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From: "Dan Wing" <dwing@cisco.com>
To: "NAKAMURA, Hidefumi" <nakamura.hidefumi@lab.ntt.co.jp>
Cc: "James M. Polk" <jmpolk@cisco.com>, "IETF MMUSIC" <mmusic@ietf.org>
Subject: RE: [MMUSIC] IPv4 and IPv6 interworking
Date: Wed, 1 Oct 2003 11:58:52 -0700
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Thanks, I'll take a look at using that to our advantage with
IPv4/v6 interworking.

-d

> -----Original Message-----
> From: NAKAMURA, Hidefumi [mailto:nakamura.hidefumi@lab.ntt.co.jp]
> Sent: Monday, September 29, 2003 9:44 PM
> To: Dan Wing
> Cc: James M. Polk; IETF MMUSIC
> Subject: Re: [MMUSIC] IPv4 and IPv6 interworking
> 
> 
> Hi.
> 
> If you establish a session with SIP, I think RFC3312 mechanism may help
> to solve this problem though some extension will be needed.
> (and also in case that the SIP messages are exchanged in IPv4 packets)
> 
> RFC3312 describes how to negotiate SDPs between two endpoints without
> sending independent INVITE messages twice though it is for the QoS
> negotiation.
> 
> > > At 01:14 PM 9/29/2003 -0700, Dan Wing wrote:
> > > >Let's say a Alice capable of sourcing/sinking both IPv4 and 
> > > >IPv6, and Alice
> > > >wants to establish a session with Bob.  Alice doesn't know, a 
> > > >priori, if Bob
> > > >supports IPv4 or IPv6.  Let's also say that Bob only supports IPv4.
> > > 
> > > What's the normal response of an IPv4 device that receives an 
> > > IPv6 packet? 
> > > This seems like a good place to start...
> > 
> > Yes, but it didn't receive an IPv6 packet -- it receives a SIP 
> > packet from its IPv4-only SIP proxy, but the SDP describes an 
> > IPv6 session.
> > 
> > The normal reaction to an SDP session you can't support is well
> > understood.  I'm asking if that's all folks expect out of IPv4/
> > IPv6 interworking.
> > 
> > -d
> > 
> > > >Does Alice have to send an Offer with IPv6, have Bob reject the 
> > > Offer, and
> > > >then Alice has to send another Offer for IPv4?
> > > >
> > > >RFC3266 describes SDP for IPv6, but it appears it isn't possible to
> > > >simultaneously offer both IPv4 and IPv6 in SDP.
> > > >
> > > >-d
> > > >
> > > >
> > > >_______________________________________________
> > > >mmusic mailing list
> > > >mmusic@ietf.org
> > > >https://www1.ietf.org/mailman/listinfo/mmusic
> > > 
> > > 
> > > cheers,
> > > James
> > > 
> > >                                 *******************
> > >                  Truth is not to be argued... it is to be presented
> > > 
> > 
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www1.ietf.org/mailman/listinfo/mmusic
> 
> -- 
> NAKAMURA, Hidefumi 
> NTT ServiceIntegration /NetworkServiceSystems Labs.
> tel: +81(422)59 3904; fax:+81(422)60 4012
> ---
> 

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From: "Dan Wing" <dwing@cisco.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
Cc: "IETF MMUSIC" <mmusic@ietf.org>
Subject: RE: [MMUSIC] IPv4 and IPv6 interworking
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Thanks, Jonathan.

-d

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Monday, September 29, 2003 11:55 PM
> To: Dan Wing
> Cc: IETF MMUSIC
> Subject: Re: [MMUSIC] IPv4 and IPv6 interworking
> 
> 
> The right way to do this is with the alt mechanism:
> http://www.ietf.org/internet-drafts/draft-camarillo-mmusic-alt-01.txt
> 
> in concert with ICE:
> http://www.ietf.org/internet-drafts/draft-rosenberg-sipping-ice-01.txt
> 
> THe alt mechanism allows Alice to tell Bob about both the v4 and v6 
> addresses, and the ICE mechanism (which will be revised as an mmusic 
> work item per the last ietf meeting) allows for connectivity in more 
> complex cases than the one you describe below. In particular, when 
> Alice is v6 only and there is a 6to4 nat.
> 
> -Jonathan R.
> 
> Dan Wing wrote:
> 
> > Let's say a Alice capable of sourcing/sinking both IPv4 and 
> IPv6, and Alice
> > wants to establish a session with Bob.  Alice doesn't know, a 
> priori, if Bob
> > supports IPv4 or IPv6.  Let's also say that Bob only supports IPv4.
> > 
> > Does Alice have to send an Offer with IPv6, have Bob reject the 
> Offer, and
> > then Alice has to send another Offer for IPv4?
> > 
> > RFC3266 describes SDP for IPv6, but it appears it isn't possible to
> > simultaneously offer both IPv4 and IPv6 in SDP.
> > 
> > -d
> > 
> > 
> > _______________________________________________
> > 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, 02 Oct 2003 16:47:27 +0200
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [MMUSIC] RTSP: Proposal to remove ready-nm state
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Hi,

I have suggested and got support in yesterdays discussion to remove the 
ready-nm state from the RTSP state machine and the functionality 
required to make this state work. See below link for further comments on 
this.
http://rtsp.org/bug795295

Unless anyone has any comment about this within a week or so I will 
start removing it from the text.


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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Subject: [MMUSIC] RTSP: Proposal to remove SETUP and TEARDOWN of individual media streams
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Hi,

Another issue that was discussed on yesterdays telecon was the removal 
of the part specifying that usage of setup and teardown on individual 
media stream in PLAY state is possible. This does not effect the 
functionality for:

- SETUP and Teardown in ready state
- The usage of SETUP to change transport parameters.
- Usage of aggregated Teardown in play state to end session.

The removal of this functionality will be done in such way that it will 
be possible in the future to specify this as extension functionality.

If no comments are received on this within a week I will start making 
these changes. And comments for keeping this functionality should state 
a strong use cases for this.

See also:
http://rtsp.org/bug810666

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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Subject: [MMUSIC] RTSP: Lack of aggregated control URI means non-aggregated control
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Hi,

In yesterdays telecon the issue around http://rtsp.org/bug640272
was discussed. The remaining issue is if the lack of an aggregated 
control URI in SDP means that a session "SHOULD" or "SHALL" be setup in 
non-aggregated control. Both viewpoints was expressed in Vienna and the 
the discussion needs closure. My personal view is that a SHOULD is 
sufficient. Therefore I have drafted the below text for the SETUP 
section of the RTSP draft. The URI rules are based on the ones Aravind 
drafted.

"If an Aggregate control URI is not specified in the session
description, it is probably a indication that non-aggregated
control should be used. However a client MAY try to SETUP the
session in aggregated control. If the server refuse to aggregate
the specified media, the server SHALL use the 459 error code.
If the server allows the aggregation, then the client MUST
create an URI for aggregate control of the session. This URI
MUST contain the servers host address and MUST contain the
port, if applicable (e.g. not default port). Once an URI is used
to refer to an aggregation for a given session, that URI MUST
be used to refer to that aggregation for the duration of the
session. If the contents of the aggregation change, then a
different aggregate control URI SHOULD be used."

Please give your comments.

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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Subject: [MMUSIC] RTSP: Synchronization information in regards to changed transport
 parameters
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Hi,

The previous week we had a discussion on the issue if there is 
meaningful to allow for the possibility to include Range and RTP-Info in 
SETUP response which changes the transport parameters for a media stream.

There where a good use reasons mentioned by Anders Klemets why this 
information should be present. As of a result of the discussion the 
below text has been drafted by me. Please review and comment if it is 
appropriate.

"A client {\MAY} issue a {\SETUP} request for a stream that is
already set up or playing in the session to change transport
parameters, which a server {\MAY} allow.  If it does not allow
this, it {\MUST} respond with error 455 (Method Not Valid In This
State). Reasons to support changing transport parameters, is to
allow for application layer mobility and flexibility to utilize
the best available transport as it becomes available.

In a {\SETUP} response for a request to change the transport
parameters while in Play state, the server {\SHOULD} include the
\header{Range} to indicate from what point the new transport
parameters are used. Further if RTP is used for delivery the
server {\SHOULD} also include the \header{RTP-Info} header to
indicate from what timestamp and RTP sequence number the change
has taken place. If both \header{RTP-Info} and \header{Range} is
included in the response the "rtp\_time" parameter and range
{\MUST} be for the corresponding time, i.e. be used in the same
way as for {\PLAY} to ensure the correct synchronization
information is present.

If the transport parameter change while in play state results in a
change of synchronization related information, for example
changing RTP SSRC, the server {\MUST} provide in the {\SETUP}
response the necessary synchronization information. However the
server is {\RECOMMENDED} to avoid changing the synchronization
information if possible."


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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Date: Thu, 2 Oct 2003 09:23:04 -0700
From: Tom Marshall <tmarshall@real.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Cc: IETF MMUSIC WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] RTSP: Proposal to remove SETUP and TEARDOWN of individual media streams in PLAY state
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Together with the removal of ready-nm state, does mean that a successful
TEARDOWN always deletes the session?

On Thu, Oct 02, 2003 at 04:54:40PM +0200, Magnus Westerlund wrote:
> Hi,
>=20
> Another issue that was discussed on yesterdays telecon was the removal=20
> of the part specifying that usage of setup and teardown on individual=20
> media stream in PLAY state is possible. This does not effect the=20
> functionality for:
>=20
> - SETUP and Teardown in ready state
> - The usage of SETUP to change transport parameters.
> - Usage of aggregated Teardown in play state to end session.
>=20
> The removal of this functionality will be done in such way that it will=
=20
> be possible in the future to specify this as extension functionality.
>=20
> If no comments are received on this within a week I will start making=20
> these changes. And comments for keeping this functionality should state=
=20
> a strong use cases for this.
>=20
> See also:
> http://rtsp.org/bug810666
>=20
> Best Regards
>=20
> Magnus Westerlund
>=20
> 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
>=20
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Subject: Re: [MMUSIC] RTSP: Proposal to remove SETUP and TEARDOWN of indiv
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No,

Not if you do teardown on a individual stream in Ready state for a 
session with more than one media.

Tom Marshall wrote:

> Together with the removal of ready-nm state, does mean that a successful
> TEARDOWN always deletes the session?
> 

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: [MMUSIC] Notes for 10/01/03 RTSP Telecon
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Meeting Notes for 10/01 RTSP Concall

Attendees:

  Magnus Westerlund
  Thomas Zeng
  Aravind Narasimhan
  Greg Sherwood
  Colin Perkins
  Sean Sheedy

Detailed Minutes:

Thomas:  Working on strawman draft for End of Stream
Sean:    Filed end of stream 2 bug years ago.
Magnus:  In feature request area.
Magnus:  3.a. Range header bug.  We have text,  Let's close

Magnus:  3.b. PAUSE in ready state.
Magnus:  Text available in bis04. Any controversy? Take it to list

Magnus   3.c. Multiple SSRC. Let's close
Aravind: has been through multiple iterations in meetings and spec.

Magnus:  3.d. RTP timestamp and pause
Aravind: There is text in 3GPP.
Magnus:  I proof read it.  It should be good.  But we need a good
example in the RTSP Spec.

Magnus:  3.e. "a=range" is missing IANA registration
Magnus:  Has been written up.  Can be closed.  Please review text in bis04

Magnus:  3.f. PLAY at end of media
Magnus:  Written up this month.  Needs to be reviewed.

Magnus:  3.g: Implicit REDIRECT
Magnus:  Resolution in Vienna was a new 300 redirect code so that if you
understand the SDP and 300 code you can go immediately to the new server
and use the SDP.  If you dont understand the code then just redirect and
do an describe.
Greg:    Sounds good.

Vienna Discussion Items:

Colin Joined the call.

Magnus:  3.h. Aggregate URL discussion
Magnus:  Session declarations lacking a session level control URL,
SHOULD be used in non-aggregated control.
Aravind: The main issue is what the rules are for generating the
aggregate control URL.
Magnus:  The wording proposed to have client create aggregate control
URI  in the current text is good.
Aravind: I belive Bill May had an issue with this. Not sure if it is
resolved.
Magnus:  Will post to the mailing list the proposed wording.
Magnus:  I think the rules available which Aravind wrote into bis04
are acceptable. But lets resolve this when I post to the mailing
list the new SETUP text.

Magnus:  3.i.  a=range in SDP to indicate live media
Magnus:  Has been adopted in 3GPP.  I added the wording to CVS.
Thomas:  I dont see any problems with it


Magnus:  3.j. How does REDIRECT method work
Magnus:  Sessions will be terminated by successful REDIRECT
Thomas:  I dont object to this.  But I want to point out that this is
the only server -> client request that changes state
Thomas:  The spirit of not requiring TEARDOWN after redirect is that the
server can get rid of the session.  By having the client doing a
teardown the cleint maintains control of the state machine
Magnus:  It doesn't make sense to have the client send TEARDOWN when the
server needs to go away
Greg:    If the server wants to drop the session but do it
intelligently, it can do so by doing a REDIRECT without location header
which basically indicates to the client that the server is going away.
Greg:    Is it ok for the server to close the connecting when it sends a
REDIRECT?
Magnus:  Connection:close can't be put into work until the complete
transaction is performed. In other words the response must be sent
before closing the TCP connection.
Greg:    The situation where a server would send a REDIRECT is where the
server would have to go down in short order.  So it may want to close
the connection.
Thomas:  Why don't we do that instead of having a redirect always imply
a teardown.
Magnus:  After the client answers RTSP 200 Ok, that can have connection
close header.
Thomas:  Does the server still stay around to get 200 ok.
Greg:    The server may send the connection close header and may not do
anything else and my not stay around for the response.
Thomas:  If the server needs to go away then this wont work.
Greg:    In the case where the server has to go away the server should
wait for the 200k but it is just a courtesy to the client.
Magnus:  There are other considerations especially with proxies in the
middle.  If there are multiple sessions over a single TCP conneciton.
Thomas:  Isnt there a way to do a REDIRECT with out a session header
which tells the server that all the session are going away.  That will
take care of this concern.
Greg:    Please clarify the proxy issue. It is just replicating the
connections to the server, no?
Magnus:  1 TCP connection from proxy to server may be handling multiple
sessions.
Magnus:  Any volunteers to write up?  No.  Leave it waiting.

Magnus:  3.k Indicating end of stream
Thomas:  End of stream strawman draft is almost done.
Sean:    We had a proposal 2 years back based on the way we did it.  It
was a hack based on ANNOUCE, but there are bits of information that the
client wants.  Will Send to thomas for him to incoporate in to his write up
Mangus:  Client will want to know sequence number of the last packet on
the stream.
Thomas:  Specific to RTP-Info: The BNF does allow one of  rtptime or seqnum,
or both, so for response to END_OF_STREAM just seqnum is enough to tell
client what is the ending packet.
Magnus:  Take discussion to mailing list.

Magnus:  3.m Request to response time is not limited.
Magnus:  Response time limit for requests.  There is no indication in
spec right now.  Was pointed out by SIP folks.
Sean:    Would like to second that. Nobody is willing to wait for an
arbitrary amount of time for a response in order to retry.  10 seconsds
seems reasonable.
Thomas:  We need a header parameter in the client request indicating how
long the it is going to wait.
Sean:    If the client sends a request a server, how long should it wait?
Sean:    Should just have a resonable value.
Thomas:  Do you see an issue in having the client send a parameter
indicating client time out?
Sean:    Too complicated.
Greg:    Is there already a precedence for this in SIP?
Magnus:  Had an impression that there is something is SIP.
Collin:  It wouldn't surprise me if SIP has it.  But it may not be the
same thing that is useful for RTSP.
Thomas:  It is the client that knows what it wants to expect.  And since
the server can tell the client of timeout, why not have client do it to
server.
Magnus:  10 seconds should be enough for all purposes here.
Greg:    The server must respond in 10 seconds?
Thomas:  or send 100 response saying I'm working on it.
Greg:    What if the client pipelines hundereds of requests all at the
same time?
Collin:  There are trasnmission delays.  Can only deal with this with
Timestamps.
Magnus:  The requester waits up to 10 second. The responder must answer
within 5 seconds. That way some numbers on transport delay is present.
Thomas:  10 seconds without regard to TCP delays.
Collin:  Does the spec have a section with discussion when it is
appropriate to pipeline commands.
Thomas:  There is no congestions control dicussion on pipeline.
Magnus:  You can pipline n setups, multiple non-aggregated plays or
pause, OPTIONS and DESCRIBE.
Thomas:  Need to handle congestion control too. . .
Collin:  One is cogestgion, the other is that it is going to take a long
time
Sean:    If the server knows it needs to do some serious processing,
then it will send a 100 response.  If no 100 reponse it sent, then there
is a congestion problem somewhere
Sean:    Will attempt to write this up

Magnus:  3.n.  Version number
Mangus:  Aravind is taking a look at the spec for change
Aravind: I am cataloging the spec for changes so that we can have a good
understanding of the scope of the changes.  IT was clear at the Vienna
meeting that we did not have a good handle on this.  Should have a
preliminary list by sometime next week.

Magnus:  4.a TEARDOWN using * and the Ready-nm state.
Magnus:  No need to keep this complication in the state machine.  If you
have no media in your session, your session is gone.
Greg:    Why was this added:
Magnus:  Added a year ago.  It was mainly to preserve some resources on
the server side.  But this seems less useful now.
Aravind: Agreed.
Thomas:  Leaving it in will require more error handling.  What happens
when you do PLAY on a Ready-nm state?
Mangus:  Let's remove this state
Aravind: It complicates the state machine for the normal case.
Magnus:  I will mail the question to the mailing list before removing it

Magnus:  4.b. How to handle sync in re-SETUP
Magnus:  Will use Range and RTP-Info for syncing after RESETUP.  Will
post to list.

Magnus:  4.c. Re-SETUP of transport parameters
Magnus:  There was discussion on the mailing list. Geetha agreed to
write it up.

Magnus:  4.d. Removal of SETUP in play state
Magnus:  SETUP in play state will be removed from the core spec and will
post a mailing list.  It is better to leave this as an extension.
Thomas:  Yes it works better as an extension with the stream switching
stuff.
Aravind: Let's remove it, but let's make sure that we don't make  it
hard to add as an extension.


Magnus:  Next Meeting will be Tuesday, October 14th. That will give use
a week and half to finish up changes to the spec for the November meeting.
Magnus:  October 27th is the deadline for submission to November IETF
meeting.






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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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To: IETF MMUSIC WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] RTSP: Lack of aggregated control URI means non-aggregated
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References: <3F7C3E04.3080204@ericsson.com>
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Hi,

I got a comment that made me aware of the inappropriateness in my view 
of the last sentence in the below text, i.e. "If the contents of the 
aggregation change, then a different aggregate control URI SHOULD be 
used.". This sentence is not clear what it means. I think it is much 
simpler and clearer to simply require the client and server to continue 
to use the created URI for this session. Therefore I propose to remove 
the last sentence.


Best Regards

Magnus

Magnus Westerlund wrote:

> Hi,
> 
> In yesterdays telecon the issue around http://rtsp.org/bug640272
> was discussed. The remaining issue is if the lack of an aggregated 
> control URI in SDP means that a session "SHOULD" or "SHALL" be setup in 
> non-aggregated control. Both viewpoints was expressed in Vienna and the 
> the discussion needs closure. My personal view is that a SHOULD is 
> sufficient. Therefore I have drafted the below text for the SETUP 
> section of the RTSP draft. The URI rules are based on the ones Aravind 
> drafted.
> 
> "If an Aggregate control URI is not specified in the session
> description, it is probably a indication that non-aggregated
> control should be used. However a client MAY try to SETUP the
> session in aggregated control. If the server refuse to aggregate
> the specified media, the server SHALL use the 459 error code.
> If the server allows the aggregation, then the client MUST
> create an URI for aggregate control of the session. This URI
> MUST contain the servers host address and MUST contain the
> port, if applicable (e.g. not default port). Once an URI is used
> to refer to an aggregation for a given session, that URI MUST
> be used to refer to that aggregation for the duration of the
> session. If the contents of the aggregation change, then a
> different aggregate control URI SHOULD be used."
> 
> Please give your comments.
> 
> 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
> 

-- 

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: Aravind Narasimhan <aravind.narasimhan@sun.com>
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] Notes for 10/01/03 RTSP Telecon
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I still don't recall anyone explaining why this is an issue.=20
Implementations are free to "give up" at any point and, of course, the
end-user ultimately should decide the acceptable delay.  I think the most
sensible way to handle this is to recommend that clients have a default
timeout of X seconds (I would say X no smaller than 60 but that's me) and
the timeout should (must?) be configurable by the user.

The reason I'm so opposed to this idea is that RTSP runs on top of TCP.  TCP
can, and does, mitigate packet loss with retransmissions.  You cannot assume
that just because the client sent a request at time T, the server received
it at (T+RTT/2), and likewise for the server's response.  I've personally
run into high loss networks where responses were delayed for much longer
than 10 seconds due to retransmissions but still wanted the content enough
to wait.  I've also run into physical layer problems (disconnected network
cable, timed out VPN) where correcting the problem allowed the TCP
connection to continue smoothly.  I don't think we should gratuitously break
the RTSP layer sooner than the TCP layer gives up or try to second guess the
TCP layer, especially since RTSP above TCP in the stack.

> Magnus:  3.m Request to response time is not limited.
> Magnus:  Response time limit for requests.  There is no indication in
> spec right now.  Was pointed out by SIP folks.
> Sean:    Would like to second that. Nobody is willing to wait for an
> arbitrary amount of time for a response in order to retry.  10 seconsds
> seems reasonable.
> Thomas:  We need a header parameter in the client request indicating how
> long the it is going to wait.
> Sean:    If the client sends a request a server, how long should it wait?
> Sean:    Should just have a resonable value.
> Thomas:  Do you see an issue in having the client send a parameter
> indicating client time out?
> Sean:    Too complicated.
> Greg:    Is there already a precedence for this in SIP?
> Magnus:  Had an impression that there is something is SIP.
> Collin:  It wouldn't surprise me if SIP has it.  But it may not be the
> same thing that is useful for RTSP.
> Thomas:  It is the client that knows what it wants to expect.  And since
> the server can tell the client of timeout, why not have client do it to
> server.
> Magnus:  10 seconds should be enough for all purposes here.
> Greg:    The server must respond in 10 seconds?
> Thomas:  or send 100 response saying I'm working on it.
> Greg:    What if the client pipelines hundereds of requests all at the
> same time?
> Collin:  There are trasnmission delays.  Can only deal with this with
> Timestamps.
> Magnus:  The requester waits up to 10 second. The responder must answer
> within 5 seconds. That way some numbers on transport delay is present.
> Thomas:  10 seconds without regard to TCP delays.
> Collin:  Does the spec have a section with discussion when it is
> appropriate to pipeline commands.
> Thomas:  There is no congestions control dicussion on pipeline.
> Magnus:  You can pipline n setups, multiple non-aggregated plays or
> pause, OPTIONS and DESCRIBE.
> Thomas:  Need to handle congestion control too. . .
> Collin:  One is cogestgion, the other is that it is going to take a long
> time
> Sean:    If the server knows it needs to do some serious processing,
> then it will send a 100 response.  If no 100 reponse it sent, then there
> is a congestion problem somewhere
> Sean:    Will attempt to write this up

--=20
One good reason why computers can do more work than people is that they
never have to stop and answer the phone.

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Hi Tom,

I understand your concerns regarding TCP, and I think your reasoning is 
valid for this part of the problem. The other part is how long a server 
should maximum wait before using a 1xx response code to indicate is the 
other part. This problem arise for example when doing setup and the time 
to get the load the media can be of more than a few seconds, for example 
due to a tape robot.

In these scenarios it is good to have a time limit that tells you how 
long you are at maximum are allowed to delay responding without giving 
any indication that the request is received. Of course in the tape robot 
example if the server knows that this is a tape robot it could usually 
send a 1xx response directly after requesting the media to be loaded.

The client side is still a problem. I guess that one needs to write some 
recommendations on how the client should behave. Having a server side 
limit allow the client to form opinion on lowest possible delay before 
time-out.

Another interesting problem is with proxies. In this case the 
transmission of the request to the proxy goes fast and fine. Then when 
the proxy tries to contact the server the TCP connection takes an long 
time. How long should the client in this case wait?

Another question: For HTTP and SIP after sending a 1xx response, with 
what time period must the server repeat this message during the wait for 
the real response?

In conclusion: There are some considerations to make in this area.

- I think a server side time-out to send a 1xx response is good.

- I think that we must resolve also how the client is recommended to 
behave. This will always be a implementation thing, but guidelines I 
think is good.

Please continue the discussion.

Cheers

Magnus

Tom Marshall wrote:

> I still don't recall anyone explaining why this is an issue. 
> Implementations are free to "give up" at any point and, of course, the
> end-user ultimately should decide the acceptable delay.  I think the
> most
> sensible way to handle this is to recommend that clients have a default
> timeout of X seconds (I would say X no smaller than 60 but that's me)
> and
> the timeout should (must?) be configurable by the user.
> 
> The reason I'm so opposed to this idea is that RTSP runs on top of TCP.
> TCP
> can, and does, mitigate packet loss with retransmissions.  You cannot
> assume
> that just because the client sent a request at time T, the server
> received
> it at (T+RTT/2), and likewise for the server's response.  I've
> personally
> run into high loss networks where responses were delayed for much longer
> than 10 seconds due to retransmissions but still wanted the content
> enough
> to wait.  I've also run into physical layer problems (disconnected
> network
> cable, timed out VPN) where correcting the problem allowed the TCP
> connection to continue smoothly.  I don't think we should gratuitously
> break
> the RTSP layer sooner than the TCP layer gives up or try to second guess
> the
> TCP layer, especially since RTSP above TCP in the stack.
> 
> 
>>Magnus:  3.m Request to response time is not limited.
>>Magnus:  Response time limit for requests.  There is no indication in
>>spec right now.  Was pointed out by SIP folks.
>>Sean:    Would like to second that. Nobody is willing to wait for an
>>arbitrary amount of time for a response in order to retry.  10
> 
> seconsds
> 
>>seems reasonable.
>>Thomas:  We need a header parameter in the client request indicating
> 
> how
> 
>>long the it is going to wait.
>>Sean:    If the client sends a request a server, how long should it
> 
> wait?
> 
>>Sean:    Should just have a resonable value.
>>Thomas:  Do you see an issue in having the client send a parameter
>>indicating client time out?
>>Sean:    Too complicated.
>>Greg:    Is there already a precedence for this in SIP?
>>Magnus:  Had an impression that there is something is SIP.
>>Collin:  It wouldn't surprise me if SIP has it.  But it may not be the
>>same thing that is useful for RTSP.
>>Thomas:  It is the client that knows what it wants to expect.  And
> 
> since
> 
>>the server can tell the client of timeout, why not have client do it
> 
> to
> 
>>server.
>>Magnus:  10 seconds should be enough for all purposes here.
>>Greg:    The server must respond in 10 seconds?
>>Thomas:  or send 100 response saying I'm working on it.
>>Greg:    What if the client pipelines hundereds of requests all at the
>>same time?
>>Collin:  There are trasnmission delays.  Can only deal with this with
>>Timestamps.
>>Magnus:  The requester waits up to 10 second. The responder must
> 
> answer
> 
>>within 5 seconds. That way some numbers on transport delay is present.
>>Thomas:  10 seconds without regard to TCP delays.
>>Collin:  Does the spec have a section with discussion when it is
>>appropriate to pipeline commands.
>>Thomas:  There is no congestions control dicussion on pipeline.
>>Magnus:  You can pipline n setups, multiple non-aggregated plays or
>>pause, OPTIONS and DESCRIBE.
>>Thomas:  Need to handle congestion control too. . .
>>Collin:  One is cogestgion, the other is that it is going to take a
> 
> long
> 
>>time
>>Sean:    If the server knows it needs to do some serious processing,
>>then it will send a 100 response.  If no 100 reponse it sent, then
> 
> there
> 
>>is a congestion problem somewhere
>>Sean:    Will attempt to write this up
> 
> 

-- 

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] Notes for 10/01/03 RTSP Telecon
From: Sean Sheedy <seans@ncube.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Cc: Tom Marshall <tmarshall@real.com>, mmusic@ietf.org
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I basically agree with Magnus.

Relying on TCP alone is not sufficient.  Although it will provide
reliability to the control connection itself, it's incapable of handling
other congestion or failure events outside of the IP domain.  For
example:
      * Delays in the server's authentication and billing back end.
      * Server failures which still allow incoming requests from the IP
        stack, but which prevent the server from processing the requests
        in any useful way.

Clients need some guidance about the expected response time of the
server to be able to detect these types of failures and deal with them
appropriately.  At the least, it's more user-friendly to be able to say
"The server is not responding" than to simply wait for the user to give
up in frustration.

In practice, every client software author I've worked with has wanted to
know the maximum response time of the server.  I think discussing the
issue in the specification and coming up with some guidelines is better
for interoperability than leaving it entirely up to each implementor.

Sean


On Mon, 2003-10-06 at 00:28, Magnus Westerlund wrote:
> Hi Tom,
> 
> I understand your concerns regarding TCP, and I think your reasoning is 
> valid for this part of the problem. The other part is how long a server 
> should maximum wait before using a 1xx response code to indicate is the 
> other part. This problem arise for example when doing setup and the time 
> to get the load the media can be of more than a few seconds, for example 
> due to a tape robot.
> 
> In these scenarios it is good to have a time limit that tells you how 
> long you are at maximum are allowed to delay responding without giving 
> any indication that the request is received. Of course in the tape robot 
> example if the server knows that this is a tape robot it could usually 
> send a 1xx response directly after requesting the media to be loaded.
> 
> The client side is still a problem. I guess that one needs to write some 
> recommendations on how the client should behave. Having a server side 
> limit allow the client to form opinion on lowest possible delay before 
> time-out.
> 
> Another interesting problem is with proxies. In this case the 
> transmission of the request to the proxy goes fast and fine. Then when 
> the proxy tries to contact the server the TCP connection takes an long 
> time. How long should the client in this case wait?
> 
> Another question: For HTTP and SIP after sending a 1xx response, with 
> what time period must the server repeat this message during the wait for 
> the real response?
> 
> In conclusion: There are some considerations to make in this area.
> 
> - I think a server side time-out to send a 1xx response is good.
> 
> - I think that we must resolve also how the client is recommended to 
> behave. This will always be a implementation thing, but guidelines I 
> think is good.
> 
> Please continue the discussion.
> 
> Cheers
> 
> Magnus
> 
> Tom Marshall wrote:
> 
> > I still don't recall anyone explaining why this is an issue. 
> > Implementations are free to "give up" at any point and, of course, the
> > end-user ultimately should decide the acceptable delay.  I think the
> > most
> > sensible way to handle this is to recommend that clients have a default
> > timeout of X seconds (I would say X no smaller than 60 but that's me)
> > and
> > the timeout should (must?) be configurable by the user.
> > 
> > The reason I'm so opposed to this idea is that RTSP runs on top of TCP.
> > TCP
> > can, and does, mitigate packet loss with retransmissions.  You cannot
> > assume
> > that just because the client sent a request at time T, the server
> > received
> > it at (T+RTT/2), and likewise for the server's response.  I've
> > personally
> > run into high loss networks where responses were delayed for much longer
> > than 10 seconds due to retransmissions but still wanted the content
> > enough
> > to wait.  I've also run into physical layer problems (disconnected
> > network
> > cable, timed out VPN) where correcting the problem allowed the TCP
> > connection to continue smoothly.  I don't think we should gratuitously
> > break
> > the RTSP layer sooner than the TCP layer gives up or try to second guess
> > the
> > TCP layer, especially since RTSP above TCP in the stack.
> > 
> > 
> >>Magnus:  3.m Request to response time is not limited.
> >>Magnus:  Response time limit for requests.  There is no indication in
> >>spec right now.  Was pointed out by SIP folks.
> >>Sean:    Would like to second that. Nobody is willing to wait for an
> >>arbitrary amount of time for a response in order to retry.  10
> > 
> > seconsds
> > 
> >>seems reasonable.
> >>Thomas:  We need a header parameter in the client request indicating
> > 
> > how
> > 
> >>long the it is going to wait.
> >>Sean:    If the client sends a request a server, how long should it
> > 
> > wait?
> > 
> >>Sean:    Should just have a resonable value.
> >>Thomas:  Do you see an issue in having the client send a parameter
> >>indicating client time out?
> >>Sean:    Too complicated.
> >>Greg:    Is there already a precedence for this in SIP?
> >>Magnus:  Had an impression that there is something is SIP.
> >>Collin:  It wouldn't surprise me if SIP has it.  But it may not be the
> >>same thing that is useful for RTSP.
> >>Thomas:  It is the client that knows what it wants to expect.  And
> > 
> > since
> > 
> >>the server can tell the client of timeout, why not have client do it
> > 
> > to
> > 
> >>server.
> >>Magnus:  10 seconds should be enough for all purposes here.
> >>Greg:    The server must respond in 10 seconds?
> >>Thomas:  or send 100 response saying I'm working on it.
> >>Greg:    What if the client pipelines hundereds of requests all at the
> >>same time?
> >>Collin:  There are trasnmission delays.  Can only deal with this with
> >>Timestamps.
> >>Magnus:  The requester waits up to 10 second. The responder must
> > 
> > answer
> > 
> >>within 5 seconds. That way some numbers on transport delay is present.
> >>Thomas:  10 seconds without regard to TCP delays.
> >>Collin:  Does the spec have a section with discussion when it is
> >>appropriate to pipeline commands.
> >>Thomas:  There is no congestions control dicussion on pipeline.
> >>Magnus:  You can pipline n setups, multiple non-aggregated plays or
> >>pause, OPTIONS and DESCRIBE.
> >>Thomas:  Need to handle congestion control too. . .
> >>Collin:  One is cogestgion, the other is that it is going to take a
> > 
> > long
> > 
> >>time
> >>Sean:    If the server knows it needs to do some serious processing,
> >>then it will send a 100 response.  If no 100 reponse it sent, then
> > 
> > there
> > 
> >>is a congestion problem somewhere
> >>Sean:    Will attempt to write this up
> > 
> > 
-- 
Sean Sheedy                                         seans@ncube.com
(503) 531-6427


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From: Tom Marshall <tmarshall@real.com>
To: Sean Sheedy <seans@ncube.com>
Cc: Magnus Westerlund <magnus.westerlund@ericsson.com>, mmusic@ietf.org
Subject: Re: [MMUSIC] Notes for 10/01/03 RTSP Telecon
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On Mon, Oct 06, 2003 at 01:18:15PM -0700, Sean Sheedy wrote:
> I basically agree with Magnus.
>=20
> Relying on TCP alone is not sufficient.  Although it will provide
> reliability to the control connection itself, it's incapable of handling
> other congestion or failure events outside of the IP domain.  For
> example:
>       * Delays in the server's authentication and billing back end.
>       * Server failures which still allow incoming requests from the IP
>         stack, but which prevent the server from processing the requests
>         in any useful way.

While I agree in principle that a method of handling delays is desirable, I
strongly suggest using _only_ positive feedback (1xx responses).  I want to
ensure that it's crystal clear that negative feedback (no response) is _not_
an indication of failure.  Just about every network application I know about
allows the TCP layer to do its job without imposing arbitrary time limits on
responses.  Try unplugging your network cable and running various programs
that use TCP sockets to see how long they take to fail.  The applications
that do impose arbitrary time limits (our player is one) have configurable
limits for the timeout period.  I'm not thrilled with that solution but at
least the user can adjust the timeout for situations where packet loss or
delay are larger than the implementor expected.

Regarding your second bullet, it does not really matter if the server can
allow incoming requests or not.  The TCP stack will happily accept several
normal sized RTSP requests and queue them even if the network cable is
unplugged.  In the BSD socket API, acceptance for delivery does not imply
anything about whether the data was actually delivered.  One could argue
that a non-BSD implementation could pass up the ACK so that the application
knew the data had been received by the other side, but that only guarantees
that the destination TCP stack has received the data.  It still doesn't say
whether the server application has received it.  If the server is indeed in
a state where it can accept requests but cannot process them for longer than
the TCP is willing to wait (such that the TCP connection is broken) and it
further cannot send 1xx responses to tell the client it's busy, I submit
that the server needs to be redesigned.

> Clients need some guidance about the expected response time of the
> server to be able to detect these types of failures and deal with them
> appropriately.  At the least, it's more user-friendly to be able to say
> "The server is not responding" than to simply wait for the user to give
> up in frustration.
>=20
> In practice, every client software author I've worked with has wanted to
> know the maximum response time of the server.  I think discussing the
> issue in the specification and coming up with some guidelines is better
> for interoperability than leaving it entirely up to each implementor.

Again, I suggest playing with different network applications to see what
implementors of other protocols have done.

This issue is as old as TCP itself.  I know it's tempting to try second
guessing the network and pop up a dialog that says the server is not
responding, please contact the provider, it's not our fault.  But it's
simply not possible to do this on top of TCP.  The only feedback that you
can give is positive.  If the user gets tired of waiting, let the user press
the "stop" button.  It's not that hard.  Really.

There is also a difference between initial delay when establishing a TCP
connection and the delay in RTSP request/response time.  The usual delay
that causes problems establishing the TCP connection is the DNS lookup.=20
When designing guidelines for clients, I think it's critical that we make
this point.  RTSP delay guidelines are in no way a guideline for the
underlying TCP connection and they must _only_ be used after the TCP
connection has been successfully established.

--=20
To err is human; to really screw things up requires the root password.

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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-09.txt
	Pages		: 18
	Date		: 2003-10-7
	
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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From: "Joel Tran" <joel_tran@hotmail.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>, "Mmusic" <mmusic@ietf.org>
Date: Tue, 7 Oct 2003 11:07:47 -0400
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Subject: [MMUSIC] Status of TURN
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Hi Jonathan,

I would like to ask you two little questions concerning TURN.

First, I would like to know what is the exact status of TURN? Will it be
adopted as a Workgroup item of MMUSIC since TURN is refeered in various
documents?

Second, I would like to ask you if there is a reason why the lastest =
version
of the draft is not on the IETF web and only available throught
http://www.jdrosen.net/papers/draft-rosenberg-midcom-turn-02.txt?

Thanks,
...J

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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Joel Tran'" <joel_tran@hotmail.com>,
        Jonathan Rosenberg
	 <jdrosen@dynamicsoft.com>,
        Mmusic <mmusic@ietf.org>
Subject: RE: [MMUSIC] Status of TURN
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The Transport Area Directors understand that this document will proceed as
an individual submission, outside the context of any of the working groups
in the Transport Area. It is clearly of interest to a number of these
working groups.

I gather that at least one more revision needs to be undertaken before this
document will be considered for publication as an RFC, but we're very
interested to see it go forward.

Jon Peterson
NeuStar, Inc.
(co-AD for Transport)

> -----Original Message-----
> From: Joel Tran [mailto:joel_tran@hotmail.com]
> Sent: Tuesday, October 07, 2003 11:08 AM
> To: Jonathan Rosenberg; Mmusic
> Subject: [MMUSIC] Status of TURN
> 
> 
> Hi Jonathan,
> 
> I would like to ask you two little questions concerning TURN.
> 
> First, I would like to know what is the exact status of TURN? 
> Will it be
> adopted as a Workgroup item of MMUSIC since TURN is refeered 
> in various
> documents?
> 
> Second, I would like to ask you if there is a reason why the 
> lastest version
> of the draft is not on the IETF web and only available throught
> http://www.jdrosen.net/papers/draft-rosenberg-midcom-turn-02.txt?
> 
> Thanks,
> ...J
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 

_______________________________________________
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Thanks for the clarification, Jon.

To answer Joel's second question - there were some mixups during the 
submission process last time, and the -02 never got into the archives. 
I've just resubmitted it so it should be there shortly.

I am also hoping to get in a real revision (resulting in a -03 
version) before the deadline. Please, if you have comments or 
questions, forward them to me.

Thanks,
Jonathan R.

Peterson, Jon wrote:

> The Transport Area Directors understand that this document will proceed as
> an individual submission, outside the context of any of the working groups
> in the Transport Area. It is clearly of interest to a number of these
> working groups.
> 
> I gather that at least one more revision needs to be undertaken before this
> document will be considered for publication as an RFC, but we're very
> interested to see it go forward.
> 
> Jon Peterson
> NeuStar, Inc.
> (co-AD for Transport)
> 
> 
>>-----Original Message-----
>>From: Joel Tran [mailto:joel_tran@hotmail.com]
>>Sent: Tuesday, October 07, 2003 11:08 AM
>>To: Jonathan Rosenberg; Mmusic
>>Subject: [MMUSIC] Status of TURN
>>
>>
>>Hi Jonathan,
>>
>>I would like to ask you two little questions concerning TURN.
>>
>>First, I would like to know what is the exact status of TURN? 
>>Will it be
>>adopted as a Workgroup item of MMUSIC since TURN is refeered 
>>in various
>>documents?
>>
>>Second, I would like to ask you if there is a reason why the 
>>lastest version
>>of the draft is not on the IETF web and only available throught
>>http://www.jdrosen.net/papers/draft-rosenberg-midcom-turn-02.txt?
>>
>>Thanks,
>>...J
>>
>>_______________________________________________
>>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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Thread-Topic: specifying the multicast interface
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Subject: [MMUSIC] specifying the multicast interface
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When creating a multicast session, it is possible to specify the
outgoing interface using the IP_MULTICAST_IF socket option.  Does RFC
2326 (or 3550/1889) prescribe a mechanism for specifying the interface?

Analogously, the IP_ADD_MEMBERSHIP socket option allows the receiver to
specify the multicast interface explicitly (via the imr_interface data
member of the ip_mreq struct).  Is there a mechanism (via the SDP
DESCRIBE, I guess) for indicating the multicast interface?

Thanks,
Matt


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--NextPart


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


        RFC 3605

        Title:      Real Time Control Protocol (RTCP) attribute in
                    Session Description Protocol (SDP)
        Author(s):  C. Huitema
        Status:     Standards Track
        Date:       October 2003
        Mailbox:    huitema@microsoft.com
        Pages:      8
        Characters: 17270
        Updates/Obsoletes/SeeAlso:    None

        I-D Tag:    draft-ietf-mmusic-sdp4nat-05.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3605.txt


The Session Description Protocol (SDP) is used to describe the
parameters of media streams used in multimedia sessions.  When a
session requires multiple ports, SDP assumes that these ports have
consecutive numbers.  However, when the session crosses a network
address translation device that also uses port mapping, the ordering
of ports can be destroyed by the translation.  To handle this, we
propose an extension attribute to SDP.

This document is a product of the Multiparty Multimedia Session
Control Working Group of the IETF.

This is now a Proposed Standard Protocol.

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

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Authors, for further information.


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

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--NextPart


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


        RFC 3605

        Title:      Real Time Control Protocol (RTCP) attribute in
                    Session Description Protocol (SDP)
        Author(s):  C. Huitema
        Status:     Standards Track
        Date:       October 2003
        Mailbox:    huitema@microsoft.com
        Pages:      8
        Characters: 17270
        Updates/Obsoletes/SeeAlso:    None

        I-D Tag:    draft-ietf-mmusic-sdp4nat-05.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3605.txt


The Session Description Protocol (SDP) is used to describe the
parameters of media streams used in multimedia sessions.  When a
session requires multiple ports, SDP assumes that these ports have
consecutive numbers.  However, when the session crosses a network
address translation device that also uses port mapping, the ordering
of ports can be destroyed by the translation.  To handle this, we
propose an extension attribute to SDP.

This document is a product of the Multiparty Multimedia Session
Control Working Group of the IETF.

This is now a Proposed Standard Protocol.

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

This announcement is sent to the IETF list and the RFC-DIST list.
Requests to be added to or deleted from the IETF distribution list
should be sent to IETF-REQUEST@IETF.ORG.  Requests to be
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Details on obtaining RFCs via FTP or EMAIL may be obtained by sending
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        To: rfc-info@RFC-EDITOR.ORG
        Subject: getting rfcs

        help: ways_to_get_rfcs

Requests for special distribution should be addressed to either the
author of the RFC in question, or to RFC-Manager@RFC-EDITOR.ORG.  Unless
specifically noted otherwise on the RFC itself, all RFCs are for
unlimited distribution.echo 
Submissions for Requests for Comments should be sent to
RFC-EDITOR@RFC-EDITOR.ORG.  Please consult RFC 2223, Instructions to RFC
Authors, for further information.


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

...

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From: Thomas Zeng <zeng@PacketVideo.COM>
To: mmusic@ietf.org
Date: Fri, 10 Oct 2003 18:05:50 -0700
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Subject: [MMUSIC] strawman draft for signalling end of stream in RTSP
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Hi:

During Vienna IETF, I took the action to write this draft, which will be an
extension of the core RTSP protocol. It's been 2 month, I finally have the
first version, which is attached.

Thanks in advance for your comments,

thomas <<rtsp_end_of_stream.txt>> 

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=0A=
MMUSIC                                                     Thomas M. =
Zeng=0A=
                                                         P. Greg =
Sherwood=0A=
Internet-Draft                                                =
PacketVideo=0A=
Expires: April 29, 2004                                     Sept 29, =
2003=0A=
=0A=
=0A=
                    Signalling End Of Stream in RTSP=0A=
                       draft-zeng-mmusic-00=0A=
=0A=
Status of this Memo=0A=
=0A=
   This document is an Internet-Draft and is in full conformance =
with=0A=
   all provisions of Section 10 of RFC2026.=0A=
=0A=
   Internet-Drafts are working documents of the Internet Engineering=0A=
   Task Force (IETF), its areas, and its working groups. Note that =
other=0A=
   groups may also distribute working documents as Internet-Drafts.=0A=
=0A=
   Internet-Drafts are draft documents valid for a maximum of six =
months=0A=
   and may be updated, replaced, or obsoleted by other documents at =
any=0A=
   time. It is inappropriate to use Internet-Drafts as reference=0A=
   material or to cite them other than as "work in progress."=0A=
=0A=
   The list of current Internet-Drafts can be accessed at http://=0A=
   www.ietf.org/ietf/1id-abstracts.txt.=0A=
=0A=
   The list of Internet-Draft Shadow Directories can be accessed at=0A=
   http://www.ietf.org/shadow.html.=0A=
=0A=
   This Internet-Draft will expire on Apr. 29, 2004.=0A=
=0A=
Copyright Notice=0A=
=0A=
   Copyright (C) The Internet Society (2003). All Rights Reserved.=0A=
=0A=
Abstract=0A=
=0A=
   This document describes a new RTSP method called END_OF_STREAM, =
which =0A=
   extends the core RTSP protocol to enable one RTSP protocol entity to =
=0A=
   signal other entity(s) =0A=
   that it has no more media data to send. This request could be issued =
either with a =0A=
   non-aggregate or aggregate control URI, the latter implies that all =
media streams =0A=
   in the aggregate session have run out of media data.=0A=
=0A=
   The receiver of this RTSP request is expected to reply with 200 OK =
response.=0A=
=0A=
   Since END_OF_STREAM represents an extension to the core RTSP =
protocol, a feature tag,=0A=
   "method.eos", is defined to facilitate capability negotiation using =
RTSP extension =0A=
   mechanism [RTSP_NEW].=0A=
=0A=
=0A=
1. Motivation=0A=
=0A=
   In the core RTSP protocol [RTSP_NEW], an RTSP client relies on =
the=0A=
   media transport mechanism  to signal end of stream.=0A=
=0A=
   When the media transport mechanism happens to be RTP over UDP, this =
=0A=
   is carried out by RTCP BYE packet [RTP_NEW]. In practice, there are =
=0A=
   some drawbacks with this approach:=0A=
     =0A=
     1. When the server sends an RTCP BYE packet with its SSRC, the =
server=0A=
        is giving up=0A=
        the SSRC (see section 8.2 in [RTP_NEW]). The server would be =
required to =0A=
        switch to a new SSRC on a subsequent PLAY of the same media =
stream.=0A=
        Since server's SSRC is only communicated in the Transport =
header of SETUP =0A=
        response, the server would not have an opportunity to send a =
new value to =0A=
        the player, and the client would have to discover the SSRC from =
the=0A=
        incoming RTP packets. =0A=
=0A=
     2. RTCP BYE packet method does not offer a simple, gurranteed =
method of delievering=0A=
        an end-of-stream announcement.=0A=
=0A=
     3. RTCP BYE packet method does not offer the option to have a =
single aggregate=0A=
        end-of-stream announcemnt for all media streams in the RTSP =
session.=0A=
=0A=
     4. Section 6.3.7 of RFC3550 stipulates that an RTP sender cannot =
send =0A=
        RTCP BYE =0A=
        before leaving the RTP session if it has not already sent at =
least one =0A=
        RTP or RTCP packet. This is a problem under error conditions. =
Consider the case =0A=
        where an RTP session has just started (i.e., RTSP PLAY has been =
=0A=
        successfully=0A=
        acknowledged with an RTSP 200 OK response), and the sender =
attempts to =0A=
        retrieve media frames from its media source. The media source =
fails =0A=
        to provide any media frame due to its internal error such as =
=0A=
        file corruption. The sender should inform its receiver(s) but =
it cannot=0A=
        send BYE packets.=0A=
    =0A=
=0A=
   The motivation to solve the above issues is particularly high for =
=0A=
   unicast streaming applications that use RTSP over TCP in the control =
plane,=0A=
   and RTP over UDP in the media transport. =0A=
=0A=
   There is also the desire to have an EOS (End Of Stream) =0A=
   signalling mechanism =0A=
   for non-RTP delivery. One such delivery is MPEG2 transport streams =
used in =0A=
   the cable TV environment. In non-IP delivery environments, the =
transport =0A=
   typically remains allocated even if no media is being delivered.  =
This=0A=
   means that a client cannot watch for the server to close the=0A=
   transport to signal the end of stream. Meanwhile,  watching for the =
=0A=
   incoming media to stop is unreliable.  Short=0A=
   timeouts can trigger a false end of media detection if the=0A=
   media flow is temporarily delayed.  Long timeouts introduce=0A=
   unacceptable latencies.  Clients are unable to distinguish=0A=
   between a normal end of stream and an error condition that=0A=
   resulted in the media delivery stopping.=0A=
=0A=
   We note that using TEARDOWN from server to client is not appropriate =
=0A=
   because:=0A=
     1. TEARDOWN is currently not allowed from server to client =
[RTSP_NEW];=0A=
     2. Even if TEARDOWN is made available in server to client =
direction, =0A=
        the definition of TEARDOWN requires that, if the request URI is =
=0A=
        aggregate, that the session must be de-allocated by the server. =
=0A=
        There are RTSP applications that use SET_PARAMETER from client =
to=0A=
        server as the means to report session QoS statistics, but if=0A=
        server uses TEARDOWN on aggregate URL to signal end of stream, =
=0A=
        the client can no longer use SET_PARAMETER with a session =
header. =0A=
     3. In general, RTSP, being a client-server protocol,=0A=
        should let client, not server to control session state. But =
TEARDOWN=0A=
        on aggregate URL will change session from PLAYING state to INIT =
state.=0A=
=0A=
   We note that using PAUSE from server to client is not appropriate =
either,=0A=
   because PAUSE will change the state of the RTSP session.=0A=
=0A=
   We note also that using ANNOUNCE from server to client is not =
suitable, for two reasons:=0A=
      1. ANNOUNCE from server to client is used in RFC2326 to update =
session description. =0A=
         Extending it to indicate end-of-stream event is awkward.=0A=
      2. ANNOUNCE is not in the core RTSP protocol [RTSP_NEW].=0A=
   =0A=
   The next section describes our solution.=0A=
=0A=
3. Solution Using END_OF_STREAM method=0A=
=0A=
   We introduce a new RTSP method to signal end of stream in =
applications=0A=
   where an persistent RTSP connection is maintained between server and =
client.=0A=
   The new method is an RTSP "extension-method" [RTSP_NEW]. It is =
called=0A=
   END_OF_STREAM, and is a request from media sender=0A=
   to receiver. In the case of RTSP client playing media streams sent =
by RTSP server,=0A=
   the END_OF_STREAM is sent from server to client. In the use case =
where RTSP server is=0A=
   recording media sent by RTSP client, the END_OF_STREAM requet could =
be sent from an=0A=
   RTSP client to server.  =0A=
=0A=
   As any other RTSP request method, END_OF_STREAM must be acknowledged =
with a response.=0A=
   =0A=
   3.1 The Definition of END_OF_STREAM=0A=
      =0A=
   The END_OF_STREAM request for a URI signals the fact that no more =
RTP packets will be=0A=
   sent. Here is an example:=0A=
=0A=
     S->C: END_OF_STREAM rtsp://foo.com/bar.avi RTSP/1.0=0A=
           CSeq: 10=0A=
           Session: 12345678=0A=
	   Range: npt=3D0-200=0A=
           RTP-Info: =
url=3D//foo.com/bar.avi/streamid=3D0;seq=3D45102,=0A=
             url=3Drtsp://foo.com/bar.avi/streamid=3D1;seq=3D30211    =
=0A=
	   Reason: End of range reached=0A=
=0A=
     C->S: RTSP/1.0 200 OK  =0A=
           CSeq: 10=0A=
           Session: 12345678=0A=
    =0A=
   An END_OF_STREAM request MUST include "CSeq", "Range" and "Session" =
headers.=0A=
   It SHOULD include "RTP-Info" header.=0A=
   The RTP-Info in server's END_OF_STREAM request=0A=
   is used to indicate the sequence number of =0A=
   the ending RTP packet for each media stream.=0A=
   =0A=
   An END_OF_STREAM requet MAY include a new "Reason" header, defined =
as a =0A=
   string, whose purpose is to allow the server to explain why stream =
has ended,=0A=
   and whose ABNF definition is given below:=0A=
    =0A=
          Reason     =3D  "Reason" ":"   Reason-Phrase CRLF=0A=
=0A=
   where:=0A=
       -- "Reason-Phrase" is a parameter defined in section 7.1.1 of =
[RTSP_NEW].=0A=
=0A=
   =0A=
   Note that the server is free to use any text as "Reason-Phrase". In =
certain=0A=
   applications, the client=0A=
   may  display the text, in its entirety, to end user to improve =
user=0A=
   friendliness.=0A=
      =0A=
   END_OF_STREAM does not remove session ID, nor does it change the =
state=0A=
   of the RTSP session.=0A=
=0A=
   END_OF_STREAM may be applied to either aggregate or non-aggregate =
URI.=0A=
   =0A=
   END_OF_STREAM does not apply if the RTSP session is not in =
PLAYING=0A=
   state. If a client receives END_OF_STREAM while it is not in PLAYING =
=0A=
   state, it must reply with a 455 response: Method Not Valid in This =
State.=0A=
=0A=
   We note that client can piggy back QoS statistics as a entity body =
in=0A=
   its 200 OK response to the END_OF_STREAM request from the server.   =
=0A=
=0A=
   3.2 Limitations=0A=
   =0A=
   END_OF_STREAM method is applicable only if the server has the =
means=0A=
   to contact the client when it reaches the end of a media stream.=0A=
   This may not be possible if the RTSP connection between server =
and=0A=
   client is not persistent. When it is time to send END_OF_STREAM =
but=0A=
   the server does not have connection to the client, the server =
will=0A=
   simply skip END_OF_STREAM. That is to say, the server will not =0A=
   queue up the END_OF_STREAM message.=0A=
   =0A=
=0A=
4. Feature tag=0A=
  =0A=
   The END_OF_STREAM feature is represented by a feature tag (see =
[RTSP_NEW]=0A=
   for how to use feature tags in capability negotiations).=0A=
=0A=
   The feature tag is:=0A=
=0A=
          method.eos=0A=
=0A=
   which means that the END_OF_STREAM method is fully supported. By =
fully=0A=
   we mean either at non-aggregate or aggregate level.=0A=
=0A=
5. Use Case=0A=
=0A=
   In this example, the server sends END_OF_STREAM request to client =
for=0A=
   one live media stream, because upstream source terminates the =
stream=0A=
   after 200 seconds. The fact that the stream has played for 200 =
seconds=0A=
   is communicated by the Range header in the END_OF_STREAM request.=0A=
=0A=
     C->S: PLAY rtsp://foo.com/bar.avi/streamid=3D0 RTSP/1.0=0A=
	   Supported: method.eos=0A=
           CSeq: 10=0A=
           Session: 12345678=0A=
	   Range: 0-200=0A=
=0A=
     S->C: RTSP/1.0 200 OK=0A=
	   Supported: method.eos=0A=
           CSeq: 10=0A=
           Session: 12345678=0A=
           RTP-Info: url=3D//foo.com/bar.avi/streamid=3D0;seq=3D0; =
rtptime=3D0 =0A=
=0A=
     S->C: END_OF_STREAM rtsp://foo.com/bar.avi/streamid=3D0 =
RTSP/1.0=0A=
           CSeq: 123=0A=
           Session: 12345678=0A=
	   Range: npt=3D0-200; bytes=3D0-200000=0A=
           RTP-Info: url=3D//foo.com/bar.avi/streamid=3D0;seq=3D45102 =
=0A=
	   Reason: End of range reached=0A=
=0A=
     C->S: RTSP/1.0 200 OK  =0A=
           CSeq: 123=0A=
           Session: 12345678=0A=
  =0A=
   In the END_OF_STREAM request, the client will learn that the server =
has =0A=
   completed the stream as requested (as indicated by the "End of range =
reached" Reason).=0A=
=0A=
   From the two RTP-Info headers, one in PLAY response, one in =
END_OF_STREAM requst,=0A=
   the client can derive the total number=0A=
   of RTP packets that the server has sent. From the npt field in the =
Range header, =0A=
   the client knows the presentation time that this stream has covered, =
from the=0A=
   server's point of view, while the byte-range field can optional =
inform the =0A=
   client how much media data, in bytes, that the server has sent. =0A=
 =0A=
5. Security Considerations=0A=
 =0A=
   Because there is only one new TEXT header, "Reason", added by our =
=0A=
   new RTSP method,=0A=
   the security considerations outlined in [RTSP_NEW] apply here as =
well.=0A=
     =0A=
6. IANA Considerations=0A=
=0A=
   A new method name, and its associated feature tag need to be =
registered =0A=
   with IANA.=0A=
    =0A=
     -- Method name: END_OF_STREAM=0A=
     =0A=
     -- feature tag: method.eos=0A=
=0A=
=0A=
=0C=0A=
=0A=
Normative References=0A=
=0A=
   [RTSP_NEW] Schulzrinne, H., Rao, A., Lanphier, R., Westerlund, M., =
=0A=
         "Real Time Streaming Protocol", =0A=
         draft-ietf-mmusic-rfc2326bis-04.txt=0A=
=0A=
   [RTP_NEW] RFC3550 "Real-time Transport Protocol", July 2003=0A=
=0A=
=0A=
Full Copyright Statement=0A=
=0A=
   Copyright (C) The Internet Society (2003). All Rights Reserved.=0A=
=0A=
   This document and translations of it may be copied and furnished =
to=0A=
   others, and derivative works that comment on or otherwise explain =
it=0A=
   or assist in its implementation may be prepared, copied, =
published=0A=
   and distributed, in whole or in part, without restriction of any=0A=
   kind, provided that the above copyright notice and this paragraph =
are=0A=
   included on all such copies and derivative works. However, this=0A=
   document itself may not be modified in any way, such as by =
removing=0A=
   the copyright notice or references to the Internet Society or =
other=0A=
   Internet organizations, except as needed for the purpose of=0A=
   developing Internet standards in which case the procedures for=0A=
   copyrights defined in the Internet Standards process must be=0A=
   followed, or as required to translate it into languages other =
than=0A=
   English.=0A=
=0A=
   The limited permissions granted above are perpetual and will not =
be=0A=
   revoked by the Internet Society or its successors or assignees.=0A=
=0A=
   This document and the information contained herein is provided on =
an=0A=
   "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET =
ENGINEERING=0A=
   TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, =
INCLUDING=0A=
   BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION=0A=
=0A=
=0A=
=0A=
=0A=
=0A=
   HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF=0A=
   MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.=0A=
=0A=
=0A=
Acknowledgement=0A=
=0A=
   Thanks to Sean Sheedy for suggesting the inclusion of "Range" =
header.=0A=
=0A=
   Funding for the RFC Editor function is currently provided by the=0A=
   Internet Society.=0A=

------_=_NextPart_000_01C38F93.D00A5B50--

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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-533691 ] Clarify Range Where t0 <= t1 (SourceForge.net)
   2. [ rtspspec-Bugs-513880 ] How to use Range when Scale is negative (SourceForge.net)
   3. [ rtspspec-Bugs-663715 ] Requirement levels for Transport:SSRC needs clarification (SourceForge.net)
   4. [ rtspspec-Bugs-663715 ] Requirement levels for Transport:SSRC needs clarification (SourceForge.net)
   5. [ rtspspec-Bugs-533691 ] Clarify Range Where t0 <= t1 (SourceForge.net)
   6. [ rtspspec-Bugs-513880 ] How to use Range when Scale is negative (SourceForge.net)
   7. [ rtspspec-Bugs-513737 ] Unclarities when doing resetup (SourceForge.net)
   8. [ rtspspec-Bugs-630920 ] RTP-Info does not support multiple SSRC (SourceForge.net)

--__--__--

Message: 1
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Wed, 01 Oct 2003 08:16:50 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-533691 ] Clarify Range Where t0 <= t1

Bugs item #533691, was opened at 2002-03-22 12:36
Message generated for change (Comment added) made by arvy_n
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=533691&group_id=23194

Category: None
>Group: Fix - Discussed/Resolved
Status: Open
Resolution: None
Priority: 5
Submitted By: Thomas M Zeng (zengtm)
>Assigned to: Aravind Narasimhan (arvy_n)
Summary: Clarify Range Where t0 <= t1

Initial Comment:
If the server receives a PLAY range from t0-t1, where

t0 < t1, should it reject the PLAY, or should the 

server play backwards as if the timescale is -1?



If t0 = t1, should the server send any media data 

after responding 200 OK?  

We've seen different servers behave differently in 

this scenario.



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

>Comment By: Aravind Narasimhan (arvy_n)
Date: 2003-10-01 11:16

Message:
Logged In: YES 
user_id=437699

The following text has been added to the spec with regards

to this bug:



By default, range intervals increase, where the second point

is larger than the first point.



   Example:



       Range: npt=10-15



However, range intervals can also decrease if the Scale

header (see section  13.34) indicates a negative scale

value. For example, this would be the case when a playback

in reverse is desired.



   Example:



       Scale: -1

       Range: npt=15-10



Decreasing ranges are still half open intervals as described

above.  Thus, For range A-B, A is closed and B is open. In

the above example, 15 is closed and 10 is open. An exception

to this rule is the case when B=0 in a decreasing range. In

this case, the range is closed on both ends, as otherwise

there would be no way to reach 0 on a reverse playback.



   Example:



       Scale: -1

       Range: npt=15-0



   In this range both 15 and 0 are closed.



A decreasing range interval without a corresponding negative

Scale header is not valid.

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

Comment By: Thomas M Zeng (zengtm)
Date: 2003-03-05 12:53

Message:
Logged In: YES 
user_id=20397

If scale positive, t1 must >= t0.



Otherwise, t1 must <= t0

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

Comment By: Thomas M Zeng (zengtm)
Date: 2002-03-22 12:41

Message:
Logged In: YES 
user_id=20397

I just came across 513880, which is closely related.

That is to say, if timescale is positive, should the play 

range to be required to have t1 > t0?

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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=533691&group_id=23194


--__--__--

Message: 2
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Wed, 01 Oct 2003 08:22:04 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-513880 ] How to use Range when Scale is negative

Bugs item #513880, was opened at 2002-02-06 13:48
Message generated for change (Comment added) made by arvy_n
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=513880&group_id=23194

Category: None
>Group: Fix - Discussed/Resolved
Status: Open
Resolution: None
Priority: 5
Submitted By: Anders Klemets (klemets)
>Assigned to: Aravind Narasimhan (arvy_n)
Summary: How to use Range when Scale is negative

Initial Comment:
I think the spec could use an example that shows how 

to use the Range header when rewinding (Scale header 

is set to a negative value.)  Or, instead of an 

example, some explanatory text could be added.



The issue that I find a little bit confusing is how 

the server should specify the end of the range. The 

end of the range would be numerically smaller than the 

start of the range.



I think the following illustrates a valid exchange:



C->S:

PLAY rtsp://server/a RTSP/1.0

Scale: -1

Range: npt=45-



S->C:

RTSP/1.0 200 OK

Scale: -1

Range: npt=45-0



This is the type of example that I think would be nice 

to have in the spec.



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

>Comment By: Aravind Narasimhan (arvy_n)
Date: 2003-10-01 11:22

Message:
Logged In: YES 
user_id=437699

The following text has been added to the spec with regards

to this bug:



When indicating a negative scale for a reverse playback, the

Range header must indicate a decreasing range as described

in section 13.29.



Example of playing in reverse at 3.5 times normal rate:

    Scale: -3.5

    Range: npt=15-10

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

Comment By: Magnus Westerlund (magwes)
Date: 2002-10-30 12:29

Message:
Logged In: YES 
user_id=302620

This needs to be discussed.

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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=513880&group_id=23194


--__--__--

Message: 3
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Wed, 01 Oct 2003 08:36:09 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-663715 ] Requirement levels for Transport:SSRC needs clarification

Bugs item #663715, was opened at 2003-01-07 09:31
Message generated for change (Comment added) made by arvy_n
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=663715&group_id=23194

Category: None
Group: Fix - Discussed/Resolved
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
>Assigned to: Aravind Narasimhan (arvy_n)
Summary: Requirement levels for Transport:SSRC needs clarification

Initial Comment:
The SSRC parameter in the transport header has a few

unclarities.

- Why should a client be allowed to propose a SSRC to

the server?

- If allowed as current text indicates, what is the

requirement level on the server to follow the proposed

value. MAY or SHOULD.



First I would like to know the motivation a client has

to suggest a SSRC. This can probably not be removed but

unless there is good reasons for it I think the

requirement on a server to follow the request is MAY.

With a good motivation  the level is SHOULD. 



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

>Comment By: Aravind Narasimhan (arvy_n)
Date: 2003-10-01 11:36

Message:
Logged In: YES 
user_id=437699

To address this bug the following text has been added to

13.40 of bis04:



ssrc: The ssrc parameter, if included in a SETUP response,

indicates the RTP SSRC [23] value that will be used by the

media server for RTP packets within the stream. It is

expressed as an eight digit hexadecimal value. If the server

does not act as a synchronization source for stream data

(for instance, server is a translator, reflector, etc.) the

value will be the "packet sender's SSRC" that would have

been used in the RTCP Receiver Reports generated by the

server, regardless of whether the server actually generates

RTCP RRs. If there are multiple sources within the stream,

the ssrc parameter only indicates the value for a single

synchronization source. Other sources must be deduced from

the actual RTP/RTCP stream.



The functionality of specifying the ssrc parameter in a

SETUP request is deprecated as it is incompatible with the

specification of RTP in RFC 1889. If the parameter is

included in the transport header of a SETUP request, the

server MAY ignore it, and choose an appropriate SSRC for the

stream. It MAY set the ssrc parameter in the transport

header of the response.



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

Comment By: Magnus Westerlund (magwes)
Date: 2003-05-12 07:31

Message:
Logged In: YES 
user_id=302620

Plan is to deprecate the behavior that a client is allowed

to specify SSRC to the server. See telecon minutes from 28

April 2003. 



the behavior on how it shall be used will also be clarified.

See discussion on multiple SSRCs. 

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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=663715&group_id=23194


--__--__--

Message: 4
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Wed, 01 Oct 2003 08:38:13 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-663715 ] Requirement levels for Transport:SSRC needs clarification

Bugs item #663715, was opened at 2003-01-07 09:31
Message generated for change (Settings changed) made by arvy_n
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=663715&group_id=23194

Category: None
>Group: Fixed in CVS/Need WG Approval
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Aravind Narasimhan (arvy_n)
Summary: Requirement levels for Transport:SSRC needs clarification

Initial Comment:
The SSRC parameter in the transport header has a few

unclarities.

- Why should a client be allowed to propose a SSRC to

the server?

- If allowed as current text indicates, what is the

requirement level on the server to follow the proposed

value. MAY or SHOULD.



First I would like to know the motivation a client has

to suggest a SSRC. This can probably not be removed but

unless there is good reasons for it I think the

requirement on a server to follow the request is MAY.

With a good motivation  the level is SHOULD. 



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

Comment By: Aravind Narasimhan (arvy_n)
Date: 2003-10-01 11:36

Message:
Logged In: YES 
user_id=437699

To address this bug the following text has been added to

13.40 of bis04:



ssrc: The ssrc parameter, if included in a SETUP response,

indicates the RTP SSRC [23] value that will be used by the

media server for RTP packets within the stream. It is

expressed as an eight digit hexadecimal value. If the server

does not act as a synchronization source for stream data

(for instance, server is a translator, reflector, etc.) the

value will be the "packet sender's SSRC" that would have

been used in the RTCP Receiver Reports generated by the

server, regardless of whether the server actually generates

RTCP RRs. If there are multiple sources within the stream,

the ssrc parameter only indicates the value for a single

synchronization source. Other sources must be deduced from

the actual RTP/RTCP stream.



The functionality of specifying the ssrc parameter in a

SETUP request is deprecated as it is incompatible with the

specification of RTP in RFC 1889. If the parameter is

included in the transport header of a SETUP request, the

server MAY ignore it, and choose an appropriate SSRC for the

stream. It MAY set the ssrc parameter in the transport

header of the response.



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

Comment By: Magnus Westerlund (magwes)
Date: 2003-05-12 07:31

Message:
Logged In: YES 
user_id=302620

Plan is to deprecate the behavior that a client is allowed

to specify SSRC to the server. See telecon minutes from 28

April 2003. 



the behavior on how it shall be used will also be clarified.

See discussion on multiple SSRCs. 

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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=663715&group_id=23194


--__--__--

Message: 5
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Wed, 01 Oct 2003 08:38:17 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-533691 ] Clarify Range Where t0 <= t1

Bugs item #533691, was opened at 2002-03-22 12:36
Message generated for change (Settings changed) made by arvy_n
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=533691&group_id=23194

Category: None
>Group: Fixed in CVS/Need WG Approval
Status: Open
Resolution: None
Priority: 5
Submitted By: Thomas M Zeng (zengtm)
Assigned to: Aravind Narasimhan (arvy_n)
Summary: Clarify Range Where t0 <= t1

Initial Comment:
If the server receives a PLAY range from t0-t1, where

t0 < t1, should it reject the PLAY, or should the 

server play backwards as if the timescale is -1?



If t0 = t1, should the server send any media data 

after responding 200 OK?  

We've seen different servers behave differently in 

this scenario.



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

Comment By: Aravind Narasimhan (arvy_n)
Date: 2003-10-01 11:16

Message:
Logged In: YES 
user_id=437699

The following text has been added to the spec with regards

to this bug:



By default, range intervals increase, where the second point

is larger than the first point.



   Example:



       Range: npt=10-15



However, range intervals can also decrease if the Scale

header (see section  13.34) indicates a negative scale

value. For example, this would be the case when a playback

in reverse is desired.



   Example:



       Scale: -1

       Range: npt=15-10



Decreasing ranges are still half open intervals as described

above.  Thus, For range A-B, A is closed and B is open. In

the above example, 15 is closed and 10 is open. An exception

to this rule is the case when B=0 in a decreasing range. In

this case, the range is closed on both ends, as otherwise

there would be no way to reach 0 on a reverse playback.



   Example:



       Scale: -1

       Range: npt=15-0



   In this range both 15 and 0 are closed.



A decreasing range interval without a corresponding negative

Scale header is not valid.

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

Comment By: Thomas M Zeng (zengtm)
Date: 2003-03-05 12:53

Message:
Logged In: YES 
user_id=20397

If scale positive, t1 must >= t0.



Otherwise, t1 must <= t0

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

Comment By: Thomas M Zeng (zengtm)
Date: 2002-03-22 12:41

Message:
Logged In: YES 
user_id=20397

I just came across 513880, which is closely related.

That is to say, if timescale is positive, should the play 

range to be required to have t1 > t0?

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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=533691&group_id=23194


--__--__--

Message: 6
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Wed, 01 Oct 2003 08:38:59 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-513880 ] How to use Range when Scale is negative

Bugs item #513880, was opened at 2002-02-06 13:48
Message generated for change (Settings changed) made by arvy_n
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=513880&group_id=23194

Category: None
>Group: Fixed in CVS/Need WG Approval
Status: Open
Resolution: None
Priority: 5
Submitted By: Anders Klemets (klemets)
Assigned to: Aravind Narasimhan (arvy_n)
Summary: How to use Range when Scale is negative

Initial Comment:
I think the spec could use an example that shows how 

to use the Range header when rewinding (Scale header 

is set to a negative value.)  Or, instead of an 

example, some explanatory text could be added.



The issue that I find a little bit confusing is how 

the server should specify the end of the range. The 

end of the range would be numerically smaller than the 

start of the range.



I think the following illustrates a valid exchange:



C->S:

PLAY rtsp://server/a RTSP/1.0

Scale: -1

Range: npt=45-



S->C:

RTSP/1.0 200 OK

Scale: -1

Range: npt=45-0



This is the type of example that I think would be nice 

to have in the spec.



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

Comment By: Aravind Narasimhan (arvy_n)
Date: 2003-10-01 11:22

Message:
Logged In: YES 
user_id=437699

The following text has been added to the spec with regards

to this bug:



When indicating a negative scale for a reverse playback, the

Range header must indicate a decreasing range as described

in section 13.29.



Example of playing in reverse at 3.5 times normal rate:

    Scale: -3.5

    Range: npt=15-10

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

Comment By: Magnus Westerlund (magwes)
Date: 2002-10-30 12:29

Message:
Logged In: YES 
user_id=302620

This needs to be discussed.

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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=513880&group_id=23194


--__--__--

Message: 7
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Wed, 01 Oct 2003 08:50:12 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-513737 ] Unclarities when doing resetup

Bugs item #513737, was opened at 2002-02-06 08:26
Message generated for change (Settings changed) made by arvy_n
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=513737&group_id=23194

Category: None
Group: Fixed in CVS/Need WG Approval
Status: Open
Resolution: Fixed
Priority: 5
Submitted By: Magnus Westerlund (magwes)
>Assigned to: Aravind Narasimhan (arvy_n)
Summary: Unclarities when doing resetup

Initial Comment:
Setup is used to change transport parameters of a already setuped stream. What happends if the 

new SETUP contains errors? Except of responding with an error message, what happends to the 

session and transport state? It should be clarified in the specification. Which of the following 

alternatives are correct?



1. Keep the old setting and states as if the second SETUP never happend.

2. Invalidate the media in the session until a correct setup has happend.



To me case 1 seems more natural but will require more from the server. 

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

Comment By: Magnus Westerlund (magwes)
Date: 2003-05-15 04:09

Message:
Logged In: YES 
user_id=302620

A clarified third paragraph in the SETUP section (11.3) has

been added. If a setup request fails the previous state

shall be left as it was, i.e. proposal 1 in the initial

declartion of this bug. 





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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=513737&group_id=23194


--__--__--

Message: 8
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Wed, 01 Oct 2003 09:00:15 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-630920 ] RTP-Info does not support multiple SSRC

Bugs item #630920, was opened at 2002-10-30 02:44
Message generated for change (Comment added) made by arvy_n
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=630920&group_id=23194

Category: General
>Group: Fixed in CVS/Need WG Approval
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
>Assigned to: Aravind Narasimhan (arvy_n)
Summary: RTP-Info does not support multiple SSRC

Initial Comment:
The RTP-Info header can't be used to synchronize

multiple RTP flows using different SSRC as a single

media stream. As each SSRC has its own RTP sequence and

timestamp space to allow synchronization it needs to be

possible to give these values for each SSRC in the

media stream. 



This bug has connection with bug# 625352.



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

>Comment By: Aravind Narasimhan (arvy_n)
Date: 2003-10-01 12:00

Message:
Logged In: YES 
user_id=437699

The following text has been added to 13.33 of bis04 to

address this bug:



The RTP-Info response-header field is used to set

RTP-specific parameters in the PLAY response. These

parameters correspond to the synchronization source

identified by the ssrc parameter of the Transport response

header in the SETUP reponse. For streams using RTP as

transport protocol the RTP-Info header SHALL be part of a

200 response to PLAY.



. . .



Additionally, the RTP-Info header parameter fields only

apply to a single SSRC within a stream (the SSRC reported in

the transport response header; see section  13.40). If there

are multiple synchronization sources present within a RTP

session, RTCP must be used to map RTP and NTP timestamps for

those sources, for both synchronization and drift-checking.

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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=630920&group_id=23194




End of Rtspspec-bugs Digest


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I will be out of the office starting  10/11/2003 and will not return until
10/25/2003.

I will respond to your message when I return.

In case of urgent matters, you can contact Thierry Draus
<Thierry.Draus@ses-astra.com>



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From exim@www1.ietf.org  Mon Oct 13 13:51:25 2003
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From: Thomas Zeng <zeng@PacketVideo.COM>
To: "'Matthew Heaney'" <mheaney@on2.com>, mmusic@ietf.org
Subject: RE: [MMUSIC] specifying the multicast interface
Date: Mon, 13 Oct 2003 10:49:41 -0700
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Dear Matthew:

Yes, you can provide the IP of the selected IF via the c= line in SDP, but
the final say is upto RTSP negotiations, ie., you can use the
Transport header in the SETUP request/response to over-ride SDP. Transport
header has the following general parameters, quoting 2326:


   unicast | multicast:
          mutually exclusive indication of whether unicast or multicast
          delivery will be attempted. Default value is multicast.
          Clients that are capable of handling both unicast and
          multicast transmission MUST indicate such capability by
          including two full transport-specs with separate parameters
          for each.

   destination:
          The address to which a stream will be sent. The client may
          specify the multicast address with the destination parameter.
          To avoid becoming the unwitting perpetrator of a remote-
          controlled denial-of-service attack, a server SHOULD
          authenticate the client and SHOULD log such attempts before
          allowing the client to direct a media stream to an address not
          chosen by the server. This is particularly important if RTSP
          commands are issued via UDP, but implementations cannot rely
          on TCP as reliable means of client identification by itself. A
          server SHOULD not allow a client to direct media streams to an
          address that differs from the address commands are coming
          from.

   source:
          If the source address for the stream is different than can be
          derived from the RTSP endpoint address (the server in playback
          or the client in recording), the source MAY be specified.

     This information may also be available through SDP. However, since
     this is more a feature of transport than media initialization, the
     authoritative source for this information should be in the SETUP
     response.

Hope this is the answer you are looking for.

- thomas

-----Original Message-----
From: Matthew Heaney [mailto:mheaney@on2.com]
Sent: Friday, October 10, 2003 12:35 PM
To: mmusic@ietf.org
Subject: [MMUSIC] specifying the multicast interface


When creating a multicast session, it is possible to specify the
outgoing interface using the IP_MULTICAST_IF socket option.  Does RFC
2326 (or 3550/1889) prescribe a mechanism for specifying the interface?

Analogously, the IP_ADD_MEMBERSHIP socket option allows the receiver to
specify the multicast interface explicitly (via the imr_interface data
member of the ip_mreq struct).  Is there a mechanism (via the SDP
DESCRIBE, I guess) for indicating the multicast interface?

Thanks,
Matt


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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. PARTNERSHIP. (DENIS DUBE)

--__--__--

Message: 1
From: DENIS DUBE <dubedube88@netscape.net>
To: rtspspec-bugs@lists.sourceforge.net
Reply-To: dubedube88@netscape.net
Date: Mon, 13 Oct 2003 22:30:22 +0200
Subject: [rtspspec-bugs] PARTNERSHIP.


This is a multi-part message in MIME format

--9854268a-fdc0-11d7-9787-444553540000
Content-Type: text/plain; charset=iso-8859-1
Content-Transfer-Encoding: quoted-printable

FROM: DENIS DUBE
E-mail:dubedube88@netscape.net
TEL:+31-630-550-312
AMSTERDAM
Dear Sir,
                    FUNDS FOR INVESTMENT
I deem it necessary to first introduce myself before putting this proposal
to you for consideration.
I am Dr.Denis Dube a principal official managing, accounting, securing and =
payment of Sierra Leone contractors.
I however. was moved to write you this letter for a joint venture for =
profitable investment of a medium term capital worth TWENTY MILLION UNITED =
STATES DOLLARS ($20 MILLION)which we have within our authority for immediate =
investment.
It is vital to inform you that this fund in question does not fall within the =
category of laundered or drug related money, but rather my colleagues and I =
are principal officials responsible for managing, accounting,securing of =
funds for various projects in Sierra Leone.We have been able to temporarily =
set aside this amount which makes up only a little fraction of the funds =
schedule for use in the year 2009 in the last phase of the six years =
nationwide project of reconstruction and intallation of infrastructural =
facilities worth Nine Hundred and Twenty Five Million dollars ($925million) =
allocated for these projects.
The two major contracting firms TRADE PARTNERS UK AND TGS-NOPEC have been =
paid motivational amount and operations have commence while major balance =
payments are schedule for the last quaters of 2009 when the last phase of the =
projects would be completed. Therefore, considering the fact that these funds =
wounld be lying idle for six yars, we decided to take out ($20million)and we =
are required to return this amount within six years back to the treasury when =
we are expected to present our audit report.
We hereby ask you to use your vast expertise in sourcing and negotiating for =
the project that you consider most lucrative, As a matter of fact we have =
successfully moved this funds to a security and finance company in Amsterdam, =
The Netherlands for safe custody and I am currently in Amsterdam for the =
purpose of securing a would be investor/partner. In the mean time, could you =
indicate your willingness to be part of this proposal by acknowledging this =
mail via e-mail or telephone number +31-630-550-312 so that we can discuss =
contructivelly all relevant details and evaluation of your proposed =
investment.
I await your immidiate response.
Kind regards,
Denis Dube(Dr)   
--9854268a-fdc0-11d7-9787-444553540000--





End of Rtspspec-bugs Digest


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Hi

Time again for RTSP telecon. Sorry for being late. You are invited
to participate in a conference the 14th of October at 18.00 CET.

The details to participate.

Telephone number: +1 972 583 9950
Participant PIN   74373
Date   10/14/2003 (MM/DD/YYYY)
Start Time   18:00 (HH:MM)  (CET)


The agenda
----------

1. Opening of meeting.

2. Administrative
    - Scribe
    - IETF in Minneapolis
      * Off-line Meeting?
      * What to present.

3. Since last meeting (1st of October).
a. http://rtsp.org/bug507882 - RTP timestamp and pause
     - Aravind to write up text

b. http://rtsp.org/bug742328 - How to handle PLAY request when at the
end of a media clip
     - Text sent to mailing list, please review.

c. http://rtsp.org/bug742348 - Usage of implicit REDIRECT.
     - Needs text

d. http://rtsp.org/bug640272 - Default way of deriving aggregate URL
    - Text sent to the mailing list

e. http://rtsp.org/bug771638 - a=range in SDP to indicate live media
    - Text sent to the mailing list.

f. http://rtsp.org/bug531189 - How does REDIRECT method work
    - Needs somebody to write up.

g. http://rtsp.org/bug810678 - How to indicate end of media stream
    - Strawman proposal available from Thomas. For discussion.

h. http://rtsp.org/bug770522 - Request to response time is not limited.
    - Some discussion on mailing list. Sean to write up text.

i. http://rtsp.org/bug810681 - 	The updated specification version number
    - Awaiting investigation results regarding changes by Aravind.

j. http://rtsp.org/bug795295 - TEARDOWN using * and the Ready-nm state
    - Posted to the mailing list, no comments. Unless comments are
      received it will be edited in.

k. http://rtsp.org/bug804219 - How to handle sync in re-SETUP
    - Text posted to the list. No comments received so far.

l. http://rtsp.org/bug806404 - Re-SETUP of transport parameters
    - Awaiting proposal text from Geetha.

m. http://rtsp.org/bug810666 - Removal of SETUP in play state
    - Proposed to be removed. Message sent to the list.

4. New Issues

a. http://rtsp.org/bug811815 - Use of c= in SDP to know possible address 
classes

b. http://rtsp.org/bug795314 - Caching of OPTIONS request

5. Any other business

6. Next Telecon: Proposal Wednesday, 3rd of December.

7. End of meeting

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 exim@www1.ietf.org  Tue Oct 14 03:38:27 2003
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [MMUSIC] RTSP: Bug 649272 - Default way of deriving aggregate URL
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Hi,

This is the proposed text in regards bug:

http://rtsp.org/bug640272 - Default way of deriving aggregate URL

The text is from the SETUP method description:

"The server {\MUST} generate a session identifier in response to a
successful {\SETUP} request, unless a {\SETUP} request to a server
includes a session identifier, in which case the server {\MUST}
bundle this setup request into the existing session (aggregated
session) or return error 459 (Aggregate Operation Not Allowed)
(see Section ~\ref{sec:error459}). An Aggregate control URI
{\MUST} be used to control an aggregated session. This URI {\MUST}
be different from the stream control URIs of the individual media
streams included in the aggregate. The Aggregate control URI
{\SHOULD} be specified by the session description since there is
no general rule for deriving it from the various stream control
URIs in the session.

If an Aggregate control URI is not specified in the session
description, it is probably a indication that non-aggregated
control should be used. However a client {\MAY} try to SETUP the
session in aggregated control. If the server refuse to aggregate
the specified media, the server {\SHALL} use the 459 error code.
If the server allows the aggregation, then the client {\MUST}
create an URI for aggregate control of the session. This URI
{\MUST} contain the servers host address and {\MUST} contain the
port, if applicable (e.g. not default port). Once an URI is used
to refer to an aggregation for a given session, that URI {\MUST}
be used to refer to that aggregation for the duration of the
session."


Please comment


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 exim@www1.ietf.org  Tue Oct 14 03:41:24 2003
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Subject: [MMUSIC] RTSP: bug 742328 - How to handle PLAY request when at the end of
 a media clip
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Hi,

This is the text proposal for solving this bug:

http://rtsp.org/bug742328 -
How to handle PLAY request when at the end of a media clip

The text belongs to the PLAY method description:

"A client desiring to play the media from the beginning {\MUST}
send a {\PLAY} request with a \header{Range} header pointing at
the beginning, e.g. npt=0-. If a {\PLAY} request is received
without a \header{Range} header when media delivery has stopped at
the end, the server {\SHOULD} respond with a 457 "Invalid Range"
error response. In that response the current pause point in a
\header{Range} header {\SHALL} be included."


Please comment!

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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Today, the voice over WLAN market is an extremely small and immature =
market but in the longer term, all users of WLAN networks could =
eventually find a business case within the context of interoperability =
between WLANs and 2.5G and 3G networks.=20
The 2004 WiFi Voice Conference will be held in Paris on May 25 to 28, =
2004.=20

A call for papers has been launched on-line, effective until November =
15, 2003.=20
More details at:
http://www.upperside.fr/wifivoice04/wifivoice04intro.htm

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<DIV><FONT size=3D2>Today, the voice over WLAN market is an extremely =
small and=20
immature market but in the longer term, all users of WLAN networks could =

eventually find a business case within the context of interoperability =
between=20
WLANs and 2.5G and 3G networks. <BR></FONT><FONT size=3D2>The <SPAN=20
class=3Dtextebold><FONT color=3D#2a9085>2004 WiFi Voice =
Conference</FONT></SPAN>=20
will be held in Paris on <SPAN class=3Dtextebold>May 25 to 28, =
2004</SPAN>.=20
<BR><BR>A call for papers has been launched on-line, effective until =
<SPAN=20
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Subject: [MMUSIC] RTSP: Bug 771638 - a=range in SDP to indicate live media
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Hi,

The below text is proposed for resolving the the following bug:

http://rtsp.org/bug771638 - a=range in SDP to indicate live media

 From Annex C:

"
Range of Presentation

The ``a=range'' attribute defines the total time range of the
stored session or an individual media. That Live sessions are
non-seekable can also be indicated, while the length of live
sessions can be deduced from the ``t'' and ``r'' parameters.

The attribute is both a session and a media level attribute. For
presentations that contains media streams of the same durations,
the range attribute {\SHOULD} only be used at session-level. In
case of different length the range attribute {\MUST} be given at
media level for all media. If the attribute is present at media
level and session level the media level values {\SHALL} be used.

The unit is specified first, followed by the value range. The
units and their values are as defined in Section \ref{sec:smpte},
\ref{sec:npt} and \ref{sec:clock}. Any open ended range (start-),
i.e. without stop range, is of unspecified duration and {\SHALL}
be considered as non-seekable content unless this property is
overridden.


This attribute is defined in ABNF \cite{rfc2234} as:

a-range-def = "a" "=" "range" ":" ranges-specifier CRLF



Examples:

   a=range:npt=0-34.4368
   a=range:clock=19971113T2115-19971113T2203
   Non seekable stream of unknown duration:
   a=range:npt=0-

"

Any comments should be given before wednesday the 22nd of October.

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: [MMUSIC] RTSP: Minutes telecon 14th of October 2003
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Hi,

Here is the preliminary minutes for todays telecon. Please comment.

RTSP Telecon the 14th of October

Participants:
Magnus Westerlund, Colin Perkins, Aravind Narasimhan, and Tom Marshall.

Agenda:

1. Opening of meeting.

2. Administrative
    - Scribe
    - IETF in Minneapolis
      * Off-line Meeting?
      * What to present.

3. Since last meeting (1st of October).
a. http://rtsp.org/bug507882 - RTP timestamp and pause
     - Aravind to write up text

b. http://rtsp.org/bug742328 - How to handle PLAY request when at the
end of a media clip
     - Text sent to mailing list, please review.

c. http://rtsp.org/bug742348 - Usage of implicit REDIRECT.
     - Needs text

d. http://rtsp.org/bug640272 - Default way of deriving aggregate URL
    - Text sent to the mailing list

e. http://rtsp.org/bug771638 - a=range in SDP to indicate live media
    - Text sent to the mailing list.

f. http://rtsp.org/bug531189 - How does REDIRECT method work
    - Needs somebody to write up.

g. http://rtsp.org/bug810678 - How to indicate end of media stream
    - Strawman proposal available from Thomas. For discussion.

h. http://rtsp.org/bug770522 - Request to response time is not limited.
    - Some discussion on mailing list. Sean to write up text.

i. http://rtsp.org/bug810681 -     The updated specification version number
    - Awaiting investigation results regarding changes by Aravind.

j. http://rtsp.org/bug795295 - TEARDOWN using * and the Ready-nm state
    - Posted to the mailing list, no comments. Unless comments are
      received it will be edited in.

k. http://rtsp.org/bug804219 - How to handle sync in re-SETUP
    - Text posted to the list. No comments received so far.

l. http://rtsp.org/bug806404 - Re-SETUP of transport parameters
    - Awaiting proposal text from Geetha.

m. http://rtsp.org/bug810666 - Removal of SETUP in play state
    - Proposed to be removed. Message sent to the list.

4. New Issues

a. http://rtsp.org/bug811815 - Use of c= in SDP to know possible address 
classes

b. http://rtsp.org/bug795314 - Caching of OPTIONS request

5. Any other business

6. Next Telecon: Proposal Wednesday, 3rd of December.

7. End of meeting

Minutes
-------

1. Meeting was opened.

2. - Magnus volunteered to write the minutes.

    - IETF meeting in Minneapolis:
      * Colin will request a conference room for a out-side of normal
        agenda meeting. A guess of on the number of participants are
        around 15.

      * The authors will perform a status update on the specification in
        the WG session. The current resolutions and open issues will be
        presented.

3.

a. Nothing has happened.

b. Await review from the WG.

c. Status not changed.

d. Await review from the WG.

e. Was not sent yet, but will be available directly after meeting.
Review comments needed.

f. The remaining issue is with implicit closing of the session. The
    previous meeting has agreement on the functionality but some minor
    issues around use of "connection" needs resolving. Best way of
    resolving this is probably to present a specification text proposal.
    Aravind will take this as a second priority issue.

g. Send comments on the document to list. The discussion resolved that
    this can be divided into several pieces:

    - We need to document in the RTSP specification regarding RTP what
      happens when a source times out a SSRC or receives a RTCP BYE for a
      source.

    - Using BYE is not suitable way to indicate the end of the stream.
      The motivation for this can be seen in Thomas proposal.

    - Look into SIP to see how they have resolved the discrepancies
      between media and signaling channel.

h. Awaits text from Sean.

i. This is a later decision when we have received a list of changes that
    are accurate.

j. Unless comments are received by Monday the 20th of October, this
    resolution will be edited in. Magnus will post a reply to the initial
    mail pointing out the deadline.

k. Will be kept in the text unless comments are review by Wednesday the
    22th of October.

l. Awaits text.

m. Will be edited in unless comments are received by Monday the 20th of
    October.

4.
a. Question raised is: Are there need for a server to announce through
    SDP or other pre SETUP mechanism which address classes it supports to
    the client?

    There was discussion on this topic and the resolution was the
    following:

    It should be assumed that a server being possible to find in DNS with
    a certain address class will support both signaling and media
    delivery for that address. Thus a RTSP request over TCP/IPv6 will
    send media using that source IPv6 address.

b. The OPTIONS request has some dependency on server state and what
content that are addressed. Therefore it seems doubtful that caching of
OPTIONS request are possible. In general it seems that the only
really cachable RTSP request/response pair is for DESCRIBE.

5. No other business.

6. Preliminary date is the 3rd of December, which is three weeks after
the IETF meeting.

7. Meeting was closed.



-- 

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,

Please provide any comments by Wednesday the 22th of October, or the 
below text will be part of the next version.

/Magnus

Magnus Westerlund wrote:

> Hi,
> 
> This is the text proposal for solving this bug:
> 
> http://rtsp.org/bug742328 -
> How to handle PLAY request when at the end of a media clip
> 
> The text belongs to the PLAY method description:
> 
> "A client desiring to play the media from the beginning {\MUST}
> send a {\PLAY} request with a \header{Range} header pointing at
> the beginning, e.g. npt=0-. If a {\PLAY} request is received
> without a \header{Range} header when media delivery has stopped at
> the end, the server {\SHOULD} respond with a 457 "Invalid Range"
> error response. In that response the current pause point in a
> \header{Range} header {\SHALL} be included."
> 
> 
> Please comment!
> 
> 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
> 
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 

-- 

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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Date: Tue, 14 Oct 2003 20:06:14 +0200
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To: IETF MMUSIC WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] RTSP: Bug 649272 - Default way of deriving aggregate
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References: <3F8BA79F.4090604@ericsson.com>
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Hi,

Please provide comments by Wednesday the 22th of October, or this text 
will be part of the next draft version.

Cheers

Magnus

Magnus Westerlund wrote:

> Hi,
> 
> This is the proposed text in regards bug:
> 
> http://rtsp.org/bug640272 - Default way of deriving aggregate URL
> 
> The text is from the SETUP method description:
> 
> "The server {\MUST} generate a session identifier in response to a
> successful {\SETUP} request, unless a {\SETUP} request to a server
> includes a session identifier, in which case the server {\MUST}
> bundle this setup request into the existing session (aggregated
> session) or return error 459 (Aggregate Operation Not Allowed)
> (see Section ~\ref{sec:error459}). An Aggregate control URI
> {\MUST} be used to control an aggregated session. This URI {\MUST}
> be different from the stream control URIs of the individual media
> streams included in the aggregate. The Aggregate control URI
> {\SHOULD} be specified by the session description since there is
> no general rule for deriving it from the various stream control
> URIs in the session.
> 
> If an Aggregate control URI is not specified in the session
> description, it is probably a indication that non-aggregated
> control should be used. However a client {\MAY} try to SETUP the
> session in aggregated control. If the server refuse to aggregate
> the specified media, the server {\SHALL} use the 459 error code.
> If the server allows the aggregation, then the client {\MUST}
> create an URI for aggregate control of the session. This URI
> {\MUST} contain the servers host address and {\MUST} contain the
> port, if applicable (e.g. not default port). Once an URI is used
> to refer to an aggregation for a given session, that URI {\MUST}
> be used to refer to that aggregation for the duration of the
> session."
> 
> 
> Please comment
> 
> 
> 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
> 
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 

-- 

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: IETF MMUSIC WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] RTSP: Proposal to remove ready-nm state
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Hi,

Unless any comments are received by Monday the 20th of October the 
proposed changes will be edited in into the next draft version.

Cheers

Magnus

Magnus Westerlund wrote:

> Hi,
> 
> I have suggested and got support in yesterdays discussion to remove the 
> ready-nm state from the RTSP state machine and the functionality 
> required to make this state work. See below link for further comments on 
> this.
> http://rtsp.org/bug795295
> 
> Unless anyone has any comment about this within a week or so I will 
> start removing it from the text.
> 
> 
> 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
> 

-- 

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,

Unless any further comments are received by Monday the 20th of October 
this change will be edited in.

Cheers

Magnus

Magnus Westerlund wrote:

> Hi,
> 
> Another issue that was discussed on yesterdays telecon was the removal 
> of the part specifying that usage of setup and teardown on individual 
> media stream in PLAY state is possible. This does not effect the 
> functionality for:
> 
> - SETUP and Teardown in ready state
> - The usage of SETUP to change transport parameters.
> - Usage of aggregated Teardown in play state to end session.
> 
> The removal of this functionality will be done in such way that it will 
> be possible in the future to specify this as extension functionality.
> 
> If no comments are received on this within a week I will start making 
> these changes. And comments for keeping this functionality should state 
> a strong use cases for this.
> 
> See also:
> http://rtsp.org/bug810666
> 
> 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
> 

-- 

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: "Joel Tran" <joel.tran@USherbrooke.ca>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        "'Peterson, Jon'" <jon.peterson@neustar.biz>
Cc: "'Mmusic'" <mmusic@ietf.org>
Subject: RE : [MMUSIC] Status of TURN
Date: Tue, 14 Oct 2003 15:07:28 -0400
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Thanks for your answers concerning TURN.

...J

Here are few comments and questions regarding TURN after my first =
reading.=20

I would also like to suggest to put the section 9 concerning the =
"Protocol
Details" before the section 7 concerning the "Server Behavior". In the
section 7 & 8, some protocol specific messages are being used with prior
introduction on the TURN messages/protocol. It is thus difficult to read =
the
document. With a proper introduction on the messages of the protocol, =
the
document will be easier to read.

[PAGE 3]  =20
   A complete solution requires a means by which a client can obtain a
   transport address from which it can receive media from any peer which
   can send packets to the public Internet. This can only be
   accomplished by relaying data though a server that resides on the
   public Internet. This specification describes Traversal Using Relay
   NAT (TURN), a protocol that allows a client to obtain IP addresses
   and ports from such a relay.

Joel : Not true. A relay server can reside in a private network (i.e. =
full
media proxy server/by setting a virtual server provided by NAT routers).

[Page 5]
      TURN Client: A TURN client (also just referred to as a client) is
      an entity that generates TURN requests. A TURN client can be an
      end system, such as a Session Initiation Protocol (SIP) [6] User
      Agent, or can be a network element, such as a Back-to-Back User
      Agent (B2BUA) SIP server. The TURN protocol will provide the STUN
                                                                   ^^^^
      client with IP addresses that route to it from the public
      Internet.
Joel : Did you mean TURN clien or really STUN client? TURN and STUN
request/protocol are not the same and thus can not respond to it.

[Page 6]
   When a client is behind a NAT, transport addresses obtained from the
   local operating system will not be publically routable, and
   therefore, not useful in these protocols. TURN allows a client to
   obtain a transport address, from a server on the public Internet,
   which can be used in protocols meeting the above criteria. However,
   the transport addresses obtained from TURN servers are not generally
                                                              ^^^^^^^^^
   useful for receiving data from anywhere.
Joel : It is ambigious to say generally. This implies that TURN purpose =
is
not ment to receive data from anywhere but can be used to receive data =
from
anywhere.

[Page 7]
   A TURN client first discovers the address of a TURN server. This can
   be preconfigured, or it can be discovered using SRV records [3] This
                                                                  ^^
   will allow for different TURN servers for UDP and TCP.
Joel : . Missing!
Joel : Phrase unclear : This will allow for different TURN servers for =
UDP
and TCP. Should it be something like : This will allow the use of =
different
TURN server for UDP and TCP.


[Page 8]

   transport address, PA, in the TURN response. The TURN server is
   responsible for guaranteeing that packets sent to PA route to the
   TURN server.
Joel : The PA is the TURN Server. How can you send data from PA to the =
TURN
Server? Should this phase be something like :  The TURN Server is then
responsible for guaranteeing that packets sent to PA route get to SA.


   TURN also allows a client to request an odd or even port when one is
   allocated, and for it to pre-allocate the next higher port. This is
   useful for securing consecutive ports for usage with the Real Time
   Transport Protocol (RTP) [5].
Joel : The phrase is unclear, the two cases should be separated to make =
it
clear: First, request for an odd or event port at the allocation time.
Second, Request for a subsequent pre-allocated port.

[Section 7.1]
Joel: It will proabaly ease the reading of this section if you list and
describe all the ERROR-CODE in a table. Furthermore, you can probably do =
a
use case diagram to ease the reading of this section.

[Page 11]
   The server proceeds to authorize the client. The means for
   authorization are outside the scope of this specification. It is
   anticipated that TURN servers will be run by providers that also
   provide an application service, such as SIP or RTSP. In that case, a
   user would be authorized to use TURN if they are authorized to use
   the application service.
Joel: You should point to the document which solves the issue concerning =
the
authorization.

[Page 12]
   the case in TURN. In TURN, the server MUST store the allocated
   username and password for a duration of at least 30 minutes. Once an
   Allocate request has been authenticated using that username and
   password, if the result was an Allocate Error Response, the username
   and password are discarded. If the result was an Allocate Response,
   resulting in the creation of a new binding, the username and password
   become associated with that binding. They can only be used to
   authenticate Allocate requests sent from the same source transport
   address in order to refresh or de-allocate that binding. Once the
   binding is deleted, the username and password are discarded.

Joel: How long should the binding in TURN last last? When does the =
binding
expires? This paragraph is concerning Share secret. It should not =
include
elements on Allocate Request. This could be move to the next section.


   Allocate requests are used to obtain an IP address and port that the
   client can use to receive UDP and TCP packets from any host on the
   network, even when the client is behind a symmetric NAT. To do this,
   a TURN server allocates a local transport address, and passes it to
   the client in an Allocate Response. When the server receives packets
   on this allocated address, it acts as a relay, and forwards them
   towards the source of the Allocate request. The server remembers the
   source transport address where that packet came from, and "locks
   down". This means that packets sent from the client to the TURN
   server are forwarded to that address.
Joel: "Locks down" what? The phrase is unclear. Locks down the transport
address from the source packet.

[Page 13]
   provided by the server. Typically, this will be XXXX, the default
   TURN port. [[OPEN ISSUE: Can we just use the STUN port?]].
Joel: I wonder if there is some security issue if we use the same port =
as
STUN. For example, how do we treat purely STUN user without =
authentication?
How STUN and TURN server can interoperate if they are not merge into a
single daemon?

   The server MUST check the Allocate Request for a MESSAGE-INTEGRITY
   attribute. If not present, the server generates a Allocate Error
   Response with an ERROR-CODE attribute with response code 401.
Joel: Another table combined with a use case will ease the reading of =
this
section.

[Page 14]
   determines whether or not it has sufficient capacity to handle a
   binding that will generate the requested bandwidth. If so, the server
   looks for the presence of the TRANSPORT-PREFERENCES attribute in the
   request. If the attribute indicates that an even port is requested,
   the server attempts to allocate a transport address with an even
   port. If the attribute indicates that an odd port is requested, the
   server attempts to allocate a transport address with an odd port. If
   the attribute indicates that there is no preference for port parity,
   or if the TRANSPORT-PREFERENCES attribute was absent, the server
   attempts to allocate a port with either parity. The server MUST NOT
   allocate ports from the well-known port range (0-1023) and MUST NOT
   allocate ports from the user registered port range (1024 through
   49151).
Joel : You should introduce the two elements TURN are looking for at =
first
and then describing them to ease the reading.=20


[Page 15]

   Request (TCP or UDP), the source IP address and port of the allocate
   five-tuple are set to the source IP address and port in the Allocate
   Request, and the destination IP address and port of the allocate
   five-tuple are set to the destination IP address and port in the
   Allocate Request). The protocol in the remote five-tuple is set to
   the protocol from the Allocate Request. The source IP address of the
   remote five-tuple is set to the interface from which the base port
   was allocated. The source port of the remote five-tuple is set to the
   base port. If the binding was allocated for TCP, the connection on
   which the Allocate request was received is associated with the
   allocate five-tuple in the binding.


   The server MUST remember the one-time username and password used to
   obtain the binding.

Joel: You should probably mention that in the remote tuples, the remote
transport address is left blank.

Joel : A little suggestion, at the creation of the request message, a 3 =
way
binding should be created with the one-time username/password, the =
allocate
and remote five-tuples. When a packet is received at the transport =
address,
the binding is changed to only the two five-tuples.

Joel: There is no case threating the case were, in the two five-tuples, =
the
protocol of the remote five-tuples does not match the protocol of the
allocate five-tuples. For example, what happen when TCP is used in
combination with UDP? Personally, I think an error should occur since it =
is
not the purpose of TURN to make such things as UDP over TCP or TCP over =
UDP.

[Page 16]

   The server MUST add a BANDWIDTH attribute to the Allocate Response.
   This MUST be equal to the attribute from the request, if one was
   present. Otherwise, it indicates a per-binding cap that the server is
   placing on the bandwidth usage on each binding. Such caps are needed
   to prevent against denial-of-service attacks (See Section 10.
Joel: Closing ) mising.

[Page 17]
   present, the server checks that this address and port has been
   pre-allocated by a previous Allocate Request. The only user
   authorized to allocate a pre-allocated address is the same one that
   generated the pre-allocation. Note that the one-time usernames for
   both requests (the pre-allocation and the final allocation) will be
   different. However, both MUST have been obtained through Shared
   Secret Requests authenticated as being sent from the same user.
Joel: Which error code is sent if the pre-allocated is not generated by =
the
user who generated the pre-allocation?


   Once a binding has been created, packets received from the client are
   generally forwarded to the remote client. However, if the binding is
   UDP, the client can send subsequent Allocate requests to the TURN
   server. To determine which packets are for the TURN server, and which
   need to be relayed, the server looks at the packet. If the packet is
   shorter than 28 bytes, it is not a TURN request. If it is longer than
   28 bytes, the server checks bytes 25-28. If these bytes are equal to
   the MAGIC-COOKIE, the request is a TURN request. Otherwise, it is a
   data packet, and is to be relayed.
Joel : Why between 25-28? This can be problematic in the future if some
extension is added to a TURN messages. I'm not familliar with Ipv6 but =
is
there some issues with IPv6 adress releam? I understand that this is =
within
the mecanism to determine which packet are TURN packet and which packet =
are
relaying packet.

All comments are welcome!


-----Message d'origine-----
De : mmusic-admin@ietf.org [mailto:mmusic-admin@ietf.org] De la part de
Jonathan Rosenberg
Envoy=E9 : 9 octobre, 2003 12:44
=C0 : Peterson, Jon
Cc : 'Joel Tran'; Mmusic
Objet : Re: [MMUSIC] Status of TURN


Thanks for the clarification, Jon.

To answer Joel's second question - there were some mixups during the=20
submission process last time, and the -02 never got into the archives.=20
I've just resubmitted it so it should be there shortly.

I am also hoping to get in a real revision (resulting in a -03=20
version) before the deadline. Please, if you have comments or=20
questions, forward them to me.

Thanks,
Jonathan R.

Peterson, Jon wrote:

> The Transport Area Directors understand that this document will=20
> proceed as an individual submission, outside the context of any of the =

> working groups in the Transport Area. It is clearly of interest to a=20
> number of these working groups.
>=20
> I gather that at least one more revision needs to be undertaken before =

> this document will be considered for publication as an RFC, but we're=20
> very interested to see it go forward.
>=20
> Jon Peterson
> NeuStar, Inc.
> (co-AD for Transport)
>=20
>=20
>>-----Original Message-----
>>From: Joel Tran [mailto:joel_tran@hotmail.com]
>>Sent: Tuesday, October 07, 2003 11:08 AM
>>To: Jonathan Rosenberg; Mmusic
>>Subject: [MMUSIC] Status of TURN
>>
>>
>>Hi Jonathan,
>>
>>I would like to ask you two little questions concerning TURN.
>>
>>First, I would like to know what is the exact status of TURN?
>>Will it be
>>adopted as a Workgroup item of MMUSIC since TURN is refeered=20
>>in various
>>documents?
>>
>>Second, I would like to ask you if there is a reason why the
>>lastest version
>>of the draft is not on the IETF web and only available throught
>>http://www.jdrosen.net/papers/draft-rosenberg-midcom-turn-02.txt?
>>
>>Thanks,
>>...J
>>
>>_______________________________________________
>>mmusic mailing list
>>mmusic@ietf.org
>>https://www1.ietf.org/mailman/listinfo/mmusic
>>
>=20
>=20

--=20
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 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
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https://www1.ietf.org/mailman/listinfo/mmusic




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



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Subject: RE: [MMUSIC] RTSP: Minutes telecon 14th of October 2003
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On item g:

    - Look into SIP to see how they have resolved the discrepancies
      between media and signaling channel.

Could someone fammiliar with SIP help me out here: I need to know what
method of SIP to look, is it SIP BYE method by any chance?

(I saw that SIP has a BYE method to terminate the call in the signalling
channel. )

thanks,
thomas



-----Original Message-----
From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
Sent: Tuesday, October 14, 2003 11:04 AM
To: IETF MMUSIC WG
Subject: [MMUSIC] RTSP: Minutes telecon 14th of October 2003


Hi,

Here is the preliminary minutes for todays telecon. Please comment.

RTSP Telecon the 14th of October

Participants:
Magnus Westerlund, Colin Perkins, Aravind Narasimhan, and Tom Marshall.

Agenda:

1. Opening of meeting.

2. Administrative
    - Scribe
    - IETF in Minneapolis
      * Off-line Meeting?
      * What to present.

3. Since last meeting (1st of October).
a. http://rtsp.org/bug507882 - RTP timestamp and pause
     - Aravind to write up text

b. http://rtsp.org/bug742328 - How to handle PLAY request when at the
end of a media clip
     - Text sent to mailing list, please review.

c. http://rtsp.org/bug742348 - Usage of implicit REDIRECT.
     - Needs text

d. http://rtsp.org/bug640272 - Default way of deriving aggregate URL
    - Text sent to the mailing list

e. http://rtsp.org/bug771638 - a=range in SDP to indicate live media
    - Text sent to the mailing list.

f. http://rtsp.org/bug531189 - How does REDIRECT method work
    - Needs somebody to write up.

g. http://rtsp.org/bug810678 - How to indicate end of media stream
    - Strawman proposal available from Thomas. For discussion.

h. http://rtsp.org/bug770522 - Request to response time is not limited.
    - Some discussion on mailing list. Sean to write up text.

i. http://rtsp.org/bug810681 -     The updated specification version number
    - Awaiting investigation results regarding changes by Aravind.

j. http://rtsp.org/bug795295 - TEARDOWN using * and the Ready-nm state
    - Posted to the mailing list, no comments. Unless comments are
      received it will be edited in.

k. http://rtsp.org/bug804219 - How to handle sync in re-SETUP
    - Text posted to the list. No comments received so far.

l. http://rtsp.org/bug806404 - Re-SETUP of transport parameters
    - Awaiting proposal text from Geetha.

m. http://rtsp.org/bug810666 - Removal of SETUP in play state
    - Proposed to be removed. Message sent to the list.

4. New Issues

a. http://rtsp.org/bug811815 - Use of c= in SDP to know possible address 
classes

b. http://rtsp.org/bug795314 - Caching of OPTIONS request

5. Any other business

6. Next Telecon: Proposal Wednesday, 3rd of December.

7. End of meeting

Minutes
-------

1. Meeting was opened.

2. - Magnus volunteered to write the minutes.

    - IETF meeting in Minneapolis:
      * Colin will request a conference room for a out-side of normal
        agenda meeting. A guess of on the number of participants are
        around 15.

      * The authors will perform a status update on the specification in
        the WG session. The current resolutions and open issues will be
        presented.

3.

a. Nothing has happened.

b. Await review from the WG.

c. Status not changed.

d. Await review from the WG.

e. Was not sent yet, but will be available directly after meeting.
Review comments needed.

f. The remaining issue is with implicit closing of the session. The
    previous meeting has agreement on the functionality but some minor
    issues around use of "connection" needs resolving. Best way of
    resolving this is probably to present a specification text proposal.
    Aravind will take this as a second priority issue.

g. Send comments on the document to list. The discussion resolved that
    this can be divided into several pieces:

    - We need to document in the RTSP specification regarding RTP what
      happens when a source times out a SSRC or receives a RTCP BYE for a
      source.

    - Using BYE is not suitable way to indicate the end of the stream.
      The motivation for this can be seen in Thomas proposal.

    - Look into SIP to see how they have resolved the discrepancies
      between media and signaling channel.

h. Awaits text from Sean.

i. This is a later decision when we have received a list of changes that
    are accurate.

j. Unless comments are received by Monday the 20th of October, this
    resolution will be edited in. Magnus will post a reply to the initial
    mail pointing out the deadline.

k. Will be kept in the text unless comments are review by Wednesday the
    22th of October.

l. Awaits text.

m. Will be edited in unless comments are received by Monday the 20th of
    October.

4.
a. Question raised is: Are there need for a server to announce through
    SDP or other pre SETUP mechanism which address classes it supports to
    the client?

    There was discussion on this topic and the resolution was the
    following:

    It should be assumed that a server being possible to find in DNS with
    a certain address class will support both signaling and media
    delivery for that address. Thus a RTSP request over TCP/IPv6 will
    send media using that source IPv6 address.

b. The OPTIONS request has some dependency on server state and what
content that are addressed. Therefore it seems doubtful that caching of
OPTIONS request are possible. In general it seems that the only
really cachable RTSP request/response pair is for DESCRIBE.

5. No other business.

6. Preliminary date is the 3rd of December, which is three weeks after
the IETF meeting.

7. Meeting was closed.



-- 

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
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Subject: RE: [MMUSIC] RTSP: Bug 771638 - a=range in SDP to indicate live media
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From: "Jae-Hwan Kim" <kjh1905m@nextreaming.com>
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        "IETF MMUSIC WG" <mmusic@ietf.org>
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Hi,

Just a simple question regarding range header although it is far from an
live indicating.
I saw an example, 2 range headers at session-level SDP because the
original content contains a=3Drange in its hinted SDP area: the first =
one
is 'a=3Drange:npt=3D0-620' server made and the other is
'a=3Drange:npt=3D0-300'(or npt=3D0-) originally from the server; in case =
of
two different range attributes exist. Almost all hinted server just
seems to get and forward SDP and RTP data.

How can we, as a player, understand this SDP?=20
Just I think the possible and right behavior is as followings:
1) If there are two or more a=3Drange header at a session-level or
media-level, the maximum one shall be used.
2) But if there is npt=3D0-, it means also this content is not seekable.

I would appreciate it for your advice.

Cheers,
Jae-Hwan


> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]=20
> Sent: Wednesday, October 15, 2003 2:37 AM
> To: IETF MMUSIC WG
> Subject: [MMUSIC] RTSP: Bug 771638 - a=3Drange in SDP to=20
> indicate live media
>=20
>=20
> Hi,
>=20
> The below text is proposed for resolving the the following bug:
>=20
> http://rtsp.org/bug771638 - a=3Drange in SDP to indicate live media
>=20
>  From Annex C:
>=20
> "
> Range of Presentation
>=20
> The ``a=3Drange'' attribute defines the total time range of the=20
> stored session or an individual media. That Live sessions are=20
> non-seekable can also be indicated, while the length of live=20
> sessions can be deduced from the ``t'' and ``r'' parameters.
>=20
> The attribute is both a session and a media level attribute.=20
> For presentations that contains media streams of the same=20
> durations, the range attribute {\SHOULD} only be used at=20
> session-level. In case of different length the range=20
> attribute {\MUST} be given at media level for all media. If=20
> the attribute is present at media level and session level the=20
> media level values {\SHALL} be used.
>=20
> The unit is specified first, followed by the value range. The=20
> units and their values are as defined in Section=20
> \ref{sec:smpte}, \ref{sec:npt} and \ref{sec:clock}. Any open=20
> ended range (start-), i.e. without stop range, is of=20
> unspecified duration and {\SHALL} be considered as=20
> non-seekable content unless this property is overridden.
>=20
>=20
> This attribute is defined in ABNF \cite{rfc2234} as:
>=20
> a-range-def =3D "a" "=3D" "range" ":" ranges-specifier CRLF
>=20
>=20
>=20
> Examples:
>=20
>    a=3Drange:npt=3D0-34.4368
>    a=3Drange:clock=3D19971113T2115-19971113T2203
>    Non seekable stream of unknown duration:
>    a=3Drange:npt=3D0-
>=20
> "
>=20
> Any comments should be given before wednesday the 22nd of October.
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> 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
>=20
>=20
>=20
> _______________________________________________
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>=20

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Date: Wed, 15 Oct 2003 00:52:53 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Subject: Re: [MMUSIC] RTSP: Minutes telecon 14th of October 2003
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Yes, SIP BYE is used to terminate the call, which also results in the 
termination of the media sessions.

-Jonathan R.

Thomas Zeng wrote:

> On item g:
> 
>     - Look into SIP to see how they have resolved the discrepancies
>       between media and signaling channel.
> 
> Could someone fammiliar with SIP help me out here: I need to know what
> method of SIP to look, is it SIP BYE method by any chance?
> 
> (I saw that SIP has a BYE method to terminate the call in the signalling
> channel. )
> 
> thanks,
> thomas
> 
> 
> 
> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Tuesday, October 14, 2003 11:04 AM
> To: IETF MMUSIC WG
> Subject: [MMUSIC] RTSP: Minutes telecon 14th of October 2003
> 
> 
> Hi,
> 
> Here is the preliminary minutes for todays telecon. Please comment.
> 
> RTSP Telecon the 14th of October
> 
> Participants:
> Magnus Westerlund, Colin Perkins, Aravind Narasimhan, and Tom Marshall.
> 
> Agenda:
> 
> 1. Opening of meeting.
> 
> 2. Administrative
>     - Scribe
>     - IETF in Minneapolis
>       * Off-line Meeting?
>       * What to present.
> 
> 3. Since last meeting (1st of October).
> a. http://rtsp.org/bug507882 - RTP timestamp and pause
>      - Aravind to write up text
> 
> b. http://rtsp.org/bug742328 - How to handle PLAY request when at the
> end of a media clip
>      - Text sent to mailing list, please review.
> 
> c. http://rtsp.org/bug742348 - Usage of implicit REDIRECT.
>      - Needs text
> 
> d. http://rtsp.org/bug640272 - Default way of deriving aggregate URL
>     - Text sent to the mailing list
> 
> e. http://rtsp.org/bug771638 - a=range in SDP to indicate live media
>     - Text sent to the mailing list.
> 
> f. http://rtsp.org/bug531189 - How does REDIRECT method work
>     - Needs somebody to write up.
> 
> g. http://rtsp.org/bug810678 - How to indicate end of media stream
>     - Strawman proposal available from Thomas. For discussion.
> 
> h. http://rtsp.org/bug770522 - Request to response time is not limited.
>     - Some discussion on mailing list. Sean to write up text.
> 
> i. http://rtsp.org/bug810681 -     The updated specification version number
>     - Awaiting investigation results regarding changes by Aravind.
> 
> j. http://rtsp.org/bug795295 - TEARDOWN using * and the Ready-nm state
>     - Posted to the mailing list, no comments. Unless comments are
>       received it will be edited in.
> 
> k. http://rtsp.org/bug804219 - How to handle sync in re-SETUP
>     - Text posted to the list. No comments received so far.
> 
> l. http://rtsp.org/bug806404 - Re-SETUP of transport parameters
>     - Awaiting proposal text from Geetha.
> 
> m. http://rtsp.org/bug810666 - Removal of SETUP in play state
>     - Proposed to be removed. Message sent to the list.
> 
> 4. New Issues
> 
> a. http://rtsp.org/bug811815 - Use of c= in SDP to know possible address 
> classes
> 
> b. http://rtsp.org/bug795314 - Caching of OPTIONS request
> 
> 5. Any other business
> 
> 6. Next Telecon: Proposal Wednesday, 3rd of December.
> 
> 7. End of meeting
> 
> Minutes
> -------
> 
> 1. Meeting was opened.
> 
> 2. - Magnus volunteered to write the minutes.
> 
>     - IETF meeting in Minneapolis:
>       * Colin will request a conference room for a out-side of normal
>         agenda meeting. A guess of on the number of participants are
>         around 15.
> 
>       * The authors will perform a status update on the specification in
>         the WG session. The current resolutions and open issues will be
>         presented.
> 
> 3.
> 
> a. Nothing has happened.
> 
> b. Await review from the WG.
> 
> c. Status not changed.
> 
> d. Await review from the WG.
> 
> e. Was not sent yet, but will be available directly after meeting.
> Review comments needed.
> 
> f. The remaining issue is with implicit closing of the session. The
>     previous meeting has agreement on the functionality but some minor
>     issues around use of "connection" needs resolving. Best way of
>     resolving this is probably to present a specification text proposal.
>     Aravind will take this as a second priority issue.
> 
> g. Send comments on the document to list. The discussion resolved that
>     this can be divided into several pieces:
> 
>     - We need to document in the RTSP specification regarding RTP what
>       happens when a source times out a SSRC or receives a RTCP BYE for a
>       source.
> 
>     - Using BYE is not suitable way to indicate the end of the stream.
>       The motivation for this can be seen in Thomas proposal.
> 
>     - Look into SIP to see how they have resolved the discrepancies
>       between media and signaling channel.
> 
> h. Awaits text from Sean.
> 
> i. This is a later decision when we have received a list of changes that
>     are accurate.
> 
> j. Unless comments are received by Monday the 20th of October, this
>     resolution will be edited in. Magnus will post a reply to the initial
>     mail pointing out the deadline.
> 
> k. Will be kept in the text unless comments are review by Wednesday the
>     22th of October.
> 
> l. Awaits text.
> 
> m. Will be edited in unless comments are received by Monday the 20th of
>     October.
> 
> 4.
> a. Question raised is: Are there need for a server to announce through
>     SDP or other pre SETUP mechanism which address classes it supports to
>     the client?
> 
>     There was discussion on this topic and the resolution was the
>     following:
> 
>     It should be assumed that a server being possible to find in DNS with
>     a certain address class will support both signaling and media
>     delivery for that address. Thus a RTSP request over TCP/IPv6 will
>     send media using that source IPv6 address.
> 
> b. The OPTIONS request has some dependency on server state and what
> content that are addressed. Therefore it seems doubtful that caching of
> OPTIONS request are possible. In general it seems that the only
> really cachable RTSP request/response pair is for DESCRIBE.
> 
> 5. No other business.
> 
> 6. Preliminary date is the 3rd of December, which is three weeks after
> the IETF meeting.
> 
> 7. Meeting was closed.
> 
> 
> 

-- 
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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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
CC: Thomas Zeng <zeng@PacketVideo.COM>, IETF MMUSIC WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] RTSP: Minutes telecon 14th of October 2003
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Thanks Jonathan,

However this is more regarding the other way around. How does SIP handle 
if it receives RTCP BYE or times out a SSRC in an RTP session part of 
the currnet media session?

Cheers

Magnus

Jonathan Rosenberg wrote:

> Yes, SIP BYE is used to terminate the call, which also results in the 
> termination of the media sessions.
> 
> -Jonathan R.
> 
> Thomas Zeng wrote:
> 
>> On item g:
>>
>>     - Look into SIP to see how they have resolved the discrepancies
>>       between media and signaling channel.
>>
>> Could someone fammiliar with SIP help me out here: I need to know what
>> method of SIP to look, is it SIP BYE method by any chance?
>>
>> (I saw that SIP has a BYE method to terminate the call in the signalling
>> channel. )
>>
>> thanks,
>> thomas
>>
>>
>>
>> -----Original Message-----
>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
>> Sent: Tuesday, October 14, 2003 11:04 AM
>> To: IETF MMUSIC WG
>> Subject: [MMUSIC] RTSP: Minutes telecon 14th of October 2003
>>
>>
>> Hi,
>>
>> Here is the preliminary minutes for todays telecon. Please comment.
>>
>> RTSP Telecon the 14th of October
>>
>> Participants:
>> Magnus Westerlund, Colin Perkins, Aravind Narasimhan, and Tom Marshall.
>>
>> Agenda:
>>
>> 1. Opening of meeting.
>>
>> 2. Administrative
>>     - Scribe
>>     - IETF in Minneapolis
>>       * Off-line Meeting?
>>       * What to present.
>>
>> 3. Since last meeting (1st of October).
>> a. http://rtsp.org/bug507882 - RTP timestamp and pause
>>      - Aravind to write up text
>>
>> b. http://rtsp.org/bug742328 - How to handle PLAY request when at the
>> end of a media clip
>>      - Text sent to mailing list, please review.
>>
>> c. http://rtsp.org/bug742348 - Usage of implicit REDIRECT.
>>      - Needs text
>>
>> d. http://rtsp.org/bug640272 - Default way of deriving aggregate URL
>>     - Text sent to the mailing list
>>
>> e. http://rtsp.org/bug771638 - a=range in SDP to indicate live media
>>     - Text sent to the mailing list.
>>
>> f. http://rtsp.org/bug531189 - How does REDIRECT method work
>>     - Needs somebody to write up.
>>
>> g. http://rtsp.org/bug810678 - How to indicate end of media stream
>>     - Strawman proposal available from Thomas. For discussion.
>>
>> h. http://rtsp.org/bug770522 - Request to response time is not limited.
>>     - Some discussion on mailing list. Sean to write up text.
>>
>> i. http://rtsp.org/bug810681 -     The updated specification version 
>> number
>>     - Awaiting investigation results regarding changes by Aravind.
>>
>> j. http://rtsp.org/bug795295 - TEARDOWN using * and the Ready-nm state
>>     - Posted to the mailing list, no comments. Unless comments are
>>       received it will be edited in.
>>
>> k. http://rtsp.org/bug804219 - How to handle sync in re-SETUP
>>     - Text posted to the list. No comments received so far.
>>
>> l. http://rtsp.org/bug806404 - Re-SETUP of transport parameters
>>     - Awaiting proposal text from Geetha.
>>
>> m. http://rtsp.org/bug810666 - Removal of SETUP in play state
>>     - Proposed to be removed. Message sent to the list.
>>
>> 4. New Issues
>>
>> a. http://rtsp.org/bug811815 - Use of c= in SDP to know possible 
>> address classes
>>
>> b. http://rtsp.org/bug795314 - Caching of OPTIONS request
>>
>> 5. Any other business
>>
>> 6. Next Telecon: Proposal Wednesday, 3rd of December.
>>
>> 7. End of meeting
>>
>> Minutes
>> -------
>>
>> 1. Meeting was opened.
>>
>> 2. - Magnus volunteered to write the minutes.
>>
>>     - IETF meeting in Minneapolis:
>>       * Colin will request a conference room for a out-side of normal
>>         agenda meeting. A guess of on the number of participants are
>>         around 15.
>>
>>       * The authors will perform a status update on the specification in
>>         the WG session. The current resolutions and open issues will be
>>         presented.
>>
>> 3.
>>
>> a. Nothing has happened.
>>
>> b. Await review from the WG.
>>
>> c. Status not changed.
>>
>> d. Await review from the WG.
>>
>> e. Was not sent yet, but will be available directly after meeting.
>> Review comments needed.
>>
>> f. The remaining issue is with implicit closing of the session. The
>>     previous meeting has agreement on the functionality but some minor
>>     issues around use of "connection" needs resolving. Best way of
>>     resolving this is probably to present a specification text proposal.
>>     Aravind will take this as a second priority issue.
>>
>> g. Send comments on the document to list. The discussion resolved that
>>     this can be divided into several pieces:
>>
>>     - We need to document in the RTSP specification regarding RTP what
>>       happens when a source times out a SSRC or receives a RTCP BYE for a
>>       source.
>>
>>     - Using BYE is not suitable way to indicate the end of the stream.
>>       The motivation for this can be seen in Thomas proposal.
>>
>>     - Look into SIP to see how they have resolved the discrepancies
>>       between media and signaling channel.
>>
>> h. Awaits text from Sean.
>>
>> i. This is a later decision when we have received a list of changes that
>>     are accurate.
>>
>> j. Unless comments are received by Monday the 20th of October, this
>>     resolution will be edited in. Magnus will post a reply to the initial
>>     mail pointing out the deadline.
>>
>> k. Will be kept in the text unless comments are review by Wednesday the
>>     22th of October.
>>
>> l. Awaits text.
>>
>> m. Will be edited in unless comments are received by Monday the 20th of
>>     October.
>>
>> 4.
>> a. Question raised is: Are there need for a server to announce through
>>     SDP or other pre SETUP mechanism which address classes it supports to
>>     the client?
>>
>>     There was discussion on this topic and the resolution was the
>>     following:
>>
>>     It should be assumed that a server being possible to find in DNS with
>>     a certain address class will support both signaling and media
>>     delivery for that address. Thus a RTSP request over TCP/IPv6 will
>>     send media using that source IPv6 address.
>>
>> b. The OPTIONS request has some dependency on server state and what
>> content that are addressed. Therefore it seems doubtful that caching of
>> OPTIONS request are possible. In general it seems that the only
>> really cachable RTSP request/response pair is for DESCRIBE.
>>
>> 5. No other business.
>>
>> 6. Preliminary date is the 3rd of December, which is three weeks after
>> the IETF meeting.
>>
>> 7. Meeting was closed.
>>
>>
>>
> 

-- 

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: Bug 771638 - a=range in SDP to indicate live media
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Hi Jae-Hwan,

I am not certain I understand your concern.

You have cases where the SDP has the following attributes in it:
a=range:npt=0-620
a=range:npt=0-300

In my opinion this is an erroneous SDP. The server creating this SDP has 
made a mistake. The only concern I have with limiting the SDP attribute 
to only contain a single range is that the following case is not 
possible to express:

Range: npt=0-300, npt=450-600

That is possible to express in the RTSP range header. And is maybe even 
more useful when used with "clock" to inform what parts that are 
available of yesterdays recorded TV programs.

If the above was not what you meant, please clarify.

Cheers

Magnus


Jae-Hwan Kim wrote:

> Hi,
> 
> Just a simple question regarding range header although it is far from an
> live indicating.
> I saw an example, 2 range headers at session-level SDP because the
> original content contains a=range in its hinted SDP area: the first one
> is 'a=range:npt=0-620' server made and the other is
> 'a=range:npt=0-300'(or npt=0-) originally from the server; in case of
> two different range attributes exist. Almost all hinted server just
> seems to get and forward SDP and RTP data.
> 
> How can we, as a player, understand this SDP? 
> Just I think the possible and right behavior is as followings:
> 1) If there are two or more a=range header at a session-level or
> media-level, the maximum one shall be used.
> 2) But if there is npt=0-, it means also this content is not seekable.
> 
> I would appreciate it for your advice.
> 
> Cheers,
> Jae-Hwan
> 
> 
> 
>>-----Original Message-----
>>From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com] 
>>Sent: Wednesday, October 15, 2003 2:37 AM
>>To: IETF MMUSIC WG
>>Subject: [MMUSIC] RTSP: Bug 771638 - a=range in SDP to 
>>indicate live media
>>
>>
>>Hi,
>>
>>The below text is proposed for resolving the the following bug:
>>
>>http://rtsp.org/bug771638 - a=range in SDP to indicate live media
>>
>> From Annex C:
>>
>>"
>>Range of Presentation
>>
>>The ``a=range'' attribute defines the total time range of the 
>>stored session or an individual media. That Live sessions are 
>>non-seekable can also be indicated, while the length of live 
>>sessions can be deduced from the ``t'' and ``r'' parameters.
>>
>>The attribute is both a session and a media level attribute. 
>>For presentations that contains media streams of the same 
>>durations, the range attribute {\SHOULD} only be used at 
>>session-level. In case of different length the range 
>>attribute {\MUST} be given at media level for all media. If 
>>the attribute is present at media level and session level the 
>>media level values {\SHALL} be used.
>>
>>The unit is specified first, followed by the value range. The 
>>units and their values are as defined in Section 
>>\ref{sec:smpte}, \ref{sec:npt} and \ref{sec:clock}. Any open 
>>ended range (start-), i.e. without stop range, is of 
>>unspecified duration and {\SHALL} be considered as 
>>non-seekable content unless this property is overridden.
>>
>>
>>This attribute is defined in ABNF \cite{rfc2234} as:
>>
>>a-range-def = "a" "=" "range" ":" ranges-specifier CRLF
>>
>>
>>
>>Examples:
>>
>>   a=range:npt=0-34.4368
>>   a=range:clock=19971113T2115-19971113T2203
>>   Non seekable stream of unknown duration:
>>   a=range:npt=0-
>>
>>"
>>
>>Any comments should be given before wednesday the 22nd of October.
>>
>>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
>>
>>
>>
>>_______________________________________________
>>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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Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com


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Does our current effort to summarize multicast service discovery =
requirements in MMUSIC meet your needs?

Internet Media Guide requirements: =
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-img-req-00.txt

Clearly, exchanging/announcing session/etc parameters effects joining =
(SSM and ASM), involves security risks and enable multicast content =
services - like streaming media and file delivery. So any comments, =
criticisms, ideas and beers are welcome. (Speaking of service =
announcements, does Minneapolis offer the same excellent range of Ales =
as San Francisco?)

Cheers,
Rod.

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Thread-Topic: Service/Session Announcement for Multicast Apps
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Does anyone have a problem with a new and better announcement protocol =
than SAP?

We need a few things which SAP doesn't handle too well, which is one of =
the things covered by the Internet Media Guide effort in MMUSIC: =
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-img-req-00.txt

I guess anyone involved in applications/services over multicast could be =
interested, especially where point-to-point isn't desirable.

I wanted to get an idea of any strong opinions before I stick out head =
and propose a real protocol task for MMUSIC next month.

Cheers, Rod.

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From exim@www1.ietf.org  Wed Oct 15 08:47:33 2003
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From: "Pete Cordell" <pete@tech-know-ware.com>
To: "Peterson, Jon" <jon.peterson@neustar.biz>, "Mmusic" <mmusic@ietf.org>
Cc: <midcom@ietf.org>, "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
References: <A9DECB0B8A01A54DBECC03B25D29513C0A7C32@stntexch03.va.neustar.com>
Subject: Re: [MMUSIC] Status of TURN
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Dear Jon,

I'm very surprised to see that TURN is being encouraged to be put forward as
an individual submission.  We believe that TURN has major security flaws, is
incomplete, and there are better ways of achieving Symmetric NAT traversal.
Suggesting it should become an RFC without debate and full consideration
would be wrong.

If you remember, TURN was just one of the candidates put forward as a
pre-Midcom solution that at that time was put aside in favour of developing
a consensus protocol.   Despite our commitment, consensus wasn't reached,
mainly due to the amount of effort that was being put into SIP (RFC3261) by
other important stake holders.

We feel that to arbitrarily promote TURN at this time is short cutting the
consensus making process.  Surely, if there is sufficient interest in a
standard of this nature, the correct way forward is to assign it as a work
item in a suitable group and re-start the consensus process. If so, we
remain willing to help with time, resource and considerable technical
know-how. Several significant developments have meant the world of Symmetric
NAT traversal has advanced considerably since the pre-Midcom days.

Regards,

Pete.

=============================================
Pete Cordell
Ridgeway
=============================================

----- Original Message -----
From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Joel Tran'" <joel_tran@hotmail.com>; "Jonathan Rosenberg"
<jdrosen@dynamicsoft.com>; "Mmusic" <mmusic@ietf.org>
Sent: 09 October 2003 17:08
Subject: RE: [MMUSIC] Status of TURN


>
> The Transport Area Directors understand that this document will proceed as
> an individual submission, outside the context of any of the working groups
> in the Transport Area. It is clearly of interest to a number of these
> working groups.
>
> I gather that at least one more revision needs to be undertaken before
this
> document will be considered for publication as an RFC, but we're very
> interested to see it go forward.
>
> Jon Peterson
> NeuStar, Inc.
> (co-AD for Transport)
>
> > -----Original Message-----
> > From: Joel Tran [mailto:joel_tran@hotmail.com]
> > Sent: Tuesday, October 07, 2003 11:08 AM
> > To: Jonathan Rosenberg; Mmusic
> > Subject: [MMUSIC] Status of TURN
> >
> >
> > Hi Jonathan,
> >
> > I would like to ask you two little questions concerning TURN.
> >
> > First, I would like to know what is the exact status of TURN?
> > Will it be
> > adopted as a Workgroup item of MMUSIC since TURN is refeered
> > in various
> > documents?
> >
> > Second, I would like to ask you if there is a reason why the
> > lastest version
> > of the draft is not on the IETF web and only available throught
> > http://www.jdrosen.net/papers/draft-rosenberg-midcom-turn-02.txt?
> >
> > Thanks,
> > ...J
> >
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www1.ietf.org/mailman/listinfo/mmusic
> >
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic


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Subject: RE: [MMUSIC] specifying the multicast interface
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> -----Original Message-----
> From: Thomas Zeng [mailto:zeng@PacketVideo.COM]=20
> Sent: Monday, October 13, 2003 1:50 PM
>=20
> Yes, you can provide the IP of the selected IF via the c=3D=20
> line in SDP,=20

But doesn't that specify the multicast address?  The interface address
is different -- it's the IP of the NIC used to multicast the session.



> but the final say is upto RTSP negotiations,=20
> ie., you can use the Transport header in the SETUP=20
> request/response to over-ride SDP. Transport header has the=20
> following general parameters, quoting 2326:
>=20
>=20
>    destination:
>           The address to which a stream will be sent. The client may
>           specify the multicast address with the destination=20
> parameter.

But isn't this the multicast address?  This is not the same as the
interface address.


>=20
>    source:
>           If the source address for the stream is different=20
> than can be
>           derived from the RTSP endpoint address (the server=20
> in playback
>           or the client in recording), the source MAY be specified.

Not sure whether this qualifies as the interface address.


> Hope this is the answer you are looking for.

I don't think so, but thanks anyway.

-Matt


_______________________________________________
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From: Thomas Zeng <zeng@PacketVideo.COM>
To: "'Matthew Heaney'" <mheaney@on2.com>, mmusic@ietf.org
Subject: RE: [MMUSIC] specifying the multicast interface
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Why does one need to publish "the IP of the NIC used to multicast the
session" ?

-----Original Message-----
From: Matthew Heaney [mailto:mheaney@on2.com]
Sent: Wednesday, October 15, 2003 8:27 AM
To: Thomas Zeng; mmusic@ietf.org
Subject: RE: [MMUSIC] specifying the multicast interface




> -----Original Message-----
> From: Thomas Zeng [mailto:zeng@PacketVideo.COM] 
> Sent: Monday, October 13, 2003 1:50 PM
> 
> Yes, you can provide the IP of the selected IF via the c= 
> line in SDP, 

But doesn't that specify the multicast address?  The interface address
is different -- it's the IP of the NIC used to multicast the session.



> but the final say is upto RTSP negotiations, 
> ie., you can use the Transport header in the SETUP 
> request/response to over-ride SDP. Transport header has the 
> following general parameters, quoting 2326:
> 
> 
>    destination:
>           The address to which a stream will be sent. The client may
>           specify the multicast address with the destination 
> parameter.

But isn't this the multicast address?  This is not the same as the
interface address.


> 
>    source:
>           If the source address for the stream is different 
> than can be
>           derived from the RTSP endpoint address (the server 
> in playback
>           or the client in recording), the source MAY be specified.

Not sure whether this qualifies as the interface address.


> Hope this is the answer you are looking for.

I don't think so, but thanks anyway.

-Matt

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



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Subject: RE: [MMUSIC] specifying the multicast interface
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Because a customer asked us to.

A multicast session is started on a TrueCast server using a web-based
admin interface.  The admin user specifies the multicast address and
port he wants, the name of the presentation, etc. =20

Currently, our admin tool doesn't allow you to specify the IP of the
interface card, even though explicitly specifying the interface over
which to multicast the session is allowed per the BSD socket API.  So we
have to fix it.

For now I'll just use a GET_PARAMETER to get the interface list, and
then augment the SETUP request with an "X-multicast-interface" header.



> -----Original Message-----
> From: Thomas Zeng [mailto:zeng@PacketVideo.COM]=20
> Sent: Wednesday, October 15, 2003 8:55 PM
> To: Matthew Heaney; mmusic@ietf.org
> Subject: RE: [MMUSIC] specifying the multicast interface
>=20
>=20
> Why does one need to publish "the IP of the NIC used to=20
> multicast the session" ?
>=20
> -----Original Message-----
> From: Matthew Heaney [mailto:mheaney@on2.com]
> Sent: Wednesday, October 15, 2003 8:27 AM
> To: Thomas Zeng; mmusic@ietf.org
> Subject: RE: [MMUSIC] specifying the multicast interface
>=20
>=20
>=20
>=20
> > -----Original Message-----
> > From: Thomas Zeng [mailto:zeng@PacketVideo.COM]
> > Sent: Monday, October 13, 2003 1:50 PM
> >=20
> > Yes, you can provide the IP of the selected IF via the c=3D
> > line in SDP,=20
>=20
> But doesn't that specify the multicast address?  The=20
> interface address is different -- it's the IP of the NIC used=20
> to multicast the session.
>=20
>=20
>=20
> > but the final say is upto RTSP negotiations,
> > ie., you can use the Transport header in the SETUP=20
> > request/response to over-ride SDP. Transport header has the=20
> > following general parameters, quoting 2326:
> >=20
> >=20
> >    destination:
> >           The address to which a stream will be sent. The client may
> >           specify the multicast address with the destination
> > parameter.
>=20
> But isn't this the multicast address?  This is not the same=20
> as the interface address.
>=20
>=20
> >=20
> >    source:
> >           If the source address for the stream is different
> > than can be
> >           derived from the RTSP endpoint address (the server=20
> > in playback
> >           or the client in recording), the source MAY be specified.
>=20
> Not sure whether this qualifies as the interface address.
>=20
>=20
> > Hope this is the answer you are looking for.
>=20
> I don't think so, but thanks anyway.
>=20
> -Matt
>=20

_______________________________________________
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        midcom@ietf.org
Subject: Re: [MMUSIC] Status of TURN
References: <A9DECB0B8A01A54DBECC03B25D29513C0A7C32@stntexch03.va.neustar.com> <003301c3931a$781e72c0$0200000a@tkw>
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Just because its an individual submission does not mean that comments 
and criticisms are not welcome. Please feel free to send me a note.

Thanks,
Jonathan R.

Pete Cordell wrote:

> Dear Jon,
> 
> I'm very surprised to see that TURN is being encouraged to be put forward as
> an individual submission.  We believe that TURN has major security flaws, is
> incomplete, and there are better ways of achieving Symmetric NAT traversal.
> Suggesting it should become an RFC without debate and full consideration
> would be wrong.
> 
> If you remember, TURN was just one of the candidates put forward as a
> pre-Midcom solution that at that time was put aside in favour of developing
> a consensus protocol.   Despite our commitment, consensus wasn't reached,
> mainly due to the amount of effort that was being put into SIP (RFC3261) by
> other important stake holders.
> 
> We feel that to arbitrarily promote TURN at this time is short cutting the
> consensus making process.  Surely, if there is sufficient interest in a
> standard of this nature, the correct way forward is to assign it as a work
> item in a suitable group and re-start the consensus process. If so, we
> remain willing to help with time, resource and considerable technical
> know-how. Several significant developments have meant the world of Symmetric
> NAT traversal has advanced considerably since the pre-Midcom days.
> 
> Regards,
> 
> Pete.
> 
> =============================================
> Pete Cordell
> Ridgeway
> =============================================
> 
> ----- Original Message -----
> From: "Peterson, Jon" <jon.peterson@neustar.biz>
> To: "'Joel Tran'" <joel_tran@hotmail.com>; "Jonathan Rosenberg"
> <jdrosen@dynamicsoft.com>; "Mmusic" <mmusic@ietf.org>
> Sent: 09 October 2003 17:08
> Subject: RE: [MMUSIC] Status of TURN
> 
> 
> 
>>The Transport Area Directors understand that this document will proceed as
>>an individual submission, outside the context of any of the working groups
>>in the Transport Area. It is clearly of interest to a number of these
>>working groups.
>>
>>I gather that at least one more revision needs to be undertaken before
> 
> this
> 
>>document will be considered for publication as an RFC, but we're very
>>interested to see it go forward.
>>
>>Jon Peterson
>>NeuStar, Inc.
>>(co-AD for Transport)
>>
>>
>>>-----Original Message-----
>>>From: Joel Tran [mailto:joel_tran@hotmail.com]
>>>Sent: Tuesday, October 07, 2003 11:08 AM
>>>To: Jonathan Rosenberg; Mmusic
>>>Subject: [MMUSIC] Status of TURN
>>>
>>>
>>>Hi Jonathan,
>>>
>>>I would like to ask you two little questions concerning TURN.
>>>
>>>First, I would like to know what is the exact status of TURN?
>>>Will it be
>>>adopted as a Workgroup item of MMUSIC since TURN is refeered
>>>in various
>>>documents?
>>>
>>>Second, I would like to ask you if there is a reason why the
>>>lastest version
>>>of the draft is not on the IETF web and only available throught
>>>http://www.jdrosen.net/papers/draft-rosenberg-midcom-turn-02.txt?
>>>
>>>Thanks,
>>>...J
>>>
>>>_______________________________________________
>>>mmusic mailing list
>>>mmusic@ietf.org
>>>https://www1.ietf.org/mailman/listinfo/mmusic
>>>
>>
>>_______________________________________________
>>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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From: Thomas Zeng <zeng@PacketVideo.COM>
To: "'Matthew Heaney'" <mheaney@on2.com>, Thomas Zeng <zeng@PacketVideo.COM>,
        mmusic@ietf.org
Subject: RE: [MMUSIC] specifying the multicast interface
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Interesting...

So if we add a new parameter (since we are revising RFC2326, it is not
impossible to as for an optional parameter, say, "multicast-interface") to
the Transport header in SETUP request/response, you no longer has to use the
"x-multicast-interface" parameter which is not interoperable?

But you may want to convince people that others might be interested in this
new parameter ...

Just a thought.
thomas

> -----Original Message-----
> From: Matthew Heaney [mailto:mheaney@on2.com]
> Sent: Thursday, October 16, 2003 4:13 PM
> To: Thomas Zeng; mmusic@ietf.org
> Subject: RE: [MMUSIC] specifying the multicast interface
> 
> 
> Because a customer asked us to.
> 
> A multicast session is started on a TrueCast server using a web-based
> admin interface.  The admin user specifies the multicast address and
> port he wants, the name of the presentation, etc.  
> 
> Currently, our admin tool doesn't allow you to specify the IP of the
> interface card, even though explicitly specifying the interface over
> which to multicast the session is allowed per the BSD socket 
> API.  So we
> have to fix it.
> 
> For now I'll just use a GET_PARAMETER to get the interface list, and
> then augment the SETUP request with an "X-multicast-interface" header.
> 
> 
> 
> > -----Original Message-----
> > From: Thomas Zeng [mailto:zeng@PacketVideo.COM] 
> > Sent: Wednesday, October 15, 2003 8:55 PM
> > To: Matthew Heaney; mmusic@ietf.org
> > Subject: RE: [MMUSIC] specifying the multicast interface
> > 
> > 
> > Why does one need to publish "the IP of the NIC used to 
> > multicast the session" ?
> > 
> > -----Original Message-----
> > From: Matthew Heaney [mailto:mheaney@on2.com]
> > Sent: Wednesday, October 15, 2003 8:27 AM
> > To: Thomas Zeng; mmusic@ietf.org
> > Subject: RE: [MMUSIC] specifying the multicast interface
> > 
> > 
> > 
> > 
> > > -----Original Message-----
> > > From: Thomas Zeng [mailto:zeng@PacketVideo.COM]
> > > Sent: Monday, October 13, 2003 1:50 PM
> > > 
> > > Yes, you can provide the IP of the selected IF via the c=
> > > line in SDP, 
> > 
> > But doesn't that specify the multicast address?  The 
> > interface address is different -- it's the IP of the NIC used 
> > to multicast the session.
> > 
> > 
> > 
> > > but the final say is upto RTSP negotiations,
> > > ie., you can use the Transport header in the SETUP 
> > > request/response to over-ride SDP. Transport header has the 
> > > following general parameters, quoting 2326:
> > > 
> > > 
> > >    destination:
> > >           The address to which a stream will be sent. The 
> client may
> > >           specify the multicast address with the destination
> > > parameter.
> > 
> > But isn't this the multicast address?  This is not the same 
> > as the interface address.
> > 
> > 
> > > 
> > >    source:
> > >           If the source address for the stream is different
> > > than can be
> > >           derived from the RTSP endpoint address (the server 
> > > in playback
> > >           or the client in recording), the source MAY be 
> specified.
> > 
> > Not sure whether this qualifies as the interface address.
> > 
> > 
> > > Hope this is the answer you are looking for.
> > 
> > I don't think so, but thanks anyway.
> > 
> > -Matt
> > 
> 

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



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Hi Matthew,

I have some problem of understanding why you need to specify the 
outgoing interface to use. Is this multihomes servers that you have an 
interest of ensuring that the transmissions are made using the right 
sender address? A little more explanation on the needs and advantages to 
specify the sending interface would be appreciated. Especially if we 
should consider adding anything to the RTSP standard.

Then I would also recommend you to skip the "X-" from your parameter 
name, as if it would be standardized in the future you don't have to 
change the implementations, to also support the new name. The only 
consideration is that if someone in the future registers this name is 
that the proprietary use are responsible for updating there feature or 
giving it a new name.

cheers

Magnus

Matthew Heaney wrote:

> Because a customer asked us to.
> 
> A multicast session is started on a TrueCast server using a web-based
> admin interface.  The admin user specifies the multicast address and
> port he wants, the name of the presentation, etc.  
> 
> Currently, our admin tool doesn't allow you to specify the IP of the
> interface card, even though explicitly specifying the interface over
> which to multicast the session is allowed per the BSD socket API.  So we
> have to fix it.
> 
> For now I'll just use a GET_PARAMETER to get the interface list, and
> then augment the SETUP request with an "X-multicast-interface" header.
> 
> 
> 
> 
>>-----Original Message-----
>>From: Thomas Zeng [mailto:zeng@PacketVideo.COM] 
>>Sent: Wednesday, October 15, 2003 8:55 PM
>>To: Matthew Heaney; mmusic@ietf.org
>>Subject: RE: [MMUSIC] specifying the multicast interface
>>
>>
>>Why does one need to publish "the IP of the NIC used to 
>>multicast the session" ?
>>
>>-----Original Message-----
>>From: Matthew Heaney [mailto:mheaney@on2.com]
>>Sent: Wednesday, October 15, 2003 8:27 AM
>>To: Thomas Zeng; mmusic@ietf.org
>>Subject: RE: [MMUSIC] specifying the multicast interface
>>
>>
>>
>>
>>
>>>-----Original Message-----
>>>From: Thomas Zeng [mailto:zeng@PacketVideo.COM]
>>>Sent: Monday, October 13, 2003 1:50 PM
>>>
>>>Yes, you can provide the IP of the selected IF via the c=
>>>line in SDP, 
>>
>>But doesn't that specify the multicast address?  The 
>>interface address is different -- it's the IP of the NIC used 
>>to multicast the session.
>>
>>
>>
>>
>>>but the final say is upto RTSP negotiations,
>>>ie., you can use the Transport header in the SETUP 
>>>request/response to over-ride SDP. Transport header has the 
>>>following general parameters, quoting 2326:
>>>
>>>
>>>   destination:
>>>          The address to which a stream will be sent. The client may
>>>          specify the multicast address with the destination
>>>parameter.
>>
>>But isn't this the multicast address?  This is not the same 
>>as the interface address.
>>
>>
>>
>>>   source:
>>>          If the source address for the stream is different
>>>than can be
>>>          derived from the RTSP endpoint address (the server 
>>>in playback
>>>          or the client in recording), the source MAY be specified.
>>
>>Not sure whether this qualifies as the interface address.
>>
>>
>>
>>>Hope this is the answer you are looking for.
>>
>>I don't think so, but thanks anyway.
>>
>>-Matt
>>
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 

-- 

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



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> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]=20
> Sent: Friday, October 17, 2003 10:13 AM
>=20
> I have some problem of understanding why you need to specify the=20
> outgoing interface to use. Is this multihomes servers that=20
> you have an=20
> interest of ensuring that the transmissions are made using the right=20
> sender address? A little more explanation on the needs and=20
> advantages to=20
> specify the sending interface would be appreciated. Especially if we=20
> should consider adding anything to the RTSP standard.

I don't know the exact reasons why our customer asked for the feature,
but we do have machines here with two interface cards.  In our case each
card is on a completely different network. =20

Suppose a client on one network is interested in joining the multicast
session.  If the server does not specify the local interface from which
to send, then do we have any guarantee that the datagrams will reach the
client?  (I still have to run this test.) =20

Suppose the "default" interface (the one chosen by the kernel if we
don't explicitly specify) is the one attached to the other network.
Does this mean that the client (who isn't on the network to which the
"default" interface is attached) doesn't ever see any of those packets?

It just seems odd that RTSP allows you to specify the TTL
(IP_MULTICAST_TTL), but provides no way to specify the local interface
for sending (IP_MULTICAST_IF) or whether local loopback occurs
(IP_MULTICAST_LOOP).


> Then I would also recommend you to skip the "X-" from your parameter=20
> name, as if it would be standardized in the future you don't have to=20
> change the implementations, to also support the new name. The only=20
> consideration is that if someone in the future registers this name is=20
> that the proprietary use are responsible for updating there=20
> feature or=20
> giving it a new name.

Server mods are easy enough for me.

I just realized, though, that it might be nice to allow the admin user
to specify either the IP of the interface or its name.  Maybe I need to
two headers?  Here are some ideas:

interface
local-interface
multicast-interface
interface-address
interface-name
multicast-interface-address
multicast-interface-name
etc

-Matt


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From: Tom Marshall <tmarshall@real.com>
To: Matthew Heaney <mheaney@on2.com>
Cc: Magnus Westerlund <magnus.westerlund@ericsson.com>,
        Thomas Zeng <zeng@PacketVideo.COM>, mmusic@ietf.org
Subject: Re: [MMUSIC] specifying the multicast interface
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> Suppose the "default" interface (the one chosen by the kernel if we
> don't explicitly specify) is the one attached to the other network.
> Does this mean that the client (who isn't on the network to which the
> "default" interface is attached) doesn't ever see any of those packets?
>=20
> It just seems odd that RTSP allows you to specify the TTL
> (IP_MULTICAST_TTL), but provides no way to specify the local interface
> for sending (IP_MULTICAST_IF) or whether local loopback occurs
> (IP_MULTICAST_LOOP).

I'm not really familiar with your application, but our server is written to
always use the same interface for the transport and control connections.=20
This should solve the problem of the "default" interface.  Do you or your
customer have other reasons for wanting to specify the interface?

--=20
Beware of Programmers who carry screwdrivers.
        -- Leonard Brandwein

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To: "Tom Marshall" <tmarshall@real.com>
Cc: "Magnus Westerlund" <magnus.westerlund@ericsson.com>,
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> -----Original Message-----
> From: Tom Marshall [mailto:tmarshall@real.com]=20
> Sent: Friday, October 17, 2003 1:01 PM
>=20
> I'm not really familiar with your application, but our server=20
> is written to always use the same interface for the transport=20
> and control connections.=20
> This should solve the problem of the "default" interface. =20

The server sends RTP packets out over one interface (the "default"
interface).  The client is on a network attached to another interface.
If the client doesn't get any RTP packets, then I still have a problem.


> Do=20
> you or your customer have other reasons for wanting to=20
> specify the interface?

I assume my customer has his reasons for asking, but he didn't specify
why.  I can try to get more info.


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From: Tom Marshall <tmarshall@real.com>
To: Matthew Heaney <mheaney@on2.com>
Cc: Magnus Westerlund <magnus.westerlund@ericsson.com>,
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On Fri, Oct 17, 2003 at 04:08:03PM -0400, Matthew Heaney wrote:
>=20
>=20
> > -----Original Message-----
> > From: Tom Marshall [mailto:tmarshall@real.com]=20
> > Sent: Friday, October 17, 2003 1:01 PM
> >=20
> > I'm not really familiar with your application, but our server=20
> > is written to always use the same interface for the transport=20
> > and control connections.=20
> > This should solve the problem of the "default" interface. =20
>=20
> The server sends RTP packets out over one interface (the "default"
> interface).  The client is on a network attached to another interface.
> If the client doesn't get any RTP packets, then I still have a problem.

Yes, and the implication was that if the server sent RTP packets out on the
same interface that has the control connection instead of the "default"
interface, it may solve the problem.  I"m interested to see if it does.

--=20
Research is what I'm doing when I don't know what I'm doing.
        -- Wernher von Braun

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From: Deepak Kotian <dkotian3@vsnl.net>
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Reply-to: Deepak Kotian <dkotian3@vsnl.net>
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Subject: [MMUSIC] The initial audio lags video by around 17-20 seconds and after that it
 is OK on RealOne via RTSP/RTP
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Hi,

I have a problem,
The initial audio lags video by around 17-20 seconds and after that it is OK
on RealOne via RTSP/RTP.
Basically the audio comes after 17-20 seconds against video.
i.e. the initial audio packets are not played at all. The audio packets
are sent at all time.
We have implemented RFC 2250 for MPEG1 Video and Audio
and RTSP minimal server.

Any suggestion on this one please.

Thanks and Regards
Deepak



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From: Deepak Kotian <dkotian3@vsnl.net>
To: IETF MMUSIC WG <mmusic@ietf.org>
Reply-to: Deepak Kotian <dkotian3@vsnl.net>
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Subject: [MMUSIC] Re: The initial audio lags video by around 17-20 seconds and after
 that it is OK on RealOne via RTSP/RTP
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Forgot to add, The player is realOne.

Thanks and Regards
Deepak

----- Original Message -----
From: "Deepak Kotian" <dkotian3@vsnl.net>
To: "IETF MMUSIC WG" <mmusic@ietf.org>
Sent: Sunday, October 19, 2003 1:22 AM
Subject: The initial audio lags video by around 17-20 seconds and after that
it is OK on RealOne via RTSP/RTP


> Hi,
>
> I have a problem,
> The initial audio lags video by around 17-20 seconds and after that it is
OK
> on RealOne via RTSP/RTP.
> Basically the audio comes after 17-20 seconds against video.
> i.e. the initial audio packets are not played at all. The audio packets
> are sent at all time.
> We have implemented RFC 2250 for MPEG1 Video and Audio
> and RTSP minimal server.
>
> Any suggestion on this one please.
>
> Thanks and Regards
> Deepak
>



_______________________________________________
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mmusic@ietf.org
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From: Thomas Zeng <zeng@PacketVideo.COM>
To: "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>,
        IETF MMUSIC WG
	 <mmusic@ietf.org>
Subject: RE: [MMUSIC] RTSP: Bug 649272 - Default way of deriving aggregate
	 URL
Date: Sun, 19 Oct 2003 22:37:46 -0700
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Magnus:

It might help to show an example where the aggregate control URI is not in
SDP, yet
the client conjures up one which the server then accepts.

Thanks,
- thomas

> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Tuesday, October 14, 2003 11:06 AM
> To: IETF MMUSIC WG
> Subject: Re: [MMUSIC] RTSP: Bug 649272 - Default way of 
> deriving aggregate URL
> 
> 
> Hi,
> 
> Please provide comments by Wednesday the 22th of October, or 
> this text 
> will be part of the next draft version.
> 
> Cheers
> 
> Magnus
> 
> Magnus Westerlund wrote:
> 
> > Hi,
> > 
> > This is the proposed text in regards bug:
> > 
> > http://rtsp.org/bug640272 - Default way of deriving aggregate URL
> > 
> > The text is from the SETUP method description:
> > 
> > "The server {\MUST} generate a session identifier in response to a
> > successful {\SETUP} request, unless a {\SETUP} request to a server
> > includes a session identifier, in which case the server {\MUST}
> > bundle this setup request into the existing session (aggregated
> > session) or return error 459 (Aggregate Operation Not Allowed)
> > (see Section ~\ref{sec:error459}). An Aggregate control URI
> > {\MUST} be used to control an aggregated session. This URI {\MUST}
> > be different from the stream control URIs of the individual media
> > streams included in the aggregate. The Aggregate control URI
> > {\SHOULD} be specified by the session description since there is
> > no general rule for deriving it from the various stream control
> > URIs in the session.
> > 
> > If an Aggregate control URI is not specified in the session
> > description, it is probably a indication that non-aggregated
> > control should be used. However a client {\MAY} try to SETUP the
> > session in aggregated control. If the server refuse to aggregate
> > the specified media, the server {\SHALL} use the 459 error code.
> > If the server allows the aggregation, then the client {\MUST}
> > create an URI for aggregate control of the session. This URI
> > {\MUST} contain the servers host address and {\MUST} contain the
> > port, if applicable (e.g. not default port). Once an URI is used
> > to refer to an aggregation for a given session, that URI {\MUST}
> > be used to refer to that aggregation for the duration of the
> > session."
> > 
> > 
> > Please comment
> > 
> > 
> > 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
> > 
> > 
> > 
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www1.ietf.org/mailman/listinfo/mmusic
> > 
> 
> -- 
> 
> 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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From: "Pete Cordell" <pete@tech-know-ware.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
Cc: "Peterson, Jon" <jon.peterson@neustar.biz>, "Mmusic" <mmusic@ietf.org>,
        <midcom@ietf.org>
References: <A9DECB0B8A01A54DBECC03B25D29513C0A7C32@stntexch03.va.neustar.com> <003301c3931a$781e72c0$0200000a@tkw> <3F8F42F2.6030402@dynamicsoft.com>
Subject: Re: [midcom] Re: [MMUSIC] Status of TURN
Date: Mon, 20 Oct 2003 15:01:05 +0100
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Hi Jonathan,

That's generous of you, but I basically feel that there is either a need for
this work, in which case it should be done as part of the normal open IETF
process, or there isn't a need, in which case it should go no further.

As I've said elsewhere, I think the reason the effort failed last time was
because people were working on the SIP RFC.  I know you're superhuman (I am
impressed by your output), but I think even you were saturated at that time.
And I actually agree with your prioritisation at the time (there's no point
in having a traversal solution if there is nothing that needs traversing!)

However, to put it bluntly, I would feel aggrieved if those circumstances
were later used to justify fast tracking a proposal that did not represent
consensus of the interested parties involved.

Pete.

=============================================
Pete Cordell
+44 1473 635863
=============================================

----- Original Message -----
From: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
To: "Pete Cordell" <pete@tech-know-ware.com>
Cc: "Peterson, Jon" <jon.peterson@neustar.biz>; "Mmusic" <mmusic@ietf.org>;
<midcom@ietf.org>
Sent: 17 October 2003 02:16
Subject: [midcom] Re: [MMUSIC] Status of TURN


> Just because its an individual submission does not mean that comments
> and criticisms are not welcome. Please feel free to send me a note.
>
> Thanks,
> Jonathan R.
>
> Pete Cordell wrote:
>
> > Dear Jon,
> >
> > I'm very surprised to see that TURN is being encouraged to be put
forward as
> > an individual submission.  We believe that TURN has major security
flaws, is
> > incomplete, and there are better ways of achieving Symmetric NAT
traversal.
> > Suggesting it should become an RFC without debate and full consideration
> > would be wrong.
> >
> > If you remember, TURN was just one of the candidates put forward as a
> > pre-Midcom solution that at that time was put aside in favour of
developing
> > a consensus protocol.   Despite our commitment, consensus wasn't
reached,
> > mainly due to the amount of effort that was being put into SIP (RFC3261)
by
> > other important stake holders.
> >
> > We feel that to arbitrarily promote TURN at this time is short cutting
the
> > consensus making process.  Surely, if there is sufficient interest in a
> > standard of this nature, the correct way forward is to assign it as a
work
> > item in a suitable group and re-start the consensus process. If so, we
> > remain willing to help with time, resource and considerable technical
> > know-how. Several significant developments have meant the world of
Symmetric
> > NAT traversal has advanced considerably since the pre-Midcom days.
> >
> > Regards,
> >
> > Pete.
> >
> > =============================================
> > Pete Cordell
> > Ridgeway
> > =============================================
> >
> > ----- Original Message -----
> > From: "Peterson, Jon" <jon.peterson@neustar.biz>
> > To: "'Joel Tran'" <joel_tran@hotmail.com>; "Jonathan Rosenberg"
> > <jdrosen@dynamicsoft.com>; "Mmusic" <mmusic@ietf.org>
> > Sent: 09 October 2003 17:08
> > Subject: RE: [MMUSIC] Status of TURN
> >
> >
> >
> >>The Transport Area Directors understand that this document will proceed
as
> >>an individual submission, outside the context of any of the working
groups
> >>in the Transport Area. It is clearly of interest to a number of these
> >>working groups.
> >>
> >>I gather that at least one more revision needs to be undertaken before
> >
> > this
> >
> >>document will be considered for publication as an RFC, but we're very
> >>interested to see it go forward.
> >>
> >>Jon Peterson
> >>NeuStar, Inc.
> >>(co-AD for Transport)
> >>
> >>
> >>>-----Original Message-----
> >>>From: Joel Tran [mailto:joel_tran@hotmail.com]
> >>>Sent: Tuesday, October 07, 2003 11:08 AM
> >>>To: Jonathan Rosenberg; Mmusic
> >>>Subject: [MMUSIC] Status of TURN
> >>>
> >>>
> >>>Hi Jonathan,
> >>>
> >>>I would like to ask you two little questions concerning TURN.
> >>>
> >>>First, I would like to know what is the exact status of TURN?
> >>>Will it be
> >>>adopted as a Workgroup item of MMUSIC since TURN is refeered
> >>>in various
> >>>documents?
> >>>
> >>>Second, I would like to ask you if there is a reason why the
> >>>lastest version
> >>>of the draft is not on the IETF web and only available throught
> >>>http://www.jdrosen.net/papers/draft-rosenberg-midcom-turn-02.txt?
> >>>
> >>>Thanks,
> >>>...J
> >>>
> >>>_______________________________________________
> >>>mmusic mailing list
> >>>mmusic@ietf.org
> >>>https://www1.ietf.org/mailman/listinfo/mmusic
> >>>
> >>
> >>_______________________________________________
> >>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
>
>
> _______________________________________________
> midcom mailing list
> midcom@ietf.org
> https://www1.ietf.org/mailman/listinfo/midcom


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Message-ID: <003a01c39721$709a9640$0200000a@tkw>
From: "Pete Cordell" <pete@tech-know-ware.com>
To: "Melinda Shore" <mshore@cisco.com>
Cc: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>,
        "Peterson, Jon" <jon.peterson@neustar.biz>, "Mmusic" <mmusic@ietf.org>,
        <midcom@ietf.org>
References: <86BB4302-0309-11D8-A84F-000A95E35274@cisco.com>
Subject: Re: [midcom] Re: [MMUSIC] Status of TURN
Date: Mon, 20 Oct 2003 16:47:12 +0100
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Melinda,

I would argue that individual submissions are part of the IETF process, but
are not necessarily open.  If we and Jonathan were to evolve TURN (or
something similar) outside of a working group item, that would effectively
be a behind closed doors effort, and would not be open in the sense that
most IETF working group work is.

Publishing TURN as an individual submission would also set a bad precedent.
There's been a significant boom in the number of traversal solution
providers lately.  Publishing TURN would set the precedent that it is OK to
bypass the open process (at least in the traversal space) and use the
individual submission process to get that attractive RFC number (marketing
people and buyers don't really care what type it is).  True, the individual
process requires a lot of patience.  However, the open process requires a
thick skin, a willingness to compromise, re-writing code, possibly delayed
launch dates and ... a lot of patience!

Pete.

=============================================
Pete Cordell
+44 1473 635863
=============================================

----- Original Message -----
From: "Melinda Shore" <mshore@cisco.com>
To: "Pete Cordell" <pete@tech-know-ware.com>
Cc: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>; "Peterson, Jon"
<jon.peterson@neustar.biz>; "Mmusic" <mmusic@ietf.org>; <midcom@ietf.org>
Sent: 20 October 2003 15:27
Subject: Re: [midcom] Re: [MMUSIC] Status of TURN


> On Monday, October 20, 2003, at 10:01 AM, Pete Cordell wrote:
> > That's generous of you, but I basically feel that there is either a
> > need for
> > this work, in which case it should be done as part of the normal open
> > IETF
> > process, or there isn't a need, in which case it should go no further.
>
> I hope it's understood that the publication of individual
> submissions (i.e. not working group output) is in fact part of
> the normal open IETF process, available to anybody who wishes
> to jump through the various hoops to do it.  RFC 2026 mentions
> it but probably doesn't make it as clear as it should.
>
> Melinda
>
>
> _______________________________________________
> midcom mailing list
> midcom@ietf.org
> https://www1.ietf.org/mailman/listinfo/midcom


_______________________________________________
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Date: Mon, 20 Oct 2003 21:42:42 +0530
From: Deepak Kotian <dkotian3@vsnl.net>
To: avt@ietf.org, IETF MMUSIC WG <mmusic@ietf.org>
Reply-to: Deepak Kotian <dkotian3@vsnl.net>
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Subject: [MMUSIC] Re: ipv6 multicast problem on Linux kernel 2.4.18-3 while doing a
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Hi,

Also an additional information.
F001:0:0:0:0:0:0:43 MULTICast address works
fine with sendto, but FF12::224.5.6.7 does not
work.
Also, the errno is 99 , the errno menitioned in the 
earlier mail was not OK.

Any inputs from anyone on this now.

Thanks and Regards
Deepak

  ----- Original Message ----- 
  From: Deepak Kotian 
  To: avt@ietf.org 
  Sent: Sunday, October 19, 2003 5:15 PM
  Subject: ipv6 multicast problem on Linux kernel 2.4.18-3 while doing a sendto 


  Hi,    

  We  are currently facing a problem when
  we try mutlicast IP FF12::224.5.6.7 on Linux.
  The function inet_pton() returns > 0, that means the IP is OK, but
  when we do a sendto(), it gives an error, EADDRINVAL.

  The kernel used is 2.4.18-3 and ipv6 is enable. 
  Can one please let me know, how to make this multicast IPV6
  sendto() functionality work properly. 

  I think, this is not the right forum, but if some has worked on it,
  can one please share his experience on it.
  Or does anyone know the right forum for this query.

           
  Thanks and Regards
  Deepak


--Boundary_(ID_FKut3SW8i0b9+1uwDNuT9Q)
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
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<BODY bgColor=#ffffff>
<DIV><FONT face=Arial size=2>Hi,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>Also an additional information.</FONT></DIV>
<DIV><FONT face=Arial size=2>F001:0:0:0:0:0:0:43 MULTICast address 
works</FONT></DIV>
<DIV><FONT face=Arial size=2>fine with sendto, but FF12::224.5.6.7 does 
not</FONT></DIV>
<DIV><FONT face=Arial size=2>work.</FONT></DIV>
<DIV><FONT face=Arial size=2>Also, the errno is 99 , the errno menitioned in the 
</FONT></DIV>
<DIV><FONT face=Arial size=2>earlier mail was not OK.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>Any inputs from anyone on this now.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>Thanks and Regards</FONT></DIV>
<DIV><FONT face=Arial size=2>Deepak</FONT></DIV>
<DIV>&nbsp;</DIV>
<BLOCKQUOTE dir=ltr 
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
  <DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
  <DIV 
  style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B> 
  <A title=dkotian3@vsnl.net href="mailto:dkotian3@vsnl.net">Deepak Kotian</A> 
  </DIV>
  <DIV style="FONT: 10pt arial"><B>To:</B> <A title=avt@ietf.org 
  href="mailto:avt@ietf.org">avt@ietf.org</A> </DIV>
  <DIV style="FONT: 10pt arial"><B>Sent:</B> Sunday, October 19, 2003 5:15 
  PM</DIV>
  <DIV style="FONT: 10pt arial"><B>Subject:</B> ipv6 multicast problem on Linux 
  kernel 2.4.18-3 while doing a sendto </DIV>
  <DIV><BR></DIV>
  <DIV><FONT face=Arial size=2>Hi,&nbsp;&nbsp;&nbsp; </FONT></DIV>
  <DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV>
  <DIV><FONT face=Arial size=2>We&nbsp; are currently facing a problem 
  when</FONT></DIV>
  <DIV><FONT face=Arial size=2>we try mutlicast IP FF12::224.5.6.7 on 
  Linux.</FONT></DIV>
  <DIV><FONT face=Arial size=2>The&nbsp;function inet_pton() returns &gt; 0, 
  that means the IP is OK, but</FONT></DIV>
  <DIV><FONT face=Arial size=2>when we do a sendto(), it gives an error, 
  EADDRINVAL.</FONT></DIV>
  <DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV>
  <DIV><FONT face=Arial size=2>The kernel used is 2.4.18-3 and ipv6 is enable. 
  </FONT></DIV>
  <DIV><FONT face=Arial size=2>Can one please let me know, how to make this 
  multicast IPV6</FONT></DIV>
  <DIV><FONT face=Arial size=2>sendto() functionality work properly. 
  </FONT></DIV>
  <DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV>
  <DIV><FONT face=Arial size=2>I think, this is not the right forum, but if some 
  has worked on it,</FONT></DIV>
  <DIV><FONT face=Arial size=2>can one please share his experience on 
  it.</FONT></DIV>
  <DIV><FONT face=Arial size=2>Or does anyone know the right forum for this 
  query.</FONT></DIV>
  <DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV>
  <DIV><FONT face=Arial 
  size=2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</FONT><FONT 
  face=Arial size=2></FONT></DIV>
  <DIV><FONT face=Arial size=2>Thanks and Regards</FONT></DIV>
  <DIV><FONT face=Arial size=2>Deepak</FONT></DIV>
  <DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV></BLOCKQUOTE></BODY></HTML>

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Message-ID: <000b01c3972d$6e4a6e40$0200000a@tkw>
From: "Pete Cordell" <pete@tech-know-ware.com>
To: "Melinda Shore" <mshore@cisco.com>
Cc: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>,
        "Peterson, Jon" <jon.peterson@neustar.biz>, "Mmusic" <mmusic@ietf.org>,
        <midcom@ietf.org>
References: <07AB2C20-0318-11D8-A84F-000A95E35274@cisco.com>
Subject: Re: [midcom] Re: [MMUSIC] Status of TURN
Date: Mon, 20 Oct 2003 18:13:02 +0100
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I agree that the process allows this.  However, I believe it would be unwise
to take this route.

The IESG can recommend that a draft not be published if (among other
reasons) it conflicts with work in the IETF.

The effort was stopped because people were too busy doing other things,
rather than it was not possible to reach consensus, hence it is not a failed
effort in that sense.  Now that people have more time, it seems that there
is still interest in it from more than one group.  Hence the correct thing
to do is see whether the work should be resumed.  (If you like, the work
should have been considered paused rather than stopped.)

It seems an act of considerable bad faith to stop a piece of work going
forward because you are busy, and then later claim that your individual
submission should be adopted because the work was stopped.

There seems only two options: the work item is re-started, or all work is
stopped.  To do otherwise would be an abuse of the spirit of the IETF even
if it is not in violation of the written rules.

Pete.

=============================================
Pete Cordell
+44 1473 635863
=============================================

----- Original Message -----
From: "Melinda Shore" <mshore@cisco.com>
To: "Pete Cordell" <pete@tech-know-ware.com>
Cc: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>; "Peterson, Jon"
<jon.peterson@neustar.biz>; "Mmusic" <mmusic@ietf.org>; <midcom@ietf.org>
Sent: 20 October 2003 17:11
Subject: Re: [midcom] Re: [MMUSIC] Status of TURN


> On Monday, October 20, 2003, at 11:47 AM, Pete Cordell wrote:
> > I would argue that individual submissions are part of the IETF
> > process, but
> > are not necessarily open.  If we and Jonathan were to evolve TURN (or
> > something similar) outside of a working group item, that would
> > effectively
> > be a behind closed doors effort, and would not be open in the sense
> > that
> > most IETF working group work is.
>
> That's one way to look at it, I suppose, but the point really
> is that publication of individual submissions is a route open
> to anybody and that there's no process subversion here.  Note
> as well that it's being published in draft form and he's
> solicited input.
>
> I'm actually not very happy about this, myself, and would
> oppose pretty much any relaying technology.  Aside from the
> technical issues, among the unfortunate consequences of all
> this pre-midcom stuff has been that it's taken the pressure
> off of firewall/NAT vendors to do something useful.  That's
> not good.  But regardless of how I feel personally about all
> this stuff I really think that advancing an individual draft
> towards publication is not even remotely an abuse of IETF
> process.
>
> Melinda
>
>
> _______________________________________________
> midcom mailing list
> midcom@ietf.org
> https://www1.ietf.org/mailman/listinfo/midcom


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Thread-Topic: Status of draft-ietf-mmusic-sdescriptions-01.txt
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From: "Barany, Pete" <pbarany@qualcomm.com>
To: <mmusic@ietf.org>
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Subject: [MMUSIC] Status of draft-ietf-mmusic-sdescriptions-01.txt
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Hi,

Could someone please tell me what is the status of the following I-D?

      <draft-ietf-mmusic-sdescriptions-01.txt>

Thanks in advance.

Regards,

Pete

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Subject: Re: [midcom] Re: [MMUSIC] Status of TURN
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On Monday, October 20, 2003, at 10:01 AM, Pete Cordell wrote:
> That's generous of you, but I basically feel that there is either a 
> need for
> this work, in which case it should be done as part of the normal open 
> IETF
> process, or there isn't a need, in which case it should go no further.

I hope it's understood that the publication of individual
submissions (i.e. not working group output) is in fact part of
the normal open IETF process, available to anybody who wishes
to jump through the various hoops to do it.  RFC 2026 mentions
it but probably doesn't make it as clear as it should.

Melinda


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Subject: RE: [midcom] Re: [MMUSIC] Status of TURN
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From: "Christian Huitema" <huitema@windows.microsoft.com>
To: "Pete Cordell" <pete@tech-know-ware.com>,
        "Melinda Shore" <mshore@cisco.com>
Cc: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>,
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Individual submissions are a well established part of the process, and
should be considered a fine way to make progress when a working group
would not. There is a catch, however. Individual contributions are very
seldom published as proposed standards -- they normally end up as either
experimental or informational. The criteria for publishing something as
informational ought to be, demonstration of at least some significant
interest in the community -- but not necessarily demonstration of
consensus. The bar for refusing publication in the presence of enough
interest has to be somewhat high -- basically, the group has to be
convinced that implementing the proposal as is would damage the
Internet.=20

Obviously, publication as a proposed standard would require the
equivalent of WG consensus. I don't believe there is such consensus on
TURN.

-- Christian Huitema

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Subject: Re: [midcom] Re: [MMUSIC] Status of TURN
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Cc: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>,
        "Peterson, Jon" <jon.peterson@neustar.biz>, "Mmusic" <mmusic@ietf.org>,
        <midcom@ietf.org>
To: "Pete Cordell" <pete@tech-know-ware.com>
From: Melinda Shore <mshore@cisco.com>
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On Monday, October 20, 2003, at 11:47 AM, Pete Cordell wrote:
> I would argue that individual submissions are part of the IETF 
> process, but
> are not necessarily open.  If we and Jonathan were to evolve TURN (or
> something similar) outside of a working group item, that would 
> effectively
> be a behind closed doors effort, and would not be open in the sense 
> that
> most IETF working group work is.

That's one way to look at it, I suppose, but the point really
is that publication of individual submissions is a route open
to anybody and that there's no process subversion here.  Note
as well that it's being published in draft form and he's
solicited input.

I'm actually not very happy about this, myself, and would
oppose pretty much any relaying technology.  Aside from the
technical issues, among the unfortunate consequences of all
this pre-midcom stuff has been that it's taken the pressure
off of firewall/NAT vendors to do something useful.  That's
not good.  But regardless of how I feel personally about all
this stuff I really think that advancing an individual draft
towards publication is not even remotely an abuse of IETF
process.

Melinda


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Date: Tue, 21 Oct 2003 13:33:44 -0400
From: Flemming Andreasen <fandreas@cisco.com>
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CC: mmusic@ietf.org
Subject: Re: [MMUSIC] Status of draft-ietf-mmusic-sdescriptions-01.txt
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There will be an -02 version before the end of this week.

-- Flemming

"Barany, Pete" wrote:

> Hi,
>
> Could someone please tell me what is the status of the following I-D?
>
>       <draft-ietf-mmusic-sdescriptions-01.txt>
>
> Thanks in advance.
>
> Regards,
>
> Pete
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic


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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Security Preconditions for Session Description 
			  Protocol Media Streams
	Author(s)	: F. Andreasen, et. al.
	Filename	: draft-andreasen-mmusic-securityprecondition-00.txt
	Pages		: 8
	Date		: 2003-10-20
	
This document defines a new security precondition for the Session 
Description Protocol precondition framework described in RFC 3312.  
A security precondition can be used to delay session establishment 
or modification until media stream security has been negotiated 
successfully.

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In off-list discussion with Magnus and Gonzalo, is was asked if it is 
allowed to have multiple c= lines in a m= block, but not at session 
level, since RFC 2327 is not entirely clear. The following 
clarification was proposed for the end of Section 5.7 of 
draft-ietf-mmusic-sdp-new-14:

OLD:
    Multiple addresses or "c=" lines MAY be specified on a per-media
    basis. They MUST NOT be specified for a session-level "c=" field.

NEW:
    Multiple addresses or "c=" lines MAY be specified on a per-media
    basis only if they provide multicast addresses for different layers
    in a hierarchical or layered encoding scheme. They MUST NOT be
    specified for a session-level "c=" field.

Any comments?

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


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Subject: Re: [MMUSIC] SDP b= clarification
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From: Colin Perkins <csp@csperkins.org>
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On Tuesday, Sep 2, 2003, at 22:58 Europe/London, Tom Marshall wrote:
> On Tue, Sep 02, 2003 at 09:58:17PM +0100, Colin Perkins wrote:
>> On Thursday, Aug 28, 2003, at 20:38 Europe/London, Tom Marshall wrote:
>>> In RFC3556, the example SDP shows multiple b=lines in the media
>>> sections.
>>> In RFC2327 (and the updated draft sdp-new-13), the b= 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= lines are
>>> presumedly desirable, I think the sdp-new draft should be updated to
>>> reflect
>>> this.
>>
>> The ABNF in draft-ietf-mmusic-sdp-new-13.txt explicitly allows
>> repetition.
>>
>>   bandwidth-fields =    *("b=" bwtype ":" bandwidth CRLF)
>>   bwtype =              token
>>   bandwidth =           1*DIGIT
>
> Cool.  Could we get a note in the text that corresponds with the ABNF?

Provided there are no objections, if you suggest explicit wording 
changes, I can incorporate them into -15.

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


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From: Thomas Zeng <zeng@PacketVideo.COM>
To: "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>
Cc: IETF MMUSIC WG <mmusic@ietf.org>
Subject: RE: [MMUSIC] RTSP: Minutes telecon 14th of October 2003
Date: Tue, 21 Oct 2003 16:12:14 -0700
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Magnus:

During one of the telecon for RTSP, I recall that a decision was made to 
deprecate "time=" parameter in Range header.

We have found a use case in the context of playlist support, where we would
like to use
"time=" parameter in Range header in the PLAY request as designed in
RFC2326, quoted below:

10.5 PLAY

...

   The Range header may also contain a time parameter. This parameter
   specifies a time in UTC at which the playback should start. If the
   message is received after the specified time, playback is started
   immediately. The time parameter may be used to aid in synchronization
   of streams obtained from different sources.



In light of this, would it be possible to retain this "time parameter"?

thanks,
thomas
-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent: Tuesday, October 14, 2003 9:53 PM
To: Thomas Zeng
Cc: 'Magnus Westerlund'; IETF MMUSIC WG
Subject: Re: [MMUSIC] RTSP: Minutes telecon 14th of October 2003


Yes, SIP BYE is used to terminate the call, which also results in the 
termination of the media sessions.

-Jonathan R.

Thomas Zeng wrote:

> On item g:
> 
>     - Look into SIP to see how they have resolved the discrepancies
>       between media and signaling channel.
> 
> Could someone fammiliar with SIP help me out here: I need to know what
> method of SIP to look, is it SIP BYE method by any chance?
> 
> (I saw that SIP has a BYE method to terminate the call in the signalling
> channel. )
> 
> thanks,
> thomas
> 
> 
> 
> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Tuesday, October 14, 2003 11:04 AM
> To: IETF MMUSIC WG
> Subject: [MMUSIC] RTSP: Minutes telecon 14th of October 2003
> 
> 
> Hi,
> 
> Here is the preliminary minutes for todays telecon. Please comment.
> 
> RTSP Telecon the 14th of October
> 
> Participants:
> Magnus Westerlund, Colin Perkins, Aravind Narasimhan, and Tom Marshall.
> 
> Agenda:
> 
> 1. Opening of meeting.
> 
> 2. Administrative
>     - Scribe
>     - IETF in Minneapolis
>       * Off-line Meeting?
>       * What to present.
> 
> 3. Since last meeting (1st of October).
> a. http://rtsp.org/bug507882 - RTP timestamp and pause
>      - Aravind to write up text
> 
> b. http://rtsp.org/bug742328 - How to handle PLAY request when at the
> end of a media clip
>      - Text sent to mailing list, please review.
> 
> c. http://rtsp.org/bug742348 - Usage of implicit REDIRECT.
>      - Needs text
> 
> d. http://rtsp.org/bug640272 - Default way of deriving aggregate URL
>     - Text sent to the mailing list
> 
> e. http://rtsp.org/bug771638 - a=range in SDP to indicate live media
>     - Text sent to the mailing list.
> 
> f. http://rtsp.org/bug531189 - How does REDIRECT method work
>     - Needs somebody to write up.
> 
> g. http://rtsp.org/bug810678 - How to indicate end of media stream
>     - Strawman proposal available from Thomas. For discussion.
> 
> h. http://rtsp.org/bug770522 - Request to response time is not limited.
>     - Some discussion on mailing list. Sean to write up text.
> 
> i. http://rtsp.org/bug810681 -     The updated specification version
number
>     - Awaiting investigation results regarding changes by Aravind.
> 
> j. http://rtsp.org/bug795295 - TEARDOWN using * and the Ready-nm state
>     - Posted to the mailing list, no comments. Unless comments are
>       received it will be edited in.
> 
> k. http://rtsp.org/bug804219 - How to handle sync in re-SETUP
>     - Text posted to the list. No comments received so far.
> 
> l. http://rtsp.org/bug806404 - Re-SETUP of transport parameters
>     - Awaiting proposal text from Geetha.
> 
> m. http://rtsp.org/bug810666 - Removal of SETUP in play state
>     - Proposed to be removed. Message sent to the list.
> 
> 4. New Issues
> 
> a. http://rtsp.org/bug811815 - Use of c= in SDP to know possible address 
> classes
> 
> b. http://rtsp.org/bug795314 - Caching of OPTIONS request
> 
> 5. Any other business
> 
> 6. Next Telecon: Proposal Wednesday, 3rd of December.
> 
> 7. End of meeting
> 
> Minutes
> -------
> 
> 1. Meeting was opened.
> 
> 2. - Magnus volunteered to write the minutes.
> 
>     - IETF meeting in Minneapolis:
>       * Colin will request a conference room for a out-side of normal
>         agenda meeting. A guess of on the number of participants are
>         around 15.
> 
>       * The authors will perform a status update on the specification in
>         the WG session. The current resolutions and open issues will be
>         presented.
> 
> 3.
> 
> a. Nothing has happened.
> 
> b. Await review from the WG.
> 
> c. Status not changed.
> 
> d. Await review from the WG.
> 
> e. Was not sent yet, but will be available directly after meeting.
> Review comments needed.
> 
> f. The remaining issue is with implicit closing of the session. The
>     previous meeting has agreement on the functionality but some minor
>     issues around use of "connection" needs resolving. Best way of
>     resolving this is probably to present a specification text proposal.
>     Aravind will take this as a second priority issue.
> 
> g. Send comments on the document to list. The discussion resolved that
>     this can be divided into several pieces:
> 
>     - We need to document in the RTSP specification regarding RTP what
>       happens when a source times out a SSRC or receives a RTCP BYE for a
>       source.
> 
>     - Using BYE is not suitable way to indicate the end of the stream.
>       The motivation for this can be seen in Thomas proposal.
> 
>     - Look into SIP to see how they have resolved the discrepancies
>       between media and signaling channel.
> 
> h. Awaits text from Sean.
> 
> i. This is a later decision when we have received a list of changes that
>     are accurate.
> 
> j. Unless comments are received by Monday the 20th of October, this
>     resolution will be edited in. Magnus will post a reply to the initial
>     mail pointing out the deadline.
> 
> k. Will be kept in the text unless comments are review by Wednesday the
>     22th of October.
> 
> l. Awaits text.
> 
> m. Will be edited in unless comments are received by Monday the 20th of
>     October.
> 
> 4.
> a. Question raised is: Are there need for a server to announce through
>     SDP or other pre SETUP mechanism which address classes it supports to
>     the client?
> 
>     There was discussion on this topic and the resolution was the
>     following:
> 
>     It should be assumed that a server being possible to find in DNS with
>     a certain address class will support both signaling and media
>     delivery for that address. Thus a RTSP request over TCP/IPv6 will
>     send media using that source IPv6 address.
> 
> b. The OPTIONS request has some dependency on server state and what
> content that are addressed. Therefore it seems doubtful that caching of
> OPTIONS request are possible. In general it seems that the only
> really cachable RTSP request/response pair is for DESCRIBE.
> 
> 5. No other business.
> 
> 6. Preliminary date is the 3rd of December, which is three weeks after
> the IETF meeting.
> 
> 7. Meeting was closed.
> 
> 
> 

-- 
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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Message-ID: <004d01c3985b$b1f25340$0200000a@tkw>
From: "Pete Cordell" <pete@tech-know-ware.com>
To: "Christian Huitema" <huitema@windows.microsoft.com>,
        "Melinda Shore" <mshore@cisco.com>
Cc: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>,
        "Peterson, Jon" <jon.peterson@neustar.biz>, "Mmusic" <mmusic@ietf.org>,
        <midcom@ietf.org>
References: <DAC3FCB50E31C54987CD10797DA511BA05BD5D4A@WIN-MSG-10.wingroup.windeploy.ntdev.microsoft.com>
Subject: Re: [midcom] Re: [MMUSIC] Status of TURN
Date: Wed, 22 Oct 2003 06:14:03 +0100
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What you say Christian is technically absolutely correct.

However this communities appetite for implementing and labelling their
products with even -00 draft material suggests to me that many implementers
(and probably their customers) are not bothered by these subtleties of IETF
process.

Certainly it would help to know whether informational or proposed standard
is
being sought for this draft.

Pete.

=============================================
Pete Cordell
+44 1473 635863
=============================================

----- Original Message -----
From: "Christian Huitema" <huitema@windows.microsoft.com>
To: "Pete Cordell" <pete@tech-know-ware.com>; "Melinda Shore"
<mshore@cisco.com>
Cc: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>; "Peterson, Jon"
<jon.peterson@neustar.biz>; "Mmusic" <mmusic@ietf.org>; <midcom@ietf.org>
Sent: 20 October 2003 20:27
Subject: RE: [midcom] Re: [MMUSIC] Status of TURN


Individual submissions are a well established part of the process, and
should be considered a fine way to make progress when a working group
would not. There is a catch, however. Individual contributions are very
seldom published as proposed standards -- they normally end up as either
experimental or informational. The criteria for publishing something as
informational ought to be, demonstration of at least some significant
interest in the community -- but not necessarily demonstration of
consensus. The bar for refusing publication in the presence of enough
interest has to be somewhat high -- basically, the group has to be
convinced that implementing the proposal as is would damage the
Internet.

Obviously, publication as a proposed standard would require the
equivalent of WG consensus. I don't believe there is such consensus on
TURN.

-- Christian Huitema

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From: Geetha Srikantan <Geetha.Srikantan@Sun.COM>
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Hello everyone,

Here is the promised write-up on RTSP transport negotiation..sorry for
the delay. If there is agreement on the content of this note, I can
write up text for the RTSP draft.

thanks
geetha


Topic: RTSP Transport negotiation using SETUP during play state.

Introduction

The RTSP specification allows for negotiation of transport parameters
via the SETUP method, while the RTSP session is in play state. The
transport parameters that can be negotiated and the exact form of this
negotiation is under-specified in the RTSP documents. This note is an
attempt to clarify some of the issues around transport negotiation for
inclusion in the ongoing revision of RTSP.

The recent interest in mobility for RTSP implementations also impacts
transport parameters used by the session. For instance, a wireless
RTSP client could be assigned a new IP address, or may experience
change of network characteristics, while it is in the midst of a RTSP
session. These issues are also addressed in this note.


Outline:

1. Objective
2. Enumeration of all Transport Parameters
3. Transport Parameters Excluded from Negotiation
4. Negotiable Transport Parameters
5. Impact of Transport Negotiation
6. Examples
	6.1 Client IP Address Changes
	6.2 UDP -> TCP
	6.3 TCP -> UDP
7. Other transport related changes to RTSP draft
8. Summary



1. Objective: 

The main goals of this document are to
	* Provide an enumeration of transport parameters.
	* Identify transport parameters that cannot be negotiated
	  with minimal impact on other sessions.
	* Propose transport parameters that can be negotiated during
	  play state.
	* Determine implications of changing these parameters via a
	  SETUP request while the session is in play state.
	* Provide examples demonstrating transport negotiation.


2. Enumeration of all Transport Parameters

   The following is a list of all transport parameters in the current
   RTSP draft.

	unicast/multicast
	destination
	source
	mode (=PLAY)
	interleaved
	layers (for multicast)
	ttl (for multicast
	port (for multicast)
	client_port (for RTP)
	server_port (for RTP)
	ssrc (for RTP)
	url


3. Transport Parameters Excluded from Negotiation

   It is recommended that the following parameters be excluded from the
   negotiation process.

	mode
	url
	unicast/multicast
	layers
	ttl
	ssrc
	destination
	source


    As the "mode" parameter is restricted to a single value in the
current draft, it can be excluded.

    As per the recent proposal to not support addition of urls via SETUP
during play state, the "url" parameter can be excluded. 

    Changing from multicast to unicast based on a single RTSP session's
SETUP, would impact all the other clients accessing the multicast
session. 
Several issues need to be considered in this case, including transport
negotiation. Similarly, changing from unicast to multicast also has
implications beyond a single RTSP session. Changes to other
multicast-related parameters, "layers", "ssrc", "ttl" most likely
requires specification beyond transport negotiation.
The recommendation is to *not* specify changing the transport from
"unicast" to "multicast" or vice versa, and also not specify changes
to the "layers" and "ttl" parameters, in this draft. It is recommedned 
that negotiation of multicast-related transport parameters should be
deferred to an RTSP-extension document. 

    Once an RTSP session has been initiated and the "destination" and
"source" have been established - i.e. a multicast "destination" has
been identified, and the stream "source" has been identified, there
should be no need to change these parameters.  It is recommended that
the "destination" and "source" parameters not be negotiated when the
session is in play state.  The case of the client IP address change is
signaled in a different manner, outlined in the next section.


4. Negotiable Transport Parameters

   Hence there are three main conditions that require transport
negotiation for unicast:

	* client IP address changes
	* client wants to switch transport from UDP -> TCP 
	* client wants to switch transport from TCP -> UDP

   Changes in the client IP address can be indicated using a new
"client_ip" parameter. Hence the parameters that are impacted by
transport negotiation are:

	client_ip
	interleaved
	client_port
	server_port

 
5. Impact of Transport Negotiation

   The first column describes the primary change, the second and third
columns describe any secondary transport parameters that MUST or 
MAY change as a result of the primary change.


Changed			MUST			MAY	
-------			----			---
	
Client IP address	client_ip		client_port
						server_port

UDP -> TCP		interleaved		Server may change
			Drop client_port	interleaved channel ids
			Drop server_port	in response.

TCP -> UDP		client_port in request	client_ip
			server_port in response client_port in response	
			Drop interleaved



6. Examples

6.1 Client IP Address Changes

     C->S: DESCRIBE rtsp://foo/twister.sdp RTSP/1.1
	   CSeq: 1

     S->C: RTSP/1.0 200 OK
	   CSeq: 1
           Content-Type: application/sdp
	   Content-Length: 164

	   v=0
           o=- 2890844256 2890842807 IN IP4 172.16.2.93
           s=RTSP Session
           i=An Example of RTSP Session Usage
           a=control:rtsp://foo/twister
           t=0 0
           m=audio 0 RTP/AVP 0
           a=control:rtsp://foo/twister/audio
           m=video 0 RTP/AVP 26
           a=control:rtsp://foo/twister/video

     C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
           CSeq: 2
           Transport: RTP/AVP;unicast;client_port=8000-8001

     S->C: RTSP/1.0 200 OK
           CSeq: 2
           Transport: RTP/AVP;unicast;client_port=8000-8001;
                      server_port=9000-9001
           Session: 12345678

     C->S: SETUP rtsp://foo/twister/video RTSP/1.0
           CSeq: 3
           Transport: RTP/AVP;unicast;client_port=8002-8003
           Session: 12345678

     S->C: RTSP/1.0 200 OK
           CSeq: 3
           Transport: RTP/AVP;unicast;client_port=8002-8003;
                      server_port=9004-9005
           Session: 12345678

     C->S: PLAY rtsp://foo/twister RTSP/1.0
           CSeq: 4
           Range: npt=0-
           Session: 12345678

     S->C: RTSP/1.0 200 OK
           CSeq: 4
           Session: 12345678
           RTP-Info: url=rtsp://foo/twister/video;
             seq=9810092;rtptime=3450012,
	   url=rtsp://foo/twister/audio;seq=5923;rtptime=22234432
	   Range: npt=0-30.76

... now change client IP address ....

     C->S: SETUP rtsp://foo/twister/video RTSP/1.0
           CSeq: 5
	   Transport: RTP/AVP;unicast;client_ip=<new IP address;
	              client_port=8002-8003;server_port=9004-9005
	   Range: npt=5-
           Session: 12345678
			
     S->C: RTSP/1.0 200 OK
           CSeq: 5
	   Transport: RTP/AVP;unicast;client_ip=<new IP address>;
		      client_port=8002-8003;server_port=9004-9005
	   RTP-Info: url=rtsp://foo/twister/video;
		    seq=<new>;rtptime=<new>,
	   Range: npt=5-30.76
	   Session: 12345678


     C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
           CSeq: 6
	   Transport:RTP/AVP;unicast;client_ip=<new IP address>;
		     client_port=8000-8001;server_port=9000-9001
	   Range: npt=5-
           Session: 12345678

     S->C: RTSP/1.0 200 OK
           CSeq: 6
	   Transport: RTP/AVP;unicast;client_ip=<new IP address>;
		      client_port=8000-8001;server_port=9000-9001
	   RTP-Info:url=rtsp://foo/twister/audio;
		    seq=<new>;rtptime=<new>
	   Range: npt=5-30.76	
	   Session: 12345678



     
6.2 UDP -> TCP

     C->S: DESCRIBE rtsp://foo/twister.sdp RTSP/1.0
	   CSeq: 1

     S->C: RTSP/1.0 200 OK
	   Content-Base: rtsp://foo/twister/
	   Content-Length: 230
	   Content-Type: application/sdp
	   Cseq: 1

	   v=0
	   o=- 3262117360 3262117360 IN IP4 0.0.0.0
	   s=RTSP Session
	   c=IN IP4 0.0.0.0
	   t=0 0
	   a=control:*
	   a=range:npt=0-10.566000
	   m=video 0 RTP/AVP 26
	   a=control:rtsp://foo/twister/video
	   m=audio 0 RTP/AVP 0
	   a=control:rtsp://foo/twister/audio

     C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
	   CSeq: 2
	   Transport: RTP/AVP/UDP;unicast;client_port=8000-8001

     S->C: RTSP/1.0 200 OK
	   CSeq: 2
	   Session: 12345678
	   Transport: RTP/ACP/UDP;unicast;client_port=8000-8001;
		      server_port=9000-9001

     C->S: PLAY rtsp://foo/twister RTSP/1.0
	   CSeq: 3
	   Session: 12345678

     S->C: RTSP/1.0 200 OK
	   Content-Base: rtsp://foo/twister/
	   CSeq: 3
	   RTP-Info: url=rtsp://foo/twister/audio;seq=7028;rtptime=1339949552
	   Range: npt=0-
	   Session: 12345678

... now request TCP ...

     C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
	   CSeq: 4
	   Transport: RTP/AVP/TCP;interleaved=0-1

     S->C: RTSP/1.0 200 OK
	   Cseq: 4
	   Transport: RTP/AVP/TCP;unicast;
		      client_ip=<client IP>;interleaved=0-1
	   RTP-Info: url=rtsp://foo/twister/audio;seq=<new>;rtptime=<new>
	   Range: npt=0-10.56600
	   Session: 12345678



6.3 TCP -> UDP

     C->S: DESCRIBE rtsp://foo/twister.sdp RTSP/1.0
	   CSeq: 1

     S->C: RTSP/1.0 200 OK
	   Content-Base: rtsp://foo/twister/
	   Content-Length: 230
	   Content-Type: application/sdp
	   Cseq: 1

	   v=0
	   o=- 3262117360 3262117360 IN IP4 0.0.0.0
	   s=RTSP Session
	   c=IN IP4 0.0.0.0
	   t=0 0
	   a=control:*
	   a=range:npt=0-10.566000
	   m=video 0 RTP/AVP 26
	   a=control:rtsp://foo/twister/video
	   m=audio 0 RTP/AVP 0
	   a=control:rtsp://foo/twister/audio

     C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
	   CSeq: 2
	   Transport: RTP/AVP/TCP;interleaved=0-1

     S->C: RTSP/1.0 200 OK
	   Cseq: 2
	   Session: 12345678
	   Transport: RTP/AVP/TCP;unicast;client_ip=<client IP>;interleaved=0-1

     C->S: PLAY rtsp://foo/twister RTSP/1.0
	   CSeq: 3
	   Session: 12345678

     S->C: RTSP/1.0 200 OK
	   Content-Base: rtsp://foo/twister/
	   Cseq: 3
	   RTP-Info: url=rtsp://foo/twister/audio;seq=7028;rtptime=1339949552
	   Range: npt=0-
	   Session: 12345678

... now request UDP ...


     C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
	   CSeq: 4
	   Transport: RTP/AVP/UDP;unicast;client_port=8000-8001
	   Range: npt=0-

     S->C: RTSP/1.0 200 OK
	   CSeq: 4
	   Transport: RTP/ACP/UDP;unicast;client_port=8000-8001;
		      server_port=9000-9001
	   RTP-Info: url=rtsp://foo/twister/audio;seq=<new>;rtptime=<new>
	   Range: npt=0-10.56600
	   Session: 12345678




7. Other transport related changes to RTSP draft

A few comments on draft-ietf-mmusic-rfc2326bis-04.txt relating to
transport parameters.

(a) Section 13.40:
    Page 81: Are "src_addr" and "dest_addr" parameters
    going away? If these are relevant for NAT, perhaps they belong in
    the NAT document?

    Page 83-83: BNF indicates "client_port" and
    "server_port" will be indicated as 'port1-port2'. This doesn't seem
    adequate for the case where the RTP and RTCP ports are non-adjacent
    (per the new RTP spec rfc 3550). 

    Proposal:
	client_port: rtp=<port>, rtcp=<port>
	server_port: rtp=<port>, rtcp=<port>

(b) Appendix B, B.1.1
    Page 115: Mentions "src_addresses" and
    "dest_addresses" - what are these?

    Page 116: "non-continuous" ports - might want to rephrase to
    "non-adjacent" (or "non-continguous") ports?



8. Summary


This note recommends negotiation of a few transport parameters and
specifies mechanisms to do so. One new transport parameter "client_ip" 
is recommended for use in cases where the client's IP address changes
during the session. 

It is also recommended that the client_port and server_port parameters
be specified further with the use of tags "rtp_port" and "rtcp_port"
to handle the case where these ports are non-contiguous.


-- 
Geetha Srikantan <Geetha.Srikantan@Sun.COM>


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Hi Thomas,

If you have a use cases for this, I think it is possible to keep the 
timed functionality, under the condition that you provide all necessary 
specification text to solve the problems with timed PLAYs.

I would also like to have some more information about use case.

The big problem with the timed requests are the following:

- Are they allowed to be stacked, are multiple possible to be requested 
for later execution? If yes, the real problems exist.

- If you use timed requests how does the interaction happens with the 
state machine. When are the state transition made, when the request is 
received or when it is executed.

- How is errors at the time of execution handled?

- If multiple requests are allowed, how does one handle if two request 
collide.

- How do you stop a timed request, between acceptance and execution?

These are probably not all the problems, simply the ones I can remember 
of the top of my head.

So please provide a solution proposal so we can discuss and review if it 
solves the problems in a satisfactory way.

cheers

Magnus

Thomas Zeng wrote:

> Magnus:
> 
> During one of the telecon for RTSP, I recall that a decision was made to 
> deprecate "time=" parameter in Range header.
> 
> We have found a use case in the context of playlist support, where we would
> like to use
> "time=" parameter in Range header in the PLAY request as designed in
> RFC2326, quoted below:
> 
> 10.5 PLAY
> 
> ...
> 
>    The Range header may also contain a time parameter. This parameter
>    specifies a time in UTC at which the playback should start. If the
>    message is received after the specified time, playback is started
>    immediately. The time parameter may be used to aid in synchronization
>    of streams obtained from different sources.
> 
> 
> 
> In light of this, would it be possible to retain this "time parameter"?
> 
-- 

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: Colin Perkins <csp@csperkins.org>
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] SDP b= clarification
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> >>The ABNF in draft-ietf-mmusic-sdp-new-13.txt explicitly allows
> >>repetition.
> >>
> >>  bandwidth-fields =3D    *("b=3D" bwtype ":" bandwidth CRLF)
> >>  bwtype =3D              token
> >>  bandwidth =3D           1*DIGIT
> >
> >Cool.  Could we get a note in the text that corresponds with the ABNF?
>=20
> Provided there are no objections, if you suggest explicit wording=20
> changes, I can incorporate them into -15.

This would be the simplest change:

  --- draft-ietf-mmusic-sdp-new-14.txt    2003-10-22 12:51:26.000000000 -07=
00
  +++ draft-ietf-mmusic-sdp-new-14-edited.txt     2003-10-22 13:18:57.00000=
0000 -0700
  @@ -495,7 +495,7 @@
            p=3D* (phone number)
            c=3D* (connection information - not required if included in
                 all media)
  -         b=3D* (bandwidth information)
  +         b=3D* (zero or more bandwidth information lines)
            One or more time descriptions (see below)
  =20
  =20
  @@ -519,7 +519,7 @@
            i=3D* (media title)
            c=3D* (connection information - optional if included at
                 session-level)
  -         b=3D* (bandwidth information)
  +         b=3D* (zero or more bandwidth information lines)
            k=3D* (encryption key)
            a=3D* (zero or more media attribute lines)
  =20
--=20
Football combines the two worst features of American life.
It is violence punctuated by committee meetings.
        -- George F. Will

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Subject: Re: [MMUSIC] RTSP: Proposal to remove SETUP and TEARDOWN of individual media streams in PLAY state
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On Thu, Oct 02, 2003 at 04:54:40PM +0200, Magnus Westerlund wrote:
> Hi,
>=20
> Another issue that was discussed on yesterdays telecon was the removal=20
> of the part specifying that usage of setup and teardown on individual=20
> media stream in PLAY state is possible. This does not effect the=20
> functionality for:
>=20
> - SETUP and Teardown in ready state
> - The usage of SETUP to change transport parameters.
> - Usage of aggregated Teardown in play state to end session.
>=20
> The removal of this functionality will be done in such way that it will=
=20
> be possible in the future to specify this as extension functionality.

I have a use case for this functionality, although I think it is probably
best left to an extension tag.  I'm interested in how it might be specified
such that it's compatible with the core spec.

--=20
Nothing is illegal if a hundred businessmen decide to do it, and that's true
anywhere in the world.
        -- Andrew Young

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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-810666 ] Removal of SETUP in play state (SourceForge.net)
   3. [ rtspspec-Bugs-810666 ] Removal of SETUP in play state (SourceForge.net)
   4. [ rtspspec-Bugs-585865 ] Use of Date, and Host in RTSP (SourceForge.net)
   5. [ rtspspec-Bugs-795295 ] TEARDOWN using * and the Ready-nm state (SourceForge.net)

--__--__--

Message: 1
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Wed, 22 Oct 2003 09:04:36 -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 (Comment added) made by magwes
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: Magnus Westerlund (magwes)
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. 

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

>Comment By: Magnus Westerlund (magwes)
Date: 2003-10-22 18:04

Message:
Logged In: YES 
user_id=302620

As no comments has been raised, this will be edited out. 

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

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: Wed, 22 Oct 2003 09:03:52 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-810666 ] Removal of SETUP in play state

Bugs item #810666, was opened at 2003-09-22 17:30
Message generated for change (Comment added) made by magwes
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=810666&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: Magnus Westerlund (magwes)
Summary: Removal of SETUP in play state

Initial Comment:
On MMUSIC there is a proposal to remove the
functionality for setup in play state to add a new
individual media stream. This not about changing
transport parameters!

First the need for this is not clear. The only use
cases seems to be to do switching between different
media streams variants. As this doesn't look as the
best method and secondly that switching at all is an
add on functionality this seem to not belong in core spec.

Secondly removing this will simplify the specification
writing.

Third, if the method is need in the future it will be
possible to write an extension specification. 


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

>Comment By: Magnus Westerlund (magwes)
Date: 2003-10-22 18:03

Message:
Logged In: YES 
user_id=302620

No comment against this removal has been raised. Therefore
this will be editied.

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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=810666&group_id=23194


--__--__--

Message: 3
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Wed, 22 Oct 2003 09:30:01 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-810666 ] Removal of SETUP in play state

Bugs item #810666, was opened at 2003-09-22 17:30
Message generated for change (Comment added) made by magwes
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=810666&group_id=23194

Category: None
>Group: Resolved - Added to text
Status: Open
>Resolution: Accepted
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Magnus Westerlund (magwes)
Summary: Removal of SETUP in play state

Initial Comment:
On MMUSIC there is a proposal to remove the
functionality for setup in play state to add a new
individual media stream. This not about changing
transport parameters!

First the need for this is not clear. The only use
cases seems to be to do switching between different
media streams variants. As this doesn't look as the
best method and secondly that switching at all is an
add on functionality this seem to not belong in core spec.

Secondly removing this will simplify the specification
writing.

Third, if the method is need in the future it will be
possible to write an extension specification. 


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

>Comment By: Magnus Westerlund (magwes)
Date: 2003-10-22 18:30

Message:
Logged In: YES 
user_id=302620

The removal has now been performed and checked in in the
following commit:

/cvsroot/rtspspec/changes.tex,v  <--  changes.tex
new revision: 1.18; previous revision: 1.17
/cvsroot/rtspspec/iana.tex,v  <--  iana.tex
new revision: 1.14; previous revision: 1.13
/cvsroot/rtspspec/machine.tex,v  <--  machine.tex
new revision: 1.13; previous revision: 1.12
/cvsroot/rtspspec/methods.tex,v  <--  methods.tex
new revision: 1.33; previous revision: 1.32


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

Comment By: Magnus Westerlund (magwes)
Date: 2003-10-22 18:03

Message:
Logged In: YES 
user_id=302620

No comment against this removal has been raised. Therefore
this will be editied.

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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=810666&group_id=23194


--__--__--

Message: 4
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Wed, 22 Oct 2003 09:43:08 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-585865 ] Use of Date, and Host in RTSP

Bugs item #585865, was opened at 2002-07-24 13:22
Message generated for change (Comment added) made by magwes
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=585865&group_id=23194

Category: General
>Group: Resolved - Added to text
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Magnus Westerlund (magwes)
Summary: Use of Date, and Host in RTSP

Initial Comment:
There exist a number of HTTP headers that are just
referenced into the specification. Several is hard to
interpret on how they relate to RTSP and what
requirement level that is needed. 

Date:
The date header MUST be sent by origin server in
responses. What is an origin server and what exceptions
exist for RTSP? For RTSP does that include requests also?

Host:
The comment is that this should not be used in RTSP.
This should be reflected in the heade table. 

Other headers that are simply referenced needs most
probably some specification on applicability and
exceptions.


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

>Comment By: Magnus Westerlund (magwes)
Date: 2003-10-22 18:43

Message:
Logged In: YES 
user_id=302620

The Host and Date header has both been clarified on there
usage. Edits are checked in for both. The Date has been
there for some versions, while the host text is new. 

If some other headers also needs edit, has not been
evaluated at this time. 

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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=585865&group_id=23194


--__--__--

Message: 5
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Wed, 22 Oct 2003 09:30:27 -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 (Comment added) made by magwes
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=795295&group_id=23194

Category: None
>Group: Resolved - Added to text
Status: Open
>Resolution: Accepted
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Magnus Westerlund (magwes)
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. 

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

>Comment By: Magnus Westerlund (magwes)
Date: 2003-10-22 18:30

Message:
Logged In: YES 
user_id=302620

The removal has now been performed and checked in in the
following commit:

/cvsroot/rtspspec/changes.tex,v  <--  changes.tex
new revision: 1.18; previous revision: 1.17
/cvsroot/rtspspec/iana.tex,v  <--  iana.tex
new revision: 1.14; previous revision: 1.13
/cvsroot/rtspspec/machine.tex,v  <--  machine.tex
new revision: 1.13; previous revision: 1.12
/cvsroot/rtspspec/methods.tex,v  <--  methods.tex
new revision: 1.33; previous revision: 1.32


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

Comment By: Magnus Westerlund (magwes)
Date: 2003-10-22 18:04

Message:
Logged In: YES 
user_id=302620

As no comments has been raised, this will be edited out. 

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

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




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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-810678 ] How to indicate end of media stream (SourceForge.net)
   2. [ rtspspec-Bugs-513753 ] RTSP extensions and IANA (SourceForge.net)
   3. [ rtspspec-Bugs-640272 ] Default way of deriving aggregate URL (SourceForge.net)
   4. [ rtspspec-Bugs-493265 ] Pause - Play behaviour clarification (SourceForge.net)
   5. [ rtspspec-Bugs-531189 ] How does REDIRECT method work (SourceForge.net)
   6. [ rtspspec-Bugs-500868 ] Require: option-tag values unclear (SourceForge.net)
   7. [ rtspspec-Bugs-500906 ] Client liveness criteria vague (SourceForge.net)

--__--__--

Message: 1
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Thu, 23 Oct 2003 07:44:24 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-810678 ] How to indicate end of media stream

Bugs item #810678, was opened at 2003-09-22 17:48
Message generated for change (Comment added) made by magwes
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=810678&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: How to indicate end of media stream

Initial Comment:
discussed at the 57th IETF meeting 16th of July 2003.

Issue:  Possible methods for signalling end of
stream/media (not session) in RTP.

Discussion: Need to signal that a stream/media has
ended exist mostly in unbound sessions. Proposal
brought up to indicate this is RTCP BYE when the sender
stops sending and has waited some time it can remove
the SSRC of that source and send the BYE to indicate
this. Another proposal was to send this information in
RTSP methods. One could use sever to client TEARDOWN to
indicate this session really are over, nothing more can
be done. PAUSE can be sent when the media has been
served but the client may rewind do other operations.
The TEARDOWN idea was not liked as certain applications
require session data to be sent from client to server
using SET_PARAMETER. A new method should be evaluated.
There might also exist significant synchronization
problems in this.

Resolution: Propose a RTSP extension with a new method.
Thomas Zeng to write strawman proposal for evaluation
to mailing list.


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

>Comment By: Magnus Westerlund (magwes)
Date: 2003-10-23 16:44

Message:
Logged In: YES 
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A strawman proposal is avail in this MMUSIC messge:
https://www1.ietf.org/mail-archive/working-groups/mmusic/current/msg01978.html


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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=810678&group_id=23194


--__--__--

Message: 2
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Thu, 23 Oct 2003 07:38:59 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-513753 ] RTSP extensions and IANA

Bugs item #513753, was opened at 2002-02-06 15:05
Message generated for change (Comment added) made by magwes
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=513753&group_id=23194

Category: None
Group: Fixed in CVS/Need WG Approval
>Status: Closed
>Resolution: Accepted
Priority: 5
Submitted By: Magnus Westerlund (magwes)
>Assigned to: Magnus Westerlund (magwes)
Summary: RTSP extensions and IANA

Initial Comment:
Today the RFC 2326 talks about extensions to RTSP and also has a number of headers that uses 
parameters. In the next version the following clarifications should be done:
- Setup a IANA register for RTSP parameters.
- Rules how to register different types of parameters.
- What type of documents that are needed for extending RTSP. IANA registering of parameters.  
IETF RFC for new headers(?), methods and future protocol versions.
 

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

>Comment By: Magnus Westerlund (magwes)
Date: 2003-10-23 16:38

Message:
Logged In: YES 
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The IANA section is clearly needed. As it is has existed for
quite a while this bug will be closed. 

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

Comment By: Magnus Westerlund (magwes)
Date: 2002-10-30 18:17

Message:
Logged In: YES 
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The IANA section has been exteneded to include what is
currently all found values that need regestries. The rules
and there levels needs further input. 

All x- things has been removed due to there evilness.  

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

Comment By: Magnus Westerlund (magwes)
Date: 2002-05-27 16:52

Message:
Logged In: YES 
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Most of the issues describe in this bug are now adressed in
the new CVS repository file:
http://cvs.sourceforge.net/cgi-bin/viewcvs.cgi/rtspspec/iana.tex

Some further work regarding parameters are also required.

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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=513753&group_id=23194


--__--__--

Message: 3
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Thu, 23 Oct 2003 07:49:06 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-640272 ] Default way of deriving aggregate URL

Bugs item #640272, was opened at 2002-11-18 21:19
Message generated for change (Comment added) made by magwes
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=640272&group_id=23194

Category: General
>Group: Fixed in CVS/Need WG Approval
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
>Assigned to: Magnus Westerlund (magwes)
Summary: Default way of deriving aggregate URL

Initial Comment:
Currently there is very little said on which aggregated
control URL to use. The only given case is the
a=control attribute in SDP at session level. In cases
when one aggregates media stream that have no common
SDP declaration or dependencies in other ways the URL
can't be derived. Examples of such cases is when you
would like to aggregate a session multiplexed
retransmission stream with the original data stream,
without requiring aggregatation between all media types
part of a stream. Another example is aggregation of
streams that are lacking a common SDP description but
still can be intressting to aggregate. 

To fix this a way to derive what a valid aggregate URL
is needs to be specified. 

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

>Comment By: Magnus Westerlund (magwes)
Date: 2003-10-23 16:49

Message:
Logged In: YES 
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Currently I have written in the specification text that a
client not having received a aggregated control URI SHOULD
consider it as no aggregation available. However it does not
rule out that one tries to create aggregated control. The
rules for how to create the URI is present. 

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

Comment By: Magnus Westerlund (magwes)
Date: 2003-09-22 17:40

Message:
Logged In: YES 
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In the IETF in Vienna discussion July 16th 2003:

Issue: Missing session level control URI in SDP. IF the
session control URI is missing, should we allow the client
to try to to aggregate these? If aggregated which session
control URI must be used.

Discussion: This needs to be resolved. Tom Marshall did not
thought that we should prevent the use case to aggregate
unless necessary for interoperability. One possible
resolution is to say that any media description missing
session level control URI must be setup with individual RTSP
session for each media stream and any synchronization
between media be done on the receiver side. A practical
resolution resolution could be: A client SHOULD follow the
above statement but MAY try to aggregate. The server MAY
allow this aggregation but MAY also refuse it using 4xx
(Only non aggregated operation allowed).

Resolution?: Session declarations lacking a session level
control URL, MUST be used in non-aggregated control.
Further discussion needed? 


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

Comment By: Rob Lanphier (robla)
Date: 2003-05-28 18:34

Message:
Logged In: YES 
user_id=3796

From 14 May 2003 telecon
(http://rtsp.org/2003/telecon/minutes14May.html):
# Magnus: this is problem when aggregate control done in not
obvious way
# Magnus: what happens when you try to aggregate things that
aren't designed to be aggregated
# Joerg: I think this would require a lot of work to make
this work correctly
# Magnus: solution is to solve this with non-aggregate control
# Tom: how does client know
# Magnus: reuse session id
# Tom: what happens when you pass the session id with a
different URL
# Rob: it should be undefined
# Magnus: it should return error
# Rob: that's no more backward compatible
# Magnus: do you have proposed wording 11.4 second paragraph
# RESOLUTION: Tom to write up wording for 11.4 second paragraph

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

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https://sourceforge.net/tracker/?func=detail&atid=377744&aid=640272&group_id=23194


--__--__--

Message: 4
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Thu, 23 Oct 2003 07:29:40 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-493265 ] Pause - Play behaviour clarification

Bugs item #493265, was opened at 2001-12-14 12:55
Message generated for change (Comment added) made by magwes
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=493265&group_id=23194

Category: None
>Group: Fixed in CVS/Need WG Approval
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
>Assigned to: Magnus Westerlund (magwes)
Summary: Pause - Play behaviour clarification

Initial Comment:
The scenario is the following:
A media stream is ready to start playing when the client does the following A-C

A: PLAY range npt:10-30
B: @npt:22-> PAUSE
C: PLAY (no range)

This will result in resuming playing at npt:22. 
The unclearity is which of the following behaviour is the correct one?
1. Play until 30, i.e the second play is equal to PLAY range:22-30.
2. Play until end of media, i.e. the second play is equal to PLAY range:22-

The RFC states the following in Section 10.6 third paragraph:
  "A PAUSE request discards all queued PLAY requests. However, the pause
   point in the media stream MUST be maintained. A subsequent PLAY
   request without Range header resumes from the pause point."

The main unclarity is in what resumes means when also taking the first sentence into account.

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

>Comment By: Magnus Westerlund (magwes)
Date: 2003-10-23 16:29

Message:
Logged In: YES 
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This comment has been addressed by the general clarification
of PLAY and PAUSE handling plus some furhter
clarificiations. The specification text indicates that
unless the second play request specify a range the server
will play from the pause point of 22 second until the end of
media. 



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

Comment By: Rob Lanphier (robla)
Date: 2001-12-17 21:32

Message:
Logged In: YES 
user_id=3796

I would be inclined toward the 22-end interpretation (your option 2).  Won't change the spec until it's been 
covered by the group, though.

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

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--__--__--

Message: 5
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Thu, 23 Oct 2003 06:10:28 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-531189 ] How does REDIRECT method work

Bugs item #531189, was opened at 2002-03-18 05:14
Message generated for change (Comment added) made by magwes
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=531189&group_id=23194

Category: None
>Group: Fixed in CVS/Need WG Approval
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Magnus Westerlund (magwes)
Summary: How does REDIRECT method work

Initial Comment:
In the spec there is conflicting wording on what will happend when the server sends a REDIRECT to 
the client. In the state machine part the RFC says that the server should go to init state. Section 
10.10 REDIRECT says that the client must do TEARDOWN on the session. 

How is it supposed to work espically combined with RANGE needs to be clarified

also see #513891



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

>Comment By: Magnus Westerlund (magwes)
Date: 2003-10-23 15:10

Message:
Logged In: YES 
user_id=302620

At the telecons the 1 nov and 14th of nov 2003 the issue of
implicit session removal was discussed. It was agreed that
having it to remove the state was okay. However the handling
of the signalling transport connection was more of an issue. 

A specification text proposal is now edited in. Please
review to see if it is found acceptable.

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

Comment By: Magnus Westerlund (magwes)
Date: 2003-09-22 17:51

Message:
Logged In: YES 
user_id=302620

From the 57th IETF meeting discussion 16th of July 2003: 

Issue: Should REDIRECT automatically close session.

Discussion: This does not concern 3rr responses. As a
connection related REDIRECT will terminate the session on
the server side it is reasonable the same thing happens for
REDIRECT request directed at the session. The exception to
this is timed REDIRECTS that will continue exist until the
specified time. The client should do TEARDOWN prior to this
point to move the session to the alternative session.

Resolution: Clarify that session will be terminated by a
successful REDIRECT, independently of its connection or
session. 

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

Comment By: Magnus Westerlund (magwes)
Date: 2003-06-25 21:11

Message:
Logged In: YES 
user_id=302620

There has been updated text on the use of REDIRECT. The
usage of Range has also been clarified and solution for all
use cases exist. 

Two issue remains regarding redirect.
1. Should acknowleging a REDIRECT request mean that the
session is closed? Currently there exist on special case
regarding the use of * in the request that indicates that
transport link shall be closed, the session should be closed
automatically. However for the other cases when sessions is
targeted directly they are supposed to send a TEARDOWN for
that session. 

2. How should the client handle any outstanding request when
it receives a REDIRECT request with *?

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

Comment By: Magnus Westerlund (magwes)
Date: 2002-09-18 17:12

Message:
Logged In: YES 
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The proposed solution for ranges does not work for content
that uses UTC for positioning, like surveilance tapes.
Suggestions on how to solve this, please.

In case that the location header only contains a server
adress the client may assume that the media is identical and
does not need to send a DESCRIBE before setting up a new
session. Otherwise it needs to do it as media may be different.




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

Comment By: Magnus Westerlund (magwes)
Date: 2002-09-12 15:12

Message:
Logged In: YES 
user_id=302620

One more problem is that a client does not know if it should
issue a play with the range where it was redirected or start
from the begining. In the case of a media on deman and it
gets identical media from the new server it seems more valid
to resume at the position it was redirected at. However if
it gets redirected to another media it should start from the
beginng. And in the case of live media it should simply not
use any range in its PLAY request.

A mechanism needs to be added that solves this problem.

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

Comment By: Magnus Westerlund (magwes)
Date: 2002-09-12 15:08

Message:
Logged In: YES 
user_id=302620

The result of a MMUSIC discussion is:

3xx Responses:
The server should use this to signal the client that it
should go find the media somewhere else. This answer can be
used on the methods that do inquires regarding the media or
tries to establish a session, i.e. OPTIONS (with media URL),
DESCRIBE, SETUP, and perhaps ANNOUNCE. It should not be used
when a server has an already established session and needs
to move that session to another server, i.e. don't use them
on PAUSE, PLAY, RECORD.

The client after receiving a REDIRECT must restart its work
on getting session information and setting up the session.

REDIRECT request:
This method shall be used when the server needs to move
already established session from this server. So it shall
not be used unless there is a session identifier to
reference. The redirect request SHOULD use contain the
session control URL and MUST include a session header. This
enables the client a unambiguous identification of which
session it should move.

I don't know if the server should go directly to INIT after
receiving the response from the client on the REDIRECT. This
would allow the server to reclaim any resources as fast as
possible. Alternatively the server should wait for a
teardown, but this seem to waste a round trip of time. But
as both behaviors are hinted in RFC 2326 it needs clarification.

When a client has received a REDIRECT request and
acknowledge the redirect it must assume that the session
does no longer exist. The exception is if a range header was
part of the request, and in that case it will be without a
session at the specified moment. So after receiving a
redirect the client must start from scratch to establish a
new session at the given location.

Use of location header: My opinion is that is should be a
complete RTSP URL.

Use of Range header: Also how the Range header shall be
interpreted together with REDIRECT needs to be clarified. As
the desired behavior may vary I think we need to be a little
bit cleaver here. If a range header is included and contains
a time in UTC it shall be interpreted that the server will
serve this client until that UTC time, and after that time
the client needs to establish a new session at the given
location. If a SMPTE or NPT time is given the session will
end when the playback reaches the given point in the media.
So in the case the client rewinds to the start before
reaching specified media point the session will remain. By
using UTC time the server can signal that I can't handle
your request further than e.g. until 18.00 today. Using NPT
will allow a server to redirect a user due to limitations of
the media, e.g. charging for rest of movie.




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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=531189&group_id=23194


--__--__--

Message: 6
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Thu, 23 Oct 2003 07:33:16 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-500868 ] Require: option-tag values unclear

Bugs item #500868, was opened at 2002-01-08 17:09
Message generated for change (Comment added) made by magwes
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=500868&group_id=23194

Category: None
>Group: Resolved - Added to text
Status: Open
Resolution: None
Priority: 5
Submitted By: Sean Sheedy (seansh)
>Assigned to: Magnus Westerlund (magwes)
Summary: Require: option-tag values unclear

Initial Comment:
The proper option-tag values to use in the Require:
header when checking for optional features are not
always clear.  Some are more easily guessed then
others.

For example, to ask whether a server supports the
Scale: header, I would guess that the proper option-tag
would be "scale".  However, how does a client ask
whether the server supports the smpte time format in
the Range: header?

The document should include a table of optional
features and their corresponding option-tag values.

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

>Comment By: Magnus Westerlund (magwes)
Date: 2003-10-23 16:33

Message:
Logged In: YES 
user_id=302620

The feature tags will be defined in the necessary
specifications. All official tags will be stored in an IANA
managed repository according the rules available in the IANA
chapter of the core specification. 

The SMPTE range header question will be solved thorugh the
use of Accept-Ranges instead of feature tags.


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

Comment By: Rob Lanphier (robla)
Date: 2002-01-29 04:26

Message:
Logged In: YES 
user_id=3796

This should probably be a separately defined document.  The Require header was intended as a way of 
querying for features outside of the scope of RTSP, rather than for features in RTSP.

It is too late to require that existing implementations support new feature tags, but I agree it would be 
useful thing to have around.  A candidate list may fall out of the list of features in RTSP.  Sean, would you 
mind taking a first crack at creating the list of features tags you'd like to see?


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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=500868&group_id=23194


--__--__--

Message: 7
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Thu, 23 Oct 2003 07:37:06 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-500906 ] Client liveness criteria vague

Bugs item #500906, was opened at 2002-01-08 18:14
Message generated for change (Comment added) made by magwes
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=500906&group_id=23194

Category: None
Group: Fixed in CVS/Need WG Approval
Status: Open
>Resolution: Accepted
Priority: 5
Submitted By: Sean Sheedy (seansh)
>Assigned to: Magnus Westerlund (magwes)
Summary: Client liveness criteria vague

Initial Comment:
The current standard is vague about what constitutes a
client "liveness" indication sufficient to prevent the
media server from freeing a client's session.  It also
does not address under what circumstances a media
server may close an inactive persistent control
connection from a client.

These areas should be more explicitly specified so that
clients can reliably maintain their sessions with
minimum liveness detection overhead.

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

>Comment By: Magnus Westerlund (magwes)
Date: 2003-10-23 16:37

Message:
Logged In: YES 
user_id=302620

As no comments on this bug has been raised in a year it will
be closed. 

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

Comment By: Magnus Westerlund (magwes)
Date: 2002-10-30 14:39

Message:
Logged In: YES 
user_id=302620

The criteria for closing connections has been clarified in
the chapter connections.  

The liveness signaling has also been clarified in the
session header chapter. 

Changes are checked in.

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

Comment By: Magnus Westerlund (magwes)
Date: 2002-01-14 11:13

Message:
Logged In: YES 
user_id=302620

Primary source of livness information is RTCP, which should be clarified. 
Secondly the new method PING could be used, see bug report #477400. 


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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=500906&group_id=23194




End of Rtspspec-bugs Digest


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Thread-Topic: [ rtspspec-Bugs-810678 ] How to indicate end of media stream
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> -----Original Message-----
> From: rtspspec-bugs-request@lists.sourceforge.net=20
> [mailto:rtspspec-bugs-request@lists.sourceforge.net]=20
> Sent: Thursday, October 23, 2003 11:22 PM
>=20
> Issue:  Possible methods for signalling end of
> stream/media (not session) in RTP.
>=20
> Discussion: Need to signal that a stream/media has
> ended exist mostly in unbound sessions. Proposal
> brought up to indicate this is RTCP BYE when the sender
> stops sending and has waited some time it can remove
> the SSRC of that source and send the BYE to indicate
> this. Another proposal was to send this information in
> RTSP methods. One could use sever to client TEARDOWN to=20
> indicate this session really are over, nothing more can be=20
> done. PAUSE can be sent when the media has been served but=20
> the client may rewind do other operations. The TEARDOWN idea=20
> was not liked as certain applications require session data to=20
> be sent from client to server using SET_PARAMETER. A new=20
> method should be evaluated. There might also exist=20
> significant synchronization problems in this.
>=20
> Resolution: Propose a RTSP extension with a new method.
> Thomas Zeng to write strawman proposal for evaluation
> to mailing list.


We do this already in TrueCast using a special RTP header extension
word.

The RTSP spec says that when the play request has completed, you're
supposed to automatically transition to the paused state, per RFC 2326
10.5 (p. 35).  However, this doesn't really work, because it's
impossible for a client to know which frame corresponds to the endpoint
of the Range header of the PLAY response.

In other words, if the client sees a delay in the receipt of a frame, it
can't know whether this is because of a server issue, a network issue,
or just that there are no more frames coming.

So what we do instead is to send a special end-of-stream RTP packet,
containing a single header extension word, once we've sent all the
frames in the range specified in the original PLAY request.  We remain
in the PLAY state, and continue to send the special packet every 100ms,
until the client either explicitly issues a PAUSE request, or sends a
TEARDOWN.


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Cc: mmusic@ietf.org, zeng@PacketVideo.COM, magnus.westerlund@ericsson.com
Subject: Re: [MMUSIC] [ rtspspec-Bugs-810678 ] How to indicate end of media stream
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> > Discussion: Need to signal that a stream/media has
> > ended exist mostly in unbound sessions. Proposal
> > brought up to indicate this is RTCP BYE when the sender
> > stops sending and has waited some time it can remove
> > the SSRC of that source and send the BYE to indicate
> > this. Another proposal was to send this information in
> > RTSP methods. One could use sever to client TEARDOWN to=20
> > indicate this session really are over, nothing more can be=20
> > done. PAUSE can be sent when the media has been served but=20
> > the client may rewind do other operations. The TEARDOWN idea=20
> > was not liked as certain applications require session data to=20
> > be sent from client to server using SET_PARAMETER. A new=20
> > method should be evaluated. There might also exist=20
> > significant synchronization problems in this.
> >=20
> > Resolution: Propose a RTSP extension with a new method.
> > Thomas Zeng to write strawman proposal for evaluation
> > to mailing list.
>=20
>=20
> We do this already in TrueCast using a special RTP header extension
> word.
>=20
> The RTSP spec says that when the play request has completed, you're
> supposed to automatically transition to the paused state, per RFC 2326
> 10.5 (p. 35).  However, this doesn't really work, because it's
> impossible for a client to know which frame corresponds to the endpoint
> of the Range header of the PLAY response.
>=20
> In other words, if the client sees a delay in the receipt of a frame, it
> can't know whether this is because of a server issue, a network issue,
> or just that there are no more frames coming.
>=20
> So what we do instead is to send a special end-of-stream RTP packet,
> containing a single header extension word, once we've sent all the
> frames in the range specified in the original PLAY request.  We remain
> in the PLAY state, and continue to send the special packet every 100ms,
> until the client either explicitly issues a PAUSE request, or sends a
> TEARDOWN.

We also have an end-of-stream packet for our RDT transport.  I'm not sure
what we do with RTP but I can check if you're interested.

It seems logical to me that this be communicated over the transport channel,
not the control channel.  A simple RTP extension like you mention would make
a great proposal.  Obviously not everyone here shares that view though...

--=20
Research is what I'm doing when I don't know what I'm doing.
        -- Wernher von Braun

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Subject: RE: [MMUSIC] [ rtspspec-Bugs-810678 ] How to indicate end of media stream
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Thread-Topic: [MMUSIC] [ rtspspec-Bugs-810678 ] How to indicate end of media stream
Thread-Index: AcOaaOiMG10GnAjxQB+mDun5UK/RDQAAP8MQ
From: "Matthew Heaney" <mheaney@on2.com>
To: "Tom Marshall" <tmarshall@real.com>
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        <magnus.westerlund@ericsson.com>
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> -----Original Message-----
> From: Tom Marshall [mailto:tmarshall@real.com]=20
> Sent: Friday, October 24, 2003 3:55 PM
>=20
> We also have an end-of-stream packet for our RDT transport. =20
> I'm not sure
> what we do with RTP but I can check if you're interested.
>=20
> It seems logical to me that this be communicated over the=20
> transport channel,
> not the control channel.  A simple RTP extension like you=20
> mention would make
> a great proposal.  Obviously not everyone here shares that=20
> view though...

It's not as simple as sending a normal RTCP packet with no other info
(say, a BYE msg), because UDP packets can arrive in any order.  Nor is
it clear how the RTP and RTCP streams would even be synchronized in UDP.
Even in TCP it would be kind of a pain.

It might be possible to use an RTCP packet (with an APP command maybe?)
to specify the sequence number and/or RTP timestamp of the last frame.
But this might get hairy if the last frame can be consumed prior to
receipt of the RTCP indicator.

So I'm in agreement with you that using RTP to indicate EOS is probably
simpler than the RTCP alternative.

Note that this is typically how things are done anyway, ie.

void f(const List& list)
{
   ListNode* node =3D list.m_head;

   while (node !=3D 0)
   {
     // do something with node->m_item
    =20
     node =3D node.m_next;
   }
}

Or at a higher level of abstraction:

void g(const std::list<int>& list)
{
   typedef std::list<int>::const_iterator iter_t;

   iter_t i =3D list.begin();
   const iter_t j =3D list.end();

   while (i !=3D j)=20
   {
      // do something with *i

      ++i;
   }
}


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Subject: RE: [MMUSIC] [ rtspspec-Bugs-810678 ] How to indicate end of medi
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my comments embedded.

-----Original Message-----
From: Tom Marshall [mailto:tmarshall@real.com]
Sent: Friday, October 24, 2003 12:55 PM
To: Matthew Heaney
Cc: mmusic@ietf.org; zeng@PacketVideo.COM;
magnus.westerlund@ericsson.com
Subject: Re: [MMUSIC] [ rtspspec-Bugs-810678 ] How to indicate end of
media stream


> > Discussion: Need to signal that a stream/media has
> > ended exist mostly in unbound sessions. Proposal
> > brought up to indicate this is RTCP BYE when the sender
> > stops sending and has waited some time it can remove
> > the SSRC of that source and send the BYE to indicate
> > this. Another proposal was to send this information in
> > RTSP methods. One could use sever to client TEARDOWN to 
> > indicate this session really are over, nothing more can be 
> > done. PAUSE can be sent when the media has been served but 
> > the client may rewind do other operations. The TEARDOWN idea 
> > was not liked as certain applications require session data to 
> > be sent from client to server using SET_PARAMETER. A new 
> > method should be evaluated. There might also exist 
> > significant synchronization problems in this.
> > 
> > Resolution: Propose a RTSP extension with a new method.
> > Thomas Zeng to write strawman proposal for evaluation
> > to mailing list.
> 
> 
> We do this already in TrueCast using a special RTP header extension
> word.
> 
> The RTSP spec says that when the play request has completed, you're
> supposed to automatically transition to the paused state, per RFC 2326
> 10.5 (p. 35).  However, this doesn't really work, because it's
> impossible for a client to know which frame corresponds to the endpoint
> of the Range header of the PLAY response.
> 
> In other words, if the client sees a delay in the receipt of a frame, it
> can't know whether this is because of a server issue, a network issue,
> or just that there are no more frames coming.
> 
> So what we do instead is to send a special end-of-stream RTP packet,
> containing a single header extension word, once we've sent all the
> frames in the range specified in the original PLAY request.  We remain
> in the PLAY state, and continue to send the special packet every 100ms,
> until the client either explicitly issues a PAUSE request, or sends a
> TEARDOWN.

[tz] Just curious: why not send BYE every 100 ms instead of a special
packet? Is it because
     RTCP channel may not be allocated?

We also have an end-of-stream packet for our RDT transport.  I'm not sure
what we do with RTP but I can check if you're interested.

It seems logical to me that this be communicated over the transport channel,
not the control channel.  A simple RTP extension like you mention would make
a great proposal.  Obviously not everyone here shares that view though...
[tz] A special RTP packet would work, provided re-transmission mechanism is 
     in place for RTP over UDP. It is also required for each track. For an 
     RTSP session with many tracks, it is not elegant not being able to 
     signal end-of-stream for the aggregate session.

     Also if the transport is not RTP, what do we do?  I don't think 
     RTSP is tied to using RTP as transport, for example REAL uses RDT as 
     transport.  
     So if RTSP does not provide end-of-stream signalling, every potential
     transport would have to reinvent the wheel.

     Therefore I still think that although both RTSP end-of-stream
signalling and special
     RTP extension can solve the problem, RTSP end-of-stream is more
succinct and more
     "re-usable".  The drawback, of course, is that RTSP end-of-stream
requires persistent
     RTSP connection.
-- 
Research is what I'm doing when I don't know what I'm doing.
        -- Wernher von Braun

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Subject: Re: [MMUSIC] [ rtspspec-Bugs-810678 ] How to indicate end of media
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The RealServer sends it's own end of stream APP packet over the RTCP 
channel. This packet contains the sequence number and timestamp of the 
last packet in the stream. The sequence number allows the client to know 
when it has received the last packet of the stream. This can help it 
avoid shutting down the stream early if the EOS packet happens to arrive 
before the last packet in the stream. This is not a common case, but it 
could happen.

I think the RealServer only sends five or less of the end of stream 
packets. I like the idea of periodically sending them until a PAUSE or 
TEARDOWN occurs. That seems like a better way to insure the client gets 
the packet.

Should this EOS packet be on the RTP or RTCP channel? Currently the 
RealServer uses the RTCP channel because it had a mechanism to allow 
extensions that would be ignored by clients that don't understand the 
packet. I don't think there a mechanism for putting this type of packet on 
the RTP channel.

Aaron

On Fri, 24 Oct 2003, Tom Marshall wrote:

> > > Discussion: Need to signal that a stream/media has
> > > ended exist mostly in unbound sessions. Proposal
> > > brought up to indicate this is RTCP BYE when the sender
> > > stops sending and has waited some time it can remove
> > > the SSRC of that source and send the BYE to indicate
> > > this. Another proposal was to send this information in
> > > RTSP methods. One could use sever to client TEARDOWN to 
> > > indicate this session really are over, nothing more can be 
> > > done. PAUSE can be sent when the media has been served but 
> > > the client may rewind do other operations. The TEARDOWN idea 
> > > was not liked as certain applications require session data to 
> > > be sent from client to server using SET_PARAMETER. A new 
> > > method should be evaluated. There might also exist 
> > > significant synchronization problems in this.
> > > 
> > > Resolution: Propose a RTSP extension with a new method.
> > > Thomas Zeng to write strawman proposal for evaluation
> > > to mailing list.
> > 
> > 
> > We do this already in TrueCast using a special RTP header extension
> > word.
> > 
> > The RTSP spec says that when the play request has completed, you're
> > supposed to automatically transition to the paused state, per RFC 2326
> > 10.5 (p. 35).  However, this doesn't really work, because it's
> > impossible for a client to know which frame corresponds to the endpoint
> > of the Range header of the PLAY response.
> > 
> > In other words, if the client sees a delay in the receipt of a frame, it
> > can't know whether this is because of a server issue, a network issue,
> > or just that there are no more frames coming.
> > 
> > So what we do instead is to send a special end-of-stream RTP packet,
> > containing a single header extension word, once we've sent all the
> > frames in the range specified in the original PLAY request.  We remain
> > in the PLAY state, and continue to send the special packet every 100ms,
> > until the client either explicitly issues a PAUSE request, or sends a
> > TEARDOWN.
> 
> We also have an end-of-stream packet for our RDT transport.  I'm not sure
> what we do with RTP but I can check if you're interested.
> 
> It seems logical to me that this be communicated over the transport channel,
> not the control channel.  A simple RTP extension like you mention would make
> a great proposal.  Obviously not everyone here shares that view though...
> 
> 


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Date: Fri, 24 Oct 2003 14:03:05 -0700
From: Tom Marshall <tmarshall@real.com>
To: Thomas Zeng <zeng@PacketVideo.COM>
Cc: Matthew Heaney <mheaney@on2.com>, mmusic@ietf.org,
        magnus.westerlund@ericsson.com
Subject: Re: [MMUSIC] [ rtspspec-Bugs-810678 ] How to indicate end of medi a stream
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> > So what we do instead is to send a special end-of-stream RTP packet,
> > containing a single header extension word, once we've sent all the
> > frames in the range specified in the original PLAY request.  We remain
> > in the PLAY state, and continue to send the special packet every 100ms,
> > until the client either explicitly issues a PAUSE request, or sends a
> > TEARDOWN.
>=20
> [tz] Just curious: why not send BYE every 100 ms instead of a special
> packet? Is it because
>      RTCP channel may not be allocated?

My understanding is that if you send BYE, the RTP session is closed.  The
player should be allowed to continue using the session with another PLAY
request.

> We also have an end-of-stream packet for our RDT transport.  I'm not sure
> what we do with RTP but I can check if you're interested.
>=20
> It seems logical to me that this be communicated over the transport chann=
el,
> not the control channel.  A simple RTP extension like you mention would m=
ake
> a great proposal.  Obviously not everyone here shares that view though...
> [tz] A special RTP packet would work, provided re-transmission mechanism =
is=20
>      in place for RTP over UDP. It is also required for each track. For a=
n=20
>      RTSP session with many tracks, it is not elegant not being able to=
=20
>      signal end-of-stream for the aggregate session.

Why is this not elegant?  It keeps transport signalling in the transport and
control signalling in the control channel.  I think mixing them is less
elegant.

In my experience, the tracks don't always end at exactly the same time
anyway (though they are usually well within one second of each other).

>      Also if the transport is not RTP, what do we do?  I don't think=20
>      RTSP is tied to using RTP as transport, for example REAL uses RDT as=
=20
>      transport. =20
>      So if RTSP does not provide end-of-stream signalling, every potential
>      transport would have to reinvent the wheel.

That should be the least of the transport inventors' worries.  It just
slipped by in RTP because RTP was invented primarily for multicast live and
predates RTSP by a couple of years.

>      Therefore I still think that although both RTSP end-of-stream
> signalling and special
>      RTP extension can solve the problem, RTSP end-of-stream is more
> succinct and more
>      "re-usable".  The drawback, of course, is that RTSP end-of-stream
> requires persistent
>      RTSP connection.

I thought the current spec was steering us away from persistent connections=
?=20
Requiring a persistent connection to properly receive an end-of-stream
notification seems to be a step backward.

--=20
To use violence is to already be defeated.
        -- Chinese proverb

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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-522602 ] Should Allow S->C  "Teardown" (SourceForge.net)
   2. [ rtspspec-Bugs-829528 ] SDP appendix and RFC 3388 (SourceForge.net)
   3. [ rtspspec-Bugs-539001 ] Transport header should override m= (SourceForge.net)
   4. [ rtspspec-Bugs-500875 ] State diagram clarifications (SourceForge.net)
   5. [ rtspspec-Bugs-556295 ] Client issues PAUSE while server ready (SourceForge.net)
   6. [ rtspspec-Bugs-662436 ] RTP/AVP/TCP w/o interleaved (SourceForge.net)
   7. [ rtspspec-Bugs-625352 ] Synchronising multiple SSRCs (SourceForge.net)

--__--__--

Message: 1
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Thu, 23 Oct 2003 07:41:57 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-522602 ] Should Allow S->C  "Teardown"

Bugs item #522602, was opened at 2002-02-25 20:38
Message generated for change (Comment added) made by magwes
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=522602&group_id=23194

Category: General
Group: Fix for DS - Need WG Input
>Status: Closed
Resolution: None
Priority: 5
Submitted By: Thomas M Zeng (zengtm)
>Assigned to: Magnus Westerlund (magwes)
Summary: Should Allow S->C  "Teardown" 

Initial Comment:
Table 2 in section 10 of RFC2326 seem to disallow S->C 
Teardown. And section 10.7 is vague on whether S->C 
teardown is OK.

This should be relaxed, because Server might have to 
use TEARDOWN to tell client a live session is over.

 

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

>Comment By: Magnus Westerlund (magwes)
Date: 2003-10-23 16:41

Message:
Logged In: YES 
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As it is being take care of and new tracking bug 810678 will
be used for this issues instead. 

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

Comment By: Thomas M Zeng (zengtm)
Date: 2003-09-30 06:10

Message:
Logged In: YES 
user_id=20397

Since during Vienna IETF, we have decided to add 
END_OF_STREAM, Teardown from S->C is not necessary.
We can close this issue.

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

Comment By: Magnus Westerlund (magwes)
Date: 2002-10-30 18:52

Message:
Logged In: YES 
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To be discussed

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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=522602&group_id=23194


--__--__--

Message: 2
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Fri, 24 Oct 2003 04:38:46 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-829528 ] SDP appendix and RFC 3388

Bugs item #829528, was opened at 2003-10-24 13:38
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=829528&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: SDP appendix and RFC 3388

Initial Comment:
Should we add text in the SDP chapter on how to handle
the Grouping of media lines for SDP, RFC 3388? 



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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=829528&group_id=23194


--__--__--

Message: 3
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Fri, 24 Oct 2003 05:20:08 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-539001 ] Transport header should override m=

Bugs item #539001, was opened at 2002-04-04 01:32
Message generated for change (Comment added) made by magwes
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=539001&group_id=23194

Category: General
>Group: Fixed in CVS/Need WG Approval
Status: Open
Resolution: None
Priority: 5
Submitted By: Rob Lanphier (robla)
>Assigned to: Magnus Westerlund (magwes)
Summary: Transport header should override m=

Initial Comment:
Section C.1.7 ("Connection Information") states the
following:  "for on-demand unicast streams and some
multicast streams, the destination is specified by the
client via the SETUP request".  The rest of this
section mainly refers to the "c=" line in SDP, but it's
not clear that the SETUP Transport header also
overrides the port given in the m= line.  Clarification
of this is needed.

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

>Comment By: Magnus Westerlund (magwes)
Date: 2003-10-24 14:20

Message:
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This has been clarified in the SDP appendix. There has been
alignment between what is possible between ports and addresses. 

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

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--__--__--

Message: 4
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Thu, 23 Oct 2003 07:35:20 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-500875 ] State diagram clarifications

Bugs item #500875, was opened at 2002-01-08 17:17
Message generated for change (Comment added) made by magwes
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=500875&group_id=23194

Category: None
Group: Fixed in CVS/Need WG Approval
>Status: Closed
Resolution: Fixed
Priority: 5
Submitted By: Sean Sheedy (seansh)
>Assigned to: Magnus Westerlund (magwes)
Summary: State diagram clarifications

Initial Comment:
My understanding is that the state diagram in Appendix
A is per-session, not per-client.  However, this is not
explicitly stated.  This should be made clear in the
description.

Errors within the media server (series 5xx) often leave
a stream in an unplayable, unrecoverable state in which
the client's only option is to tear down the session
and start over.  This should be reflected in the state
diagram.

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

>Comment By: Magnus Westerlund (magwes)
Date: 2003-10-23 16:35

Message:
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I hope that the resolution in the state machine chapter is
satisfactory. AS no comments on the resolution has been
given in the last year I hereby close the bug. 

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

Comment By: Magnus Westerlund (magwes)
Date: 2002-10-30 13:30

Message:
Logged In: YES 
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The state machine has been updated and clearly represent a
RTSP session. The effect 5xx error codes may have has also
been clarified. A 5xx response is in general recoverable
from. But the server is allowed to terminate the session for
errors of unrecoerable nature. The client will detect this
when issuing another request to that session and receive a
454 (Session Not Found) as response. 



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

Comment By: Magnus Westerlund (magwes)
Date: 2002-01-14 11:08

Message:
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This one is realted to bug report #500803

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

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--__--__--

Message: 5
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Fri, 24 Oct 2003 04:51:03 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-556295 ] Client issues PAUSE while server ready

Bugs item #556295, was opened at 2002-05-15 14:12
Message generated for change (Comment added) made by magwes
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=556295&group_id=23194

Category: General
>Group: Fixed in CVS/Need WG Approval
Status: Open
Resolution: None
Priority: 5
Submitted By: Nobody/Anonymous (nobody)
>Assigned to: Magnus Westerlund (magwes)
Summary: Client issues PAUSE while server ready

Initial Comment:
If a client issues a PAUSE and the server acknowledges
and enters ready state, what is proper server response
if the player issues another PAUSE?

There appear to be two possible correct behaviors:

1. server responds with OK because it does not change
state and does no harm.

2. server responds with error 455 because PAUSE does
not make sense in this state.

Both interpretations have been implemented.  The latter
appears to be more correct, but a client that assumes
the first may misbehave if the server implements the
second.

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

>Comment By: Magnus Westerlund (magwes)
Date: 2003-10-24 13:51

Message:
Logged In: YES 
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This clarification of behavior has now beend added to the
text and will be published in v5. The text is in the state
machine and pause method section.

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

Comment By: Magnus Westerlund (magwes)
Date: 2003-05-14 17:17

Message:
Logged In: YES 
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The working assumption is that a 200 OK should be sent even
if already in ready state. Also the pause point needs to be
included in the response. 

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

Comment By: Magnus Westerlund (magwes)
Date: 2002-10-30 19:16

Message:
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This should be clarified. Input from implementors wanted.
The resolution should be made evident in the state machine. 

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

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--__--__--

Message: 6
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Fri, 24 Oct 2003 09:14:10 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-662436 ] RTP/AVP/TCP w/o interleaved

Bugs item #662436, was opened at 2003-01-05 03:01
Message generated for change (Settings changed) made by magwes
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=662436&group_id=23194

>Category: Feature Request
>Group: Fix for DS - Need WG Input
Status: Open
Resolution: None
Priority: 5
Submitted By: Nobody/Anonymous (nobody)
Assigned to: Nobody/Anonymous (nobody)
Summary: RTP/AVP/TCP w/o interleaved

Initial Comment:

If a Transport line specifies RTP/AVP/TCP
and does not also use the interleaved
parameter, the transport setup details
seem undocumented (maybe I'm not looking
in the right place ...).

I'm thinking of the level of detail the
comedia I-D provides for TCP: which
side accept()s and which side connect()s,
how many connections to set up, framing, 
etc.

Are there pre-existing implementations 
that use this transport mode? I happened
to stumble onto this in writing the TCP
appendix for the MWPP "implementation
guidelines" ... which may spawn off a
generic I-D specifying how to use comedia
with RTP (framing, specifying the fmt
tokens after TCP, etc). So, one option
is to add RTP/AVP/TCP w/o interleaving
details to that I-D, if adding it to
to the RTSP I-D isn't desirable for
some reason ...


john lazzaro -- lazzaro@cs.berkeley.edu


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

Comment By: Rob Lanphier (robla)
Date: 2003-05-28 18:34

Message:
Logged In: YES 
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From 14 May 2003 telecon
(http://rtsp.org/2003/telecon/minutes14May.html):
#  Magnus: we should leave this until RTP framing is done
# Magnus: this is basically a feature request
# Magnus: this might well be a separate draft
# Rob: agreed
# Aravind: agreed
# Tom: we'll have to deal with this issue when DCCP comes along
# Anders: I saw this in use
# Tom: they might be using interleaving or not
# RESOLUTION: separate draft for non-interleaving; narrow
spec for      interleaving 

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

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--__--__--

Message: 7
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Fri, 24 Oct 2003 05:07:50 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-625352 ] Synchronising multiple SSRCs

Bugs item #625352, was opened at 2002-10-18 20:41
Message generated for change (Comment added) made by magwes
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=625352&group_id=23194

>Category: General
>Group: Fixed in CVS/Need WG Approval
Status: Open
Resolution: None
Priority: 5
Submitted By: Go Hori (ghori)
>Assigned to: Magnus Westerlund (magwes)
>Summary: Synchronising multiple SSRCs

Initial Comment:
taken from a thread in mmusic

We have a problem in RTSP when a media stream 
needs to contain multiple
RTP SSRCs. This can't be expressed today clearly.

When one is only interested in controlling the media and 
not each
individual source the following scenarios should be 
supported:
A SINGLE SETUP request can result in one of the 
following:
1. A media stream with a single SSRC.
2. A media stream with retransmission with SSRC 
multiplexing. This
requires a SSRC for the original and one for the 
retransmission packets.
It is an advantage if both can be explicitly signalled, but 
is no
requirement.
3. Live session with any number of participants and 
unlimited use of
dynamic chosen SSRCs.

Due to clients capability the different scenarios need to 
be explicitly
signaled. Scenario 2 will be that from an SDP 
description and support of
retransmission will be signaled by requesting AVPF in 
the transport
header. However today there exist no way to signal if it 
is scenario 1
or 3. So this needs to be resolved in a nice backwards 
compatible way. I
think that support for 3 can require support for any extra 
signaling
while scenario 1 must work without any extra parts.

By doing all these scenarios with a single SETUP per 
media stream we
keep the signaling to a minimum and avoid further 
requirements on SDP.
It might be advantageous in the future to create a SDP 
attribute
declaring the streams type.

If one desires to control the individual source one either 
declare
individual media streams for each source. This will 
require use of
ANNOUNCE in dynamic settings. The alternative would 
be to create a new
media resource control protocol which is suited for this. 
However I
don't see real need to solve this problem in standard 
RTSP better than
by using ANNOUNCE.


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

>Comment By: Magnus Westerlund (magwes)
Date: 2003-10-24 14:07

Message:
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This was discussed in the telecon the 28th April 2003.

The resolution was to clarify in RTSP that only a single
SSRC one can be given in RTSP. Any furhter SSRC may be
present in the stream, however they must be synchronised
using RTCP. 

This has been clarified in the specification. 

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

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To: dccp@ietf.org
cc: mmusic@ietf.org
Date: Fri, 24 Oct 2003 23:13:22 -0700
From: Eddie Kohler <kohler@icir.org>
Subject: [MMUSIC] DCCP, TFRC, and VoIP issue summary
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Hi all,

Several months ago, we had a pretty lively discussion on these lists about
why TFRC and/or DCCP might be inappropriate for VoIP and other apps. I
wanted to summarize again the problems that people have brought up, and
where we stand on solutions. Comments welcome.

A. Complexity

   The existing DCCP specification is simply too complex for resource-
   constrained implementations, which might be common for voice.

   * This complexity seems important and not onerous for host stacks. We
   hope that CCID 3-Thin shows it's possible to simplify DCCP's
   requirements for resource-constrained implementations without radical
   surgery on the base protocol.

B. Slow-start

   TCP-friendly congestion control algorithms must slow-start up to their
   sustained rate. Voice apps might like to start with their full bandwidth
   right away. Plus, the initial TFRC rate seems very slow (1 packet per
   RTT).

   * We have updated the CCID 3 draft to allow initial TFRC rates of up to
   4 packets per RTT (or 4380 payload bytes per RTT), in line with recent
   TCP practice (RFC 3390). For a This supports a 40ms _initial_
   packetization interval for any RTT up to 160ms: not bad at all.  And we
   might consider allowing a larger initial window when packet payloads are
   small; this would require a bit of research on whether many small
   packets is actually equivalent to few large packets in terms of
   congestion.

C. Rate slows down during idle periods

   The TFRC allowed sending rate will drop down to something very small
   (2-4 packets/RTT) when the sender is idle. This may cause problems
   particualrly with VoIP conversations, which are naturally idle-one-way
   then idle-the-other-way. The rampup from the small rate to the fair rate
   may take several to many RTTs, ouch.

   * The solution for slow start applies here. Beyond that, we recommend
   sending probe packets if any slowdown during an idle period is
   unacceptable; you don't get to reserve bandwidth in the best-effort
   internet. These probe packets could easily consist of required
   acknowledgements.

D. Rate will not increase during app-limited period

   If the application is sending at rate R, the allowed rate will not
   increase much beyond R (say, more than 2R) until the app "probes" by
   transmitting at a faster rate. The reason: DCCP determines the allowable
   maximum rate by monitoring packet drops. Thus, DCCP "can not tell the
   application when to increase its rate unless the application 'probes'
   the network first by ... increasing its rate."  (Phillipe Gentric)

   * A related issue, and again, we can't do much better in the current
   internet than having the application send probe packets: you don't get
   to reserve bandwidth you aren't using. Nevertheless, there shouldn't be
   much problem with the sender abruptly doubling its sending rate in this
   situation, assuming TFRC's observed loss event rate is low enough to
   allow it.

E. Congestion control means variable rate

   Must an application with discrete sending rates send at a rate strictly
   bounded by TFRC's allowed rate? Could we allow some flexibility, so that
   it could send at, say, any rate <= 1.8*TFRC_allowed_rate?

   * There probably wouldn't be any problem with this, but it should be
   explored in a separate document. Does anyone want to write up a draft
   about CCID 3 adapted for a sender with discrete sending rates?

F. Rate changes/oscillations

   There are application costs, possibly significant ones, associated with
   changing rates. Current codecs switch rates poorly; actual devices tend
   to swap in a new codec. Also, for audio at least, people find the
   glitches caused by frequent rate changes distracting. Thus, if the
   application has to change its rate frequently, because of CC, that's a
   problem.

   * A solution to issue D would apply here. Furthermore, we should not
   solve this problem generically; different applications will want
   different behavior. For example, Tom Phelan described a procedure where
   bandwidth is probed at the beginning of the connection, then a codec
   starts at a lower rate than the probe rate; if congestion causes the
   rate to drop below this, the connection terminates.

G. Multiple streams

   An application that had both audio and video to send might use 2 DCCP
   connections. Could the application choose how its bandwidth is allocated
   among those connections?  (Phillipe Gentric)

   * We haven't made progress on this. It's at least somewhat problematic
   since the end nodes can't necessarily assume that the two connections
   take the same path, with the same quality of service on all parts of the
   path.

Here are some non-issues: things people have brought up that I think the
DCCP spec already handles well.

H. Low delay

   It's very important to minimize packetization overhead and effects of
   the send buffer.

   * An API issue, and DCCP was designed with this in mind.

I. Control over the send buffer

   What happens when packets need to be dropped (because TFRC demands a
   lower rate than the application wants)? Some would like packets to be
   dropped by the TFRC stack, either by dropping the first packet, or the
   last packet, or the lowest-priority packet; others would like the
   application to be in direct control.

   * This is an API issue, and DCCP seems flexible enough to handle any of
   those APIs, or others.

J. Pushing against the maximum rate (streaming media)

   TFRC always pushes against its maximum rate until a loss, which will
   force a decrease in the allowable rate. This induces periodic
   distortion. Can we minimize this by providing TFRC with a maximum rate,
   beyond which it will not push? (Damon Lanphear)

   * Absolutely---TFRC will not send more data than the application
   provides.

Eddie

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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:

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



Today's Topics:

   1. [ rtspspec-Bugs-640848 ] umbiguity in aggregate/non-aggregate pla (SourceForge.net)
   2. [ rtspspec-Bugs-464462 ] Content-Length header wrong in examples (SourceForge.net)
   3. [ rtspspec-Bugs-625339 ] need clarification for stand alone SDP (SourceForge.net)
   4. [ rtspspec-Bugs-803064 ] Teardown text needs rewrite (SourceForge.net)
   5. [ rtspspec-Bugs-772330 ] Clarify that RTCP shall be sent during whole session (SourceForge.net)
   6. [ rtspspec-Bugs-619421 ] Range header in PLAY response (SourceForge.net)
   7. [ rtspspec-Bugs-494533 ] The use of the word require in the RFC. (SourceForge.net)
   8. [ rtspspec-Bugs-507882 ] RTP timestamps when playing after pause (SourceForge.net)
   9. [ rtspspec-Bugs-625342 ] DESCRIBE Res needed in C.2 and C.3 (SourceForge.net)
  10. [ rtspspec-Bugs-563153 ] Appendix C: agg/non-agg mixup (SourceForge.net)

--__--__--

Message: 1
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Fri, 24 Oct 2003 09:11:54 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-640848 ] umbiguity in aggregate/non-aggregate pla

Bugs item #640848, was opened at 2002-11-19 19:24
Message generated for change (Comment added) made by magwes
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=640848&group_id=23194

Category: None
>Group: Fixed in CVS/Need WG Approval
Status: Open
Resolution: None
Priority: 5
Submitted By: Go Hori (ghori)
>Assigned to: Magnus Westerlund (magwes)
Summary: umbiguity in aggregate/non-aggregate pla

Initial Comment:
It is not clear to me who's choice it is to pick
aggregate play vs. non-aggregate play.  


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

>Comment By: Magnus Westerlund (magwes)
Date: 2003-10-24 18:11

Message:
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No comments has been given for this since my last comment. I
have edited this into the specification. Will be available
in v5. 

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

Comment By: Magnus Westerlund (magwes)
Date: 2003-03-12 15:45

Message:
Logged In: YES 
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It is the clients. With the updated specification the client
chooses this by setting up multiple streams or not in a each
session. 

This has no impact on a server that supports aggregated
control. 

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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=640848&group_id=23194


--__--__--

Message: 2
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Fri, 24 Oct 2003 01:40:56 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-464462 ] Content-Length header wrong in examples

Bugs item #464462, was opened at 2001-09-24 18:09
Message generated for change (Settings changed) made by magwes
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=464462&group_id=23194

>Category: Editorial
Group: Fix for DS - Typo/Thinko
Status: Open
Resolution: None
Priority: 5
Submitted By: Nobody/Anonymous (nobody)
Assigned to: Nobody/Anonymous (nobody)
Summary: Content-Length header wrong in examples

Initial Comment:
In most of the examples the Content-Length header is not the right value for the actual content of the header.

like the one below,which I copied from RTSP 14.5, this number doeas not match: In this case here they write "44", but if you count (together with CRLF, or even just LF) 103 chars + 8 for CRLF or 4 for just LF.

M-C: 
RTSP/1.0 200 1 OK 
Content-type: application/sdp 
Content-Length: 44 

v=0 
o=- 2890844526 2890842807 IN IP4 192.16.24.202 
s=RTSP Session i=See above t=0 0
m=audio 0 RTP/AVP 0 

C-M: 
SETUP rtsp://server.example.com/demo/548/sound RTSP/1.0 CSeq: 2 
Transport: RTP/AVP;multicast;destination=225.219.201
..............................


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

Comment By: Magnus Westerlund (magwes)
Date: 2002-10-30 10:02

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These must be fixed in all examples as soon as the updates
becomes more stable. 

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--__--__--

Message: 3
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Fri, 24 Oct 2003 06:39:15 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-625339 ] need clarification for stand alone SDP

Bugs item #625339, was opened at 2002-10-18 19:57
Message generated for change (Comment added) made by magwes
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=625339&group_id=23194

>Category: General
>Group: Fixed in CVS/Need WG Approval
Status: Open
Resolution: None
Priority: 5
Submitted By: Go Hori (ghori)
>Assigned to: Magnus Westerlund (magwes)
>Summary: need clarification for stand alone SDP 

Initial Comment:
When a SDP file is download other than via RTSP (e.g. 
HTTP, FTP), control strings need to be all absolute.  
Should probably put clarification or give an example for 
this case to avoid confusion.

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

>Comment By: Magnus Westerlund (magwes)
Date: 2003-10-24 15:39

Message:
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There has been added a new section in the SDP appendix which
writes some clarifying notes on the topic. Text will become
available in v5.

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

Comment By: Magnus Westerlund (magwes)
Date: 2002-10-30 19:38

Message:
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Yes, this is another pice of information that belong to
appendix C.

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Message: 4
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Fri, 24 Oct 2003 09:03:02 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-803064 ] Teardown text needs rewrite

Bugs item #803064, was opened at 2003-09-09 13:20
Message generated for change (Comment added) made by magwes
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Category: None
>Group: Fixed in CVS/Need WG Approval
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
>Assigned to: Magnus Westerlund (magwes)
Summary: Teardown text needs rewrite

Initial Comment:
The text for the teardown section needs a serious
rewrite. The current text is not very clear on several
matters in regards to aggregated and non-aggregated
teardown.

Should probably await resolution of bug 795295.



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

>Comment By: Magnus Westerlund (magwes)
Date: 2003-10-24 18:03

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I have now updated the text about the TEARDOWN section. It
needs review. 

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Message: 5
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Fri, 24 Oct 2003 08:42:30 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-772330 ] Clarify that RTCP shall be sent during whole session

Bugs item #772330, was opened at 2003-07-16 16:09
Message generated for change (Comment added) made by magwes
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=772330&group_id=23194

Category: None
>Group: Fixed in CVS/Need WG Approval
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
>Assigned to: Magnus Westerlund (magwes)
Summary: Clarify that RTCP shall be sent during whole session

Initial Comment:
Appendix B should be clarified to point out that RTCP
is expected to be sent from succesful response of SETUP
until TEARDOWN. During PAUSE the server and client
should both continue to send RTCP as normal. 

Motivations to this behavior is:
- to avoid timing out SSRC, 
- ensure detection of packet loss at the end of stream.
- ensure that RTCP syncrhonization can work (necessary
for multiple SSRC). 
- Ensure that RTP stream work prior to PLAY if there is
some delay between them. 



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

>Comment By: Magnus Westerlund (magwes)
Date: 2003-10-24 17:42

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A clarification about this has been written into appendix B. 

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

Comment By: Magnus Westerlund (magwes)
Date: 2003-07-16 16:12

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At the same time it should be made clear the RTCP BYE is
sent when the session is terminated, i.e. at the time of
teardown.



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--__--__--

Message: 6
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Fri, 24 Oct 2003 05:27:55 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-619421 ] Range header in PLAY response

Bugs item #619421, was opened at 2002-10-07 01:14
Message generated for change (Comment added) made by magwes
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Category: General
>Group: Fixed in CVS/Need WG Approval
Status: Open
Resolution: None
Priority: 5
Submitted By: Nobody/Anonymous (nobody)
>Assigned to: Magnus Westerlund (magwes)
Summary: Range header in PLAY response

Initial Comment:
Current RTSP RFC makes Range header in PLAY response
from server optional.

If the PLAY request includes a Range header indicating
a possible seek, then it is probably necessary to
include Range in the response since it is possible that
the requested seek position could not be honored and
playback will resume at the nearest seekable point
which is not known to the client.

Note:  To fix this issue in 3GPP, Range in a PLAY
response was made mandatory for all PLAY responses.



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

>Comment By: Magnus Westerlund (magwes)
Date: 2003-10-24 14:27

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The inclusion of the Range header in PLAY responses is now a
MUST. We haven't found any problem with this. If anybody
finds this, please raise it. 

I have also put in Maureen into the acknowledgement section. 

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

Comment By: Magnus Westerlund (magwes)
Date: 2003-01-31 13:49

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As discussed in the teleconf the 22 jan 2003:
We should look at strengting the requirement on having the
Range header also for live streams when using RTP. A
solution based on Range:npt=now- with optional time or using
Range:clock=x can work. But careful wording is required.

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

Comment By: Rob Lanphier (robla)
Date: 2002-11-20 21:06

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One other thing...put Maureen Chesire in the acknowlegements
for raising this....

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

Comment By: Rob Lanphier (robla)
Date: 2002-11-20 21:02

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Per a conversation with Magnus and I, we think we should
soften the language somewhat (strong SHOULD instead of MUST).  

One case to consider is a live stream, which one would think
would obviate the need for this header.  However, the
rationale for having it in live is when dealing with RTP. 
Since RTP audio and video streams can have a completely
different timestamp base, there needs to be an out-of-band
technique for synchronization.  The two techniques used in
RTSP are:

1.  RTSP "Range:"<->RTSP "RTP-Info:"<->RTP timestamp synch
2.  RTCP timestamp<->NTP mapping

Eventually, a well-behaved RTSP client will synch the
streams based on
the RTCP timestamps (when received).  However, there's a
period before the first RTCP packet arrives that there's no
sync, unless a Range header is known.

So, that also points to what the Range header should look
like.  The
Range header should include an open-ended wallclock range,
like so:
   Range: clock=20021120T133205Z- 
   RTP-Info:url=rtsp://example.com/fizzle/audiotrack;
            seq=5712;rtptime=934207921,
            url=rtsp://example.com/fizzle/videotrack;
            seq=57654;rtptime=2792482193

The spec should make this clear.

An interesting consideration to make here is a restriction
on Range response. The current rtsp spec states that "The
unit of the range in the reply is the same as that in the
request".  Perhaps an alternate syntax, if the PLAY request
has a Range: header in npt is as follows:

   Range: npt=now;time=20021120T133205Z

The "time" allows for mapping to occur.



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

Comment By: Magnus Westerlund (magwes)
Date: 2002-10-30 19:36

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Yes, the range header is need for RTP synchonization. The
draft specification has been updated to require this when
using RTP to make synchonization possible. 

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--__--__--

Message: 7
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Fri, 24 Oct 2003 01:42:20 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-494533 ] The use of the word require in the RFC.

Bugs item #494533, was opened at 2001-12-18 11:16
Message generated for change (Comment added) made by magwes
You can respond by visiting: 
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>Category: Editorial
Group: Fix for DS - Large task
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: The use of the word require in the RFC.

Initial Comment:
The RFC contains different usage of the word require. Sometimes it means that a implementation 
must interpret and understand a header or method. Othertimes the a certain header must be sent 
togheter with a method. 

One fix is to create a better table for the headers that clarifies the two different cases. Another is 
looking over the methods and use cases and clearly write when they must be included. 

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

>Comment By: Magnus Westerlund (magwes)
Date: 2003-10-24 10:42

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There are also a great need to look at the usage of
normative language according to RFC 2119. 

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

Comment By: Rob Lanphier (robla)
Date: 2001-12-19 02:29

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We need to rework that table to clarify the difference between the two.

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--__--__--

Message: 8
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Thu, 23 Oct 2003 06:12:43 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-507882 ] RTP timestamps when playing after pause

Bugs item #507882, was opened at 2002-01-24 09:30
Message generated for change (Comment added) made by magwes
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=507882&group_id=23194

Category: General
Group: Resolved - Need write up
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
>Assigned to: Aravind Narasimhan (arvy_n)
Summary: RTP timestamps when playing after pause

Initial Comment:
What happends to the RTP timestamp when you issue a play after being paused for some time?

The following was my interpretation of the proposal for a solution. The
RTP timestamp is equal to display time. So it you do pause, wait five
seconds and then press play the timestamp will be 5 seconds later for
the first packet when resuming.

This behavior works fine when the play contains a range, for queued play
following after each other, but not for resuming play. The reason is the
delay between a user on the client side pressing pause and until the
client stops receiving data packets. If the client then does resume
play, i.e. play without range, the server will continue after the last
packet sent. In this case the client needs to show first the packet
arriving directly after pressing pause (which according to the used
model is past time), and then correctly synchronize the new packet which
has a significant offset in timestamp corresponding to the period of
pause the server has seen. The solution to this is to not use resume
play and instead say play from the point where user pressed pause.

On the other side I don't believe any other model will not have worse
problems.

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

>Comment By: Magnus Westerlund (magwes)
Date: 2003-10-23 15:12

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Aravind has agreed to write up the example and necessary
text. Transfering bug to him.

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

Comment By: Magnus Westerlund (magwes)
Date: 2003-09-22 10:48

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A Clear example of RTP Timestamp and synchronization will be
added in appendix B - RTP transport.

The discussion in MMUSIC has made it clear that a continous
time model is needed to make RTCP work correctly. It has to
be accpeted that for on-demand media this will result in
slightly more code for the clients and servers. However all
pices to makes things work is already present. 

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

Comment By: Magnus Westerlund (magwes)
Date: 2003-05-23 16:05

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This has been further discussed in the MMUSIC mailing list. 

A important point is that RTP and RTCP to function as
intended in this environment must have a timestamp space
that increase continously with real time. 

A client handling the RTSP range header and RTP-Info can
always synchronize the data correctly independtly of which
case the server sent out. 

Using a continous RTP Timestamp space allows the same
timestamp model for both on-demand and live media.

The proposed resolution is to clrafiy in the RTP appendix of
the specification that a RTP timestamp space continous in
regards to real-time SHALL be used. Further write a warning
that this has been missunderstood and clients is RECOMMENDED
to handle both cases. 


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Message: 9
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Fri, 24 Oct 2003 07:32:52 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-625342 ] DESCRIBE Res needed in C.2 and C.3

Bugs item #625342, was opened at 2002-10-18 20:00
Message generated for change (Comment added) made by magwes
You can respond by visiting: 
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>Category: General
>Group: Fixed in CVS/Need WG Approval
Status: Open
Resolution: None
Priority: 5
Submitted By: Go Hori (ghori)
>Assigned to: Magnus Westerlund (magwes)
Summary: DESCRIBE Res needed in C.2 and C.3

Initial Comment:
In the examples for aggregate control SDP and 
subsequent SETUP request, it is confusing how a URL 
for SETUP is created.  It is missing a DESCRIBE 
response and gives an impression that a sesseion level 
control string was used as a base URL.


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

>Comment By: Magnus Westerlund (magwes)
Date: 2003-10-24 16:32

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I agree that the example was not clear. This has now been
corrected by including a DESCRIBE response for the
aggregated example which has defines a Content-Base, which
is the most  reasonable solution to this problem. 



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

Comment By: Magnus Westerlund (magwes)
Date: 2002-10-31 13:24

Message:
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We may also want to define a dedault rule to create control
URLs based on the first media setup so that sessions missing
session level control URLs are still possible to set up in
aggregates.

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Message: 10
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Fri, 24 Oct 2003 04:37:03 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-563153 ] Appendix C: agg/non-agg mixup

Bugs item #563153, was opened at 2002-06-01 03:42
Message generated for change (Comment added) made by magwes
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Category: General
Group: Fixed in CVS/Need WG Approval
Status: Open
Resolution: None
Priority: 5
Submitted By: Rob Lanphier (robla)
>Assigned to: Magnus Westerlund (magwes)
Summary: Appendix C: agg/non-agg mixup

Initial Comment:
The initial descriptions of aggregate and non-aggregate
contol are reversed, and poorly written.

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

>Comment By: Magnus Westerlund (magwes)
Date: 2003-10-24 13:37

Message:
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I have went over the SDP appendix once more and found some
more strangeness regarding aggregated and non-aggregated.
This is now fixed. 

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

Comment By: Rob Lanphier (robla)
Date: 2002-06-06 22:24

Message:
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It turns out this had already been fixed in the draft-00
version.  However, the wording was still very confusing. 
This has been fixed in the version checked into CVS.

http://cvs.sourceforge.net/cgi-bin/viewcvs.cgi/rtspspec/sdp.tex.diff?r1=1.3&r2=1.4

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Subject: Re: [MMUSIC] SDP b= clarification
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On Wednesday, Oct 22, 2003, at 21:20 Europe/London, Tom Marshall wrote:
>>>> The ABNF in draft-ietf-mmusic-sdp-new-13.txt explicitly allows
>>>> repetition.
>>>>
>>>>  bandwidth-fields =    *("b=" bwtype ":" bandwidth CRLF)
>>>>  bwtype =              token
>>>>  bandwidth =           1*DIGIT
>>>
>>> Cool.  Could we get a note in the text that corresponds with the 
>>> ABNF?
>>
>> Provided there are no objections, if you suggest explicit wording
>> changes, I can incorporate them into -15.
>
> This would be the simplest change:
>
>   --- draft-ietf-mmusic-sdp-new-14.txt    2003-10-22 
> 12:51:26.000000000 -0700
>   +++ draft-ietf-mmusic-sdp-new-14-edited.txt     2003-10-22 
> 13:18:57.000000000 -0700
>   @@ -495,7 +495,7 @@
>             p=* (phone number)
>             c=* (connection information - not required if included in
>                  all media)
>   -         b=* (bandwidth information)
>   +         b=* (zero or more bandwidth information lines)
>             One or more time descriptions (see below)
>
>
>   @@ -519,7 +519,7 @@
>             i=* (media title)
>             c=* (connection information - optional if included at
>                  session-level)
>   -         b=* (bandwidth information)
>   +         b=* (zero or more bandwidth information lines)
>             k=* (encryption key)
>             a=* (zero or more media attribute lines)

Will be in -15

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


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From: Thomas Zeng <zeng@PacketVideo.COM>
To: "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>,
        Thomas Zeng
	 <zeng@PacketVideo.COM>
Cc: IETF MMUSIC WG <mmusic@ietf.org>,
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Subject: RE: [MMUSIC] RTSP: Minutes telecon 14th of October 2003
Date: Sun, 26 Oct 2003 22:44:48 -0800
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Magnus:

Our use case is best handled by a feature-tag, I think.  Let me clarify:
even if "time=" is deprecated
in Range header in the core spec, we can still "resurrect" it in a extension
to RTSP protocol, as long as
we address any ambuguity (I believe we can)?

More embedded.

thanks,
thomas

> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Wednesday, October 22, 2003 1:33 AM
> To: Thomas Zeng
> Cc: IETF MMUSIC WG
> Subject: Re: [MMUSIC] RTSP: Minutes telecon 14th of October 2003
> 
> 
> Hi Thomas,
> 
> If you have a use cases for this, I think it is possible to keep the 
> timed functionality, under the condition that you provide all 
> necessary 
> specification text to solve the problems with timed PLAYs.

We will either propose timed PLAY or timed PLAYLIST_PLAY.
> 
> I would also like to have some more information about use case.
> 
The use case involves server-side playlist play, where one may desire to
have the option to jump to a new
item in the playlist, but only at the end of the current item.
> The big problem with the timed requests are the following:
> 
> - Are they allowed to be stacked, are multiple possible to be 
> requested 
> for later execution? If yes, the real problems exist.

No, only outstanding timed request is allowed. That is, the stack size is 1.
> 
> - If you use timed requests how does the interaction happens with the 
> state machine. When are the state transition made, when the 
> request is 
> received or when it is executed.

The first PLAY is always executed immediately. So state transition is no
issue.
In another words, timed PLAY is only in PLAYING state for clip transition.
> 
> - How is errors at the time of execution handled?
Error will be reported via event report (say via SET_PARAMETER or ANNOUNCE).
> 
> - If multiple requests are allowed, how does one handle if 
> two request 
> collide.

Again, only one timed request is allowed.
> 
> - How do you stop a timed request, between acceptance and execution?
> 
By sending another PLAY or PLAYLIST_PLAY.
> These are probably not all the problems, simply the ones I 
> can remember 
> of the top of my head.
> 
> So please provide a solution proposal so we can discuss and 
> review if it 
> solves the problems in a satisfactory way.
> 
> cheers
> 
> Magnus
> 
> Thomas Zeng wrote:
> 
> > Magnus:
> > 
> > During one of the telecon for RTSP, I recall that a 
> decision was made to 
> > deprecate "time=" parameter in Range header.
> > 
> > We have found a use case in the context of playlist 
> support, where we would
> > like to use
> > "time=" parameter in Range header in the PLAY request as designed in
> > RFC2326, quoted below:
> > 
> > 10.5 PLAY
> > 
> > ...
> > 
> >    The Range header may also contain a time parameter. This 
> parameter
> >    specifies a time in UTC at which the playback should 
> start. If the
> >    message is received after the specified time, playback is started
> >    immediately. The time parameter may be used to aid in 
> synchronization
> >    of streams obtained from different sources.
> > 
> > 
> > 
> > In light of this, would it be possible to retain this "time 
> parameter"?
> > 
> -- 
> 
> 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 Thomas,

I don't think there will be any problem with later writing a extension 
draft resolving the problems and how one likes this to work. Through the 
use of a feature tag and a the Require header the client can ensure that 
the server interprets the timed field correctly in such requests. Please 
note that I am not removing the syntactical element, rather declaring it 
as shall not be used in the core spec for other purposes then REDIRECT.

I hope you look over the possible solutions for playlist to find what is 
a good solution.

If one take the simple approach only having the desire to delay the 
start of the playback the problems would not be to hard to solve. 
However it would be simpler if one haves a separate extension draft for 
this rather than using the core specification, which I interprets as 
being your intention.

Cheers

Magnus


Thomas Zeng wrote:

> Magnus:
> 
> Our use case is best handled by a feature-tag, I think.  Let me clarify:
> even if "time=" is deprecated
> in Range header in the core spec, we can still "resurrect" it in a extension
> to RTSP protocol, as long as
> we address any ambuguity (I believe we can)?
> 
> More embedded.
> 
> thanks,
> thomas
> 
> 
>>-----Original Message-----
>>From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
>>Sent: Wednesday, October 22, 2003 1:33 AM
>>To: Thomas Zeng
>>Cc: IETF MMUSIC WG
>>Subject: Re: [MMUSIC] RTSP: Minutes telecon 14th of October 2003
>>
>>
>>Hi Thomas,
>>
>>If you have a use cases for this, I think it is possible to keep the 
>>timed functionality, under the condition that you provide all 
>>necessary 
>>specification text to solve the problems with timed PLAYs.
> 
> 
> We will either propose timed PLAY or timed PLAYLIST_PLAY.
> 
>>I would also like to have some more information about use case.
>>
> 
> The use case involves server-side playlist play, where one may desire to
> have the option to jump to a new
> item in the playlist, but only at the end of the current item.
> 
>>The big problem with the timed requests are the following:
>>
>>- Are they allowed to be stacked, are multiple possible to be 
>>requested 
>>for later execution? If yes, the real problems exist.
> 
> 
> No, only outstanding timed request is allowed. That is, the stack size is 1.
> 
>>- If you use timed requests how does the interaction happens with the 
>>state machine. When are the state transition made, when the 
>>request is 
>>received or when it is executed.
> 
> 
> The first PLAY is always executed immediately. So state transition is no
> issue.
> In another words, timed PLAY is only in PLAYING state for clip transition.
> 
>>- How is errors at the time of execution handled?
> 
> Error will be reported via event report (say via SET_PARAMETER or ANNOUNCE).
> 
>>- If multiple requests are allowed, how does one handle if 
>>two request 
>>collide.
> 
> 
> Again, only one timed request is allowed.
> 
>>- How do you stop a timed request, between acceptance and execution?
>>
> 
> By sending another PLAY or PLAYLIST_PLAY.
> 



-- 

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,

You having this discussion on a friday evening (my time) makes it kind 
of hard of give comments into the discussion :-). So I guess it falls on 
my lot to summarize and give comments on what I see as important 
considerations.

There are several possible solutions that all have advantages and 
drawbacks.

1. Using RTCP BYE
- bye sending a RTCP bye the server will signal that it is leaving the 
session with the indicated SSRC(s). As this is probably not the intended 
effect in many cases RTCP bye is not semantically correct.
- Killing the SSRC prevents the continues reporting of RTCP sender 
reports allowing a client not supporting the end of stream mechanism to 
detect that it is lost.
- The intention of RTCP BYE is to work as hint to help keeping the 
accurate group size, therefore it is not very robust. RTCP bye are only 
sent once and any receiver not getting it will time-out the source instead.

Personal comment: I am not in favour of using this mechanism.

2. Using an RTCP extension, APP or other
- By including SSRC and RTP sequence number of the last packet to be 
sent the server the RTCP packet can be sent both earlier and later than 
the transmission of the last RTP packet.
- Requires the usage of RTCP, which in my opinion in most cases is good. 
I know there might be use cases where RTCP may not be possible to use 
even from server to client(s).
- RTCP is not robust, message will be need to be repeated. Due to RTCP 
bandwidth limitation this might result in that it takes some time to get 
the message across.

3. Using the RTSP signalling channel with an END_OF_STREAM method.
- Sent in the RTSP signalling channel, so if TCP is used it is robust. 
Will get the message over as soon as TCP can transmit it. However due to 
packet losses this time is not bound, it might still arrive late.
- Is not RTP specific, through the usage of other transport protocols 
equivalent to RTP-Info they could inform what the last packet that 
belongs to the request is.
- Due to the use of RTSP signalling channel it may go through one or 
more proxies. Thus taking longer time than the media path.
- If the client drops the signalling connection it will not be possible 
to get over.
- One message will cover all streams in the same RTSP session.

4. Use a RTP extension.
This solution can either be implemented through a new RTP payload format 
including the necessary information, or through some RTP header extension.
- Will go in the media channel as media rather than signalling.
- Can't be sent in advance for certain media requiring continuos RTP 
sequence numbers if it requires its own packets.
- Is not robust will require that one either uses some type of ACKs and 
adaptive transmission of them or do a repetition accomplishing high 
enough delivery probability.
- The use of header extension would prevent other header extensions.
- The use of a payload format would either require its own RTP packet or 
  use of RFC 2198.


General comments:

- Please note that any end of stream indication will not be 100% 
reliable and guaranteed to arrive in time before the last packet arrives.

- Regarding messages from server to client. The client are not required 
to keep the TCP connection open, however it is strongly recommended. My 
intention with the current text around connections is that the protocol 
shall survive the loss of the signalling channel. That this allows for 
clients to close the TCP connection has to be accepted.

- Regarding the requirement about state transition as an indication. 
This is not suitable in all use cases. I think this mechanism if located 
in RTP/RTCP then it should work also for multicast. Thus one needs to 
look into the separate use cases.

I think we should further explore the pro and cons of proposal 2 to 4. 
Thus trying to decide what the desired solution is. There are probably 
further considerations that I have not yet thought of.


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: Rod.Walsh@nokia.com
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Subject: [MMUSIC] time-lined static media with FLUTE & IMG
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Hi Magnus, Ross (and everyone)

1=BD-2 years back there was a draft on time-lined static media =
(draft-westerlund-avt-rtp-static-media-01) - basically to send files, =
text (etc) with timed presentation along with AV media streams. Since =
the basic requirements haven't gone away, but RMT has made progress on =
usable protocols, what do you think about using FLUTE =
(draft-ietf-rmt-flute-02) for the same job? With a quick comparison I =
think it should be pretty straight forward, although you may have some =
preferences (or extensions to the FLUTE-FDT) if you already have a =
working RTP-payload'ed implementation.

Ross mentioned sending ID-3 tags in-band with audio streams. Any ideas =
you have on doing this with FLUTE (ID3) and RTP (music) could be pretty =
useful for the Internet Media Guide (IMG) work in MMUSIC - how to move =
around (and announce) metadata for 'foo' multicast and unicast sessions.

Cheers, Rod.

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



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From: Colin Perkins <csp@csperkins.org>
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I've just submitted draft-ietf-mmusic-sdp-new-15.txt to address IESG 
concerns and other comments raised on the mailing list. Changes since 
-14 include:

- Update references:
         RFC 1889 -> RFC 3550
         RFC 1890 -> RFC 3551
         draft-ietf-mmusic-sdp4nat-05.txt -> RFC 3605
- Clarify that b= may be duplicated
- Clarify that multiple addresses or "c=" lines can only be specified if
   they provide multicast addresses for different layers
- Clarify use of b= with layered coding:
         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.
- Clarify which ports are associated with an m= line when layered coding
   is used
- Remove extra ")" from the definition of m1 in the ABNF
- Add reference to RFC2396 in definition of u=
- Add MIME registration to the IANA Considerations section, and note 
that
   this draft updates RFC 2327
- In the description of "k=", note that S/MIME email and TLS protected 
HTTP
   and SIP might be appropriate secure transports for SDP. Also note 
that it
   is necessary to "ensure that the secure channel is with the party 
that is
   authorized to join the session, not an intermediary" and "If a caching
   server is used, there ought to be a way to keep the server from 
accessing
   the key"
- The recommendation that systems "SHOULD NOT deliver the user into an
   interactive multimedia session without the user being aware that this
   will happen" in the Security Considerations section was strengthened 
to
   "MUST NOT deliver the user into an interactive multimedia session 
unless
   the user has explicitly pre-authorized such action"
- A comment in the ABNF for "token-char" allowed a larger set of 
characters
   than the actual definition. The comment has been removed, leaving the 
set
   of allowed characters as specified in the ABNF
- Clarify that UTF-8 normalization is not needed because comparisons are
   not done on fields with UTF-8.
- Clarify that k= is NOT RECOMMENDED and give pointers to the 
sdescriptions
   and key-mgmt drafts.
- Clarify that k= does not include an algorithm identifier, key type, or
   any other information about the key.
- Clarify that k= does not support the transfer of two keys
- Remove reference to RFC 3551/1890 from the description of k=, since 
that
   nowhere specified how k= was used.
- Note that a session description may contain URIs which may be 
dereferenced.
   This makes the SDP non-self contained.
- Remove "Changes from RFC 2327" section, since it was very inaccurate 
and
   outdated.

Feedback on these changes would be appreciated.

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

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--- sdp-14.xml	Fri Sep  5 12:45:27 2003
+++ sdp.xml	Mon Oct 27 12:21:58 2003
@@ -1,10 +1,11 @@
 <?xml version="1.0"?>
 <?rfc toc="yes"?>
+<?rfc tocdepth="2"?>
 <?rfc strict="yes"?>
 <?rfc compact="yes"?>
 <!DOCTYPE rfc SYSTEM "rfc2629.dtd">
 <rfc obsoletes="2327, 3266" category="std" ipr="full2026" 
-     docName="draft-ietf-mmusic-sdp-new-14.txt">
+     docName="draft-ietf-mmusic-sdp-new-15.txt">
 <front>
 <title abbrev="SDP"> SDP: Session Description Protocol </title>
   <author initials="M." surname="Handley" fullname="Mark Handley">
@@ -44,7 +45,7 @@
       <email>csp@csperkins.org</email>
     </address>
   </author>
-  <date day="4" month="September" year="2003" />
+  <date day="27" month="October" year="2003" />
   <abstract>
   <t> 
     This memo defines the Session Description Protocol (SDP). SDP is
@@ -58,6 +59,9 @@
 <!---------------------------------------------------------------------------->
 <section title="Introduction">
 
+<t> [Note to RFC Editor: All references to RFC XXXX should be replaced by 
+    the RFC number of this document, when published.] </t>
+
 <t> When initiating multimedia teleconferences, voice-over-IP calls,
     streaming video, or other real-time sessions, there is a requirement 
     to convey media details, transport addresses, and other session
@@ -67,10 +71,11 @@
     irrespective of how that information is transported.  SDP is purely a
     format for session description - it does not incorporate a transport
     protocol, and is intended to use different transport protocols as
-    appropriate, including the Session Announcement Protocol [RFC2974],
-    Session Initiation Protocol [RFC3261], Real-Time Streaming Protocol
-    [RFC2326], electronic mail using the MIME extensions, and the Hypertext
-    Transport Protocol. </t>
+    appropriate, including the Session Announcement Protocol 
+    <xref target="RFC2974" />, Session Initiation Protocol 
+    <xref target="RFC3261" />, Real-Time Streaming Protocol
+    <xref target="RFC2326" />, electronic mail using the MIME extensions, 
+    and the Hypertext Transport Protocol. </t>
 
 <t> SDP is intended to be general purpose so that it can be used in a wide 
     range of network environments and applications.  However, it is not
@@ -104,7 +109,8 @@
 
 <t> The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
     "SHOULD", "SHOULD NOT", "RECOMMENDED",  "MAY", and "OPTIONAL" in this 
-    document are to be interpreted as described in RFC 2119 [RFC2119]. </t>
+    document are to be interpreted as described in 
+    <xref target="RFC2119">RFC 2119</xref>. </t>
 
 </section>
 <!---------------------------------------------------------------------------->
@@ -122,23 +128,23 @@
       required to participate in the session. </t>
 
   <t> One protocol commonly used to implement such a distributed directory
-      is the Session Announcement Protocol, SAP [RFC2974]. SDP provides
+      is the <xref target="RFC2974">Session Announcement Protocol, SAP</xref>. SDP provides
       the recommended session description format for such announcements. </t>
   </section>
 
   <section title="Session Initiation">
-  <t> The Session Initiation Protocol, SIP [RFC3261] is an application layer 
+  <t> The <xref target="RFC3261">Session Initiation Protocol, SIP</xref> is an application layer 
       control protocol for creating, modifying and terminating sessions
       such as Internet multimedia conferences, Internet telephone calls and
       multimedia distribution.  The SIP messages used to create sessions
       carry session descriptions which allow participants to agree on a set
       of compatible media types. These session descriptions are commonly
-      formatted using SDP. When used with SIP, the offer/answer model
-      [RFC3264] provides a limited framework for negotiation using SDP. </t>
+      formatted using SDP. When used with SIP, the <xref target="RFC3264">
+      offer/answer model</xref> provides a limited framework for negotiation using SDP. </t>
   </section>
 
   <section title="Streaming media">
-  <t> The Real Time Streaming Protocol, RTSP [RFC2326], is an application-
+  <t> The <xref target="RFC2326">Real Time Streaming Protocol, RTSP</xref>, is an application-
       level protocol for control over the delivery of data with real-time
       properties.  RTSP provides an extensible framework to enable
       controlled, on-demand delivery of real-time data, such as audio and
@@ -250,7 +256,8 @@
         typically conveyed by encrypting the session description during
         distribution.  The details of how encryption is performed are
         dependent on the mechanism used to convey SDP - e.g. mechanisms are
-        defined for SDP transported using SAP [RFC2974] and SIP [RFC3261]. </t>
+        defined for SDP transported using <xref target="RFC2974">SAP</xref>
+	and <xref target="RFC3261">SIP</xref>. </t>
 
     <t> If a session announcement is private it is possible to use that
         private announcement to convey encryption keys necessary to decode
@@ -275,7 +282,7 @@
 
     <section title="Internationalization">
     <t> The SDP specification recommends the use of the ISO 10646 character
-        sets in the UTF-8 encoding [RFC2279] to allow many different languages 
+        sets in the UTF-8 encoding <xref target="RFC2279" /> to allow many different languages 
 	to be represented.  However, to assist in compact representations, 
 	SDP also allows other character sets such as ISO 8859-1 to be used 
 	when desired.  Internationalization only applies to free-text fields 
@@ -290,7 +297,9 @@
 <t> SDP session descriptions are entirely textual using the ISO 10646
     character set in UTF-8 encoding.  SDP field names and attribute names
     use only the US-ASCII subset of UTF-8, but textual fields and
-    attribute values MAY use the full ISO 10646 character set.  The
+    attribute values MAY use the full ISO 10646 character set. Field and 
+    attribute values which use the full UTF-8 character set are never directly 
+    compared, hence there is no requirement for UTF-8 normalization.  The
     textual form, as opposed to a binary encoding such as ASN.1 or XDR,
     was chosen to enhance portability, to enable a variety of transports
     to be used (e.g, session description in a MIME email message) and to
@@ -306,6 +315,11 @@
     discarded.  This also allows rapid discarding of encrypted
     announcements for which a receiver does not have the correct key. </t>
 
+<t> An SDP session description may contain URIs which reference 
+    external content in the "u=", "k=" and "a=" lines. These URIs may be
+    dereferenced in some cases, making the session description non-self
+    contained. </t>
+
 <t> An SDP session description consists of a number of lines of text of
     the form: </t>
 
@@ -346,7 +360,7 @@
       p=* (phone number)
       c=* (connection information - not required if included in
            all media)
-      b=* (bandwidth information)
+      b=* (zero or more bandwidth information lines)
       One or more time descriptions (see below)
       z=* (time zone adjustments)
       k=* (encryption key)
@@ -362,7 +376,7 @@
       i=* (media title)
       c=* (connection information - optional if included at
            session-level)
-      b=* (bandwidth information)
+      b=* (zero or more bandwidth information lines)
       k=* (encryption key)
       a=* (zero or more media attribute lines)
 </artwork>
@@ -442,7 +456,7 @@
 	    type> and &lt;address> form a globally unique identifier for the 
 	    session.  The method of session id allocation is up to the creating 
 	    tool, but it has been suggested that a Network Time Protocol (NTP)
-            format timestamp be used to ensure uniqueness [RFC1305]. </t>
+            format timestamp be used to ensure uniqueness <xref target="RFC1305" />. </t>
 
         <t hangText="&lt;version>"> is a version number for this announcement.
 	    It is needed for proxy announcements to detect which of several
@@ -526,7 +540,8 @@
     </artwork>
     </figure>
  
-    <t> A URI is a Universal Resource Identifier as used by WWW clients.
+    <t> A URI is a Universal Resource Identifier as used by WWW clients 
+        <xref target="RFC2396"/>.
         The URI should be a pointer to additional information about the
         conference. This field is OPTIONAL, but if it is present it MUST
         be specified before the first media field. No more than one URI
@@ -715,8 +730,9 @@
         </t>
 
     <t> Multiple addresses or "c=" lines MAY be specified on a per-media
-        basis. They MUST NOT be specified for a session-level "c=" field.
-	</t>
+        basis only if they provide multicast addresses for different layers
+	in a hierarchical or layered encoding scheme. They MUST NOT be
+	specified for a session-level "c=" field. </t>
 
     <t> The slash notation described above MUST NOT be used for IP unicast
         addresses. </t>
@@ -737,29 +753,30 @@
         modifiers defined in this memo use the default units). </t>
 
     <t> The &lt;modifier> is a single alphanumeric word giving the meaning 
-        of the bandwidth figure. Two modifiers are initially defined: </t>
+        of the bandwidth figure. Two modifiers are initially defined: 
 
-    <figure>
-    <artwork>
-   CT  Conference Total
+    <list style="hanging">
+    <t hangText="CT">  
        If the bandwidth of a session or media in a session is
        different from the bandwidth implicit from the scope, a
        "b=CT:..." line should be supplied for the session giving
        the proposed upper limit to the bandwidth used.  The primary
        purpose of this is to give an approximate idea as to whether
-       two or more sessions can co-exist simultaneously.
+       two or more sessions can co-exist simultaneously. 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. </t>
 
-   AS  Application-Specific Maximum
+    <t hangText="AS">
        The bandwidth is interpreted to be application-specific (it
        will be the application's concept of maximum bandwidth).
        Normally this will coincide with what is set on the
        application's "maximum bandwidth" control if applicable.
        For RTP based applications, AS gives the RTP "session
-       bandwidth" as defined in section 6.2 of [RFC1889].
-    </artwork>
-    </figure>
+       bandwidth" as defined in section 6.2 of <xref target="RFC3550"/>.</t>
+    </list>
 
-    <t> Note that CT gives a total bandwidth figure for all the media at
+        Note that CT gives a total bandwidth figure for all the media at
         all sites.  AS gives a bandwidth figure for a single media at a
         single site, although there may be many sites sending
         simultaneously.</t>
@@ -798,7 +815,7 @@
 
     <t> The first and second sub-fields give the start and stop times for
         the session respectively. These values are the decimal representation 
-	of Network Time Protocol (NTP) time values in seconds [RFC1305].  
+	of Network Time Protocol (NTP) time values in seconds <xref target="RFC1305"/>.  
 	To convert these values to UNIX time, subtract decimal 2208988800.
 	</t>
 
@@ -941,13 +958,28 @@
     </figure>
 
     <t> If transported over a secure and trusted channel, the session
-        description protocol MAY be used to convey encryption keys.  A key
-        field is permitted before the first media entry (in which case it
-        applies to all media in the session), or for each media entry as
-        required. </t>
-
-    <t> The format of keys and their usage is outside the scope of this
-        document, but see [RFC1890] for an example. </t>
+        description protocol MAY be used to convey encryption keys. A
+	simple mechanism for key exchange is provided by the key field
+	("k=") although this is primarily supported for compatibility
+	with older implementations and its use is NOT RECOMMENDED. Work 
+	is in progress to define new key exchange mechanisms for use 
+	with SDP 
+	<xref target="I-D.ietf-mmusic-sdescriptions"/>
+	<xref target="I-D.ietf-mmusic-kmgmt-ext"/>
+	and it is expected that new applications will use those mechanisms. 
+	</t>
+	
+    <t> A key field is permitted before the first media entry (in which
+        case it applies to all media in the session), or for each media
+	entry as required.  The format of keys and their usage is outside
+	the scope of this document, and the key field provides no way to
+	indicate the encryption algorithm to be used, key type, or other
+	information about the key: this is assumed to be provided by the
+	higher-level protocol using SDP. If there is a need to convey this
+	information within SDP, the extensions mentioned previously SHOULD
+	be used.  Many security protocols require two keys, one for
+	confidentiality and another for integrity. This specification does
+	not support the transfer of two keys.   </t>
 
     <t> The method indicates the mechanism to be used to obtain a usable
         key by external means, or from the encoded encryption key given.
@@ -972,9 +1004,10 @@
 
       A Universal Resource Identifier is included in the key field.
       The URI refers to the data containing the key, and may require
-      additional authentication before the key can be returned.  When
-      a request is made to the given URI, the MIME content-type of
-      the reply specifies the encoding for the key in the reply.
+      additional authentication before the key can be returned. When
+      a request is made to the given URI, the reply should specify
+      the encoding for the key. The URI is often a secure HTTP URI, 
+      although this is not required.
 
    k=prompt
 
@@ -990,8 +1023,14 @@
 
     <t> The key field MUST NOT be used unless it can be guaranteed that
         the SDP is conveyed over a secure and trusted channel. An example
-        of such a channel might be SDP embedded inside an S/MIME message.
-	</t>
+        of such a channel might be SDP embedded inside an S/MIME message
+	or a TLS protected HTTP or SIP session. It is important to ensure
+	that the secure channel is with the party that is authorized to
+	join the session, not an intermediary: if a caching proxy server is
+	used, it is important to ensure that the proxy is either trusted or
+	unable to access the SDP. Definition of appropriate security measures 
+	is beyond the scope of this specification, and should be defined by 
+	the users of SDP. </t>
     </section>
 
     <section title="Attributes (&quot;a=&quot;)">
@@ -1080,9 +1119,9 @@
         network being used as specified in the relevant "c=" field, and on
         the transport protocol defined in the third sub-field.  Other
         ports used by the media application (such as the RTCP port
-        [RFC1889]) MAY be derived algorithmically from the base media port
+        <xref target="RFC3550"/>) MAY be derived algorithmically from the base media port
         or MAY be specified in a separate attribute (e.g. "a=rtcp:" as
-        defined in [RTCPSDP]). </t>
+        defined in <xref target="RFC3605"/>). </t>
 
     <t> For applications where hierarchically encoded streams are being
         sent to a unicast address, it may be necessary to specify multiple
@@ -1097,8 +1136,9 @@
 
     <t> In such a case, the ports used depend on the transport protocol.
         For RTP, the default is that only the even numbered ports are used
-        for data and the corresponding one-higher odd port is used for
-        RTCP.  For example: </t>
+        for data with the corresponding one-higher odd ports used for the
+        RTCP belonging to the RTP session, and the &lt;number of ports>
+        denoting the number of RTP sessions.  For example: </t>
 
     <figure>
     <artwork>
@@ -1110,7 +1150,7 @@
         and 49172 and 49173 form the second RTP/RTCP pair. RTP/AVP is the
         transport protocol and 31 is the format (see below). If non-
         contiguous ports are required, they must be signalled using a
-        separate attribute (e.g. "a=rtcp:" as defined in [RTCPSDP]). </t>
+        separate attribute (e.g. "a=rtcp:" as defined in <xref target="RFC3605"/>). </t>
 
     <t> If multiple addresses are specified in the "c=" field and multiple
         ports are specified in the "m=" field, a one-to-one mapping from
@@ -1162,8 +1202,8 @@
         RTP PCM audio.  In addition, relays and monitoring tools that are
         transport-protocol-specific but format-independent are possible.</t>
 
-    <t> For RTP media streams operating under the RTP Audio/Video Profile
-        [RFC1890], the protocol field is "RTP/AVP".  Should other RTP
+    <t> For RTP media streams operating under the <xref target="RFC3551">RTP 
+        Audio/Video Profile</xref>, the protocol field is "RTP/AVP".  Should other RTP
         profiles be defined in the future, their profiles will be
         specified in the same way.  For example, the protocol field
         "RTP/XYZ" would specify RTP operating under a profile whose short
@@ -1258,7 +1298,7 @@
         defined in the RTP Audio/Video Profile) packetisation is required,
         the "ptime" attribute is used as given below. </t>
 
-    <t> For more details on RTP audio and video formats, see [RFC1890]. </t>
+    <t> For more details on RTP audio and video formats, see <xref target="RFC3551"/>. </t>
 
     <t> Predefined application formats for the UDP protocol with non-RTP
         media are as below: </t>
@@ -1393,7 +1433,7 @@
 
       Specifying the attribute "type:H332" indicates that this
       loosely coupled session is part of a H.332 session as defined
-      in the ITU H.332 specification [H.332].  Media tools should be
+      in the ITU H.332 specification [15].  Media tools should be
       started "recvonly".
 
       Specifying the attribute "type:test" is suggested as a hint
@@ -1452,7 +1492,7 @@
       allowed although NOT RECOMMENDED.
 
       The "sdplang" attribute value must be a single RFC 3066
-      language tag in US-ASCII [RFC3066].  It is not dependent on
+      language tag in US-ASCII [6].  It is not dependent on
       the charset attribute.  An "sdplang" attribute SHOULD be
       specified when a session is of sufficient scope to cross
       geographic boundaries where the language of recipients cannot
@@ -1473,7 +1513,7 @@
       session or media from most important to least important.
 
       The "lang" attribute value must be a single RFC 3066 language
-      tag in US-ASCII [RFC3066].  It is not dependent on the charset
+      tag in US-ASCII [6].  It is not dependent on the charset
       attribute.  A "lang" attribute SHOULD be specified when a
       session is of sufficient scope to cross geographic boundaries
       where the language of recipients cannot be assumed, or where
@@ -1602,9 +1642,9 @@
     appropriate to participate in multimedia sessions, without the user
     first being informed that such software will be started and giving
     their consent.  Thus a session description arriving by session
-    announcement, email, session invitation, or WWW page SHOULD NOT
-    deliver the user into an interactive multimedia session without the
-    user being aware that this will happen.  As it is not always simple
+    announcement, email, session invitation, or WWW page MUST NOT
+    deliver the user into an interactive multimedia session unless the
+    user has explicitly pre-authorized such action.  As it is not always simple
     to tell whether a session is interactive or not, applications that
     are unsure should assume sessions are interactive. </t>
 
@@ -1636,6 +1676,61 @@
 </section>
 <!---------------------------------------------------------------------------->
 <section title="IANA Considerations">
+<section title="The &quot;application/sdp&quot; media type">
+
+<t> One new MIME type is to be registered, as defined below. This updates 
+    the previous definition from RFC 2327. </t>
+
+<figure>
+<artwork>
+
+   To: ietf-types@iana.org
+   Subject: Registration of MIME media type application/sdp
+
+   MIME media type name: application
+
+   MIME subtype name: sdp
+
+   Required parameters: None.
+
+   Optional parameters: None.
+
+   Encoding considerations: 
+      See section 5 of RFC XXXX
+
+   Security considerations:
+      See section 8 of RFC XXXX
+
+   Interoperability considerations:
+      See RFC XXXX
+
+   Published specification:
+      RFC XXXX
+
+   Applications which use this media type:
+      Voice over IP, video teleconferencing, streaming media, instant
+      messaging, etc. See also section 3 of RFC XXXX.
+
+   Additional information:
+
+   Magic number(s):   None.
+   File extension(s): The extension ".sdp" is commonly used.
+   Macintosh File Type Code(s):	
+
+   Person & email address to contact for further information:
+      Colin Perkins &lt;csp@csperkins.org>
+      IETF MMUSIC working group
+
+   Intended usage: COMMON
+
+   Author/Change controller:
+      Authors of RFC XXXX
+      IETF MMUSIC working group
+
+</artwork>
+</figure>
+</section>
+<section title="Registration of Parameters">
 
 <t> There are seven field names that may be registered with IANA. Using
     the terminology in the SDP specification BNF, they are "media",
@@ -1651,16 +1746,16 @@
         a standards-track RFC MUST be produced for a new top-level content
         type to be registered, and the registration MUST provide good
         justification why no existing media name is appropriate (the
-        "Standards Action" policy of RFC 2434 [RFC2434]). </t>
+        "Standards Action" policy of <xref target="RFC2434">RFC 2434</xref>. </t>
 
     </section>
     <section title="Transport protocols (&quot;proto&quot;)">
 
     <t> The "proto" field describes the transport protocol used. This
         SHOULD reference a standards-track protocol RFC.  This memo
-        registers three values: "RTP/AVP" is a reference to RTP [RFC1889]
-        used under the RTP Profile for Audio and Video Conferences with
-        Minimal Control [RFC1890]) running over UDP/IP; "TCP" denotes an
+        registers three values: "RTP/AVP" is a reference to <xref target="RFC3550">RTP</xref>
+        used under the <xref target="RFC3551">RTP Profile for Audio and Video Conferences with
+        Minimal Control</xref> running over UDP/IP; "TCP" denotes an
         unspecified format over TCP; and "udp" indicates an unspecified
         format over UDP. </t>
 
@@ -1804,11 +1899,12 @@
 	will be made. </t>
     </section>
 </section>
+</section>
 <!---------------------------------------------------------------------------->
 <appendix title="SDP Grammar">
 
 <t> This appendix provides an Augmented BNF grammar for SDP.  ABNF is
-    defined in [RFC2234].</t>
+    defined in <xref target="RFC2234"/>.</t>
 
 <figure>
 <artwork><![CDATA[
@@ -1989,7 +2085,7 @@
                          ; range 224.0.0.0 to 239.255.255.255
 
    m1 =                  ("22" ("4"/"5"/"6"/"7"/"8"/"9")) /
-                         ("23" DIGIT ))
+                         ("23" DIGIT )
 
    IP6-multicast =       hexpart [ "/" integer ]
                          ; IPv6 address starting with FF
@@ -2032,10 +2128,6 @@
 
    token-char =          %x21 / %x23-27 / %x2A-2B / %x2D-2E / %x30-39
                          / %x41-5A / %x5E-7E
-                         ; definition from RFC 2045 -
-                         ; "any (US-ASCII) CHAR except SPACE, CTLs,
-                         ; or tspecials".
-                         ; the tspecials are ()<>@,;:
 
    token =               1*(token-char)
 
@@ -2064,70 +2156,6 @@
 </figure>
 </appendix>
 <!---------------------------------------------------------------------------->
-<appendix title="Changes from RFC 2327">
-
-<t>
-<list style="symbols">
-<t> Deprecate X- notation for experimental parameters </t>
-
-<t> Clarify that a=recvonly does NOT mean that you don't send RTCP, and
-    similarly for sendonly and inactive. These only effect the RTP stream. </t>
-
-<t> Rewrite and correct the ABNF syntax (thanks to Jonathan Lennox) </t>
-
-<t> Update BNF to support IPv6. </t>
-
-<t> Add a=inactive attribute. </t>
-
-<t> Add a=maxptime attribute. </t>
-
-<t> RFC 2327 mandated that either e= or p= was required. Both are now
-    optional, to reflect actual usage. </t>
-
-<t> Removed references to "conference" from the description of the t= line,
-    to make it less SAP oriented. </t>
-
-<t> Note about wrap-around of NTP timestamps in t= </t>
-
-<t> References have been updated and split into normative and informative
-    sections. </t>
-
-<t> Section 3.1 was replaced with a reference to RFC 2119, and the memo has
-    been updated to use the RFC 2119 terminology (MUST, SHOULD, etc). </t>
-
-<t> Use of "application/sdp" as MIME a type for SDP files is now "MUST" 
-    rather than "SHOULD". </t>
-
-<t> Many sections have been updated to be less SAP specific, and to reference 
-    other current uses of SDP such as RTSP and SIP. </t>
-
-<t> The introduction and background has been rewritten, to remove references 
-    to the Mbone, reflecting current use of SDP. </t>
-
-<t> The section on concatenation of session descriptions (which was not
-    allowed in SAP, but allowed in other cases) has been removed. It is
-    assumed that transports of SDP specify will specify this. </t>
-
-<t> The description of the c= line has been updated to reflect common usage
-    of SDP, rather than Mbone conferencing with SAP. </t>
-
-<t> The b= line no longer makes a normative reference to the Mbone FAQ for
-    bandwidth limits at various TTLs. The AS modifier to b= is noted as
-    being the RTP session bandwidth. </t>
-
-<t> Define relation between the m= line and MIME types </t>
-
-<t> Note use of s= in sessions with no meaningful name </t>
-
-<t> Allow a=rtpmap to be a session level attribute, in addition to a media
-    level attribute </t>
-
-<t> Clarify the limitations of the k= field </t>
-
-<t> Clarify IANA considerations </t>
-</list></t>
-</appendix>
-<!---------------------------------------------------------------------------->
 <appendix title="Acknowledgments">
 
 <t> Many people in the IETF MMUSIC working group have made comments and
@@ -2144,20 +2172,24 @@
 <?rfc include="reference.RFC.2119.xml"?>
 <?rfc include="reference.RFC.2234.xml"?>
 <?rfc include="reference.RFC.2279.xml"?>
+<?rfc include="reference.RFC.2396.xml"?>
 <?rfc include="reference.RFC.2434.xml"?>
 <?rfc include="reference.RFC.3066.xml"?>
 </references>
 
 <references title="Informative References">
 <?rfc include="reference.RFC.1305.xml"?>
-<?rfc include="reference.RFC.1889.xml"?>
-<?rfc include="reference.RFC.1890.xml"?>
 <?rfc include="reference.RFC.2974.xml"?>
 <?rfc include="reference.RFC.3261.xml"?>
 <?rfc include="reference.RFC.2326.xml"?>
 <?rfc include="reference.RFC.3264.xml"?>
-<?rfc include="reference.I-D.ietf-mmusic-sdp4nat.xml"?>
-<?ref include="reference.ITU.H332.1998.xml"?>
+<?rfc include="reference.RFC.3550.xml"?>
+<?rfc include="reference.RFC.3551.xml"?>
+<?rfc include="reference.RFC.3605.xml"?>
+<?rfc include="reference.ITU.H332.1998.xml"?>
+<?rfc include="reference.I-D.ietf-mmusic-kmgmt-ext.xml"?>
+<?rfc include="reference.I-D.ietf-mmusic-sdescriptions.xml"?>
+
 
 </references>
 </back>

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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.

	Title		: SDP Security Descriptions for Media Streams
	Author(s)	: F. Andreasen, M. Baugher, D. Wing
	Filename	: draft-ietf-mmusic-sdescriptions-02.txt
	Pages		: 36
	Date		: 2003-10-24
	
This document defines a Session Description Protocol (SDP) 
cryptographic attribute for media streams.  The attribute describes 
a cryptographic key and other parameters, which serve to configure 
security for a media stream.  This document defines the Secure Real-
time Transport Protocol (SRTP) parameters for the attribute.  The 
SDP crypto attribute requires the services of a data security 
protocol to secure the SDP message.

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To: "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>,
        Thomas Zeng
	 <zeng@PacketVideo.COM>
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Subject: RE: [MMUSIC] RTSP: Minutes telecon 14th of October 2003
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> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Monday, October 27, 2003 1:00 AM
> To: Thomas Zeng
> Cc: IETF MMUSIC WG; Greg Sherwood; David Kosiba
> Subject: Re: [MMUSIC] RTSP: Minutes telecon 14th of October 2003
> 
> 
> Hi Thomas,
> 
> I don't think there will be any problem with later writing a 
> extension 
> draft resolving the problems and how one likes this to work. 
> Through the 
> use of a feature tag and a the Require header the client can 
> ensure that 
> the server interprets the timed field correctly in such 
> requests. Please 
> note that I am not removing the syntactical element, rather 
> declaring it 
> as shall not be used in the core spec for other purposes then 
> REDIRECT.
> 
> I hope you look over the possible solutions for playlist to 
> find what is 
> a good solution.
> 
> If one take the simple approach only having the desire to delay the 
> start of the playback the problems would not be to hard to solve. 
> However it would be simpler if one haves a separate extension 
> draft for 
> this rather than using the core specification, which I interprets as 
> being your intention.

Yes, indeed. 
> 
> Cheers
> 
> Magnus
> 
> 
> Thomas Zeng wrote:
> 
> > Magnus:
> > 
> > Our use case is best handled by a feature-tag, I think.  
> Let me clarify:
> > even if "time=" is deprecated
> > in Range header in the core spec, we can still "resurrect" 
> it in a extension
> > to RTSP protocol, as long as
> > we address any ambuguity (I believe we can)?
> > 
> > More embedded.
> > 
> > thanks,
> > thomas
> > 
> > 
> >>-----Original Message-----
> >>From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> >>Sent: Wednesday, October 22, 2003 1:33 AM
> >>To: Thomas Zeng
> >>Cc: IETF MMUSIC WG
> >>Subject: Re: [MMUSIC] RTSP: Minutes telecon 14th of October 2003
> >>
> >>
> >>Hi Thomas,
> >>
> >>If you have a use cases for this, I think it is possible to 
> keep the 
> >>timed functionality, under the condition that you provide all 
> >>necessary 
> >>specification text to solve the problems with timed PLAYs.
> > 
> > 
> > We will either propose timed PLAY or timed PLAYLIST_PLAY.
> > 
> >>I would also like to have some more information about use case.
> >>
> > 
> > The use case involves server-side playlist play, where one 
> may desire to
> > have the option to jump to a new
> > item in the playlist, but only at the end of the current item.
> > 
> >>The big problem with the timed requests are the following:
> >>
> >>- Are they allowed to be stacked, are multiple possible to be 
> >>requested 
> >>for later execution? If yes, the real problems exist.
> > 
> > 
> > No, only outstanding timed request is allowed. That is, the 
> stack size is 1.
> > 
> >>- If you use timed requests how does the interaction 
> happens with the 
> >>state machine. When are the state transition made, when the 
> >>request is 
> >>received or when it is executed.
> > 
> > 
> > The first PLAY is always executed immediately. So state 
> transition is no
> > issue.
> > In another words, timed PLAY is only in PLAYING state for 
> clip transition.
> > 
> >>- How is errors at the time of execution handled?
> > 
> > Error will be reported via event report (say via 
> SET_PARAMETER or ANNOUNCE).
> > 
> >>- If multiple requests are allowed, how does one handle if 
> >>two request 
> >>collide.
> > 
> > 
> > Again, only one timed request is allowed.
> > 
> >>- How do you stop a timed request, between acceptance and execution?
> >>
> > 
> > By sending another PLAY or PLAYLIST_PLAY.
> > 
> 
> 
> 
> -- 
> 
> 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] [ rtspspec-Bugs-810678 ] How to indicate end of medi a stream
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Thread-Index: AcOcbzMTe31BP+h3SBOlH2InDnNXeAAQYDNg
From: "Matthew Heaney" <mheaney@on2.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>,
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> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]=20
> Sent: Monday, October 27, 2003 4:46 AM
> To: Tom Marshall
>=20
> 4. Use a RTP extension.
> This solution can either be implemented through a new RTP=20
> payload format=20
> including the necessary information, or through some RTP=20
> header extension.
> - Will go in the media channel as media rather than signalling.
> - Can't be sent in advance for certain media requiring continuos RTP=20
> sequence numbers if it requires its own packets.
> - Is not robust will require that one either uses some type=20
> of ACKs and=20
> adaptive transmission of them or do a repetition accomplishing high=20
> enough delivery probability.
> - The use of header extension would prevent other header extensions.
> - The use of a payload format would either require its own=20
> RTP packet or=20
>   use of RFC 2198.

Other issues:

o What is the rtp timestamp of the EOS indication?

o If the EOS indication increments (rtp timestamps are "monotonically
increasing"), then by how much should the timestamp increment for each
EOS sent?


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Subject: RE: [MMUSIC] [ rtspspec-Bugs-810678 ] How to indicate end of medi a stream
Date: Mon, 27 Oct 2003 14:19:02 -0500
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I suppose I should also bring up the issue of end-of-stream vs.
end-of-play. =20

Perhaps it would be useful for the server to indicate that it's done
playing all the streams implied by the aggregate play request.

(This actually requires some work in the TrueCast case, since each
stream is more or less independent.  Right now I leave it up to the
client to decide when to issue the (aggregate) PAUSE or TEARDOWN request
following receipt of one or more EOS indications.)

-Matt


> -----Original Message-----
> From: Matthew Heaney=20
> Sent: Monday, October 27, 2003 12:37 PM
> To: Magnus Westerlund; Tom Marshall
> Cc: mmusic@ietf.org; zeng@PacketVideo.COM
> Subject: RE: [MMUSIC] [ rtspspec-Bugs-810678 ] How to=20
> indicate end of medi a stream
>=20
>=20
>=20
>=20
> > -----Original Message-----
> > From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> > Sent: Monday, October 27, 2003 4:46 AM
> > To: Tom Marshall
> >=20
> > 4. Use a RTP extension.
> > This solution can either be implemented through a new RTP
> > payload format=20
> > including the necessary information, or through some RTP=20
> > header extension.
> > - Will go in the media channel as media rather than signalling.
> > - Can't be sent in advance for certain media requiring=20
> continuos RTP=20
> > sequence numbers if it requires its own packets.
> > - Is not robust will require that one either uses some type=20
> > of ACKs and=20
> > adaptive transmission of them or do a repetition accomplishing high=20
> > enough delivery probability.
> > - The use of header extension would prevent other header extensions.
> > - The use of a payload format would either require its own=20
> > RTP packet or=20
> >   use of RFC 2198.
>=20
> Other issues:
>=20
> o What is the rtp timestamp of the EOS indication?
>=20
> o If the EOS indication increments (rtp timestamps are=20
> "monotonically increasing"), then by how much should the=20
> timestamp increment for each EOS sent?
>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
>=20

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> -----Original Message-----
> From: Matthew Heaney [mailto:mheaney@on2.com]
> Sent: Monday, October 27, 2003 11:19 AM
> To: Matthew Heaney; Magnus Westerlund; Tom Marshall
> Cc: mmusic@ietf.org; zeng@PacketVideo.COM
> Subject: RE: [MMUSIC] [ rtspspec-Bugs-810678 ] How to 
> indicate end of medi a stream
> 
> 
> I suppose I should also bring up the issue of end-of-stream vs.
> end-of-play.  


> 
> Perhaps it would be useful for the server to indicate that it's done
> playing all the streams implied by the aggregate play request.

Do you imply "end-of-play" should be signalled by RTSP server, then?

> 
> (This actually requires some work in the TrueCast case, since each
> stream is more or less independent.  Right now I leave it up to the
> client to decide when to issue the (aggregate) PAUSE or 
> TEARDOWN request
> following receipt of one or more EOS indications.)
> 
> -Matt
> 
> 
> > -----Original Message-----
> > From: Matthew Heaney 
> > Sent: Monday, October 27, 2003 12:37 PM
> > To: Magnus Westerlund; Tom Marshall
> > Cc: mmusic@ietf.org; zeng@PacketVideo.COM
> > Subject: RE: [MMUSIC] [ rtspspec-Bugs-810678 ] How to 
> > indicate end of medi a stream
> > 
> > 
> > 
> > 
> > > -----Original Message-----
> > > From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> > > Sent: Monday, October 27, 2003 4:46 AM
> > > To: Tom Marshall
> > > 
> > > 4. Use a RTP extension.
> > > This solution can either be implemented through a new RTP
> > > payload format 
> > > including the necessary information, or through some RTP 
> > > header extension.
> > > - Will go in the media channel as media rather than signalling.
> > > - Can't be sent in advance for certain media requiring 
> > continuos RTP 
> > > sequence numbers if it requires its own packets.
> > > - Is not robust will require that one either uses some type 
> > > of ACKs and 
> > > adaptive transmission of them or do a repetition 
> accomplishing high 
> > > enough delivery probability.
> > > - The use of header extension would prevent other header 
> extensions.
> > > - The use of a payload format would either require its own 
> > > RTP packet or 
> > >   use of RFC 2198.
> > 
> > Other issues:
> > 
> > o What is the rtp timestamp of the EOS indication?
> > 
> > o If the EOS indication increments (rtp timestamps are 
> > "monotonically increasing"), then by how much should the 
> > timestamp increment for each EOS sent?
> > 
> > 
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www1.ietf.org/mailman/listinfo/mmusic
> > 
> 

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Subject: RE: [MMUSIC] [ rtspspec-Bugs-810678 ] How to indicate end of medi a stream
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> -----Original Message-----
> From: Thomas Zeng [mailto:zeng@PacketVideo.COM]=20
> Sent: Monday, October 27, 2003 2:24 PM
>=20
> > -----Original Message-----
> > From: Matthew Heaney [mailto:mheaney@on2.com]
> > Sent: Monday, October 27, 2003 11:19 AM
> >=20
> > Perhaps it would be useful for the server to indicate that it's done
> > playing all the streams implied by the aggregate play request.
>=20
> Do you imply "end-of-play" should be signalled by RTSP server, then?

I was thinking that if we treat the original play request as "atomic,"
that perhaps the server should indicate that the entire play request has
finished.  By "finished" I mean all the streams that entered the playing
state as a result of the aggregate play request are done.

If you send an EOS indicator for each stream separately, then it's up to
the client to figure out when to move to another aggregate play state.
Right now he must synthesize this information himself, based on how many
streams were reported back in the RTP-Info of the PLAY response, and how
many different EOS indicators he has received.


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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		: SDP Security Descriptions for Media Streams
	Author(s)	: F. Andreasen, M. Baugher, D. Wing
	Filename	: draft-ietf-mmusic-sdescriptions-02.txt
	Pages		: 36
	Date		: 2003-10-24
	
This document defines a Session Description Protocol (SDP) 
cryptographic attribute for media streams.  The attribute describes 
a cryptographic key and other parameters, which serve to configure 
security for a media stream.  This document defines the Secure Real-
time Transport Protocol (SRTP) parameters for the attribute.  The 
SDP crypto attribute requires the services of a data security 
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From: Thomas Zeng <zeng@PacketVideo.COM>
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	 <magnus.westerlund@ericsson.com>,
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Dear AVT and MMUSIC members:

Last night, I submitted revision 01 of the strawman End-Of-Stream proposal,
which is the action assigned to me during IETF-57.
See forwarded email.

In this revision, I added "Appendix A" to describe a new RTCP packet for
singalling EOS: the RTCP EOS packet.
My intention is to  present both RTSP EOS and RTCP EOS, and open to the
group to debate whether one or both of them (or none
of them) should be standardized.

Note that in my view defining a special RTP packet is inferior to new RTCP
EOS packet, because RTCP is more suitable for control purposes.
Therefor I didn't present an RTP based EOS solution. The final decision is
up to the group.

As always, your comments are highly appreciated.

Given the differing opinions on EOS signalling, I'd like to request a 15
minutes time slot during MMUSIC session on Nov.14 to
present these alternatives to the group. I will send a separate email to
MMUSIC chairs shortly.

Another note:
   In Appendix A, the sequence number of the ending RTP packet is given for
each SSRC/CSRC in the new RTCP EOS packet. 
It appears to be enough and I don't see a need to also put RTP time of the
ending RTP timestamp in the RTCP EOS packet.

AVT mailing list is CC'ed since if we opt for a new RTCP EOS packet, AVT
would be the WG to get approval.

Thanks,
- thomas


>  -----Original Message-----
> From: 	Thomas Zeng  
> Sent:	Sunday, October 26, 2003 10:34 PM
> To:	'internet-drafts@ietf.org'
> Subject:	draft-zeng-mmusic-rtsp-end-of-stream-01.txt
> 
>  <<draft-zeng-mmusic-rtsp-end-of-stream-01.txt>> 

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=0A=
MMUSIC                                                     Thomas M. =
Zeng=0A=
                                                         P. Greg =
Sherwood=0A=
Internet-Draft                                                =
PacketVideo=0A=
Expires: April 29, 2004                                     Oct  26, =
2003=0A=
=0A=
=0A=
              Signalling End Of Stream in Real Time Streaming=0A=
                draft-zeng-mmusic-end-of-stream-01.txt=0A=
=0A=
Status of this Memo=0A=
=0A=
   This document is an Internet-Draft and is in full conformance =
with=0A=
   all provisions of Section 10 of RFC2026.=0A=
=0A=
   Internet-Drafts are working documents of the Internet Engineering=0A=
   Task Force (IETF), its areas, and its working groups. Note that =
other=0A=
   groups may also distribute working documents as Internet-Drafts.=0A=
=0A=
   Internet-Drafts are draft documents valid for a maximum of six =
months=0A=
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any=0A=
   time. It is inappropriate to use Internet-Drafts as reference=0A=
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=0A=
   The list of current Internet-Drafts can be accessed at http://=0A=
   www.ietf.org/ietf/1id-abstracts.txt.=0A=
=0A=
   The list of Internet-Draft Shadow Directories can be accessed at=0A=
   http://www.ietf.org/shadow.html.=0A=
=0A=
   This Internet-Draft will expire on Apr. 29, 2004.=0A=
=0A=
Copyright Notice=0A=
=0A=
   Copyright (C) The Internet Society (2003). All Rights Reserved.=0A=
=0A=
Abstract=0A=
=0A=
   This document describes a new RTSP method called END_OF_STREAM, =
which =0A=
   extends the core RTSP protocol to enable one RTSP protocol entity to =
=0A=
   signal other entity(s) =0A=
   that it has no more media data to send. This request could be issued =
=0A=
   either with a =0A=
   non-aggregate or aggregate control URI, the latter implies that all =
=0A=
   media streams =0A=
   in the aggregate session have run out of media data.=0A=
=0A=
   The receiver of this RTSP request is expected to reply with 200 OK =
=0A=
   response.=0A=
=0A=
   Since END_OF_STREAM represents an extension to the core RTSP =
protocol, =0A=
   a feature tag,=0A=
   "method.eos", is defined to facilitate capability negotiation using =
=0A=
   RTSP extension =0A=
   mechanism [RTSP_NEW].=0A=
=0A=
=0A=
1. Motivation=0A=
=0A=
   In the core RTSP protocol [RTSP_NEW], an RTSP client relies on =
the=0A=
   media transport mechanism  to signal end of stream.=0A=
=0A=
   When the media transport mechanism happens to be RTP over UDP, this =
=0A=
   is carried out by RTCP BYE packet [RTP_NEW]. In practice, there are =
=0A=
   some drawbacks with this approach:=0A=
     =0A=
     1. When the server sends an RTCP BYE packet with its SSRC, the =0A=
        server is giving up=0A=
        the SSRC (see section 8.2 in [RTP_NEW]). The server would be =
=0A=
        required to =0A=
        switch to a new SSRC on a subsequent PLAY of the same media =0A=
        stream.=0A=
        Since server's SSRC is only communicated in the Transport =
header =0A=
        of SETUP =0A=
        response, the server would not have an opportunity to send a =
new=0A=
        value to =0A=
        the player, and the client would have to discover the SSRC from =
=0A=
        the incoming RTP packets. =0A=
=0A=
     2. RTCP BYE packet method does not offer a simple, gurranteed =0A=
        method of delievering=0A=
        an end-of-stream announcement.=0A=
=0A=
     3. RTCP BYE packet method does not offer the option to have a =
single=0A=
        aggregate=0A=
        end-of-stream announcemnt for all media streams in the RTSP =0A=
        session.=0A=
=0A=
     4. Section 6.3.7 of RFC3550 stipulates that an RTP sender cannot =
=0A=
        send RTCP BYE =0A=
        before leaving the RTP session if it has not already sent at =
=0A=
        least one RTP or RTCP packet. This is a problem under =0A=
        error conditions. Consider the case =0A=
        where an RTP session has just started (i.e., RTSP PLAY has =0A=
        been =0A=
        successfully acknowledged with an RTSP 200 OK response), and =
the =0A=
        sender attempts to =0A=
        retrieve media frames from its media source. The media source =
=0A=
        fails to provide any media frame due to its internal error such =
=0A=
        as file corruption. The sender should inform its receiver(s) =
=0A=
        but it cannot send BYE packets.=0A=
=0A=
   The motivation to solve the above issues is particularly high for =
=0A=
   unicast streaming applications that use RTSP over TCP in the control =
=0A=
   plane, and RTP over UDP in the media transport. =0A=
=0A=
   There is also the desire to have an EOS (End Of Stream) =0A=
   signalling mechanism =0A=
   for non-RTP delivery. One such delivery is MPEG2 transport streams =
=0A=
   used in the cable TV environment. In non-IP delivery environments, =
=0A=
   the transport typically remains allocated even if no media is being =
=0A=
   delivered.  This=0A=
   means that a client cannot watch for the server to close the=0A=
   transport to signal the end of stream. Meanwhile,  watching for the =
=0A=
   incoming media to stop is unreliable.  Short=0A=
   timeouts can trigger a false end of media detection if the=0A=
   media flow is temporarily delayed.  Long timeouts introduce=0A=
   unacceptable latencies.  Clients are unable to distinguish=0A=
   between a normal end of stream and an error condition that=0A=
   resulted in the media delivery stopping.=0A=
=0A=
   We note that using TEARDOWN from server to client is not =0A=
   appropriate because:=0A=
     1. TEARDOWN is currently not allowed from server to =0A=
        client [RTSP_NEW];=0A=
     2. Even if TEARDOWN is made available in server to client =0A=
        direction, =0A=
        the definition of TEARDOWN requires that, if the request =0A=
        URI is =0A=
        aggregate, that the session must be de-allocated by the =0A=
        server. =0A=
        There are RTSP applications that use SET_PARAMETER from =0A=
        client to=0A=
        server as the means to report session QoS statistics, but if=0A=
        server uses TEARDOWN on aggregate URL to signal end of stream, =
=0A=
        the client can no longer use SET_PARAMETER with a =0A=
        session header. =0A=
     3. In general, RTSP, being a client-server protocol,=0A=
        should let client, not server to control session state. But =0A=
        TEARDOWN=0A=
        on aggregate URL will change session from PLAYING state =0A=
        to INIT state.=0A=
=0A=
   We note that using PAUSE from server to client is not appropriate =
=0A=
   either, because PAUSE will change the state of the RTSP session.=0A=
=0A=
   We note also that using ANNOUNCE from server to client is not =0A=
   suitable, for two reasons:=0A=
      1. ANNOUNCE from server to client is used in RFC2326 to update =
=0A=
         session description. =0A=
         Extending it to indicate end-of-stream event is awkward.=0A=
      2. ANNOUNCE is not in the core RTSP protocol [RTSP_NEW].=0A=
=0A=
   There has been proposals to extend RTP or RTCP protocol to =
introduce=0A=
   in-band signalling of end-of-stream events. Appendex A explains=0A=
   one such proposal. We note that in practice both approaches =0A=
   (using an RTSP method or a special RTP/RTCP packet to signal=0A=
    end-of-stream) have shown up in propriatery implemtations. =0A=
   Therefore it is our view that allowing both approaches =0A=
   is a reasonable compromise until a concensus is reached.   =0A=
=0A=
   The next section describes our solution.=0A=
=0A=
3. Solution Using END_OF_STREAM method=0A=
=0A=
   We introduce a new RTSP method to signal end of stream in =
applications=0A=
   where an persistent RTSP connection is maintained between server =0A=
   and client.=0A=
   The new method is an RTSP "extension-method" [RTSP_NEW]. It is =
called=0A=
   END_OF_STREAM, and is a request from media sender=0A=
   to receiver. In the case of RTSP client playing media streams sent =
by =0A=
   RTSP server,=0A=
   the END_OF_STREAM is sent from server to client. In the use case =
where =0A=
   RTSP server is=0A=
   recording media sent by RTSP client, the END_OF_STREAM requet could =
=0A=
   be sent from an=0A=
   RTSP client to server.  =0A=
=0A=
   As any other RTSP request method, END_OF_STREAM must be acknowledged =
=0A=
   with a response.=0A=
   =0A=
   3.1 The Definition of END_OF_STREAM=0A=
      =0A=
   The END_OF_STREAM request for a URI signals the fact that no more =
=0A=
   RTP packets will be=0A=
   sent. Here is an example:=0A=
=0A=
     S->C: END_OF_STREAM rtsp://foo.com/bar.avi RTSP/1.0=0A=
           CSeq: 10=0A=
           Session: 12345678=0A=
	   Range: npt=3D0-200=0A=
           RTP-Info: =
url=3D//foo.com/bar.avi/streamid=3D0;seq=3D45102,=0A=
             url=3Drtsp://foo.com/bar.avi/streamid=3D1;seq=3D30211    =
=0A=
	   Reason: End of range reached=0A=
=0A=
     C->S: RTSP/1.0 200 OK  =0A=
           CSeq: 10=0A=
           Session: 12345678=0A=
    =0A=
   An END_OF_STREAM request MUST include "CSeq", "Range" and "Session" =
=0A=
   headers.=0A=
   It SHOULD include "RTP-Info" header.=0A=
   The RTP-Info in server's END_OF_STREAM request=0A=
   is used to indicate the sequence number of =0A=
   the ending RTP packet for each media stream.=0A=
   =0A=
   An END_OF_STREAM requet MAY include a new "Reason" header, =0A=
   defined as a =0A=
   string, whose purpose is to allow the server to explain why stream =
=0A=
   has ended, and whose ABNF definition is given below:=0A=
    =0A=
          Reason     =3D  "Reason" ":"   Reason-Phrase CRLF=0A=
=0A=
   where:=0A=
       -- "Reason-Phrase" is a parameter defined in section 7.1.1 of =
=0A=
          [RTSP_NEW].=0A=
=0A=
   =0A=
   Note that the server is free to use any text as "Reason-Phrase". In =
=0A=
   certain applications, the client=0A=
   may  display the text, in its entirety, to end user to improve =
user=0A=
   friendliness.=0A=
      =0A=
   END_OF_STREAM does not remove session ID, nor does it change the =0A=
   state of the RTSP session.=0A=
=0A=
   END_OF_STREAM may be applied to either aggregate or non-aggregate =
URI.=0A=
   =0A=
   END_OF_STREAM does not apply if the RTSP session is not in =
PLAYING=0A=
   state. If a client receives END_OF_STREAM while it is not in PLAYING =
=0A=
   state, it must reply with a 455 response: Method Not Valid in This =
=0A=
   State.=0A=
=0A=
   We note that client can piggy back QoS statistics as a entity body =
in=0A=
   its 200 OK response to the END_OF_STREAM request from the server.   =
=0A=
=0A=
   3.2 Limitations=0A=
   =0A=
   END_OF_STREAM method is applicable only if the server has the =
means=0A=
   to contact the client when it reaches the end of a media stream.=0A=
   This may not be possible if the RTSP connection between server =
and=0A=
   client is not persistent. When it is time to send END_OF_STREAM =
but=0A=
   the server does not have connection to the client, the server =
will=0A=
   simply skip END_OF_STREAM. That is to say, the server will not =0A=
   queue up the END_OF_STREAM message.=0A=
   =0A=
=0A=
4. Feature tag=0A=
  =0A=
   The END_OF_STREAM feature is represented by a feature tag (see =0A=
   [RTSP_NEW]=0A=
   for how to use feature tags in capability negotiations).=0A=
=0A=
   The feature tag is:=0A=
=0A=
          method.eos=0A=
=0A=
   which means that the END_OF_STREAM method is fully supported. By =0A=
   fully we mean either at non-aggregate or aggregate level.=0A=
=0A=
5. Use Case=0A=
=0A=
   In this example, the server sends END_OF_STREAM request to client =
for=0A=
   one live media stream, because upstream source terminates the =
stream=0A=
   after 200 seconds. The fact that the stream has played for 200 =
seconds=0A=
   is communicated by the Range header in the END_OF_STREAM request.=0A=
=0A=
     C->S: PLAY rtsp://foo.com/bar.avi/streamid=3D0 RTSP/1.0=0A=
	   Supported: method.eos=0A=
           CSeq: 10=0A=
           Session: 12345678=0A=
	   Range: 0-200=0A=
=0A=
     S->C: RTSP/1.0 200 OK=0A=
	   Supported: method.eos=0A=
           CSeq: 10=0A=
           Session: 12345678=0A=
           RTP-Info: url=3D//foo.com/bar.avi/streamid=3D0;seq=3D0; =
rtptime=3D0 =0A=
=0A=
     S->C: END_OF_STREAM rtsp://foo.com/bar.avi/streamid=3D0 =
RTSP/1.0=0A=
           CSeq: 123=0A=
           Session: 12345678=0A=
	   Range: npt=3D0-200; bytes=3D0-200000=0A=
           RTP-Info: url=3D//foo.com/bar.avi/streamid=3D0;seq=3D45102 =
=0A=
	   Reason: End of range reached=0A=
=0A=
     C->S: RTSP/1.0 200 OK  =0A=
           CSeq: 123=0A=
           Session: 12345678=0A=
  =0A=
   In the END_OF_STREAM request, the client will learn that the server =
has =0A=
   completed the stream as requested (as indicated by the "End of range =
=0A=
   reached" Reason).=0A=
=0A=
   From the two RTP-Info headers, one in PLAY response, one in =0A=
   END_OF_STREAM requst, the client can derive the total number=0A=
   of RTP packets that the server has sent. From the npt field in the =
=0A=
   Range header, the client knows the presentation time that this =
stream =0A=
   has covered, from the server's point of view, while the byte-range =
field =0A=
   can optional inform the =0A=
   client how much media data, in bytes, that the server has sent. =0A=
 =0A=
5. Security Considerations=0A=
 =0A=
   Because there is only one new TEXT header, "Reason", added by our =
=0A=
   new RTSP method,=0A=
   the security considerations outlined in [RTSP_NEW] apply here as =
well.=0A=
     =0A=
6. IANA Considerations=0A=
=0A=
   A new method name, and its associated feature tag need to be =
registered =0A=
   with IANA.=0A=
    =0A=
     -- Method name: END_OF_STREAM=0A=
     =0A=
     -- feature tag: method.eos=0A=
=0A=
=0A=
=0C=0A=
=0A=
Appendix A: Signalling End-Of-Streaming With RTCP EOD Packet=0A=
=0A=
In this appendix, a new RTCP packet type is defined to signal=0A=
end-of-stream event. =0A=
=0A=
A new RTCP packet type is prefered over profile specific extensions=0A=
(see section 6.4.1 of  [RTP-NEW])=0A=
because this feature is profile independent.=0A=
=0A=
1. EOS: End-Of-Stream RTCP Packet=0A=
=0A=
       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 =
1=0A=
      =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+=0A=
      |V=3D2|P|    SC   |   PT=3DEOS=3D205  |             length        =
    |=0A=
      =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+=0A=
      |                           SSRC/CSRC                           =
|=0A=
      =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+=0A=
      |        extended RTP sequence number                           | =
=0A=
      =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+=0A=
      :                              ...                              =
:=0A=
      =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D=
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=0A=
(opt) |     length    |               reason for EOS                =
...=0A=
      =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+=0A=
=0A=
  The EOS packet indicates that one or more sources have exhausted =
its=0A=
  media data and will not send any new RTP packet, yet the source is =
=0A=
  not about to leave the session.=0A=
=0A=
  packet type (PT):  8 bits=0A=
     Contains the constant 205 to idnentify this as an RTCP EOS =
packet.=0A=
 =0A=
  source count (SC): 5 bits=0A=
     The number of SSRC/CSRC identifiers included in this EOS =
packet.=0A=
     A count value of zero is valid, but useless. For each one of =0A=
     them, an extended RTP sequence number is included. See below.=0A=
=0A=
  extended RTP sequence number: 32 bit=0A=
    The extended sequence number of the ending RTP packet from the =0A=
    source.=0A=
    The low 16 bits contain the sequence number of the ending RTP=0A=
    data packet from source SSRC_n, and the most significant 16=0A=
    bits extend that sequence number with the corresponding count of=0A=
    sequence number cycles. The RTP receiver needs to track the=0A=
    extended sequence number to properly generate RTCP=0A=
    RR packets, hence it can easily determine which RTP packet is =
the=0A=
    ending RTP packet.=0A=
=0A=
2. Rules of sending RTCP EOS packet=0A=
=0A=
  RTCP EOS packet should be sent if the source runs out of media =
data,=0A=
  but is NOT about to leave the session. It may, for instance, wait=0A=
  for the receiver to send anther RTSP PLAY request (if RTSP is the=0A=
  session contorl protocol in use) and then re-use the SSRC to send=0A=
  RTP packets.=0A=
=0A=
  If an EOS packet is received by a mixer, the mixer SHOULD forward=0A=
  the EOS packet with its SSRC/CSRC identifier(s) unchanged.=0A=
=0A=
  Optionally,  the EOS packet MAY include an 8-bit octet count =0A=
  followed by that many=0A=
  octets of text indicating the reason for ending, e.g., "End of file" =
=0A=
  or "Upstream EOS received".  The string has the same=0A=
  encoding as that described for SDES.  If the string fills the =
packet=0A=
  to the next 32-bit boundary, the string is not null terminated.  =
If=0A=
  not, the EOS packet MUST be padded with null octets to the next =
32-=0A=
  bit boundary.  This padding is separate from that indicated by the =
P=0A=
  bit in the RTCP header.=0A=
  =0A=
  When RTP is carried over un-reliable transport (e.g., UDP), an RTP =
=0A=
  source may repeatedly send RTCP EOS packets until it is assured =
that=0A=
  the receiver(s) have got the event notification -- e.g., it has=0A=
  received another RTSP PLAY request from the RTP receiver.  =0A=
=0A=
^L=0A=
Normative References=0A=
=0A=
   [RTSP_NEW] Schulzrinne, H., Rao, A., Lanphier, R., Westerlund, M., =
=0A=
         "Real Time Streaming Protocol", =0A=
         draft-ietf-mmusic-rfc2326bis-04.txt=0A=
=0A=
   [RTP_NEW] RFC3550 "Real-time Transport Protocol", July 2003=0A=
=0A=
=0A=
Full Copyright Statement=0A=
=0A=
   Copyright (C) The Internet Society (2003). All Rights Reserved.=0A=
=0A=
   This document and translations of it may be copied and furnished =
to=0A=
   others, and derivative works that comment on or otherwise explain =
it=0A=
   or assist in its implementation may be prepared, copied, =
published=0A=
   and distributed, in whole or in part, without restriction of any=0A=
   kind, provided that the above copyright notice and this paragraph =
are=0A=
   included on all such copies and derivative works. However, this=0A=
   document itself may not be modified in any way, such as by =
removing=0A=
   the copyright notice or references to the Internet Society or =
other=0A=
   Internet organizations, except as needed for the purpose of=0A=
   developing Internet standards in which case the procedures for=0A=
   copyrights defined in the Internet Standards process must be=0A=
   followed, or as required to translate it into languages other =
than=0A=
   English.=0A=
=0A=
   The limited permissions granted above are perpetual and will not =
be=0A=
   revoked by the Internet Society or its successors or assignees.=0A=
=0A=
   This document and the information contained herein is provided on =
an=0A=
   "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET =
ENGINEERING=0A=
   TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, =
INCLUDING=0A=
   BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION=0A=
=0A=
=0A=
=0A=
=0A=
=0A=
   HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF=0A=
   MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.=0A=
=0A=
=0A=
Acknowledgement=0A=
=0A=
   Thanks to Sean Sheedy for suggesting the inclusion of "Range" =
header=0A=
   and for contributing some text in the motivations section.=0A=
=0A=
   Funding for the RFC Editor function is currently provided by the=0A=
   Internet Society.=0A=
=0A=
Author Addresses=0A=
=0A=
   Thomas Zeng=0A=
   PacketVideo Corp.=0A=
   10350 Science Center Dr., Suite 210=0A=
   San Diego, CA 92127=0A=
   email: zeng@pv.com=0A=
=0A=
   P. Greg Sherwood=0A=
   PacketVideo Corp.=0A=
   10350 Science Center Dr., Suite 210=0A=
   San Diego, CA 92127=0A=
   email: sherwood@pv.com
------_=_NextPart_000_01C39D0A.68A03730--

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Subject: RE: [MMUSIC] [ rtspspec-Bugs-810678 ] How to indicate end of medi a stream
Date: Tue, 28 Oct 2003 11:02:21 +0200
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Thread-Topic: [MMUSIC] [ rtspspec-Bugs-810678 ] How to indicate end of medi a stream
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From: "Amir Wolf" <Amir.Wolf@emblaze.com>
To: "Matthew Heaney" <mheaney@on2.com>, "Thomas Zeng" <zeng@PacketVideo.COM>,
        "Magnus Westerlund" <magnus.westerlund@ericsson.com>,
        "Tom Marshall" <tmarshall@real.com>
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Hi All,
I think that signaling EOS per stream has more value since one of the =
most helpful uses of this feature is when you play a very low fps video =
e.g. average of 1/4 fps and audio.
It is important for the player to know when video ends, while it can =
easily identify end of session by getting the required duration.=20

> -----Original Message-----
> From: mmusic-admin@ietf.org [mailto:mmusic-admin@ietf.org]On=20
> Behalf Of Matthew Heaney
> Sent: Monday, October 27, 2003 10:19 PM
> To: Thomas Zeng; Magnus Westerlund; Tom Marshall
> Cc: mmusic@ietf.org
> Subject: RE: [MMUSIC] [ rtspspec-Bugs-810678 ] How to=20
> indicate end of medi a stream
>=20
>=20
>=20
>=20
> > -----Original Message-----
> > From: Thomas Zeng [mailto:zeng@PacketVideo.COM]=20
> > Sent: Monday, October 27, 2003 2:24 PM
> >=20
> > > -----Original Message-----
> > > From: Matthew Heaney [mailto:mheaney@on2.com]
> > > Sent: Monday, October 27, 2003 11:19 AM
> > >=20
> > > Perhaps it would be useful for the server to indicate=20
> that it's done
> > > playing all the streams implied by the aggregate play request.
> >=20
> > Do you imply "end-of-play" should be signalled by RTSP server, then?
>=20
> I was thinking that if we treat the original play request as "atomic,"
> that perhaps the server should indicate that the entire play=20
> request has
> finished.  By "finished" I mean all the streams that entered=20
> the playing
> state as a result of the aggregate play request are done.
>=20
> If you send an EOS indicator for each stream separately, then=20
> it's up to
> the client to figure out when to move to another aggregate play state.
> Right now he must synthesize this information himself, based=20
> on how many
> streams were reported back in the RTP-Info of the PLAY=20
> response, and how
> many different EOS indicators he has received.
>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
>=20

_______________________________________________
mmusic mailing list
mmusic@ietf.org
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [MMUSIC] Re: FW: draft-zeng-mmusic-rtsp-end-of-stream-01.txt
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Hi,

If you have comments on this, lets discuss this on the MMUSIC list. So 
no future cross postings.

Cheers

Magnus

Thomas Zeng wrote:

> Dear AVT and MMUSIC members:
> 
> Last night, I submitted revision 01 of the strawman End-Of-Stream proposal,
> which is the action assigned to me during IETF-57.
> See forwarded email.
> 
> In this revision, I added "Appendix A" to describe a new RTCP packet for
> singalling EOS: the RTCP EOS packet.
> My intention is to  present both RTSP EOS and RTCP EOS, and open to the
> group to debate whether one or both of them (or none
> of them) should be standardized.
> 
> Note that in my view defining a special RTP packet is inferior to new RTCP
> EOS packet, because RTCP is more suitable for control purposes.
> Therefor I didn't present an RTP based EOS solution. The final decision is
> up to the group.
> 
> As always, your comments are highly appreciated.
> 
> Given the differing opinions on EOS signalling, I'd like to request a 15
> minutes time slot during MMUSIC session on Nov.14 to
> present these alternatives to the group. I will send a separate email to
> MMUSIC chairs shortly.
> 
> Another note:
>    In Appendix A, the sequence number of the ending RTP packet is given for
> each SSRC/CSRC in the new RTCP EOS packet. 
> It appears to be enough and I don't see a need to also put RTP time of the
> ending RTP timestamp in the RTCP EOS packet.
> 
> AVT mailing list is CC'ed since if we opt for a new RTCP EOS packet, AVT
> would be the WG to get approval.
> 
> Thanks,
> - thomas
> 
> 
> 
>> -----Original Message-----
>>From: 	Thomas Zeng  
>>Sent:	Sunday, October 26, 2003 10:34 PM
>>To:	'internet-drafts@ietf.org'
>>Subject:	draft-zeng-mmusic-rtsp-end-of-stream-01.txt
>>
>> <<draft-zeng-mmusic-rtsp-end-of-stream-01.txt>> 
> 
> 
> 
> ------------------------------------------------------------------------
> 
> 
> MMUSIC                                                     Thomas M. Zeng
>                                                          P. Greg Sherwood
> Internet-Draft                                                PacketVideo
> Expires: April 29, 2004                                     Oct  26, 2003
> 
> 
>               Signalling End Of Stream in Real Time Streaming
>                 draft-zeng-mmusic-end-of-stream-01.txt
> 
> Status of this Memo
> 
>    This document is an Internet-Draft and is in full conformance with
>    all provisions of Section 10 of RFC2026.
> 
>    Internet-Drafts are working documents of the Internet Engineering
>    Task Force (IETF), its areas, and its working groups. Note that other
>    groups may also distribute working documents as Internet-Drafts.
> 
>    Internet-Drafts are draft documents valid for a maximum of six months
>    and may be updated, replaced, or obsoleted by other documents at any
>    time. It is inappropriate to use Internet-Drafts as reference
>    material or to cite them other than as "work in progress."
> 
>    The list of current Internet-Drafts can be accessed at http://
>    www.ietf.org/ietf/1id-abstracts.txt.
> 
>    The list of Internet-Draft Shadow Directories can be accessed at
>    http://www.ietf.org/shadow.html.
> 
>    This Internet-Draft will expire on Apr. 29, 2004.
> 
> Copyright Notice
> 
>    Copyright (C) The Internet Society (2003). All Rights Reserved.
> 
> Abstract
> 
>    This document describes a new RTSP method called END_OF_STREAM, which 
>    extends the core RTSP protocol to enable one RTSP protocol entity to 
>    signal other entity(s) 
>    that it has no more media data to send. This request could be issued 
>    either with a 
>    non-aggregate or aggregate control URI, the latter implies that all 
>    media streams 
>    in the aggregate session have run out of media data.
> 
>    The receiver of this RTSP request is expected to reply with 200 OK 
>    response.
> 
>    Since END_OF_STREAM represents an extension to the core RTSP protocol, 
>    a feature tag,
>    "method.eos", is defined to facilitate capability negotiation using 
>    RTSP extension 
>    mechanism [RTSP_NEW].
> 
> 
> 1. Motivation
> 
>    In the core RTSP protocol [RTSP_NEW], an RTSP client relies on the
>    media transport mechanism  to signal end of stream.
> 
>    When the media transport mechanism happens to be RTP over UDP, this 
>    is carried out by RTCP BYE packet [RTP_NEW]. In practice, there are 
>    some drawbacks with this approach:
>      
>      1. When the server sends an RTCP BYE packet with its SSRC, the 
>         server is giving up
>         the SSRC (see section 8.2 in [RTP_NEW]). The server would be 
>         required to 
>         switch to a new SSRC on a subsequent PLAY of the same media 
>         stream.
>         Since server's SSRC is only communicated in the Transport header 
>         of SETUP 
>         response, the server would not have an opportunity to send a new
>         value to 
>         the player, and the client would have to discover the SSRC from 
>         the incoming RTP packets. 
> 
>      2. RTCP BYE packet method does not offer a simple, gurranteed 
>         method of delievering
>         an end-of-stream announcement.
> 
>      3. RTCP BYE packet method does not offer the option to have a single
>         aggregate
>         end-of-stream announcemnt for all media streams in the RTSP 
>         session.
> 
>      4. Section 6.3.7 of RFC3550 stipulates that an RTP sender cannot 
>         send RTCP BYE 
>         before leaving the RTP session if it has not already sent at 
>         least one RTP or RTCP packet. This is a problem under 
>         error conditions. Consider the case 
>         where an RTP session has just started (i.e., RTSP PLAY has 
>         been 
>         successfully acknowledged with an RTSP 200 OK response), and the 
>         sender attempts to 
>         retrieve media frames from its media source. The media source 
>         fails to provide any media frame due to its internal error such 
>         as file corruption. The sender should inform its receiver(s) 
>         but it cannot send BYE packets.
> 
>    The motivation to solve the above issues is particularly high for 
>    unicast streaming applications that use RTSP over TCP in the control 
>    plane, and RTP over UDP in the media transport. 
> 
>    There is also the desire to have an EOS (End Of Stream) 
>    signalling mechanism 
>    for non-RTP delivery. One such delivery is MPEG2 transport streams 
>    used in the cable TV environment. In non-IP delivery environments, 
>    the transport typically remains allocated even if no media is being 
>    delivered.  This
>    means that a client cannot watch for the server to close the
>    transport to signal the end of stream. Meanwhile,  watching for the 
>    incoming media to stop is unreliable.  Short
>    timeouts can trigger a false end of media detection if the
>    media flow is temporarily delayed.  Long timeouts introduce
>    unacceptable latencies.  Clients are unable to distinguish
>    between a normal end of stream and an error condition that
>    resulted in the media delivery stopping.
> 
>    We note that using TEARDOWN from server to client is not 
>    appropriate because:
>      1. TEARDOWN is currently not allowed from server to 
>         client [RTSP_NEW];
>      2. Even if TEARDOWN is made available in server to client 
>         direction, 
>         the definition of TEARDOWN requires that, if the request 
>         URI is 
>         aggregate, that the session must be de-allocated by the 
>         server. 
>         There are RTSP applications that use SET_PARAMETER from 
>         client to
>         server as the means to report session QoS statistics, but if
>         server uses TEARDOWN on aggregate URL to signal end of stream, 
>         the client can no longer use SET_PARAMETER with a 
>         session header. 
>      3. In general, RTSP, being a client-server protocol,
>         should let client, not server to control session state. But 
>         TEARDOWN
>         on aggregate URL will change session from PLAYING state 
>         to INIT state.
> 
>    We note that using PAUSE from server to client is not appropriate 
>    either, because PAUSE will change the state of the RTSP session.
> 
>    We note also that using ANNOUNCE from server to client is not 
>    suitable, for two reasons:
>       1. ANNOUNCE from server to client is used in RFC2326 to update 
>          session description. 
>          Extending it to indicate end-of-stream event is awkward.
>       2. ANNOUNCE is not in the core RTSP protocol [RTSP_NEW].
> 
>    There has been proposals to extend RTP or RTCP protocol to introduce
>    in-band signalling of end-of-stream events. Appendex A explains
>    one such proposal. We note that in practice both approaches 
>    (using an RTSP method or a special RTP/RTCP packet to signal
>     end-of-stream) have shown up in propriatery implemtations. 
>    Therefore it is our view that allowing both approaches 
>    is a reasonable compromise until a concensus is reached.   
> 
>    The next section describes our solution.
> 
> 3. Solution Using END_OF_STREAM method
> 
>    We introduce a new RTSP method to signal end of stream in applications
>    where an persistent RTSP connection is maintained between server 
>    and client.
>    The new method is an RTSP "extension-method" [RTSP_NEW]. It is called
>    END_OF_STREAM, and is a request from media sender
>    to receiver. In the case of RTSP client playing media streams sent by 
>    RTSP server,
>    the END_OF_STREAM is sent from server to client. In the use case where 
>    RTSP server is
>    recording media sent by RTSP client, the END_OF_STREAM requet could 
>    be sent from an
>    RTSP client to server.  
> 
>    As any other RTSP request method, END_OF_STREAM must be acknowledged 
>    with a response.
>    
>    3.1 The Definition of END_OF_STREAM
>       
>    The END_OF_STREAM request for a URI signals the fact that no more 
>    RTP packets will be
>    sent. Here is an example:
> 
>      S->C: END_OF_STREAM rtsp://foo.com/bar.avi RTSP/1.0
>            CSeq: 10
>            Session: 12345678
> 	   Range: npt=0-200
>            RTP-Info: url=//foo.com/bar.avi/streamid=0;seq=45102,
>              url=rtsp://foo.com/bar.avi/streamid=1;seq=30211    
> 	   Reason: End of range reached
> 
>      C->S: RTSP/1.0 200 OK  
>            CSeq: 10
>            Session: 12345678
>     
>    An END_OF_STREAM request MUST include "CSeq", "Range" and "Session" 
>    headers.
>    It SHOULD include "RTP-Info" header.
>    The RTP-Info in server's END_OF_STREAM request
>    is used to indicate the sequence number of 
>    the ending RTP packet for each media stream.
>    
>    An END_OF_STREAM requet MAY include a new "Reason" header, 
>    defined as a 
>    string, whose purpose is to allow the server to explain why stream 
>    has ended, and whose ABNF definition is given below:
>     
>           Reason     =  "Reason" ":"   Reason-Phrase CRLF
> 
>    where:
>        -- "Reason-Phrase" is a parameter defined in section 7.1.1 of 
>           [RTSP_NEW].
> 
>    
>    Note that the server is free to use any text as "Reason-Phrase". In 
>    certain applications, the client
>    may  display the text, in its entirety, to end user to improve user
>    friendliness.
>       
>    END_OF_STREAM does not remove session ID, nor does it change the 
>    state of the RTSP session.
> 
>    END_OF_STREAM may be applied to either aggregate or non-aggregate URI.
>    
>    END_OF_STREAM does not apply if the RTSP session is not in PLAYING
>    state. If a client receives END_OF_STREAM while it is not in PLAYING 
>    state, it must reply with a 455 response: Method Not Valid in This 
>    State.
> 
>    We note that client can piggy back QoS statistics as a entity body in
>    its 200 OK response to the END_OF_STREAM request from the server.   
> 
>    3.2 Limitations
>    
>    END_OF_STREAM method is applicable only if the server has the means
>    to contact the client when it reaches the end of a media stream.
>    This may not be possible if the RTSP connection between server and
>    client is not persistent. When it is time to send END_OF_STREAM but
>    the server does not have connection to the client, the server will
>    simply skip END_OF_STREAM. That is to say, the server will not 
>    queue up the END_OF_STREAM message.
>    
> 
> 4. Feature tag
>   
>    The END_OF_STREAM feature is represented by a feature tag (see 
>    [RTSP_NEW]
>    for how to use feature tags in capability negotiations).
> 
>    The feature tag is:
> 
>           method.eos
> 
>    which means that the END_OF_STREAM method is fully supported. By 
>    fully we mean either at non-aggregate or aggregate level.
> 
> 5. Use Case
> 
>    In this example, the server sends END_OF_STREAM request to client for
>    one live media stream, because upstream source terminates the stream
>    after 200 seconds. The fact that the stream has played for 200 seconds
>    is communicated by the Range header in the END_OF_STREAM request.
> 
>      C->S: PLAY rtsp://foo.com/bar.avi/streamid=0 RTSP/1.0
> 	   Supported: method.eos
>            CSeq: 10
>            Session: 12345678
> 	   Range: 0-200
> 
>      S->C: RTSP/1.0 200 OK
> 	   Supported: method.eos
>            CSeq: 10
>            Session: 12345678
>            RTP-Info: url=//foo.com/bar.avi/streamid=0;seq=0; rtptime=0 
> 
>      S->C: END_OF_STREAM rtsp://foo.com/bar.avi/streamid=0 RTSP/1.0
>            CSeq: 123
>            Session: 12345678
> 	   Range: npt=0-200; bytes=0-200000
>            RTP-Info: url=//foo.com/bar.avi/streamid=0;seq=45102 
> 	   Reason: End of range reached
> 
>      C->S: RTSP/1.0 200 OK  
>            CSeq: 123
>            Session: 12345678
>   
>    In the END_OF_STREAM request, the client will learn that the server has 
>    completed the stream as requested (as indicated by the "End of range 
>    reached" Reason).
> 
>    From the two RTP-Info headers, one in PLAY response, one in 
>    END_OF_STREAM requst, the client can derive the total number
>    of RTP packets that the server has sent. From the npt field in the 
>    Range header, the client knows the presentation time that this stream 
>    has covered, from the server's point of view, while the byte-range field 
>    can optional inform the 
>    client how much media data, in bytes, that the server has sent. 
>  
> 5. Security Considerations
>  
>    Because there is only one new TEXT header, "Reason", added by our 
>    new RTSP method,
>    the security considerations outlined in [RTSP_NEW] apply here as well.
>      
> 6. IANA Considerations
> 
>    A new method name, and its associated feature tag need to be registered 
>    with IANA.
>     
>      -- Method name: END_OF_STREAM
>      
>      -- feature tag: method.eos
> 
> 
> 
> 
> Appendix A: Signalling End-Of-Streaming With RTCP EOD Packet
> 
> In this appendix, a new RTCP packet type is defined to signal
> end-of-stream event. 
> 
> A new RTCP packet type is prefered over profile specific extensions
> (see section 6.4.1 of  [RTP-NEW])
> because this feature is profile independent.
> 
> 1. EOS: End-Of-Stream RTCP Packet
> 
>        0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
>       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>       |V=2|P|    SC   |   PT=EOS=205  |             length            |
>       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>       |                           SSRC/CSRC                           |
>       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>       |        extended RTP sequence number                           | 
>       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>       :                              ...                              :
>       +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
> (opt) |     length    |               reason for EOS                ...
>       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
> 
>   The EOS packet indicates that one or more sources have exhausted its
>   media data and will not send any new RTP packet, yet the source is 
>   not about to leave the session.
> 
>   packet type (PT):  8 bits
>      Contains the constant 205 to idnentify this as an RTCP EOS packet.
>  
>   source count (SC): 5 bits
>      The number of SSRC/CSRC identifiers included in this EOS packet.
>      A count value of zero is valid, but useless. For each one of 
>      them, an extended RTP sequence number is included. See below.
> 
>   extended RTP sequence number: 32 bit
>     The extended sequence number of the ending RTP packet from the 
>     source.
>     The low 16 bits contain the sequence number of the ending RTP
>     data packet from source SSRC_n, and the most significant 16
>     bits extend that sequence number with the corresponding count of
>     sequence number cycles. The RTP receiver needs to track the
>     extended sequence number to properly generate RTCP
>     RR packets, hence it can easily determine which RTP packet is the
>     ending RTP packet.
> 
> 2. Rules of sending RTCP EOS packet
> 
>   RTCP EOS packet should be sent if the source runs out of media data,
>   but is NOT about to leave the session. It may, for instance, wait
>   for the receiver to send anther RTSP PLAY request (if RTSP is the
>   session contorl protocol in use) and then re-use the SSRC to send
>   RTP packets.
> 
>   If an EOS packet is received by a mixer, the mixer SHOULD forward
>   the EOS packet with its SSRC/CSRC identifier(s) unchanged.
> 
>   Optionally,  the EOS packet MAY include an 8-bit octet count 
>   followed by that many
>   octets of text indicating the reason for ending, e.g., "End of file" 
>   or "Upstream EOS received".  The string has the same
>   encoding as that described for SDES.  If the string fills the packet
>   to the next 32-bit boundary, the string is not null terminated.  If
>   not, the EOS packet MUST be padded with null octets to the next 32-
>   bit boundary.  This padding is separate from that indicated by the P
>   bit in the RTCP header.
>   
>   When RTP is carried over un-reliable transport (e.g., UDP), an RTP 
>   source may repeatedly send RTCP EOS packets until it is assured that
>   the receiver(s) have got the event notification -- e.g., it has
>   received another RTSP PLAY request from the RTP receiver.  
> 
> ^L
> Normative References
> 
>    [RTSP_NEW] Schulzrinne, H., Rao, A., Lanphier, R., Westerlund, M., 
>          "Real Time Streaming Protocol", 
>          draft-ietf-mmusic-rfc2326bis-04.txt
> 
>    [RTP_NEW] RFC3550 "Real-time Transport Protocol", July 2003
> 
> 
> Full Copyright Statement
> 
>    Copyright (C) The Internet Society (2003). All Rights Reserved.
> 
>    This document and translations of it may be copied and furnished to
>    others, and derivative works that comment on or otherwise explain it
>    or assist in its implementation may be prepared, copied, published
>    and distributed, in whole or in part, without restriction of any
>    kind, provided that the above copyright notice and this paragraph are
>    included on all such copies and derivative works. However, this
>    document itself may not be modified in any way, such as by removing
>    the copyright notice or references to the Internet Society or other
>    Internet organizations, except as needed for the purpose of
>    developing Internet standards in which case the procedures for
>    copyrights defined in the Internet Standards process must be
>    followed, or as required to translate it into languages other than
>    English.
> 
>    The limited permissions granted above are perpetual and will not be
>    revoked by the Internet Society or its successors or assignees.
> 
>    This document and the information contained herein is provided on an
>    "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
>    TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
>    BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
> 
> 
> 
> 
> 
>    HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
>    MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
> 
> 
> Acknowledgement
> 
>    Thanks to Sean Sheedy for suggesting the inclusion of "Range" header
>    and for contributing some text in the motivations section.
> 
>    Funding for the RFC Editor function is currently provided by the
>    Internet Society.
> 
> Author Addresses
> 
>    Thomas Zeng
>    PacketVideo Corp.
>    10350 Science Center Dr., Suite 210
>    San Diego, CA 92127
>    email: zeng@pv.com
> 
>    P. Greg Sherwood
>    PacketVideo Corp.
>    10350 Science Center Dr., Suite 210
>    San Diego, CA 92127
>    email: sherwood@pv.com

-- 

Magnus Westerlund

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


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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		: Real Time Streaming Protocol (RTSP)
	Author(s)	: H. Schulzrinne
	Filename	: draft-ietf-mmusic-rfc2326bis-05.txt,.ps,.pdf
	Pages		: 147
	Date		: 2003-10-27
	
This memorandum is a revision of RFC 2326, which is currently a Pro-
posed Standard.
The Real Time Streaming Protocol, or RTSP, is an application-level
protocol for control over the delivery of data with real-time
properties. RTSP provides an extensible framework to enable con-
trolled, on-demand delivery of real-time data, such as audio and
video. Sources of data can include both live data feeds and stored
clips. This protocol is intended to control multiple data delivery
sessions, provide a means for choosing delivery channels such as UDP,
multicast UDP and TCP, and provide a means for choosing delivery
mechanisms based upon RTP (RFC 1889).

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Subject: RE: [MMUSIC] [ rtspspec-Bugs-810678 ] How to indicate end of medi a stream
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To: "Amir Wolf" <Amir.Wolf@emblaze.com>, "Thomas Zeng" <zeng@PacketVideo.COM>,
        "Magnus Westerlund" <magnus.westerlund@ericsson.com>,
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> -----Original Message-----
> From: Amir Wolf [mailto:Amir.Wolf@emblaze.com]
> Sent: Tuesday, October 28, 2003 4:02 AM

> I think that signaling EOS per stream has more value since
> one of the most helpful uses of this feature is when you play=20
> a very low fps video e.g. average of 1/4 fps and audio. It is=20
> important for the player to know when video ends, while it=20
> can easily identify end of session by getting the required duration.=20

What do you mean by "required duration"?  The value of the RTSP Range
header, or something else?

The problem is a PLAY request.  At least in the current spec, if a
play-with-range request is received while a previous play request is
still playing, the server is supposed to queue the request.  A
play-sans-range request is a query if the server is still playing, but
causes a state change if the server is not playing.

The problem is defining what "still playing" means.  In the normal case
the audio and video streams are never exactly the same duration, so one
stream can easily finish before another.  (The is especially true in the
TrueCast case, because I try to stay 5 sec. ahead of real-time for
audio, so that stream will definitely finish first.)

So if one stream finishes, does this mean the original play request is
"still playing," because not all of the streams are done?  This is what
I was alluding to in my previous post about a play request being
"atomic."

The issue is that if the client must synthesize whether the server
thinks it's done, by keeping track of the different EOS indicators, then
the client might reach a conclusion that's different from the server.
If the server says that it's done, then there is no ambiguity.

This can be a problem if the client issues a play request (say, sans
range header), because he doesn't know apriori whether the server will
treat this as a no-op query, or actually start playing, unless the
client knows the exact state of the server.

The RTSP spec isn't very helpful on this point, because it doesn't
clearly define what "still playing" means for an aggregate play request.


> > -----Original Message-----
> > From: mmusic-admin@ietf.org [mailto:mmusic-admin@ietf.org]On Behalf=20
> > Of Matthew Heaney
> > Sent: Monday, October 27, 2003 10:19 PM
> > To: Thomas Zeng; Magnus Westerlund; Tom Marshall
> > Cc: mmusic@ietf.org
> > Subject: RE: [MMUSIC] [ rtspspec-Bugs-810678 ] How to
> > indicate end of medi a stream
> >=20
> >=20
> >=20
> >=20
> > > -----Original Message-----
> > > From: Thomas Zeng [mailto:zeng@PacketVideo.COM]
> > > Sent: Monday, October 27, 2003 2:24 PM
> > >=20
> > > > -----Original Message-----
> > > > From: Matthew Heaney [mailto:mheaney@on2.com]
> > > > Sent: Monday, October 27, 2003 11:19 AM
> > > >=20
> > > > Perhaps it would be useful for the server to indicate
> > that it's done
> > > > playing all the streams implied by the aggregate play request.
> > >=20
> > > Do you imply "end-of-play" should be signalled by RTSP
> server, then?
> >=20
> > I was thinking that if we treat the original play request
> as "atomic,"
> > that perhaps the server should indicate that the entire
> play request
> > has finished.  By "finished" I mean all the streams that entered the

> > playing state as a result of the aggregate play request are done.
> >=20
> > If you send an EOS indicator for each stream separately, then it's=20
> > up to the client to figure out when to move to another aggregate
> play state.
> > Right now he must synthesize this information himself, based
> > on how many
> > streams were reported back in the RTP-Info of the PLAY=20
> > response, and how
> > many different EOS indicators he has received.
> >=20
> >=20
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org https://www1.ietf.org/mailman/listinfo/mmusic
> >=20
>=20

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Hi Matthew,

I can do one simplification for you. When we are discussing an End of 
stream or play request we are doing this in the context of the updated 
specification. However I do see that there is probably need to make some 
short considerations on how this will interoperate with RFC 2326 
implementation.

However if you are not familiar with the currently ongoing work on 
updating RTSP you can start with looking at the latest draft version. 
However I will already now make it clear that the draft is not 
publication ready. We authors are hoping to get a version out before 
christmas that actual incorporates all decided fixes to the protocol.

http://www.ietf.org/internet-drafts/draft-ietf-mmusic-rfc2326bis-05.txt

The current RTSP is a protocol stripping away all functionality that was 
not implemented in RFC 2326. So the following functionality has been 
removed:

- Queued play
- Record and Announce methods.
- Use of non-aggregated play and pause control in Aggregated sessions.

More information about this work can be seen on:
http://rtsp.org
http://rtspspec.sourceforge.net

In regards to the end of media/play request fulfilled signalling this 
does not change much. However your arguments about separating initial 
and any queued play is probably not a argument we would give much 
attention to.

This allows me to bring up a discussion on why this signalling is need. 
Reasons I see:

- For living signalling without a predestined end of the session. In 
this case it seems the main goal is to inform the client that session 
has ended. Possibly a user readable reason is desirable to be given, 
Like "Tonight's jam session has now ended, come back next Tuesday for 
more interesting music!". The timing of this is that the message should 
be possible to give as soon as the last media has been played out.

- To allow the transport handler in the client to know what the last 
packet it will receive to prevent unsecurity about buffering handling, 
retransmission. Thus it is desired to know before the last packet is 
received that it will be the last one. This is true independently of the 
session type.

- To inform the client about that the play request has been fulfilled 
and all media streams will be delivered when the given packets has been 
received. However for media streams that has a determined length and for 
PLAY requests with closed ranges this information is derivable to a high 
degree. The exact sequence number can't be determined, however the RTP 
timestamp for the end of the request can be determined.

- Something new which we haven't discussed so far is: Error information. 
If the server fails to fulfil the request during the playout. Using a 
server->client message the server can inform the client that the 
delivery has failed, why, and at which point in the media stream.

Are there any more reasons for having something like this?

I would also like to make people aware that depending on what usage are 
the most important the preferred choice of delivery is different.

Cheers

Magnus



Matthew Heaney wrote:

> 
>>-----Original Message-----
>>From: Amir Wolf [mailto:Amir.Wolf@emblaze.com]
>>Sent: Tuesday, October 28, 2003 4:02 AM
> 
> 
>>I think that signaling EOS per stream has more value since
>>one of the most helpful uses of this feature is when you play 
>>a very low fps video e.g. average of 1/4 fps and audio. It is 
>>important for the player to know when video ends, while it 
>>can easily identify end of session by getting the required duration. 
> 
> 
> What do you mean by "required duration"?  The value of the RTSP Range
> header, or something else?
> 
> The problem is a PLAY request.  At least in the current spec, if a
> play-with-range request is received while a previous play request is
> still playing, the server is supposed to queue the request.  A
> play-sans-range request is a query if the server is still playing, but
> causes a state change if the server is not playing.
> 
> The problem is defining what "still playing" means.  In the normal case
> the audio and video streams are never exactly the same duration, so one
> stream can easily finish before another.  (The is especially true in the
> TrueCast case, because I try to stay 5 sec. ahead of real-time for
> audio, so that stream will definitely finish first.)
> 
> So if one stream finishes, does this mean the original play request is
> "still playing," because not all of the streams are done?  This is what
> I was alluding to in my previous post about a play request being
> "atomic."
> 
> The issue is that if the client must synthesize whether the server
> thinks it's done, by keeping track of the different EOS indicators, then
> the client might reach a conclusion that's different from the server.
> If the server says that it's done, then there is no ambiguity.
> 
> This can be a problem if the client issues a play request (say, sans
> range header), because he doesn't know apriori whether the server will
> treat this as a no-op query, or actually start playing, unless the
> client knows the exact state of the server.
> 
> The RTSP spec isn't very helpful on this point, because it doesn't
> clearly define what "still playing" means for an aggregate play request.
> 
> 
> 
-- 

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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I've been given the task of recoding the RTSP versioning support in our
server (it currently is pretty much nonexistent).  Since the draft spec is
nearing completion, this might be a good time to clarify how versioning
should be handled.  I've collected a few datapoints...

* RFC 2326 3.1 references RFC 2068 3.1.  It says, in part:

    The <minor> number is incremented when the changes made to the protocol
    add features which do not change the general message parsing algorithm,
    but which may add to the message semantics and imply additional
    capabilities of the sender. The <major> number is incremented when the
    format of a message within the protocol is changed.

* Apache (both 1.3.x and 2.0.x) will respond v1.1 for a v1.0+ request.  eg:

    GET / HTTP/1.0
    Host: localhost

    HTTP/1.1 200 OK
    [...]

    GET / HTTP/2.0
    Host: localhost

    HTTP/1.1 200 OK
    [...]

* IIS 6.0 will respond v1.0 for a v1.0 request and v1.1 for a v1.1+ request=
=2E=20
  eg:

    GET / HTTP/1.0
    Host: www.microsoft.com

    HTTP/1.0 200 OK
    [...]

    GET / HTTP/1.1
    Host: www.microsoft.com

    HTTP/1.1 200 OK
    [...]

    GET / HTTP/1.2
    Host: www.microsoft.com

    HTTP/1.1 200 OK
    [...]

    GET / HTTP/2.0
    Host: www.microsoft.com

    HTTP/1.1 200 OK
    [...]

That is really all the information we have for RTSP in the strictest sense,
because RFC 2616 was written after RFC 2326 and RFC 2145 is never referenced
in RFC 2326.  However, since RFC 2145 applies to HTTP/1.1 (and thus RTSP/1.0
by extension), and it was written prior to RFC 2326, it might provide some
valuable information for us.  According to section 2.3, it appears to me
that Apache properly implements the recommendation and IIS implements a
compliant but non-recommended behavior.

The biggest question that remains in my mind is the proper response to major
version changes.  What does it mean when the format of a message changes?=
=20
Would it be possible for HTTP/2.0 to use, say, XML for its headers so that
it's not even parseable by a HTTP/1.x implementation?  If so, shouldn't a
1.x implementation always respond with 505 if the major version is higher
than it understands?

The new RTSP draft references RFC 2616, which in turn references RFC 2145.=
=20
I assume all of the above will apply to it as well.  Does everyone feel that
the short reference to RFC 2616 in section 3.1 of the draft is sufficient?

--=20
The nice thing about standards is that there are so many of them to choose
=66rom.
        -- Andrew Tanenbaum

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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		: Session Description and Capability Negotiation
	Author(s)	: D. Kutscher, J. Ott, C. Bormann
	Filename	: draft-ietf-mmusic-sdpng-07.txt
	Pages		: 65
	Date		: 2003-10-28
	
This document defines a language for describing multimedia sessions
with respect to configuration parameters and capabilities of end
systems. 
This document is a product of the Multiparty Multimedia Session
Control (MMUSIC) working group of the Internet Engineering Task
Force. Comments are solicited and should be addressed to the working
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.

	Title		: SDP: Session Description Protocol
	Author(s)	: M. Handley, V. Jacobson, C. Perkins
	Filename	: draft-ietf-mmusic-sdp-new-15.txt
	Pages		: 45
	Date		: 2003-10-28
	
This memo defines the Session Description Protocol (SDP). SDP is
intended for describing multimedia sessions for the purposes of
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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-831553 ] Error in rfc2326bis-04 example 15.4 (SourceForge.net)

--__--__--

Message: 1
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Mon, 27 Oct 2003 21:20:21 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-831553 ] Error in rfc2326bis-04 example 15.4

Bugs item #831553, was opened at 2003-10-28 05:20
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=831553&group_id=23194

Category: Editorial
Group: None
Status: Open
Resolution: None
Priority: 5
Submitted By: Thomas M Zeng (zengtm)
Assigned to: Nobody/Anonymous (nobody)
Summary: Error in rfc2326bis-04 example 15.4

Initial Comment:
The last two messages in this example is:

   C->M: PLAY rtsp://live.example.com/concert/audio 
RTSP/1.0
         CSeq: 3
         Session: 0456804596

   M->C: RTSP/1.0 200 OK
         CSeq: 3
         Session: 0456804596
         Range:npt=now-


But I wonder how the receiver could perform the sync
between RTP timestamp and NPT, without a RTP-Info 
header in the PLAY response? I wouldn't like to see 
such RTSP conversation in our applicatoin.

A real implementation would likely to conduct the PLAY 
request/response including RTP-Info, for example: 

C->M: PLAY rtsp://live.example.com/concert/audio 
RTSP/1.0
         CSeq: 3
         Session: 0456804596
         Range: now-

   M->C: RTSP/1.0 200 OK
         CSeq: 3
         Session: 0456804596
         Range:npt=1000-
         RTP-Info: 
url=rtsp://live.example.com/concert/audio; seq=1473; 
rtptime=80000

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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=831553&group_id=23194




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From exim@www1.ietf.org  Wed Oct 29 02:17:26 2003
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Message-ID: <3F9F6933.3010508@dynamicsoft.com>
Date: Wed, 29 Oct 2003 02:16:03 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Subject: [MMUSIC] Updated ICE specification
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Folks,

I have submitted an update to ICE. Based on agreement at the last 
ietf, it is now a work item of MMUSIC. Until the I-D appears in the 
archives, you can pick up a copy at:

http://www.jdrosen.net/papers/draft-ietf-mmusic-ice-00.txt
http://www.jdrosen.net/papers/draft-ietf-mmusic-ice-00.html

There were many changes since the last revision:

* The draft is now SIP and offer/answer independent. The way I 
achieved this was to describe ICE as an abstract protocol, and then 
define mappings from that abstract protocol to concrete ones. A 
mapping is included for SIP, using the SDP alt parameter extension and 
Gonzalo's new IPV group semantic.

* The spec now mandates that a client, in addition to be configured 
with one or more relay servers (i.e., turn), needs to know which of 
those are on the public internet. This would be through configuration. 
This information is needed only for backwards compatibility with 
non-ice clients.

* In an update to TURN, I added a Send mechanism which allows relay of 
outbound requests. ICE now includes usage of this mechanism. Its 
needed for inter-enterprise calls where the media needs to pass 
through two relays, one in each domain.

* I added some text on backwads compatibility processing when the 
caller realizes that the callee doesnt support ICE.

* peer derived transport addresses learned by Sending (using the TURN 
Send primitive) a STUN request through a TURN relay have the same 
priority as turn derived transport addresses.


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



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Subject: [MMUSIC] Error in rfc2326bis-04 example 15.4
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Hi,

As Thomas has logged a bug about some examples, I like to point out that 
the examples has not been corrected to follow the specification text. 
Therefore I strongly suggest that people ignore the examples unless they 
are willing to send us complete proposals for how they should look.

Thanks Thomas for the proposal for how to fix it, however I have a 
comment in regards to the example. See below:

> Bugs item #831553, was opened at 2003-10-28 05:20
> 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=831553&group_id=23194
> 
> Category: Editorial
> Group: None
> Status: Open
> Resolution: None
> Priority: 5
> Submitted By: Thomas M Zeng (zengtm)
> Assigned to: Nobody/Anonymous (nobody)
> Summary: Error in rfc2326bis-04 example 15.4
> 
> Initial Comment:
> The last two messages in this example is:
> 
>    C->M: PLAY rtsp://live.example.com/concert/audio 
> RTSP/1.0
>          CSeq: 3
>          Session: 0456804596
> 
>    M->C: RTSP/1.0 200 OK
>          CSeq: 3
>          Session: 0456804596
>          Range:npt=now-
> 
> 
> But I wonder how the receiver could perform the sync
> between RTP timestamp and NPT, without a RTP-Info 
> header in the PLAY response? I wouldn't like to see 
> such RTSP conversation in our applicatoin.
> 
> A real implementation would likely to conduct the PLAY 
> request/response including RTP-Info, for example: 
> 
> C->M: PLAY rtsp://live.example.com/concert/audio 
> RTSP/1.0
>          CSeq: 3
>          Session: 0456804596
>          Range: now-
> 
>    M->C: RTSP/1.0 200 OK
>          CSeq: 3
>          Session: 0456804596
>          Range:npt=1000-
>          RTP-Info: 
> url=rtsp://live.example.com/concert/audio; seq=1473; 
> rtptime=80000
> 

Yes, it is correct that RTP-Info are need. However there are several 
options in this example:

- Have a Range header in the Play request?

- Should the server specify the current point since session start, or 
only give a now, which doesn't provide the client any reference to where 
in the live session the stream is in, but provides the minimum 
information about what is needed for synchronization.

If you have any comments. Please respond.

Cheers

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
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Subject: Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-rfc2326bis-05.txt,.ps,.pdf
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Hi,

I would like to make you aware of that the 05 text versions contains a 
number of erroneous formatted tables. This is the known occurrences:

Page 24, status codes.
Page 26, table lines
Page 51, Adjusted Packet diagram does not fit on one page.
Page 62, 63, Header table.
Page 99, Message syntax
Page 100, 101, Message syntax
page 114, Table 7, init state
Page 115, Table 8, Ready State
Page 116, Table 9, Play State

You will have to live with these errors or read the pdf or PS version. 
Please note that the change bars in the documents are in relation to the 
04 draft version. The major changes are in the methods chapter where a 
lot of rewrite has been performed.

We will post a new corrected version after the meeting.

Cheers

Magnus

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		: Real Time Streaming Protocol (RTSP)
> 	Author(s)	: H. Schulzrinne
> 	Filename	: draft-ietf-mmusic-rfc2326bis-05.txt,.ps,.pdf
> 	Pages		: 147
> 	Date		: 2003-10-27
> 	
> This memorandum is a revision of RFC 2326, which is currently a Pro-
> posed Standard.
> The Real Time Streaming Protocol, or RTSP, is an application-level
> protocol for control over the delivery of data with real-time
> properties. RTSP provides an extensible framework to enable con-
> trolled, on-demand delivery of real-time data, such as audio and
> video. Sources of data can include both live data feeds and stored
> clips. This protocol is intended to control multiple data delivery
> sessions, provide a means for choosing delivery channels such as UDP,
> multicast UDP and TCP, and provide a means for choosing delivery
> mechanisms based upon RTP (RFC 1889).
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-rfc2326bis-05.txt
> 
> To remove yourself from the IETF Announcement list, send a message to 
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> Internet-Drafts are also available by anonymous FTP. Login with the username
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> 
> A list of Internet-Drafts directories can be found in
> http://www.ietf.org/shadow.html 
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> 
> 
> Internet-Drafts can also be obtained by e-mail.
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> Send a message to:
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-- 

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
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Subject: RE: [MMUSIC] RTSP transport negotiation
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Dear Geetha,

>... now change client IP address ....
>
>     C->S: SETUP rtsp://foo/twister/video RTSP/1.0
>           CSeq: 5
>	   Transport: RTP/AVP;unicast;client_ip=3D<new IP address;
>	              client_port=3D8002-8003;server_port=3D9004-9005
>	   Range: npt=3D5-
>           Session: 12345678


I'm not sure the Range field can be aplicated to the SETUP request.
From the specification it seems the Range is applicable on the SETUP =
response
but not in the SETUP request.

   Header              Where  Proxy DES OPT SETUP PLAY PAUSE TRD =20

   Range                 R           -   -    -    o     o   -
   Range                 r           -   -    c    m*   m*   -


Best Regards,
Enrico




------------------------------------------------------------------
Ing Enrico M. Balatti
CEFRIEL - Politecnico di Milano
ICT Center of Excellence For Research, Innovation,
Education & industrial Labs partnership
via Fucini 2, 20133 Milano (ITALY)
voice: +39 0223954327 fax: +39 0223954527
e-mail: balatti@cefriel.it <mailto:balatti@cefriel.it>   www.cefriel.it =
<http://www.cefriel.it>=20
------------------------------------------------------------------


-----Original Message-----
From: Geetha Srikantan [mailto:Geetha.Srikantan@Sun.COM]
Sent: mercoled=EC 22 ottobre 2003 7.26
To: mmusic@ietf.org
Subject: [MMUSIC] RTSP transport negotiation


Hello everyone,

Here is the promised write-up on RTSP transport negotiation..sorry for
the delay. If there is agreement on the content of this note, I can
write up text for the RTSP draft.

thanks
geetha


Topic: RTSP Transport negotiation using SETUP during play state.

Introduction

The RTSP specification allows for negotiation of transport parameters
via the SETUP method, while the RTSP session is in play state. The
transport parameters that can be negotiated and the exact form of this
negotiation is under-specified in the RTSP documents. This note is an
attempt to clarify some of the issues around transport negotiation for
inclusion in the ongoing revision of RTSP.

The recent interest in mobility for RTSP implementations also impacts
transport parameters used by the session. For instance, a wireless
RTSP client could be assigned a new IP address, or may experience
change of network characteristics, while it is in the midst of a RTSP
session. These issues are also addressed in this note.


Outline:

1. Objective
2. Enumeration of all Transport Parameters
3. Transport Parameters Excluded from Negotiation
4. Negotiable Transport Parameters
5. Impact of Transport Negotiation
6. Examples
	6.1 Client IP Address Changes
	6.2 UDP -> TCP
	6.3 TCP -> UDP
7. Other transport related changes to RTSP draft
8. Summary



1. Objective:=20

The main goals of this document are to
	* Provide an enumeration of transport parameters.
	* Identify transport parameters that cannot be negotiated
	  with minimal impact on other sessions.
	* Propose transport parameters that can be negotiated during
	  play state.
	* Determine implications of changing these parameters via a
	  SETUP request while the session is in play state.
	* Provide examples demonstrating transport negotiation.


2. Enumeration of all Transport Parameters

   The following is a list of all transport parameters in the current
   RTSP draft.

	unicast/multicast
	destination
	source
	mode (=3DPLAY)
	interleaved
	layers (for multicast)
	ttl (for multicast
	port (for multicast)
	client_port (for RTP)
	server_port (for RTP)
	ssrc (for RTP)
	url


3. Transport Parameters Excluded from Negotiation

   It is recommended that the following parameters be excluded from the
   negotiation process.

	mode
	url
	unicast/multicast
	layers
	ttl
	ssrc
	destination
	source


    As the "mode" parameter is restricted to a single value in the
current draft, it can be excluded.

    As per the recent proposal to not support addition of urls via SETUP
during play state, the "url" parameter can be excluded.=20

    Changing from multicast to unicast based on a single RTSP session's
SETUP, would impact all the other clients accessing the multicast
session.=20
Several issues need to be considered in this case, including transport
negotiation. Similarly, changing from unicast to multicast also has
implications beyond a single RTSP session. Changes to other
multicast-related parameters, "layers", "ssrc", "ttl" most likely
requires specification beyond transport negotiation.
The recommendation is to *not* specify changing the transport from
"unicast" to "multicast" or vice versa, and also not specify changes
to the "layers" and "ttl" parameters, in this draft. It is recommedned=20
that negotiation of multicast-related transport parameters should be
deferred to an RTSP-extension document.=20

    Once an RTSP session has been initiated and the "destination" and
"source" have been established - i.e. a multicast "destination" has
been identified, and the stream "source" has been identified, there
should be no need to change these parameters.  It is recommended that
the "destination" and "source" parameters not be negotiated when the
session is in play state.  The case of the client IP address change is
signaled in a different manner, outlined in the next section.


4. Negotiable Transport Parameters

   Hence there are three main conditions that require transport
negotiation for unicast:

	* client IP address changes
	* client wants to switch transport from UDP -> TCP=20
	* client wants to switch transport from TCP -> UDP

   Changes in the client IP address can be indicated using a new
"client_ip" parameter. Hence the parameters that are impacted by
transport negotiation are:

	client_ip
	interleaved
	client_port
	server_port

=20
5. Impact of Transport Negotiation

   The first column describes the primary change, the second and third
columns describe any secondary transport parameters that MUST or=20
MAY change as a result of the primary change.


Changed			MUST			MAY=09
-------			----			---
=09
Client IP address	client_ip		client_port
						server_port

UDP -> TCP		interleaved		Server may change
			Drop client_port	interleaved channel ids
			Drop server_port	in response.

TCP -> UDP		client_port in request	client_ip
			server_port in response client_port in response=09
			Drop interleaved



6. Examples

6.1 Client IP Address Changes

     C->S: DESCRIBE rtsp://foo/twister.sdp RTSP/1.1
	   CSeq: 1

     S->C: RTSP/1.0 200 OK
	   CSeq: 1
           Content-Type: application/sdp
	   Content-Length: 164

	   v=3D0
           o=3D- 2890844256 2890842807 IN IP4 172.16.2.93
           s=3DRTSP Session
           i=3DAn Example of RTSP Session Usage
           a=3Dcontrol:rtsp://foo/twister
           t=3D0 0
           m=3Daudio 0 RTP/AVP 0
           a=3Dcontrol:rtsp://foo/twister/audio
           m=3Dvideo 0 RTP/AVP 26
           a=3Dcontrol:rtsp://foo/twister/video

     C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
           CSeq: 2
           Transport: RTP/AVP;unicast;client_port=3D8000-8001

     S->C: RTSP/1.0 200 OK
           CSeq: 2
           Transport: RTP/AVP;unicast;client_port=3D8000-8001;
                      server_port=3D9000-9001
           Session: 12345678

     C->S: SETUP rtsp://foo/twister/video RTSP/1.0
           CSeq: 3
           Transport: RTP/AVP;unicast;client_port=3D8002-8003
           Session: 12345678

     S->C: RTSP/1.0 200 OK
           CSeq: 3
           Transport: RTP/AVP;unicast;client_port=3D8002-8003;
                      server_port=3D9004-9005
           Session: 12345678

     C->S: PLAY rtsp://foo/twister RTSP/1.0
           CSeq: 4
           Range: npt=3D0-
           Session: 12345678

     S->C: RTSP/1.0 200 OK
           CSeq: 4
           Session: 12345678
           RTP-Info: url=3Drtsp://foo/twister/video;
             seq=3D9810092;rtptime=3D3450012,
	   url=3Drtsp://foo/twister/audio;seq=3D5923;rtptime=3D22234432
	   Range: npt=3D0-30.76

... now change client IP address ....

     C->S: SETUP rtsp://foo/twister/video RTSP/1.0
           CSeq: 5
	   Transport: RTP/AVP;unicast;client_ip=3D<new IP address;
	              client_port=3D8002-8003;server_port=3D9004-9005
	   Range: npt=3D5-
           Session: 12345678
		=09
     S->C: RTSP/1.0 200 OK
           CSeq: 5
	   Transport: RTP/AVP;unicast;client_ip=3D<new IP address>;
		      client_port=3D8002-8003;server_port=3D9004-9005
	   RTP-Info: url=3Drtsp://foo/twister/video;
		    seq=3D<new>;rtptime=3D<new>,
	   Range: npt=3D5-30.76
	   Session: 12345678


     C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
           CSeq: 6
	   Transport:RTP/AVP;unicast;client_ip=3D<new IP address>;
		     client_port=3D8000-8001;server_port=3D9000-9001
	   Range: npt=3D5-
           Session: 12345678

     S->C: RTSP/1.0 200 OK
           CSeq: 6
	   Transport: RTP/AVP;unicast;client_ip=3D<new IP address>;
		      client_port=3D8000-8001;server_port=3D9000-9001
	   RTP-Info:url=3Drtsp://foo/twister/audio;
		    seq=3D<new>;rtptime=3D<new>
	   Range: npt=3D5-30.76=09
	   Session: 12345678



    =20
6.2 UDP -> TCP

     C->S: DESCRIBE rtsp://foo/twister.sdp RTSP/1.0
	   CSeq: 1

     S->C: RTSP/1.0 200 OK
	   Content-Base: rtsp://foo/twister/
	   Content-Length: 230
	   Content-Type: application/sdp
	   Cseq: 1

	   v=3D0
	   o=3D- 3262117360 3262117360 IN IP4 0.0.0.0
	   s=3DRTSP Session
	   c=3DIN IP4 0.0.0.0
	   t=3D0 0
	   a=3Dcontrol:*
	   a=3Drange:npt=3D0-10.566000
	   m=3Dvideo 0 RTP/AVP 26
	   a=3Dcontrol:rtsp://foo/twister/video
	   m=3Daudio 0 RTP/AVP 0
	   a=3Dcontrol:rtsp://foo/twister/audio

     C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
	   CSeq: 2
	   Transport: RTP/AVP/UDP;unicast;client_port=3D8000-8001

     S->C: RTSP/1.0 200 OK
	   CSeq: 2
	   Session: 12345678
	   Transport: RTP/ACP/UDP;unicast;client_port=3D8000-8001;
		      server_port=3D9000-9001

     C->S: PLAY rtsp://foo/twister RTSP/1.0
	   CSeq: 3
	   Session: 12345678

     S->C: RTSP/1.0 200 OK
	   Content-Base: rtsp://foo/twister/
	   CSeq: 3
	   RTP-Info: =
url=3Drtsp://foo/twister/audio;seq=3D7028;rtptime=3D1339949552
	   Range: npt=3D0-
	   Session: 12345678

... now request TCP ...

     C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
	   CSeq: 4
	   Transport: RTP/AVP/TCP;interleaved=3D0-1

     S->C: RTSP/1.0 200 OK
	   Cseq: 4
	   Transport: RTP/AVP/TCP;unicast;
		      client_ip=3D<client IP>;interleaved=3D0-1
	   RTP-Info: url=3Drtsp://foo/twister/audio;seq=3D<new>;rtptime=3D<new>
	   Range: npt=3D0-10.56600
	   Session: 12345678



6.3 TCP -> UDP

     C->S: DESCRIBE rtsp://foo/twister.sdp RTSP/1.0
	   CSeq: 1

     S->C: RTSP/1.0 200 OK
	   Content-Base: rtsp://foo/twister/
	   Content-Length: 230
	   Content-Type: application/sdp
	   Cseq: 1

	   v=3D0
	   o=3D- 3262117360 3262117360 IN IP4 0.0.0.0
	   s=3DRTSP Session
	   c=3DIN IP4 0.0.0.0
	   t=3D0 0
	   a=3Dcontrol:*
	   a=3Drange:npt=3D0-10.566000
	   m=3Dvideo 0 RTP/AVP 26
	   a=3Dcontrol:rtsp://foo/twister/video
	   m=3Daudio 0 RTP/AVP 0
	   a=3Dcontrol:rtsp://foo/twister/audio

     C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
	   CSeq: 2
	   Transport: RTP/AVP/TCP;interleaved=3D0-1

     S->C: RTSP/1.0 200 OK
	   Cseq: 2
	   Session: 12345678
	   Transport: RTP/AVP/TCP;unicast;client_ip=3D<client =
IP>;interleaved=3D0-1

     C->S: PLAY rtsp://foo/twister RTSP/1.0
	   CSeq: 3
	   Session: 12345678

     S->C: RTSP/1.0 200 OK
	   Content-Base: rtsp://foo/twister/
	   Cseq: 3
	   RTP-Info: =
url=3Drtsp://foo/twister/audio;seq=3D7028;rtptime=3D1339949552
	   Range: npt=3D0-
	   Session: 12345678

... now request UDP ...


     C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
	   CSeq: 4
	   Transport: RTP/AVP/UDP;unicast;client_port=3D8000-8001
	   Range: npt=3D0-

     S->C: RTSP/1.0 200 OK
	   CSeq: 4
	   Transport: RTP/ACP/UDP;unicast;client_port=3D8000-8001;
		      server_port=3D9000-9001
	   RTP-Info: url=3Drtsp://foo/twister/audio;seq=3D<new>;rtptime=3D<new>
	   Range: npt=3D0-10.56600
	   Session: 12345678




7. Other transport related changes to RTSP draft

A few comments on draft-ietf-mmusic-rfc2326bis-04.txt relating to
transport parameters.

(a) Section 13.40:
    Page 81: Are "src_addr" and "dest_addr" parameters
    going away? If these are relevant for NAT, perhaps they belong in
    the NAT document?

    Page 83-83: BNF indicates "client_port" and
    "server_port" will be indicated as 'port1-port2'. This doesn't seem
    adequate for the case where the RTP and RTCP ports are non-adjacent
    (per the new RTP spec rfc 3550).=20

    Proposal:
	client_port: rtp=3D<port>, rtcp=3D<port>
	server_port: rtp=3D<port>, rtcp=3D<port>

(b) Appendix B, B.1.1
    Page 115: Mentions "src_addresses" and
    "dest_addresses" - what are these?

    Page 116: "non-continuous" ports - might want to rephrase to
    "non-adjacent" (or "non-continguous") ports?



8. Summary


This note recommends negotiation of a few transport parameters and
specifies mechanisms to do so. One new transport parameter "client_ip"=20
is recommended for use in cases where the client's IP address changes
during the session.=20

It is also recommended that the client_port and server_port parameters
be specified further with the use of tags "rtp_port" and "rtcp_port"
to handle the case where these ports are non-contiguous.


--=20
Geetha Srikantan <Geetha.Srikantan@Sun.COM>


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From exim@www1.ietf.org  Wed Oct 29 10:45:12 2003
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Subject: RE: [MMUSIC] I-D ACTION:draft-ietf-mmusic-rfc2326bis-05.txt,.ps,.pdf
Date: Wed, 29 Oct 2003 10:30:08 -0500
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Does the -05 version have a .pdf format?  This URL doesn't work:

<http://www.ietf.org/internet-drafts/draft-ietf-mmusic-rfc2326bis-05.pdf
>

I can only find the .txt file.



> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]=20
> Sent: Wednesday, October 29, 2003 3:53 AM
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] I-D=20
> ACTION:draft-ietf-mmusic-rfc2326bis-05.txt,.ps,.pdf
>=20
>=20
> Hi,
>=20
> I would like to make you aware of that the 05 text versions=20
> contains a=20
> number of erroneous formatted tables. This is the known occurrences:
>=20
> Page 24, status codes.
> Page 26, table lines
> Page 51, Adjusted Packet diagram does not fit on one page.
> Page 62, 63, Header table.
> Page 99, Message syntax
> Page 100, 101, Message syntax
> page 114, Table 7, init state
> Page 115, Table 8, Ready State
> Page 116, Table 9, Play State
>=20
> You will have to live with these errors or read the pdf or PS=20
> version.=20
> Please note that the change bars in the documents are in=20
> relation to the=20
> 04 draft version. The major changes are in the methods=20
> chapter where a=20
> lot of rewrite has been performed.
>=20
> We will post a new corrected version after the meeting.
>=20
> Cheers
>=20
> Magnus
>=20
> Internet-Drafts@ietf.org wrote:
>=20
> > A New Internet-Draft is available from the on-line Internet-Drafts=20
> > directories. This draft is a work item of the Multiparty Multimedia=20
> > Session Control Working Group of the IETF.
> >=20
> > 	Title		: Real Time Streaming Protocol (RTSP)
> > 	Author(s)	: H. Schulzrinne
> > 	Filename	: draft-ietf-mmusic-rfc2326bis-05.txt,.ps,.pdf
> > 	Pages		: 147
> > 	Date		: 2003-10-27
> > =09
> > This memorandum is a revision of RFC 2326, which is=20
> currently a Pro-=20
> > posed Standard. The Real Time Streaming Protocol, or RTSP, is an=20
> > application-level protocol for control over the delivery of=20
> data with=20
> > real-time properties. RTSP provides an extensible framework=20
> to enable=20
> > con- trolled, on-demand delivery of real-time data, such as=20
> audio and
> > video. Sources of data can include both live data feeds and stored
> > clips. This protocol is intended to control multiple data delivery
> > sessions, provide a means for choosing delivery channels=20
> such as UDP,
> > multicast UDP and TCP, and provide a means for choosing delivery
> > mechanisms based upon RTP (RFC 1889).
> >=20
> > A URL for this Internet-Draft is:=20
> >=20
> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-rfc2326bis-05.tx
> > t
> >=20
> > To remove yourself from the IETF Announcement list, send a=20
> message to
> > ietf-announce-request with the word unsubscribe in the body=20
> of the message.
> >=20
> > Internet-Drafts are also available by anonymous FTP. Login with the=20
> > username "anonymous" and a password of your e-mail address. After=20
> > logging in, type "cd internet-drafts" and then
> > 	"get draft-ietf-mmusic-rfc2326bis-05.txt".
> >=20
> > A list of Internet-Drafts directories can be found in=20
> > http://www.ietf.org/shadow.html or=20
> > ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> >=20
> >=20
> > Internet-Drafts can also be obtained by e-mail.
> >=20
> > Send a message to:
> > 	mailserv@ietf.org.
> > In the body type:
> > 	"FILE /internet-drafts/draft-ietf-mmusic-rfc2326bis-05.txt".
> > =09
> > NOTE:	The mail server at ietf.org can return the document in
> > 	MIME-encoded form by using the "mpack" utility.  To use this
> > 	feature, insert the command "ENCODING mime" before the "FILE"
> > 	command.  To decode the response(s), you will need "munpack" or
> > 	a MIME-compliant mail reader.  Different MIME-compliant=20
> mail readers
> > 	exhibit different behavior, especially when dealing with
> > 	"multipart" MIME messages (i.e. documents which have been split
> > 	up into multiple messages), so check your local documentation on
> > 	how to manipulate these messages.
> > 	=09
> > 	=09
> > Below is the data which will enable a MIME compliant mail reader=20
> > implementation to automatically retrieve the ASCII version of the=20
> > Internet-Draft.
> >=20
>=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
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
>=20

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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		: Interactive Connectivity Establishment (ICE): 
			  A Methodology for Network Address Translator (NAT) 
			  Traversal for the Session Initiation Protocol (SIP)
	Author(s)	: J. Rosenberg
	Filename	: draft-ietf-mmusic-ice-00.txt
	Pages		: 37
	Date		: 2003-10-29
	
This document describes a methodology for Network Address Translator
   (NAT) traversal for multimedia session signaling protocols, such as
   the Session Initiation Protocol (SIP). This methodology is called
   Interactive Connectivity Establishment (ICE). ICE is not a new
   protocol, but rather makes use of existing protocols, such as Simple
   Traversal of UDP Through NAT (STUN), Traversal Using Relay NAT (TURN)
   and even Real Specific IP (RSIP). ICE works through the mutual
   cooperation of both endpoints in a session.

A URL for this Internet-Draft is:
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Date: Thu, 30 Oct 2003 13:40:16 -0500
From: Aravind Narasimhan <Aravind.Narasimhan@Sun.COM>
Subject: Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-rfc2326bis-05.txt,.ps,.pdf
In-reply-to: <AFD33B1DBD906847B460CE449A9EC53B57EE8A@exchange.alb.on2.com>
To: Matthew Heaney <mheaney@on2.com>
Cc: mmusic@ietf.org
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I only submitted the Postscript and the Text version as I was not aware 
until now that you could submit a PDF verison.  You can however find the 
PDF version on RTSP.org at http://www.rtsp.org/drafts/

Aravind

Matthew Heaney wrote:

>Does the -05 version have a .pdf format?  This URL doesn't work:
>
><http://www.ietf.org/internet-drafts/draft-ietf-mmusic-rfc2326bis-05.pdf
>  
>
>
>I can only find the .txt file.
>
>
>
>  
>
>>-----Original Message-----
>>From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com] 
>>Sent: Wednesday, October 29, 2003 3:53 AM
>>To: mmusic@ietf.org
>>Subject: Re: [MMUSIC] I-D 
>>ACTION:draft-ietf-mmusic-rfc2326bis-05.txt,.ps,.pdf
>>
>>
>>Hi,
>>
>>I would like to make you aware of that the 05 text versions 
>>contains a 
>>number of erroneous formatted tables. This is the known occurrences:
>>
>>Page 24, status codes.
>>Page 26, table lines
>>Page 51, Adjusted Packet diagram does not fit on one page.
>>Page 62, 63, Header table.
>>Page 99, Message syntax
>>Page 100, 101, Message syntax
>>page 114, Table 7, init state
>>Page 115, Table 8, Ready State
>>Page 116, Table 9, Play State
>>
>>You will have to live with these errors or read the pdf or PS 
>>version. 
>>Please note that the change bars in the documents are in 
>>relation to the 
>>04 draft version. The major changes are in the methods 
>>chapter where a 
>>lot of rewrite has been performed.
>>
>>We will post a new corrected version after the meeting.
>>
>>Cheers
>>
>>Magnus
>>
>>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		: Real Time Streaming Protocol (RTSP)
>>>	Author(s)	: H. Schulzrinne
>>>	Filename	: draft-ietf-mmusic-rfc2326bis-05.txt,.ps,.pdf
>>>	Pages		: 147
>>>	Date		: 2003-10-27
>>>	
>>>This memorandum is a revision of RFC 2326, which is 
>>>      
>>>
>>currently a Pro- 
>>    
>>
>>>posed Standard. The Real Time Streaming Protocol, or RTSP, is an 
>>>application-level protocol for control over the delivery of 
>>>      
>>>
>>data with 
>>    
>>
>>>real-time properties. RTSP provides an extensible framework 
>>>      
>>>
>>to enable 
>>    
>>
>>>con- trolled, on-demand delivery of real-time data, such as 
>>>      
>>>
>>audio and
>>    
>>
>>>video. Sources of data can include both live data feeds and stored
>>>clips. This protocol is intended to control multiple data delivery
>>>sessions, provide a means for choosing delivery channels 
>>>      
>>>
>>such as UDP,
>>    
>>
>>>multicast UDP and TCP, and provide a means for choosing delivery
>>>mechanisms based upon RTP (RFC 1889).
>>>
>>>A URL for this Internet-Draft is: 
>>>
>>>      
>>>
>>http://www.ietf.org/internet-drafts/draft-ietf-mmusic-rfc2326bis-05.tx
>>    
>>
>>>t
>>>
>>>To remove yourself from the IETF Announcement list, send a 
>>>      
>>>
>>message to
>>    
>>
>>>ietf-announce-request with the word unsubscribe in the body 
>>>      
>>>
>>of the message.
>>    
>>
>>>Internet-Drafts are also available by anonymous FTP. Login with the 
>>>username "anonymous" and a password of your e-mail address. After 
>>>logging in, type "cd internet-drafts" and then
>>>	"get draft-ietf-mmusic-rfc2326bis-05.txt".
>>>
>>>A list of Internet-Drafts directories can be found in 
>>>http://www.ietf.org/shadow.html or 
>>>ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>>>
>>>
>>>Internet-Drafts can also be obtained by e-mail.
>>>
>>>Send a message to:
>>>	mailserv@ietf.org.
>>>In the body type:
>>>	"FILE /internet-drafts/draft-ietf-mmusic-rfc2326bis-05.txt".
>>>	
>>>NOTE:	The mail server at ietf.org can return the document in
>>>	MIME-encoded form by using the "mpack" utility.  To use this
>>>	feature, insert the command "ENCODING mime" before the "FILE"
>>>	command.  To decode the response(s), you will need "munpack" or
>>>	a MIME-compliant mail reader.  Different MIME-compliant 
>>>      
>>>
>>mail readers
>>    
>>
>>>	exhibit different behavior, especially when dealing with
>>>	"multipart" MIME messages (i.e. documents which have been split
>>>	up into multiple messages), so check your local documentation on
>>>	how to manipulate these messages.
>>>		
>>>		
>>>Below is the data which will enable a MIME compliant mail reader 
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>>>Internet-Draft.
>>>
>>>      
>>>
>>-- 
>>
>>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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>mmusic@ietf.org
>https://www1.ietf.org/mailman/listinfo/mmusic
>  
>



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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		: A Transport Independent Bandwidth Modifier for the 
			  Session Description Protocol (SDP)
	Author(s)	: M. Westerlund
	Filename	: draft-ietf-mmusic-sdp-bwparam-05.txt
	Pages		: 20
	Date		: 2003-10-30
	
The existing Session Description Protocol (SDP) bandwidth modifiers
and their values include the bandwidth needed also for the transport
and IP layers. When using SDP with protocols like the Session
Announcement Protocol (SAP), the Session Initiation Protocol (SIP)
and the Real-Time Streaming Protocol (RTSP) and when the involved
hosts reside in networks running different IP versions, the
interpretation of what lower layer bandwidths are included is not
clear. This document defines a bandwidth modifier that does not
include transport overhead; instead an additional packet rate
attribute is defined. The transport independent bit-rate value
together with the maximum packet rate can then be used to calculate
the real bit-rate over the transport actually used.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-bwparam-05.txt

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Date: Fri, 31 Oct 2003 10:43:41 -0800
From: Geetha Srikantan <Geetha.Srikantan@Sun.COM>
Subject: RE: [MMUSIC] RTSP transport negotiation
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To: Enrico Balatti <balatti@cefriel.it>
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hi Enrico,

Thanks for the comments..

That's true - per the current spec we cannot have a Range header in the
SETUP request.
However, if there were no Range header in the SETUP request during play
state, the client would have to either=20
(a) send a PLAY request with a Range header and use the PLAY response=20
    for synchronization, or,
(b) live with the range selected by the server when it receives a SETUP=20
    during play state.

With (a) there is the issue of an extra PLAY request-response. It might
just be better for the client to issue a PAUSE from the old transport
setting, send a SETUP request to modify transport and PLAY to resume
playing from the Range of the client's choosing. Here the transport
negotiation happens in Ready state not Play state.

With (b) there is a high likelihood of synchronization problems when
transport parameters change - RTP packets would get delivered to the old
transport, and discarded; the server has no way of knowing the gap in
time between when it processes the SETUP during play and when the
packets it already sent didn't actually get used by the player. This
implies media glitches at the client are a real possibility.

My inclination is to go with either specifying a Range header in the
SETUP request during play state or send the SETUP request without Range,
follow it up with a PLAY request with Range header. In the latter case
we'd be violating another restriction in the spec about issuing PLAY
requests during Play state, I believe. So this implies the spec would
have to modify restrictions on either the SETUP request or the PLAY
request.

It would be good to get more comments on this, especially from the
authors.

thanks
geetha



On Wed, 2003-10-29 at 06:02, Enrico Balatti wrote:
> Dear Geetha,
>=20
> >... now change client IP address ....
> >
> >     C->S: SETUP rtsp://foo/twister/video RTSP/1.0
> >           CSeq: 5
> >	   Transport: RTP/AVP;unicast;client_ip=3D<new IP address;
> >	              client_port=3D8002-8003;server_port=3D9004-9005
> >	   Range: npt=3D5-
> >           Session: 12345678
>=20
>=20
> I'm not sure the Range field can be aplicated to the SETUP request.
> From the specification it seems the Range is applicable on the SETUP re=
sponse
> but not in the SETUP request.
>=20
>    Header              Where  Proxy DES OPT SETUP PLAY PAUSE TRD =20
>=20
>    Range                 R           -   -    -    o     o   -
>    Range                 r           -   -    c    m*   m*   -
>=20
>=20
> Best Regards,
> Enrico
>=20
>=20
>=20
>=20
> ------------------------------------------------------------------
> Ing Enrico M. Balatti
> CEFRIEL - Politecnico di Milano
> ICT Center of Excellence For Research, Innovation,
> Education & industrial Labs partnership
> via Fucini 2, 20133 Milano (ITALY)
> voice: +39 0223954327 fax: +39 0223954527
> e-mail: balatti@cefriel.it <mailto:balatti@cefriel.it>   www.cefriel.it=
 <http://www.cefriel.it>=20
> ------------------------------------------------------------------
>=20
>=20
> -----Original Message-----
> From: Geetha Srikantan [mailto:Geetha.Srikantan@Sun.COM]
> Sent: mercoled=EC 22 ottobre 2003 7.26
> To: mmusic@ietf.org
> Subject: [MMUSIC] RTSP transport negotiation
>=20
>=20
> Hello everyone,
>=20
> Here is the promised write-up on RTSP transport negotiation..sorry for
> the delay. If there is agreement on the content of this note, I can
> write up text for the RTSP draft.
>=20
> thanks
> geetha
>=20
>=20
> Topic: RTSP Transport negotiation using SETUP during play state.
>=20
> Introduction
>=20
> The RTSP specification allows for negotiation of transport parameters
> via the SETUP method, while the RTSP session is in play state. The
> transport parameters that can be negotiated and the exact form of this
> negotiation is under-specified in the RTSP documents. This note is an
> attempt to clarify some of the issues around transport negotiation for
> inclusion in the ongoing revision of RTSP.
>=20
> The recent interest in mobility for RTSP implementations also impacts
> transport parameters used by the session. For instance, a wireless
> RTSP client could be assigned a new IP address, or may experience
> change of network characteristics, while it is in the midst of a RTSP
> session. These issues are also addressed in this note.
>=20
>=20
> Outline:
>=20
> 1. Objective
> 2. Enumeration of all Transport Parameters
> 3. Transport Parameters Excluded from Negotiation
> 4. Negotiable Transport Parameters
> 5. Impact of Transport Negotiation
> 6. Examples
> 	6.1 Client IP Address Changes
> 	6.2 UDP -> TCP
> 	6.3 TCP -> UDP
> 7. Other transport related changes to RTSP draft
> 8. Summary
>=20
>=20
>=20
> 1. Objective:=20
>=20
> The main goals of this document are to
> 	* Provide an enumeration of transport parameters.
> 	* Identify transport parameters that cannot be negotiated
> 	  with minimal impact on other sessions.
> 	* Propose transport parameters that can be negotiated during
> 	  play state.
> 	* Determine implications of changing these parameters via a
> 	  SETUP request while the session is in play state.
> 	* Provide examples demonstrating transport negotiation.
>=20
>=20
> 2. Enumeration of all Transport Parameters
>=20
>    The following is a list of all transport parameters in the current
>    RTSP draft.
>=20
> 	unicast/multicast
> 	destination
> 	source
> 	mode (=3DPLAY)
> 	interleaved
> 	layers (for multicast)
> 	ttl (for multicast
> 	port (for multicast)
> 	client_port (for RTP)
> 	server_port (for RTP)
> 	ssrc (for RTP)
> 	url
>=20
>=20
> 3. Transport Parameters Excluded from Negotiation
>=20
>    It is recommended that the following parameters be excluded from the
>    negotiation process.
>=20
> 	mode
> 	url
> 	unicast/multicast
> 	layers
> 	ttl
> 	ssrc
> 	destination
> 	source
>=20
>=20
>     As the "mode" parameter is restricted to a single value in the
> current draft, it can be excluded.
>=20
>     As per the recent proposal to not support addition of urls via SETU=
P
> during play state, the "url" parameter can be excluded.=20
>=20
>     Changing from multicast to unicast based on a single RTSP session's
> SETUP, would impact all the other clients accessing the multicast
> session.=20
> Several issues need to be considered in this case, including transport
> negotiation. Similarly, changing from unicast to multicast also has
> implications beyond a single RTSP session. Changes to other
> multicast-related parameters, "layers", "ssrc", "ttl" most likely
> requires specification beyond transport negotiation.
> The recommendation is to *not* specify changing the transport from
> "unicast" to "multicast" or vice versa, and also not specify changes
> to the "layers" and "ttl" parameters, in this draft. It is recommedned=20
> that negotiation of multicast-related transport parameters should be
> deferred to an RTSP-extension document.=20
>=20
>     Once an RTSP session has been initiated and the "destination" and
> "source" have been established - i.e. a multicast "destination" has
> been identified, and the stream "source" has been identified, there
> should be no need to change these parameters.  It is recommended that
> the "destination" and "source" parameters not be negotiated when the
> session is in play state.  The case of the client IP address change is
> signaled in a different manner, outlined in the next section.
>=20
>=20
> 4. Negotiable Transport Parameters
>=20
>    Hence there are three main conditions that require transport
> negotiation for unicast:
>=20
> 	* client IP address changes
> 	* client wants to switch transport from UDP -> TCP=20
> 	* client wants to switch transport from TCP -> UDP
>=20
>    Changes in the client IP address can be indicated using a new
> "client_ip" parameter. Hence the parameters that are impacted by
> transport negotiation are:
>=20
> 	client_ip
> 	interleaved
> 	client_port
> 	server_port
>=20
> =20
> 5. Impact of Transport Negotiation
>=20
>    The first column describes the primary change, the second and third
> columns describe any secondary transport parameters that MUST or=20
> MAY change as a result of the primary change.
>=20
>=20
> Changed			MUST			MAY=09
> -------			----			---
> =09
> Client IP address	client_ip		client_port
> 						server_port
>=20
> UDP -> TCP		interleaved		Server may change
> 			Drop client_port	interleaved channel ids
> 			Drop server_port	in response.
>=20
> TCP -> UDP		client_port in request	client_ip
> 			server_port in response client_port in response=09
> 			Drop interleaved
>=20
>=20
>=20
> 6. Examples
>=20
> 6.1 Client IP Address Changes
>=20
>      C->S: DESCRIBE rtsp://foo/twister.sdp RTSP/1.1
> 	   CSeq: 1
>=20
>      S->C: RTSP/1.0 200 OK
> 	   CSeq: 1
>            Content-Type: application/sdp
> 	   Content-Length: 164
>=20
> 	   v=3D0
>            o=3D- 2890844256 2890842807 IN IP4 172.16.2.93
>            s=3DRTSP Session
>            i=3DAn Example of RTSP Session Usage
>            a=3Dcontrol:rtsp://foo/twister
>            t=3D0 0
>            m=3Daudio 0 RTP/AVP 0
>            a=3Dcontrol:rtsp://foo/twister/audio
>            m=3Dvideo 0 RTP/AVP 26
>            a=3Dcontrol:rtsp://foo/twister/video
>=20
>      C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
>            CSeq: 2
>            Transport: RTP/AVP;unicast;client_port=3D8000-8001
>=20
>      S->C: RTSP/1.0 200 OK
>            CSeq: 2
>            Transport: RTP/AVP;unicast;client_port=3D8000-8001;
>                       server_port=3D9000-9001
>            Session: 12345678
>=20
>      C->S: SETUP rtsp://foo/twister/video RTSP/1.0
>            CSeq: 3
>            Transport: RTP/AVP;unicast;client_port=3D8002-8003
>            Session: 12345678
>=20
>      S->C: RTSP/1.0 200 OK
>            CSeq: 3
>            Transport: RTP/AVP;unicast;client_port=3D8002-8003;
>                       server_port=3D9004-9005
>            Session: 12345678
>=20
>      C->S: PLAY rtsp://foo/twister RTSP/1.0
>            CSeq: 4
>            Range: npt=3D0-
>            Session: 12345678
>=20
>      S->C: RTSP/1.0 200 OK
>            CSeq: 4
>            Session: 12345678
>            RTP-Info: url=3Drtsp://foo/twister/video;
>              seq=3D9810092;rtptime=3D3450012,
> 	   url=3Drtsp://foo/twister/audio;seq=3D5923;rtptime=3D22234432
> 	   Range: npt=3D0-30.76
>=20
> ... now change client IP address ....
>=20
>      C->S: SETUP rtsp://foo/twister/video RTSP/1.0
>            CSeq: 5
> 	   Transport: RTP/AVP;unicast;client_ip=3D<new IP address;
> 	              client_port=3D8002-8003;server_port=3D9004-9005
> 	   Range: npt=3D5-
>            Session: 12345678
> 		=09
>      S->C: RTSP/1.0 200 OK
>            CSeq: 5
> 	   Transport: RTP/AVP;unicast;client_ip=3D<new IP address>;
> 		      client_port=3D8002-8003;server_port=3D9004-9005
> 	   RTP-Info: url=3Drtsp://foo/twister/video;
> 		    seq=3D<new>;rtptime=3D<new>,
> 	   Range: npt=3D5-30.76
> 	   Session: 12345678
>=20
>=20
>      C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
>            CSeq: 6
> 	   Transport:RTP/AVP;unicast;client_ip=3D<new IP address>;
> 		     client_port=3D8000-8001;server_port=3D9000-9001
> 	   Range: npt=3D5-
>            Session: 12345678
>=20
>      S->C: RTSP/1.0 200 OK
>            CSeq: 6
> 	   Transport: RTP/AVP;unicast;client_ip=3D<new IP address>;
> 		      client_port=3D8000-8001;server_port=3D9000-9001
> 	   RTP-Info:url=3Drtsp://foo/twister/audio;
> 		    seq=3D<new>;rtptime=3D<new>
> 	   Range: npt=3D5-30.76=09
> 	   Session: 12345678
>=20
>=20
>=20
>     =20
> 6.2 UDP -> TCP
>=20
>      C->S: DESCRIBE rtsp://foo/twister.sdp RTSP/1.0
> 	   CSeq: 1
>=20
>      S->C: RTSP/1.0 200 OK
> 	   Content-Base: rtsp://foo/twister/
> 	   Content-Length: 230
> 	   Content-Type: application/sdp
> 	   Cseq: 1
>=20
> 	   v=3D0
> 	   o=3D- 3262117360 3262117360 IN IP4 0.0.0.0
> 	   s=3DRTSP Session
> 	   c=3DIN IP4 0.0.0.0
> 	   t=3D0 0
> 	   a=3Dcontrol:*
> 	   a=3Drange:npt=3D0-10.566000
> 	   m=3Dvideo 0 RTP/AVP 26
> 	   a=3Dcontrol:rtsp://foo/twister/video
> 	   m=3Daudio 0 RTP/AVP 0
> 	   a=3Dcontrol:rtsp://foo/twister/audio
>=20
>      C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
> 	   CSeq: 2
> 	   Transport: RTP/AVP/UDP;unicast;client_port=3D8000-8001
>=20
>      S->C: RTSP/1.0 200 OK
> 	   CSeq: 2
> 	   Session: 12345678
> 	   Transport: RTP/ACP/UDP;unicast;client_port=3D8000-8001;
> 		      server_port=3D9000-9001
>=20
>      C->S: PLAY rtsp://foo/twister RTSP/1.0
> 	   CSeq: 3
> 	   Session: 12345678
>=20
>      S->C: RTSP/1.0 200 OK
> 	   Content-Base: rtsp://foo/twister/
> 	   CSeq: 3
> 	   RTP-Info: url=3Drtsp://foo/twister/audio;seq=3D7028;rtptime=3D13399=
49552
> 	   Range: npt=3D0-
> 	   Session: 12345678
>=20
> ... now request TCP ...
>=20
>      C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
> 	   CSeq: 4
> 	   Transport: RTP/AVP/TCP;interleaved=3D0-1
>=20
>      S->C: RTSP/1.0 200 OK
> 	   Cseq: 4
> 	   Transport: RTP/AVP/TCP;unicast;
> 		      client_ip=3D<client IP>;interleaved=3D0-1
> 	   RTP-Info: url=3Drtsp://foo/twister/audio;seq=3D<new>;rtptime=3D<new=
>
> 	   Range: npt=3D0-10.56600
> 	   Session: 12345678
>=20
>=20
>=20
> 6.3 TCP -> UDP
>=20
>      C->S: DESCRIBE rtsp://foo/twister.sdp RTSP/1.0
> 	   CSeq: 1
>=20
>      S->C: RTSP/1.0 200 OK
> 	   Content-Base: rtsp://foo/twister/
> 	   Content-Length: 230
> 	   Content-Type: application/sdp
> 	   Cseq: 1
>=20
> 	   v=3D0
> 	   o=3D- 3262117360 3262117360 IN IP4 0.0.0.0
> 	   s=3DRTSP Session
> 	   c=3DIN IP4 0.0.0.0
> 	   t=3D0 0
> 	   a=3Dcontrol:*
> 	   a=3Drange:npt=3D0-10.566000
> 	   m=3Dvideo 0 RTP/AVP 26
> 	   a=3Dcontrol:rtsp://foo/twister/video
> 	   m=3Daudio 0 RTP/AVP 0
> 	   a=3Dcontrol:rtsp://foo/twister/audio
>=20
>      C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
> 	   CSeq: 2
> 	   Transport: RTP/AVP/TCP;interleaved=3D0-1
>=20
>      S->C: RTSP/1.0 200 OK
> 	   Cseq: 2
> 	   Session: 12345678
> 	   Transport: RTP/AVP/TCP;unicast;client_ip=3D<client IP>;interleaved=3D=
0-1
>=20
>      C->S: PLAY rtsp://foo/twister RTSP/1.0
> 	   CSeq: 3
> 	   Session: 12345678
>=20
>      S->C: RTSP/1.0 200 OK
> 	   Content-Base: rtsp://foo/twister/
> 	   Cseq: 3
> 	   RTP-Info: url=3Drtsp://foo/twister/audio;seq=3D7028;rtptime=3D13399=
49552
> 	   Range: npt=3D0-
> 	   Session: 12345678
>=20
> ... now request UDP ...
>=20
>=20
>      C->S: SETUP rtsp://foo/twister/audio RTSP/1.0
> 	   CSeq: 4
> 	   Transport: RTP/AVP/UDP;unicast;client_port=3D8000-8001
> 	   Range: npt=3D0-
>=20
>      S->C: RTSP/1.0 200 OK
> 	   CSeq: 4
> 	   Transport: RTP/ACP/UDP;unicast;client_port=3D8000-8001;
> 		      server_port=3D9000-9001
> 	   RTP-Info: url=3Drtsp://foo/twister/audio;seq=3D<new>;rtptime=3D<new=
>
> 	   Range: npt=3D0-10.56600
> 	   Session: 12345678
>=20
>=20
>=20
>=20
> 7. Other transport related changes to RTSP draft
>=20
> A few comments on draft-ietf-mmusic-rfc2326bis-04.txt relating to
> transport parameters.
>=20
> (a) Section 13.40:
>     Page 81: Are "src_addr" and "dest_addr" parameters
>     going away? If these are relevant for NAT, perhaps they belong in
>     the NAT document?
>=20
>     Page 83-83: BNF indicates "client_port" and
>     "server_port" will be indicated as 'port1-port2'. This doesn't seem
>     adequate for the case where the RTP and RTCP ports are non-adjacent
>     (per the new RTP spec rfc 3550).=20
>=20
>     Proposal:
> 	client_port: rtp=3D<port>, rtcp=3D<port>
> 	server_port: rtp=3D<port>, rtcp=3D<port>
>=20
> (b) Appendix B, B.1.1
>     Page 115: Mentions "src_addresses" and
>     "dest_addresses" - what are these?
>=20
>     Page 116: "non-continuous" ports - might want to rephrase to
>     "non-adjacent" (or "non-continguous") ports?
>=20
>=20
>=20
> 8. Summary
>=20
>=20
> This note recommends negotiation of a few transport parameters and
> specifies mechanisms to do so. One new transport parameter "client_ip"=20
> is recommended for use in cases where the client's IP address changes
> during the session.=20
>=20
> It is also recommended that the client_port and server_port parameters
> be specified further with the use of tags "rtp_port" and "rtcp_port"
> to handle the case where these ports are non-contiguous.
--=20
Geetha Srikantan <Geetha.Srikantan@Sun.COM>


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