that register signalling must begin within 80 +/- 20 ms of the end of
>>the "seize" line signal.  Yet no event is defined for "seize", only =
for
>>the register signals.  It's a little hard to see how the receiving
>>gateway can propagate KPx (the initial register signal) on time if it
>>doesn't know that "seize" was playing, even if the sending end can
>>recognize and report the start of KPx within the 20 ms tolerance.  As =
I
>>see it, the present set of events is useful only to transfer the
>>register events from one gateway to another which will report them to
>>call control rather than propagating them as tones.
>>
>>These missing events raise the issue of whether proposals to add =
events
>>to RFC 2833 should be backed up by requirements documents describing
>>their use.  I have been adding brief application descriptions in my
>>rewrite of 2833bis, as justification for the normative text which I =
have
>>also been adding.  (Basically just obvious stuff, missing the first =
time
>>around and highlighted by Flemming's comments earlier this year.)  I =
am
>>hoping that this additional information prompts a critical review of =
the
>>current set of 2833/2833bis events, with a view to rounding them out =
as
>>necessary to make applications work well.
>>
>>Tom Taylor
>>
>>
>>_______________________________________________
>>Audio/Video Transport Working Group
>>avt@ietf.org
>>https://www1.ietf.org/mailman/listinfo/avt
>>
>>
>>-
>>This list is maintained by Snowshore Networks - =
http://www.snowshore.com
>>All comments on this list are the comments of the message originators =
and
>>Snowshore is not to be held responsible for any actions or comments =
found
>>on this list. The archives for this list can be found at
>>http://flyingfox.snowshore.com/toip_archive/maillist.html
>>
>=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



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Subject: RE: [dccp] Re: [AVT] Would DCCP kill Internet real-time media dis	tribution?
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Rosan you wrote:

>I'm dimly aware of what Philippe is talking about, but I don't
>think it does what the first paragraph above describes today.

have a look at things like:

http://www.nextream-online.com/nextream/documents/pdf/WP%20FLEXTREAM.pdf

the type of rate change these things do is pretty flexible, they actually change the rate
several times *inside one video frame*  and the range of changes can be pretty large.
quality is "tunable" and .... well just switch on your TV and look !

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: =?iso-8859-1?Q?Jos=E9_Rey?= <rey@panasonic.de>
To: <avt@ietf.org>
Cc: "Colin Perkins \(E-mail\)" <csp@csperkins.org>,
        "David Leon \(E-mail\)" <David.Leon@nokia.com>,
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        <hakenberg@panasonic.de>
Subject: RE: [AVT] Retransmission draft
Date: Thu, 4 Dec 2003 14:34:56 +0100
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Hi Colin, all,

in the past weeks I had some intensive internal discussions on the matter of
the potential IPRs on the RTP retransmission draft. As a WG participant I
perfectly understand and respect the comments and interests expressed by you
and others of the AVT group. In order to keep the good working spirit in AVT
it is my strong intention to adress these comments in the best possible way
by taking the interests of all parties involved in this work into account.

Matsushita has published an IPR statement that it is prepared to grant a
license on reasonable, nondiscriminatory terms and conditions (in accordance
with paragraph 10.3.3 of the RFC 2026) on any part that is included into
the RTP retransmission standard and that Matsushita possibly holds IPRs on.
Providing a royality-free license to open-source developers will
unfortunately lead to a situation where Matsushita will lose any rights on
their IPRs in most cases. Taking into account the work and effort (we
started with the retransmission draft in AVT more than 2.5 years ago) we
spent into this draft, this is not acceptable for Matsushita.

On the other hand, we believe that the situation for open-source developers
is quite easy to solve by other means. First of all they are not in any way
'forced' to implement RTP retransmission. It is just an optional repair
mechanism for the RTP protocol, as described in RFC2354. In order to solve
the problems that RTP retransmission tries to solve, there are other
solutions available from the AVT group which are not encumbered by IPRs,
e.g. RFC2733 and draft-ietf-avt-ulp-08.txt. This makes the implementation
of the RTP retransmission solution purely optional and easily avoidable to
everyone.

In order to make this situation clear to anybody who reads the RTP
retransmission draft, we suggest to add a paragraph at the very beginning of
the draft. This paragraph will make implementors aware of the situation that
the solution described in this draft is possibly encumbered by IPRs.
Furthermore it will point implementors to non-IPR-encumbered solutions as
above, which may perfectly be used instead of RTP retransmission.

Finally and FYI: 3GPP is aware of the IPR situation of the RTP
retransmission draft and has accepted the solution provided in
draft-ietf-avt-rtp-retransmission-09.txt as working assumption in their
packet based streaming service (PSS). The acceptance is conditional on the
fact that the draft will proceed on standards track and an RFC is available
before the release of the PSS specifications, (expected for March 2004).

Looking forward to your comments,

Jose

> -----Original Message-----
> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of Colin
> Perkins
> Sent: Tuesday, October 21, 2003 6:58 PM
> To: rey@panasonic.de
> Cc: avt@ietf.org; 'Magnus Westerlund'
> Subject: Re: [AVT] Retransmission draft
>
>
> On Tuesday, Oct 21, 2003, at 12:54 Europe/London, Jose Rey wrote:
> ...
> > In previous emails we have been asked to explain our patent
> > application and
> > we
> > informed AVT that the patent application itself specifies a
transport
> > method
> > using the RTP protocol where two sequence numbers are used: one to
> > specifiy
> > the transmission order (=RTP SN) and the other one defining the
> > reprodution
> > order, which in the draft corresponds with the OSN (original
sequence
> > number). We also stated that it is not possible to implement the
> > protocol
> > without infringing on the patent.
> > So far we have not received any more questions on the patent, but
are
> > happy
> > to provide them in case needed.
> >
> > On the issue of how to proceed with this RTP retransmission draft
some
> > opinions have been provided by Magnus Westerlund, Lars-Erik Jonsson
and
> > John Lazzaro. John Lazzaro (08.09.03) has suggested to publish it as
> > Informational if the patent rights are held while Magnus and
> > Lars-Erik have suggested to proceed with the document as a Proposed
> > Standard (as we understand from the emails on from 22.09.03 and
> > 09.09.03).
> >
> > Taking all the above into account and considering that
> > independent of the patent situation this work is intended to be
> > included in
> > the next release of 3GPP, we would like to ask the group to proceed
> > with the
> > draft on standards track.
> >
> > cheers,
> > Jose
>
> [takes working group chair hat off]
>
> 3GPP is only one forum which uses IETF standards, and one where
dealing
> with IPR issues is commonplace. Having patented standards is more of
an
> issue for open source developers, who are often excluded by the
> licensing terms. Would Matsushita be willing to clarify if the
> "reasonable, nondiscriminatory" terms allow open source
implementations
> to use the retransmission format (this will probably require a
> Royalty-Free license)?
>
> [puts working group chair hat on again]
>
> --
> Colin Perkins
> http://csperkins.org/
>
>
> _______________________________________________
> 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: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'philippe.gentric@philips.com'" <philippe.gentric@philips.com>
Cc: avt@ietf.org
Subject: RE: [dccp] Re: [AVT] Would DCCP kill Internet real-time media dis
		tribution?
Date: Thu, 4 Dec 2003 09:08:58 -0500 
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I think this proves my point.
>I have never heard/seen a codec that can dynamically, in real time,
>vary its rate in response to network conditions in a way that
>is not objectionable to the user.

This product cannot dynamically vary its rate in response to network
conditions.  It appears to vary rates of multiple streams to keep
the aggregate within a fixed rate, which indeed, as you say, may
portend the possibility of dynamically varying the rate in response
to network conditions.  The algorithm, as described, probably is
not suitable for interactive streams, because of delay considerations,
as I suggested might be a concern.

If we had a reasonable amount of experience using techniques such as
this one in real networks, then I might have to change my opinion.
As it stands, this is pretty far from what I would need to
change our product, or to get me to believe that any substantial
part of the VoIP market would deploy such algorithms any time soon.
If you and Eddie want to keep slogging on DCCP for this kind of
codec, please continue, but I'm still in the skeptical stage.

Brian 

> -----Original Message-----
> From: philippe.gentric@philips.com 
> [mailto:philippe.gentric@philips.com]
> Sent: Thursday, December 04, 2003 4:58 AM
> To: Rosen, Brian
> Cc: avt@ietf.org
> Subject: RE: [dccp] Re: [AVT] Would DCCP kill Internet real-time media
> dis tribution?
> 
> 
> 
> Rosan you wrote:
> 
> >I'm dimly aware of what Philippe is talking about, but I don't
> >think it does what the first paragraph above describes today.
> 
> have a look at things like:
> 
> http://www.nextream-online.com/nextream/documents/pdf/WP%20FLE
XTREAM.pdf

the type of rate change these things do is pretty flexible, they actually
change the rate
several times *inside one video frame*  and the range of changes can be
pretty large.
quality is "tunable" and .... well just switch on your TV and look !

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: [AVT] RFC 2833 tones payload and modulation
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The RFC 2833 tones payload type provides for both summation of 
frequencies and modulation by another frequency.  However, the 
modulation is missing a piece of information, because modulation of 
tones in practice isn't just a matter of multiplication.

My particular example is the V.8 ANSam tone.  This is modulated by a 15 
Hz envelope.  However, the specification of that envelope says that the 
amplitude of the total ANSam waveform varies from a minimum of 80% of 
the nominal value to 120%.

Obviously the payload description has been long established and it would 
be difficult to add the missing information now.  I see two alternatives:
   -- Recommend that modulation be applied the way it is for ANSam (i.e. 
with precisely the same 80% and 120% limits).  I'd justify this on the 
grounds that ANSam will be the most common case.  The other cases are 
tones intended for human consumption, so my second justifying argument 
is that using the ANSam boundaries might be close enough to what the 
tones have to sound like.
   -- Alternatively, recommend that the receiving gateway be aware of 
the nature of any tone presented with modulation, so it can construct 
the right envelope.

Any opinions, or is this something no one has implemented anyway?


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



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Date: Thu, 4 Dec 2003 14:59:24 +0000
From: Colin Perkins <csp@csperkins.org>
To: <rey@panasonic.de>
Cc: avt@ietf.org
Subject: Re: [AVT] Retransmission draft
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Jos=E9,

[cc list trimmed]

--> Jos=E9 Rey <rey@panasonic.de> writes:
> in the past weeks I had some intensive internal discussions on the matter
> of the potential IPRs on the RTP retransmission draft. As a WG
> participant I perfectly understand and respect the comments and interests
> expressed by you and others of the AVT group. In order to keep the good
> working spirit in AVT it is my strong intention to adress these comments
> in the best possible way by taking the interests of all parties involved
> in this work into account.
>=20
> Matsushita has published an IPR statement that it is prepared to grant a
> license on reasonable, nondiscriminatory terms and conditions (in
> accordance with paragraph 10.3.3 of the RFC 2026) on any part that is
> included into the RTP retransmission standard and that Matsushita
> possibly holds IPRs on. Providing a royality-free license to open-source
> developers will unfortunately lead to a situation where Matsushita will
> lose any rights on their IPRs in most cases. Taking into account the work
> and effort (we started with the retransmission draft in AVT more than 2.5
> years ago) we spent into this draft, this is not acceptable for
> Matsushita.
>=20
> On the other hand, we believe that the situation for open-source
> developers is quite easy to solve by other means. First of all they are
> not in any way'forced' to implement RTP retransmission. It is just an
> optional repair mechanism for the RTP protocol, as described in RFC2354.
> In order to solve the problems that RTP retransmission tries to solve,
> there are other solutions available from the AVT group which are not
> encumbered by IPRs, e.g. RFC2733 and draft-ietf-avt-ulp-08.txt. This
> makes the implementation of the RTP retransmission solution purely
> optional and easily avoidable to everyone.

Provided it is clear that the mechanism is encumbered, yes.

> In order to make this situation clear to anybody who reads the RTP
> retransmission draft, we suggest to add a paragraph at the very beginning
> of the draft. This paragraph will make implementors aware of the
> situation that the solution described in this draft is possibly
> encumbered by IPRs. Furthermore it will point implementors to
> non-IPR-encumbered solutions as above, which may perfectly be used
> instead of RTP retransmission.

Without seeing the wording of this paragraph, I can't give a definitive
answer, but that would largely address my concerns.

--=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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To: "Paul E. Jones" <paulej@packetizer.com>
CC: Arnoud van Wijk <a.vwijk@viataal.nl>, avt@ietf.org,
        gunnar.hellstrom@omnitor.se, toip@snowshore.com, gv@trace.wisc.edu
Subject: Re: Betr.: FW: [AVT] Missing events in RFC 2833
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Based on this intervention, I would propose to add the list of events 
given by Gunnar to those already given in RFC 2833.

Paul E. Jones wrote:

> Tom,
> 
> I don't think we can say that North American GWs only speak Baudot.  We must
> allow a V.21 user to call home to England, for example.  Your point about
> the use of RFC 2833 for start-up messages is valid and is likely to occur
> between two gateways, as they would have negotiated the use of certain RFC
> 2833 tones and event messages.  Such description falls into the scope of
> ITU-T V.ToIP.
> 
> However, if a gateway happened to terminate to a Pocket PC, for example, no
> such tones would be negotiated or used.  The gateway could either "eat"
> those tones or assume it's a textphone device.  The behavior needs to be
> specified somewhere... perhaps also in V.ToIP?
> 
> In any case, I think we're in agreement that RFC 2833 should not be used as
> a text transport mechanism.  It might be used as part of start-up exchanges
> before it can be determined that the calling/called device is a modem, fax,
> or TTY device.  GW-GW calls have slightly special requirements that IP
> phones do not have.
> 
> Paul
> 
> ----- Original Message ----- 
> From: "Tom Taylor" <taylor@nortelnetworks.com>
> To: "Arnoud van Wijk" <a.vwijk@viataal.nl>
> Cc: "'Paul E. Jones'" <paulej@packetizer.com>; <avt@ietf.org>;
> <gunnar.hellstrom@omnitor.se>; <toip@snowshore.com>; <gv@trace.wisc.edu>
> Sent: Wednesday, December 03, 2003 1:19 PM
> Subject: Re: Betr.: FW: [AVT] Missing events in RFC 2833
> 
> 
> 
>>OK, so in North America the gateway always emulates a Baudot terminal.
>>That means it won't be possible to negotiate alternating text and voice
>>with a V.18 terminal.  I imagine Gunnar's bitter response to that one is
>>that nobody is making V.18 terminals anyway.
>>
>>Arnoud van Wijk wrote:
>>
>>
>>>I think the question we need answered is:
>>>" Do we need to transport real-timing streaming text (aka Interactive
> 
> Text)
> 
>>>through PSTNs with RFC 2833 (a beefed up version of it)?"
>>>
>>>For Interactive Texting, gateways it will be necessary to allow backward
>>>compatibility with analogue textphones. Transcoding from T.140/RTP into
> 
> the
> 
>>>local legacy textphone protocol. For America it is then Boudot.
>>>All that the gateway has to do is simply connect to a PSTN using Boudot
> 
> over
> 
>>>the voice channel. Every PSTN connects with an Interactive Text gateway.
> 
> And
> 
>>>that IT gateway does the transcoding, so that the PSTN sees it as a
> 
> voice
> 
>>>channel as it is done today.
>>>
>>>There is then no RFC 2833 involved for the Interactive Texting.
>>>
>>>Thus when we get 2 PSTN's that are linked together over IP for a voice
> 
> call,
> 
>>>then is it much easier to send the text as RFC 2793.
>>>So, the only requirement is then that every PSTN must have a means to
>>>transcode RFC 2793 into local analogue text telephony (directly or via
> 
> an IT
> 
>>>gateway).
>>>
>>>I want to have clarity on this before starting to complicate matters
> 
> even
> 
>>>more. Analogue text telephony is a dying beast. And my goal is to kill
> 
> it as
> 
>>>fast as possible without any problems.
>>>
>>>Greetz
>>>
>>>Arnoud
>>>
>>>Drs. Arnoud A. T. van Wijk
>>>Viataal
>>>Research & Development
>>>Afdeling RDS
>>>Theerestraat 42
>>>5271 GD Sint-Michielsgestel
>>>The Netherlands.
>>>Mobile: +31651921948
>>>International text telephone: +31735588408
>>
>>
>>
>>_______________________________________________
>>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: "'Henning Schulzrinne'" <hgs@cs.columbia.edu>
Cc: avt@ietf.org
Subject: RE: [dccp] Re: [AVT] Would DCCP kill Internet real-time media dis
	 tribution?
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Hi Henning,

> I believe that there now is a suggested initial TCP windows 
> size > 1. Is 
> this correct? (If you figure a maximum acceptable network 
> delay of 100 
> ms and a typical audio/video packet spacing of 50 ms, you would be 
> sending about 4 packets per RTT.)

The current CCID3 draft (TFRC for DCCP) says the initial rate is 4 packets
per RTT.  This should pretty much eliminate slow start for speech audio.

Your points about the value of silence suppression for non-multicast
conferencing seem bang on target.  I'll remember to keep that in mind.

Tom Phelan

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



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From: =?iso-8859-1?Q?Jos=E9_Rey?= <rey@panasonic.de>
To: "'Colin Perkins'" <csp@csperkins.org>
Cc: <avt@ietf.org>
Subject: RE: [AVT] Retransmission draft
Date: Thu, 4 Dec 2003 17:52:25 +0100
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> -----Original Message-----
> From: Colin Perkins [mailto:csp@csperkins.org]
> Sent: Thursday, December 04, 2003 3:59 PM
> To: rey@panasonic.de
> Cc: avt@ietf.org
> Subject: Re: [AVT] Retransmission draft
>
>
> Jos=E9,
>
> [cc list trimmed]
>
> --> Jos=E9 Rey <rey@panasonic.de> writes:
> > in the past weeks I had some intensive internal discussions
> on the matter
> > of the potential IPRs on the RTP retransmission draft. As a WG
> > participant I perfectly understand and respect the comments
> and interests
> > expressed by you and others of the AVT group. In order to
> keep the good
> > working spirit in AVT it is my strong intention to adress
> these comments
> > in the best possible way by taking the interests of all
> parties involved
> > in this work into account.
> >
> > Matsushita has published an IPR statement that it is
> prepared to grant a
> > license on reasonable, nondiscriminatory terms and conditions (in
> > accordance with paragraph 10.3.3 of the RFC 2026) on any
> part that is
> > included into the RTP retransmission standard and that Matsushita
> > possibly holds IPRs on. Providing a royality-free license
> to open-source
> > developers will unfortunately lead to a situation where
> Matsushita will
> > lose any rights on their IPRs in most cases. Taking into
> account the work
> > and effort (we started with the retransmission draft in AVT
> more than 2.5
> > years ago) we spent into this draft, this is not acceptable for
> > Matsushita.
> >
> > On the other hand, we believe that the situation for open-source
> > developers is quite easy to solve by other means. First of
> all they are
> > not in any way'forced' to implement RTP retransmission. It
> is just an
> > optional repair mechanism for the RTP protocol, as
> described in RFC2354.
> > In order to solve the problems that RTP retransmission
> tries to solve,
> > there are other solutions available from the AVT group which are not
> > encumbered by IPRs, e.g. RFC2733 and draft-ietf-avt-ulp-08.txt. This
> > makes the implementation of the RTP retransmission solution purely
> > optional and easily avoidable to everyone.
>
> Provided it is clear that the mechanism is encumbered, yes.
>
> > In order to make this situation clear to anybody who reads the RTP
> > retransmission draft, we suggest to add a paragraph at the
> very beginning
> > of the draft. This paragraph will make implementors aware of the
> > situation that the solution described in this draft is possibly
> > encumbered by IPRs. Furthermore it will point implementors to
> > non-IPR-encumbered solutions as above, which may perfectly be used
> > instead of RTP retransmission.
>
> Without seeing the wording of this paragraph, I can't give a
> definitive
> answer, but that would largely address my concerns.
>

right, how about this:

"This document provides an RTP packet retransmission mechanism for repair=
 of
streaming media.  The purpose of this paragraph is to make the reader awa=
re
that the retransmission mechanism described in this document may be
encumbered by patent applications filed from Matsushita.  For details on =
the
IPR statement, please visit the IETF IPR web page
http://www.ietf.org/ietf/IPR.

However, packet retransmission is not the only means for streaming client=
s
and servers to cope with packet losses.  A detailed description of option=
s
for repair of streaming media is given in RFC2354.  Therefore, implemente=
rs
of this protocol shall take into account that there are in fact other
mechanisms, such as forward error correction (FEC) or packet interleaving
that represent an alternative to solve the problems packet retransmission
tries to solve.    In particular RFC2733 and RFC2198 present FEC and
redundant packet transmission schemes that might well fit to the needs of
the implementer. "

Jos=E9
> --
> 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] Retransmission draft
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With this disclaimer, I am now confused about the prespective with which I=

should read this draft as a (potential) implementator=2E I understand that=

Matsushita may have some IPR in this submission=2E But now are you saying
that you really don't know why anyone should use this scheme? Would it be
possible to give a scenario or two where the proposed mechanism may be
superior to follow? What is the point in sending this off to IETF RFC
track?=20

Thanks in advance for your explanation=2E=20

Vinay Sathe
Multirate Systems

right, how about this:

"This document provides an RTP packet retransmission mechanism for repair =
of
streaming media=2E  The purpose of this paragraph is to make the reader aw=
are
that the retransmission mechanism described in this document may be
encumbered by patent applications filed from Matsushita=2E  For details on=
 the
IPR statement, please visit the IETF IPR web page
http://www=2Eietf=2Eorg/ietf/IPR=2E

However, packet retransmission is not the only means for streaming clients=

and servers to cope with packet losses=2E  A detailed description of optio=
ns
for repair of streaming media is given in RFC2354=2E  Therefore, implement=
ers
of this protocol shall take into account that there are in fact other
mechanisms, such as forward error correction (FEC) or packet interleaving
that represent an alternative to solve the problems packet retransmission
tries to solve=2E    In particular RFC2733 and RFC2198 present FEC and
redundant packet transmission schemes that might well fit to the needs of
the implementer=2E "

Jos=E9


--------------------------------------------------------------------
mail2web - Check your email from the web at
http://mail2web=2Ecom/ =2E



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



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Subject: RE: [AVT] Retransmission draft
Date: Thu, 4 Dec 2003 11:29:33 -0800
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Thread-Topic: [AVT] Retransmission draft
Thread-Index: AcO6a6ZTzAji8JnHQemfmg+OL7R9xAAML/yw
From: "Gamze Seckin" <gamze@vidiator.com>
To: <rey@panasonic.de>, <avt@ietf.org>
Cc: "Colin Perkins (E-mail)" <csp@csperkins.org>,
        "David Leon (E-mail)" <David.Leon@nokia.com>,
        "Viktor Varsa (E-mail)" <viktor.varsa@nokia.com>,
        "Magnus Westerlund (E-mail)" <magnus.westerlund@era.ericsson.se>,
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Hi,

A quick note on the comment with regards to 3GPP's decision on this =
draft:

It has not been yet accepted yet within 3gpp. It is still under =
discussion. In the last 3gpp sa4 meeting various delegates raised a =
concern about server complexity and delay issues with regards to the =
protocol. It was said that the acceptance is conditional on concerns =
being addressed in the next meeting.=20

thanks
best regards,

gamze seckin

-----Original Message-----
From: Jos=E9 Rey [mailto:rey@panasonic.de]
Sent: Thursday, December 04, 2003 5:35 AM
To: avt@ietf.org
Cc: Colin Perkins (E-mail); David Leon (E-mail); Viktor Varsa (E-mail);
Magnus Westerlund (E-mail); hakenberg@panasonic.de
Subject: RE: [AVT] Retransmission draft


Hi Colin, all,

in the past weeks I had some intensive internal discussions on the =
matter of
the potential IPRs on the RTP retransmission draft. As a WG participant =
I
perfectly understand and respect the comments and interests expressed by =
you
and others of the AVT group. In order to keep the good working spirit in =
AVT
it is my strong intention to adress these comments in the best possible =
way
by taking the interests of all parties involved in this work into =
account.

Matsushita has published an IPR statement that it is prepared to grant a
license on reasonable, nondiscriminatory terms and conditions (in =
accordance
with paragraph 10.3.3 of the RFC 2026) on any part that is included into
the RTP retransmission standard and that Matsushita possibly holds IPRs =
on.
Providing a royality-free license to open-source developers will
unfortunately lead to a situation where Matsushita will lose any rights =
on
their IPRs in most cases. Taking into account the work and effort (we
started with the retransmission draft in AVT more than 2.5 years ago) we
spent into this draft, this is not acceptable for Matsushita.

On the other hand, we believe that the situation for open-source =
developers
is quite easy to solve by other means. First of all they are not in any =
way
'forced' to implement RTP retransmission. It is just an optional repair
mechanism for the RTP protocol, as described in RFC2354. In order to =
solve
the problems that RTP retransmission tries to solve, there are other
solutions available from the AVT group which are not encumbered by IPRs,
e.g. RFC2733 and draft-ietf-avt-ulp-08.txt. This makes the =
implementation
of the RTP retransmission solution purely optional and easily avoidable =
to
everyone.

In order to make this situation clear to anybody who reads the RTP
retransmission draft, we suggest to add a paragraph at the very =
beginning of
the draft. This paragraph will make implementors aware of the situation =
that
the solution described in this draft is possibly encumbered by IPRs.
Furthermore it will point implementors to non-IPR-encumbered solutions =
as
above, which may perfectly be used instead of RTP retransmission.

Finally and FYI: 3GPP is aware of the IPR situation of the RTP
retransmission draft and has accepted the solution provided in
draft-ietf-avt-rtp-retransmission-09.txt as working assumption in their
packet based streaming service (PSS). The acceptance is conditional on =
the
fact that the draft will proceed on standards track and an RFC is =
available
before the release of the PSS specifications, (expected for March 2004).

Looking forward to your comments,

Jose

> -----Original Message-----
> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of Colin
> Perkins
> Sent: Tuesday, October 21, 2003 6:58 PM
> To: rey@panasonic.de
> Cc: avt@ietf.org; 'Magnus Westerlund'
> Subject: Re: [AVT] Retransmission draft
>
>
> On Tuesday, Oct 21, 2003, at 12:54 Europe/London, Jose Rey wrote:
> ...
> > In previous emails we have been asked to explain our patent
> > application and
> > we
> > informed AVT that the patent application itself specifies a
transport
> > method
> > using the RTP protocol where two sequence numbers are used: one to
> > specifiy
> > the transmission order (=3DRTP SN) and the other one defining the
> > reprodution
> > order, which in the draft corresponds with the OSN (original
sequence
> > number). We also stated that it is not possible to implement the
> > protocol
> > without infringing on the patent.
> > So far we have not received any more questions on the patent, but
are
> > happy
> > to provide them in case needed.
> >
> > On the issue of how to proceed with this RTP retransmission draft
some
> > opinions have been provided by Magnus Westerlund, Lars-Erik Jonsson
and
> > John Lazzaro. John Lazzaro (08.09.03) has suggested to publish it as
> > Informational if the patent rights are held while Magnus and
> > Lars-Erik have suggested to proceed with the document as a Proposed
> > Standard (as we understand from the emails on from 22.09.03 and
> > 09.09.03).
> >
> > Taking all the above into account and considering that
> > independent of the patent situation this work is intended to be
> > included in
> > the next release of 3GPP, we would like to ask the group to proceed
> > with the
> > draft on standards track.
> >
> > cheers,
> > Jose
>
> [takes working group chair hat off]
>
> 3GPP is only one forum which uses IETF standards, and one where
dealing
> with IPR issues is commonplace. Having patented standards is more of
an
> issue for open source developers, who are often excluded by the
> licensing terms. Would Matsushita be willing to clarify if the
> "reasonable, nondiscriminatory" terms allow open source
implementations
> to use the retransmission format (this will probably require a
> Royalty-Free license)?
>
> [puts working group chair hat on again]
>
> --
> Colin Perkins
> http://csperkins.org/
>
>
> _______________________________________________
> 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 Dec  4 22:16:29 2003
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To: Qiaobing Xie <Qiaobing.Xie@motorola.com>
cc: avt@ietf.org, dccp@ietf.org
In-Reply-To: Message from Qiaobing Xie <Qiaobing.Xie@motorola.com> 
   of "Wed, 03 Dec 2003 11:38:58 CST." <3FCE1FB1.950C93C7@motorola.com> 
Date: Thu, 04 Dec 2003 19:15:32 -0800
From: Eddie Kohler <kohler@icir.org>
Subject: [AVT] Re: [dccp] Would a DCCP flow need to send 2x its preferred rate?
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> I wish I could be as optimistic as you. I wonder if anyone on the list
> has some numbers that can show what percentage of additional media data
> will be injected into the network if codecs would stop doing silence
> detection (motion detection for video).

For the app Henning mentioned (multi-party, non-multicast conferencing),
silence suppression is a huge win.  (With 100 participants and silence
suppression, you have 1-2 inbound streams' worth of data and 100 outbound
streams' worth; without it, you have 50-100 inbound streams and 100
outbound streams.)

On the other hand, say you have:
- 100 Kb/s uplink
- 100 person-to-person multimedia connections on the link
  - Each of which uses 2 Kb/s when not silent
  - and is silent 50% of the time
- and 100 TCP flows on the link (which are coming and going).

A back of the envelope analysis: (Comments welcome)

1. If the media flows use silence suppression but not congestion control,
   they will fit in the link on average, but crowd out the TCP flows.
   Useful work getting done: 100.  (1 "Work" unit == 1 TCP flow achieving
   >= 500 b/s, OR one media flow achieving its preferred nonsilent rate
   [2 Kb/s].)

2. If the media flows use congestion control but not silence suppression,
   media and TCP will share the link, but each media flow is limited to
   approximately 1/4 its preferred nonsilent bandwidth.  Useful work
   getting done: 125.

3. If the media flows use congestion control and send 1/2 rate filler data
   during silent periods, media and TCP will share the link, but each media
   flow is limited to approximately 1/3 its preferred nonsilent bandwidth.
   Useful work getting done: 133.

*None* of these results are particularly attractive.  What you really want
is 150 units of work or something like it.  But the first result is worst.

The easiest way to get 150 work units is through bandwidth reservations,
and thus QoS.  QoS is also the only reasonable way to privilege the media
traffic above TCP.

Note that "all-or-nothing" media apps, which would either send at their
preferred rate or not at all, get a tough break; given the TCP flows, the
fair rate is < 1 Kb/s, too low for any media apps to send.  (And thus
leading to 100 work units.)  I'm not sure what the right solution is.
There might not be a solution apart from QoS/bandwidth reservations.

> Fundamentally, I am concerned that this DCCP
> "app-who-sends-more-will-get-rewarded" feature will be taken advantage of
> in the wrong way (the draft that starts this discussion is the first
> attempt for this, but it won't be the last).
> 
> If this behavior of "sending filler data when no real data to send"
> becomes a recommended practice for media-over-DCCP (as you and Tom seem
> to suggest), what would stop the same recommendation be made for app over
> TCP? For an app over a long lived TCP connection, the sender can also
> benefit from sending "filler data" to keep the pipe from collapsing, so
> as to avoid the penalty of going through another slow start when it has
> real data to send next time. But I think this is hardly something that
> should be recommended for TCP based apps. Why should it be more
> acceptable for DCCP then?

It is a tough tradeoff.  

* Sending filler data is clearly bad.

* On the other hand, the Internet works not because we judge whether the
  *contents* of a data stream are worthwhile, but because we ensure that
  data streams are friendly to one another.  They do not continue
  transmitting at high rates in the face of congestion.  They do not cause
  congestion collapse.

  Sending filler data *limited by congestion control* won't cause
  congestion collapse.  The app will not send *more* filler data after a
  loss event; it will send less.

My feeling has therefore been that filler data *limited by congestion
control* is better for the Internet than non-congestion-controlled media
traffic.  But it is tough.

Eddie

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



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> Jose Rey writes:
>
> Providing a royality-free license to open-source developers will
> unfortunately lead to a situation where Matsushita will lose any
> rights on their IPRs in most cases. Taking into account the work and
> effort (we started with the retransmission draft in AVT more than 2.5
> years ago) we spent into this draft, this is not acceptable for
> Matsushita.

And likewise, an AVT standards-track RFC that is not open for free
software implementation is not acceptable to me.  So, let the record
show that there was not a unanimous consensus on this document going
to Last Call on the standards track, unless Matsushita changes its
stance and offers a royalty-free license to open-source developers.

---------------------------------------------------
John Lazzaro    http://www.cs.berkeley.edu/~lazzaro
     lazzaro [at] cs [dot] berkeley [dot] edu     
---------------------------------------------------



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



From exim@www1.ietf.org  Fri Dec  5 01:50:24 2003
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Subject: Re: [AVT] Re: [dccp] Would a DCCP flow need to send 2x its
  preferred rate?
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 <Qiaobing.Xie@motorola.com>
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On Thu, 04 Dec 2003 19:15:32 -0800
 Eddie Kohler <kohler@icir.org> wrote:
> > I wish I could be as optimistic as you. I wonder if anyone on the list
> > has some numbers that can show what percentage of additional media data
> > will be injected into the network if codecs would stop doing silence
> > detection (motion detection for video).
> 

Hello;

(Just got back from the airport, and am working backwards in e-mails)

From Stanislaw's indispensible netflows :

http://netflow.internet2.edu/weekly/20031124/

Reports on traffic on all Abilene routers

Table 7 (for the most recent week).

Http        : 10.6 % of all octets for the week
https       :  0.52 %
IPSEC + SSH :  2.68 %
Multicast   :  1.79% (for fan-in, not fan-out)
"Subset of VOIP" : 0.01 %

File Sharing : Between 15.21 and 52.08% (depending on how much of the "unknown" traffic is f.s.)

Oh, just for reference :

Table 8 

TCP 92.22 %
UDP  5.16 %

So, my answer to your question is of order 0.01% (as I do not believe that at present the majority
of VOIP is encrypted).

Regards
Marshall Eubanks

> For the app Henning mentioned (multi-party, non-multicast conferencing),
> silence suppression is a huge win.  (With 100 participants and silence
> suppression, you have 1-2 inbound streams' worth of data and 100 outbound
> streams' worth; without it, you have 50-100 inbound streams and 100
> outbound streams.)
> 
> On the other hand, say you have:
> - 100 Kb/s uplink
> - 100 person-to-person multimedia connections on the link
>   - Each of which uses 2 Kb/s when not silent
>   - and is silent 50% of the time
> - and 100 TCP flows on the link (which are coming and going).
> 
> A back of the envelope analysis: (Comments welcome)
> 
> 1. If the media flows use silence suppression but not congestion control,
>    they will fit in the link on average, but crowd out the TCP flows.
>    Useful work getting done: 100.  (1 "Work" unit == 1 TCP flow achieving
>    >= 500 b/s, OR one media flow achieving its preferred nonsilent rate
>    [2 Kb/s].)
> 
> 2. If the media flows use congestion control but not silence suppression,
>    media and TCP will share the link, but each media flow is limited to
>    approximately 1/4 its preferred nonsilent bandwidth.  Useful work
>    getting done: 125.
> 
> 3. If the media flows use congestion control and send 1/2 rate filler data
>    during silent periods, media and TCP will share the link, but each media
>    flow is limited to approximately 1/3 its preferred nonsilent bandwidth.
>    Useful work getting done: 133.
> 
> *None* of these results are particularly attractive.  What you really want
> is 150 units of work or something like it.  But the first result is worst.
> 
> The easiest way to get 150 work units is through bandwidth reservations,
> and thus QoS.  QoS is also the only reasonable way to privilege the media
> traffic above TCP.
> 
> Note that "all-or-nothing" media apps, which would either send at their
> preferred rate or not at all, get a tough break; given the TCP flows, the
> fair rate is < 1 Kb/s, too low for any media apps to send.  (And thus
> leading to 100 work units.)  I'm not sure what the right solution is.
> There might not be a solution apart from QoS/bandwidth reservations.
> 
> > Fundamentally, I am concerned that this DCCP
> > "app-who-sends-more-will-get-rewarded" feature will be taken advantage of
> > in the wrong way (the draft that starts this discussion is the first
> > attempt for this, but it won't be the last).
> > 
> > If this behavior of "sending filler data when no real data to send"
> > becomes a recommended practice for media-over-DCCP (as you and Tom seem
> > to suggest), what would stop the same recommendation be made for app over
> > TCP? For an app over a long lived TCP connection, the sender can also
> > benefit from sending "filler data" to keep the pipe from collapsing, so
> > as to avoid the penalty of going through another slow start when it has
> > real data to send next time. But I think this is hardly something that
> > should be recommended for TCP based apps. Why should it be more
> > acceptable for DCCP then?
> 
> It is a tough tradeoff.  
> 
> * Sending filler data is clearly bad.
> 
> * On the other hand, the Internet works not because we judge whether the
>   *contents* of a data stream are worthwhile, but because we ensure that
>   data streams are friendly to one another.  They do not continue
>   transmitting at high rates in the face of congestion.  They do not cause
>   congestion collapse.
> 
>   Sending filler data *limited by congestion control* won't cause
>   congestion collapse.  The app will not send *more* filler data after a
>   loss event; it will send less.
> 
> My feeling has therefore been that filler data *limited by congestion
> control* is better for the Internet than non-congestion-controlled media
> traffic.  But it is tough.
> 
> Eddie
> 
> _______________________________________________
> 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 Dec  5 04:00:13 2003
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	 Fri, 5 Dec 2003 09:57:45 +0100
From: "Rolf Hakenberg" <hakenberg@panasonic.de>
To: "Gamze Seckin" <gamze@vidiator.com>, <rey@panasonic.de>, <avt@ietf.org>
Cc: "Colin Perkins \(E-mail\)" <csp@csperkins.org>,
        "David Leon \(E-mail\)" <David.Leon@nokia.com>,
        "Viktor Varsa \(E-mail\)" <viktor.varsa@nokia.com>,
        "Magnus Westerlund \(E-mail\)" <magnus.westerlund@era.ericsson.se>
Subject: RE: [AVT] Retransmission draft
Date: Fri, 5 Dec 2003 09:57:45 +0100
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Dear Gamze and all,

Although I believe that the concerns expressed in 3GPP on server complexi=
ty
and delay are less relevant for our discussion on potential IPRs, please =
let
me clarify the agreement in 3GPP.

At the last 3GPP SA4 meeting there was a consensus on to make RTP
Retransmission a working assumption in PSS Rel-6, subject to the concerns=
 on
server complexity and delay being addressed at the next meeting.

Important for our current discussion on potential IPRs and how to progres=
s
the draft is the fact that there was no condition on the IPR issues, but =
in
fact conditions that the draft should move to RFC on standards track befo=
re
the finalisation of 3GPP Rel-6.

Best Regards,
Rolf.

> -----Original Message-----
> From: Gamze Seckin [mailto:gamze@vidiator.com]
> Sent: Thursday, December 04, 2003 8:30 PM
> To: rey@panasonic.de; avt@ietf.org
> Cc: Colin Perkins (E-mail); David Leon (E-mail); Viktor Varsa (E-mail);
> Magnus Westerlund (E-mail); hakenberg@panasonic.de
> Subject: RE: [AVT] Retransmission draft
>
>
> Hi,
>
> A quick note on the comment with regards to 3GPP's decision on this dra=
ft:
>
> It has not been yet accepted yet within 3gpp. It is still under
> discussion. In the last 3gpp sa4 meeting various delegates raised
> a concern about server complexity and delay issues with regards
> to the protocol. It was said that the acceptance is conditional
> on concerns being addressed in the next meeting.
>
> thanks
> best regards,
>
> gamze seckin
>
> -----Original Message-----
> From: Jos=E9 Rey [mailto:rey@panasonic.de]
> Sent: Thursday, December 04, 2003 5:35 AM
> To: avt@ietf.org
> Cc: Colin Perkins (E-mail); David Leon (E-mail); Viktor Varsa (E-mail);
> Magnus Westerlund (E-mail); hakenberg@panasonic.de
> Subject: RE: [AVT] Retransmission draft
>
>
> Hi Colin, all,
>
> in the past weeks I had some intensive internal discussions on
> the matter of
> the potential IPRs on the RTP retransmission draft. As a WG participant=
 I
> perfectly understand and respect the comments and interests
> expressed by you
> and others of the AVT group. In order to keep the good working
> spirit in AVT
> it is my strong intention to adress these comments in the best
> possible way
> by taking the interests of all parties involved in this work into accou=
nt.
>
> Matsushita has published an IPR statement that it is prepared to grant =
a
> license on reasonable, nondiscriminatory terms and conditions (in
> accordance
> with paragraph 10.3.3 of the RFC 2026) on any part that is included int=
o
> the RTP retransmission standard and that Matsushita possibly
> holds IPRs on.
> Providing a royality-free license to open-source developers will
> unfortunately lead to a situation where Matsushita will lose any rights=
 on
> their IPRs in most cases. Taking into account the work and effort (we
> started with the retransmission draft in AVT more than 2.5 years ago) w=
e
> spent into this draft, this is not acceptable for Matsushita.
>
> On the other hand, we believe that the situation for open-source
> developers
> is quite easy to solve by other means. First of all they are not
> in any way
> 'forced' to implement RTP retransmission. It is just an optional repair
> mechanism for the RTP protocol, as described in RFC2354. In order to so=
lve
> the problems that RTP retransmission tries to solve, there are other
> solutions available from the AVT group which are not encumbered by IPRs=
,
> e.g. RFC2733 and draft-ietf-avt-ulp-08.txt. This makes the implementati=
on
> of the RTP retransmission solution purely optional and easily avoidable=
 to
> everyone.
>
> In order to make this situation clear to anybody who reads the RTP
> retransmission draft, we suggest to add a paragraph at the very
> beginning of
> the draft. This paragraph will make implementors aware of the
> situation that
> the solution described in this draft is possibly encumbered by IPRs.
> Furthermore it will point implementors to non-IPR-encumbered solutions =
as
> above, which may perfectly be used instead of RTP retransmission.
>
> Finally and FYI: 3GPP is aware of the IPR situation of the RTP
> retransmission draft and has accepted the solution provided in
> draft-ietf-avt-rtp-retransmission-09.txt as working assumption in their
> packet based streaming service (PSS). The acceptance is conditional on =
the
> fact that the draft will proceed on standards track and an RFC is
> available
> before the release of the PSS specifications, (expected for March 2004).
>
> Looking forward to your comments,
>
> Jose
>
> > -----Original Message-----
> > From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of Coli=
n
> > Perkins
> > Sent: Tuesday, October 21, 2003 6:58 PM
> > To: rey@panasonic.de
> > Cc: avt@ietf.org; 'Magnus Westerlund'
> > Subject: Re: [AVT] Retransmission draft
> >
> >
> > On Tuesday, Oct 21, 2003, at 12:54 Europe/London, Jose Rey wrote:
> > ...
> > > In previous emails we have been asked to explain our patent
> > > application and
> > > we
> > > informed AVT that the patent application itself specifies a
> transport
> > > method
> > > using the RTP protocol where two sequence numbers are used: one to
> > > specifiy
> > > the transmission order (=3DRTP SN) and the other one defining the
> > > reprodution
> > > order, which in the draft corresponds with the OSN (original
> sequence
> > > number). We also stated that it is not possible to implement the
> > > protocol
> > > without infringing on the patent.
> > > So far we have not received any more questions on the patent, but
> are
> > > happy
> > > to provide them in case needed.
> > >
> > > On the issue of how to proceed with this RTP retransmission draft
> some
> > > opinions have been provided by Magnus Westerlund, Lars-Erik Jonsson
> and
> > > John Lazzaro. John Lazzaro (08.09.03) has suggested to publish it a=
s
> > > Informational if the patent rights are held while Magnus and
> > > Lars-Erik have suggested to proceed with the document as a Proposed
> > > Standard (as we understand from the emails on from 22.09.03 and
> > > 09.09.03).
> > >
> > > Taking all the above into account and considering that
> > > independent of the patent situation this work is intended to be
> > > included in
> > > the next release of 3GPP, we would like to ask the group to proceed
> > > with the
> > > draft on standards track.
> > >
> > > cheers,
> > > Jose
> >
> > [takes working group chair hat off]
> >
> > 3GPP is only one forum which uses IETF standards, and one where
> dealing
> > with IPR issues is commonplace. Having patented standards is more of
> an
> > issue for open source developers, who are often excluded by the
> > licensing terms. Would Matsushita be willing to clarify if the
> > "reasonable, nondiscriminatory" terms allow open source
> implementations
> > to use the retransmission format (this will probably require a
> > Royalty-Free license)?
> >
> > [puts working group chair hat on again]
> >
> > --
> > Colin Perkins
> > http://csperkins.org/
> >
> >
> > _______________________________________________
> > 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  Fri Dec  5 04:07:04 2003
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To: "Rosen, Brian" <Brian.Rosen@marconi.com>
cc: "'Eddie Kohler'" <kohler@icir.org>, avt@ietf.org, dccp@ietf.org
From: Sally Floyd <floyd@icir.org>
Date: Thu, 04 Dec 2003 15:02:29 -0800
Subject: [AVT] Re: [dccp] RE: Would DCCP kill Internet real-time media distribution?
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Brian -

>Slow start is not
>a practical thing for interactive use, sorry.

I have not read all of the earlier email, so I might be repeating
things that have already been said.

Currently, slow-start for TCP allows an initial window of up to
four packets.  The reason that the Internet still works plausibly
well even with interactive traffic that does not obey slow-start
is that the vast majority of the packets in the Internet still use
TCP, and still abide by slow-start and the other mechanisms of TCP
congestion control.  If the majority of the traffic on a congested
link was interactive traffic that did not use slow-start, and that
also did not use any admissions control or explicit feedback from
routers allowing high initial windows, I don't think that any of
the interactive users would be very happy with the results.  This
would particularly be true, I would conjecture, in a network with
only interactive traffic, with no TCP traffic to back off and make
space for the interactive traffic.

For a best-effort network that includes TCP traffic, there is the
additional concern of the fairness for the TCP users.  There is no
reason why best-effort interactive UDP traffic has a higher right
to bandwidth or to low delay that does the best-effort TCP traffic
(whose users are paying the same monthly fees to their service
providers).  Remember that the TCP traffic could easily be someone
doing a web transaction that is perfectly important to them or to
their business, and for which they are expecting the same prompt
response times from the network that the interactive users are
expecting.

My own prediction is that, if there seems to be a danger of congestion
in best-effort links due to best-effort UDP traffic that is not
TCP-friendly, the routers at the congested links would end up
preferentially dropping the UDP traffic in times of high congestion
(since this is easier than more precisely-targetted forms of
preferential dropping at the router).  But maybe things will work
out better than that, and routers will have mechanisms to precisely
identify those flows that aren't using decent end-to-end congestion
control in a time of high congestion, and preferentially drop packets
from only those flows.  No telling...

- Sally
http://www.icir.org/floyd/




_______________________________________________
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: "'Sally Floyd'" <floyd@icir.org>
Cc: "'Eddie Kohler'" <kohler@icir.org>, avt@ietf.org, dccp@ietf.org
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Subject: [AVT] RE: [dccp] RE: Would DCCP kill Internet real-time media distribut
 ion?
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Sally

I think the primary control for media streams is going to be 
admission control.  There is a missing link to make this work,
which is that there is no feedback path from the media flow to the
signalling (RTP to SIP/RTSP), but fundamentally, if a media flow
experiences congestion, it will be downshifted to a lower rate 
(once, maybe) or shut down.  It's pretty reasonable to expect that
such congestion control could be widely deployed.  I think that
is the way congestion collapse should be addressed.

It's harder to deal with the relative fairness of TCP vs media flows.
None of the mechanisms that I have heard proposed will work in my
opinion.  Work means that you get acceptable voice/video quality up 
to the point where congestion is bad enough that signalling shuts 
down the flow.  Media flows can't deal with "fair" in the sense that 
TCP does.  There is no "fair", there is only "all or nothing". 

One possible avenue of exploration is to look more into aggregate
behavior; you CAN probably assert a more TCP friendly "rate control"
if you deal with the aggregate of media traffic, even if the individual
components of the traffic experience all or nothing flow control.

Ultimately, however, the actual media flow can't do anything interesting
with existing TCP mechanisms.  It only has a very gross rate control,
it can't probe, it's silly to send filler, and it can't use any form 
of a window.  Similarly, TCP doesn't have any way to know if the media 
flow is present, and thus adjust it's behavior appropriately.  
"appropriately" means not being greedy; the biggest actual problem that 
exists today is that TCP destroys voice, not that voice destroys TCP.
I think, naively perhaps, that it is possible to design mechanisms 
that would work together, but it is highly unlikely to be possible to
do this from one side; TCP must change as well as UDP (or a replacement).

I suspect that if a service provider is providing both a data service
and a voice service over that data service, that it will drop all the
TCP traffic before it drops any voice traffic, because, so far, the
pricing for a voice bit is several orders of magnitude larger than
a data bit.  If the data service provider is not also the voice service
provider, it's not clear to me why he would care which customer bits
get treated differently.  Conventional wisdom in carriers is that
flat rate Internet service makes small profits, while voice/video 
makes a lot more.  Don't know if that's really true, or will remain true,
but I'll bet you will scream much more loudly if your voice/video
sounds/looks bad than if your browser slows down.

Of course, deployment of QoS mechanisms would be helpful. 

Brian

> -----Original Message-----
> From: Sally Floyd [mailto:floyd@icir.org]
> Sent: Thursday, December 04, 2003 6:02 PM
> To: Rosen, Brian
> Cc: 'Eddie Kohler'; avt@ietf.org; dccp@ietf.org
> Subject: Re: [dccp] RE: Would DCCP kill Internet real-time media
> distribution? 
> 
> 
> Brian -
> 
> >Slow start is not
> >a practical thing for interactive use, sorry.
> 
> I have not read all of the earlier email, so I might be repeating
> things that have already been said.
> 
> Currently, slow-start for TCP allows an initial window of up to
> four packets.  The reason that the Internet still works plausibly
> well even with interactive traffic that does not obey slow-start
> is that the vast majority of the packets in the Internet still use
> TCP, and still abide by slow-start and the other mechanisms of TCP
> congestion control.  If the majority of the traffic on a congested
> link was interactive traffic that did not use slow-start, and that
> also did not use any admissions control or explicit feedback from
> routers allowing high initial windows, I don't think that any of
> the interactive users would be very happy with the results.  This
> would particularly be true, I would conjecture, in a network with
> only interactive traffic, with no TCP traffic to back off and make
> space for the interactive traffic.
> 
> For a best-effort network that includes TCP traffic, there is the
> additional concern of the fairness for the TCP users.  There is no
> reason why best-effort interactive UDP traffic has a higher right
> to bandwidth or to low delay that does the best-effort TCP traffic
> (whose users are paying the same monthly fees to their service
> providers).  Remember that the TCP traffic could easily be someone
> doing a web transaction that is perfectly important to them or to
> their business, and for which they are expecting the same prompt
> response times from the network that the interactive users are
> expecting.
> 
> My own prediction is that, if there seems to be a danger of congestion
> in best-effort links due to best-effort UDP traffic that is not
> TCP-friendly, the routers at the congested links would end up
> preferentially dropping the UDP traffic in times of high congestion
> (since this is easier than more precisely-targetted forms of
> preferential dropping at the router).  But maybe things will work
> out better than that, and routers will have mechanisms to precisely
> identify those flows that aren't using decent end-to-end congestion
> control in a time of high congestion, and preferentially drop packets
> from only those flows.  No telling...
> 
> - Sally
> http://www.icir.org/floyd/
> 
> 
> 

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



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To: "Rosen, Brian" <Brian.Rosen@marconi.com>
cc: "'Sally Floyd'" <floyd@icir.org>, avt@ietf.org, dccp@ietf.org
In-Reply-To: Message from "Rosen, Brian" <Brian.Rosen@marconi.com> 
   of "Fri, 05 Dec 2003 09:42:35 EST." <313680C9A886D511A06000204840E1CF070B618C@whq-msgusr-02.pit.comms.marconi.com> 
Date: Fri, 05 Dec 2003 11:35:25 -0800
From: Eddie Kohler <kohler@icir.org>
Subject: [AVT] Re: [dccp] RE: Would DCCP kill Internet real-time media distribut ion?
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> It's harder to deal with the relative fairness of TCP vs media flows.
> None of the mechanisms that I have heard proposed will work in my
> opinion.  Work means that you get acceptable voice/video quality up 
> to the point where congestion is bad enough that signalling shuts 
> down the flow.  Media flows can't deal with "fair" in the sense that 
> TCP does.  There is no "fair", there is only "all or nothing". 

I think the key question for the DCCP folk is, What does it mean for
congestion to be "bad enough that signalling shuts down the flow"?

1. The loss rate is greater than some fixed constant.  (Independent of flow
   rate or RTT.)
2. Loss rate plus RTT indicate that the TCP-friendly rate has dropped below
   the app's transmit rate.
3. Loss rate plus RTT indicate that the TCP-friendly rate has dropped below
   some fraction -- say half -- of the app's preferred transmit rate.

If your app wants (1), then we are not the right protocol for you.
Arguably you are bad Homer and should get no donut on the public
best-effort Internet.

On the other hand, if (2) or (3) is the case, then we *want* to be your
protocol.  We provide a signalling path for the receiver to verifiably
inform the sender of the loss/mark rate.  We provide different mechanisms
for calculating the TCP-friendly rate.  The app can decide how to react to
the rate; such as by reducing once, then cutting off.  (DCCP does not
require apps with smoothly varying rates!  It provides a smoothly varying
_reference_ rate; the app's rate need not smoothly vary, as long as it
stays within some factor of the reference rate.)

As we've discussed, the more interactive, CBR-flavored apps would find it
difficult to use DCCP now.  The issue seems to be slow start.  ("Filler
data", for example, is really about slow start: filler data is one
mechanism for avoiding slow start after an idle period.)

So, thought experiment for interactive, CBR-flavored apps:
1. Is slow start in fact the only problem?
2. What would these apps give up in return for avoiding slow start?  (Maybe
   the apps would limit their transmit rate to below some absolute number,
   or shut off quickly if a "fast start" caused loss.  In return, maybe the
   app would avoid *any* slow start, or maybe it'd avoid slow start after
   idle periods.)
3. Would any such compromise be acceptable for the DCCP folk?  (Need to
   think more.)

> "appropriately" means not being greedy; the biggest actual problem that 
> exists today is that TCP destroys voice, not that voice destroys TCP.

I assume you see this happen.  We've been throwing around assumptions about
when -- do you agree?  Does it happen mostly if the congested link has low
statmux?  Any references?

Eddie

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



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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Eddie Kohler'" <kohler@icir.org>
Cc: "'Sally Floyd'" <floyd@icir.org>, avt@ietf.org, dccp@ietf.org
Subject: RE: [AVT] Re: [dccp] RE: Would DCCP kill Internet real-time media
	 distribut ion?
Date: Fri, 5 Dec 2003 15:15:10 -0500 
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> I think the key question for the DCCP folk is, What does it mean for
> congestion to be "bad enough that signalling shuts down the flow"?
> 
> 1. The loss rate is greater than some fixed constant.  
> (Independent of flow
>    rate or RTT.)
> 2. Loss rate plus RTT indicate that the TCP-friendly rate has 
> dropped below
>    the app's transmit rate.
> 3. Loss rate plus RTT indicate that the TCP-friendly rate has 
> dropped below
>    some fraction -- say half -- of the app's preferred transmit rate.
It's closest to (1).  There really isn't a "fixed" constant, because
what's acceptable quality is in the eye of the beholder, but probably,
the signalling system would establish a fixed loss rate, below which
it would take action.  It would not matter what else was happening;
that is, whatever the reason for the loss, the quality of the media
is below the point where it should be maintained. 

It will then lower the rate, or drop the flow.

For voice, the loss rate is something in the single digit percent
range for high quality, and I don't know, maybe 10%-15% or so for 
low quality services.  I don't think anything approaching 30% is
useful, even if it's free.  You get to do some averaging.

> 
> If your app wants (1), then we are not the right protocol for you.
> Arguably you are bad Homer and should get no donut on the public
> best-effort Internet.
Agree that DCCP is probably not useful for interactive media.
For non-interactive media, there is some hope, because slow-start
is probably not much of a problem.  However, I still don't see
how you get to maintain a fixed rate as TCP flows come and go,
even with a lot of buffering.  Won't TFRC in the face of
congestion lower your rate?  That would not work; the rate must
be pretty constant (averaged over the buffer length).

Disagree that media is a bad Homer.  TCP is broken.
Nowhere is it written that TCP=best effort.

> 
> On the other hand, if (2) or (3) is the case, then we *want* 
> to be your
> protocol.  We provide a signalling path for the receiver to verifiably
> inform the sender of the loss/mark rate.  We provide 
> different mechanisms
> for calculating the TCP-friendly rate.  The app can decide 
> how to react to
> the rate; such as by reducing once, then cutting off.  (DCCP does not
> require apps with smoothly varying rates!  It provides a 
> smoothly varying
> _reference_ rate; the app's rate need not smoothly vary, as long as it
> stays within some factor of the reference rate.)
I think this is all foolish.  You have to assume that the TCP flows
are coming and going, and the media flow is pretty fixed.  I don't 
see how DCCP will maintain the original TCP-friendly rate, even if
that rate was forced to be acceptable.  You can say that it
varies the TCP-friendly rate slowly, but not zero, right?
Time scales here are minutes to hours.  You need constant rate
over minutes to hours.

> 
> As we've discussed, the more interactive, CBR-flavored apps 
> would find it
> difficult to use DCCP now.  The issue seems to be slow start. 
>  ("Filler
> data", for example, is really about slow start: filler data is one
> mechanism for avoiding slow start after an idle period.)
Well, it's one of a couple of problems.
All the current apps are CBR flavored, Philippe's suggestions
notwithstanding.  Slow start is a killer for interactive, for sure.

> 
> So, thought experiment for interactive, CBR-flavored apps:
> 1. Is slow start in fact the only problem?
No, I think there is no way to maintain a constant rate
over the time period.

> 2. What would these apps give up in return for avoiding slow 
> start?  (Maybe
>    the apps would limit their transmit rate to below some 
> absolute number,
>    or shut off quickly if a "fast start" caused loss.  In 
> return, maybe the
>    app would avoid *any* slow start, or maybe it'd avoid slow 
> start after
>    idle periods.)
It could do any form of signalling of its intentions

It can limit its rate to a fixed number; it can even negotiate
that number, and, if necessary, once, it can lower the number.

It can react to loss by lowering the rate (once) or shutting down


> 3. Would any such compromise be acceptable for the DCCP folk? 
>  (Need to
>    think more.)
> 
> > "appropriately" means not being greedy; the biggest actual 
> problem that 
> > exists today is that TCP destroys voice, not that voice 
> destroys TCP.
> 
> I assume you see this happen.  We've been throwing around 
> assumptions about
> when -- do you agree?  Does it happen mostly if the congested 
> link has low
> statmux?  Any references?
I have no reference data.  My sense is that it happens under
both low and high statmux, but I'm not really sure its the same
mechanism in both cases.  Need more data I suspect.

Brian 

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



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--NextPart

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

	Title		: RTP payload format for a 64 kbit/s transparent call
	Author(s)	: R. Kreuter
	Filename	: draft-ietf-avt-rtp-clearmode-04.txt
	Pages		: 7
	Date		: 2003-12-5
	
This document describes how to carry 64 kbit/s data streams 
transparently in RTP packets, using a pseudo-codec called 
'Clearmode'.  It also serves as registration for a related MIME type 
called 'audio/clearmode'. 
'Clearmode' is a basic feature of VoIP media gateways.

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



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To: "Rosen, Brian" <Brian.Rosen@marconi.com>
cc: "'Sally Floyd'" <floyd@icir.org>, avt@ietf.org
Subject: Re: [AVT] Re: [dccp] RE: Would DCCP kill Internet real-time media distribut ion? 
In-Reply-To: Message from "Rosen, Brian" <Brian.Rosen@marconi.com> 
   of "Fri, 05 Dec 2003 15:15:10 EST." <313680C9A886D511A06000204840E1CF070B6190@whq-msgusr-02.pit.comms.marconi.com> 
Date: Fri, 05 Dec 2003 13:22:35 -0800
From: Eddie Kohler <kohler@icir.org>
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> However, I still don't see
> how you get to maintain a fixed rate as TCP flows come and go,
> even with a lot of buffering.  Won't TFRC in the face of
> congestion lower your rate?

Not necessarily. The TCP equation takes in an RTT and a loss event rate,
and produces a reference rate. This reference rate can be thought of as
"what TCP would achieve sending full blast."

If you were originally sending less than what TCP would have sent (possibly
the normal case?), then slightly increased loss rates would not reduce the
reference rate below your actual transmit rate.  Continue sending at your
preferred rate.

Caveat: The reference rate *is* limited to (2 * your recent transmit
rate).  But the factor of 2 is pretty wide wiggle room.

Eddie

_______________________________________________
Audio/Video Transport Working Group
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From: Mark Handley <M.Handley@cs.ucl.ac.uk>
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To: qli <qli@brainmediaco.com>
cc: avt <avt@ietf.org>, dccp <dccp@ietf.org>
Subject: Re: [AVT] RE: [dccp] Would a DCCP flow need to send 2x its preferred rate? 
In-reply-to: Your message of "Wed, 03 Dec 2003 09:32:40 EST."
             <86C0B7B413918E45A4B0A97F95657A0A152586@ceruleus.brainmediaco.com> 
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>> If the congested link has low levels of statmux -- like one  DCCP flow and
>> one TCP flow -- then yes, the TCP would try for the available bandwidth.
>> And during a silent period, the TFRC flow would have to transmit something
>> so it wasn't crowded out entirely.  But this amount would be *half* the
>> nominal rate, or perhaps less.  Not twice.

>Huh?  You seem to be saying that if you have a lot of TCP streams on a link
>they won't use all of the available bandwidth.  I thought just the opposite.
>With a low level of multiplexing there's more chance of synchronizing losses
>and cutbacks, and not fully utilizing the link.  But with a large number
>of flows enough of them will be blasting away while others are cut back.

No this isn't what Eddie was saying.  Say the link has capacity C.
Assume every flow has the same RTT for the sake of simplicity:

In the low statmux case:
  The TFRC flow achieves rate C because there's no other traffic.  Then
  it pauses, and a TCP starts which achieves rate C.  Then TFRC starts
  up again at rate C, and 50% loss briefly ensures before everything
  settles down.

In the high statmux case (n flows): 

  The TFRC flow achieves rate C/n, and loss rate p1.  Then it
  pauses. The other TCP flows rapidly use up the available capacity,
  each averaging rate C/(n-1).  Then TFRC restarts with rate C/n, and
  briefly sees loss rate p2.  If n >> 1 then C/n ~= C/(n-1) and p1 ~=
  p2.  Thus the *startup* of the TFRC flow doesn't significantly
  affect the loss rate, and it is going at approximately the correct
  rate.

  Of course the number of flows can change while the TFRC flow is
  idle.  But the worst that can happen is that the *additional* loss
  caused by the TFRC restarting is bounded to p = 1/(n-1).  For large
  n, this will likely be dominated by the loss caused by the TCPs
  themselves doing AIMD.

Thus the restart issue is only a serious problem when there is little
statmuxing at the bottleneck link (or where many TFRC restart
simultaneously, synchronized by some external event).

The moral of this is to use RSVP/Intserv on your 56K modem link, and
ignore the problem elsewhere.

Cheers,
	Mark

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



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From: Guy T Almes <almes@internet2.edu>
To: slow@caltech.edu, "Rosen, Brian" <Brian.Rosen@marconi.com>
Cc: "'Sally Floyd'" <floyd@icir.org>, "'Eddie Kohler'" <kohler@icir.org>,
        avt@ietf.org, dccp@ietf.org, fast-support@cs.caltech.edu,
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Subject: [AVT] Re: [dccp] RE: Would DCCP kill Internet real-time media distribut
 ion?
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Steven,
  What does seem clear is that, *if* using delay+loss signals together you 
can maintain modest-sized and stable-sized queues at the bottleneck link, 
then there are many positive implications for the net, both for the 
routers/switches and their engineering and for the applications, 
particularly jitter-sensitive ones, that share the bottleneck link.

  It would seem that the challenge is to accomplish the task of maintaining 
such queues in a highly dynamic environment with many and many kinds of 
flows.
  With an increasing reach of the international advanced production 
networks (e.g., Internet2/Abilene in the US, Geant in Europe) being made up 
of 10 Gb/s links, this challenge is ever more crucial.

  I look forward to hearing from others on the thread.

  Regards,
        -- Guy

--On Friday, December 05, 2003 09:51:56 -0800 Steven Low <slow@caltech.edu> 
wrote:

>
> Thanks everyone for the intresting discussions from various
> perspectives.   What are your expert opinions on the coexistence
> of a delay-based TCP with (let's assume, essentially) fixed rates
> of media flows?   Granted that there are many challenges in filtering
> out noise in delay measurements, etc, but these seem more
> surmountable than the fundamental difficulty of a loss-based
> scheme where TCP must periodically push the queue to overflow,
> thus squeezing out other flows that are not as (or more)
> aggressive than TCP.    In a delay-based scheme where (e.g.) a
> TCP flow will stabilize when it maintains a certain amount of queue
> (instead of pushing it over the cliff), it will then coexist peacefully
> with media flows, each sharing the link at a rate that's proportional
> to their buffer shares, without the protocol itself producing severe
> oscillations.
>
> Again, there are of course challenges to filter delay noise, ensure
> fairness, etc, but it seems to avoid a more difficult problem of a
> loss-based scheme.
>
> Steven
>
>
>
> Rosen, Brian wrote:
>
>> Sally
>>
>> I think the primary control for media streams is going to be
>> admission control.  There is a missing link to make this work,
>> which is that there is no feedback path from the media flow to the
>> signalling (RTP to SIP/RTSP), but fundamentally, if a media flow
>> experiences congestion, it will be downshifted to a lower rate
>> (once, maybe) or shut down.  It's pretty reasonable to expect that
>> such congestion control could be widely deployed.  I think that
>> is the way congestion collapse should be addressed.
>>
>> It's harder to deal with the relative fairness of TCP vs media flows.
>> None of the mechanisms that I have heard proposed will work in my
>> opinion.  Work means that you get acceptable voice/video quality up
>> to the point where congestion is bad enough that signalling shuts
>> down the flow.  Media flows can't deal with "fair" in the sense that
>> TCP does.  There is no "fair", there is only "all or nothing".
>>
>> One possible avenue of exploration is to look more into aggregate
>> behavior; you CAN probably assert a more TCP friendly "rate control"
>> if you deal with the aggregate of media traffic, even if the individual
>> components of the traffic experience all or nothing flow control.
>>
>> Ultimately, however, the actual media flow can't do anything interesting
>> with existing TCP mechanisms.  It only has a very gross rate control,
>> it can't probe, it's silly to send filler, and it can't use any form
>> of a window.  Similarly, TCP doesn't have any way to know if the media
>> flow is present, and thus adjust it's behavior appropriately.
>> "appropriately" means not being greedy; the biggest actual problem that
>> exists today is that TCP destroys voice, not that voice destroys TCP.
>> I think, naively perhaps, that it is possible to design mechanisms
>> that would work together, but it is highly unlikely to be possible to
>> do this from one side; TCP must change as well as UDP (or a replacement).
>>
>> I suspect that if a service provider is providing both a data service
>> and a voice service over that data service, that it will drop all the
>> TCP traffic before it drops any voice traffic, because, so far, the
>> pricing for a voice bit is several orders of magnitude larger than
>> a data bit.  If the data service provider is not also the voice service
>> provider, it's not clear to me why he would care which customer bits
>> get treated differently.  Conventional wisdom in carriers is that
>> flat rate Internet service makes small profits, while voice/video
>> makes a lot more.  Don't know if that's really true, or will remain true,
>> but I'll bet you will scream much more loudly if your voice/video
>> sounds/looks bad than if your browser slows down.
>>
>> Of course, deployment of QoS mechanisms would be helpful.
>>
>> Brian
>>
>>
>>
>>> -----Original Message-----
>>> From: Sally Floyd [mailto:floyd@icir.org]
>>> Sent: Thursday, December 04, 2003 6:02 PM
>>> To: Rosen, Brian
>>> Cc: 'Eddie Kohler'; avt@ietf.org; dccp@ietf.org
>>> Subject: Re: [dccp] RE: Would DCCP kill Internet real-time media
>>> distribution?
>>>
>>>
>>> Brian -
>>>
>>>
>>>
>>>> Slow start is not
>>>> a practical thing for interactive use, sorry.
>>>>
>>>>
>>> I have not read all of the earlier email, so I might be repeating
>>> things that have already been said.
>>>
>>> Currently, slow-start for TCP allows an initial window of up to
>>> four packets.  The reason that the Internet still works plausibly
>>> well even with interactive traffic that does not obey slow-start
>>> is that the vast majority of the packets in the Internet still use
>>> TCP, and still abide by slow-start and the other mechanisms of TCP
>>> congestion control.  If the majority of the traffic on a congested
>>> link was interactive traffic that did not use slow-start, and that
>>> also did not use any admissions control or explicit feedback from
>>> routers allowing high initial windows, I don't think that any of
>>> the interactive users would be very happy with the results.  This
>>> would particularly be true, I would conjecture, in a network with
>>> only interactive traffic, with no TCP traffic to back off and make
>>> space for the interactive traffic.
>>>
>>> For a best-effort network that includes TCP traffic, there is the
>>> additional concern of the fairness for the TCP users.  There is no
>>> reason why best-effort interactive UDP traffic has a higher right
>>> to bandwidth or to low delay that does the best-effort TCP traffic
>>> (whose users are paying the same monthly fees to their service
>>> providers).  Remember that the TCP traffic could easily be someone
>>> doing a web transaction that is perfectly important to them or to
>>> their business, and for which they are expecting the same prompt
>>> response times from the network that the interactive users are
>>> expecting.
>>>
>>> My own prediction is that, if there seems to be a danger of congestion
>>> in best-effort links due to best-effort UDP traffic that is not
>>> TCP-friendly, the routers at the congested links would end up
>>> preferentially dropping the UDP traffic in times of high congestion
>>> (since this is easier than more precisely-targetted forms of
>>> preferential dropping at the router).  But maybe things will work
>>> out better than that, and routers will have mechanisms to precisely
>>> identify those flows that aren't using decent end-to-end congestion
>>> control in a time of high congestion, and preferentially drop packets
>>> from only those flows.  No telling...
>>>
>>> - Sally
>>> http://www.icir.org/floyd/
>>>
>>>
>>>
>>>
>>>
>>
>> _______________________________________________
>> 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
>>
>>
>>
>
> --
> _____________________________________________
> Steven Low                assoc prof, cs & ee
> http://netlab.caltech.edu tel: (626) 395-6767
> slow@caltech.edu          fax: (626) 568-3603
>
>
>
>



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



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Date: Fri, 05 Dec 2003 09:51:56 -0800
From: Steven Low <slow@caltech.edu>
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Thanks everyone for the intresting discussions from various
perspectives.   What are your expert opinions on the coexistence
of a delay-based TCP with (let's assume, essentially) fixed rates
of media flows?   Granted that there are many challenges in filtering
out noise in delay measurements, etc, but these seem more
surmountable than the fundamental difficulty of a loss-based
scheme where TCP must periodically push the queue to overflow,
thus squeezing out other flows that are not as (or more)
aggressive than TCP.    In a delay-based scheme where (e.g.) a
TCP flow will stabilize when it maintains a certain amount of queue
(instead of pushing it over the cliff), it will then coexist peacefully
with media flows, each sharing the link at a rate that's proportional
to their buffer shares, without the protocol itself producing severe
oscillations.

Again, there are of course challenges to filter delay noise, ensure
fairness, etc, but it seems to avoid a more difficult problem of a
loss-based scheme.

Steven



Rosen, Brian wrote:

>Sally
>
>I think the primary control for media streams is going to be 
>admission control.  There is a missing link to make this work,
>which is that there is no feedback path from the media flow to the
>signalling (RTP to SIP/RTSP), but fundamentally, if a media flow
>experiences congestion, it will be downshifted to a lower rate 
>(once, maybe) or shut down.  It's pretty reasonable to expect that
>such congestion control could be widely deployed.  I think that
>is the way congestion collapse should be addressed.
>
>It's harder to deal with the relative fairness of TCP vs media flows.
>None of the mechanisms that I have heard proposed will work in my
>opinion.  Work means that you get acceptable voice/video quality up 
>to the point where congestion is bad enough that signalling shuts 
>down the flow.  Media flows can't deal with "fair" in the sense that 
>TCP does.  There is no "fair", there is only "all or nothing". 
>
>One possible avenue of exploration is to look more into aggregate
>behavior; you CAN probably assert a more TCP friendly "rate control"
>if you deal with the aggregate of media traffic, even if the individual
>components of the traffic experience all or nothing flow control.
>
>Ultimately, however, the actual media flow can't do anything interesting
>with existing TCP mechanisms.  It only has a very gross rate control,
>it can't probe, it's silly to send filler, and it can't use any form 
>of a window.  Similarly, TCP doesn't have any way to know if the media 
>flow is present, and thus adjust it's behavior appropriately.  
>"appropriately" means not being greedy; the biggest actual problem that 
>exists today is that TCP destroys voice, not that voice destroys TCP.
>I think, naively perhaps, that it is possible to design mechanisms 
>that would work together, but it is highly unlikely to be possible to
>do this from one side; TCP must change as well as UDP (or a replacement).
>
>I suspect that if a service provider is providing both a data service
>and a voice service over that data service, that it will drop all the
>TCP traffic before it drops any voice traffic, because, so far, the
>pricing for a voice bit is several orders of magnitude larger than
>a data bit.  If the data service provider is not also the voice service
>provider, it's not clear to me why he would care which customer bits
>get treated differently.  Conventional wisdom in carriers is that
>flat rate Internet service makes small profits, while voice/video 
>makes a lot more.  Don't know if that's really true, or will remain true,
>but I'll bet you will scream much more loudly if your voice/video
>sounds/looks bad than if your browser slows down.
>
>Of course, deployment of QoS mechanisms would be helpful. 
>
>Brian
>
>  
>
>>-----Original Message-----
>>From: Sally Floyd [mailto:floyd@icir.org]
>>Sent: Thursday, December 04, 2003 6:02 PM
>>To: Rosen, Brian
>>Cc: 'Eddie Kohler'; avt@ietf.org; dccp@ietf.org
>>Subject: Re: [dccp] RE: Would DCCP kill Internet real-time media
>>distribution? 
>>
>>
>>Brian -
>>
>>    
>>
>>>Slow start is not
>>>a practical thing for interactive use, sorry.
>>>      
>>>
>>I have not read all of the earlier email, so I might be repeating
>>things that have already been said.
>>
>>Currently, slow-start for TCP allows an initial window of up to
>>four packets.  The reason that the Internet still works plausibly
>>well even with interactive traffic that does not obey slow-start
>>is that the vast majority of the packets in the Internet still use
>>TCP, and still abide by slow-start and the other mechanisms of TCP
>>congestion control.  If the majority of the traffic on a congested
>>link was interactive traffic that did not use slow-start, and that
>>also did not use any admissions control or explicit feedback from
>>routers allowing high initial windows, I don't think that any of
>>the interactive users would be very happy with the results.  This
>>would particularly be true, I would conjecture, in a network with
>>only interactive traffic, with no TCP traffic to back off and make
>>space for the interactive traffic.
>>
>>For a best-effort network that includes TCP traffic, there is the
>>additional concern of the fairness for the TCP users.  There is no
>>reason why best-effort interactive UDP traffic has a higher right
>>to bandwidth or to low delay that does the best-effort TCP traffic
>>(whose users are paying the same monthly fees to their service
>>providers).  Remember that the TCP traffic could easily be someone
>>doing a web transaction that is perfectly important to them or to
>>their business, and for which they are expecting the same prompt
>>response times from the network that the interactive users are
>>expecting.
>>
>>My own prediction is that, if there seems to be a danger of congestion
>>in best-effort links due to best-effort UDP traffic that is not
>>TCP-friendly, the routers at the congested links would end up
>>preferentially dropping the UDP traffic in times of high congestion
>>(since this is easier than more precisely-targetted forms of
>>preferential dropping at the router).  But maybe things will work
>>out better than that, and routers will have mechanisms to precisely
>>identify those flows that aren't using decent end-to-end congestion
>>control in a time of high congestion, and preferentially drop packets
>>from only those flows.  No telling...
>>
>>- Sally
>>http://www.icir.org/floyd/
>>
>>
>>
>>    
>>
>
>_______________________________________________
>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
>
>  
>

-- 
_____________________________________________
Steven Low                assoc prof, cs & ee
http://netlab.caltech.edu tel: (626) 395-6767
slow@caltech.edu          fax: (626) 568-3603




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



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Date: Sun, 7 Dec 2003 14:18:39 +0000
From: Colin Perkins <csp@csperkins.org>
To: avt@ietf.org
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Subject: [AVT] WG last call: iLBC payload format and codec
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This is to announce a working group last call on the RTP Payload Format and
codec for iLBC speech:

	draft-ietf-avt-rtp-ilbc-04.txt
	draft-ietf-avt-ilbc-codec-04.txt

Please send any final comments on these drafts to the mailing list by 19th
December 2003.  If no substantive comments are received by that time, we
will ask the IESG to publish the codec as an Experimental RFC and the RTP
payload format as a Proposed Standard RFC.

-- 
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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Subject: [AVT] WG last call: draft-ietf-avt-rtp-clearmode-04.txt
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This is to announce a working group last call on the RTP payload format for
64 kbit/s transparent calls (draft-ietf-avt-rtp-clearmode-04.txt).  Please
send any final comments on this draft to the mailing list by 19th December
2003. If no substantive comments are received by that time, we will ask the
IESG to publish this draft as a Proposed Standard RFC. 

-- 
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>
To: avt@ietf.org
Cc: magnus.westerlund@ericsson.com
Subject: Re: [AVT] Last call for review: draft-jones-avt-audio-t38-02.txt
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The only comments received on this draft were minor editorial issues, which
are corrected in the -03 revision. Accordingly, we will recommend the IESG
publish draft-jones-avt-audio-t38-03.txt as a Proposed Standard RFC, under
the condition that the amended version of T.38 containing the RTP payload
format is published by the ITU.

Colin



--> Colin Perkins <csp@csperkins.org> writes:
> 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


-- 
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  Sun Dec  7 14:14:23 2003
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Date: Sun, 7 Dec 2003 19:13:02 +0000
From: Colin Perkins <csp@csperkins.org>
To: "Rolf Hakenberg" <hakenberg@panasonic.de>
Cc: gamze@vidiator.com, rey@panasonic.de, avt@ietf.org, David.Leon@nokia.com,
        viktor.varsa@nokia.com, magnus.westerlund@era.ericsson.se
Subject: Re: [AVT] Retransmission draft
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--> "Rolf Hakenberg" <hakenberg@panasonic.de> writes:
> Dear Gamze and all,
> 
> Although I believe that the concerns expressed in 3GPP on server
> complexity and delay are less relevant for our discussion on potential
> IPRs, please let me clarify the agreement in 3GPP.
> 
> At the last 3GPP SA4 meeting there was a consensus on to make RTP
> Retransmission a working assumption in PSS Rel-6, subject to the concerns
> on server complexity and delay being addressed at the next meeting.

Do you expect addressing these concerns to require changes to the draft?

> Important for our current discussion on potential IPRs and how to
> progress the draft is the fact that there was no condition on the IPR
> issues, but in fact conditions that the draft should move to RFC on
> standards track before the finalisation of 3GPP Rel-6.

Moving to the standards track requires resolution of the IPR issue within
AVT, of course.

-- 
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: Sun, 7 Dec 2003 19:23:06 +0000
From: Colin Perkins <csp@csperkins.org>
To: <rey@panasonic.de>
Cc: avt@ietf.org
Subject: Re: [AVT] Retransmission draft
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--> Jos=E9 Rey <rey@panasonic.de> writes:
...
> > > In order to make this situation clear to anybody who reads the RTP
> > > retransmission draft, we suggest to add a paragraph at the very
> > > beginning of the draft. This paragraph will make implementors aware
> > > of the situation that the solution described in this draft is
> > > possibly encumbered by IPRs. Furthermore it will point implementors
> > > to non-IPR-encumbered solutions as above, which may perfectly be used
> > > instead of RTP retransmission.
> >
> > Without seeing the wording of this paragraph, I can't give a
> > definitive answer, but that would largely address my concerns.
>=20
> right, how about this:
>=20
> "This document provides an RTP packet retransmission mechanism for repair
> of streaming media.  The purpose of this paragraph is to make the reader
> aware that the retransmission mechanism described in this document may be
> encumbered by patent applications filed from Matsushita.  For details on
> the IPR statement, please visit the IETF IPR web page
> http://www.ietf.org/ietf/IPR.
>=20
> However, packet retransmission is not the only means for streaming
> clients and servers to cope with packet losses.  A detailed description
> of options for repair of streaming media is given in RFC2354.  Therefore,
> implementers of this protocol shall take into account that there are in
> fact other mechanisms, such as forward error correction (FEC) or packet
> interleaving that represent an alternative to solve the problems packet
> retransmission tries to solve.    In particular RFC2733 and RFC2198
> present FEC and redundant packet transmission schemes that might well fit
> to the needs of the implementer. "

[takes working group chair hat off]

This, at least, would stop accidental infringement which is one of my main
concerns. I would still strongly prefer a royalty-free license, however.

Colin

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



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Subject: [AVT] 9th Brazilian Workshop on Network & Services Operations & Management
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* Apologies if you receive multiple copies of this announcement *

WGRS - Workshop on Network & Services Operations & Management

The 9th Brazilian Workshop on Network & Services Operations & Management
(WGRS'2004) is an annual event that has been held since 1995. Previously
known as WTMN (Workshop on Telecommunication Networks & Services
Management),
the event has been held either with the Brazilian Symposium on Computer
Networks (SBRC) or with the Latin American Network Operations and
Management
Symposium (LANOMS).

The main goal of WGRS is to provide a Brazilian forum in the area of
network management focused on the local audience, involving networking
and telecommunication companies, academic institutions, service providers
and equipment vendors.=20

This year, in its 9th edition, WGRS will take place in Gramado - Rio
Grande do Sul, during the 22th SBRC <http://www.sbrc2004.ufrgs.br>.
Besides the presentation of papers describing research results, on-going
projects and practical experience reports, a number of invited keynotes
from the industry is also planned.


Topics of interest include, but are not limited to:

   * Integrating Intrusion Detection and Network Management
   * Intrusion Policy Management
   * Web Services Security
   * Mobile/Wireless Network Management
   * Business and Service Management
   * Management Platforms, Interoperability, Standards and Protocols
   * Management of Networks and Services: VPNs, Optical Networks, CATV
Networks,
     Mobile and Ad-Hoc Networks, Voice over IP, Transaction- Oriented
Services,
     Electronic Commerce, Active and Programmable Networks, Sensor
Networks
   * Management Technologies and Frameworks (Web, XML, CORBA, DEN, etc)
   * Management Aspects of Service Pricing, Accounting and Billing
   * Management of Service Level Agreements and Quality of Service
   * Data Warehousing and Data Mining in Management
   * Programmable, Mobile and Intelligent Agents in Management
   * Network and Systems Monitoring
   * Fault and Performance Management
   * Security Management
   * Configuration Management and Policy-Driven Management
   * Information Modeling
   * Traffic and Performance Management
   * Open-Source Management Software
   * Management of Grid Computing, Clusters, Peer-to-Peer Applications,
and
     Ubiquitous Computing Environments
   * Operations Support Systems
   * Business Processes for Network and Service Management
   * Peer-to-peer Network Technologies for Management
   * Management of Distributed Systems
   * Case Studies and Experiences in Management

Paper Submission

Submitted manuscripts must be written in either English, Spanish
or Portuguese and the text should not exceed 8 pages, including the
abstract, figures, diagrams, references, etc (very small figures, as
well as bitmap format should be avoided). The page format should be A4
(29.7 x 21.0 cm), font size 12 pts, Times, lateral margins 2.5 cm,
upper margin 3.0 cm, lower margin 2.5 cm, with 12pt Times fonts,
single column page with no more than 45 lines.  The paper can be
prepared using one of the templates avaliable in the SBC webpage
(<a href=3D"http://www.sbc.org.br">http://www.sbc.org.br</a>). The
paper must be submitted eletronically through EDAS (at <a
href=3D"http://soyuz.metropoa.tche.br/sbrc2004_wgrs">
http://soyuz.metropoa.tche.br/sbrc2004_wgrs</a>).
The file must be necessarily in the postscript format (use generic
postscript) or PDF (Portable Document Format).</p>

Program Committee

   * Ant=F4nio A. F. Loureiro (UFMG)  loureiro@dcc.ufmg.br
   * Bruno Schulze (LNCC)  schulze@lncc.br
   * Carlos B. Westphall (UFSC)  westphal@lrg.ufsc.br
   * Carlos de Castro Goulart (UFV)  goulart@dpi.ufv.br
   * Edmundo R. M. Madeira (UNICAMP)  edmundo@ic.unicamp.br
   * Elias P. Duarte Jr. (UFPR) elias@inf.ufpr.br=20
   * Elizabeth S. Specialski (UFSC)  beth@inf.ufsc.br
   * Iara Machado (RNP)  iara@rnp.br
   * Jos=E9 N. de Souza (UFC)  neuman@ufc.br
   * Jos=E9 Marcos Silva Nogueira jmarcos@dcc.ufmg.br
   * Keiko V. O. Fonseca (CEFET/PR)  keiko@cpgei.cefetpr.br
   * Liane M. R. Tarouco (UFRGS) liane.tarouco@ufrgs.br
   * Lisandro Z. Granville (UFRGS)  granville@inf.ufrgs.br
   * Luciano P. Gaspary (UNISINOS)  paschoal@exatas.unisinos.br
   * Luiz Nacamura Jr. (CEFET/PR)  nacamura@cpgei.cefetpr.br

Important Dates=20

   * Paper Submission March 11 2004
   * Notification     April 8  2004
   * Camera-Ready due April 15 2004

Workshop Chair

   * Elias P. Duarte Jr. (UFPR)



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



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From: "Jose Rey" <rey@panasonic.de>
To: <vsathe@san.rr.com>
Cc: <avt@ietf.org>
Subject: RE: [AVT] Retransmission draft
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Hi Vinay,

thanks for your feedback. Let me try to clarify: the statement says there
are other alternatives to solve the problems the retransmission addresses.
However, this doesn't mean that these other solutions are better or worse.
It is a matter of trade-off between, e.g., client/server complexity, buff=
er
space and bit rate available.  And these depend on your scenario (!).
Typically, FEC and packet interleaving schemes offer less protection for =
the
same available bitrate, as pro-active schemes send redundant packets in
advance. You also need to XOR-encode and -decode packets.  FEC and packet
interleaving have a fixed repetition scheme, which means that packets are
repeated a fixed number of times (whether they are lost or not!) whereas
retransmission is more flexible here by allowing each lost packet to be
retransmitted as often as necessary within the given timing constraints. =
 An
advantage of pro-active schemes might be that they don't require a feedba=
ck
channel. On the other hand, retransmission of packets will use just the
bandwidth necessary to re-send lost packets.  You can also incorporate
packet scheduling into your retransmission server, which may enhance the
quality at the client.  In summary, you shall carefully address your need=
s
and evaluate which solution delivers the best compromise. You may want to
read RFC2354 which describes how to approach such a decision.

Hope this helps,

Jos=E9


> -----Original Message-----
> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of
> vsathe@san.rr.com
> Sent: Thursday, December 04, 2003 6:50 PM
> To: avt@ietf.org
> Subject: RE: [AVT] Retransmission draft
>
>
> With this disclaimer, I am now confused about the prespective
> with which I
> should read this draft as a (potential) implementator. I
> understand that
> Matsushita may have some IPR in this submission. But now are
> you saying
> that you really don't know why anyone should use this scheme?
> Would it be
> possible to give a scenario or two where the proposed mechanism may be
> superior to follow? What is the point in sending this off to IETF RFC
> track?
>
> Thanks in advance for your explanation.
>
> Vinay Sathe
> Multirate Systems
>
> right, how about this:
>
> "This document provides an RTP packet retransmission
> mechanism for repair of
> streaming media.  The purpose of this paragraph is to make
> the reader aware
> that the retransmission mechanism described in this document may be
> encumbered by patent applications filed from Matsushita.  For
> details on the
> IPR statement, please visit the IETF IPR web page
> http://www.ietf.org/ietf/IPR.
>
> However, packet retransmission is not the only means for
> streaming clients
> and servers to cope with packet losses.  A detailed
> description of options
> for repair of streaming media is given in RFC2354.
> Therefore, implementers
> of this protocol shall take into account that there are in fact other
> mechanisms, such as forward error correction (FEC) or packet
> interleaving
> that represent an alternative to solve the problems packet
> retransmission
> tries to solve.    In particular RFC2733 and RFC2198 present FEC and
> redundant packet transmission schemes that might well fit to
> the needs of
> the implementer. "
>
> Jos=E9
>
>
> --------------------------------------------------------------------
> mail2web - Check your email from the web at
> http://mail2web.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, 8 Dec 2003 10:46:01 +0000
From: Colin Perkins <csp@csperkins.org>
To: <rey@panasonic.de>
Cc: vsathe@san.rr.com, avt@ietf.org
Subject: Re: [AVT] Retransmission draft
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Jose,

--> "Jose Rey" <rey@panasonic.de> writes:
> thanks for your feedback. Let me try to clarify: the statement says there
> are other alternatives to solve the problems the retransmission
> addresses. However, this doesn't mean that these other solutions are
> better or worse. It is a matter of trade-off between, e.g., client/server
> complexity, buffer space and bit rate available.  And these depend on
> your scenario (!). Typically, FEC and packet interleaving schemes offer
> less protection for the same available bitrate, as pro-active schemes
> send redundant packets in advance. You also need to XOR-encode and
> -decode packets.  FEC and packet interleaving have a fixed repetition
> scheme, which means that packets are repeated a fixed number of times
> (whether they are lost or not!) whereas retransmission is more flexible
> here by allowing each lost packet to be retransmitted as often as
> necessary within the given timing constraints.  An advantage of
> pro-active schemes might be that they don't require a feedback channel.

Note that there's nothing in RFC 2733 that requires FEC to be sent with a
fixed repetition scheme. One could imagine a scenario where the receiver
used the RTCP feedback extensions to inform the sender of loss, and the
sender then generated FEC packets on demand to correct that loss...

Colin

_______________________________________________
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: "'mallman@icir.org'" <mallman@icir.org>
Cc: "'Eddie Kohler'" <kohler@icir.org>, avt@ietf.org, dccp@ietf.org
Subject: RE: [dccp] RE: [AVT] Would DCCP kill Internet real-time media dis
	tribution? 
Date: Mon, 8 Dec 2003 11:06:14 -0500 
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This is why slow start won't work.  The media starts with constant
rate.  If you have to slow start, you have to either drop packets
or you have to buffer them, and as I have pointed out, once
you buffer CBR, you keep it.  The delay introduced is worse than
the buffer, so you will drop packets to make slow start, and clip
the "hello". 

What you are saying, it seems to me, is that we have solved 
existing problems by deploying slow start.  All the 
applications so far have judged slow start as acceptable.  
Along comes real time interactive media, and we say sorry,
slow start is not acceptable.

And you want to claim then that our applications can't be 
used on the Internet?

I would say that you haven't a prayer of stopping us.

We can address congestion collapse another way, so you can't 
use that as an excuse over why we need to move off of UDP.  
I think those of us with real time interactive media are 
very willing to adopt additional congestion control mechanisms 
that attempt to deal with "fairness" (which is not viewed the 
same way by TCP and real time media), so long as they don't 
destroy the app.  However, we aren't likely to adopt anything 
replacing UDP if it does destroy the app.

Brian



> -----Original Message-----
> From: Mark Allman [mailto:mallman@icir.org]
> Sent: Monday, December 08, 2003 10:00 AM
> To: Rosen, Brian
> Cc: 'Eddie Kohler'; avt@ietf.org; dccp@ietf.org
> Subject: Re: [dccp] RE: [AVT] Would DCCP kill Internet real-time media
> distribution? 
> 
> 
> 
> (gluing together two things that I can't get to jive together in my
> head) 
> 
> > A typical RTT on the internet these days is 50-100 ms.  If you
> > experience 30% loss in the first 3 RTTs of voice, you will clip the
> > "hello" pretty badly.  If it's 7-8 RTTs, you will definitely have a
> > very poor experience.
> 
> I do not understand where this 30% loss number comes from.  If a host
> experiences 30% loss the network is *quite* congested (recent
> measurments I have done show that loss rate over 10% are very rare in
> the Internet) or the connection is crossing a very ratty error-prone
> link.  I think the latter is unlikely in the general case.  So, how
> would this:
> 
> > Yep; bang! 0 to 4Mbit in two packets.
> 
> be useful?  If you're already clipping "hello" then this is going to
> provide a very lousy experience, right?
> 
> Others have said it already, so I'll just briefly agree: In a
> best-effort network I do not think any particular flow (TCP or DCCP or
> UDP or SCTP or whatever) should be able to just start sending 
> at a "high
> rate" (where "high rate" is undefined, but 4Mbps is certainly 
> high).  I
> think it would not be terribly difficult to show that such behavior
> could easily cause problems (to competing TCP, to competing streaming
> flows and to the flow that is doing this "instant on").  Slow start is
> the current consensus for how to start from zero.  Could 
> there be other
> schemes?  Sure.  Are those here and fully fleshed out such 
> that we could
> even consider them as potential solutions?  Not right now.  
> 
> My hunch is that sending DCCP (/anything) through without 
> slow start is
> not going to work (i.e., I think consensus would be against 
> that).  So,
> if your app can't deal with that then I think we might be stuck ...  A
> more general wondering of mine is how many streaming apps require this
> "instant on" behavior (i.e., is your app an anomaly or is 
> there a whole
> class of apps that want instant-on over best-effort networks?)?
> 
> Thanks!
> 
> allman
> 
> 
> --
> Mark Allman -- ICIR -- http://www.icir.org/mallman/
> 
> _______________________________________________
> 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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 Sub-type  Registration' to Proposed Standard
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FYI

------- Forwarded message:
>Date: Mon, 08 Dec 2003 11:02:05 -0500
>From: The IESG <iesg-secretary@ietf.org>
>To: IETF-Announce :;
>Cc: 
>Subject: Last Call: 'Real-Time Facsimile (T.38) - audio/t38 MIME Sub-type 
>Registration' to Proposed Standard 
>
> The IESG has received a request from an individual submitter to consider 
> the following document:
> 
> - 'Real-Time Facsimile (T.38) - audio/t38 MIME Sub-type Registration '
>    <draft-jones-avt-audio-t38-03.txt> as a Proposed Standard
> 
> Though this is an individual submission, it has been reviewed and
> supported by the Audio Video Transport Working Group.
> 
> The IESG plans to make a decision in the next few weeks, and solicits
> final comments on this action.  Please send any comments to the
> iesg@ietf.org or ietf@ietf.org mailing lists by 2004-01-05.
> 
> The file can be obtained via
> http://www.ietf.org/internet-drafts/draft-jones-avt-audio-t38-03.txt
> 
------- End of Forwarded Message

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



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(gluing together two things that I can't get to jive together in my
head) 

> A typical RTT on the internet these days is 50-100 ms.  If you
> experience 30% loss in the first 3 RTTs of voice, you will clip the
> "hello" pretty badly.  If it's 7-8 RTTs, you will definitely have a
> very poor experience.

I do not understand where this 30% loss number comes from.  If a host
experiences 30% loss the network is *quite* congested (recent
measurments I have done show that loss rate over 10% are very rare in
the Internet) or the connection is crossing a very ratty error-prone
link.  I think the latter is unlikely in the general case.  So, how
would this:

> Yep; bang! 0 to 4Mbit in two packets.

be useful?  If you're already clipping "hello" then this is going to
provide a very lousy experience, right?

Others have said it already, so I'll just briefly agree: In a
best-effort network I do not think any particular flow (TCP or DCCP or
UDP or SCTP or whatever) should be able to just start sending at a "high
rate" (where "high rate" is undefined, but 4Mbps is certainly high).  I
think it would not be terribly difficult to show that such behavior
could easily cause problems (to competing TCP, to competing streaming
flows and to the flow that is doing this "instant on").  Slow start is
the current consensus for how to start from zero.  Could there be other
schemes?  Sure.  Are those here and fully fleshed out such that we could
even consider them as potential solutions?  Not right now.  

My hunch is that sending DCCP (/anything) through without slow start is
not going to work (i.e., I think consensus would be against that).  So,
if your app can't deal with that then I think we might be stuck ...  A
more general wondering of mine is how many streaming apps require this
"instant on" behavior (i.e., is your app an anomaly or is there a whole
class of apps that want instant-on over best-effort networks?)?

Thanks!

allman


--
Mark Allman -- ICIR -- http://www.icir.org/mallman/

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



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Mark,

very interesting reading this - in paticular, your comment:

"A more general wondering of mine is how many streaming apps require =
this
"instant on" behavior (i.e., is your app an anomaly or is there a whole
class of apps that want instant-on over best-effort networks?)?"

I totally agree. Whilst "instant on" is really important [at the =
application layer] for streaming / VoD applications with trick-modes =
such as fast-forward, rewind, etc. (don't want buffering every time you =
want to scan / hop channel etc.) it should be up to the application to =
provide that type of functionality. It should not be up to the network / =
transport to cater for this.

With the new video codecs that are coming on-line (H.264 / MPEG4 pt10) =
and the associated streaming profiles (which enable a streaming =
application to switch between encoded data-rates)  it should be =
reasonable to design streaming apps that will provide near-instant on, =
but still adhere to the slow-start rules of whatever the transport =
protocol offers.

So, it is my belief that streaming apps do not require instant-on =
throughput, and should adhere to the same rules as other applications.

rgds

Matt Walker=20

-----Original Message-----
From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of Mark
Allman
Sent: 08 December 2003 15:00
To: Rosen, Brian
Cc: 'Eddie Kohler'; avt@ietf.org; dccp@ietf.org
Subject: Re: [dccp] RE: [AVT] Would DCCP kill Internet real-time media
distribution?=20



(gluing together two things that I can't get to jive together in my
head)=20

> A typical RTT on the internet these days is 50-100 ms.  If you
> experience 30% loss in the first 3 RTTs of voice, you will clip the
> "hello" pretty badly.  If it's 7-8 RTTs, you will definitely have a
> very poor experience.

I do not understand where this 30% loss number comes from.  If a host
experiences 30% loss the network is *quite* congested (recent
measurments I have done show that loss rate over 10% are very rare in
the Internet) or the connection is crossing a very ratty error-prone
link.  I think the latter is unlikely in the general case.  So, how
would this:

> Yep; bang! 0 to 4Mbit in two packets.

be useful?  If you're already clipping "hello" then this is going to
provide a very lousy experience, right?

Others have said it already, so I'll just briefly agree: In a
best-effort network I do not think any particular flow (TCP or DCCP or
UDP or SCTP or whatever) should be able to just start sending at a "high
rate" (where "high rate" is undefined, but 4Mbps is certainly high).  I
think it would not be terribly difficult to show that such behavior
could easily cause problems (to competing TCP, to competing streaming
flows and to the flow that is doing this "instant on").  Slow start is
the current consensus for how to start from zero.  Could there be other
schemes?  Sure.  Are those here and fully fleshed out such that we could
even consider them as potential solutions?  Not right now. =20

My hunch is that sending DCCP (/anything) through without slow start is
not going to work (i.e., I think consensus would be against that).  So,
if your app can't deal with that then I think we might be stuck ...  A
more general wondering of mine is how many streaming apps require this
"instant on" behavior (i.e., is your app an anomaly or is there a whole
class of apps that want instant-on over best-effort networks?)?

Thanks!

allman


--
Mark Allman -- ICIR -- http://www.icir.org/mallman/

_______________________________________________
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



From exim@www1.ietf.org  Mon Dec  8 12:08:28 2003
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Hi,

> And, I'd still love to hear from others as to whether this "instant-on"
> is a requirement across the board or whether you're a special case.

Okay, my view on this topic about the instant-on is the following.

Streaming applications, i.e. with buffers, should do reasonably well 
with a slow start as it will most likely only effect the initial 
buffering delay.

When it comes to interactive real-time applications, i.e. without 
buffers, I think the Brian's view is mostly correct. All applications in 
this area does need to try to transmit at there lowest supported 
bit-rate initially. Then the question is what the lowest bit-rate is. It 
is not 4 mbit as somebody claimed for a high-rate conferencing 
application. Because in that case there exist techniques that will allow 
one to use a slow starting type of performance, which trades initial 
video quality against bandwidth during the slow start. For the 4 mbit 
video stream the initial bit-rate may be 256-512 kbits or possibly even 
lower. This will be a very coarse quality. but as one continues to 
transmit during the slow start and discover more bit-rate one can 
increase the quality. And I don't see this as having the same user 
problems as bit-rate switching is claimed to have. I think that any user 
of a circuit switched video telephony client are rather used to this 
behaviour, when major changes in video happens.

It will be a bigger problem for certain high quality audio codecs that 
are used. If the codec does not support to do bit-rate/quality 
trade-offs then it probably not very suitable for usage in these best 
effort networks. There exist codecs that will allow reasonable bit-rate 
steps to be used.

I think that the realistic thing in this environment is to be reasonable 
in both directions. I think that one has to understand that one a best 
effort network the media apps can't continue to behave irresponsible and 
simply blast away with UDP. It is better to start accept the fact that 
there will be need for a trade off. Then, if one needs to blast away, 
then one needs to develop more solutions or use resource reservation.

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



From exim@www1.ietf.org  Mon Dec  8 12:30:46 2003
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> What you are saying, it seems to me, is that we have solved existing
> problems by deploying slow start.  All the applications so far have
> judged slow start as acceptable.  Along comes real time interactive
> media, and we say sorry, slow start is not acceptable.

No, I was not saying that slow start has solved all the problems.  Are
there potential other methods to do startup?  Certainly there are.  Are
there new reasons to think about different ways to startup?  Probably.
But, I don't think any new startup method is currently baked enough
(and, certainly I don't think any of them have been introduced in the
IETF as baked and ready to go).  That does not mean that I think slow
start is the way to go forever.  That is just a pragmatic statement of
my feeling that the consensus of the community is that slow start is the
way to initiate transmission in the current best-effort Internet.

> And you want to claim then that our applications can't be used on the
> Internet?
> 
> I would say that you haven't a prayer of stopping us.

Look, I'm not trying to pick a fight.  This Internet thing is a big
cooperative.  You are writing an app.  I cannot stop you from writing it
or running it on the public Internet.  But, if the app does bad things
to networks then the operators can and will do something about it.
(E.g., look at what has been done to p2p stuff on college campuses.)
And, we know that CBR UDP apps have the potential to do a bunch of
damage.  So, it seems that DCCP has its place.

> We can address congestion collapse another way, so you can't use that
> as an excuse over why we need to move off of UDP.  I think those of us
> with real time interactive media are very willing to adopt additional
> congestion control mechanisms that attempt to deal with "fairness"
> (which is not viewed the same way by TCP and real time media), so long
> as they don't destroy the app.  However, we aren't likely to adopt
> anything replacing UDP if it does destroy the app.

"Destroy the app" is one end of the spectrum, I think.  "Not impact the
app at all" is the other.  And, the answer is likely somewhere in
between. 

In the end, congestion control is not about getting exactly what each
app desires.  It's about keeping the network working in a reasonable
fashion so that all the apps get reasonable performance.

And, I'd still love to hear from others as to whether this "instant-on"
is a requirement across the board or whether you're a special case.

allman


--
Mark Allman -- ICIR -- http://www.icir.org/mallman/

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



From exim@www1.ietf.org  Mon Dec  8 12:52:17 2003
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Date: Mon, 08 Dec 2003 09:47:22 -0800
To: matthew.d.walker@bt.com
From: Stephan Wenger <stewe@cs.tu-berlin.de>
Subject: RE: [dccp] RE: [AVT] Would DCCP kill Internet real-time media
  distribution? 
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Matthew,

At 09:04 AM 12/8/2003, matthew.d.walker@bt.com wrote:

>With the new video codecs that are coming on-line (H.264 / MPEG4 pt10) and 
>the associated streaming profiles (which enable a streaming application to 
>switch between encoded data-rates)  it should be reasonable to design 
>streaming apps that will provide near-instant on, but still adhere to the 
>slow-start rules of whatever the transport protocol offers.

While I'm a big fan of the extended profile of H.264, and particularly the 
SI/SP picture technology, I do not believe that there will be any 
significant deployment of these technologies in the next few years.  And 
even if there is, the problem of varying perceived qualities due to rate 
changes, and the related user acceptance problems, still exist.  (The same 
is true for all these fancy layered coding concepts that are around since 
10 years or so.  There was never any significant deployment.  I wonder, why.).

Stephan



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From exim@www1.ietf.org  Mon Dec  8 13:27:30 2003
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>, mallman@icir.org
Cc: "'Eddie Kohler'" <kohler@icir.org>, avt@ietf.org, dccp@ietf.org
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Magnus

You saw the message from Henning on the effect of slow start on
conferencing applications didn't you?  What makes you think
the video is any different?

In a conference application, streams come and go as different speakers
come and go.  Even in a centrally mixed conference, which is not
as demanding as an end-system mixed conference, most participants
are not talking, and not sending video.  When they start talking,
the start sending media, voice and video.  To send media all the
time unnecessarily uses bandwidth; a lot of it.

We don't do this now, we are end system mixed.

Even your basic assumption that coarse video is acceptable for
a while is not true; it looks very poor.

Actually, I'm not aware of any conferencing video systems that ramp
bandwidth.  The reason is that the codecs are parameterized for a given
quality level/bandwidth, and you can't smoothly change the parameters -
you set them and restart the codec.  I suspect that you could design
a codec to smoothly ramp, but that's not what we have available to us
now.  What you do see sometimes is a codec which has a fixed bandwidth
but the algorithm itself sends coarse image more often than fine image,
and you see that effect on startup.  That is very undesirable, but
it exists.

Brian

> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Monday, December 08, 2003 12:07 PM
> To: mallman@icir.org
> Cc: Rosen, Brian; 'Eddie Kohler'; avt@ietf.org; dccp@ietf.org
> Subject: Re: [dccp] RE: [AVT] Would DCCP kill Internet real-time media
> dis tribution?
> 
> 
> Hi,
> 
> > And, I'd still love to hear from others as to whether this 
> "instant-on"
> > is a requirement across the board or whether you're a special case.
> 
> Okay, my view on this topic about the instant-on is the following.
> 
> Streaming applications, i.e. with buffers, should do reasonably well 
> with a slow start as it will most likely only effect the initial 
> buffering delay.
> 
> When it comes to interactive real-time applications, i.e. without 
> buffers, I think the Brian's view is mostly correct. All 
> applications in 
> this area does need to try to transmit at there lowest supported 
> bit-rate initially. Then the question is what the lowest 
> bit-rate is. It 
> is not 4 mbit as somebody claimed for a high-rate conferencing 
> application. Because in that case there exist techniques that 
> will allow 
> one to use a slow starting type of performance, which trades initial 
> video quality against bandwidth during the slow start. For the 4 mbit 
> video stream the initial bit-rate may be 256-512 kbits or 
> possibly even 
> lower. This will be a very coarse quality. but as one continues to 
> transmit during the slow start and discover more bit-rate one can 
> increase the quality. And I don't see this as having the same user 
> problems as bit-rate switching is claimed to have. I think 
> that any user 
> of a circuit switched video telephony client are rather used to this 
> behaviour, when major changes in video happens.
> 
> It will be a bigger problem for certain high quality audio 
> codecs that 
> are used. If the codec does not support to do bit-rate/quality 
> trade-offs then it probably not very suitable for usage in these best 
> effort networks. There exist codecs that will allow 
> reasonable bit-rate 
> steps to be used.
> 
> I think that the realistic thing in this environment is to be 
> reasonable 
> in both directions. I think that one has to understand that 
> one a best 
> effort network the media apps can't continue to behave 
> irresponsible and 
> simply blast away with UDP. It is better to start accept the 
> fact that 
> there will be need for a trade off. Then, if one needs to blast away, 
> then one needs to develop more solutions or use resource reservation.
> 
> 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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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'mallman@icir.org'" <mallman@icir.org>
Cc: "'Eddie Kohler'" <kohler@icir.org>, avt@ietf.org, dccp@ietf.org
Subject: RE: [dccp] RE: [AVT] Would DCCP kill Internet real-time media dis
	 tribution? 
Date: Mon, 8 Dec 2003 13:47:37 -0500 
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Henning had a message on these threads that pointed out that in
conferencing,
you don't send until you talk.  On a conference with 100 participants, you
only see 1-4 or so actually sending, but which 1-4 changes, sometimes
rapidly.
You can't slow start them when they start talking or you clip.

Another example is Push To Talk.

Another example of non-interactive media is channel surfing.
This one depends somewhat on implementation, but many implementations
have different channels coming from different servers.  

I don't want to seem unwilling to change our app, I am very willing
to change the app to better work on the Internet, but not if it
destroys the user experience.  I don't see a way to do that 
without modifying TCP.  If you assume TCP is fixed, and you need 
media apps to live with TCP as it is, I don't see how to do that, 
but I'm certainly eager to be shown how. 

My app is the 4M app, and there aren't many of us at the moment,
but if you saw it, you'd want it, and there is no reason it shouldn't
work on the Internet.  It does now, after a fashion.

And, as I've pointed out, since these kinds of apps represent much higher
revenue per bit that the TCP-type apps, it's much more likely that the
network operators would deploy TCP stompers than media stompers.
YMMV.

Brian

> -----Original Message-----
> From: Mark Allman [mailto:mallman@icir.org]
> Sent: Monday, December 08, 2003 11:36 AM
> To: Rosen, Brian
> Cc: 'Eddie Kohler'; avt@ietf.org; dccp@ietf.org
> Subject: Re: [dccp] RE: [AVT] Would DCCP kill Internet real-time media
> dis tribution? 
> 
> 
>  
> > What you are saying, it seems to me, is that we have solved existing
> > problems by deploying slow start.  All the applications so far have
> > judged slow start as acceptable.  Along comes real time interactive
> > media, and we say sorry, slow start is not acceptable.
> 
> No, I was not saying that slow start has solved all the problems.  Are
> there potential other methods to do startup?  Certainly there 
> are.  Are
> there new reasons to think about different ways to startup?  Probably.
> But, I don't think any new startup method is currently baked enough
> (and, certainly I don't think any of them have been introduced in the
> IETF as baked and ready to go).  That does not mean that I think slow
> start is the way to go forever.  That is just a pragmatic statement of
> my feeling that the consensus of the community is that slow 
> start is the
> way to initiate transmission in the current best-effort Internet.
> 
> > And you want to claim then that our applications can't be 
> used on the
> > Internet?
> > 
> > I would say that you haven't a prayer of stopping us.
> 
> Look, I'm not trying to pick a fight.  This Internet thing is a big
> cooperative.  You are writing an app.  I cannot stop you from 
> writing it
> or running it on the public Internet.  But, if the app does bad things
> to networks then the operators can and will do something about it.
> (E.g., look at what has been done to p2p stuff on college campuses.)
> And, we know that CBR UDP apps have the potential to do a bunch of
> damage.  So, it seems that DCCP has its place.
> 
> > We can address congestion collapse another way, so you 
> can't use that
> > as an excuse over why we need to move off of UDP.  I think 
> those of us
> > with real time interactive media are very willing to adopt 
> additional
> > congestion control mechanisms that attempt to deal with "fairness"
> > (which is not viewed the same way by TCP and real time 
> media), so long
> > as they don't destroy the app.  However, we aren't likely to adopt
> > anything replacing UDP if it does destroy the app.
> 
> "Destroy the app" is one end of the spectrum, I think.  "Not 
> impact the
> app at all" is the other.  And, the answer is likely somewhere in
> between. 
> 
> In the end, congestion control is not about getting exactly what each
> app desires.  It's about keeping the network working in a reasonable
> fashion so that all the apps get reasonable performance.
> 
> And, I'd still love to hear from others as to whether this 
> "instant-on"
> is a requirement across the board or whether you're a special case.
> 
> allman
> 
> 
> --
> Mark Allman -- ICIR -- http://www.icir.org/mallman/
> 

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



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Hi Brian,

> In a conference application, streams come and go as different speakers
> come and go.  Even in a centrally mixed conference, which is not
> as demanding as an end-system mixed conference, most participants
> are not talking, and not sending video.  When they start talking,
> the start sending media, voice and video.  To send media all the
> time unnecessarily uses bandwidth; a lot of it.

I guess I've never used more-than-two site video conferencing.  You're
saying that when a site stops talking its video goes dark (or still)?

Tom P.

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



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> I totally agree. Whilst "instant on" is really important [at 
> the application layer] for streaming / VoD applications with 
> trick-modes such as fast-forward, rewind, etc. (don't want 
> buffering every time you want to scan / hop channel etc.) it 
> should be up to the application to provide that type of 
> functionality. It should not be up to the network / transport 
> to cater for this.
Fast forward/rewind functions can probably maintain an existing
stream, or even slow start.  What wouldn't work is channel surfing.
We know from experience that unless this is very fast, it's
a big problem.  The digital cable guys know that even the startup
time of the MPEG codec is a problem for them, let alone additional
delay caused by slow start.  Usually, not all channels are served
from the same server, so channel surfing has to start/stop streams
fast.

Brian

_______________________________________________
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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: avt@ietf.org, dccp@ietf.org
Subject: RE: [dccp] RE: [AVT] Would DCCP kill Internet real-time media dis
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Depends on the implementation.

If you have 100 participants, not all of them are on the screen,
only 2-10 or so are.  When one of the participants that are not
on screen start talking, they show up.  Optimized implementations
would not send the video unless you start talking in such a case.
Imagine the effect of 100 participants even at 512K if 1/10 of them
sent video vs all of them.

Brian

> -----Original Message-----
> From: Phelan, Tom [mailto:tphelan@sonusnet.com]
> Sent: Monday, December 08, 2003 1:51 PM
> To: 'Rosen, Brian'
> Cc: avt@ietf.org; dccp@ietf.org
> Subject: RE: [dccp] RE: [AVT] Would DCCP kill Internet real-time media
> dis tribution?
> 
> 
> Hi Brian,
> 
> > In a conference application, streams come and go as 
> different speakers
> > come and go.  Even in a centrally mixed conference, which is not
> > as demanding as an end-system mixed conference, most participants
> > are not talking, and not sending video.  When they start talking,
> > the start sending media, voice and video.  To send media all the
> > time unnecessarily uses bandwidth; a lot of it.
> 
> I guess I've never used more-than-two site video conferencing.  You're
> saying that when a site stops talking its video goes dark (or still)?
> 
> 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



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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'mallman@icir.org'" <mallman@icir.org>
Cc: "'Eddie Kohler'" <kohler@icir.org>, avt@ietf.org, dccp@ietf.org
Subject: RE: [dccp] RE: [AVT] Would DCCP kill Internet real-time media dis
	 tribution? 
Date: Mon, 8 Dec 2003 17:57:18 -0500 
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Mark

The reality of networks these days is that plain jane best effort 
Internet doesn't make much money.  Premium services do.  So the 
trend, which is clearly not universal, but is pretty obvious, 
is that SPs are trying to bring new bandwidth users onto their 
networks where the extra bandwidth is not just more bits, but 
for which they can charge for service beyond best effort packet.

Voice is one of those services, and video is one of those 
services.  You don't see a whole lot of investment these days 
in plain Internet service, and you do in voice/video over IP 
networks.  So, while there are pure play best effort operators 
out there, and they will be there, the money is in the premium 
services.  Even if an ISP does not directly provide premium 
services, other SPs will, using the underlying packet carrier
as the pipe, and the USERS of the premium services will yell 
loudly if their premium services are not delivering what they 
pay for.  If a customer of voice/video service gets told that 
his pipe provider is intentionally degrading his service, what 
do you think he will do with his pipe business?

Everyone agrees, I think, the most desirable situation is 
one in which we have a true converged network, not overlay 
networks, not separate networks, for these new services.  
I'm sure that there will be a lot of overlay networks around 
for quite a while, because we don't have the mechanisms we are 
discussing here - ways for classic best effort TCP to co-exist 
well with real time media.  Everyone wins if we figure out good 
ways for this to happen.  Everyone loses if any SP starts 
intentionally degrading one kind of service to make another
kind of service work better.  That's why I don't think it's 
going to be common to stomp on media OR TCP.  Rather, I think 
everyone expects us to do our jobs well and make both kinds of 
streams work well together.

We aren't going to solve this problem with anything done on one side
folks; both sides are going to have to have new mechanisms.

Brian

> -----Original Message-----
> From: Mark Allman [mailto:mallman@icir.org]
> Sent: Monday, December 08, 2003 5:02 PM
> To: Rosen, Brian
> Cc: 'Eddie Kohler'; avt@ietf.org; dccp@ietf.org
> Subject: Re: [dccp] RE: [AVT] Would DCCP kill Internet real-time media
> dis tribution? 
> 
> 
> 
> Brian-
> 
> > And, as I've pointed out, since these kinds of apps represent much
> > higher revenue per bit that the TCP-type apps, it's much more likely
> > that the network operators would deploy TCP stompers than media
> > stompers.  YMMV.
> 
> I saw your assertion in response to Sally a while back in this thread.
> 
> But, I think we have a disconnect.  I am generally thinking 
> in terms of
> best-effort networks (as I think Sally was).  Further, a network that
> has no priorities or traffic classes, but rather just tries 
> its hardest
> to get all the data across.  This is the kind of network where
> congestion control is important.  And, this is the kind of 
> network where
> I think you'll see "stompers" that attempt to surpress flows that are
> attempting to use a disproportionate amount of the bandwdith (whether
> that be streaming or mis-behaving TCP or ddos floods or whatever).
> 
> Your claim above does not match my model of a best effort network.
> I.e., if people are willing to pay more for the traffic of some
> particular app and you provision appropriately then you no longer have
> just a dumb best-effort network.  If you can engineer a network and
> provision appropriately then by all means use UDP and go from zero to
> 4Mbps instantly.  However, I don't see that as supporting the 
> assertion
> that DCCP can't be used for streaming across best-effort networks that
> are not engineered to give different levels of support to different
> classes of traffic.
> 
> allman
> 
> 
> --
> Mark Allman -- ICIR -- http://www.icir.org/mallman/
> 
> _______________________________________________
> 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  Mon Dec  8 19:50:22 2003
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I think the whole notion of fairness and value also needs to be looked 
at in a more differentiated fashion. Some of this has been alluded to, 
by Mark and others, but there seem to be at least three bottleneck cases:

- single subscriber bottlenecks, such as my DSL uplink; there, it should 
be my decision as to whether I'd like to trash my TCP performance to 
benefit UDP or not - I paid for that link. Having a queueing mechanism 
that ensures that TCP does not get starved, but can use available 
bandwidth, would seem to address most needs, without an explicit 
reservation mechanism. If I asked for an N Mb/s video stream, I 
presumably know what I was doing and really wanted this. If I don't like 
the webbrowsing speed while watching the video, I'll turn it off.

- multiple subscriber, but local bottlenecks, e.g., the access link 
owned by the business or university. Single-session TCP fairness makes 
very little sense since it simply encourages the use of multiple TCP 
sessions. Also, from a user experience perspective, why should 
instantaneous fairness be the right metric? I suspect that most people 
would consider long-term metrics as far more appropriate, e.g., their 
monthly bit budget. After all, this is what determines the overall 
expenditure of resources, at least for large multiplexing.

For example, at Columbia, we switched from limiting the instantaneous 
bandwidth available to dorms to a monthly bit budget. Exceed it, and get 
put into the low-priority queue. This seems to work pretty well and 
avoids having to distinguish 'good' from 'bad' traffic. I suspect that a 
simple end system or access router policer that rate-limits 
non-responsive traffic will do ok for large user populations. Thus, no 
sending of UDP at 100 Mb/s to the Internet or I2, but few applications 
seem to need that. This again requires no resource reservation if the 
traffic engineering can be made to work.

- bottlenecks elsewhere: there, session-level instantaneous fairness is 
probably the best we can do. However, US backbone providers claim very 
low average packet loss rates (below 1%), so it doesn't seem to be the 
network part throttling TCP at this point. If this is not true, we have 
a real problem that requires either an explicit business decision of the 
kind that Brian advocates or, conversely, penalty measures that 
discourage high-rate UDP flows.


Rosen, Brian wrote:

> Mark
> 
> The reality of networks these days is that plain jane best effort 
> Internet doesn't make much money.  Premium services do.  So the 
> trend, which is clearly not universal, but is pretty obvious, 
> is that SPs are trying to bring new bandwidth users onto their 
> networks where the extra bandwidth is not just more bits, but 
> for which they can charge for service beyond best effort packet.
> 
> Voice is one of those services, and video is one of those 
> services.  You don't see a whole lot of investment these days 
> in plain Internet service, and you do in voice/video over IP 
> networks.  So, while there are pure play best effort operators 
> out there, and they will be there, the money is in the premium 
> services.  Even if an ISP does not directly provide premium 
> services, other SPs will, using the underlying packet carrier
> as the pipe, and the USERS of the premium services will yell 
> loudly if their premium services are not delivering what they 
> pay for.  If a customer of voice/video service gets told that 
> his pipe provider is intentionally degrading his service, what 
> do you think he will do with his pipe business?
> 
> Everyone agrees, I think, the most desirable situation is 
> one in which we have a true converged network, not overlay 
> networks, not separate networks, for these new services.  
> I'm sure that there will be a lot of overlay networks around 
> for quite a while, because we don't have the mechanisms we are 
> discussing here - ways for classic best effort TCP to co-exist 
> well with real time media.  Everyone wins if we figure out good 
> ways for this to happen.  Everyone loses if any SP starts 
> intentionally degrading one kind of service to make another
> kind of service work better.  That's why I don't think it's 
> going to be common to stomp on media OR TCP.  Rather, I think 
> everyone expects us to do our jobs well and make both kinds of 
> streams work well together.
> 
> We aren't going to solve this problem with anything done on one side
> folks; both sides are going to have to have new mechanisms.
> 
> Brian
> 


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



From exim@www1.ietf.org  Mon Dec  8 20:50:25 2003
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In-Reply-To: Message from "Rosen, Brian" <Brian.Rosen@marconi.com> 
   of "Mon, 08 Dec 2003 13:47:37 EST." <313680C9A886D511A06000204840E1CF070B61A6@whq-msgusr-02.pit.comms.marconi.com> 
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Brian:
> I am very willing to change the app to better work on the Internet,

News to me. What kinds of changes are you willing to make?

Henning:
> - single subscriber bottlenecks, such as my DSL uplink; there, it should 
> be my decision as to whether I'd like to trash my TCP performance to 
> benefit UDP or not - I paid for that link.

> - multiple subscriber, but local bottlenecks, e.g., the access link 
> owned by the business or university.

> - bottlenecks elsewhere: there, session-level instantaneous fairness is 
> probably the best we can do.

Of course a single flow might encounter multiple bottlenecks.  In
well-provisioned backbones there may or may not be loss, but a lot of the
world doesn't work that way, and best-effort backbones probably won't grow
monotonically less lossy over time.

Eddie

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



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From: Eddie Kohler <kohler@icir.org>
Subject: [AVT] Faster Start after idle periods
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[Replies to dccp@ietf.org only please.]

Sally and I talked today about the slow-starting after an idle period.  We
have a potential solution that doesn't involve sending filler traffic to
keep the pipe open.

Say your app had a sending rate of K packets/RTT. Then you have an idle
period, possibly up to several minutes; and you want to start sending
again.
(1) Start sending with a window of 8 packets/RTT (not 4).
(2) Increase the window size by 4x after each lossless RTT (not 2x).

This is twice as fast as slow start.  But the multiplicative factor means
that if slow start reached a rate in N RTTs, faster start would reach the
same rate in sqrt(N) RTTs.  A window of 128 packets/RTT is reached after 2
lossless RTTs (for a large RTT of 100ms and maximum-size packets, this is
>14Mb/s after .2s).

The theoretical justification for this is that the connection recently
proved it could sustain rate K on this path.

Caveats:
(1) You must switch out of faster start (to slow start) once you reach the
    original rate K.
(2) You must switch out of faster start (to AI) after any congestion event.
(3) You can't use faster start if your last faster-start encountered
    congestion, and didn't achieve the original rate K.  (Slow start
    instead.)
(4) You probably can't use faster start if you didn't actually sustain rate
    K for long before.
(5) Possibly others.

It's not immediate-start.  It doesn't change the slow start that the
connection must go through initially.  It does greatly improve the dynamics
of DCCP connections that go idle (push-to-talk, interactive media), without
requiring filler traffic.

Details remain to be ironed out.  Help or simulations welcome.  Comments
welcome too.

Eddie

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



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> Caveats:
> ...
> (5) Possibly others.

I forgot to mention:
(4.5) You'll have to send "pings" -- one packet an RTT or so -- to verify
    that connectivity still exists.  In many cases the acks you send for
    the other endpoint's data will suffice.

_______________________________________________
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----- Original Message -----
From: "Mark Allman" <mallman@icir.org>
To: "Rosen, Brian" <Brian.Rosen@marconi.com>
Cc: "'Eddie Kohler'" <kohler@icir.org>; <avt@ietf.org>; <dccp@ietf.org>
Sent: Monday, December 08, 2003 8:29 PM
Subject: Re: [dccp] RE: [AVT] Would DCCP kill Internet real-time media
distribution?


>
> (gluing together two things that I can't get to jive together in my
> head)
>
> > A typical RTT on the internet these days is 50-100 ms.  If you
> > experience 30% loss in the first 3 RTTs of voice, you will clip the
> > "hello" pretty badly.  If it's 7-8 RTTs, you will definitely have a
> > very poor experience.
>
> I do not understand where this 30% loss number comes from.  If a host
> experiences 30% loss the network is *quite* congested (recent
> measurments I have done show that loss rate over 10% are very rare in
> the Internet) or the connection is crossing a very ratty error-prone
> link.  I think the latter is unlikely in the general case.  So, how
> would this:
>
> > Yep; bang! 0 to 4Mbit in two packets.
>
> be useful?  If you're already clipping "hello" then this is going to
> provide a very lousy experience, right?
>
> Others have said it already, so I'll just briefly agree: In a
> best-effort network I do not think any particular flow (TCP or DCCP or
> UDP or SCTP or whatever) should be able to just start sending at a "high
> rate" (where "high rate" is undefined, but 4Mbps is certainly high).  I
> think it would not be terribly difficult to show that such behavior
> could easily cause problems (to competing TCP, to competing streaming
> flows and to the flow that is doing this "instant on").  Slow start is
> the current consensus for how to start from zero.  Could there be other
> schemes?  Sure.  Are those here and fully fleshed out such that we could
> even consider them as potential solutions?  Not right now.
>
> My hunch is that sending DCCP (/anything) through without slow start is
> not going to work (i.e., I think consensus would be against that).  So,
> if your app can't deal with that then I think we might be stuck ...  A
> more general wondering of mine is how many streaming apps require this
> "instant on" behavior (i.e., is your app an anomaly or is there a whole
> class of apps that want instant-on over best-effort networks?)?
>
> Thanks!
>
> allman
>
>
> --
> Mark Allman -- ICIR -- http://www.icir.org/mallman/
>
> _______________________________________________
> 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 Dec  9 03:46:26 2003
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Subject: Re: [dccp] RE: [AVT] Would DCCP kill Internet real-time media dis tribution? 
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Brian-

> And, as I've pointed out, since these kinds of apps represent much
> higher revenue per bit that the TCP-type apps, it's much more likely
> that the network operators would deploy TCP stompers than media
> stompers.  YMMV.

I saw your assertion in response to Sally a while back in this thread.

But, I think we have a disconnect.  I am generally thinking in terms of
best-effort networks (as I think Sally was).  Further, a network that
has no priorities or traffic classes, but rather just tries its hardest
to get all the data across.  This is the kind of network where
congestion control is important.  And, this is the kind of network where
I think you'll see "stompers" that attempt to surpress flows that are
attempting to use a disproportionate amount of the bandwdith (whether
that be streaming or mis-behaving TCP or ddos floods or whatever).

Your claim above does not match my model of a best effort network.
I.e., if people are willing to pay more for the traffic of some
particular app and you provision appropriately then you no longer have
just a dumb best-effort network.  If you can engineer a network and
provision appropriately then by all means use UDP and go from zero to
4Mbps instantly.  However, I don't see that as supporting the assertion
that DCCP can't be used for streaming across best-effort networks that
are not engineered to give different levels of support to different
classes of traffic.

allman


--
Mark Allman -- ICIR -- http://www.icir.org/mallman/

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



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Hi Brian,



Rosen, Brian wrote:

> Magnus
> 
> You saw the message from Henning on the effect of slow start on
> conferencing applications didn't you?  What makes you think
> the video is any different?

I wasn't thinking of conferencing. I was more thinking of single 
conversational cases. I agree that the leg from a new sender to the 
conference focus will be problematic, and every switch would need to do 
the slow start which would effect the media badly.

The reason I think video is different from other media like audio is: at 
least in a conference application, the amount of bandwidth. You can get 
high quality voice using for example AMR-WB with a source bit rate of 
23.35 kbit/s + IP/UDP/RTP overhead giving a total around 40 kbit/s. 
While 40 kbit/s video gives either a very small picture or very bad 
quality. To get a comparable quality in the video with a least a CIF 
size screen you must go much higher.
> 
> In a conference application, streams come and go as different speakers
> come and go.  Even in a centrally mixed conference, which is not
> as demanding as an end-system mixed conference, most participants
> are not talking, and not sending video.  When they start talking,
> the start sending media, voice and video.  To send media all the
> time unnecessarily uses bandwidth; a lot of it.
> 
I agree that sending all the time is stupid.

> We don't do this now, we are end system mixed.
> 
> Even your basic assumption that coarse video is acceptable for
> a while is not true; it looks very poor.
> 

Yes, my point was not that it would not look bad initially, my point is 
that is that it is possible to make the video codec go from a very bad 
initial quality up to as good as you have bandwidth. In an application 
that keeps on sending the bad quality is only in the initial start-up.

> Actually, I'm not aware of any conferencing video systems that ramp
> bandwidth.  The reason is that the codecs are parameterized for a given
> quality level/bandwidth, and you can't smoothly change the parameters -
> you set them and restart the codec.  I suspect that you could design
> a codec to smoothly ramp, but that's not what we have available to us
> now.  What you do see sometimes is a codec which has a fixed bandwidth
> but the algorithm itself sends coarse image more often than fine image,
> and you see that effect on startup.  That is very undesirable, but
> it exists.
> 

I am not surprised that there is no system that does this. As I said in 
my previous mail it is time for media application developers to leave 
their heaven of no congestion awareness and start developing things.

Regarding the video codecs and their parameter sets. I know that if you 
need to switch parameter sets it can create problems in some codecs. 
However I know of least one codec, H.264, that is design and supports 
the use of multiple parameter sets. Also within a parameter sets the 
video has rather large differences in possible bit-rates. Even if no 
person has been using a video codec in the fashion of frequently turning 
on the bit-rate limit to the control, it is possible.

Still the main point I would like to make is: For best effort services, 
we and the users will probably need to accept that quality can vary or 
be changed during a session, due to congestion. If one can't accept 
this, then one must start using resource reservation.

Then media applications can encourage the network and congestion people 
to continue develop solutions that can get rid of slow start to be able 
to improve quality in the future.

I also liked the proposal from Eddie about the speeded up slow start 
that will lower the media impact.

Is someone keeping track of the different applications that will have 
problem and the different proposed solutions.


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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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Rosen, Brian'" <Brian.Rosen@marconi.com>,
        "'mallman@icir.org'"
	 <mallman@icir.org>
Cc: "'Eddie Kohler'" <kohler@icir.org>, avt@ietf.org, dccp@ietf.org
Subject: RE: [dccp] RE: [AVT] Would DCCP kill Internet real-time media dis
	 tribution? 
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Brian,

While some service providers may offer various QoS schemes in order to
squeeze extra revenue out of every bit they push, surely for the foreseeable
future there will continue to be providers who do not - who merely offer
best-effort Internet as a commodity. By failing to standardize a responsible
real-time protocol, we in fact encourage the sustained deployment of the
sorts of overlay networks you describe below. While the IETF does not always
see eye-to-eye with the marketplace, we do have a responsibility to offer
the marketplace credible alternatives to replacing the Internet with a
labyrinth of admissions control systems.

While I understand that the '4Megs or bust' philosophy springs from the
admirable goal of providing a consistent user experience, if I owned one of
your videophones (you know, various annual gift-giving holidays are just
around the corner...) I'd still like to be able to use it over my more
modest home office connection. I'd like a device that isn't hardwired to
just shut down if it can't offer me its ideal codec. Congestion control can
offer feedback to the device to help it know the transport-layer rate that
it can practically use, which could in turn influence codec selection, and
so on. The more RSVP-like approach of reserving the optimal amount of
bandwidth also has its place, and the IETF has long since acknowledged the
right of both of these approaches to co-exist, if uneasily. The IETF does
not hold a gun to any vendor's head, forcing them to do either or both.

There are forms of real-time communications that today's best-effort
Internet is unlikely to support in their full glory. But is this seriously
an argument that there is no point in standardizing a congestion control
scheme for unreliable datagrams for use over a best-effort network? That a
protocol instantiating such would have no applicability to real-time
communications?

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> Sent: Monday, December 08, 2003 2:57 PM
> To: 'mallman@icir.org'
> Cc: 'Eddie Kohler'; avt@ietf.org; dccp@ietf.org
> Subject: RE: [dccp] RE: [AVT] Would DCCP kill Internet real-time media
> dis tribution? 
> 
> 
> Mark
> 
> The reality of networks these days is that plain jane best effort 
> Internet doesn't make much money.  Premium services do.  So the 
> trend, which is clearly not universal, but is pretty obvious, 
> is that SPs are trying to bring new bandwidth users onto their 
> networks where the extra bandwidth is not just more bits, but 
> for which they can charge for service beyond best effort packet.
> 
> Voice is one of those services, and video is one of those 
> services.  You don't see a whole lot of investment these days 
> in plain Internet service, and you do in voice/video over IP 
> networks.  So, while there are pure play best effort operators 
> out there, and they will be there, the money is in the premium 
> services.  Even if an ISP does not directly provide premium 
> services, other SPs will, using the underlying packet carrier
> as the pipe, and the USERS of the premium services will yell 
> loudly if their premium services are not delivering what they 
> pay for.  If a customer of voice/video service gets told that 
> his pipe provider is intentionally degrading his service, what 
> do you think he will do with his pipe business?
> 
> Everyone agrees, I think, the most desirable situation is 
> one in which we have a true converged network, not overlay 
> networks, not separate networks, for these new services.  
> I'm sure that there will be a lot of overlay networks around 
> for quite a while, because we don't have the mechanisms we are 
> discussing here - ways for classic best effort TCP to co-exist 
> well with real time media.  Everyone wins if we figure out good 
> ways for this to happen.  Everyone loses if any SP starts 
> intentionally degrading one kind of service to make another
> kind of service work better.  That's why I don't think it's 
> going to be common to stomp on media OR TCP.  Rather, I think 
> everyone expects us to do our jobs well and make both kinds of 
> streams work well together.
> 
> We aren't going to solve this problem with anything done on one side
> folks; both sides are going to have to have new mechanisms.
> 
> Brian
> 
> > -----Original Message-----
> > From: Mark Allman [mailto:mallman@icir.org]
> > Sent: Monday, December 08, 2003 5:02 PM
> > To: Rosen, Brian
> > Cc: 'Eddie Kohler'; avt@ietf.org; dccp@ietf.org
> > Subject: Re: [dccp] RE: [AVT] Would DCCP kill Internet 
> real-time media
> > dis tribution? 
> > 
> > 
> > 
> > Brian-
> > 
> > > And, as I've pointed out, since these kinds of apps represent much
> > > higher revenue per bit that the TCP-type apps, it's much 
> more likely
> > > that the network operators would deploy TCP stompers than media
> > > stompers.  YMMV.
> > 
> > I saw your assertion in response to Sally a while back in 
> this thread.
> > 
> > But, I think we have a disconnect.  I am generally thinking 
> > in terms of
> > best-effort networks (as I think Sally was).  Further, a 
> network that
> > has no priorities or traffic classes, but rather just tries 
> > its hardest
> > to get all the data across.  This is the kind of network where
> > congestion control is important.  And, this is the kind of 
> > network where
> > I think you'll see "stompers" that attempt to surpress 
> flows that are
> > attempting to use a disproportionate amount of the 
> bandwdith (whether
> > that be streaming or mis-behaving TCP or ddos floods or whatever).
> > 
> > Your claim above does not match my model of a best effort network.
> > I.e., if people are willing to pay more for the traffic of some
> > particular app and you provision appropriately then you no 
> longer have
> > just a dumb best-effort network.  If you can engineer a network and
> > provision appropriately then by all means use UDP and go 
> from zero to
> > 4Mbps instantly.  However, I don't see that as supporting the 
> > assertion
> > that DCCP can't be used for streaming across best-effort 
> networks that
> > are not engineered to give different levels of support to different
> > classes of traffic.
> > 
> > allman
> > 
> > 
> > --
> > Mark Allman -- ICIR -- http://www.icir.org/mallman/
> > 
> > _______________________________________________
> > 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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Subject: RE: [dccp] RE: [AVT] Would DCCP kill Internet real-time media distribution?
Date: Tue, 9 Dec 2003 16:08:40 +0200
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From: "Yaakov Stein" <yaakov_s@rad.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>,
        "Rosen, Brian" <Brian.Rosen@marconi.com>
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There is another application (other than video, that is)
that is potentially high-bandwidth and has problems with
slow-start.

I am speaking of TDM pseudowires
(as being standardized in the PWE3 WG).

One can carry an E1 or T1 (or several of them)
over the public Internet, although higher rate TDM services
probably require dedicated networks.

Not only is TDM an inherent constant rate service,
it would be difficult to start it at anything other
than its full rate, since the other side requires the full
bandwidth for clock recovery, without which nothing works.

(Although I could imagine a special mechanism that sends the full
number of packets per second, but partially suppresses packet content
for the first few seconds.)

However, I am not sure whether TDM is an important case here
as the setup or pull-down of a TDM service is not a frequent event.

Y(J)S



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



From exim@www1.ietf.org  Tue Dec  9 09:16:17 2003
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From: "Phelan, Tom" <tphelan@sonusnet.com>
To: "'Yaakov Stein'" <yaakov_s@rad.com>
Cc: avt@ietf.org, dccp@ietf.org
Subject: RE: [dccp] RE: [AVT] Would DCCP kill Internet real-time media dis
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Hi Yaakov,

It seems to me that the TDM connection could deal with connection setup
delays caused by a ramp-up period at the beginning of the connection.  The
connection would really begin once the sufficient rate was reached.  Since
after that the transmission would be at constant rate, there wouldn't be any
more start up problems.

Tom Phelan

> -----Original Message-----
> From: Yaakov Stein [mailto:yaakov_s@rad.com]
> Sent: Tuesday, December 09, 2003 9:09 AM
> To: Magnus Westerlund; Rosen, Brian
> Cc: avt@ietf.org; dccp@ietf.org; Eddie Kohler
> Subject: RE: [dccp] RE: [AVT] Would DCCP kill Internet real-time media
> distribution?
> 
> 
> 
> There is another application (other than video, that is)
> that is potentially high-bandwidth and has problems with
> slow-start.
> 
> I am speaking of TDM pseudowires
> (as being standardized in the PWE3 WG).
> 
> One can carry an E1 or T1 (or several of them)
> over the public Internet, although higher rate TDM services
> probably require dedicated networks.
> 
> Not only is TDM an inherent constant rate service,
> it would be difficult to start it at anything other
> than its full rate, since the other side requires the full
> bandwidth for clock recovery, without which nothing works.
> 
> (Although I could imagine a special mechanism that sends the full
> number of packets per second, but partially suppresses packet content
> for the first few seconds.)
> 
> However, I am not sure whether TDM is an important case here
> as the setup or pull-down of a TDM service is not a frequent event.
> 
> Y(J)S
> 
> 
> 
> _______________________________________________
> 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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        "'mallman@icir.org'"
	 <mallman@icir.org>, avt@ietf.org, dccp@ietf.org
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Jon

While I certainly think what I'm doing is something everyone would
really, truly, want to have, I'm certainly aware that won't happen
for some time.  My app is simply a good example of what the long
term future holds.  The near term is more likely the simple voice 
conferencing application that Henning pointed out, and the channel 
surfing app for video (which is usually the better part of a megabit,
or more).

I think that the discussion of who carriers will penalize is not
very fruitful, but I'll keep swinging on those pitches as long as
people conjecture that SPs are motivated more by TCP fairness
than premium traffic revenue.  As I said, I really don't see any
carrier penalizing TCP or UDP unless it gets excessive, in the manner
Henning suggests.  I have long advocated charging for what you
deliver rather than prohibiting/firewalling/protecting-the-network.

I don't foresee a labyrinth of admission control; I see some very
simple mechanisms that will have pretty wide deployment.  What we
are missing for them is the feedback of loss.  I certainly think
that codec selection for current conditions is appropriate, downshifting
(once) depending on conditions is appropriate, keep alives, or
any new kinds of mechanisms that don't destroy the user experience
are fine with me.

I'm happy to work on, and help advocate, newer transports for media
that are better at coexisting with TCP.  However, I think I understand
what is an acceptable user experience and what isn't; I've spent the
better part of the last 4 years studying the effect of these kinds of
decisions on user experience, some of which is directly applicable
to less-than-4Mbit media streams.  I claim that as long as UDP
works, and DCCP doesn't, we aren't going to move apps to DCCP.
Interactive media streams start and stop abruptly, and the codecs
we have are close to constant rate.  You know that.  UDP can deliver
that, the current proposals for DCCP can't.

Brian

> -----Original Message-----
> From: Peterson, Jon [mailto:jon.peterson@neustar.biz]
> Sent: Monday, December 08, 2003 7:57 PM
> To: 'Rosen, Brian'; 'mallman@icir.org'
> Cc: 'Eddie Kohler'; avt@ietf.org; dccp@ietf.org
> Subject: RE: [dccp] RE: [AVT] Would DCCP kill Internet real-time media
> dis tribution? 
> 
> 
> 
> Brian,
> 
> While some service providers may offer various QoS schemes in order to
> squeeze extra revenue out of every bit they push, surely for 
> the foreseeable
> future there will continue to be providers who do not - who 
> merely offer
> best-effort Internet as a commodity. By failing to 
> standardize a responsible
> real-time protocol, we in fact encourage the sustained 
> deployment of the
> sorts of overlay networks you describe below. While the IETF 
> does not always
> see eye-to-eye with the marketplace, we do have a 
> responsibility to offer
> the marketplace credible alternatives to replacing the Internet with a
> labyrinth of admissions control systems.
> 
> While I understand that the '4Megs or bust' philosophy 
> springs from the
> admirable goal of providing a consistent user experience, if 
> I owned one of
> your videophones (you know, various annual gift-giving 
> holidays are just
> around the corner...) I'd still like to be able to use it over my more
> modest home office connection. I'd like a device that isn't 
> hardwired to
> just shut down if it can't offer me its ideal codec. 
> Congestion control can
> offer feedback to the device to help it know the 
> transport-layer rate that
> it can practically use, which could in turn influence codec 
> selection, and
> so on. The more RSVP-like approach of reserving the optimal amount of
> bandwidth also has its place, and the IETF has long since 
> acknowledged the
> right of both of these approaches to co-exist, if uneasily. 
> The IETF does
> not hold a gun to any vendor's head, forcing them to do 
> either or both.
> 
> There are forms of real-time communications that today's best-effort
> Internet is unlikely to support in their full glory. But is 
> this seriously
> an argument that there is no point in standardizing a 
> congestion control
> scheme for unreliable datagrams for use over a best-effort 
> network? That a
> protocol instantiating such would have no applicability to real-time
> communications?
> 
> Jon Peterson
> NeuStar, Inc.
> 
> > -----Original Message-----
> > From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> > Sent: Monday, December 08, 2003 2:57 PM
> > To: 'mallman@icir.org'
> > Cc: 'Eddie Kohler'; avt@ietf.org; dccp@ietf.org
> > Subject: RE: [dccp] RE: [AVT] Would DCCP kill Internet 
> real-time media
> > dis tribution? 
> > 
> > 
> > Mark
> > 
> > The reality of networks these days is that plain jane best effort 
> > Internet doesn't make much money.  Premium services do.  So the 
> > trend, which is clearly not universal, but is pretty obvious, 
> > is that SPs are trying to bring new bandwidth users onto their 
> > networks where the extra bandwidth is not just more bits, but 
> > for which they can charge for service beyond best effort packet.
> > 
> > Voice is one of those services, and video is one of those 
> > services.  You don't see a whole lot of investment these days 
> > in plain Internet service, and you do in voice/video over IP 
> > networks.  So, while there are pure play best effort operators 
> > out there, and they will be there, the money is in the premium 
> > services.  Even if an ISP does not directly provide premium 
> > services, other SPs will, using the underlying packet carrier
> > as the pipe, and the USERS of the premium services will yell 
> > loudly if their premium services are not delivering what they 
> > pay for.  If a customer of voice/video service gets told that 
> > his pipe provider is intentionally degrading his service, what 
> > do you think he will do with his pipe business?
> > 
> > Everyone agrees, I think, the most desirable situation is 
> > one in which we have a true converged network, not overlay 
> > networks, not separate networks, for these new services.  
> > I'm sure that there will be a lot of overlay networks around 
> > for quite a while, because we don't have the mechanisms we are 
> > discussing here - ways for classic best effort TCP to co-exist 
> > well with real time media.  Everyone wins if we figure out good 
> > ways for this to happen.  Everyone loses if any SP starts 
> > intentionally degrading one kind of service to make another
> > kind of service work better.  That's why I don't think it's 
> > going to be common to stomp on media OR TCP.  Rather, I think 
> > everyone expects us to do our jobs well and make both kinds of 
> > streams work well together.
> > 
> > We aren't going to solve this problem with anything done on one side
> > folks; both sides are going to have to have new mechanisms.
> > 
> > Brian
> > 
> > > -----Original Message-----
> > > From: Mark Allman [mailto:mallman@icir.org]
> > > Sent: Monday, December 08, 2003 5:02 PM
> > > To: Rosen, Brian
> > > Cc: 'Eddie Kohler'; avt@ietf.org; dccp@ietf.org
> > > Subject: Re: [dccp] RE: [AVT] Would DCCP kill Internet 
> > real-time media
> > > dis tribution? 
> > > 
> > > 
> > > 
> > > Brian-
> > > 
> > > > And, as I've pointed out, since these kinds of apps 
> represent much
> > > > higher revenue per bit that the TCP-type apps, it's much 
> > more likely
> > > > that the network operators would deploy TCP stompers than media
> > > > stompers.  YMMV.
> > > 
> > > I saw your assertion in response to Sally a while back in 
> > this thread.
> > > 
> > > But, I think we have a disconnect.  I am generally thinking 
> > > in terms of
> > > best-effort networks (as I think Sally was).  Further, a 
> > network that
> > > has no priorities or traffic classes, but rather just tries 
> > > its hardest
> > > to get all the data across.  This is the kind of network where
> > > congestion control is important.  And, this is the kind of 
> > > network where
> > > I think you'll see "stompers" that attempt to surpress 
> > flows that are
> > > attempting to use a disproportionate amount of the 
> > bandwdith (whether
> > > that be streaming or mis-behaving TCP or ddos floods or whatever).
> > > 
> > > Your claim above does not match my model of a best effort network.
> > > I.e., if people are willing to pay more for the traffic of some
> > > particular app and you provision appropriately then you no 
> > longer have
> > > just a dumb best-effort network.  If you can engineer a 
> network and
> > > provision appropriately then by all means use UDP and go 
> > from zero to
> > > 4Mbps instantly.  However, I don't see that as supporting the 
> > > assertion
> > > that DCCP can't be used for streaming across best-effort 
> > networks that
> > > are not engineered to give different levels of support to 
> different
> > > classes of traffic.
> > > 
> > > allman
> > > 
> > > 
> > > --
> > > Mark Allman -- ICIR -- http://www.icir.org/mallman/
> > > 
> > > _______________________________________________
> > > 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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Subject: RE: [AVT] Would DCCP kill Internet real-time media distribution?
Date: Tue, 9 Dec 2003 16:50:23 +0200
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From: "Yaakov Stein" <yaakov_s@rad.com>
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> Hi Yaakov,
>=20
> It seems to me that the TDM connection could deal with=20
> connection setup delays caused by a ramp-up period at the beginning of =
the=20
> connection.  The connection would really begin once the sufficient =
rate was=20
> reached.  Since after that the transmission would be at constant rate, =
there=20
> wouldn't be any more start up problems.
>=20
> Tom Phelan

There are several problem with that, as I said in my earlier email.

First, what exactly does one send before the TDM is "up to full rate"?
Arbitrarily chopped packets?=20
For the structure-aware case one could enable timeslots one at a time,
but the PWE3 WG is mandating the structure-agnostic mode as the
standards-track one, and here it is not clear how one could meaningfully =

reduce the rate.

Second, the far end uses the initial period of TDM to lock onto
the TDM source clock's frequency and phase.
This task has to be relatively fast, since before then the
TDM output does not comply with the international standards
for jitter and wander, and can cause problems in the far-end TDM =
network.

The only thing I can see that would work is to send dummy packets
(marked as such) at a rate determined by the local TDM clock,=20
and increasing their size until full bandwidth is attained.
Thereafter the dummy marking is removed and the true data sent.

By using such a mechanism the TDM clock could be acquired
without meaningful TDM data being lost or corrupting the far-end =
network.

Y(J)S

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



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Mangus

> I wasn't thinking of conferencing. I was more thinking of single 
> conversational cases. I agree that the leg from a new sender to the 
> conference focus will be problematic, and every switch would 
> need to do 
> the slow start which would effect the media badly.
I'm pretty sure that a point-to-point only solution is not a solution
even if the incidence of point to point is 100 x more common.

> 
> The reason I think video is different from other media like 
> audio is: at 
> least in a conference application, the amount of bandwidth. 
> You can get 
> high quality voice using for example AMR-WB with a source bit rate of 
> 23.35 kbit/s + IP/UDP/RTP overhead giving a total around 40 kbit/s. 
> While 40 kbit/s video gives either a very small picture or very bad 
> quality. To get a comparable quality in the video with a least a CIF 
> size screen you must go much higher.
Why do you think that the bandwidth matters in this discussion?
While bigger bandwidth causes congestion sooner, you can get congestion
with a 40kbit stream on a 256K uplink pretty easily.  The solution
is not very bandwidth sensitive I think.  Because of the effect
of packetization periods on overall delay, we see that as codec
bandwidth goes up, packet size goes up.  My 4M streams are extreme,
because you reach MTU before you get a big packetization period,
but that is not true with slower video streams.


> Yes, my point was not that it would not look bad initially, 
> my point is 
> that is that it is possible to make the video codec go from a 
> very bad 
> initial quality up to as good as you have bandwidth. In an 
> application 
> that keeps on sending the bad quality is only in the initial start-up.
But that presents a poor user experience; an experience they do not
get on existing transports.  If you degrade the user experience when
using a data network, you won't get many users.

> 
> > Actually, I'm not aware of any conferencing video systems that ramp
> > bandwidth.  The reason is that the codecs are parameterized 
> for a given
> > quality level/bandwidth, and you can't smoothly change the 
> parameters -
> > you set them and restart the codec.  I suspect that you could design
> > a codec to smoothly ramp, but that's not what we have 
> available to us
> > now.  What you do see sometimes is a codec which has a 
> fixed bandwidth
> > but the algorithm itself sends coarse image more often than 
> fine image,
> > and you see that effect on startup.  That is very undesirable, but
> > it exists.
> > 
> 
> I am not surprised that there is no system that does this. As 
> I said in 
> my previous mail it is time for media application developers to leave 
> their heaven of no congestion awareness and start developing things.
You might want to go talk to the chip vendors, as we mostly see
hardware encoders for video applications except for toys.  It's a
chip level problem, not software.


> 
> Regarding the video codecs and their parameter sets. I know 
> that if you 
> need to switch parameter sets it can create problems in some codecs. 
> However I know of least one codec, H.264, that is design and supports 
> the use of multiple parameter sets. Also within a parameter sets the 
> video has rather large differences in possible bit-rates. Even if no 
> person has been using a video codec in the fashion of 
> frequently turning 
> on the bit-rate limit to the control, it is possible.
I don't yet have enough experience with high quality H.264 to confirm
or deny that this is possible.  I haven't seen that, but implementations
are still very new.

> 
> Still the main point I would like to make is: For best effort 
> services, 
> we and the users will probably need to accept that quality 
> can vary or 
> be changed during a session, due to congestion. If one can't accept 
> this, then one must start using resource reservation.
Actually, I don't agree.  I think a combination of admission control
and some new mechanisms on both TCP and UDP-or-replacement can
achieve quite acceptable results without resource reservation mechanisms.

Brian


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



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From: "David R. Oran" <oran@cisco.com>
To: dccp@ietf.org
cc: avt@ietf.org
Subject: Re: [AVT] Faster Start after idle periods
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Define "idle period".

If I'm on a conference call, announce the speaker, then listen for 2 hours=20
of rambling, and then speak up to ask "any questions"? is that restart=20
after idle?

It seems to me there are some very tricky time constants here.

how does this differ from HTTP tearing down a TCP connection to a web site=20
and then putting up another one to the same site on the next click and=20
claiming that it doesn't need to do slow start because the browser is "just =

coming back from an idle period".

If DCCP is allowed to keep congestion state across idle periods, then=20
shouldn't TCP too?

We keep cycling around the core issue which is the claims by some in the=20
community that realtime interactive media does not work acceptably well if=20
it has to be congestion-responsive in its transmission rate, and the=20
contention by others that some clever adaptive rate control can work well=20
enough.

I think we need existence proofs of a system that does rate-adapt=20
successfully before recommending the use of DCCP in preference to UDP.

What would be even MORE helpful would be practical policing/dropping=20
algorithms on bottleneck links that dropped ALL packets of a subset of the=20
flows rather than RED or WRED across all the flows. That way a=20
measurement-based admission control scheme might have a chance of keeping a =

subset of the realtime flows going with acceptable loss/delay during=20
congestion.

Dave.

--On Monday, December 8, 2003 6:10 PM -0800 Eddie Kohler <kohler@icir.org>=20
wrote:

> [Replies to dccp@ietf.org only please.]
>
> Sally and I talked today about the slow-starting after an idle period.  =
We
> have a potential solution that doesn't involve sending filler traffic to
> keep the pipe open.
>
> Say your app had a sending rate of K packets/RTT. Then you have an idle
> period, possibly up to several minutes; and you want to start sending
> again.
> (1) Start sending with a window of 8 packets/RTT (not 4).
> (2) Increase the window size by 4x after each lossless RTT (not 2x).
>
> This is twice as fast as slow start.  But the multiplicative factor means
> that if slow start reached a rate in N RTTs, faster start would reach the
> same rate in sqrt(N) RTTs.  A window of 128 packets/RTT is reached after =
2
> lossless RTTs (for a large RTT of 100ms and maximum-size packets, this is
>> 14Mb/s after .2s).
>
> The theoretical justification for this is that the connection recently
> proved it could sustain rate K on this path.
>
> Caveats:
> (1) You must switch out of faster start (to slow start) once you reach =
the
>     original rate K.
> (2) You must switch out of faster start (to AI) after any congestion
> event. (3) You can't use faster start if your last faster-start
> encountered     congestion, and didn't achieve the original rate K.
> (Slow start     instead.)
> (4) You probably can't use faster start if you didn't actually sustain
> rate     K for long before.
> (5) Possibly others.
>
> It's not immediate-start.  It doesn't change the slow start that the
> connection must go through initially.  It does greatly improve the
> dynamics of DCCP connections that go idle (push-to-talk, interactive
> media), without requiring filler traffic.
>
> Details remain to be ironed out.  Help or simulations welcome.  Comments
> welcome too.
>
> Eddie
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt




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



From exim@www1.ietf.org  Tue Dec  9 22:12:22 2003
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From: Qiaobing Xie <Qiaobing.Xie@motorola.com>
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......

> I think we need existence proofs of a system that does rate-adapt
> successfully before recommending the use of DCCP in preference to UDP.

Not only that, we need to also show that RTP/DCCP can outperform RTP/UDP in
whatever terms before the vendors and SPs can see any incentives to spend money
for the switch. This is not going to be easy considering what's been happening
for IPv6. There needs to be an IETF "official" list of incentives for adopting
DCCP composed eventually. Just telling the vendors and SPs that UDP is
unfriendly to TCP traffic won't cut it, IMO.

AVT probably should wait to see such an "incentive list" composed before diving
into migrating payload RFCs to DCCP.

regards,
-Qiaobing


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



From exim@www1.ietf.org  Wed Dec 10 03:54:42 2003
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Subject: Re: [dccp] Re: [AVT] Faster Start after idle periods
References: <200312090210.hB92ARsD083394@coyote.icir.org> <2147483647.1070968618@[10.32.245.157]> <3FD68EFD.30B770E5@motorola.com>
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Hi Qiaobing,

I reacted to this:

Qiaobing Xie wrote:
> AVT probably should wait to see such an "incentive list" composed before diving
> into migrating payload RFCs to DCCP.

There should be no reason to do any changes to the RTP payload formats 
due to that one transport RTP over DCCP instead of UDP. So far the 
discussion has not shown any need to do any changes to RTP at all.

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



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From: Colin Perkins <csp@csperkins.org>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Cc: Qiaobing.Xie@motorola.com, avt@ietf.org
Subject: Re: [dccp] Re: [AVT] Faster Start after idle periods
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[since this is now about payload formats, I'll limit the cc to avt - please
 try to avoid cc'ing both lists]

--> Magnus Westerlund <magnus.westerlund@ericsson.com> writes:
> Qiaobing Xie wrote:
> > AVT probably should wait to see such an "incentive list" composed
> > before diving into migrating payload RFCs to DCCP.
> 
> There should be no reason to do any changes to the RTP payload formats 
> due to that one transport RTP over DCCP instead of UDP. So far the 
> discussion has not shown any need to do any changes to RTP at all.

I can certainly imagine that some payload formats could use the improved
loss feedback that DCCP provides. We yet haven't identified any need for
changes to RTP payload formats, but I wouldn't necessarily rule them out.

Colin

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



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Hello AVT:

Has anyone submitted an I-D in the past which defines a method of 
providing trickplay over HTTP (much like what RTSP can do)? For 
example, adding HTTP headers for playback position, speed, direction, 
etc.

-John Gildred
Vice President of Engineering
Pioneer Research Center USA, Inc.
A Division of Pioneer Electronics
101 Metro Drive, Suite 264
San Jose, California 95110
Office (408) 437-1800 Ext. 105
Mobile (510) 295-7770
john@pioneer-pra.com


_______________________________________________
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From exim@www1.ietf.org  Thu Dec 11 12:43:39 2003
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> John Gildred writes:
>
> Has anyone submitted an I-D in the past which defines a method of 
> providing trickplay over HTTP (much like what RTSP can do)?

From the MMUSIC minutes of IETF 51 (London):

========================================================================
10. Tunneling RTSP/RTP/RTCP in HTTP (draft-gentric-avt-rtsp-http-00.txt) 
========================================================================

RTSP describes a TCP-based streaming mode. This includes interleaved
RTP/RTCP packets. The draft proposal is to extend this to HTTP-only
firewalls.

Source of proposal: Internet Streaming Media Alliance.

The basic proposal is to use HTTP to create a channel in each
direction. The authors are looking for more ideas and thoughts on how
to proceed.

Melinda Shore (mshore@cisco.com) expressed her disagreement with the
proposal, because it raises serious security concerns.  Its whole
thrust is to violating site security policy.  It would be more
consistent with network policy to use Jonathan's SIP approach. If the
draft is accepted, HTTP might become characterized as "Horrible
Trashcan Transport Protocol". Jonathan Rosenberg noted that it is too
late to salvage HTTP's reputation: RFC 3093 "IP Over HTTP" already
describes how to do it. Mark Handley also expressed his concern with
the security implications. Port 554 is allocated for RTSP; managers
can open it if they want to. Apparently some firewall vendors already
block strange uses of HTTP. Site administrators are aware of this sort
of tunneling and are asking firewall vendors for controls.

Joerg Ott ruled that the responses he had heard make it clear that the
proposal is going nowhere in the MMUSIC WG.

---

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



From exim@www1.ietf.org  Thu Dec 11 14:39:40 2003
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Hi everyone,

	Several EE Times stories today covering the Bluetooth Americas
conference, which talks about RTP over Bluetooth work underway:

http://www.eetimes.com/sys/news/OEG20031211S0019
http://www.commsdesign.com/story/OEG20031210S0010

	Given that wireless MIDI controllers are one of the target
areas of use for RTP MIDI, I thought I should check on the list -- if
the folks involved in the aforementioned work are listening in, is
there any work in sight on the IETF side to support Bluetooth uses for
RTP?  Or, like RTP over 801.11, is the hope that the RTP <-> UDP <->
Bluetooth layering abstraction will "just work", isolating AVT from
new Bluetooth-related work items?

---------------------------------------------------
John Lazzaro    http://www.cs.berkeley.edu/~lazzaro
     lazzaro [at] cs [dot] berkeley [dot] edu     
---------------------------------------------------


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



From exim@www1.ietf.org  Thu Dec 11 18:45:34 2003
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Date: Thu, 11 Dec 2003 23:44:16 +0000
From: Colin Perkins <csp@csperkins.org>
To: avt@ietf.org
Cc: magnus.westerlund@ericsson.com
Subject: Re: [AVT] WG last call: iLBC payload format and codec
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I draw the attention of the group to the revised IPR statement for this
draft available at:
http://www.ietf.org/ietf/IPR/global-ip-ipr-draft-ietf-avt-ilbc-codec.txt

Colin



--> Colin Perkins <csp@csperkins.org> writes:
> This is to announce a working group last call on the RTP Payload Format
> and codec for iLBC speech:
> 
> 	draft-ietf-avt-rtp-ilbc-04.txt
> 	draft-ietf-avt-ilbc-codec-04.txt
> 
> Please send any final comments on these drafts to the mailing list by
> 19th December 2003.  If no substantive comments are received by that
> time, we will ask the IESG to publish the codec as an Experimental RFC
> and the RTP payload format as a Proposed Standard RFC.
> 
> -- 
> Colin Perkins
> http://csperkins.org/

_______________________________________________
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From exim@www1.ietf.org  Fri Dec 12 13:32:22 2003
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Subject: [AVT] Profile specification
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Hi all,

In RFC3550 section 5.3.1, it is stated that "The format of these 16 bits is 
to be defined by the profile specification under which the implementations 
are operating."

May I know what is the meaning of profile in this case? Is this "profile 
specification" the same profile under which the PT type can be defined (see 
section 5.1 under payload type)? Does a profile specify the format of the 
Marker Bit, the Payload Type and use of Header Extension?

Best Regards,
Fedrick

_________________________________________________________________
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http://www.msn.co.uk/messenger


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From exim@www1.ietf.org  Fri Dec 12 14:40:32 2003
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Date: Fri, 12 Dec 2003 11:37:25 -0800 (PST)
From: Stephen Casner <casner@acm.org>
To: Mainka Fedrick <fedmain@hotmail.com>
cc: avt@ietf.org
Subject: Re: [AVT] Profile specification
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On Fri, 12 Dec 2003, Mainka Fedrick wrote:

> In RFC3550 section 5.3.1, it is stated that "The format of these 16 bits is
> to be defined by the profile specification under which the implementations
> are operating."
>
> May I know what is the meaning of profile in this case? Is this "profile
> specification" the same profile under which the PT type can be defined (see
> section 5.1 under payload type)?

The term "profile" is defined in Section 1, and I believe that term is
used consistently throughout the document.  That is, there should be
no other meanings.  Therefore, the short answer to your question is
yes.

> Does a profile specify the format of the Marker Bit, the Payload
> Type and use of Header Extension?

Yes.  Section 13 lists aspects that may be defined in a profile.  So
far, all of the profiles that have been written simply follow the
default format in the main RTP spec for the format of the Marker and
PT byte, and none have defined Header Extensions.  Only one profile
has been published as an RFC (that's 3551 which updates 1890), but two
more are about to be published.

                                                        -- Steve

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



From exim@www1.ietf.org  Mon Dec 15 10:58:41 2003
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Subject: [AVT] 2833bis default event support
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Suppose a receiver indicates support for the telephone-events payload 
type but does not indicate which events it supports.  Do we continue to 
require support for events 0 to 13 in all implementations?

Tom Taylor


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



From exim@www1.ietf.org  Tue Dec 16 08:03:43 2003
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From: "Rolf Hakenberg" <hakenberg@panasonic.de>
To: "Colin Perkins" <csp@csperkins.org>
Cc: <gamze@vidiator.com>, <rey@panasonic.de>, <avt@ietf.org>,
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Subject: RE: [AVT] Retransmission draft
Date: Tue, 16 Dec 2003 14:02:15 +0100
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Dear Colin and all,

> > At the last 3GPP SA4 meeting there was a consensus on to make RTP
> > Retransmission a working assumption in PSS Rel-6, subject to
> the concerns
> > on server complexity and delay being addressed at the next meeting.
>
> Do you expect addressing these concerns to require changes to the draft?

No, I do not expect this to happen.

The comments and concerns expressed at 3GPP have been quite specific to the
3GPP PSS (packet based streaming)environment and proposed use of RTP RTX in
this requirement. One question for example was the correlation of RTP RTX
with retransmission in the radio link control (RLC) layer considering the
RLC discard timer and possible additional delay introduced by RTP RTX.

Questions on complexity are very much implementation dependent and the draft
already provides good general information about this issue in chapter 10. I
do not see a need to elaborate any further on this issue in the draft.

FYI: If 3GPP finds any information important for an implementor of RTP RTX
in a 3GPP PSS environment, it has means to put this information as
"informative parts" into the 3GPP specifiction.


Best Regards,
Rolf.


>
> > Important for our current discussion on potential IPRs and how to
> > progress the draft is the fact that there was no condition on the IPR
> > issues, but in fact conditions that the draft should move to RFC on
> > standards track before the finalisation of 3GPP Rel-6.
>
> Moving to the standards track requires resolution of the IPR issue within
> AVT, of course.
>
> --
> 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 Dec 16 12:56:36 2003
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From: Flemming Andreasen <fandreas@cisco.com>
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Subject: Re: [AVT] 2833bis default event support
References: <3FDDD9CF.2080402@nortelnetworks.com>
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I don't see any reason to require (0-15 btw). Implementations can support
whatever events they like and they MUST declare support for all the events
they want to use (0-15 included). It may be worthwhile saying, that for
2833 backwards compatibility, if you receive an SDP with 2833 events and no
fmtp parameter, that you assume events 0-15 are supported.

-- Flemming

Tom Taylor wrote:

> Suppose a receiver indicates support for the telephone-events payload
> type but does not indicate which events it supports.  Do we continue to
> require support for events 0 to 13 in all implementations?
>
> 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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Sounds good to me.

Flemming Andreasen wrote:

> I don't see any reason to require (0-15 btw). Implementations can support
> whatever events they like and they MUST declare support for all the events
> they want to use (0-15 included). It may be worthwhile saying, that for
> 2833 backwards compatibility, if you receive an SDP with 2833 events and no
> fmtp parameter, that you assume events 0-15 are supported.
> 
> -- Flemming
> 
> Tom Taylor wrote:
> 
> 
>>Suppose a receiver indicates support for the telephone-events payload
>>type but does not indicate which events it supports.  Do we continue to
>>require support for events 0 to 13 in all implementations?
>>
>>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: [MMUSIC] time-lined static media with FLUTE & IMG
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Hi,

we are also thinking of an application combining live streaming of 
continuous media with time-lined static media objects. The idea is to 
transfer slides of a lecture (binary static object) synchronized with 
the spoken explaination (audio).

Looking into the FLUTE draft, it seems to already provide the necessary 
means for reliable transfer of time-lined static media: providing 
additional timing attributes to the FDT. With the latter and the RTP 
timing information, the application can synchronize both media objects.

Now since Magnus does not think there is a straightforward solution, I 
am wondering where I might be wrong?

Has there been any follow-up discussion?

> Hi Magnus,
> 
> It was good to learn of your views on Flute! We have been thinking on similar lines about the possible use SMIL for synchronizing Flute transfers with media streams. The use of SDP for describing the sessions of the FLUTE protocol has been in the plans. We have also been preliminarily considering the use of both SMIL and SDP for our work on Internet Media Guides in MMUSIC.
> 
> Looking forward to seeing you in Minneapolis!
> 
> -Pekka
> 
> -------------------
> Juha-Pekka Luoma
> Nokia Research Center
> Tampere, Finland
> -------------------
> 
> > -----Original Message-----
> > From: ext Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> > Sent: Thursday, October 30, 2003 11:42 AM
> > To: Takeshi Yoshimura
> > Cc: Walsh Rod (NRC/Tampere); finlayson@live.com; avt@ietf.org
> > Subject: Re: [MMUSIC] time-lined static media with FLUTE & IMG
> > 
> > 
> > Hi,
> > 
> > The static media draft is dead. I have no plans of updating it as I
> > currently don't think it is the right approach to the 
> > problem. The usage
> > of RTP actually seems rather unnecessary. Most of the functionality in
> > RTP is not at all needed and other things where missing. Here 
> > I find the
> > FLUTE solution more addressing the problems around binary object
> > transport for multicast. However I don't know its performance for
> > unicast, however for any unicast session HTTP is probably 
> > usable if one
> > can schedule the GET command appropriately and to know when to stop
> > requesting because it is to late.
> > 
> > The FLUTE solution are still missing many vital parts to my
> > understanding. The first is how to achieve the synchronization. One
> > could for example use SMIL however this requires one to have resolved
> > how one performs the referencing to downloaded binary objects. To my
> > understanding a defined cache utilizing FLUTEs source URI declaration
> > could be a possibility. There also seem to be shortcoming that no real
> > effort has been made into defining how the session 
> > declaration should be
> > made.
> > 
> > The usage of RFC 2733 or ULP does not seem to provide the 
> > necessary FEC
> > performance to be used in multicast delivery, at least not for objects
> > larger than a few packets. However this is not my expertise 
> > area, but I
> > assume there are good reasons why more advanced algorithms like
> > reed-solomon or tornado codes seem to be used in the area.
> > 
> > I don't think this problem space has any straightforward solutions. It
> > is hard problems, requiring well thought through solutions. There is
> > definitely need to do more work in the area.
> > 
> > My personal feeling in regards to MBMS is that the work is to 
> > hurried to
> > be able to produce a good solution.
> > 
> > As a conclusion to the original question, I would say that the
> > application of FLUTE for time-lined static media delivery as 
> > combination
> > to multicasted continuos media like audio and video requires further
> > investigation.
> > 
> > Cheers
> > 
> > Magnus
> > 
> > Takeshi Yoshimura wrote:
> > 
> > > Hi Rod and Magnus,
> > > 
> > > I'm also considering a reliable and time-lined static media 
> > delivery,
> > > because it is discussed in 3GPP as one of the services over 3G MBMS
> > > (Multimedia Broadcast/Multicast Service).
> > > 
> > > You seem to extend FLUTE for this purpose, but I think 
> > combining generic
> > > RTP format (Magnus's draft) and RTP-FEC format (e.g, RFC2733,
> > > draft-ietf-avt-ulp, draft-ietf-avt-uxp) is more straightforward way.
> > > What do you think of this approach?
> > > 
> > > Regarding this issue, what's the current status of
> > > draft-westerlund-avt-rtp-static-media-01?  Is there any technical
> > > problem with this?
> > > 
> > > Best regards,
> > > Takeshi
> > > 
> > > 
> > >>Hi Magnus, Ross (and everyone)
> > >>
> > >>1$B%9(J-2 years back there was a draft on time-lined static 
> > media (draft-westerlund-avt-rtp-static-media-01) - basically 
> > to send files, text (etc) with timed presentation along with 
> > AV media streams. Since the basic requirements haven't gone 
> > away, but RMT has made progress on usable protocols, what do 
> > you think about using FLUTE (draft-ietf-rmt-flute-02) for the 
> > same job? With a quick comparison I think it should be pretty 
> > straight forward, although you may have some preferences (or 
> > extensions to the FLUTE-FDT) if you already have a working 
> > RTP-payload'ed implementation.
> > >>
> > >>Ross mentioned sending ID-3 tags in-band with audio 
> > streams. Any ideas you have on doing this with FLUTE (ID3) 
> > and RTP (music) could be pretty useful for the Internet Media 
> > Guide (IMG) work in MMUSIC - how to move around (and 
> > announce) metadata for 'foo' multicast and unicast sessions.
> > >>
> > >>Cheers, Rod.
> > 
> > -- 
> > 
> > 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


-- 

Best regards,

+--- Ivica Rimac -----------------------------------------------+
|               KOM - Multimedia Communications                 |
| Department of Electrical Engineering & Information Technology |
|              Darmstadt University of Technology               |
|                    email: Ivica.Rimac@kom.tu-darmstadt.de     |
|     Merckstr. 25   tel.:  +49-6151-16-6112  (Fax -6152)       |
|  64283 Darmstadt   http://www.kom.e-technik.tu-darmstadt.de   |
+---------------------------------------------------------------+


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



From exim@www1.ietf.org  Wed Dec 17 05:00:51 2003
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Subject: RE: [AVT] RE: [MMUSIC] time-lined static media with FLUTE & IMG
Date: Wed, 17 Dec 2003 11:59:34 +0200
Message-ID: <D0299AFF29E01E478321564030AD69090BE3C2@trebe003.europe.nokia.com>
Thread-Topic: [AVT] RE: [MMUSIC] time-lined static media with FLUTE & IMG
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To: <ivica.rimac@KOM.tu-darmstadt.de>, <avt@ietf.org>
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Hi Ivica

Great to hear you're interested in this field.

> Has there been any follow-up discussion?

Not much so far. I think the related people are hammering away at other =
problems at the moment, so this one is queued (at least for me).

> Now since Magnus does not think there is a straightforward=20
> solution, I=20
> am wondering where I might be wrong?

I think Magnus was refering to FEC (for RTP). The references where not =
ALC FEC references. I (also) am not a FEC expert, but if you have an =
idea of what your application reliability and network loss =
characteristics might be, it should be easier to assess what kind of FEC =
scheme would suit. Fortunately ALC's designed to allow new and old =
schemes to 'plug-in' staightforwardly. You just need the FEC spec that =
meets your needs.

As I interpretted it, Magnus highlighted 3 issues needing solution:
1. How to achieve synchronisation
2. How to reference downloaded binary objects
3. Session description (declaration) of FLUTE sessions (e.g. SDP =
extension)

You have addressed (1) with "providing additional timing attributes to =
the FDT"...

> Looking into the FLUTE draft, it seems to already provide the=20
> necessary=20
> means for reliable transfer of time-lined static media: providing=20
> additional timing attributes to the FDT. With the latter and the RTP=20
> timing information, the application can synchronize both=20
> media objects.

Which makes sense. Essentially I can see three simple solutions for (1):
1. describe the timing information in the FDT as an extension (your =
suggestion)
2. describe the timing information in the FDT in a seperate object (e.g. =
in a SMIL file - Magnus's suggestion)
3. use the expires field already in the FDT as the implicit timing field =
(I not convinced whether this is a good idea)

If you've got a spec of any of these I'd be happy to distrubte and =
review it.

Irrespecive of how you solve (1), you need to solve (2) - how to =
reference "downloaded binary objects" (aka "files"). FLUTE already gives =
you URIs to identify files so the simple way is to use these. If you =
want to be globaly unique, I guess you'll choose URLs (rather than URNs) =
although there's the consideration of whether HTTP://... is going to be =
resolved to an HTTP GET by your application if it's not savvy with =
what's being multicast. Could just use relative URLs, or even define a =
new FLUTE URL. For me everything is fine as is (it's easy to reference a =
URI and take care that you understand it's uniqueness in whatever =
domain). If anyone sees some significant problems with this, please =
highlight them - especially whether we need a FLUTE URL (since that =
needs an Internet-draft).

Finally, to solve (3) you need to define a FLUTE Session Description =
Syntax/extension. It's a fair bet that SDP will be with us for some time =
ahead so an SDP extension seems to be the minimal Internet-draft for =
general Interent usage. This is on my todo list, unless someone else =
does it first.

BTW, one of the open source FLUTE implementations is available here:
http://atm.tut.fi/mad/

Cheers, Rod

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



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Date: Thu, 18 Dec 2003 17:57:59 +0000
From: Colin Perkins <csp@csperkins.org>
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Cc: magnus.westerlund@ericsson.com, Eddie Kohler <kohler@icir.org>
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Folks,

Attached are draft minutes for the AVT sessions at the 58th IETF in
Minneapolis. We're running late with these, so please send any comments
immediately, so we can meet the submission deadline.

Thanks,
Colin

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--Multipart=_Thu__18_Dec_2003_17_57_59_+0000_yT05VQVFJoR/qsPT--

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



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Date: Wed, 17 Dec 2003 11:26:48 +0100
From: Ivica Rimac <ivica.rimac@KOM.tu-darmstadt.de>
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Subject: Re: [AVT] RE: [MMUSIC] time-lined static media with FLUTE & IMG
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Hi Rod,

thank you for your answer. See my comments inline:

Rod.Walsh@nokia.com wrote:
> Hi Ivica
> 
> Great to hear you're interested in this field.
> 
> 
>>Has there been any follow-up discussion?
> 
> 
> Not much so far. I think the related people are hammering away at other problems at the moment, so this one is queued (at least for me).
> 
> 
>>Now since Magnus does not think there is a straightforward 
>>solution, I 
>>am wondering where I might be wrong?
> 
> 
> I think Magnus was refering to FEC (for RTP). The references where not ALC FEC references. I (also) am not a FEC expert, but if you have an idea of what your application reliability and network loss characteristics might be, it should be easier to assess what kind of FEC scheme would suit. Fortunately ALC's designed to allow new and old schemes to 'plug-in' staightforwardly. You just need the FEC spec that meets your needs.
> 
> As I interpretted it, Magnus highlighted 3 issues needing solution:
> 1. How to achieve synchronisation
> 2. How to reference downloaded binary objects
> 3. Session description (declaration) of FLUTE sessions (e.g. SDP extension)
> 
> You have addressed (1) with "providing additional timing attributes to the FDT"...
> 
> 
>>Looking into the FLUTE draft, it seems to already provide the 
>>necessary 
>>means for reliable transfer of time-lined static media: providing 
>>additional timing attributes to the FDT. With the latter and the RTP 
>>timing information, the application can synchronize both 
>>media objects.
> 
> 
> Which makes sense. Essentially I can see three simple solutions for (1):
> 1. describe the timing information in the FDT as an extension (your suggestion)
> 2. describe the timing information in the FDT in a seperate object (e.g. in a SMIL file - Magnus's suggestion)
> 3. use the expires field already in the FDT as the implicit timing field (I not convinced whether this is a good idea)
> 
> If you've got a spec of any of these I'd be happy to distrubte and review it.

Thought of SMIL-like attributes:

<xs:attribute 	name="Content-Begin"							 
type="xs:unsignedLong"								use="optional"/>
<xs:attribute 	name="Content-Duration"						 
type="xs:unsignedLong"								use="optional"/>

If "Content-Duration" is not provided, the object is displayed until the 
end of the session or until the beginning of a new FLUTE object with the 
same TOI.


> 
> Irrespecive of how you solve (1), you need to solve (2) - how to reference "downloaded binary objects" (aka "files"). FLUTE already gives you URIs to identify files so the simple way is to use these. If you want to be globaly unique, I guess you'll choose URLs (rather than URNs) although there's the consideration of whether HTTP://... is going to be resolved to an HTTP GET by your application if it's not savvy with what's being multicast. Could just use relative URLs, or even define a new FLUTE URL. For me everything is fine as is (it's easy to reference a URI and take care that you understand it's uniqueness in whatever domain). If anyone sees some significant problems with this, please highlight them - especially whether we need a FLUTE URL (since that needs an Internet-draft).

The idea was to use relative URLs, conforming to your thoughts, since 
all the objects are locally cached in the end-device. Thus, there is no 
need for changes. The rest is up to the application.

> 
> Finally, to solve (3) you need to define a FLUTE Session Description Syntax/extension. It's a fair bet that SDP will be with us for some time ahead so an SDP extension seems to be the minimal Internet-draft for general Interent usage. This is on my todo list, unless someone else does it first.

Agree, this seems to be an open issue.

> 
> BTW, one of the open source FLUTE implementations is available here:
> http://atm.tut.fi/mad/
> 
> Cheers, Rod

-- 

Best regards,

+--- Ivica Rimac -----------------------------------------------+
|               KOM - Multimedia Communications                 |
| Department of Electrical Engineering & Information Technology |
|              Darmstadt University of Technology               |
|                    email: Ivica.Rimac@kom.tu-darmstadt.de     |
|     Merckstr. 25   tel.:  +49-6151-16-6112  (Fax -6152)       |
|  64283 Darmstadt   http://www.kom.e-technik.tu-darmstadt.de   |
+---------------------------------------------------------------+


_______________________________________________
Audio/Video Transport Working Group
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--NextPart

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

	Title		: Registration of the text/red MIME Sub-Type
	Author(s)	: P. Jones
	Filename	: draft-ietf-avt-text-red-00.txt
	Pages		: 5
	Date		: 2003-12-18
	
This document defines the text/red MIME sub-type.  The actual RTP
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Would not it be useful to explain what "red" stands for ?


Philippe Gentric
Chief Architect
Philips Software
philippe.gentric@philips.com
http://www.software.philips.com



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Dear Steve W. and Other Authors:

I have some questions and comments on draft-ietf-avt-rtp-h264-03.txt
<http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-h264-03.txt>:

*	Section 5.4: Packetization Modes
        There is this statement:
		The non-interleaved mode is targeted for conversational 
		   systems that may not comply with ITU-T Recommendation
H.241.  

	I tried to download H.241 spec but since I don't have ITU
subscription I am not able to find it. It would be helpful to add a brief
explanation (if feasible) on why this " may not comply with ITU-T
Recommendation H.241."   

	Reading a later section, I believe that reason is that H.241 does
not allow grouping of NAL units in RTP packets. Is that the only reason?

*	Section 5.7.2: MTAPs
There is a statement, quoted below:

	      "This memo does not specify how the NAL units within an MTAP
are ordered, but, in most cases, NAL unit 
	   decoding order SHOULD be used."

	There is a case where, when partial checksum is available, from
either UDP lite or DCCP, that the best way to order NAL units within a MTAP
is by the reverse order of NRI, since NRI indicates the relative importance
of a NAL unit, and that NRI=0 means that the NAL unit is not used as
reference to any other NAL unit. Hence it makes sense to apply partial
checksum on the first group of NAL units, upto but not including the NAL
units with NRIs equal to 0. Adding such a use case would show one more
advantage of current  MTAP structures.

*	section 8.1.     MIME Registration:
		It seems that the MIME parameters init-buf-time,
interleaving-depth and max-don-diff are somehow related, when at least two
of them are explicitely singalled. For example, if init-buf-time is too
small compares to interleaving-depth (although interleaving-depth is
specified not in time units, but in terms of number of NAL units), video
images can not be displayed after init-buf-time is reached. 
	      
	So I am wondering: wouldn't it make sense to specify
interleaving-depth in terms of time units, rather than number of NAL units?
In that case, max-don-diff would provide supplemental information about the
interleaving depth in terms of number of NAL objects. 



	Thanks, 
	-Thomas


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



From exim@www1.ietf.org  Sat Dec 20 17:03:38 2003
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From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: <philippe.gentric@philips.com>, <avt@ietf.org>
Subject: RE: [AVT] I-D ACTION:draft-ietf-avt-text-red-00.txt
Date: Sat, 20 Dec 2003 23:01:57 +0100
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Philippe,
Question: Would not it be useful to explain what "red" stands for ?

Answer: In the introduction, it is for example stated "usage of RFC 2198 and
this MIME type for
   the transport of redundant text data".

Is that not clear enough?
----------------------------------------------------------------------------
---------------------------------------------------------------
Do we need to modify the Abstract as well to read:

"Abstract

   This document defines the text/red MIME sub-type for transmission of text
with redundancy in order to improve reliability.  The actual RTP
   packetization for this MIME type is specified in RFC 2198."



-------------------------------------------
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
philippe.gentric@philips.com
Sent: Friday, December 19, 2003 10:32 AM
To: avt@ietf.org
Subject: Re: [AVT] I-D ACTION:draft-ietf-avt-text-red-00.txt



Would not it be useful to explain what "red" stands for ?


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



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



From exim@www1.ietf.org  Sat Dec 20 18:08:36 2003
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To: Thomas Zeng <zeng@pvnetsolutions.com>
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Hi Thomas,

with respect to your H.241 related comments:
When referencing (informatively) standards of non-IETF bodies, it is always 
very difficult to find a good balance between being sufficient explanatory 
and making sure that one is, in the long term, correct.

H.241 includes many different sections, only one of which is concerned with 
the RTP paylaod for H.264.  The current language of H.241 specifies (in 
addition to other, unrelated subjects) an RTP payload format for H.264 
which is a small subset of our draft.  As for major features, 
fragmentation, aggregation, the MIME/SDP stuff and a few other aspects are 
not in H.241.

The main reason for being vague is that we may well add, say, aggregation 
to the next version of H.241.  Or we may, once our draft has become an RFC, 
go ahead and retire the annex where the payload spec is specified in, and 
add a reference to our then RFC, maybe with some additional language that 
disables some mechanisms of the RFC.  In the ITU-T, we can do this as long 
as the changes are technically upward compatible, and it is common practice 
to do so.  In the IETF, on the other hand, the payload spec will stay as an 
RFC unchanged, until a new RFC with a future version of the payload spec is 
issues (and that spec may or may not be upward/downward compatible).

For this reason, I'm a bit reluctant to add more explanatory language to 
our draft.  Instead, I would even consider removing the H.241 discussions 
altogether.  However, doing this is not such a great idea either, because 
there will be implemetors with previous H.241 implementation experience, 
which would likely find it beneficial to get some hint that they do not 
have to start from scratch when implementing our draft.

More comments inline.

At 11:13 PM 12/18/2003, Thomas Zeng wrote:

>*       Section 5.7.2: MTAPs
>There is a statement, quoted below:
>
>               "This memo does not specify how the NAL units within an MTAP
>are ordered, but, in most cases, NAL unit
>            decoding order SHOULD be used."
>
>         There is a case where, when partial checksum is available, from
>either UDP lite or DCCP, that the best way to order NAL units within a MTAP
>is by the reverse order of NRI, since NRI indicates the relative importance
>of a NAL unit, and that NRI=0 means that the NAL unit is not used as
>reference to any other NAL unit. Hence it makes sense to apply partial
>checksum on the first group of NAL units, upto but not including the NAL
>units with NRIs equal to 0. Adding such a use case would show one more
>advantage of current  MTAP structures.

That is why the language calls for a "SHOULD" and not a "MUST".  Doing what 
you suggests has benefits for the error resilience (when using the 
protocols you mentioned over bit error prone links) but has a negative 
impact on the latency.  I believe, most implementors of standards like this 
one should have a rough idea about the tradeoffs, and I don't think we need 
to spell it out in more detail.  Especially not when the only usage 
examples are not widely deployed (to say the least :-) protocols like UDP 
lite, or DCCP (or the modified FEC stuff).

>*       section 8.1.     MIME Registration:
>                 It seems that the MIME parameters init-buf-time,
>interleaving-depth and max-don-diff are somehow related, when at least two
>of them are explicitely singalled. For example, if init-buf-time is too
>small compares to interleaving-depth (although interleaving-depth is
>specified not in time units, but in terms of number of NAL units), video
>images can not be displayed after init-buf-time is reached.
>
>         So I am wondering: wouldn't it make sense to specify
>interleaving-depth in terms of time units, rather than number of NAL units?
>In that case, max-don-diff would provide supplemental information about the
>interleaving depth in terms of number of NAL objects.

Using time for specifying interleaving-depth would make sense if most/all 
of the NAL units of a stream were of roughly the same size.  However, this 
is not the case.  NAL units can be very small (i.e. for parameter sets, SEI 
messages, or Picture Delimiters) or large (slices).  A single picture can 
consist of dozens of NAL units -- this is not a theoretical option, but a 
very real scenario.  Interleaving of such NAL units (even within a picture) 
makes sense as it changes the error concealment properties.  Hence the 
interleaving is defined on NAL units, and it would not make sense to use 
time instead.

Thanks for your comments, and best regards,
Stephan


_______________________________________________
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: avt@ietf.org
Cc: magnus.westerlund@ericsson.com
Subject: Re: [AVT] WG last call: iLBC payload format and codec
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The last call period for the iLBC payload format and codec has concluded.
Since there were no objections, we will now request that these drafts are
published.

Colin




--> Colin Perkins <csp@csperkins.org> writes:
> This is to announce a working group last call on the RTP Payload Format
> and codec for iLBC speech:
> 
> 	draft-ietf-avt-rtp-ilbc-04.txt
> 	draft-ietf-avt-ilbc-codec-04.txt
> 
> Please send any final comments on these drafts to the mailing list by
> 19th December 2003.  If no substantive comments are received by that
> time, we will ask the IESG to publish the codec as an Experimental RFC
> and the RTP payload format as a Proposed Standard RFC.
> 
> -- 
> Colin Perkins
> http://csperkins.org/

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



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Date: Sun, 21 Dec 2003 13:28:42 +0000
From: Colin Perkins <csp@csperkins.org>
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Subject: Re: [AVT] WG last call: draft-ietf-avt-rtp-clearmode-04.txt
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The last call period for the RTP payload format for 64kbps transparant
calls has now concluded. Since there were no objections, we will request
that the draft be published as a Proposed Standard RFC.

Colin




--> Colin Perkins <csp@csperkins.org> writes:
> This is to announce a working group last call on the RTP payload format
> for 64 kbit/s transparent calls (draft-ietf-avt-rtp-clearmode-04.txt). 
> Please send any final comments on this draft to the mailing list by 19th
> December 2003. If no substantive comments are received by that time, we
> will ask the IESG to publish this draft as a Proposed Standard RFC. 
> 
> -- 
> 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  Mon Dec 22 03:30:39 2003
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Subject: RE: [AVT] I-D ACTION:draft-ietf-avt-text-red-00.txt
To: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
Cc: avt@ietf.org
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Date: Mon, 22 Dec 2003 09:26:06 +0100
X-MIMETrack: Serialize by Router on hbg001soh/H/SERVER/PHILIPS(Release 5.0.11  |July 24, 2002) at
 22/12/2003 09:31:06
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>Philippe,
>Question: Would not it be useful to explain what "red" stands for ?
>
>Answer: In the introduction, it is for example stated "usage of RFC 2198 and
>this MIME type for
>   the transport of redundant text data".
>
>Is that not clear enough?
>----------------------------------------------------------------------------
>---------------------------------------------------------------
>Do we need to modify the Abstract as well to read:
>
>"Abstract
>
>   This document defines the text/red MIME sub-type for transmission of text
>with redundancy in order to improve reliability.  The actual RTP
>   packetization for this MIME type is specified in RFC 2198."
>

actually since one may take "red" as an acronym
(hey it is capitalized in the defintion!)

why not simply process it as such:

"This document defines the text/red (red=redundant) MIME sub-type."

;-)

regards,

Philippe Gentric
Chief Architect
Philips Software
philippe.gentric@philips.com
http://www.software.philips.com


                                                                                                                                        
                                                                                                                                        
                                                    To:   Philippe Gentric/MP4-SUR/CE/PHILIPS@EMEA1                                     
                                                     <avt@ietf.org>                                                                     
                                                    cc:                                                                                 
                                                    Subject:    RE: [AVT] I-D ACTION:draft-ietf-avt-text-red-00.txt                     
               "Gunnar Hellstrom"                                                                                                       
               <gunnar.hellstrom@omnitor            Classification:                                                                     
               .se>                                                                                                                     
                                                                                                                                        
               Sent by:                                                                                                                 
               avt-admin@ietf.org                                                                                                       
                                                                                                                                        
               20/12/2003 23:01                                                                                                         
                                                                                                                                        
                                                                                                                                        




Philippe,
Question: Would not it be useful to explain what "red" stands for ?

Answer: In the introduction, it is for example stated "usage of RFC 2198 and
this MIME type for
   the transport of redundant text data".

Is that not clear enough?
----------------------------------------------------------------------------
---------------------------------------------------------------
Do we need to modify the Abstract as well to read:

"Abstract

   This document defines the text/red MIME sub-type for transmission of text
with redundancy in order to improve reliability.  The actual RTP
   packetization for this MIME type is specified in RFC 2198."



-------------------------------------------
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
philippe.gentric@philips.com
Sent: Friday, December 19, 2003 10:32 AM
To: avt@ietf.org
Subject: Re: [AVT] I-D ACTION:draft-ietf-avt-text-red-00.txt



Would not it be useful to explain what "red" stands for ?


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



_______________________________________________
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 Dec 22 12:10:39 2003
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Subject: [AVT] Last Call: 'Extended RTP Profile for RTCP-based Feedback(RTP/AVPF)
 to Proposed Standard
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The IESG has received a request from the Audio/Video Transport WG to 
consider the following documents:

- 'Extended RTP Profile for RTCP-based Feedback(RTP/AVPF) '
   <draft-ietf-avt-rtcp-feedback-07.txt> as a Proposed Standard
- 'Extended RTP Profile for RTCP-based Feedback - Results of the Timing 
   Rule Simulations '
   <draft-burmeister-avt-rtcp-feedback-sim-02.txt> as an Informational RFC

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send any comments to the
iesg@ietf.org or ietf@ietf.org mailing lists by 2004-01-09.

The files can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-avt-rtcp-feedback-07.txt
http://www.ietf.org/internet-drafts/draft-burmeister-avt-rtcp-feedback-sim-
02.txt


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



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CC: "'avt@ietf.org'" <avt@ietf.org>
Subject: Re: [AVT] Comments on draft-ietf-avt-rtp-h264-03.txt
References: <72263E8E8622D611975C0002B32C19D805A17E5B@misty.packetvideo.com>
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Just as a matter of general information, the ITU-T has changed their 
procedures in recent years to make it easier for other standards groups 
to cooperate with them.  Any individual can order up to three of their 
documents per year without charge.  Select the appropriate option at 
http://ecs.itu.ch/cgi-bin/ebookshop.

Thomas Zeng wrote:

...
> 
> 	I tried to download H.241 spec but since I don't have ITU
> subscription I am not able to find it. It would be helpful to add a brief
> explanation (if feasible) on why this " may not comply with ITU-T
> Recommendation H.241."   
> 
...>


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



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Subject: [AVT] Last Call: 'RTP payload format for a 64 kbit/s transparent call'
 to Proposed Standard
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The IESG has received a request from the Audio/Video Transport WG to 
consider the following document:

- 'RTP payload format for a 64 kbit/s transparent call '
   <draft-ietf-avt-rtp-clearmode-04.txt> as a Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send any comments to the
iesg@ietf.org or ietf@ietf.org mailing lists by 2004-01-15.

The file can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-clearmode-04.txt


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



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The IESG has received a request from the Audio/Video Transport WG 
to consider the following document:

- 'RTP Payload Format for iLBC Speech '
   <draft-ietf-avt-rtp-ilbc-04.txt> as a Proposed Standard

The document 

-  'Internet Low Bit Rate Codec'
    <draft-ietf-avt-rtp-ilbc-codec-04.txt>

specifies the codec for which this is the payload format, and will be
considered as an Experimental.

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send any comments to the
iesg@ietf.org or ietf@ietf.org mailing lists by 2004-01-15.

The file can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-ilbc-04.txt


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



From exim@www1.ietf.org  Sun Dec 28 15:41:52 2003
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Date: Tue, 23 Dec 2003 11:25:05 -0800
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From: Quy Nguyen <qhnguyen@cisco.com>
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Subject: [AVT] Ask Open source info for RTP/RTSP/MPEG2 Client
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Dear Experts,

I would like to have/find a free/open source code implementing the RTP/RTSP 
Client and working with MPEG-2 PES.

I found one but only work with MJPEG.

Best regards

MERRY CHRISTMAS and HAPPY NEW YEAR

Quy Nguyen

email :qhnguyen@cisco.com


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



