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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 Multiparty Multimedia Session Control Working Group of the IETF.

	Title		: Key Management Extensions for Session Description
                          Protocol (SDP) and Real Time Streaming Protocol (RTSP)
	Author(s)	: J. Arkko et al.
	Filename	: draft-ietf-mmusic-kmgmt-ext-08.txt
	Pages		: 16
	Date		: 2003-8-1
	
This document defines general extensions for SDP and RTSP to carry
the security information needed by a key management protocol, in
order to secure the media. These extensions are presented as a
framework, to be used by one or more key management protocols. As
such, its use is meaningful only when it is completed by the key
management protocol in use.
General guidelines are also given on how the framework should be used
together with SIP and RTSP.

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Section 3.2 of "draft-ietf-mmusic-rfc2326bis-04.txt" (the latest RTSP spec) 
notes that the "host" part of a RTSP URL is "As defined by RFC 2732".

However, RFC 2732 is "Format for Literal IPv6 Addresses in URL's", which 
probably isn't really what you want here (or, at least, is only part of 
what you want...)

	Ross.


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From exim@www1.ietf.org  Mon Aug  4 13:59:29 2003
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Aaron Falk'" <falk@isi.edu>,
        Jonathan Rosenberg
	 <jdrosen@dynamicsoft.com>
Cc: "Phelan, Tom" <tphelan@sonusnet.com>, "'dccp@ietf.org'"
	 <dccp@ietf.org>,
        "'mmusic@ietf.org'" <mmusic@ietf.org>,
        "Pasco Anderson, Jay" <jpanderson@sonusnet.com>
Subject: RE: [dccp] Re: [MMUSIC] Two-way media with DCCP and TFRC
Date: Mon, 4 Aug 2003 13:58:01 -0400 
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> So, is the right thing to do to ramp up at the beginning of the call
> to the best rate you can get, then only ramp down?  This overcomes
> Tom's concerns about probing bandwidth.
There is no point in "ramping".  You send at the rate you decide 
(however you decide).  If you have to change, you signal a change,
disrupt the flow, and change.  There isn't anything elastic here.
You send at a fixed rate.  If you don't recieve at the correct
rate you can handle a (small) amount of variation by buffering, but
mostly, you discard and hope it gets better.

Brian 

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From: "Phelan, Tom" <tphelan@sonusnet.com>
To: "'Rosen, Brian'" <Brian.Rosen@marconi.com>,
        "'Aaron Falk'"
	 <falk@isi.edu>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: "Phelan, Tom" <tphelan@sonusnet.com>, "'dccp@ietf.org'"
	 <dccp@ietf.org>,
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        "Pasco Anderson, Jay" <jpanderson@sonusnet.com>
Subject: RE: [dccp] Re: [MMUSIC] Two-way media with DCCP and TFRC
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Hi Brian

> > So, is the right thing to do to ramp up at the beginning of the call
> > to the best rate you can get, then only ramp down?  This overcomes
> > Tom's concerns about probing bandwidth.
> There is no point in "ramping".  You send at the rate you decide 
> (however you decide).  If you have to change, you signal a change,
> disrupt the flow, and change.  There isn't anything elastic here.
> You send at a fixed rate.  If you don't recieve at the correct
> rate you can handle a (small) amount of variation by buffering, but
> mostly, you discard and hope it gets better.

Unfortunately, with slow-start, you can't just start sending at an adequate
rate.  At the beginning of the connection you need to treat the rate that
TFRC sets differently from the way you treat it later, after you've
slow-started up to an adequate rate.  I think that's what Aaron was
referring to.

Tom P.

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From: Mark Handley <M.Handley@cs.ucl.ac.uk>
X-Organisation: UCL
To: "Phelan, Tom" <tphelan@sonusnet.com>
cc: "'Rosen, Brian'" <Brian.Rosen@marconi.com>, "'Aaron Falk'" <falk@isi.edu>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        "'dccp@ietf.org'" <dccp@ietf.org>,
        "'mmusic@ietf.org'" <mmusic@ietf.org>,
        "Pasco Anderson,
    Jay" <jpanderson@sonusnet.com>
Subject: Re: [dccp] Re: [MMUSIC] Two-way media with DCCP and TFRC 
In-reply-to: Your message of "Mon, 04 Aug 2003 14:08:24 EDT."
             <4CF48720B6E9D4118D040060CF2074E9082BB4BB@sonusdc3.sonusnet.com> 
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>> > So, is the right thing to do to ramp up at the beginning of the call
>> > to the best rate you can get, then only ramp down?  This overcomes
>> > Tom's concerns about probing bandwidth.
>> There is no point in "ramping".  You send at the rate you decide 
>> (however you decide).  If you have to change, you signal a change,
>> disrupt the flow, and change.  There isn't anything elastic here.
>> You send at a fixed rate.  If you don't recieve at the correct
>> rate you can handle a (small) amount of variation by buffering, but
>> mostly, you discard and hope it gets better.
>
>Unfortunately, with slow-start, you can't just start sending at an adequate
>rate.  At the beginning of the connection you need to treat the rate that
>TFRC sets differently from the way you treat it later, after you've
>slow-started up to an adequate rate.  I think that's what Aaron was
>referring to.

For VoIP, perhaps slowstart at startup is simply unnecessary.  Lets
look at what TCP does.  Assume a 1500 byte MSS, and 200ms RTT.  Thus
the starting rate for a TCP connection will be one packet/RTT, which
is 1500*8/0.2 = 60Kb/s.  And this is assuming an initial window of one
packet, which is pretty conservative these days.

So, a VoIP flow that started at anything up to 64Kb/s (including
headers) would be perfectly reasonable - it's no worse than TCP's
default behaviour.  If the RTT were a lot larger, you'd be expected to
start slower, but if it's a lot larger than 200ms, then you're going
to have interactivity problems anyway.

Now, the question would be what happens when this is too fast?  There
are two cases really:

 - Low stat mux at bottleneck.  Here TFRC will probably build a queue
   pretty rapidly, and a good TFRC implementation would slow down as a
   result of the RTT increasing.  Pretty soon you'll see some loss,
   and things will settle down to the slower steady state.

 - High stat mux at bottleneck.  TFRC won't immediately slow down -
   that will take some loss to achieve.  But the high level of stat
   muxing and the fact that each flow only sends too fast for a small
   fraction of it's duration mean that the overall effect should be
   relatively benign.

Thus neither case seems to be terribly bad.  But this only applies to
VoIP, where the bitrates are relatively benign compared to TCP's
startup behaviour, and even then if the RTT is high, you get to start
more slowly.

Cheers,
	Mark


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From: Aaron Falk <falk@isi.edu>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: "Phelan, Tom" <tphelan@sonusnet.com>, "'dccp@ietf.org'" <dccp@ietf.org>,
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Subject: Re: [dccp] Re: [MMUSIC] Two-way media with DCCP and TFRC
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<catching up...>

Jonathan Rosenberg wrote:
> 
> Firstly, as Brian has pointed out, rate adaptive codecs for VoIP are 
> very rare. Practically speaking, nearly all commercial VoIP phones use 
> single rate codecs. Many can support multiple codecs. However, 
> frequently, once a codec is selected, it cannot be changed easily 
> during the call. This is commonly the case for hardware phones, where 
> I believe the codecs are stored in flash and downloaded into RAM once 
> a codec is selected in the call. We have had a lot of folks ask how, 
> in SIP, to signal that the called party has to pick one of the N 
> offered codecs, and then stick to it. RFC 3264 describes means for 
> doing that.
> 
> Such a phone would not permit rate adaptivity by switching codecs as 
> you describe below. Thus, it would be all or nothing in terms of 
> sending rate.


Jonathan-

Looking down the road, do you think that rate adaption is a reasonable
thing to expect of future VoIP phones?  I.e., I'd like to understand
if the design is being influenced by technology limitations which will
reasonably go away.

> Even assuming there was rate adaptivity for voice, there are 
> perceptual issues when the encoding is frequently changed. It is 
> apparently more annoying for users to oscillate between high and low 
> rate codecs, than to stay constant with a medium rate codec. I seem to 
> recall some studies along these lines in the late 90s by Jean Bolot 
> and his students, though I cannot find the specific reference.


This seems a point less likely to go away.  Perhaps this is (one
reason) why rate adaption isn't in current phones.

So, is the right thing to do to ramp up at the beginning of the call
to the best rate you can get, then only ramp down?  This overcomes
Tom's concerns about probing bandwidth.

--aaron  

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Subject: Re: [dccp] Re: [MMUSIC] Two-way media with DCCP and TFRC
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Rosen, Brian wrote:
> >
> > So, is the right thing to do to ramp up at the beginning of the call
> > to the best rate you can get, then only ramp down?  This overcomes
> > Tom's concerns about probing bandwidth.
>
> There is no point in "ramping".  You send at the rate you decide 
> (however you decide).  If you have to change, you signal a change,
> disrupt the flow, and change.  There isn't anything elastic here.
> You send at a fixed rate.  If you don't recieve at the correct
> rate you can handle a (small) amount of variation by buffering, but
> mostly, you discard and hope it gets better.

Brian-

I'm concerned about the effect "sending at a fixed rate" might have on
other traffic in the network.  "Ramping", or slow start, at the
beginning of the call allows the sender to probe for the available
capacity.  Otherwise, how does the sender know what rate to start
sending at?  If he starts too high, he'll blow away a bunch of TCP
flows while he figures out his loss rate is too high.  Starting too
low will result in the minimal quality to users.  Am I missing
something here?

I do agree that, since codecs output rates are quantized, ramping down
really means jumping from one to the next lower rate.

--aaron

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cc: "'Rosen, Brian'" <Brian.Rosen@marconi.com>, "'Aaron Falk'" <falk@isi.edu>,
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From: Sally Floyd <floyd@icir.org>
Subject: Re: [dccp] Re: [MMUSIC] Two-way media with DCCP and TFRC 
Date: Mon, 04 Aug 2003 12:13:26 -0700
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>For VoIP, perhaps slowstart at startup is simply unnecessary.  Lets
>look at what TCP does.  Assume a 1500 byte MSS, and 200ms RTT.  Thus
>the starting rate for a TCP connection will be one packet/RTT, which
>is 1500*8/0.2 = 60Kb/s.  And this is assuming an initial window of one
>packet, which is pretty conservative these days.


Initial windows:

The current standard for TCP, RFC 3390 on Increasing TCP's Initial Window
(Proposed Standard), permits an initial window of
    min (4*MSS, max (2*MSS, 4380 bytes))
bytes. That is, at least two MSS, and up to four MSS, up to an initial
window of 4380 bytes.  TCP Initial windows of two packets are not uncommon,
and initial windows of three or four packets can occasionally be
observed.  ("http://www.icir.org/tbit/June2003/initwin.512.txt").

The question on the table seems to be whether RFC 3390 could be adapted
for low-rate VoIP flows using DCCP to allow initial windows of 4380 bytes
regardless of packet size.

It certainly seems reasonable to me.  (I am agreeing with Mark's previous
email on this.)

If the congested link is a low-bandwidth one, then the limitation is in
bytes per second rather than packets per second, and 4380 bytes
in small, spaced-out packets would be less of a load than 4380 bytes 
in 2-4 larger packets.

If the congested link is a high-bandwidth one, and the limitation is
in bytes per second, then the argument above still holds.

If the congested link is a high-bandwidth one and the limitation
is in packets per second, then it is still the case that the VoIP
flows would be pacing their packets out, which would limit the
damage they would do if 4380 bytes was too high for the allowed
sending rate.  So it strikes me as liveable, for a rate-limited,
low-bandwidth, long-lifetime app like VoIP, to resume with sending rates
up to 4380 bytes per RTT after a quiescent period.


Quiescent periods:

The question then would be whether best-effort VoIP could, after a long
quiescient period, resume sending with a sending rate of up to 4380 bytes
per RTT (assuming that it got to send at least that much before
the quiescent period).  Again, this seems plausible to me, for the
reasons that Mark described (though it would of course have to be
looked at a bit more).


And the diatribe about best-effort service:

I can't imagine a viable justification for a best-effort flow to
resume sending at a sending rate significantly higher than 4380
bytes per RTT after a long idle period.   My own assumption is that
there are certain rules that go along with best-effort traffic, and
one of them is that you don't trash competing TCP traffic.  Flows
that have strict requirements for QoS probably aren't going to get
their needs met with best-effort service with or without DCCP or
other forms of congestion control.

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

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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Sally Floyd'" <floyd@icir.org>, Mark Handley <M.Handley@cs.ucl.ac.uk>
Cc: "'Aaron Falk'" <falk@isi.edu>,
        Jonathan Rosenberg
	 <jdrosen@dynamicsoft.com>,
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        "'mmusic@ietf.org'" <mmusic@ietf.org>,
        "Pasco Anderson, Jay"
	 <jpanderson@sonusnet.com>
Subject: RE: [dccp] Re: [MMUSIC] Two-way media with DCCP and TFRC 
Date: Mon, 4 Aug 2003 17:06:59 -0400 
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Two comments:

All of these calculations break if it's video rather than audio.
We need reasonable rules for video also.  The same problems
occur, although some video algorithms are not constant data rate.

Also
> Quiescent periods:
> 
> The question then would be whether best-effort VoIP could, 
> after a long
> quiescient period, resume sending with a sending rate of up 
> to 4380 bytes
> per RTT (assuming that it got to send at least that much before
> the quiescent period).  Again, this seems plausible to me, for the
> reasons that Mark described (though it would of course have to be
> looked at a bit more).
This of course says that it's all right to do it.  I simply want
to point out to those not familiar that this can happen.  Most
audio codecs employ Voice Activity Detection.  They don't send
compressed data when no one is talking.  Incidentally, they do 
play locally sourced "comfort noise" to the receiver because it
turns out to be disconcerting to hear real silence.  Some codecs
send periodic messages indicating silence should continue (sort of
a keep-alive), but they are small, and spaced out.  There are
several circumstances (a large conference for example), where it
is normal to not have anything said in one direction for a very
long time.

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        Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
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From: Sally Floyd <floyd@icir.org>
Subject: Re: [dccp] Re: [MMUSIC] Two-way media with DCCP and TFRC 
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>There are
>several circumstances (a large conference for example), where it
>is normal to not have anything said in one direction for a very
>long time.

Yep.  For an illustration of this, you could see Section 4.1 on "Lessons
from audio traffic" from the following:

Comments on Measurement-based Admissions Control for Controlled-Load Services.
Floyd, S., Technical Report, July 1996.
URL "http://www.icir.org/floyd/papers.html".

- Sally

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From: Sally Floyd <floyd@icir.org>
Subject: Re: [dccp] Re: [MMUSIC] Two-way media with DCCP and TFRC 
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Jonathan -

>My take on draft-iab-congestion is that, for voip, what the draft 
>effectively implies is that you will send at your specific 
>codec-defined rate, and then when congestion hits the defined 
>percentage, stop. That is, there will be a binary choice of either 
>sending or not, as opposed to a continuum of rates.
>
>I dont think that this is a bad thing, but does mean that the 
>threshold loss rate is incredibly important. The current draft talks 
>about relatively high loss rates, ones where the voice quality is 
>probably already unacceptable. However, it is my understanding that 
>these rates are very sensitive to the assumed TCP parameters, and that 
>they could be quite a bit lower if computed differently.

Yep, but the allowed loss rates could also be quite a bit higher
with yet a different set of TCP parameters.  E.g., for a TCP connection
with reasonably fine-grained timestamps, implemented so that a
backed-off retransmit timer is reset when a retransmitted packet
makes it through and is acked.  (This would be perfectly conformant
with the standards.)  It is on my to-do list to run these simulations,
to compare with the simulations in the draft, just to illustrate
the range of possible TCP behaviors.

For all of the simulations in that draft, a useful validation would be for
someone to run experiments with a TCP connection with 100 ms round-trip
times, with dummy-net introducing random packet drops with probability p.

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

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From: "Phelan, Tom" <tphelan@sonusnet.com>
To: "'Rosen, Brian'" <Brian.Rosen@marconi.com>,
        "'Sally Floyd'"
	 <floyd@icir.org>,
        Mark Handley <M.Handley@cs.ucl.ac.uk>
Cc: "'Aaron Falk'" <falk@isi.edu>,
        Jonathan Rosenberg
	 <jdrosen@dynamicsoft.com>,
        "'dccp@ietf.org'" <dccp@ietf.org>,
        "'mmusic@ietf.org'" <mmusic@ietf.org>,
        "Pasco Anderson, Jay"
	 <jpanderson@sonusnet.com>
Subject: RE: [dccp] Re: [MMUSIC] Two-way media with DCCP and TFRC 
Date: Tue, 5 Aug 2003 09:46:32 -0400 
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Hi Brian,

> All of these calculations break if it's video rather than audio.
> We need reasonable rules for video also.  The same problems
> occur, although some video algorithms are not constant data rate.

Do we have the same problem that we have with voice?  That is, silence
suppression reduces the transmit rate to zero for many RTTs.  Does
"stillness" suppression also reduce the rate to zero?  Or is there a minimum
rate that's greater than zero?

Tom P.

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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Phelan, Tom'" <tphelan@sonusnet.com>, "'Sally Floyd'" <floyd@icir.org>,
        Mark Handley <M.Handley@cs.ucl.ac.uk>
Cc: "'Aaron Falk'" <falk@isi.edu>,
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	 <jdrosen@dynamicsoft.com>,
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Subject: RE: [dccp] Re: [MMUSIC] Two-way media with DCCP and TFRC 
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Generally, there is no "silence".  On some algorithms, if you
remain REALLY still, the data rate drops very low, but I've never
seen zero.

Brian

> -----Original Message-----
> From: Phelan, Tom [mailto:tphelan@sonusnet.com]
> Sent: Tuesday, August 05, 2003 9:47 AM
> To: 'Rosen, Brian'; 'Sally Floyd'; Mark Handley
> Cc: 'Aaron Falk'; Jonathan Rosenberg; 'dccp@ietf.org';
> 'mmusic@ietf.org'; Pasco Anderson, Jay
> Subject: RE: [dccp] Re: [MMUSIC] Two-way media with DCCP and TFRC 
> 
> 
> Hi Brian,
> 
> > All of these calculations break if it's video rather than audio.
> > We need reasonable rules for video also.  The same problems
> > occur, although some video algorithms are not constant data rate.
> 
> Do we have the same problem that we have with voice?  That is, silence
> suppression reduces the transmit rate to zero for many RTTs.  Does
> "stillness" suppression also reduce the rate to zero?  Or is 
> there a minimum
> rate that's greater than zero?
> 
> Tom P.
> 

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



From exim@www1.ietf.org  Tue Aug  5 11:45:33 2003
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Henning Schulzrinne'" <hgs@cs.columbia.edu>,
        "Phelan, Tom"
	 <tphelan@sonusnet.com>
Cc: "'Sally Floyd'" <floyd@icir.org>, Mark Handley <M.Handley@cs.ucl.ac.uk>,
        "'Aaron Falk'" <falk@isi.edu>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
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        "'mmusic@ietf.org'" <mmusic@ietf.org>,
        "Pasco Anderson, Jay" <jpanderson@sonusnet.com>
Subject: RE: [dccp] Re: [MMUSIC] Two-way media with DCCP and TFRC
Date: Tue, 5 Aug 2003 10:31:17 -0400 
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Yeah, and this is one of the most difficult scenarios to get to work.
You are not sending anything, then all of a sudden, you are sending
full rate audio and video.  In some codecs, the start of video is an
instant peak rate transfer, and if you lose some of it, it causes
retries, or really annoying visual results.  The audio start is very
important; you have to get the audio through, and you don't have
anything like a second to get it up.  The result of doing it
wrong is clipping of the initial utterance, which is really annoying
and causes people to get into really weird behavior.

Brian

> -----Original Message-----
> From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
> Sent: Tuesday, August 05, 2003 9:51 AM
> To: Phelan, Tom
> Cc: 'Rosen, Brian'; 'Sally Floyd'; Mark Handley; 'Aaron 
> Falk'; Jonathan
> Rosenberg; 'dccp@ietf.org'; 'mmusic@ietf.org'; Pasco Anderson, Jay
> Subject: Re: [dccp] Re: [MMUSIC] Two-way media with DCCP and TFRC
> 
> 
> In a conference, it is quite common that video-follows-audio, 
> i.e., only 
> the current (and maybe the most recent) speaker sends video. 
> Thus, the 
> problem is somewhat similar - when somebody starts speaking, 
> you want to 
> immediately have decent-quality video of that speaker.
> 
> Phelan, Tom wrote:
> 
> > Hi Brian,
> > 
> > 
> >>All of these calculations break if it's video rather than audio.
> >>We need reasonable rules for video also.  The same problems
> >>occur, although some video algorithms are not constant data rate.
> > 
> > 
> > Do we have the same problem that we have with voice?  That 
> is, silence
> > suppression reduces the transmit rate to zero for many RTTs.  Does
> > "stillness" suppression also reduce the rate to zero?  Or 
> is there a minimum
> > rate that's greater than zero?
> > 
> > Tom P.
> > 
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www1.ietf.org/mailman/listinfo/mmusic
> 

_______________________________________________
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Subject: Re: [dccp] Re: [MMUSIC] Two-way media with DCCP and TFRC
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In a conference, it is quite common that video-follows-audio, i.e., only 
the current (and maybe the most recent) speaker sends video. Thus, the 
problem is somewhat similar - when somebody starts speaking, you want to 
immediately have decent-quality video of that speaker.

Phelan, Tom wrote:

> Hi Brian,
> 
> 
>>All of these calculations break if it's video rather than audio.
>>We need reasonable rules for video also.  The same problems
>>occur, although some video algorithms are not constant data rate.
> 
> 
> Do we have the same problem that we have with voice?  That is, silence
> suppression reduces the transmit rate to zero for many RTTs.  Does
> "stillness" suppression also reduce the rate to zero?  Or is there a minimum
> rate that's greater than zero?
> 
> Tom P.
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic


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Subject: [MMUSIC] Call for Papers: WiOpt'04
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Apologies if you receive multiple copies of this call!

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

Note:  The submission deadline is 5th of November, 2003.


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


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


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

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

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

Plenary Speakers:

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


IMPORTANT DATES:
Submission of extended abstract: Nov.  5, 2003.
Notification of acceptance:      Jan. 15, 2003.
Camera ready copy:               Feb. 15, 2003.

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



General chairman: Imrich Chlamtac

Vice General Chair: Roger Whitaker 

Program Co-Chairs: Marco Conti and Tamer Basar

Program committee of the conference:

Eitan Altman, Venkat Anantharam, Konstantin Avrachenkov, Andrew
Campbell, A. Chockalingam, Jon Crowcroft, Silvia Giordano, Stephen
V. Hanly, Jean-Pierre Hubaux, Holger Karl, Peter Key, P. R. Kumar,
Peter Marbach, Philippe Nain, Sven Oestring, Yoshikuni Onozato, Ramesh
Rao, Ness Shroff, Violet Syrotiuk, Thierry Turletti, Adam Wolisz, Jeff
Wieselthier

Local Arrangements Co-Chair: Sven Oestring and Julian Chesterfield

Publicity Chair: Holger Karl


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



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Date: Sun, 10 Aug 2003 14:42:11 +0200
From: Joerg Ott <jo@tzi.uni-bremen.de>
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Folks,

attached are the draft MMUSIC minutes from our meeting in Vienna.

The slides are available at

http://www.dmn.tzi.org/ietf/mmusic/57/slides/57-mmusic-slides-ppt.zip
http://www.dmn.tzi.org/ietf/mmusic/57/slides/57-mmusic-slides-pdf.zip

Joerg

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

Draft Minutes of the MMUSIC WG Meeting at the 57th IETF
=======================================================

Reported by Joerg Ott and Colin Perkins
Notes taken by Colin Perkins

The MMUSIC WG met once at the 57th IETF.  The meeting was chaired by
Colin Perkins and Joerg Ott.  It was attended by some 90 participants.

Status Update
-------------

Joerg Ott reported on the status of the working and the general
progress since the 56th IETF in San Francisco:

- The new charter was finally approved.

- RFC 3524 was published on SDP extensions for reservation flows.

- Two documents are with the RFC editor: SDP extensions for NAT
   traversal and SDPng transition.

- Two documents are with the IESG, and comments have been received:

   - draft-rajeshkumar-mmusic-gpmd-03.txt was felt to overlap with work
     on capability negotiation carried out elsewhere; it was noted that
     this probably applies to most of SDP except that SDP was first.
     The chairs and authors will work with the IESG to sort this out.
   - draft-ietf-mmusic-sdp-new-13.txt received many comments which were
     discussed in a separate slot (see below).

- Two documents are under AD review:
   - draft-ietf-mmusic-sdp-srcfilter-05.txt
   - draft-ietf-mmusic-sdp-comedia-05.txt

- draft-ietf-mmusic-sdp-bwparam-04.txt is completing WGLC on 28 Jul
   2003.

- The SDP extensions for key management
   (draft-ietf-mmusic-kmgmt-ext-07.txt) have been waiting for the MIKEY
   work to complete which seems to progress.  A further revision is
   expected shortly after the IETF.

- The SDP implementation draft draft-ietf-mmusic-sdp-implem-00.txt has
   not been revised yet, but Tom Taylor is working on it.  He is trying
   to incorporate further implementation experience from inside his
   company as well as from the SIPit events.

Two potential new work items have been discussed for MMUSIC:

- Real-time fax (draft-ietf-sipping-realtimefax-01.txt) from the
   SIPPING WG is to provide implementers with guidelines on how to use
   registered SDP attributes for ITU-T T.38.  This document actually
   contains two logical parts: the use of offer/answer mechanics (which
   belongs to MMUSIC) and a discussion of SIP-specific signaling (which
   belongs to SIPPING).  It is suggested to investigate splitting the
   document to deal with each one in the respective WG.  But it was
   pointed out that this document merely contains examples only and
   hence should be done as a whole in just one WG.  To be resolved.

- Interactive connectivity establishment (ICE) also from the SIPPING WG
   (draft-rosenberg-sipping-ice-01.txt).  This is discussed later on
   the agenda.

Revised SDP (draft-ietf-mmusic-sdp-new-13.txt)
----------------------------------------------

Colin Perkins reported on the changes incorporated in the latest
revision of SDP: It was attempted to clarify when the k= attribute may
be used.  Further clarifications include: RTP may be used with
non-consecutive port numbers and a=charset is only allowed as a
session level attribute.  Also, unregistered X- attributes and media
format are deprecated.  Finally, IANA considerations and references
were updated.

Colin further reported on issues raised on revision -13 on the mailing
list, as well as from the IESG review.

- An ABNF inconsistency was found: a comment in the ABNF for
   "token-char" allowed for a larger set of characters than the actual
   definition.  One could either (1) change the comment to match the
   actual grammar (and thus reduce the set of allowed characters) or
   (2) modify the grammar to match the comment.  The group agreed to go
   with option (1), pending a check against the use of SDP in RFC 3108.
   Tom Taylor volunteered to find this out.

- The use of b= with layered coding should be clarified as follows:
   When using the CT modifier with RTP, if several RTP sessions are
   part of the conference, the conference total refers to total
   bandwidth of all RTP sessions.  This was accepted in principle but
   it was also pointed out that a bandwidth specification per layer
   would be helpful.  Further discussion is needed on the mailing
   list.

- Again with layered coding, it should be clarified which ports are
   associated with an m= line.  The clarification proposal was
   accepted.

- The k= is underspecified.  The most straightforward way would be
   to deprecate the use of this field altogether.  Opinions from
   implementers are sought.  It was reported that MS Messenger uses
   this field; the details are unknown at this moment though.  As there
   are implementation out there, just deprecating this field won't work.

   Therefore, further work needs to go into the specification of using
   k=.  In particular, when k= shall be used, a secure channel must be
   available and endpoints need to be sure that no third party can
   eavesdrop on the key.  These issues, however, need to be specified
   by those using SDP rather than SDP itself.

   The k= field currently supports an algorithm indicator, but it should
   also specify a key type.  This may be implied by the URI or the
   media protocol used.  For more complex scenarios, e.g. when more
   than one key is needed, either the sdescriptions work
   (draft-ietf-mmusic-sdescriptions-01.txt) or the key management
   (draft-ietf-mmusic-kmgmt-ext-07.txt) work should be used.

   It was proposed that the security considerations section should
   change "SHOULD NOT deliver the user into an interactive multimedia
   session..." to "MUST NOT...".  It was found that this is an issue of
   user authorization (e.g. the user may be able to pre-authorize the
   automatic joining of certain interactive sessions) and needs to be
   dealt with by the protocols using SDP.

- The suggestion to define a "IN" value for private address spaces was
   rejected because systems 1) cannot tell whether they are in a
   private address space (relative to their peers) and 2) this change
   would be not backward compatible.

- It will be clarified that UTF-8 normalization is not needed because
   comparisons are not done on fields with UTF-8.

- Proposal on a=inactive: it was agreed not to change the SDP spec here
   (this is an issue for users of SDP, which may establish conventions
   for use, not for SDP itself).

- Wording should be added that u= and k= may include URIs that may be
   dereferenced in some (but not all) cases.  Also, it should be noted
   a k= URI may typically be an http or https URI.  Both suggested
   wording additions were accepted.

Colin will fold the comments and the discussion into the document and
produce a new revision out within a couple of weeks.


SDP Security Descriptions (draft-ietf-mmusic-sdescriptions-01.txt)
------------------------------------------------------------------

Dan Wing briefly reviewed the document and the changes from the
previous revision: a new syntax is proposed for the a=crypto attribute
and the offer/answer part is revised which may need further review.
For the next steps, Colin noted that an IANA considerations section is
needed, including a discussion of the registration procedures.
Furthermore, reviewers for finishing the document were sought: Tom
Taylor, Magnus Westerlund, and Martin Euchner volunteered.


Offer/Answer Examples (draft-johnston-mmusic-offer-answer-examples-01.txt)
--------------------------------------------------------------------------

Alan Johnston presented the changes to the document: in particular, an
example for asymmetric use of a codec has been added.  This is to be
achieved using two distinct m= lines with the same port number, one
indicating "sendonly", the other "recvonly".  The proposed behavior
caused some discussion: it was felt that the mechanism was
underspecified and that (therefore) m= lines with the same port would
be a bad idea.  Yet, the right approach to achieve the desired
behavior seemed unclear.  Gonzalo Camarillo pointed out that the
RFC3388 provided a similar example, but with an explicit grouping
(which is not present in this draft).  The applicability of RFC3388 to
this problem was discussed further.  Orit Levin pointed out that the
intention clearly is to use RFC 3388 but the question is what happens
to the endpoints that don't support RFC 3388.  Gonzalo argued that
this is already defined in RFC 3388.  The chairs stated that they
would like to see the asymmetric codec behavior with two m= line
properly specified in a document.  The present offer/answer examples
draft is meant to be illustrative; the example should be kept in place
(with added grouping) but the normative text for this behavior needs
to be someplace else.  Flemming Andreasen suggested to specify part of
this practice in the revised SDP.

SDPng Update (draft-ietf-mmusic-sdpng-06.txt)
---------------------------------------------

Dirk Kutscher presented the latest progress on SDPng, addressing the
open issues from draft -06 (a new revision still needs to be
submitted).  The SDPng structure has been enhanced to decouple
capability negotiation for potential configurations from actual
configuration (and the respective session parameters).  While the
capability model has been kept, the syntax has been revised to
exclusively rely on XML mechanisms (rather than embedding own syntax
conventions for attributes) -- which is less efficient but much
cleaner.  Dirk presented the new syntax using several examples and
also showed a formal schema definition.  A further refinement from
draft -06 is to remove the concept of libraries and use inline
definitions instead -- so that SDPng documents would be self-contained.
There was little discussion.  Steve Casner asked whether this work was
getting enough traction and questioned whether people were really
preparing for a transition.  Joerg Ott argued that people are
desparately holding onto SDP (which is perceived easier with one small
extension at a time) -- but that examples show that this getting more
complicated so that SDP is likely to have only a finite lifetime.  So
it would be nice to get broader review from the community on a
replacements for SDP.


RTP DoS Attacks (draft-rosenberg-mmusic-rtp-denialofservice-00.txt)
Interactive Connectivity Establishment (draft-rosenberg-sipping-ice-01.txt)
---------------------------------------------------------------------------

Jonathan Rosenberg introduced the RTP DoS problem: An attacker may
easily direct the RTP stream(s) from an unsuspecting source to any
target by providing the target's IP address in the SDP.  This works
for SIP and RTSP -- with RTSP providing limited protection by allowing
unicast RTP streams only to the address of the originator (but NATs
make it worse as anyone behind the same NAT can easily be attacked).
As authentication mechanisms can at most help identify the attacker
(which may be very weak though), some kind of handshake mechanism is
needed so that sources do not start sending media streams unless they
know that the right receiver is actually waiting for it: a
request/response mechanism using the RTP ports is needed.  This is ICE
(see below) which has been chosen as a solution for SIP.  However, ICE
has broader applicability and may be appropriate for RTSP, too --
which needs to be determined by the group.  Jonathan reviewed the
properties of the ICE approach and briefly walked through the basic
ICE algorithm.  With ICE being signaling protocol independent, MMUSIC
is asked to take ICE up as a WG item.

ICE is felt to introduce quite some complexity for RTSP endpoints.
If, however, NAT traversal (which is an RTSP issue on its own, see
below) and security can be addressed by the same mechanism, it may be
worth the effort.  It is noted that even if we go for the ICE
mechanism for RTSP right now, this does not solve the problem as long
as old RTSP servers are still out there.

Regarding the ICE proposal as a WG item Magnus Westerlund argued that
if it is accepted, one should split the draft into several parts: the
SIP use, the ICE mechanism, the use with offer/answer, etc.
There was consensus in the group to accept ICE as new WG item, and to
fully document the RTP DoS issues as an informational document.


RTSP Update (draft-ietf-mmusic-rc2326bis-04.txt
-----------------------------------------------

Magnus Westerlund reported on the progress of the RTSP revision,
achieved by having a new co-editor joining the team, many
teleconference calls, and also quite some list discussion.  He
presented a number of changes from draft -03 (see the slides).  With
this, most known open issues are resolved, with only a few ones
remaining (see slides).  But many resolutions still need to be folded
into an updated revision of the draft.  When this is done, a second
phase of editing on the spec is needed: examples need to be fixed, the
spec's consistency needs to be checked as needs it overall
readability, and, potential new open issues need to be solved.  With
all this, the revised RTSP spec will not meet the October deadline as
currently documented in the MMUSIC charter.  Magnus suggests another
six months would be appropriate to finalize the work.


RTSP and NAT (draft-ietf-mmusic-rtsp-nat-01.txt)
------------------------------------------------

Magnus Westerlund introduced the revised draft on RTSP and NATs, the
purpose of which is to describe how to traverse NATs and firewalls
with RTSP.  It documents several possible NAT traversal approaches and
gives recommendations regarding RTSP for firewalls.  Approaches
requiring only modifications on the client side STUN, TURN, and
RTP/RTSP tunneled in RTSP.  Application-Layer Gateways (ALGs) for RTSP
will not require any modifications -- but are generally discouraged.
Magnus further discussed open issues, particurlarly the goals for RTSP
for NATs (i.e. what type of scenarios should be achievable) as well as
possible solutions (symmetric RTP, STUN, ICE, DCCP).

On the ALGs: The group generally agreed that ALGs are a bad thing and
should not be considered further (except for documentation purposes and
to communicate this clearly to outside the IETF -- and fast, as people
start deploying these).

On the document's purpose: Commenters state that, currently, the draft
does not give conclusive resolution of what to do as an implementer,
it is merely an inventory.  It is proposed to pick some solution and
write it up, e.g. in the RTSP spec.  Magnus wants to resolve the
structure of the document and then settle on a solution: one for the
server and one or more for the client side.  Jonathan Rosenberg and
Tom Marshal argue over the number of client-side solutions to be
documented: Jonathan wants only one while Tom prefers to have multiple
feasible solutions document.  In case multiple ones are documented,
Jonathan wants exactly one "MUST implement" solution to ensure
interoperability.

There was some further discussion on technical aspects of possible
solutions, but no real conclusions were reached.


Internet Media Guides (IMG)
---------------------------
draft-nomura-mmusic-img-requirements-01.txt
draft-nomura-mmusic-img-framework-01.txt

After a brief introduction by Joerg Ott, Rod Walsh presented the
progress on the IMG requirements document (which will be re-submitted
as WG draft by end of August).  He briefly reviewed the general
topology and IMG distribution aspects and then went through the
requirement categories identified in the draft: IMG description
requirements and IMG transport requirements.  The former primarily
address IMG envelopes for existing meta-data formats, the latter their
distribution in unicast and multicast, pull and push style scenarios.
For multicast push, SAPv2 is probably inadequate, so a successor is
sought, e.g. ALC from the RMT WG.  It is pointed that additions to ALC
need to be specified separately; while Rod wants to use the RMT
proposal FLUTE, Lorenzo argues that this may not be 100% appropriate
here; Lorenzo also points out that possible tasks in the RMT WG need
to happen quickly as the group is trying to wrap up.  For unicasting,
client-initiated actions could use HTTP while notifications from the
server could use SIP mechanims.

Rod discusses on the scalability requirements IMG transport and
presents a conceptual overview showing the different pieces of an IMG
framework and outlines the achieved flexibility.  He also addresses
data relationships and particularly security considerations as taken
into account in the requirements draft.

Rod finally discusses the way ahead: the requirements draft is pretty
close to finalization; the remaining efforts are mainly editorial and
consolidation.  The aim is to go for WGLC soon.  He also points to the
existing framework draft that will be revised by the end of August.





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Subject: Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-kmgmt-ext-08.txt
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On Friday, Aug 1, 2003, at 20:28 Europe/London, 
Internet-Drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts 
> directories.
> This draft is a work item of the Multiparty Multimedia Session Control 
> Working Group of the IETF.
>
> 	Title		: Key Management Extensions for Session Description
>                           Protocol (SDP) and Real Time Streaming 
> Protocol (RTSP)
> 	Author(s)	: J. Arkko et al.
> 	Filename	: draft-ietf-mmusic-kmgmt-ext-08.txt
> 	Pages		: 16
> 	Date		: 2003-8-1
> 	
> This document defines general extensions for SDP and RTSP to carry
> the security information needed by a key management protocol, in
> order to secure the media. These extensions are presented as a
> framework, to be used by one or more key management protocols. As
> such, its use is meaningful only when it is completed by the key
> management protocol in use.
> General guidelines are also given on how the framework should be used
> together with SIP and RTSP.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-kmgmt-ext-08.txt

I have a number of comments on this draft:

  - The ABNF in Sections 2.1 and 2.2 is contradictory. The key management
    data is defined as "text" in one section, and "quoted-string" in the
    other, yet these two types have different ranges of valid characters
    (and both are more liberal than the explanatory text, which implies
    that only base-64 encoded data is allowed). Similar for the protocol
    identifier: the definitions of "non-ws-string" in SDPnew and "token"
    in RTSP are not consistent.

  - It should be stated that the protocol identifiers in Section 2.1 and
    Section 2.2 refer to the same set of values, and a reference to the
    registry of legal values should be given (i.e. reference the IANA
    parameter registry defined in Section 6).

  - The draft repeatedly mentions Kerberos, and indicates that Kerberos
    tickets may be included in key management messages, but does not
    specify how this is to be achieved. Should either specify how to use
    Kerberos, or remove all references to it.

  - The second bullet point on Page 6 mentions that other parameters may
    need to be specified for certain key management protocols, but does
    not specify how an implementor knows if a particular key management
    protocol uses additional parameters. Suggest this is listed as part
    of the IANA registration for a particular key management protocol?

  - Section 3.1 seems to assume Offer/Answer usage. It is not clear how
    it applies to SDP used with, for example, SAP, HTTP, or email? What
    about one-to-many distribution of SDP? Or distributions where there
    is no back-channel? It is acceptable for the draft to be explicit
    about the limits of its scope, but it cannot ignore the wider problem
    space.

  - The discussion of authentication on Page 7 is not mentioned in the
    list of things senders and receivers of SDP must do, on pages 5 and
    6.

  - Section 3.3 does not mention authentication. Should I assume that
    there is no need for authentication when key management is used with
    RTSP?

  - Section 4 needs to provide strict guidance about the parameters and
    specifications needed for new key management protocols. The interface
    between the key management protocol and this framework should be made
    explicit.

  - I find the arguments about security of the SDP in Section 5 weak. The
    draft needs to be explicit about the threats it prevents and those it
    leaves open. Recommend re-reading RFC 3553, Section 5.

  - In section 5, how is a beat-down attack prevented when the attacker
    wishes to beat-down to no security? What is the correct response to
    an answer which suggests no security?

  - Section 6 needs to register the RTSP extensions needed, and make it
    clear that the namespace for key management protocol identifiers is
    shared between the RTSP and SDP extensions.

  - Section 6 mentions that new key management protocols specify "the
    syntax of possible additional arguments", yet this contradicts the
    ABNF which states that only a single argument is present.

The main issue behind these seems to be the desire to define new 
parameters which can be left open for future uses. As was mentioned in 
the context of the "a=gpmd:" attribute, this is not acceptable: there 
needs to be a precise definition of the semantics of any new SDP 
attribute/RTSP header before the draft can advance.

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


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I have question related to dynamic payloads. If a session is estblished =
with a dynamic payload type number mapped to a particular codec, can the =
number change for the same codec in subsequent offers/answers in the =
same session? e.g. Lets say a session was established with a dynamic =
payload type number 96 mapped to telephone-event. Now if a new Re-Invite =
is sent out, can the response have a different payload type number for =
the telephone event?

Thanks
Sundi



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<P><FONT SIZE=3D2 FACE=3D"Arial">I have question related to dynamic =
payloads. If a session is estblished with a dynamic payload type number =
mapped to a particular codec, can the number change for the same codec =
in subsequent offers/answers in the same session? e.g. Lets say a =
session was established with a dynamic payload type number 96 mapped to =
telephone-event. Now if a new Re-Invite is sent out, can the response =
have a different payload type number for the telephone event?</FONT></P>

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

<BR><FONT SIZE=3D2 FACE=3D"Arial">Sundi</FONT>
</P>
<BR>

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I believe Section 8.3.2 of RFC 3264 directly answers your question:

The list of media formats used in the session MAY be changed.  To do
    this, the offerer creates a new media description, with the list of
    media formats in the "m=" line different from the corresponding media
    stream in the previous SDP.  This list MAY include new formats, and
    MAY remove formats present from the previous SDP.  However, in the
    case of RTP, the mapping from a particular dynamic payload type
    number to a particular codec within that media stream MUST NOT change
    for the duration of a session.  For example, if A generates an offer
    with G.711 assigned to dynamic payload type number 46, payload type
    number 46 MUST refer to G.711 from that point forward in any offers
    or answers for that media stream within the session.  However, it is
    acceptable for multiple payload type numbers to be mapped to the same
    codec, so that an updated offer could also use payload type number 72
    for G.711.


This is true for the answerer as well. Thus, in each direction, once 
PT number is bound to a codec, that PT number can not be used for a 
different codec.

-Jonathan R.

Raman, Sundereshwaran (Sundereshwaran) wrote:

> I have question related to dynamic payloads. If a session is estblished 
> with a dynamic payload type number mapped to a particular codec, can the 
> number change for the same codec in subsequent offers/answers in the 
> same session? e.g. Lets say a session was established with a dynamic 
> payload type number 96 mapped to telephone-event. Now if a new Re-Invite 
> is sent out, can the response have a different payload type number for 
> the telephone event?
> 
> Thanks
> Sundi
> 
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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From: "Hearty, John" <John.Hearty@Level3.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>,
        "Raman, Sundereshwaran (Sundereshwaran)" <ramsun@avaya.com>
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The text calls out the special case of disallowing changing the PT in
the case of RTP.  The example was for telephone events.  I suspect you
would risk interop problems if you didn't follow the RTP rule in the
text for telephone events, but technically the example is removing the
96 PT and adding a different PT mapping to telephone events which the
text seems to allow.  The rules are for a particular media stream in the
session.  It may be safer to use a whole new media stream for the new PT
if you really need to do it for some reason.

John Hearty
Level3

> -----Original Message-----
> From: mmusic-admin@ietf.org [mailto:mmusic-admin@ietf.org] On Behalf
Of
> Jonathan Rosenberg
> Sent: Wednesday, August 13, 2003 2:39 PM
> To: Raman, Sundereshwaran (Sundereshwaran)
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] (no subject)
>=20
> I believe Section 8.3.2 of RFC 3264 directly answers your question:
>=20
> The list of media formats used in the session MAY be changed.  To do
>     this, the offerer creates a new media description, with the list
of
>     media formats in the "m=3D" line different from the corresponding
media
>     stream in the previous SDP.  This list MAY include new formats,
and
>     MAY remove formats present from the previous SDP.  However, in the
>     case of RTP, the mapping from a particular dynamic payload type
>     number to a particular codec within that media stream MUST NOT
change
>     for the duration of a session.  For example, if A generates an
offer
>     with G.711 assigned to dynamic payload type number 46, payload
type
>     number 46 MUST refer to G.711 from that point forward in any
offers
>     or answers for that media stream within the session.  However, it
is
>     acceptable for multiple payload type numbers to be mapped to the
same
>     codec, so that an updated offer could also use payload type number
72
>     for G.711.
>=20
>=20
> This is true for the answerer as well. Thus, in each direction, once
> PT number is bound to a codec, that PT number can not be used for a
> different codec.
>=20
> -Jonathan R.
>=20
> Raman, Sundereshwaran (Sundereshwaran) wrote:
>=20
> > I have question related to dynamic payloads. If a session is
estblished
> > with a dynamic payload type number mapped to a particular codec, can
the
> > number change for the same codec in subsequent offers/answers in the
> > same session? e.g. Lets say a session was established with a dynamic
> > payload type number 96 mapped to telephone-event. Now if a new
Re-Invite
> > is sent out, can the response have a different payload type number
for
> > the telephone event?
> >
> > Thanks
> > Sundi
> >
> >
>=20
> --
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Technology Officer                    Parsippany, NJ 07054-2711
> dynamicsoft
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic

_______________________________________________
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Raman, Sundereshwaran (Sundereshwaran) wrote:

> Thanks for your prompt replies. I thought so too after reading the
> RFC 3264. The reason for this question was I came across two
> vendors who were changing the payload type number when a re-invite
> was sent out. Heres the scenario: Session established with 96 as
> the payload type number for telephone-event. I send out a RE-INVITE
> with no SDP in it. The response from the phone contains a different
> payload type number for the telephone event. Are they not supposed
> to send the previous agreed upon payload type number?

This is allowed. Read carefully below - it says that it IS ok for 
multiple numbers to be associated with the same codec. In other words, 
you can use PT 96 for tones, and then later PT 98 for tones. The 
opposite is NOT allowed. That is, payload type 96 cannot be used first 
for tones, and then for g.711.

-Jonathan R.


> -----Original Message----- From: Hearty, John
> [mailto:John.Hearty@Level3.com] Sent: Wednesday, August 13, 2003
> 2:49 PM To: Jonathan Rosenberg; Raman, Sundereshwaran
> (Sundereshwaran) Cc: mmusic@ietf.org Subject: RE: [MMUSIC] (no
> subject)
> 
> 
> The text calls out the special case of disallowing changing the PT
> in the case of RTP.  The example was for telephone events.  I
> suspect you would risk interop problems if you didn't follow the
> RTP rule in the text for telephone events, but technically the
> example is removing the 96 PT and adding a different PT mapping to
> telephone events which the text seems to allow.  The rules are for
> a particular media stream in the session.  It may be safer to use a
> whole new media stream for the new PT if you really need to do it
> for some reason.
> 
> John Hearty Level3
> 
> 
>> -----Original Message----- From: mmusic-admin@ietf.org
>> [mailto:mmusic-admin@ietf.org] On Behalf
> 
> Of
> 
>> Jonathan Rosenberg Sent: Wednesday, August 13, 2003 2:39 PM To:
>> Raman, Sundereshwaran (Sundereshwaran) Cc: mmusic@ietf.org 
>> Subject: Re: [MMUSIC] (no subject)
>> 
>> I believe Section 8.3.2 of RFC 3264 directly answers your
>> question:
>> 
>> The list of media formats used in the session MAY be changed.  To
>> do this, the offerer creates a new media description, with the
>> list
> 
> of
> 
>> media formats in the "m=" line different from the corresponding
> 
> media
> 
>> stream in the previous SDP.  This list MAY include new formats,
> 
> and
> 
>> MAY remove formats present from the previous SDP.  However, in
>> the case of RTP, the mapping from a particular dynamic payload
>> type number to a particular codec within that media stream MUST
>> NOT
> 
> change
> 
>> for the duration of a session.  For example, if A generates an
> 
> offer
> 
>> with G.711 assigned to dynamic payload type number 46, payload
> 
> type
> 
>> number 46 MUST refer to G.711 from that point forward in any
> 
> offers
> 
>> or answers for that media stream within the session.  However, it
>> 
> 
> is
> 
>> acceptable for multiple payload type numbers to be mapped to the
> 
> same
> 
>> codec, so that an updated offer could also use payload type
>> number
> 
> 72
> 
>> for G.711.
>> 
>> 
>> This is true for the answerer as well. Thus, in each direction,
>> once PT number is bound to a codec, that PT number can not be
>> used for a different codec.
>> 
>> -Jonathan R.
>> 
>> Raman, Sundereshwaran (Sundereshwaran) wrote:
>> 
>> 
>>> I have question related to dynamic payloads. If a session is
> 
> estblished
> 
>>> with a dynamic payload type number mapped to a particular
>>> codec, can
> 
> the
> 
>>> number change for the same codec in subsequent offers/answers
>>> in the same session? e.g. Lets say a session was established
>>> with a dynamic payload type number 96 mapped to
>>> telephone-event. Now if a new
> 
> Re-Invite
> 
>>> is sent out, can the response have a different payload type
>>> number
> 
> for
> 
>>> the telephone event?
>>> 
>>> Thanks Sundi
>>> 
>>> 
>> 
>> -- Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza 
>> Chief Technology Officer                    Parsippany, NJ
>> 07054-2711 dynamicsoft jdrosen@dynamicsoft.com
>> FAX:   (973) 952-5050 http://www.jdrosen.net
>> PHONE: (973) 952-5000 http://www.dynamicsoft.com
>> 
>> 
>> _______________________________________________ mmusic mailing
>> list mmusic@ietf.org 
>> https://www1.ietf.org/mailman/listinfo/mmusic
> 
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
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Date: Thu, 14 Aug 2003 11:20:32 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: "Raman, Sundereshwaran (Sundereshwaran)" <ramsun@avaya.com>
CC: "Hearty, John" <John.Hearty@Level3.com>, mmusic@ietf.org
Subject: Re: [MMUSIC] (no subject)
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RFC 3264 is issued and done, and what you are proposing would change 
that behavior. I dont see the benefit of doing that. Changing PT 
numbers is sometimes necessary for synchronization of other changes. 
Normally I wouldn't expect it to change randomly.

-Jonathan R.

Raman, Sundereshwaran (Sundereshwaran) wrote:
> Just a thought. I think its good to be flexible in having multiple
> payload number for the same codec. But telephone-event is a special
> codec type. It will be lot more easier if once a session is
> established with a particular payload number for the
> telephone-event, the same payload number is used for subsequent
> offers/answers. This will particularly be beneficial to the PBX
> world as many PBX features rely on the tones and if it keeps on
> changing then lot more work needs to be done. I would appreciate
> your views on this.
> 
> Thanks Sundi
> 
> -----Original Message----- From: Jonathan Rosenberg
> [mailto:jdrosen@dynamicsoft.com] Sent: Wednesday, August 13, 2003
> 3:20 PM To: Raman, Sundereshwaran (Sundereshwaran) Cc: Hearty,
> John; mmusic@ietf.org Subject: Re: [MMUSIC] (no subject)
> 
> 
> 
> 
> Raman, Sundereshwaran (Sundereshwaran) wrote:
> 
> 
>> Thanks for your prompt replies. I thought so too after reading
>> the RFC 3264. The reason for this question was I came across two 
>> vendors who were changing the payload type number when a
>> re-invite was sent out. Heres the scenario: Session established
>> with 96 as the payload type number for telephone-event. I send
>> out a RE-INVITE with no SDP in it. The response from the phone
>> contains a different payload type number for the telephone event.
>> Are they not supposed to send the previous agreed upon payload
>> type number?
> 
> 
> This is allowed. Read carefully below - it says that it IS ok for 
> multiple numbers to be associated with the same codec. In other
> words, you can use PT 96 for tones, and then later PT 98 for tones.
> The opposite is NOT allowed. That is, payload type 96 cannot be
> used first for tones, and then for g.711.
> 
> -Jonathan R.
> 
> 
> 
>> -----Original Message----- From: Hearty, John 
>> [mailto:John.Hearty@Level3.com] Sent: Wednesday, August 13, 2003 
>> 2:49 PM To: Jonathan Rosenberg; Raman, Sundereshwaran 
>> (Sundereshwaran) Cc: mmusic@ietf.org Subject: RE: [MMUSIC] (no 
>> subject)
>> 
>> 
>> The text calls out the special case of disallowing changing the
>> PT in the case of RTP.  The example was for telephone events.  I 
>> suspect you would risk interop problems if you didn't follow the 
>> RTP rule in the text for telephone events, but technically the 
>> example is removing the 96 PT and adding a different PT mapping
>> to telephone events which the text seems to allow.  The rules are
>> for a particular media stream in the session.  It may be safer to
>> use a whole new media stream for the new PT if you really need to
>> do it for some reason.
>> 
>> John Hearty Level3
>> 
>> 
>> 
>>> -----Original Message----- From: mmusic-admin@ietf.org 
>>> [mailto:mmusic-admin@ietf.org] On Behalf
>> 
>> Of
>> 
>> 
>>> Jonathan Rosenberg Sent: Wednesday, August 13, 2003 2:39 PM To:
>>>  Raman, Sundereshwaran (Sundereshwaran) Cc: mmusic@ietf.org 
>>> Subject: Re: [MMUSIC] (no subject)
>>> 
>>> I believe Section 8.3.2 of RFC 3264 directly answers your 
>>> question:
>>> 
>>> The list of media formats used in the session MAY be changed.
>>> To do this, the offerer creates a new media description, with
>>> the list
>> 
>> of
>> 
>> 
>>> media formats in the "m=" line different from the corresponding
>>> 
>> 
>> media
>> 
>> 
>>> stream in the previous SDP.  This list MAY include new formats,
>>> 
>> 
>> and
>> 
>> 
>>> MAY remove formats present from the previous SDP.  However, in 
>>> the case of RTP, the mapping from a particular dynamic payload 
>>> type number to a particular codec within that media stream MUST
>>>  NOT
>> 
>> change
>> 
>> 
>>> for the duration of a session.  For example, if A generates an
>> 
>> offer
>> 
>> 
>>> with G.711 assigned to dynamic payload type number 46, payload
>> 
>> type
>> 
>> 
>>> number 46 MUST refer to G.711 from that point forward in any
>> 
>> offers
>> 
>> 
>>> or answers for that media stream within the session.  However,
>>> it
>>> 
>> 
>> is
>> 
>> 
>>> acceptable for multiple payload type numbers to be mapped to
>>> the
>> 
>> same
>> 
>> 
>>> codec, so that an updated offer could also use payload type 
>>> number
>> 
>> 72
>> 
>> 
>>> for G.711.
>>> 
>>> 
>>> This is true for the answerer as well. Thus, in each direction,
>>>  once PT number is bound to a codec, that PT number can not be 
>>> used for a different codec.
>>> 
>>> -Jonathan R.
>>> 
>>> Raman, Sundereshwaran (Sundereshwaran) wrote:
>>> 
>>> 
>>> 
>>>> I have question related to dynamic payloads. If a session is
>> 
>> estblished
>> 
>> 
>>>> with a dynamic payload type number mapped to a particular 
>>>> codec, can
>> 
>> the
>> 
>> 
>>>> number change for the same codec in subsequent offers/answers
>>>>  in the same session? e.g. Lets say a session was established
>>>>  with a dynamic payload type number 96 mapped to 
>>>> telephone-event. Now if a new
>> 
>> Re-Invite
>> 
>> 
>>>> is sent out, can the response have a different payload type 
>>>> number
>> 
>> for
>> 
>> 
>>>> the telephone event?
>>>> 
>>>> Thanks Sundi
>>>> 
>>>> 
>>> 
>>> -- Jonathan D. Rosenberg, Ph.D.                600 Lanidex
>>> Plaza Chief Technology Officer                    Parsippany,
>>> NJ 07054-2711 dynamicsoft jdrosen@dynamicsoft.com FAX:   (973)
>>> 952-5050 http://www.jdrosen.net PHONE: (973) 952-5000
>>> http://www.dynamicsoft.com
>>> 
>>> 
>>> _______________________________________________ mmusic mailing 
>>> list mmusic@ietf.org 
>>> https://www1.ietf.org/mailman/listinfo/mmusic
>> 
>> 
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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Hi Colin,

and thanks for the comments.

> 
> I have a number of comments on this draft:
> 
>   - The ABNF in Sections 2.1 and 2.2 is contradictory. The key management
>     data is defined as "text" in one section, and "quoted-string" in the
>     other, yet these two types have different ranges of valid characters
>     (and both are more liberal than the explanatory text, which implies
>     that only base-64 encoded data is allowed). Similar for the protocol
>     identifier: the definitions of "non-ws-string" in SDPnew and "token"
>     in RTSP are not consistent.

This is as a result from the comments we received before, when we actually 
had consistent strings. At that time the comment was that we should use 
what was used in SDP and RTSP respectively (and the recommendations was 
to use non-ws-string and token). Please propose what to use.

 
>   - It should be stated that the protocol identifiers in Section 2.1 and
>     Section 2.2 refer to the same set of values, and a reference to the
>     registry of legal values should be given (i.e. reference the IANA
>     parameter registry defined in Section 6).

Ok.

> 
>   - The draft repeatedly mentions Kerberos, and indicates that Kerberos
>     tickets may be included in key management messages, but does not
>     specify how this is to be achieved. Should either specify how to use
>     Kerberos, or remove all references to it.

It was included as a result of a comment that we should give examples 
of more than one protocol that might utilize the framework. It is not 
our intention to specify how it can be used. We can agree that there are 
places where we could remove the reference. However, it might be good 
at least to keep the most general one in Section 3 to provide an overview
of the possibilities.  
 
>   - The second bullet point on Page 6 mentions that other parameters may
>     need to be specified for certain key management protocols, but does
>     not specify how an implementor knows if a particular key management
>     protocol uses additional parameters. Suggest this is listed as part
>     of the IANA registration for a particular key management protocol?

I don't see why IANA should handle this. However, it sounds reasonable 
that it should be documented in the key management draft (or a companion 
draft).
 
>   - Section 3.1 seems to assume Offer/Answer usage. It is not clear how
>     it applies to SDP used with, for example, SAP, HTTP, or email? What
>     about one-to-many distribution of SDP? Or distributions where there
>     is no back-channel? It is acceptable for the draft to be explicit
>     about the limits of its scope, but it cannot ignore the wider problem
>     space.

Ok, clarifications will be added. 
 
>   - The discussion of authentication on Page 7 is not mentioned in the
>     list of things senders and receivers of SDP must do, on pages 5 and
>     6.

Ok, we can mention it there as well.
 
>   - Section 3.3 does not mention authentication. Should I assume that
>     there is no need for authentication when key management is used with
>     RTSP?

Also in the RTSP case, the protocol name authentication described in 
section 3.1 should be applied. We will clarify this by explicitly 
state it. 
 
>   - Section 4 needs to provide strict guidance about the parameters and
>     specifications needed for new key management protocols. The interface
>     between the key management protocol and this framework should be made
>     explicit.

That is what we have tried to address. Please give examples of what
you think is lacking of that?

 
>   - I find the arguments about security of the SDP in Section 5 weak. The
>     draft needs to be explicit about the threats it prevents and those it
>     leaves open. Recommend re-reading RFC 3553, Section 5.

Ok.
 
>   - In section 5, how is a beat-down attack prevented when the attacker
>     wishes to beat-down to no security? What is the correct response to
>     an answer which suggests no security?

That would be to terminate/abort the session setup. 
 
>   - Section 6 needs to register the RTSP extensions needed, and make it
>     clear that the namespace for key management protocol identifiers is
>     shared between the RTSP and SDP extensions.

Ok.
 
>   - Section 6 mentions that new key management protocols specify "the
>     syntax of possible additional arguments", yet this contradicts the
>     ABNF which states that only a single argument is present.

Ok, that was a bad formulation. We'll fix that. 

> 
> The main issue behind these seems to be the desire to define new 
> parameters which can be left open for future uses. As was mentioned in 
> the context of the "a=gpmd:" attribute, this is not acceptable: there 
> needs to be a precise definition of the semantics of any new SDP 
> attribute/RTSP header before the draft can advance.

Ok, but doesn't fixing the above issues make this problem go away?


Fredrik

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

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Date: Fri, 15 Aug 2003 09:59:37 +0200
From: Joerg Ott <jo@tzi.uni-bremen.de>
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Subject: [MMUSIC] Final MMUSIC Minutes from 57th IETF
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Folks,

since we have not received any comments, we consider the minutes
final.

The slides are available at

http://www.dmn.tzi.org/ietf/mmusic/57/slides/57-mmusic-slides-ppt.zip
http://www.dmn.tzi.org/ietf/mmusic/57/slides/57-mmusic-slides-pdf.zip

Joerg

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

Minutes of the MMUSIC WG Meeting at the 57th IETF
=================================================

Reported by Joerg Ott and Colin Perkins
Notes taken by Colin Perkins

The MMUSIC WG met once at the 57th IETF.  The meeting was chaired by
Colin Perkins and Joerg Ott.  It was attended by some 90 participants.

Status Update
-------------

Joerg Ott reported on the status of the working and the general
progress since the 56th IETF in San Francisco:

- The new charter was finally approved.

- RFC 3524 was published on SDP extensions for reservation flows.

- Two documents are with the RFC editor: SDP extensions for NAT
    traversal and SDPng transition.

- Two documents are with the IESG, and comments have been received:

    - draft-rajeshkumar-mmusic-gpmd-03.txt was felt to overlap with work
      on capability negotiation carried out elsewhere; it was noted that
      this probably applies to most of SDP except that SDP was first.
      The chairs and authors will work with the IESG to sort this out.
    - draft-ietf-mmusic-sdp-new-13.txt received many comments which were
      discussed in a separate slot (see below).

- Two documents are under AD review:
    - draft-ietf-mmusic-sdp-srcfilter-05.txt
    - draft-ietf-mmusic-sdp-comedia-05.txt

- draft-ietf-mmusic-sdp-bwparam-04.txt is completing WGLC on 28 Jul
    2003.

- The SDP extensions for key management
    (draft-ietf-mmusic-kmgmt-ext-07.txt) have been waiting for the MIKEY
    work to complete which seems to progress.  A further revision is
    expected shortly after the IETF.

- The SDP implementation draft draft-ietf-mmusic-sdp-implem-00.txt has
    not been revised yet, but Tom Taylor is working on it.  He is trying
    to incorporate further implementation experience from inside his
    company as well as from the SIPit events.

Two potential new work items have been discussed for MMUSIC:

- Real-time fax (draft-ietf-sipping-realtimefax-01.txt) from the
    SIPPING WG is to provide implementers with guidelines on how to use
    registered SDP attributes for ITU-T T.38.  This document actually
    contains two logical parts: the use of offer/answer mechanics (which
    belongs to MMUSIC) and a discussion of SIP-specific signaling (which
    belongs to SIPPING).  It is suggested to investigate splitting the
    document to deal with each one in the respective WG.  But it was
    pointed out that this document merely contains examples only and
    hence should be done as a whole in just one WG.  To be resolved.

- Interactive connectivity establishment (ICE) also from the SIPPING WG
    (draft-rosenberg-sipping-ice-01.txt).  This is discussed later on
    the agenda.

Revised SDP (draft-ietf-mmusic-sdp-new-13.txt)
----------------------------------------------

Colin Perkins reported on the changes incorporated in the latest
revision of SDP: It was attempted to clarify when the k= attribute may
be used.  Further clarifications include: RTP may be used with
non-consecutive port numbers and a=charset is only allowed as a
session level attribute.  Also, unregistered X- attributes and media
format are deprecated.  Finally, IANA considerations and references
were updated.

Colin further reported on issues raised on revision -13 on the mailing
list, as well as from the IESG review.

- An ABNF inconsistency was found: a comment in the ABNF for
    "token-char" allowed for a larger set of characters than the actual
    definition.  One could either (1) change the comment to match the
    actual grammar (and thus reduce the set of allowed characters) or
    (2) modify the grammar to match the comment.  The group agreed to go
    with option (1), pending a check against the use of SDP in RFC 3108.
    Tom Taylor volunteered to find this out.

- The use of b= with layered coding should be clarified as follows:
    When using the CT modifier with RTP, if several RTP sessions are
    part of the conference, the conference total refers to total
    bandwidth of all RTP sessions.  This was accepted in principle but
    it was also pointed out that a bandwidth specification per layer
    would be helpful.  Further discussion is needed on the mailing
    list.

- Again with layered coding, it should be clarified which ports are
    associated with an m= line.  The clarification proposal was
    accepted.

- The k= is underspecified.  The most straightforward way would be
    to deprecate the use of this field altogether.  Opinions from
    implementers are sought.  It was reported that MS Messenger uses
    this field; the details are unknown at this moment though.  As there
    are implementation out there, just deprecating this field won't work.

    Therefore, further work needs to go into the specification of using
    k=.  In particular, when k= shall be used, a secure channel must be
    available and endpoints need to be sure that no third party can
    eavesdrop on the key.  These issues, however, need to be specified
    by those using SDP rather than SDP itself.

    The k= field currently supports an algorithm indicator, but it should
    also specify a key type.  This may be implied by the URI or the
    media protocol used.  For more complex scenarios, e.g. when more
    than one key is needed, either the sdescriptions work
    (draft-ietf-mmusic-sdescriptions-01.txt) or the key management
    (draft-ietf-mmusic-kmgmt-ext-07.txt) work should be used.

    It was proposed that the security considerations section should
    change "SHOULD NOT deliver the user into an interactive multimedia
    session..." to "MUST NOT...".  It was found that this is an issue of
    user authorization (e.g. the user may be able to pre-authorize the
    automatic joining of certain interactive sessions) and needs to be
    dealt with by the protocols using SDP.

- The suggestion to define a "IN" value for private address spaces was
    rejected because systems 1) cannot tell whether they are in a
    private address space (relative to their peers) and 2) this change
    would be not backward compatible.

- It will be clarified that UTF-8 normalization is not needed because
    comparisons are not done on fields with UTF-8.

- Proposal on a=inactive: it was agreed not to change the SDP spec here
    (this is an issue for users of SDP, which may establish conventions
    for use, not for SDP itself).

- Wording should be added that u= and k= may include URIs that may be
    dereferenced in some (but not all) cases.  Also, it should be noted
    a k= URI may typically be an http or https URI.  Both suggested
    wording additions were accepted.

Colin will fold the comments and the discussion into the document and
produce a new revision out within a couple of weeks.


SDP Security Descriptions (draft-ietf-mmusic-sdescriptions-01.txt)
------------------------------------------------------------------

Dan Wing briefly reviewed the document and the changes from the
previous revision: a new syntax is proposed for the a=crypto attribute
and the offer/answer part is revised which may need further review.
For the next steps, Colin noted that an IANA considerations section is
needed, including a discussion of the registration procedures.
Furthermore, reviewers for finishing the document were sought: Tom
Taylor, Magnus Westerlund, and Martin Euchner volunteered.


Offer/Answer Examples (draft-johnston-mmusic-offer-answer-examples-01.txt)
--------------------------------------------------------------------------

Alan Johnston presented the changes to the document: in particular, an
example for asymmetric use of a codec has been added.  This is to be
achieved using two distinct m= lines with the same port number, one
indicating "sendonly", the other "recvonly".  The proposed behavior
caused some discussion: it was felt that the mechanism was
underspecified and that (therefore) m= lines with the same port would
be a bad idea.  Yet, the right approach to achieve the desired
behavior seemed unclear.  Gonzalo Camarillo pointed out that the
RFC3388 provided a similar example, but with an explicit grouping
(which is not present in this draft).  The applicability of RFC3388 to
this problem was discussed further.  Orit Levin pointed out that the
intention clearly is to use RFC 3388 but the question is what happens
to the endpoints that don't support RFC 3388.  Gonzalo argued that
this is already defined in RFC 3388.  The chairs stated that they
would like to see the asymmetric codec behavior with two m= line
properly specified in a document.  The present offer/answer examples
draft is meant to be illustrative; the example should be kept in place
(with added grouping) but the normative text for this behavior needs
to be someplace else.  Flemming Andreasen suggested to specify part of
this practice in the revised SDP.

SDPng Update (draft-ietf-mmusic-sdpng-06.txt)
---------------------------------------------

Dirk Kutscher presented the latest progress on SDPng, addressing the
open issues from draft -06 (a new revision still needs to be
submitted).  The SDPng structure has been enhanced to decouple
capability negotiation for potential configurations from actual
configuration (and the respective session parameters).  While the
capability model has been kept, the syntax has been revised to
exclusively rely on XML mechanisms (rather than embedding own syntax
conventions for attributes) -- which is less efficient but much
cleaner.  Dirk presented the new syntax using several examples and
also showed a formal schema definition.  A further refinement from
draft -06 is to remove the concept of libraries and use inline
definitions instead -- so that SDPng documents would be self-contained.
There was little discussion.  Steve Casner asked whether this work was
getting enough traction and questioned whether people were really
preparing for a transition.  Joerg Ott argued that people are
desparately holding onto SDP (which is perceived easier with one small
extension at a time) -- but that examples show that this getting more
complicated so that SDP is likely to have only a finite lifetime.  So
it would be nice to get broader review from the community on a
replacements for SDP.


RTP DoS Attacks (draft-rosenberg-mmusic-rtp-denialofservice-00.txt)
Interactive Connectivity Establishment (draft-rosenberg-sipping-ice-01.txt)
---------------------------------------------------------------------------

Jonathan Rosenberg introduced the RTP DoS problem: An attacker may
easily direct the RTP stream(s) from an unsuspecting source to any
target by providing the target's IP address in the SDP.  This works
for SIP and RTSP -- with RTSP providing limited protection by allowing
unicast RTP streams only to the address of the originator (but NATs
make it worse as anyone behind the same NAT can easily be attacked).
As authentication mechanisms can at most help identify the attacker
(which may be very weak though), some kind of handshake mechanism is
needed so that sources do not start sending media streams unless they
know that the right receiver is actually waiting for it: a
request/response mechanism using the RTP ports is needed.  This is ICE
(see below) which has been chosen as a solution for SIP.  However, ICE
has broader applicability and may be appropriate for RTSP, too --
which needs to be determined by the group.  Jonathan reviewed the
properties of the ICE approach and briefly walked through the basic
ICE algorithm.  With ICE being signaling protocol independent, MMUSIC
is asked to take ICE up as a WG item.

ICE is felt to introduce quite some complexity for RTSP endpoints.
If, however, NAT traversal (which is an RTSP issue on its own, see
below) and security can be addressed by the same mechanism, it may be
worth the effort.  It is noted that even if we go for the ICE
mechanism for RTSP right now, this does not solve the problem as long
as old RTSP servers are still out there.

Regarding the ICE proposal as a WG item Magnus Westerlund argued that
if it is accepted, one should split the draft into several parts: the
SIP use, the ICE mechanism, the use with offer/answer, etc.
There was consensus in the group to accept ICE as new WG item, and to
fully document the RTP DoS issues as an informational document.


RTSP Update (draft-ietf-mmusic-rc2326bis-04.txt
-----------------------------------------------

Magnus Westerlund reported on the progress of the RTSP revision,
achieved by having a new co-editor joining the team, many
teleconference calls, and also quite some list discussion.  He
presented a number of changes from draft -03 (see the slides).  With
this, most known open issues are resolved, with only a few ones
remaining (see slides).  But many resolutions still need to be folded
into an updated revision of the draft.  When this is done, a second
phase of editing on the spec is needed: examples need to be fixed, the
spec's consistency needs to be checked as needs it overall
readability, and, potential new open issues need to be solved.  With
all this, the revised RTSP spec will not meet the October deadline as
currently documented in the MMUSIC charter.  Magnus suggests another
six months would be appropriate to finalize the work.


RTSP and NAT (draft-ietf-mmusic-rtsp-nat-01.txt)
------------------------------------------------

Magnus Westerlund introduced the revised draft on RTSP and NATs, the
purpose of which is to describe how to traverse NATs and firewalls
with RTSP.  It documents several possible NAT traversal approaches and
gives recommendations regarding RTSP for firewalls.  Approaches
requiring only modifications on the client side STUN, TURN, and
RTP/RTSP tunneled in RTSP.  Application-Layer Gateways (ALGs) for RTSP
will not require any modifications -- but are generally discouraged.
Magnus further discussed open issues, particurlarly the goals for RTSP
for NATs (i.e. what type of scenarios should be achievable) as well as
possible solutions (symmetric RTP, STUN, ICE, DCCP).

On the ALGs: The group generally agreed that ALGs are a bad thing and
should not be considered further (except for documentation purposes and
to communicate this clearly to outside the IETF -- and fast, as people
start deploying these).

On the document's purpose: Commenters state that, currently, the draft
does not give conclusive resolution of what to do as an implementer,
it is merely an inventory.  It is proposed to pick some solution and
write it up, e.g. in the RTSP spec.  Magnus wants to resolve the
structure of the document and then settle on a solution: one for the
server and one or more for the client side.  Jonathan Rosenberg and
Tom Marshal argue over the number of client-side solutions to be
documented: Jonathan wants only one while Tom prefers to have multiple
feasible solutions document.  In case multiple ones are documented,
Jonathan wants exactly one "MUST implement" solution to ensure
interoperability.

There was some further discussion on technical aspects of possible
solutions, but no real conclusions were reached.


Internet Media Guides (IMG)
---------------------------
draft-nomura-mmusic-img-requirements-01.txt
draft-nomura-mmusic-img-framework-01.txt

After a brief introduction by Joerg Ott, Rod Walsh presented the
progress on the IMG requirements document (which will be re-submitted
as WG draft by end of August).  He briefly reviewed the general
topology and IMG distribution aspects and then went through the
requirement categories identified in the draft: IMG description
requirements and IMG transport requirements.  The former primarily
address IMG envelopes for existing meta-data formats, the latter their
distribution in unicast and multicast, pull and push style scenarios.
For multicast push, SAPv2 is probably inadequate, so a successor is
sought, e.g. ALC from the RMT WG.  It is pointed that additions to ALC
need to be specified separately; while Rod wants to use the RMT
proposal FLUTE, Lorenzo argues that this may not be 100% appropriate
here; Lorenzo also points out that possible tasks in the RMT WG need
to happen quickly as the group is trying to wrap up.  For unicasting,
client-initiated actions could use HTTP while notifications from the
server could use SIP mechanims.

Rod discusses on the scalability requirements IMG transport and
presents a conceptual overview showing the different pieces of an IMG
framework and outlines the achieved flexibility.  He also addresses
data relationships and particularly security considerations as taken
into account in the requirements draft.

Rod finally discusses the way ahead: the requirements draft is pretty
close to finalization; the remaining efforts are mainly editorial and
consolidation.  The aim is to go for WGLC soon.  He also points to the
existing framework draft that will be revised by the end of August.






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



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Thanks for your prompt replies. I thought so too after reading the RFC =
3264.
The reason for this question was I came across two vendors who were =
changing the payload type number when a re-invite was sent out. Heres =
the scenario:
Session established with 96 as the payload type number for =
telephone-event.
I send out a RE-INVITE with no SDP in it. The response from the phone =
contains a different payload type number for the telephone event. Are =
they not supposed to send the previous agreed upon payload type number?  =


Thanks
Sundi

-----Original Message-----
From: Hearty, John [mailto:John.Hearty@Level3.com]
Sent: Wednesday, August 13, 2003 2:49 PM
To: Jonathan Rosenberg; Raman, Sundereshwaran (Sundereshwaran)
Cc: mmusic@ietf.org
Subject: RE: [MMUSIC] (no subject)


The text calls out the special case of disallowing changing the PT in
the case of RTP.  The example was for telephone events.  I suspect you
would risk interop problems if you didn't follow the RTP rule in the
text for telephone events, but technically the example is removing the
96 PT and adding a different PT mapping to telephone events which the
text seems to allow.  The rules are for a particular media stream in the
session.  It may be safer to use a whole new media stream for the new PT
if you really need to do it for some reason.

John Hearty
Level3

> -----Original Message-----
> From: mmusic-admin@ietf.org [mailto:mmusic-admin@ietf.org] On Behalf
Of
> Jonathan Rosenberg
> Sent: Wednesday, August 13, 2003 2:39 PM
> To: Raman, Sundereshwaran (Sundereshwaran)
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] (no subject)
>=20
> I believe Section 8.3.2 of RFC 3264 directly answers your question:
>=20
> The list of media formats used in the session MAY be changed.  To do
>     this, the offerer creates a new media description, with the list
of
>     media formats in the "m=3D" line different from the corresponding
media
>     stream in the previous SDP.  This list MAY include new formats,
and
>     MAY remove formats present from the previous SDP.  However, in the
>     case of RTP, the mapping from a particular dynamic payload type
>     number to a particular codec within that media stream MUST NOT
change
>     for the duration of a session.  For example, if A generates an
offer
>     with G.711 assigned to dynamic payload type number 46, payload
type
>     number 46 MUST refer to G.711 from that point forward in any
offers
>     or answers for that media stream within the session.  However, it
is
>     acceptable for multiple payload type numbers to be mapped to the
same
>     codec, so that an updated offer could also use payload type number
72
>     for G.711.
>=20
>=20
> This is true for the answerer as well. Thus, in each direction, once
> PT number is bound to a codec, that PT number can not be used for a
> different codec.
>=20
> -Jonathan R.
>=20
> Raman, Sundereshwaran (Sundereshwaran) wrote:
>=20
> > I have question related to dynamic payloads. If a session is
estblished
> > with a dynamic payload type number mapped to a particular codec, can
the
> > number change for the same codec in subsequent offers/answers in the
> > same session? e.g. Lets say a session was established with a dynamic
> > payload type number 96 mapped to telephone-event. Now if a new
Re-Invite
> > is sent out, can the response have a different payload type number
for
> > the telephone event?
> >
> > Thanks
> > Sundi
> >
> >
>=20
> --
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Technology Officer                    Parsippany, NJ 07054-2711
> dynamicsoft
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic

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Subject: RE: [MMUSIC] (no subject)
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From: "Raman, Sundereshwaran (Sundereshwaran)" <ramsun@avaya.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
Cc: "Hearty, John" <John.Hearty@Level3.com>, <mmusic@ietf.org>
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Just a thought. I think its good to be flexible in having multiple =
payload number for the same codec. But telephone-event is a special =
codec type. It will be lot more easier if once a session is established =
with a particular payload number for the telephone-event, the same =
payload number is used for subsequent offers/answers. This will =
particularly be beneficial to the PBX world as many PBX features rely on =
the tones and if it keeps on changing then lot more work needs to be =
done. I would appreciate your views on this.

Thanks
Sundi

-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent: Wednesday, August 13, 2003 3:20 PM
To: Raman, Sundereshwaran (Sundereshwaran)
Cc: Hearty, John; mmusic@ietf.org
Subject: Re: [MMUSIC] (no subject)




Raman, Sundereshwaran (Sundereshwaran) wrote:

> Thanks for your prompt replies. I thought so too after reading the
> RFC 3264. The reason for this question was I came across two
> vendors who were changing the payload type number when a re-invite
> was sent out. Heres the scenario: Session established with 96 as
> the payload type number for telephone-event. I send out a RE-INVITE
> with no SDP in it. The response from the phone contains a different
> payload type number for the telephone event. Are they not supposed
> to send the previous agreed upon payload type number?

This is allowed. Read carefully below - it says that it IS ok for=20
multiple numbers to be associated with the same codec. In other words,=20
you can use PT 96 for tones, and then later PT 98 for tones. The=20
opposite is NOT allowed. That is, payload type 96 cannot be used first=20
for tones, and then for g.711.

-Jonathan R.


> -----Original Message----- From: Hearty, John
> [mailto:John.Hearty@Level3.com] Sent: Wednesday, August 13, 2003
> 2:49 PM To: Jonathan Rosenberg; Raman, Sundereshwaran
> (Sundereshwaran) Cc: mmusic@ietf.org Subject: RE: [MMUSIC] (no
> subject)
>=20
>=20
> The text calls out the special case of disallowing changing the PT
> in the case of RTP.  The example was for telephone events.  I
> suspect you would risk interop problems if you didn't follow the
> RTP rule in the text for telephone events, but technically the
> example is removing the 96 PT and adding a different PT mapping to
> telephone events which the text seems to allow.  The rules are for
> a particular media stream in the session.  It may be safer to use a
> whole new media stream for the new PT if you really need to do it
> for some reason.
>=20
> John Hearty Level3
>=20
>=20
>> -----Original Message----- From: mmusic-admin@ietf.org
>> [mailto:mmusic-admin@ietf.org] On Behalf
>=20
> Of
>=20
>> Jonathan Rosenberg Sent: Wednesday, August 13, 2003 2:39 PM To:
>> Raman, Sundereshwaran (Sundereshwaran) Cc: mmusic@ietf.org=20
>> Subject: Re: [MMUSIC] (no subject)
>>=20
>> I believe Section 8.3.2 of RFC 3264 directly answers your
>> question:
>>=20
>> The list of media formats used in the session MAY be changed.  To
>> do this, the offerer creates a new media description, with the
>> list
>=20
> of
>=20
>> media formats in the "m=3D" line different from the corresponding
>=20
> media
>=20
>> stream in the previous SDP.  This list MAY include new formats,
>=20
> and
>=20
>> MAY remove formats present from the previous SDP.  However, in
>> the case of RTP, the mapping from a particular dynamic payload
>> type number to a particular codec within that media stream MUST
>> NOT
>=20
> change
>=20
>> for the duration of a session.  For example, if A generates an
>=20
> offer
>=20
>> with G.711 assigned to dynamic payload type number 46, payload
>=20
> type
>=20
>> number 46 MUST refer to G.711 from that point forward in any
>=20
> offers
>=20
>> or answers for that media stream within the session.  However, it
>>=20
>=20
> is
>=20
>> acceptable for multiple payload type numbers to be mapped to the
>=20
> same
>=20
>> codec, so that an updated offer could also use payload type
>> number
>=20
> 72
>=20
>> for G.711.
>>=20
>>=20
>> This is true for the answerer as well. Thus, in each direction,
>> once PT number is bound to a codec, that PT number can not be
>> used for a different codec.
>>=20
>> -Jonathan R.
>>=20
>> Raman, Sundereshwaran (Sundereshwaran) wrote:
>>=20
>>=20
>>> I have question related to dynamic payloads. If a session is
>=20
> estblished
>=20
>>> with a dynamic payload type number mapped to a particular
>>> codec, can
>=20
> the
>=20
>>> number change for the same codec in subsequent offers/answers
>>> in the same session? e.g. Lets say a session was established
>>> with a dynamic payload type number 96 mapped to
>>> telephone-event. Now if a new
>=20
> Re-Invite
>=20
>>> is sent out, can the response have a different payload type
>>> number
>=20
> for
>=20
>>> the telephone event?
>>>=20
>>> Thanks Sundi
>>>=20
>>>=20
>>=20
>> -- Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza=20
>> Chief Technology Officer                    Parsippany, NJ
>> 07054-2711 dynamicsoft jdrosen@dynamicsoft.com
>> FAX:   (973) 952-5050 http://www.jdrosen.net
>> PHONE: (973) 952-5000 http://www.dynamicsoft.com
>>=20
>>=20
>> _______________________________________________ mmusic mailing
>> list mmusic@ietf.org=20
>> https://www1.ietf.org/mailman/listinfo/mmusic
>=20
>=20

--=20
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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Magnus, Dan
   Thanks to Magnus for the careful review.  Pardon my late response but we 
are now about ready to revise the I-D and can benefit from an exchange on 
these issues.   I have a few comments below.

At 04:21 PM 7/30/2003 -0700, Dan Wing wrote:
> > Here is my comments on this draft.
>
>Thanks for your comments.  Sorry for the delay, I have been out of
>the office.
>
> > 1. As a WG draft you should have the short copyright statement and not
> > only the long one at the end of the draft.
>
>Yes, sorry we didn't correct this, I see you commented on this same
>issue when you reviewed -00.
>
> > 2. In section 1 I would like that you clarify that RFC 2828 contains
> > security terminology.
>
>Okay.
>
> > 3. Section 3.1. This section needs to be improved. As I understand it
> > the a=crypto attribute can be used with any transport and media.
> > Therefore the section should not start with defining that this defines
> > the value for RTP/SAVP. The actual crypto suits will be required to be
> > registered and defined for each individual transport. There might in
> > fact be even further restrictions that apply to future crypto suits.
>
>-sdescriptions- is only trying to describe a=crypto for the RTP/SAVP
>(SRTP) transport.

Yes and this is a shift from draft 0.  Here is our thought process:  We 
will not be implementing sdescriptions for anything but the RTP/SAVP (we 
never planned to specify security descriptions for media, which is 
something that I'd expect to see in the fmtp).  I think we'd be making a 
mistake to specify something that we don't plan to implement and the result 
will likely have a lot of errors, which will likely only be discovered when 
someone tries to implement it.  This is what I think happened with k=.  If 
k= was implemented, it was too narrow and the result is that it needs to be 
replaced.

That being said, I believe this framework is general enough to be used by 
other SDP transports.  But we really are only providing the definitions for 
RTP/SAVP.


> > 4. Section 3.4, I think the session parameters are actually specific to
> > the crypto suit, primarily.
>
>But "encrypted_srtcp" can be specified for any and all crypto suites.  Are
>there session parameters that are specific to a cryptographic suite?

Yes, session parameters are distinct from key parameters.  The crypto-suite 
are specific to the key parameters and independent from the session parameters.


> > Secondarily it can be dependent on more things.
>
>Could you explain?
>
> > 5. Section 3 and 4. In general I think the basic definition of the
> > a=crypto attribute can be more cleanly separated from the specific
> > application for SRTP.
>
>You mean, for example, to allow -sdescriptions- to define the keys
>used by some non-SRTP application (whiteboarding, perhaps)?  We're
>trying to only define usage with SRTP (RTP/SAVP).

I think we'll be more future-proof if we just face the fact that encryption 
and authentication transforms can often be specialized to the 
transport.  That is what happened to counter mode in IPsec versus 
SRTP.  TBS, I don't see any reason why a specification for a specific 
transport other than RTP/SAVP cannot reference an RTP/SAVP 
crypto-suite.  This will give us the benefit of reuse without trying to 
force a single definition on all transports.


> > 6. Section 4.1, Is usage of multiple a=crypto lines allowed. IF yes this
> > section contradict this.
>
>I agree some different wording would clarify this section.  Will work
>on it.
>
> > 7. Section 4.1.4, last sentence: How can one add a new name that is
> > identical and still carry any difference. Please reword.
>
>Okay.
>
> > 8. Section 4.2, I think there is a large shortcoming regarding the uri:
> > parameter. I think one must clearly define the format one retrieves the
> > keys when using the URI. Of course this format can be crypto suit
> > dependent.
>
>We're strongly considering dropping the URI parameter.  If anyone
>feels strongly about keeping it, speak up.

We are discussing this.  I imagine that the URI will take a person to a web 
page for the purposes of authorization.  After that, the key can be 
obtained by application-specific means, e.g. it could be sent in email.  Or 
as Magnus has pointed out, the format can be crypto-suite dependent and the 
format of the key can be as described in the sdescriptions 
draft.  Personally, I don't think we need to define this further but leave 
it to an application.  It's more flexible that way.


> > 9. Section 4.3.1, second and third paragraph. I think the definition of
> > the SSRC, ROC, SEQ, could be better done. As I understand it any of the
> > three is sufficient, why not express it that way. Then one MAY have the
> > parameter anyway.
>
>We'll tighten up that wording, thanks.

I don't think any one of the three are always sufficient.  The fact that we 
need to intrude into RTP processing in this way is a drawback to SRTP.  The 
SSRC is needed to identify the crypto context prior to the start of the RTP 
session.  In a pair-wise session, or if there is only a single sender, the 
SSRC might not be needed.  In general, it will be.  The ROC and SEQ are 
needed to generate the SRTP keystream for AES-CM and f8.  If the receiver 
does not join the session after session commencement, the ROC and SEQ might 
not be needed.  But because the SEQ starts at a random value in RTP, we 
need to account for boundary cases where the SEQ starts very close to its 
maximum value.  All of this is very messy.  But in a pairwise session of 
short duration, where the SEQ starts at zero, the SEQ and ROC do not need 
to be signaled.


> > 10. Section 5.4, second paragraph: I think the limitation you would like
> > to declare here needs some discussion. First you forbid the usage of
> > multiple m= lines on the same port, after last weeks discussion I don't
> > know if this is acceptable, at least not without further motivation. I
> > would also write some recommendations/considerations regarding grouping
> >   of media lines here.
>
>In this I-D, we will be removing all references to multiple media lines
>with the same port.  A completely separate, unrelated, Internet Draft
>needs to be written -- with consensus from the WG -- describing how multiple
>media lines with the same UDP port are supposed to be interpreted.
>
>(As an aside, the behavior for multiple media lines with the same port
>hasn't been specified elsewhere, so it is ambiguous.  I personally feel we
>need to create an unambigiuos method to express the desired behavior when
>an Offer contains multiple media lines with the same port.)
>
> > 11. Section 5.7, I am not certain that session parameters can't be
> > restrained from being changed. For example the SRC (ROC) parameter seems
> > one that might need to be included and changed in an offer for rekeying.
>
>Good point.  We haven't put enough thought into rekeying yet as it's
>unnecessary with voice applications and the maximum key lifetimes of
>the crypto suites defined in SRTP.
>
> >   Are there any synchronization problem that can arise when doing
> > rekeying? There seems to be need to define from what seq the new key are
> > valid.
>
>That seems to be more a problem with SRTP than -sdescriiptions-.
>
>To resolve the synchronization problem you have to use a per-packet MKI
>(which adds a few bytes to each SRTP packet) or the "from,to" mechanism
>(which doesn't add bytes to each SRTP packet); both MKI and "from,to"
>are described in SRTP itself.
>
>Do you feel putting like the above paragraph might be useful in
>-sdescriptions-?

I'd prefer that we just point to the SRTP specification.


> > 12. When it comes to SRTPs protection mechanism it seems necessary to be
> > able to define the window size using the a=crypto parameter. There are a
> > lot of difference between a uncompressed video stream running at a
> > couple of hundreds of Mbyte/s using 100 000 packets each second. This
> > will require a much larger window then a 64 kbit stream that sends 15
> > packets every second.
>
>SRTP (draft-ietf-avt-09.txt) says that the SRTP-WINDOW-SIZE is a
>receiver-side, implementation-dependent parameter.  If it is signaled
>in a unicast SDP, SRTP-WINDOW-SIZE would become a sender-side parameter.
>
>Is that your intent of having the SRTP-WINDOW-SIZE indicated in the
>a=crypto line?
>
> > 13. Section 6.1 First paragraph, last sentence. The attack is only
> > avoided if there is one master key per SSRC, not media stream. I think
> > that there is some ambiguity in the use of media stream, which needs to
> > be clarified to avoid misunderstanding.
>
>Okay, we'll examine that.  Thanks.
>
> > 14. Section 6.2, second paragraph, first sentence: The attribute is
> > crypto not SRTP as it says in the text.
>
>Okay.
>
> > 15. Section 7.: The ABNF does not define the extension space. For
> > example the crypto-suite definition should have an extra alternative
> > crypto-suit-extension which is VCHAR. I think the whole ABNF needs to be
> > thought through to be more complete in the extension sense.
>
>Yes, it does.
>
>In Vienna it was also pointed out that we need IANA registration of
>all of the stuff in -sdescriptions-, as well.  We'll be adding that
>to the next I-D.
>
> > 16. Section 7. Why have the 2^ optional value if it does not matter.
> > Please simplify this.
>
>At an earlier review, Mark was asked to have a method to specify
>power-of-two key lifetimes.  This seemed the easiest way to allow
>specifying power-of-two key lifetimes.
>
>But rereading the draft, I see we never added text explaining the
>purpose of "2^"!  We'll add that at the next revision.  Thanks.
>
> > 17. Section 7, The absolute URI should not rely only on RFC 2396, there
> > is an extension to that document, please see RFC 2732.
>
>Thanks.
>
> > 18. Last page, has not the normal number of lines.
>
>Again, thanks for your review.

yes, thanks.

Mark


>-d
>
> > 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
> >
> >
> >
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www1.ietf.org/mailman/listinfo/mmusic



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Hi Dan,

My clarifications and further comments below:

Dan Wing wrote:

>>3. Section 3.1. This section needs to be improved. As I understand it
>>the a=crypto attribute can be used with any transport and media.
>>Therefore the section should not start with defining that this defines
>>the value for RTP/SAVP. The actual crypto suits will be required to be
>>registered and defined for each individual transport. There might in
>>fact be even further restrictions that apply to future crypto suits.
> 
> 
> -sdescriptions- is only trying to describe a=crypto for the RTP/SAVP
> (SRTP) transport.
> 

Okay, the question is if this is acceptable. As I can see it there are 
two ways when defining "a=crypto"

1. As a future safe replacement of a=k with possibility to declare new 
ways of transporting the keys by using a place holder. That way one 
could in the future define a=crypto to be valid also for other profiles 
and media types.

2. Only define a=crypto for SRTP. However there are still need to for 
the possibility to extend it for future crypto suites for SRTP.

Personally I don't have any strong view on the alternatives. However I 
feel there is need for some discussion on what the correct way forward is.

There also seem to exist possibilities that some of the k= usage should 
actually be defined using a=crypto to take care of all concerns 
regarding what encryption method is used etc.

> 
>>4. Section 3.4, I think the session parameters are actually specific to
>>the crypto suit, primarily.
> 
> 
> But "encrypted_srtcp" can be specified for any and all crypto suites.  Are
> there session parameters that are specific to a cryptographic suite?
> 
> 
>>Secondarily it can be dependent on more things.
> 
> 
> Could you explain?
> 

I think you helped explain the problem here. First, the session 
parameters that can be used are always in relation to the crypto suit. 
This means that each crypto suit needs to define which parameters that 
are valid for the crypto suit. Secondly there are certain parameters 
that will be present in all SRTP suites so far defined.

Taking the "encrypted_srtcp" as an example. Can't a future crypto suit 
define that RTCP will always be encrypted and the need for the parameter 
is removed. This may not be a good way of defining it, however it is 
possible.

So the point I try to make is that each crypto suit should define what 
parameters it supports.

> 
>>5. Section 3 and 4. In general I think the basic definition of the
>>a=crypto attribute can be more cleanly separated from the specific
>>application for SRTP.
> 
> 
> You mean, for example, to allow -sdescriptions- to define the keys
> used by some non-SRTP application (whiteboarding, perhaps)?  We're
> trying to only define usage with SRTP (RTP/SAVP).

Depending on what is the agreed on path, this comment either goes away 
or needs to be addressed.

>>10. Section 5.4, second paragraph: I think the limitation you would like
>>to declare here needs some discussion. First you forbid the usage of
>>multiple m= lines on the same port, after last weeks discussion I don't
>>know if this is acceptable, at least not without further motivation. I
>>would also write some recommendations/considerations regarding grouping
>>  of media lines here.
> 
> 
> In this I-D, we will be removing all references to multiple media lines
> with the same port.  A completely separate, unrelated, Internet Draft
> needs to be written -- with consensus from the WG -- describing how multiple
> media lines with the same UDP port are supposed to be interpreted.
> 
> (As an aside, the behavior for multiple media lines with the same port
> hasn't been specified elsewhere, so it is ambiguous.  I personally feel we
> need to create an unambigiuos method to express the desired behavior when
> an Offer contains multiple media lines with the same port.)
> 

I don't see a problem with defining it this way. However there is need 
to clearly define that this specification does not cover the case with 
multiple m= line for the same port.

>>  Are there any synchronization problem that can arise when doing
>>rekeying? There seems to be need to define from what seq the new key are
>>valid.
> 
> 
> That seems to be more a problem with SRTP than -sdescriiptions-.
> 
> To resolve the synchronization problem you have to use a per-packet MKI
> (which adds a few bytes to each SRTP packet) or the "from,to" mechanism
> (which doesn't add bytes to each SRTP packet); both MKI and "from,to"
> are described in SRTP itself.
> 
> Do you feel putting like the above paragraph might be useful in
> -sdescriptions-?
> 

Yes, and explanation on how the parameters a=crypto are used for this 
purpose.

> 
>>12. When it comes to SRTPs protection mechanism it seems necessary to be
>>able to define the window size using the a=crypto parameter. There are a
>>lot of difference between a uncompressed video stream running at a
>>couple of hundreds of Mbyte/s using 100 000 packets each second. This
>>will require a much larger window then a 64 kbit stream that sends 15
>>packets every second.
> 
> 
> SRTP (draft-ietf-avt-09.txt) says that the SRTP-WINDOW-SIZE is a
> receiver-side, implementation-dependent parameter.  If it is signaled
> in a unicast SDP, SRTP-WINDOW-SIZE would become a sender-side parameter.
> 
> Is that your intent of having the SRTP-WINDOW-SIZE indicated in the
> a=crypto line?

As SRTP-WINDOW-SIZE is application dependent there might be need for an 
application to hint how big this window should be to work satisfactory. 
However there might be enough information given in SDP attributes like 
a=maxprate and the bandwidth modifiers to allow a receiving side to 
derive the parameters. One could allow the parameter as a hint to the 
application.


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


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Hi Rod,

Thanks for the review.

I don't see a problem with the usage on session level you propose. The 
original intent of the change was to allow for stream combinations that 
has there bit-rate peaks at different places to be correctly expressed 
when given as combination. If an application uses the necessary SDP 
extension to understand that certain media tracks are alternatives to 
other I don't see any problem.

The type will be fixed in AUTH48 unless other comments result in a new 
version (I hope not).

Best Regards

Magnus


Rod.Walsh@nokia.com wrote:

> Hi Magnus
> 
> -bwparam-04 is an excellent document - thanks for doing the hard work!
> 
> The ID allows TIAS to be less than the sum of the total bandwidth as media bandwdith may vary diferently (makes sense). Also, this feature can be useful for when an application specifically expects/requires that all the media will not be used. Such as in a "internationalised TV program" - e.g. a single video stream and 2 audio streams of different languages to end-receivers required to decode only 1 video and 1 audio stream. In my eyes, this works without any changes at all to the ID (there's no need to explicitly mention this use case). However, I'd be interested to know if you see that there are any problems in this usage?
> 
> I also spotted one trivial typo - first word, last paragraph, section 6.3 - missing "
> 
> I don't think either of these should have an effect last call since the ID is already so mature.
> 
> Cheers, Rod.
> 
> 
> 
>>-----Original Message-----
>>From: ext Colin Perkins [mailto:csp@csperkins.org]
>>Sent: Thursday, July 17, 2003 9:01 PM
>>To: mmusic@ietf.org
>>Cc: jo@tzi.uni-bremen.de
>>Subject: [MMUSIC] WG Last call: draft-ietf-mmusic-sdp-bwparam-04.txt
>>
>>
>>As announced in the MMUSIC session this morning, the SDP bandwidth
>>parameters draft is now in working group last call. Please 
>>send any final
>>comments on this draft to the mailing list by 28th July. If 
>>no substantive
>>comments are received by that time, we propose to submit this 
>>draft to the
>>IESG for publication as a proposed standard RFC.
>>
>>-- 
>>Colin Perkins
>>csp@csperkins.org
>>
>>_______________________________________________
>>mmusic mailing list
>>mmusic@ietf.org
>>https://www1.ietf.org/mailman/listinfo/mmusic
>>
> 
> 

-- 

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
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Hi Enrico,

I have not had time to read the paper by Schulzrinne. However from what=20
I understand this should be possible to do in RTSP. However it touches=20
upon grey zones in the specification in two ways. The connection between =

RTSP message transport and session, that should not exist. And the usage =

of SETUP to change transport parameters. Therefore I am not surprised=20
that the Darwin server does not work in your experiments.

I think the Darwin server connects the TCP connection to the session and =

that the session actually is deleted if the TCP connection is removed.

There is no discussion of how to use RTSP for application layer=20
mobility. Are there a need for such a chapter in the specification or as =

a separate draft?

Best Regards

Magnus

Enrico Balatti wrote:

> Dear Ashutosh,
>=20
> thank you for you quickly reply.
>=20
> I tried to implement what the paper says about RTSP mobility.
> The Client now send the SETUP after the handoff, but the RTSP
> server (Darwin Streaming Server) doesn't respond with any message.
>=20
> Looking the Server source code I think that the Server doesn't
> accept the incoming SETUP message because it doesn't belong to
> any opend RTSP session (it is like there is a relation between=20
> the session and the transport socket and the session accepts packet
> from its client socket). Therefore if the packet arrive with a source=20
> address different (due to mobility), the packet is rejected.
>=20
> I don't know if this mean that the Server is not RTSP standard complian=
t.
>=20
> Do you know where the RTSP specification=20
> talks about the RTSP application-mobility?
>=20
> Thanks,
> Enrico
>=20
>=20
> -----Original Message-----
> From: Ashutosh Dutta [mailto:adutta@research.telcordia.com]
> Sent: marted=EC 22 luglio 2003 18.49
> To: Enrico Balatti
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] RTSP and mobility=20
>=20
>=20
>=20
> Enrico,
>        please see the paper
>       =20
>        http://www.cs.columbia.edu/~hgs/papers/Schu0007_Application.pdf,=

>        it talks about some issues related to RTSP and mobility when IP
>        address  changes on the  mobile because  of subnet  movement or
>        domain movement.
>=20
> Thanks
> Ashutosh
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
>=20

--=20

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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To: Magnus Westerlund <magnus.westerlund@ericsson.com>
cc: Enrico Balatti <balatti@cefriel.it>, mmusic@ietf.org
Subject: Re: [MMUSIC] RTSP and mobility 
In-reply-to: Your message of "Tue, 19 Aug 2003 13:57:13 +0200."
             <3F421099.8010508@ericsson.com> 
Date: Tue, 19 Aug 2003 08:56:26 -0400
From: Ashutosh Dutta <adutta@research.telcordia.com>
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A combination of  SETUP and PLAY as described in 10.4  and 10.5 of RFC
2326 should be able to handle the mobility of RTP stream. It should be
similar to  using SIP Re-INVITE  to take care of  mid-session mobility
for RTP stream. It  could be that SETUP message has to  be over UDP to
make it work instead of TCP in case of IP address change?

Thanks
Ashutosh

> Hi Enrico,
> 
> I have not had time to read the paper by Schulzrinne. However from what=20
> I understand this should be possible to do in RTSP. However it touches=20
> upon grey zones in the specification in two ways. The connection between =
> 
> RTSP message transport and session, that should not exist. And the usage =
> 
> of SETUP to change transport parameters. Therefore I am not surprised=20
> that the Darwin server does not work in your experiments.
> 
> I think the Darwin server connects the TCP connection to the session and =
> 
> that the session actually is deleted if the TCP connection is removed.
> 
> There is no discussion of how to use RTSP for application layer=20
> mobility. Are there a need for such a chapter in the specification or as =
> 
> a separate draft?
> 
> Best Regards
> 
> Magnus
> 
> Enrico Balatti wrote:
> 
> > Dear Ashutosh,
> >=20
> > thank you for you quickly reply.
> >=20
> > I tried to implement what the paper says about RTSP mobility.
> > The Client now send the SETUP after the handoff, but the RTSP
> > server (Darwin Streaming Server) doesn't respond with any message.
> >=20
> > Looking the Server source code I think that the Server doesn't
> > accept the incoming SETUP message because it doesn't belong to
> > any opend RTSP session (it is like there is a relation between=20
> > the session and the transport socket and the session accepts packet
> > from its client socket). Therefore if the packet arrive with a source=20
> > address different (due to mobility), the packet is rejected.
> >=20
> > I don't know if this mean that the Server is not RTSP standard complian=
> t.
> >=20
> > Do you know where the RTSP specification=20
> > talks about the RTSP application-mobility?
> >=20
> > Thanks,
> > Enrico
> >=20
> >=20
> > -----Original Message-----
> > From: Ashutosh Dutta [mailto:adutta@research.telcordia.com]
> > Sent: marted=EC 22 luglio 2003 18.49
> > To: Enrico Balatti
> > Cc: mmusic@ietf.org
> > Subject: Re: [MMUSIC] RTSP and mobility=20
> >=20
> >=20
> >=20
> > Enrico,
> >        please see the paper
> >       =20
> >        http://www.cs.columbia.edu/~hgs/papers/Schu0007_Application.pdf,=
> 
> >        it talks about some issues related to RTSP and mobility when IP
> >        address  changes on the  mobile because  of subnet  movement or
> >        domain movement.
> >=20
> > Thanks
> > Ashutosh
> >=20
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www1.ietf.org/mailman/listinfo/mmusic
> >=20
> 
> --=20
> 
> 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
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic

Ashutosh 

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Date: Tue, 19 Aug 2003 11:15:53 -0400
From: Henning Schulzrinne <hgs@cs.columbia.edu>
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Subject: Re: [MMUSIC] RTSP and mobility
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Since this is a source of interoperability problems, some basic=20
discussions on transport behavior seems helpful for the base spec. You=20
can't fix the problems with an additional spec. I don't think this has=20
to be complicated, but address changes are now not uncommon for=20
long-lived sessions (e.g., due to multi-homed hosts, DHCP or moving=20
around between 802.11 APs.)

Magnus Westerlund wrote:

> Hi Enrico,
>=20
> I have not had time to read the paper by Schulzrinne. However from what=
=20
> I understand this should be possible to do in RTSP. However it touches=20
> upon grey zones in the specification in two ways. The connection betwee=
n=20
> RTSP message transport and session, that should not exist. And the usag=
e=20
> of SETUP to change transport parameters. Therefore I am not surprised=20
> that the Darwin server does not work in your experiments.
>=20
> I think the Darwin server connects the TCP connection to the session an=
d=20
> that the session actually is deleted if the TCP connection is removed.
>=20
> There is no discussion of how to use RTSP for application layer=20
> mobility. Are there a need for such a chapter in the specification or a=
s=20
> a separate draft?
>=20
> Best Regards
>=20
> Magnus
>=20
> Enrico Balatti wrote:
>=20
>> Dear Ashutosh,
>>
>> thank you for you quickly reply.
>>
>> I tried to implement what the paper says about RTSP mobility.
>> The Client now send the SETUP after the handoff, but the RTSP
>> server (Darwin Streaming Server) doesn't respond with any message.
>>
>> Looking the Server source code I think that the Server doesn't
>> accept the incoming SETUP message because it doesn't belong to
>> any opend RTSP session (it is like there is a relation between the=20
>> session and the transport socket and the session accepts packet
>> from its client socket). Therefore if the packet arrive with a source=20
>> address different (due to mobility), the packet is rejected.
>>
>> I don't know if this mean that the Server is not RTSP standard complia=
nt.
>>
>> Do you know where the RTSP specification talks about the RTSP=20
>> application-mobility?
>>
>> Thanks,
>> Enrico
>>
>>
>> -----Original Message-----
>> From: Ashutosh Dutta [mailto:adutta@research.telcordia.com]
>> Sent: marted=EC 22 luglio 2003 18.49
>> To: Enrico Balatti
>> Cc: mmusic@ietf.org
>> Subject: Re: [MMUSIC] RTSP and mobility
>>
>>
>> Enrico,
>>        please see the paper
>>              =20
>> http://www.cs.columbia.edu/~hgs/papers/Schu0007_Application.pdf,
>>        it talks about some issues related to RTSP and mobility when IP
>>        address  changes on the  mobile because  of subnet  movement or
>>        domain movement.
>>
>> Thanks
>> Ashutosh
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www1.ietf.org/mailman/listinfo/mmusic
>>
>=20



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



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Subject: Re: [MMUSIC] Comments on draft-ietf-mmusic-sdescription-01.txt
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Magnus,

At 01:39 PM 8/19/2003 +0200, Magnus Westerlund wrote:
>Hi Dan,
>
>My clarifications and further comments below:
>
>Dan Wing wrote:
>
>>>3. Section 3.1. This section needs to be improved. As I understand it
>>>the a=crypto attribute can be used with any transport and media.
>>>Therefore the section should not start with defining that this defines
>>>the value for RTP/SAVP. The actual crypto suits will be required to be
>>>registered and defined for each individual transport. There might in
>>>fact be even further restrictions that apply to future crypto suits.
>>
>>-sdescriptions- is only trying to describe a=crypto for the RTP/SAVP
>>(SRTP) transport.
>
>Okay, the question is if this is acceptable. As I can see it there are two 
>ways when defining "a=crypto"
>
>1. As a future safe replacement of a=k

Do you mean k=?

>with possibility to declare new ways of transporting the keys by using a 
>place holder. That way one could in the future define a=crypto to be valid 
>also for other profiles and media types.
>
>2. Only define a=crypto for SRTP. However there are still need to for the 
>possibility to extend it for future crypto suites for SRTP.
>
>Personally I don't have any strong view on the alternatives. However I 
>feel there is need for some discussion on what the correct way forward is.
>
>There also seem to exist possibilities that some of the k= usage should 
>actually be defined using a=crypto to take care of all concerns regarding 
>what encryption method is used etc.

I think that a=crypto does everything that k= does.  My concern is that 
there might be an unforeseen need for extension for a particular transport.



>>>4. Section 3.4, I think the session parameters are actually specific to
>>>the crypto suit, primarily.
>>
>>But "encrypted_srtcp" can be specified for any and all crypto suites.  Are
>>there session parameters that are specific to a cryptographic suite?
>>
>>>Secondarily it can be dependent on more things.
>>
>>Could you explain?
>
>I think you helped explain the problem here. First, the session parameters 
>that can be used are always in relation to the crypto suit.

No, this is not the case:  They are specific to the transport.  The 
crypto-suite may or may not be specific to the transport (see below).

>This means that each crypto suit needs to define which parameters that are 
>valid for the crypto suit. Secondly there are certain parameters that will 
>be present in all SRTP suites so far defined.
>
>Taking the "encrypted_srtcp" as an example. Can't a future crypto suit 
>define that RTCP will always be encrypted and the need for the parameter 
>is removed. This may not be a good way of defining it, however it is possible.

I suppose it is but it would be a very brittle way to define the crypto 
suite, which *may be* orthogonal to the transport.  In the first submission 
of this work, we expected the crypto suite to be transport 
independent.  The reason it is not is because we have seen examples of the 
crypto suite being tailored and optimized for a particular transport.  For 
example, AES counter mode for SRTP re-uses the RTP sequence number to 
obviate the need to add a nonce to the header and save precious bits on low 
bandwidth links.  IPsec decided to put a nonce into the IPsec transport 
header for AES counter mode - the IV field - rather than overload the 
sequence number.  For sdescriptions, it seemed to us to be safer to not try 
and develop a single set of crypto suites for all transports since the 
namespace could become unwieldy (e.g. AES-CTR-128-HMAC-SHA1-80-SRTP, 
AES-CTR-128-HMAC-SHA1-80-IPSEC).

You seem to be taking this to its logical conclusion of putting all 
transport-specific parameters into the crypto suite, but I don't think we 
want to do that.  For one thing, it will result in too many crypto 
suites.  A bigger problem is that some parameter support a range of values, 
such as the size of the MKI.


>So the point I try to make is that each crypto suit should define what 
>parameters it supports.

We made a distinction of parameters that are specific to the key, these are 
the domain of the crypto suite, versus parameters related to the transport, 
these are the session parameters.



>>>5. Section 3 and 4. In general I think the basic definition of the
>>>a=crypto attribute can be more cleanly separated from the specific
>>>application for SRTP.
>>
>>You mean, for example, to allow -sdescriptions- to define the keys
>>used by some non-SRTP application (whiteboarding, perhaps)?  We're
>>trying to only define usage with SRTP (RTP/SAVP).
>
>Depending on what is the agreed on path, this comment either goes away or 
>needs to be addressed.

I think we created this confusion by changing course on the WG.  We 
originally wanted a replacement for k= because of its deficiencies, which 
we describe in the I-D.  We developed some misgivings on this course when 
working through the O/A, which is very transport specific.  It did not seem 
prudent to define secure transports, O/A, etc. for fax, whiteboard, modem, 
etc. if we are not planning on testing these specifications on real-world 
implementations.

I guess we are in a bit of a muddle on this important point.  Sorry about 
that.  What we feel confident doing here is defining the best possible 
specification for SRTP.


>>>10. Section 5.4, second paragraph: I think the limitation you would like
>>>to declare here needs some discussion. First you forbid the usage of
>>>multiple m= lines on the same port, after last weeks discussion I don't
>>>know if this is acceptable, at least not without further motivation. I
>>>would also write some recommendations/considerations regarding grouping
>>>  of media lines here.
>>
>>In this I-D, we will be removing all references to multiple media lines
>>with the same port.  A completely separate, unrelated, Internet Draft
>>needs to be written -- with consensus from the WG -- describing how multiple
>>media lines with the same UDP port are supposed to be interpreted.
>>(As an aside, the behavior for multiple media lines with the same port
>>hasn't been specified elsewhere, so it is ambiguous.  I personally feel we
>>need to create an unambigiuos method to express the desired behavior when
>>an Offer contains multiple media lines with the same port.)
>
>I don't see a problem with defining it this way. However there is need to 
>clearly define that this specification does not cover the case with 
>multiple m= line for the same port.
>
>>>  Are there any synchronization problem that can arise when doing
>>>rekeying? There seems to be need to define from what seq the new key are
>>>valid.
>>
>>That seems to be more a problem with SRTP than -sdescriiptions-.
>>To resolve the synchronization problem you have to use a per-packet MKI
>>(which adds a few bytes to each SRTP packet) or the "from,to" mechanism
>>(which doesn't add bytes to each SRTP packet); both MKI and "from,to"
>>are described in SRTP itself.
>>Do you feel putting like the above paragraph might be useful in
>>-sdescriptions-?
>
>Yes, and explanation on how the parameters a=crypto are used for this purpose.

I think that a non-zero MKI explains it all.



>>>12. When it comes to SRTPs protection mechanism it seems necessary to be
>>>able to define the window size using the a=crypto parameter. There are a
>>>lot of difference between a uncompressed video stream running at a
>>>couple of hundreds of Mbyte/s using 100 000 packets each second. This
>>>will require a much larger window then a 64 kbit stream that sends 15
>>>packets every second.
>>
>>SRTP (draft-ietf-avt-09.txt) says that the SRTP-WINDOW-SIZE is a
>>receiver-side, implementation-dependent parameter.  If it is signaled
>>in a unicast SDP, SRTP-WINDOW-SIZE would become a sender-side parameter.
>>Is that your intent of having the SRTP-WINDOW-SIZE indicated in the
>>a=crypto line?
>
>As SRTP-WINDOW-SIZE is application dependent there might be need for an 
>application to hint how big this window should be to work satisfactory. 
>However there might be enough information given in SDP attributes like 
>a=maxprate and the bandwidth modifiers to allow a receiving side to derive 
>the parameters. One could allow the parameter as a hint to the application.

ok.  I think the outstanding problem that we need to resolve with the WG is 
to what degree we generalize a=crypto beyond the SRTP transport.

thanks, Mark



>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



_______________________________________________
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To: Henning Schulzrinne <hgs@cs.columbia.edu>
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Subject: Re: [MMUSIC] RTSP and mobility
References: <6F5FB8313FE82C4EA10DA4DA40B90949B0658A@swing.cefriel.it> <3F421099.8010508@ericsson.com> <3F423F29.5030708@cs.columbia.edu>
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Hi Henning & Enrico

Yes, I don't see a problem with this. In the updated specification this=20
is the main reason why we maintain and strength the requirement that the =

RTSP session shall be independent of the RTSP message transport.

Henning, you are probably correct that the it only a few paragraphs to=20
explain how to achieve application layer mobility and that way provide=20
robustness.

Unless some one objects I would propose that we add some text on this=20
issue in the RTSP core specification.

Cheers

Magnus



Henning Schulzrinne wrote:
> Since this is a source of interoperability problems, some basic=20
> discussions on transport behavior seems helpful for the base spec. You =

> can't fix the problems with an additional spec. I don't think this has =

> to be complicated, but address changes are now not uncommon for=20
> long-lived sessions (e.g., due to multi-homed hosts, DHCP or moving=20
> around between 802.11 APs.)
>=20
> Magnus Westerlund wrote:
>=20
>> Hi Enrico,
>>
>> I have not had time to read the paper by Schulzrinne. However from=20
>> what I understand this should be possible to do in RTSP. However it=20
>> touches upon grey zones in the specification in two ways. The=20
>> connection between RTSP message transport and session, that should not=
=20
>> exist. And the usage of SETUP to change transport parameters.=20
>> Therefore I am not surprised that the Darwin server does not work in=20
>> your experiments.
>>
>> I think the Darwin server connects the TCP connection to the session=20
>> and that the session actually is deleted if the TCP connection is=20
>> removed.
>>
>> There is no discussion of how to use RTSP for application layer=20
>> mobility. Are there a need for such a chapter in the specification or =

>> as a separate draft?
>>
>> Best Regards
>>
>> Magnus
>>
>> Enrico Balatti wrote:
>>
>>> Dear Ashutosh,
>>>
>>> thank you for you quickly reply.
>>>
>>> I tried to implement what the paper says about RTSP mobility.
>>> The Client now send the SETUP after the handoff, but the RTSP
>>> server (Darwin Streaming Server) doesn't respond with any message.
>>>
>>> Looking the Server source code I think that the Server doesn't
>>> accept the incoming SETUP message because it doesn't belong to
>>> any opend RTSP session (it is like there is a relation between the=20
>>> session and the transport socket and the session accepts packet
>>> from its client socket). Therefore if the packet arrive with a source=
=20
>>> address different (due to mobility), the packet is rejected.
>>>
>>> I don't know if this mean that the Server is not RTSP standard=20
>>> compliant.
>>>
>>> Do you know where the RTSP specification talks about the RTSP=20
>>> application-mobility?
>>>
>>> Thanks,
>>> Enrico
>>>
>>>
>>> -----Original Message-----
>>> From: Ashutosh Dutta [mailto:adutta@research.telcordia.com]
>>> Sent: marted=EC 22 luglio 2003 18.49
>>> To: Enrico Balatti
>>> Cc: mmusic@ietf.org
>>> Subject: Re: [MMUSIC] RTSP and mobility
>>>
>>>
>>> Enrico,
>>>        please see the paper
>>>              =20
>>> http://www.cs.columbia.edu/~hgs/papers/Schu0007_Application.pdf,
>>>        it talks about some issues related to RTSP and mobility when I=
P
>>>        address  changes on the  mobile because  of subnet  movement o=
r
>>>        domain movement.
>>>
>>> Thanks
>>> Ashutosh
>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/mmusic
>>>
>>
>=20
>=20

--=20

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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From: Tom Marshall <tmarshall@real.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Cc: Henning Schulzrinne <hgs@cs.columbia.edu>,
        Enrico Balatti <balatti@cefriel.it>, mmusic@ietf.org
Subject: Re: [MMUSIC] RTSP and mobility
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References: <6F5FB8313FE82C4EA10DA4DA40B90949B0658A@swing.cefriel.it> <3F421099.8010508@ericsson.com> <3F423F29.5030708@cs.columbia.edu> <3F42551B.9010503@ericsson.com>
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In order to make this work, don't you also need to s/MAY/MUST/ in the first
paragraph of 10.4?  This would seem to be a controversial change.

On Tue, Aug 19, 2003 at 06:49:31PM +0200, Magnus Westerlund wrote:
> Hi Henning & Enrico
>=20
> Yes, I don't see a problem with this. In the updated specification this=
=20
> is the main reason why we maintain and strength the requirement that the=
=20
> RTSP session shall be independent of the RTSP message transport.
>=20
> Henning, you are probably correct that the it only a few paragraphs to=20
> explain how to achieve application layer mobility and that way provide=20
> robustness.
>=20
> Unless some one objects I would propose that we add some text on this=20
> issue in the RTSP core specification.
>=20
> Cheers
>=20
> Magnus
>=20
>=20
>=20
> Henning Schulzrinne wrote:
> >Since this is a source of interoperability problems, some basic=20
> >discussions on transport behavior seems helpful for the base spec. You=
=20
> >can't fix the problems with an additional spec. I don't think this has=
=20
> >to be complicated, but address changes are now not uncommon for=20
> >long-lived sessions (e.g., due to multi-homed hosts, DHCP or moving=20
> >around between 802.11 APs.)
> >
> >Magnus Westerlund wrote:
> >
> >>Hi Enrico,
> >>
> >>I have not had time to read the paper by Schulzrinne. However from=20
> >>what I understand this should be possible to do in RTSP. However it=20
> >>touches upon grey zones in the specification in two ways. The=20
> >>connection between RTSP message transport and session, that should not=
=20
> >>exist. And the usage of SETUP to change transport parameters.=20
> >>Therefore I am not surprised that the Darwin server does not work in=20
> >>your experiments.
> >>
> >>I think the Darwin server connects the TCP connection to the session=20
> >>and that the session actually is deleted if the TCP connection is=20
> >>removed.
> >>
> >>There is no discussion of how to use RTSP for application layer=20
> >>mobility. Are there a need for such a chapter in the specification or=
=20
> >>as a separate draft?
> >>
> >>Best Regards
> >>
> >>Magnus
> >>
> >>Enrico Balatti wrote:
> >>
> >>>Dear Ashutosh,
> >>>
> >>>thank you for you quickly reply.
> >>>
> >>>I tried to implement what the paper says about RTSP mobility.
> >>>The Client now send the SETUP after the handoff, but the RTSP
> >>>server (Darwin Streaming Server) doesn't respond with any message.
> >>>
> >>>Looking the Server source code I think that the Server doesn't
> >>>accept the incoming SETUP message because it doesn't belong to
> >>>any opend RTSP session (it is like there is a relation between the=20
> >>>session and the transport socket and the session accepts packet
> >>>from its client socket). Therefore if the packet arrive with a source=
=20
> >>>address different (due to mobility), the packet is rejected.
> >>>
> >>>I don't know if this mean that the Server is not RTSP standard=20
> >>>compliant.
> >>>
> >>>Do you know where the RTSP specification talks about the RTSP=20
> >>>application-mobility?
> >>>
> >>>Thanks,
> >>>Enrico
> >>>
> >>>
> >>>-----Original Message-----
> >>>From: Ashutosh Dutta [mailto:adutta@research.telcordia.com]
> >>>Sent: marted=EC 22 luglio 2003 18.49
> >>>To: Enrico Balatti
> >>>Cc: mmusic@ietf.org
> >>>Subject: Re: [MMUSIC] RTSP and mobility
> >>>
> >>>
> >>>Enrico,
> >>>       please see the paper
> >>>             =20
> >>>http://www.cs.columbia.edu/~hgs/papers/Schu0007_Application.pdf,
> >>>       it talks about some issues related to RTSP and mobility when IP
> >>>       address  changes on the  mobile because  of subnet  movement or
> >>>       domain movement.
> >>>
> >>>Thanks
> >>>Ashutosh
> >>>
> >>>_______________________________________________
> >>>mmusic mailing list
> >>>mmusic@ietf.org
> >>>https://www1.ietf.org/mailman/listinfo/mmusic
> >>>
> >>
> >
> >
>=20
> --=20
>=20
> Magnus Westerlund
>=20
> Multimedia Technologies, Ericsson Research EAB/TVA/A
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
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Hi Tom,

Yes, you are correct that the current MAY would need to be changed to 
MUST to always allow it. Performing this change may be a bit to much 
change. I think that it would be better to explain why support of this 
functionality is good, based on that it makes Application layer mobility 
possible. If there is need to learn the support for this, we should 
define a feature tag for it.

Is anybody willing to look through the specification to see if there are 
any more conflicts or limitations and provide a draft informational text 
about this?

Best Regards

Magnus



Tom Marshall wrote:

> In order to make this work, don't you also need to s/MAY/MUST/ in the
> first
> paragraph of 10.4?  This would seem to be a controversial change.
> 


Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
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Hi Mark

More comments below.

Mark Baugher wrote:

>> Okay, the question is if this is acceptable. As I can see it there are 
>> two ways when defining "a=crypto"
>>
>> 1. As a future safe replacement of a=k
> 
> 
> Do you mean k=?

Yes.

> 
> 
> I think that a=crypto does everything that k= does.  My concern is that 
> there might be an unforeseen need for extension for a particular transport.
> 

Not as currently specified. Can't k= be used to specify the use of DES 
on complete RTP and RTCP packets? This is certainly something different 
than what so far is defined. However I don't know if there is any need 
for including this.


> 
> I suppose it is but it would be a very brittle way to define the crypto 
> suite, which *may be* orthogonal to the transport.  In the first 
> submission of this work, we expected the crypto suite to be transport 
> independent.  The reason it is not is because we have seen examples of 
> the crypto suite being tailored and optimized for a particular 
> transport.  For example, AES counter mode for SRTP re-uses the RTP 
> sequence number to obviate the need to add a nonce to the header and 
> save precious bits on low bandwidth links.  IPsec decided to put a nonce 
> into the IPsec transport header for AES counter mode - the IV field - 
> rather than overload the sequence number.  For sdescriptions, it seemed 
> to us to be safer to not try and develop a single set of crypto suites 
> for all transports since the namespace could become unwieldy (e.g. 
> AES-CTR-128-HMAC-SHA1-80-SRTP, AES-CTR-128-HMAC-SHA1-80-IPSEC).
> 
> You seem to be taking this to its logical conclusion of putting all 
> transport-specific parameters into the crypto suite, but I don't think 
> we want to do that.  For one thing, it will result in too many crypto 
> suites.  A bigger problem is that some parameter support a range of 
> values, such as the size of the MKI.

What do you mean with transport here? Depending on what level you mean I 
think that one must in fact make the crypto suits transport dependent.

If you with transport means for example RTP and the used profile the 
crypto suit must in fact be transport dependent as I think your example 
crypto suits AES-CTR-128-HMAC-SHA1-80-SRTP, 
AES-CTR-128-HMAC-SHA1-80-IPSEC visualize. Even if the same crypto 
algorithm is used by both SRTP and IPSEC the other parameters are not 
identical I would assume, or are they? There is need to know that IPSEC 
should be applied and that SRTP is not needed. Therefore I would assume 
that the crypto suits must be in fact be dependent on encryption method 
which might imply certain transports.

What we are still discussing in fact are if the a=crypto attribute only 
should be applicable for SRTP independent on lower layers that transport 
them, or we should make it possible for future extensions that might 
apply for usage with other crypto methods?

> 
> 
>> So the point I try to make is that each crypto suit should define what 
>> parameters it supports.
> 
> 
> We made a distinction of parameters that are specific to the key, these 
> are the domain of the crypto suite, versus parameters related to the 
> transport, these are the session parameters.
> 
> 

In fact I would instead say that sessions parameters are SRTP parameters 
that are not depending on the crypto suit. Are you certain that 
SEQ/ROC/SSRC is session parameters. I think they would be in fact 
useless if one develops a crypto suit that are based on explicit IVs 
instead of the current implicit ones.

As a result of that all session parameters are in direct relation to 
SRTP usage I don't see a problem with tying them to the crypto suit as 
these crypto suits are directly related SRTP and no other protocol. None 
of the listed parameters in section 4.3 would care if the RTP is 
transported over IP/UDP, IP/DCCP, or something completely different as 
long as the packet borders are recoverable.

> 
> I think we created this confusion by changing course on the WG.  We 
> originally wanted a replacement for k= because of its deficiencies, 
> which we describe in the I-D.  We developed some misgivings on this 
> course when working through the O/A, which is very transport specific.  
> It did not seem prudent to define secure transports, O/A, etc. for fax, 
> whiteboard, modem, etc. if we are not planning on testing these 
> specifications on real-world implementations.
> 
> I guess we are in a bit of a muddle on this important point.  Sorry 
> about that.  What we feel confident doing here is defining the best 
> possible specification for SRTP.

I agree about this. I hope that this message exchange has made it clear 
to the general public that you would like to change the intention. As 
this being a WG document you are in fact required or at least strongly 
encourage to ask the group for input on such changes.

I don't think you need to consider all strange, non used methods. 
However what I think the WG needs to consider is if it should be 
possible to extend "a=crypto" to carry keys for, DES and 3DES of the 
complete RTP packet as defined in the RTP spec, or something else that 
people see need of?

If the result of that question is a yes, then I think that the a=crypto 
attribute must either put in a method indicator first or make the crypto 
suite carry implicit information about the method and the structure of 
the rest of the a=crypto line.


>>> That seems to be more a problem with SRTP than -sdescriiptions-.
>>> To resolve the synchronization problem you have to use a per-packet MKI
>>> (which adds a few bytes to each SRTP packet) or the "from,to" mechanism
>>> (which doesn't add bytes to each SRTP packet); both MKI and "from,to"
>>> are described in SRTP itself.
>>> Do you feel putting like the above paragraph might be useful in
>>> -sdescriptions-?
>>
>>
>> Yes, and explanation on how the parameters a=crypto are used for this 
>> purpose.
> 
> 
> I think that a non-zero MKI explains it all.
> 

What about the usage of <From, To>? I couldn't found anything about it. 
What I think is important is to resolve how a rekeying with this 
mechanism will happen.

I don't think that the following section actually explains how to 
perform the rekeying to a point of interoperability.

Section 5.7:
   "When an active SRTP session is rekeyed, this is indicated by sending
    a new SDP.  Rekeying is done following the rules described for a
    normal Offer/Answer exchange.  The Answerer can take this
    opportunity to rekey the traffic it is sending, if the Answerer
    desires.  During rekeying, the session parameters cannot be changed
    and MUST NOT be specified in the Offer or the Answer. "

I think that you must in fact clarify that the key for rekeying must be 
connected to a MKI or "from,to" interval to indicate when valid.


> ok.  I think the outstanding problem that we need to resolve with the WG 
> is to what degree we generalize a=crypto beyond the SRTP transport.
> 

Agreed, I hope that some other people can give there opinion about this.


Best Regards

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
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Hi,

During the 57th IETF meeting we discussed the meaning of open intervals 
for the SDP attribute a=range, e.g. "a=range:npt=0-". My memory and 
notes indicates that the conclusions made was:

1. An open range indicates a media with unknown duration.

2. There seem to be need for more attributes expressing things like: 
seekability.

The start of the discussion on this topic was the usability of an open 
range to indicate that media is live and not seekable. There where 
comments that even though a session is live it might be seekable, at 
least into the past, i.e. behind "now". This is can be accomplished by 
on the fly recording servers. However there is a lack of possibility to 
express this capability.

There are currently implementations that interpret an open range as not 
being seekable, for example the quicktime 6.3 player. So to rehash the 
question in this forum:

Would it be acceptable to define an open interval range attribute as 
being live and non seekable unless other attributes give further 
properties, like seekable?

If you own a commercial implementation of a RTSP client, could you 
please inform how it would react to an open range?

Looking forward to your comments.

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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From: "Christer Holmberg (SE/LMF)" <christer.holmberg@ericsson.com>
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Hi,

This mail is related to an issue which has came up at the SIPit interop event.

Page 39 in sdp-new-13 says the following:

"RTP payload formats under the "RTP/AVP" protocol that have been
assigned static payload types MUST use the static payload type as
their "fmt" value.  For payload formats under "RTP/AVP" that have
a dynamic payload type number, the dynamic payload type number is
given as the "fmt" and an additional "rtpmap" attribute specifies
registration for the payload format."


My question is: is it allowed, when using a dynamic codec, to "override" a static payload type, using rtpmap, for that codec.

For example, 0 is the statically defined payload type for PCMU. Now, is the following allowed:

m=audio 99999 0
a=rtpmap: 0 DYNAMIC_CODEC/8000

I can't find anywhere in the sdp draft text saying that this is forbidden (I appologise if it's there and I've just missed it), and RFC3551 (chapter 3) doesn't forbid it either, so...

Regards,

Christer Holmberg
Ericsson Finland



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From: "Hearty, John" <John.Hearty@Level3.com>
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This RFC 3551 text seems to answer your question most directly, the last
MAY allows it.:

3.  Registering Additional Encodings
<snip>
   This profile reserves payload type numbers in the range 96-127
   exclusively for dynamic assignment.  Applications SHOULD first use
   values in this range for dynamic payload types.  Those applications
   which need to define more than 32 dynamic payload types MAY bind
   codes below 96, in which case it is RECOMMENDED that unassigned
   payload type numbers be used first.  However, the statically assigned
   payload types are default bindings and MAY be dynamically bound to
   new encodings if needed.  Redefining payload types below 96 may cause
   incorrect operation if an attempt is made to join a session without
   obtaining session description information that defines the dynamic
   payload types.


John


> -----Original Message-----
> From: mmusic-admin@ietf.org [mailto:mmusic-admin@ietf.org] On Behalf
Of
> Christer Holmberg (SE/LMF)
> Sent: Thursday, August 21, 2003 9:13 AM
> To: 'mmusic@ietf.org'
> Subject: [MMUSIC] Using static payload types for dynamic codecs
>=20
>=20
> Hi,
>=20
> This mail is related to an issue which has came up at the SIPit
interop
> event.
>=20
> Page 39 in sdp-new-13 says the following:
>=20
> "RTP payload formats under the "RTP/AVP" protocol that have been
> assigned static payload types MUST use the static payload type as
> their "fmt" value.  For payload formats under "RTP/AVP" that have
> a dynamic payload type number, the dynamic payload type number is
> given as the "fmt" and an additional "rtpmap" attribute specifies
> registration for the payload format."
>=20
>=20
> My question is: is it allowed, when using a dynamic codec, to
"override" a
> static payload type, using rtpmap, for that codec.
>=20
> For example, 0 is the statically defined payload type for PCMU. Now,
is
> the following allowed:
>=20
> m=3Daudio 99999 0
> a=3Drtpmap: 0 DYNAMIC_CODEC/8000
>=20
> I can't find anywhere in the sdp draft text saying that this is
forbidden
> (I appologise if it's there and I've just missed it), and RFC3551
(chapter
> 3) doesn't forbid it either, so...
>=20
> Regards,
>=20
> Christer Holmberg
> Ericsson Finland
>=20
>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic

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From: "Christer Holmberg (SE/LMF)" <christer.holmberg@ericsson.com>
To: "'Hearty, John'" <John.Hearty@Level3.com>, mmusic@ietf.org
Subject: RE: [MMUSIC] Using static payload types for dynamic codecs
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Hi John,

Yes, RFC 3551 allows it. The question is if SDP allows it. One COULD use SDP for another transport than RTP, so I really think the SDP spec should define if it's allowd or not in SDP.

Thanks for your comments!

Regards,

Christer Holmberg
Ericsson Finland



> -----Original Message-----
> From: Hearty, John [mailto:John.Hearty@Level3.com]
> Sent: 21. elokuuta 2003 19:06
> To: Christer Holmberg (SE/LMF); mmusic@ietf.org
> Subject: RE: [MMUSIC] Using static payload types for dynamic codecs
> 
> 
> 
> This RFC 3551 text seems to answer your question most 
> directly, the last
> MAY allows it.:
> 
> 3.  Registering Additional Encodings
> <snip>
>    This profile reserves payload type numbers in the range 96-127
>    exclusively for dynamic assignment.  Applications SHOULD first use
>    values in this range for dynamic payload types.  Those applications
>    which need to define more than 32 dynamic payload types MAY bind
>    codes below 96, in which case it is RECOMMENDED that unassigned
>    payload type numbers be used first.  However, the 
> statically assigned
>    payload types are default bindings and MAY be dynamically bound to
>    new encodings if needed.  Redefining payload types below 
> 96 may cause
>    incorrect operation if an attempt is made to join a session without
>    obtaining session description information that defines the dynamic
>    payload types.
> 
> 
> John
> 
> 
> > -----Original Message-----
> > From: mmusic-admin@ietf.org [mailto:mmusic-admin@ietf.org] On Behalf
> Of
> > Christer Holmberg (SE/LMF)
> > Sent: Thursday, August 21, 2003 9:13 AM
> > To: 'mmusic@ietf.org'
> > Subject: [MMUSIC] Using static payload types for dynamic codecs
> > 
> > 
> > Hi,
> > 
> > This mail is related to an issue which has came up at the SIPit
> interop
> > event.
> > 
> > Page 39 in sdp-new-13 says the following:
> > 
> > "RTP payload formats under the "RTP/AVP" protocol that have been
> > assigned static payload types MUST use the static payload type as
> > their "fmt" value.  For payload formats under "RTP/AVP" that have
> > a dynamic payload type number, the dynamic payload type number is
> > given as the "fmt" and an additional "rtpmap" attribute specifies
> > registration for the payload format."
> > 
> > 
> > My question is: is it allowed, when using a dynamic codec, to
> "override" a
> > static payload type, using rtpmap, for that codec.
> > 
> > For example, 0 is the statically defined payload type for PCMU. Now,
> is
> > the following allowed:
> > 
> > m=audio 99999 0
> > a=rtpmap: 0 DYNAMIC_CODEC/8000
> > 
> > I can't find anywhere in the sdp draft text saying that this is
> forbidden
> > (I appologise if it's there and I've just missed it), and RFC3551
> (chapter
> > 3) doesn't forbid it either, so...
> > 
> > Regards,
> > 
> > Christer Holmberg
> > Ericsson Finland
> > 
> > 
> > 
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www1.ietf.org/mailman/listinfo/mmusic
> 

_______________________________________________
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Date: Thu, 21 Aug 2003 11:37:26 -0700 (PDT)
From: Geetha Srikantan <Geetha.Srikantan@Sun.COM>
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I meant to send this in earlier..here it is now..

Questions and comments (some typos too):

Page 10: Transport Negotiation
Can transport parameters of an existing session be changed, mid-stream?
For instance, is it possible to change the server's RTP/RTCP ports
mid-stream? 
If so, exactly how does the transition to using the new ports happen? Do some 
mechanisms need to be specified to ensure synchronization etc after the cutover?
If not, it would be good to specify the states in which transport negotiation 
can be done.

Page 10, last paragraph section 1.5 - are 'stream identifiers' the
same as 'session identifiers'? If so, it might help to use the same
term throughout the document.

Page 14, section 2 - typo after last paragraph 'b'?

Page 15, 1st paragraph, last sentence - typo?

Page 15, 3rd paragraph, it might help to rephrase to say that FQDNs
are preferred, especially for IPv6, (maybe even 'required for IPv6 ?)
however IP addresses should also be supported (for backward
compatibility). 

Page 18, section 3.7, 2nd sentence - typo 'in' appears twice.

Page 20, section 5 - adding Expires header?
Page 27, section 7.1.2 - add Expires header?

Page 30, section 9.3, 3rd paragraph, last sentence - typo?
Page 31, section 9.3, Is play.basic a superset of *not* con.persistent?

Page 31, section 9.4, 1st paragraph, 4th sentence s/must/should/?
I.e. 'This requires that any RTSP URL pointing tat a IPv6 host should
use fully qualified domain name and not a IPv6 address.'

Page 31, section 10, 1st paragraph, 2nd sentence - typo 'too'?

Page 31-32, section 10,
Should responses to OPTIONS be cached? If so, is there an expiry on
this and any specification on how to refresh?
Are there rules about determining capability in the presence of a
proxy?
Does this section specify all that is required for each endpoint to
determine capability?

Page 32, 1st paragraph, last sentence - typo? Should it say 'Public
header tag which declares...'?

Page 32, 2nd paragraph, typos? Should it be 
'It is not necessary to use OPTIONS to discover support of a method,
it is possible to simply try it. If the receiver of the request does
not support the method it will respond with an error code indicating
that the method is either not implemented (501) or does not apply for
the resource (405). ..'

Page 32, 3rd paragraph, 3rd sentence, typo? Should it be:
'The amount of functionality  that a feature-tag represents can vary
significantly.' 

Page 32, 5th paragrapgh, typo? Should it be:
'Supported: The supported header is used..'

Page 34, 1st paragraph - typo? "SHOULD NOT" instead of "SHOULD not"?

Page 36, section 11.3, 2nd paragraph - the part that mentions that the
media shall start to play in sync.
Does a=group apply to this discussion? 
What is the scope of synchronization here - does it mean that packets
are sent on the different tracks according the the movie timeline that
is specified for all packets in the track?

Also, does 'setup.playing' apply to a specific URL or request/session
or is it a statement about the server in general?

Page 37, 3rd paragraph 'Since SETUP.." Isn't it likely that there
would be intermediary devices that would be interested in seeing the
b= line in the DESCRIBE response as well?

Page 38: liveness:
Should there be a mention of signs of liveness - no RTCP and no TCP
KeepAlive implies the session is gone?

Page 39, 4th paragraph, about the Range header. There is a mention
in section 3.5, 2nd paragraph, that 'now' should only be used for
live. This applies even in this section, right? I.e. the
recommendation to use npt=now- is for live?

Page 39, 5th paragraph - If there is no Range header in the request,
what should the response use as units for the Range header?
There was a suggestion that, if you do have the Range in npt, it can
be used, if not in the SETUP response there should be an indication of
the Range specifications that apply to the stream.

Also, it seems like the specification here is only for old
implementations, as newer implementations the Range header must be
sent in the request according to Page 41.


Page 39, last paragraph - typo? Should it be 
'After playing the desired range, the presentation is NOT
automatically paused, media delivery simply stops. A PAUSE must be
issued before another PLAY request can be issued."

Page 41, 2nd and 3rd paragraph: Here it says PLAY without range is not
ok - does this apply for stored content only? This seems to contradict
the text in the 3rd paragraph..

Page 43, 1st paragraph - is very confusing. Does this refer to the
case where the PAUSE request comes after the first occurence? Again it
seems like PLAY without a range header is ok...confusing.

Page 44, section 11.6, 1st paragraph, suggested rewording of the last
few sentences:

'However the session and any other media not torn down yet, remains,
and any valid request, e.g. PLAY or SETUP, can be issued. As an
optional feature a server MAY keep a session even after the last media
on it has been torn down by use of TEARDOWN with a media URI for the
last media. In this cases, when the server continues to maintain a
valid session, after processing a TEARDOWN request on this session, the
server MUST return the session identifier in the TEARDOWN response.'

Page 44, section 11.6, last paragraph: It would help to move these
sentences to earlier in the section - to explain the motivation for
what's discussed in the sentences above.


That's it for now,
geetha



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To: Magnus Westerlund <magnus.westerlund@ericsson.com>
From: Mark Baugher <mbaugher@cisco.com>
Subject: Re: [MMUSIC] Comments on draft-ietf-mmusic-sdescription-01.txt
Cc: Dan Wing <dwing@cisco.com>, Flemming Andreasen <fandreas@cisco.com>,
        IETF MMUSIC WG <mmusic@ietf.org>
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hi Magnus,
At 07:05 PM 8/20/2003 +0200, Magnus Westerlund wrote:
<...>
>>I think that a=crypto does everything that k= does.  My concern is that 
>>there might be an unforeseen need for extension for a particular transport.
>
>Not as currently specified. Can't k= be used to specify the use of DES on 
>complete RTP and RTCP packets? This is certainly something different than 
>what so far is defined. However I don't know if there is any need for 
>including this.

I didn't precisely say what I intended:  One can *extend* sdescriptions to 
do anything that k= does.  Adding DES or 3DES, for example, is done by 
adding a new crypto-suite.



>>I suppose it is but it would be a very brittle way to define the crypto 
>>suite, which *may be* orthogonal to the transport.  In the first 
>>submission of this work, we expected the crypto suite to be transport 
>>independent.  The reason it is not is because we have seen examples of 
>>the crypto suite being tailored and optimized for a particular 
>>transport.  For example, AES counter mode for SRTP re-uses the RTP 
>>sequence number to obviate the need to add a nonce to the header and save 
>>precious bits on low bandwidth links.  IPsec decided to put a nonce into 
>>the IPsec transport header for AES counter mode - the IV field - rather 
>>than overload the sequence number.  For sdescriptions, it seemed to us to 
>>be safer to not try and develop a single set of crypto suites for all 
>>transports since the namespace could become unwieldy (e.g. 
>>AES-CTR-128-HMAC-SHA1-80-SRTP, AES-CTR-128-HMAC-SHA1-80-IPSEC).
>>You seem to be taking this to its logical conclusion of putting all 
>>transport-specific parameters into the crypto suite, but I don't think we 
>>want to do that.  For one thing, it will result in too many crypto 
>>suites.  A bigger problem is that some parameter support a range of 
>>values, such as the size of the MKI.
>
>What do you mean with transport here? Depending on what level you mean I 
>think that one must in fact make the crypto suits transport dependent.

Sorry, I meant transport (e.g. RTP/SAVP) in the SDP sense,
m=<media> <port> <transport> <fmt list>.
There is really no reason why AES-CM-128-HMAC-SHA1 could not serve multiple 
transports, in principle, because it uses AES counter mode and HMAC-SHA1, 
both are standard definitions.  But they can diverge:  SRTP permits a 
128-bit input key for deriving HMAC-SHA1 whereas IPsec requires a 160-bit 
input, IPsec uses a longer counter, etc.  There is no reason why future 
crypto suites, however, can't have identical definitions.  I think that 
this is preferable.

But we can't have identical definitions for the session parameters because 
we're talking about different SDP transport protocols, e.g. unlike SRTP, 
IPsec has no notion of a variable-length key index (probably owing to the 
differences between a kernel and application implementation), and of course 
no notion of SRTCP which can be encrypted or not encrypted.


>If you with transport means for example RTP and the used profile the 
>crypto suit must in fact be transport dependent as I think your example 
>crypto suits AES-CTR-128-HMAC-SHA1-80-SRTP, AES-CTR-128-HMAC-SHA1-80-IPSEC 
>visualize. Even if the same crypto algorithm is used by both SRTP and 
>IPSEC the other parameters are not identical I would assume, or are they? 
>There is need to know that IPSEC should be applied and that SRTP is not 
>needed. Therefore I would assume that the crypto suits must be in fact be 
>dependent on encryption method which might imply certain transports.
>
>What we are still discussing in fact are if the a=crypto attribute only 
>should be applicable for SRTP independent on lower layers that transport 
>them, or we should make it possible for future extensions that might apply 
>for usage with other crypto methods?

I want sdescriptions to be extensible to other SDP transports.  It 
definitely has to support extension with other crypto methods.



>>>So the point I try to make is that each crypto suit should define what 
>>>parameters it supports.
>>
>>We made a distinction of parameters that are specific to the key, these 
>>are the domain of the crypto suite, versus parameters related to the 
>>transport, these are the session parameters.
>>
>>put in a method indicator first or make the crypto suite carry implicit 
>>information about the method and the structure of the rest of the 
>>a=crypto line.
>
>
>>>>That seems to be more a problem with SRTP than -sdescriiptions-.
>>>>To resolve the synchronization problem you have to use a per-packet MKI
>>>>(which adds a few bytes to each SRTP packet) or the "from,to" mechanism
>>>>(which doesn't add bytes to each SRTP packet); both MKI and "from,to"
>>>>are described in SRTP itself.
>>>>Do you feel putting like the above paragraph might be useful in
>>>>-sdescriptions-?
>>>
>>>
>>>Yes, and explanation on how the parameters a=crypto are used for this 
>>>purpose.
>>
>>I think that a non-zero MKI explains it all.
>
>What about the usage of <From, To>? I couldn't found anything about it. 
>What I think is important is to resolve how a rekeying with this mechanism 
>will happen.

This can be done with the current definitions based on the sequence number, 
MKI, and lifetime.


>I don't think that the following section actually explains how to perform 
>the rekeying to a point of interoperability.
>
>Section 5.7:
>   "When an active SRTP session is rekeyed, this is indicated by sending
>    a new SDP.  Rekeying is done following the rules described for a
>    normal Offer/Answer exchange.  The Answerer can take this
>    opportunity to rekey the traffic it is sending, if the Answerer
>    desires.  During rekeying, the session parameters cannot be changed
>    and MUST NOT be specified in the Offer or the Answer. "
>
>I think that you must in fact clarify that the key for rekeying must be 
>connected to a MKI or "from,to" interval to indicate when valid.

I think Dan, Flemming and I need to discuss this further.



>>ok.  I think the outstanding problem that we need to resolve with the WG 
>>is to what degree we generalize a=crypto beyond the SRTP transport.
>
>Agreed, I hope that some other people can give there opinion about this.

I hope so too.  As it stands now, this I-D is a framework for all 
transports but with definitions only for SRTP.  We feel we can generalize 
O/A for all transports.  But the session parameters are SRTP only as are 
the cryptographic-key parameters.  If this is okay with the WG, then there 
is no problem for us.

Thanks, Mark



>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
>
>
>_______________________________________________
>mmusic mailing list
>mmusic@ietf.org
>https://www1.ietf.org/mailman/listinfo/mmusic



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Subject: RE: [MMUSIC] RTSP and mobility
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Hi Magnus, Tom, Henning and Ashutosh,

I agree with you about some "Application Mobility" paragraphs=20
in the RTSP core specification and I also agree on the change=20
from MAY to MUST in section 10.4.

According to other commitments, I'll try to look through the =
specification=20
searching any more conflicts or limitations.

Best Regards,
Enrico



-----Original Message-----
From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
Sent: mercoled=EC 20 agosto 2003 9.36
To: Tom Marshall
Cc: Henning Schulzrinne; Enrico Balatti; mmusic@ietf.org
Subject: Re: [MMUSIC] RTSP and mobility


Hi Tom,

Yes, you are correct that the current MAY would need to be changed to=20
MUST to always allow it. Performing this change may be a bit to much=20
change. I think that it would be better to explain why support of this=20
functionality is good, based on that it makes Application layer mobility =

possible. If there is need to learn the support for this, we should=20
define a feature tag for it.

Is anybody willing to look through the specification to see if there are =

any more conflicts or limitations and provide a draft informational text =

about this?

Best Regards

Magnus



Tom Marshall wrote:

> In order to make this work, don't you also need to s/MAY/MUST/ in the
> first
> paragraph of 10.4?  This would seem to be a controversial change.
>=20


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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Hi Mark,

To try to summarize the last mails in the discussion:

1. You are going to create a framework that may be used in the future 
also for other transports then RTP/SAVP. However you will only define 
crypto suits that apply for RTP/SAVP.

2. There is need to resolve how "a=crypto" is used for doing rekeying in 
a synchronized way.

3. Input from others on these issues (especially the first) are desired.


I have some comments upon the first thing. I think that this intention 
is a good one. In respect to the last version I think there are some 
work on clarifying how this will be done. The big issue that needs to be 
resolved is:
- How to connect a crypto suit to transport? Either you need to define a 
crypto suit for each transport, or in cases where there might be 
multiple transport that can use the crypto suit ensure that the 
transport context will be enough to resolve this.

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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Subject: Re: [MMUSIC] RTSP and mobility
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Hi Enrico,

Enrico Balatti wrote:

> Hi Magnus, Tom, Henning and Ashutosh,
> 
> I agree with you about some "Application Mobility" paragraphs 
> in the RTSP core specification and I also agree on the change 
> from MAY to MUST in section 10.4.

Please not jump ahead of things. Any change of this MAY to a MUST is in 
need of more extended discussion and consideration. I think I am against 
  changing this due to backwards compatibility reasons. If we would 
change it and a updated client assumes that all server support this 
functionality, while all ones does not, it will create problems. In my 
opinion the most reasonable solution of this problem is to clearly 
motivate why you should implement this functionality. This can be done 
in a satisfactory way by creating a application mobility section. To 
ensure that a client can determine the servers support of the mobility 
features a feature tag to use in supported header can be defined. That 
way the client can determine if it will work before attempting it.

> 
> According to other commitments, I'll try to look through the specification 
> searching any more conflicts or limitations.

Very Good!


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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Magnus Westerlund wrote:

> Hi Mark,
>
> To try to summarize the last mails in the discussion:
>
> 1. You are going to create a framework that may be used in the future
> also for other transports then RTP/SAVP. However you will only define
> crypto suits that apply for RTP/SAVP.
>

We probably need to elaborate on this one a bit:

The original intent with the draft was to define a mechanism that can be
used to describe and negotiate (using offer/answer) SRTP parameters for SRTP
media streams. As always, it is desirable to define a general mechanism to
do this rather than just a specific one, however I do have some concerns
about scope explosion here as we are not trying to define a mechanism that
can be used to negotiate security parameters for any kind of stream, e.g.
establish a TLS session or setup an IPSec SA. If it turns out the mechanism
can be used to do that, then great, however this in itself should not be a
goal. I think a reasonable constraint is that the security parameters must
either be such they than just be advertised (e.g. for SAP) or they can be
negotiated in a single offer/answer exchange. Anything that doesn't fit
within those constraints is out of scope.

As far as the security mechanisms are concerned, we need to make a
distinction between:
1. The "method" being used for security and any parameters associated with
that
2. The crypto-suites being used and any parameters associated with that.

By method, I mean for example SRTP, RFC 3550 Section 9, etc. Not all methods
can be used with all transports, for example RFC 3550 Section 9 cannot be
used for RTP/SAVP, but there could conceivably be more than one method used
for a particular transport. Where negotiation is involved, it would be
desirable to be able to specify different methods, at least for the initial
offer/answer exchange. For subsequent offer/answer exchanges it could be
more complicated as the media stream packets themselves may not necessarily
have an indication of which method is being used and a graceful switchover
thus becomes complicated (unless you use a different port).

The crypto-suites will be interpreted within the context of a method (as you
indicated earlier) and will have certain parameters associated with them.

Going back to offer/answer, we can probably come up with a set of generic
rules that apply to all methods and crypto-suites, however I suspect that
for most, there will be a need to further define method-specific and maybe
even crypto-suite-specific offer/answer rules, with SRTP being one such
case. Thus, the draft will define a framework for advertising and
negotiating security parameters and provides the details for how to do it
for SRTP, but in order to use it for anything other than SRTP, additional
specification would be needed (in a seperate RFC).

How does this sound ?

>
> 2. There is need to resolve how "a=crypto" is used for doing rekeying in
> a synchronized way.
>

Agreed.


>
> 3. Input from others on these issues (especially the first) are desired.
>
> I have some comments upon the first thing. I think that this intention
> is a good one. In respect to the last version I think there are some
> work on clarifying how this will be done. The big issue that needs to be
> resolved is:
> - How to connect a crypto suit to transport?

I think crypto-suites should be defined within the context of the method
(which includes transport).

> Either you need to define a
> crypto suit for each transport, or in cases where there might be
> multiple transport that can use the crypto suit ensure that the
> transport context will be enough to resolve this.
>

I prefer being explicit and defining it for each method/transport; it also
makes the name space issue a lot cleaner.

-- Flemming



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Today's Topics:

   1. [ rtspspec-Bugs-792973 ] Clarify changing of transport parameters (SourceForge.net)

--__--__--

Message: 1
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Fri, 22 Aug 2003 00:21:19 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-792973 ] Clarify changing of transport parameters

Bugs item #792973, was opened at 2003-08-22 09:21
Message generated for change (Tracker Item Submitted) made by Item Submitter
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=792973&group_id=23194

Category: None
Group: Fix DS - Needs clarificiation
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: Clarify changing of transport parameters

Initial Comment:
The use of SETUP during a session to change the
transprot parameters should be clarified in several
aspects:

- Clarify the text about in which states one can do it.
Also the result of failed requests impact, etc. 

- Clarify the adavantage of this by adding a
application mobility section as motiviation. 

- consider if a feature tag for this is necessary?

- Consider if there are any problems of allowing all
parameters to be changed. 


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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=792973&group_id=23194




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Hi Geetha,

And thanks for the review. I have fixed the typos, however there are a 
few things that may need discussion.

If you are replying to any particular section could you please remove 
everything except that topic and change the subject. Please also one 
topic per email.

Geetha Srikantan wrote:

> I meant to send this in earlier..here it is now..
> 
> Questions and comments (some typos too):
> 
> Page 10: Transport Negotiation
> Can transport parameters of an existing session be changed, mid-stream?
> For instance, is it possible to change the server's RTP/RTCP ports
> mid-stream? 

Yes, If the server implements it and allows it. As seen by the 
application mobility discussion this functionality can be used for this.

> If so, exactly how does the transition to using the new ports happen? Do some 
> mechanisms need to be specified to ensure synchronization etc after the cutover?

Upon the successful response to a SETUP request to change the transport 
parameters the server will start using the new parameters. The client to 
avoid losing any packets should keep the old ports/transport open for a 
couple of seconds. The client should also have opened its ports for the 
new transport prior to sending the request. So I don't believe there are 
any synchronization problem.

> If not, it would be good to specify the states in which transport negotiation 
> can be done.

This is currently specified in the state machine. If supported is must 
be possible in READY state, and MAY be supported in PLAY state. This can 
probably be a clarified and simplified by tightening the requirements a 
bit. The text in section 11.3 can probably be improved to be more clear 
about what is allowed in this respect.

> 
> Page 10, last paragraph section 1.5 - are 'stream identifiers' the
> same as 'session identifiers'? If so, it might help to use the same
> term throughout the document.

In some sense. The occurrence here of stream identifiers should probably 
be replaced by session ID and stream and control URLs.
> 
> Page 14, section 2 - typo after last paragraph 'b'?

Definitely, probably a result of the document generation.

> 
> Page 15, 1st paragraph, last sentence - typo?

Definitely a bad sentence. Is now corrected.

> 
> Page 15, 3rd paragraph, it might help to rephrase to say that FQDNs
> are preferred, especially for IPv6, (maybe even 'required for IPv6 ?)
> however IP addresses should also be supported (for backward
> compatibility).

I think we have a good level now. First of all the requirement to use 
FQDNs to handle IPv6 hosts only applies to RFC 2326 servers, not the 
ones in accordance with this specification. There are also reasons why 
one might use literal IP addresses. For example in destination fields, a 
FQDN name that resolve only to the targeted host may not exist. Also 
there are some performance issues if proxies needs to do DNS requests 
for all messages it is forwarding.

> 
> Page 18, section 3.7, 2nd sentence - typo 'in' appears twice.

Fixed

> 
> Page 20, section 5 - adding Expires header?
> Page 27, section 7.1.2 - add Expires header?

The Expires header is a Entity header, see section 8.

> 
> Page 30, section 9.3, 3rd paragraph, last sentence - typo?
Fixed

> Page 31, section 9.3, Is play.basic a superset of *not* con.persistent?

play.basic is not a superset of con.peristent. A RTSP entity supporting 
play.basic must handle non-persistent connections. The con.peristent is 
defined to allow any service requiring a persistent connection to use 
this in a Required header. However looking at this I would in fact 
propose that this is removed. I can't see any clear need and therefore 
it is unnecessary, and if it is need in the future it can simply be 
define in relation to the necessary user of the feature tag.

> 
> Page 31, section 9.4, 1st paragraph, 4th sentence s/must/should/?
> I.e. 'This requires that any RTSP URL pointing tat a IPv6 host should
> use fully qualified domain name and not a IPv6 address.'
> 
It is a informal must, not a normative one. This paragraph is only for 
information on how to make this work if the other party does not support 
  the updated specification. Are there any problem with this?

> Page 31, section 10, 1st paragraph, 2nd sentence - typo 'too'?
Fixed

> 
> Page 31-32, section 10,
> Should responses to OPTIONS be cached? 
Maybe, it is actually a problem of what OPTIONS is used for. As long it 
is only used for determine server capability it is not a problem. 
However if it is used to maintain session liveness, i.e. session ID 
present, no. Also depending on the URI and the possibilities to cache ma 
y vary. Needs further consideration.

If so, is there an expiry on
> this and any specification on how to refresh?
This is a tough problem. I don't think there are. The Expires header is 
an entity header, but the text in RFC 2616 seems to mean the whole 
request. SO this is something that needs consideration.

> Are there rules about determining capability in the presence of a
> proxy?
Yes, I would say so. Options can either be targeted at the next hop. Or 
to the server by including host name. However I think the usage of 
supported header will not reveal anything about the proxy. Also 
something that might need some more consideration.

> Does this section specify all that is required for each endpoint to
> determine capability?
Actually this section is totally informal. The specification of the 
operations is actually dispersed around the document. See the options 
method and the headers for specification.

> 
> Page 32, 1st paragraph, last sentence - typo? Should it say 'Public
> header tag which declares...'?
Yes, fixed.

> 
> Page 32, 2nd paragraph, typos? Should it be 
> 'It is not necessary to use OPTIONS to discover support of a method,
> it is possible to simply try it. If the receiver of the request does
> not support the method it will respond with an error code indicating
> that the method is either not implemented (501) or does not apply for
> the resource (405). ..'
> 
Yes, fixed.

> Page 32, 3rd paragraph, 3rd sentence, typo? Should it be:
> 'The amount of functionality  that a feature-tag represents can vary
> significantly.' 
Yes, fixed.
> 
> Page 32, 5th paragrapgh, typo? Should it be:
> 'Supported: The supported header is used..'
Yes, fixed.
> 
> Page 34, 1st paragraph - typo? "SHOULD NOT" instead of "SHOULD not"?
Yes, fixed.
> 
> Page 36, section 11.3, 2nd paragraph - the part that mentions that the
> media shall start to play in sync.

This is only intended to resolve the synchronization problem when doing 
SETUP in PLAY state. The newly setuped media must be playing in synch 
with already present media, otherwise this would break the aggregation 
policy we apply.
> Does a=group apply to this discussion? 
I have not considered if a=group has any impact at all. I would not 
thinks so, because such a SETUP as described in the second paragraph in 
it self requires that the server plays out the media in synch.

> What is the scope of synchronization here - does it mean that packets
> are sent on the different tracks according the the movie timeline that
> is specified for all packets in the track?
The session has an internal clock that runs during play state. When a 
media is added to the session that media must also play out according to 
that clock.

0----5----10---15---20---25 Session playout time
0xxxx5xxxx10xxx15xxx20xxx25 First media stream
           10***15***20***25 Second media stream starts at 10 seconds
                             into the media.

In this example a SETUP request to add a second media stream is received 
at time 10. Then the second media will start playout from time point 10.

I hope this clarifies this. Are there need to improve the text in that 
chapter? Is the functionality (SETUP during PLAY state) needed, or could 
we throw it away. In my opinion it is not a really necessary and such a 
extension could anyway be added later. In my eyes it will anyway need 
extensions.
> 
> Also, does 'setup.playing' apply to a specific URL or request/session
> or is it a statement about the server in general? 

A Feature tag is used in a context, depending on the context it can 
either relate to a session, media stream, or the complete server.

> 
> Page 37, 3rd paragraph 'Since SETUP.." Isn't it likely that there
> would be intermediary devices that would be interested in seeing the
> b= line in the DESCRIBE response as well?

This is probably something the original authors should answer on. 
However in my opinion the design is made to allow an middle box to 
understand that the individual flows exist and which protocols they use. 
It is not designed to allow for conveying QoS properties for the stream. 
Anyway I don't think that b= is of much more use then to provide some 
basic limits. A media stream today is not of fixed bit-rate, it may very 
well be adapted to available resources for best effort traffic. For 
streams with QoS agreements they will have there own policy on policing, 
control and admission control that does not belong in RTSP.

> 
> Page 38: liveness:
> Should there be a mention of signs of liveness - no RTCP and no TCP
> KeepAlive implies the session is gone?

How do you interpret the second last paragraph and bullet list of 
section 11.3? There are also a reference to section 13.37 that discuss 
this further.

 From the text of section 11.3:

"Signs of liveness for a RTSP session are:

      + Any RTSP request from a client which includes a Session header
        with that session's ID.

      + If RTP is used as a transport for the underlying media streams,
        an RTCP sender or receiver report from the client for any of the
        media streams in that RTSP session."

> 
> Page 39, 4th paragraph, about the Range header. There is a mention
> in section 3.5, 2nd paragraph, that 'now' should only be used for
> live. This applies even in this section, right? I.e. the
> recommendation to use npt=now- is for live?

Yes. Is it not understandable that this sentence is in relation to the 
context of the previous sentence? Have you proposal for improvements of 
the text?

> 
> Page 39, 5th paragraph - If there is no Range header in the request,
> what should the response use as units for the Range header?
> There was a suggestion that, if you do have the Range in npt, it can
> be used, if not in the SETUP response there should be an indication of
> the Range specifications that apply to the stream.

Yes, this is a bit unclear. I propose it to clarify it in the following way:

For a on-demand stream, the server {\MUST} reply with the actual
range that will be played back.  This may differ from the
requested range if alignment of the requested range to valid frame
boundaries is required for the media source.  If no range is
specified in the request, the start position {\SHALL} still be
returned in the reply.  The unit of the range in the reply
{\SHOULD} be the same as that in the request. If no range header
was present in the request the server {\MAY} use any of the
supported formats, however it is {\RECOMMENDED} that a format the
client has shown support for is used. If the media part of an
aggregate has different lengths the {\PLAY} request and any
\header{Range} {\SHALL} be performed as long it is valid for the
longest media. Media will be sent whenever it is available for the
given play-out point.

> 
> Also, it seems like the specification here is only for old
> implementations, as newer implementations the Range header must be
> sent in the request according to Page 41.

No, it is for the updated specification. There is no requirement that a 
client includes a range header in a request. The range header MUST be 
present in responses. Please quote the section you have interpreted as 
requiring Range in PLAY requests.

> 
> 
> Page 39, last paragraph - typo? Should it be 
> 'After playing the desired range, the presentation is NOT
> automatically paused, media delivery simply stops. A PAUSE must be
> issued before another PLAY request can be issued."

Yes, type except that I would like to keep "request" in the last 
sentence after pause:

"A PAUSE request must be issued before another PLAY request can be issued."

> 
> Page 41, 2nd and 3rd paragraph: Here it says PLAY without range is not
> ok - does this apply for stored content only? This seems to contradict
> the text in the 3rd paragraph..

My intention with the two following paragraphs:

   "The queued play functionality described in RFC 2326 [21] is removed
    and multiple ranges can be used to achieve a similar performance. If
    a server receives a PLAY request while in the PLAY state, the server
    SHALL respond using the error code 455 (Method Not Valid In This
    State). This will signal the client that queued play are not sup-
    ported.

    The use of PLAY for keep-alive signaling, i.e. PLAY request without a
    range header, has also been depreciated.  Instead a client can use,
    PING, SET_PARAMETER or OPTIONS for keep alive. A server receiving a
    PLAY keep alive SHALL respond with the 455 error code."

Is that it is informs that while in PLAY state, a PLAY request is not 
allowed. This was in RFC 2326 used for two different purposes depending 
on if range was present or not (queued play, and keep alive). These two 
paragraphs try to explain what to do if one would receive a request that 
tries to perform such an operation. I will add a "in PLAY state" to the 
second paragraphs first sentence to be clearer.

> 
> Page 43, 1st paragraph - is very confusing. Does this refer to the
> case where the PAUSE request comes after the first occurence? Again it
> seems like PLAY without a range header is ok...confusing.

I guess it is not that confusing clearing out the previous confusion.

> 
> Page 44, section 11.6, 1st paragraph, suggested rewording of the last
> few sentences:
> 
> 'However the session and any other media not torn down yet, remains,
> and any valid request, e.g. PLAY or SETUP, can be issued. As an
> optional feature a server MAY keep a session even after the last media
> on it has been torn down by use of TEARDOWN with a media URI for the
> last media. In this cases, when the server continues to maintain a
> valid session, after processing a TEARDOWN request on this session, the
> server MUST return the session identifier in the TEARDOWN response.'
> 
Yes, you where correct that the sentences need to be reworded. However I 
was not completely happy with your proposal either. So I propose the 
following:

The {\TEARDOWN} request stops the stream delivery for the given
URI, freeing the resources associated with it. If the URI is the
aggregated control URI for this presentation, any RTSP session
identifier associated with the session is no longer valid. The use
of "*" as URI in {\TEARDOWN} will also result in that the session
is removed independent of the number of media that was part of
it. If the URI in the request was for a media within an aggregated
session that media is removed from the aggregate. However the
session and any other media not torn down yet, remains, and any
valid request, e.g. PLAY or SETUP, can be issued. The server
{\MAY} keep the session when the last media is torn down with a
{\TEARDOWN} request specifying the media URI. The server {\SHALL}
always include in a {\TEARDOWN} response the session header, if a
valid session state exist after the processing of the request.

However I have logged a bug suggesting that we remove the Ready-NM 
state. That way we can get rid of the "*" teardowns that I feel will not 
work satisfactory through proxies.

> Page 44, section 11.6, last paragraph: It would help to move these
> sentences to earlier in the section - to explain the motivation for
> what's discussed in the sentences above.

I don't think this provides any motivation. However it might be 
beneficial to change the the complete section into something that is 
more readable and more clearly explains things. One can also discuss the 
need for having TEARDOWN during PLAY state. Please provide input on this 
topic.




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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* Apologies if you receive multiple copies of this announcement *

LANOMS'03
3rd Latin American Network Operations and Management Symposium
Iguassu Falls, Brazil
September 4-6, 2003

IEEE ComSoc CNOM Technically Co-sponsored

http://www.inf.ufpr.br/lanoms


CALL FOR PARTICIPATION
----------------------

Registration is now open.  Please  register soon: there  is a limit of
150 participants.

Preliminary Program is now available at:

http://www.inf.ufpr.br/lanoms/program.shtml

Invited Talks:

Evolution of IP/eBusiness Management - New Paradigm and SLA
Masayoshi Ejiri, Fujitsu, Japan

Management Challenges in an on demand environment
Germ=E1n Goldszmidt, IBM, U.S.A.

Integrating  Latin American   and   European  Research and   Education
Networks through the Alice Project
Michael Stanton, RNP, Brazil

Tutorials:

Security and Mobility Management in the Embedded Internet
Guy Pujole, Universite de Versailles, France

Management for Next Generation of Wireless Networks and Services
Mehmet Ulema, Manhattan College, New York, U.S.A.

Over-the-air Device Management
Paul Oommen, Nokia Research Center, U.S.A.

Technical Sessions:

Security Management
Management of Mobile/Wireless and Ad-Hoc Networks
Management Platforms and Frameworks
Management of Networks and Services
Network and Systems Monitoring
Management of Service Level Agreements and Quality of Service

The symposium will have  only single-track Technical Sessions. Details
can be found at:

http://www.inf.ufpr.br/lanoms/technical_sessions.shtml
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This is an automatic digest of bugs submitted and changed on the RTSP bug tracker.  To see a list of bugs, visit:

http://sf.net/projects/rtspspec

When replying, please make your Subject line more specific than "Re: Rtspspec-bugs digest"



Today's Topics:

   1. [ rtspspec-Bugs-795295 ] TEARDOWN using * and the Ready-nm state (SourceForge.net)
   2. [ rtspspec-Bugs-795314 ] Caching of OPTIONS request (SourceForge.net)

--__--__--

Message: 1
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Tue, 26 Aug 2003 05:08:24 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-795295 ] TEARDOWN using * and the Ready-nm state

Bugs item #795295, was opened at 2003-08-26 14:08
Message generated for change (Tracker Item Submitted) made by Item Submitter
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=795295&group_id=23194

Category: None
Group: Fix for DS - Need WG Input
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: TEARDOWN using * and the Ready-nm state

Initial Comment:
The 20 february 2002 we discussed

http://rtsp.org/bug506412, for conclusion of that

discussion see minutes:

http://rtsp.org/2002/telecon/minutes20Feb.html



However to me it seems that the need for state Ready-nm

is not that great. Also the use of "*" as a URL will

not work when proxies are present, unless requirements

are put on them to maintain certain state in regards to

sessions. The use of the presentation URL also is

problematic as some sessions may not have a defined

session URL, like for example for sessions intended to

be used in non-aggregted mode. 



Therefore my suggestion is that the Ready-nm state is

removed and that teardown using "*" is dissallowed.

Comments please. 

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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=795295&group_id=23194


--__--__--

Message: 2
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Tue, 26 Aug 2003 05:27:47 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-795314 ] Caching of OPTIONS request

Bugs item #795314, was opened at 2003-08-26 14:27
Message generated for change (Tracker Item Submitted) made by Item Submitter
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=795314&group_id=23194

Category: None
Group: None
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: Caching of OPTIONS request

Initial Comment:
Geetha Srikantan thought of the question if OPTIONS

request are cachable, and if, how to determine the

expiration of such cached command. 



My answer

Maybe, it is actually a problem of what OPTIONS is used

for. 

As long it is only used for determing server capability

it is not a problem. However if it is used to maintain

session liveness, i.e. session ID present, no. Also

depending on the URI and the possibilities to cache ma

y vary. Needs further consideration.



This is a tough problem. I don't think there are. The

Expires header is an entity header, but the text in RFC

2616 seems to mean the whole request. So this is also

something that needs consideration. 



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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=795314&group_id=23194




End of Rtspspec-bugs Digest


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From: Thomas Zeng <zeng@PacketVideo.COM>
To: "'mmusic@ietf.org'" <mmusic@ietf.org>
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Subject: [MMUSIC] Interpreting RFC2327: SDP whitespace problem in "s=" line
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Hello All:

During recent interop testing, one issue has come up regarding whether it is
valid to have "s=[single space]" in
an SDP as the subject of a presentation.

In section 6 of RFC2327 (see
http://www.ietf.org/rfc/rfc2327.txt?number=2327):

   An SDP session description consists of a number of lines of text of
   the form <type>=<value> <type> is always exactly one character and is
   case-significant.  <value> is a structured text string whose format
   depends on <type>.  It also will be case-significant unless a
   specific field defines otherwise.  Whitespace is not permitted either
   side of the `=' sign. In general <value> is either a number of fields
   delimited by a single space character or a free format string.

Taken off context, this paragraph seems to be prohibiting
'<type>[space]=<value>'  or
'<type>=[space]<value>' .  However, we don't think the intent was to place
restrictions on <value>.  If <value> is a free-format string, then it should
be able to start with a space. (And this is what the ABNF definition says,
for the value in the 's=' case).

We checked
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-new-13.txt as
well, which clearly states that
if there is no meaningful subject, a single space should be used in "s="
line, yet it apparently has contracdictory statement in section 5.

We hereby would like to confirm with this mailing list, that "s=[single
space]" is a legal subject line. 
We would also appreciate clarification in
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-new-13.txt , 
section 5, to indicate that:

   " An SDP session description consists of a number of lines of text of
   the form:

      <type>=<value>

   where <type> MUST be exactly one case-significant character and
   <value> is structured text whose format depends on <type>.  In
   general <value> is either a number of fields delimited by a single
   space character, or a free format string.   [modified text]  Whitespace
MUST NOT be used either side of the "=" sign, except if the free    format
<vallue> string  begins with a space or contains a single space. An exampel
is: "s=[single space]", which is a   legal SDP line."

Thanks in advance for your feedback.

thomas
PacketVideo Corp.

www.pv.com




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From: Tom Marshall <tmarshall@real.com>
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Subject: [MMUSIC] SDP b= clarification
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In RFC3556, the example SDP shows multiple b=3Dlines in the media sections.=
=20
In RFC2327 (and the updated draft sdp-new-13), the b=3D line is specified
without any corresponding note that the line may be repeated.  I've always
assumed this meant exactly one instance was allowed.  If that is the case,
the example SDP in 3556 is noncompliant.  Since multiple b=3D lines are
presumedly desirable, I think the sdp-new draft should be updated to reflect
this.

--=20
Majority, n.:
        That quality that distinguishes a crime from a law.

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From: Colin Perkins <csp@csperkins.org>
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Hi,

On Thursday, Aug 14, 2003, at 17:08 Europe/London, Fredrik Lindholm 
(KI/EAB) wrote:
...
>> I have a number of comments on this draft:
>>
>>   - The ABNF in Sections 2.1 and 2.2 is contradictory. The key 
>> management
>>     data is defined as "text" in one section, and "quoted-string" in 
>> the
>>     other, yet these two types have different ranges of valid 
>> characters
>>     (and both are more liberal than the explanatory text, which 
>> implies
>>     that only base-64 encoded data is allowed). Similar for the 
>> protocol
>>     identifier: the definitions of "non-ws-string" in SDPnew and 
>> "token"
>>     in RTSP are not consistent.
>
> This is as a result from the comments we received before, when we 
> actually
> had consistent strings. At that time the comment was that we should use
> what was used in SDP and RTSP respectively (and the recommendations was
> to use non-ws-string and token). Please propose what to use.

If base-64 encoded text is a requirement, be explicit about that. 
Whatever is chosen, you will need to give an explicit ABNF definition, 
and include a comment stating that it is consistent with 
"non-ws-string" in the SDP section, and consistent with "token" in the 
RTSP section.

>>   - The draft repeatedly mentions Kerberos, and indicates that 
>> Kerberos
>>     tickets may be included in key management messages, but does not
>>     specify how this is to be achieved. Should either specify how to 
>> use
>>     Kerberos, or remove all references to it.
>
> It was included as a result of a comment that we should give examples
> of more than one protocol that might utilize the framework. It is not
> our intention to specify how it can be used. We can agree that there 
> are
> places where we could remove the reference. However, it might be good
> at least to keep the most general one in Section 3 to provide an 
> overview
> of the possibilities.

Including examples of other protocols which might use the framework in 
future is appropriate. The problem is that the draft states that 
Kerberos is supported ("Other protocols MAY use the described attribute 
and header, e.g. Kerberos"). It should say that the framework is 
intended to support other key management protocols, for example 
Kerberos, but that this is subject to future standardisation.

Be clear what is a possible future extension, and what is allowed today.

>>   - The second bullet point on Page 6 mentions that other parameters 
>> may
>>     need to be specified for certain key management protocols, but 
>> does
>>     not specify how an implementor knows if a particular key 
>> management
>>     protocol uses additional parameters. Suggest this is listed as 
>> part
>>     of the IANA registration for a particular key management protocol?
>
> I don't see why IANA should handle this. However, it sounds reasonable
> that it should be documented in the key management draft (or a 
> companion
> draft).

Each new key management protocol that uses this framework needs to 
provide this information. The way of ensuring this is to mandate that 
all new protocols that use the framework are registered with IANA, and 
that IANA require this information be provided.

>>   - Section 4 needs to provide strict guidance about the parameters 
>> and
>>     specifications needed for new key management protocols. The 
>> interface
>>     between the key management protocol and this framework should be 
>> made
>>     explicit.
>
> That is what we have tried to address. Please give examples of what
> you think is lacking of that?

The draft asks for "the information the key management needs from SDP 
and RTSP", but doesn't indicate how that information is to be 
specified. For example, does one extract certain fields from the SDP? 
Is some information taken from an RTSP method or header? Does this data 
have to be put into a canonical form? etc. There needs to be almost a 
"checklist" the protocol designer and IANA can look at, to decide if 
the new protocol is completely specified.

>>   - In section 5, how is a beat-down attack prevented when the 
>> attacker
>>     wishes to beat-down to no security? What is the correct response 
>> to
>>     an answer which suggests no security?
>
> That would be to terminate/abort the session setup.

Okay, but that needs to be specified in the draft.

>> The main issue behind these seems to be the desire to define new
>> parameters which can be left open for future uses. As was mentioned in
>> the context of the "a=gpmd:" attribute, this is not acceptable: there
>> needs to be a precise definition of the semantics of any new SDP
>> attribute/RTSP header before the draft can advance.
>
> Ok, but doesn't fixing the above issues make this problem go away?

Hopefully. The above issues were specific instances of a general 
problem with the draft.

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


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