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From: philippe.gentric@philips.com
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Subject: Re: [MMUSIC] RTSP server behind NAT
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: Magnus Westerlund <magnus.westerlund@era.ericsson.se>, mmusic@ietf.org,
        Thomas Zeng <zeng@packetvideo.com>
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>NATs on both sides does not always require a relay on the public
>Internet. It only does if both NATs are symmetric. If at least one is
>full cone, port restricted, etc., then the relay can be avoided. ICE
>will make this determination for you, and only use TURN if connectivity
>cannot be established directly.

I understand: one always need at least one public server,
but in some cases that server does not have to "relay" i.e. process the data,
instead the system is configured so that data flows directly from peer to peer.

> ...consider the model where SIP is used to establish
> the session, and then RTSP is introduced for the control aspects.
.........
>Thoughts?

Implementing SIP looks like a *lot* of work !-)

regards,



Philippe Gentric
Software Architect
Philips MP4Net
philippe.gentric@philips.com
http://www.platform4.philips.com

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Subject: RE: [MMUSIC] RTSP teleconference, have or not?
To: Thomas Zeng <zeng@packetvideo.com>
Cc: Magnus Westerlund <magnus.westerlund@era.ericsson.se>,
        IETF MMUSIC WG <mmusic@ietf.org>, mmusic-admin@ietf.org,
        "'Tom Marshall'" <tmarshall@real.com>
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Date: Tue, 1 Apr 2003 12:09:39 -0500
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I don't mind skipping this Wed's telecon, either.


Andrea Basso
NMS Communications
(732) 936-2118



|---------+---------------------------->
|         |           Thomas Zeng      |
|         |           <zeng@packetvideo|
|         |           .com>            |
|         |           Sent by:         |
|         |           <mmusic-admin@iet|
|         |           f.org>           |
|         |                            |
|         |                            |
|         |           03/31/2003 01:56 |
|         |           PM               |
|         |                            |
|---------+---------------------------->
  >----------------------------------------------------------------------------------------------------------------|
  |                                                                                                                |
  |       To:       "'Tom Marshall'" <tmarshall@real.com>, Magnus Westerlund <magnus.westerlund@era.ericsson.se>   |
  |       cc:       IETF MMUSIC WG <mmusic@ietf.org>                                                               |
  |       Subject:  RE: [MMUSIC] RTSP teleconference, have or not?                                                 |
  >----------------------------------------------------------------------------------------------------------------|




Happy Birtheday, Tom.

I don't mind skipping this Wed's telecon, either. I am behind in RTSP
related work after SF meetings.

- thomas


-----Original Message-----
From: Tom Marshall [mailto:tmarshall@real.com]
Sent: Monday, March 31, 2003 10:18 AM
To: Magnus Westerlund
Cc: IETF MMUSIC WG
Subject: Re: [MMUSIC] RTSP teleconference, have or not?


I would not mind cancelling or postponing the call, as it would be nice to
be able to sleep in on my b'day.  :-)

On Mon, Mar 31, 2003 at 01:10:49PM +0200, Magnus Westerlund wrote:
> Hi,
>
> It was planned to have a RTSP telecon on Wednesday (2 April). I am
> somewhat short of time but I are willing to have this meeting at 18.00
> CET, which after having changed to daylight savings are GMT+2. As I
> understand it US are changing to daylight saving next sunday and
> therefore the conference will be one hour earlier than usual. This I
> have heard is a bit difficult for the people on the US west cost. Due to
> these considerations I would like to ask your opinion on what to do. Are
> we going to:
> 1. Cancel the meeting and having it next week instead?
> 2. Have the meeting at the proposed time.
>
> Please send me your view before 12.00 tuesday (CET).

--
Coding is easy; All you do is sit staring at a terminal until the drops of
blood form on your forehead.
        -- Anonymous
_______________________________________________
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https://www1.ietf.org/mailman/listinfo/mmusic





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Subject: RE: [MMUSIC] RTSP teleconference, have or not?
From: Rob Lanphier <robla@real.com>
To: mmusic@ietf.org,
        Magnus WesterlundMagnusWesterlund <magnus.westerlund@era.ericsson.se>
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Agreed.  Should we just plan on April 16 for our next call?

Rob

On Tue, 2003-04-01 at 09:09, Andrea Basso wrote:
> I don't mind skipping this Wed's telecon, either.
> 
> 
> Andrea Basso
> NMS Communications
> (732) 936-2118
> 
> 
> 
> |---------+---------------------------->
> |         |           Thomas Zeng      |
> |         |           <zeng@packetvideo|
> |         |           .com>            |
> |         |           Sent by:         |
> |         |           <mmusic-admin@iet|
> |         |           f.org>           |
> |         |                            |
> |         |                            |
> |         |           03/31/2003 01:56 |
> |         |           PM               |
> |         |                            |
> |---------+---------------------------->
>   >----------------------------------------------------------------------------------------------------------------|
>   |                                                                                                                |
>   |       To:       "'Tom Marshall'" <tmarshall@real.com>, Magnus Westerlund <magnus.westerlund@era.ericsson.se>   |
>   |       cc:       IETF MMUSIC WG <mmusic@ietf.org>                                                               |
>   |       Subject:  RE: [MMUSIC] RTSP teleconference, have or not?                                                 |
>   >----------------------------------------------------------------------------------------------------------------|
> 
> 
> 
> 
> Happy Birtheday, Tom.
> 
> I don't mind skipping this Wed's telecon, either. I am behind in RTSP
> related work after SF meetings.
> 
> - thomas
> 
> 
> -----Original Message-----
> From: Tom Marshall [mailto:tmarshall@real.com]
> Sent: Monday, March 31, 2003 10:18 AM
> To: Magnus Westerlund
> Cc: IETF MMUSIC WG
> Subject: Re: [MMUSIC] RTSP teleconference, have or not?
> 
> 
> I would not mind cancelling or postponing the call, as it would be nice to
> be able to sleep in on my b'day.  :-)
> 
> On Mon, Mar 31, 2003 at 01:10:49PM +0200, Magnus Westerlund wrote:
> > Hi,
> >
> > It was planned to have a RTSP telecon on Wednesday (2 April). I am
> > somewhat short of time but I are willing to have this meeting at 18.00
> > CET, which after having changed to daylight savings are GMT+2. As I
> > understand it US are changing to daylight saving next sunday and
> > therefore the conference will be one hour earlier than usual. This I
> > have heard is a bit difficult for the people on the US west cost. Due to
> > these considerations I would like to ask your opinion on what to do. Are
> > we going to:
> > 1. Cancel the meeting and having it next week instead?
> > 2. Have the meeting at the proposed time.
> >
> > Please send me your view before 12.00 tuesday (CET).
> 
> --
> Coding is easy; All you do is sit staring at a terminal until the drops of
> blood form on your forehead.
>         -- Anonymous
> _______________________________________________
> 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
-- 
Rob Lanphier, Helix Community Coordinator - RealNetworks
http://helixcommunity.org http://rtsp.org http://realnetworks.com

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Subject: RE: [MMUSIC] RTSP teleconference, have or not?
To: Rob Lanphier <robla@real.com>
Cc: Magnus WesterlundMagnusWesterlund <magnus.westerlund@era.ericsson.se>,
        mmusic@ietf.org, mmusic-admin@ietf.org
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From: "Andrea Basso" <Andrea_Basso@nmss.com>
Date: Tue, 1 Apr 2003 15:18:52 -0500
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16 of April  is fine with me.

-A

Andrea Basso
NMS Communications
(732) 936-2118



                                                                                                                   
                      Rob Lanphier                                                                                 
                      <robla@real.com>         To:       <mmusic@ietf.org>, Magnus WesterlundMagnusWesterlund      
                      Sent by:                  <magnus.westerlund@era.ericsson.se>                                
                      <mmusic-admin@iet        cc:                                                                 
                      f.org>                   Subject:  RE: [MMUSIC] RTSP teleconference, have or not?            
                                                                                                                   
                                                                                                                   
                      04/01/2003 12:31                                                                             
                      PM                                                                                           
                                                                                                                   
                                                                                                                   




Agreed.  Should we just plan on April 16 for our next call?

Rob

On Tue, 2003-04-01 at 09:09, Andrea Basso wrote:
> I don't mind skipping this Wed's telecon, either.
>
>
> Andrea Basso
> NMS Communications
> (732) 936-2118
>
>
>
> |---------+---------------------------->
> |         |           Thomas Zeng      |
> |         |           <zeng@packetvideo|
> |         |           .com>            |
> |         |           Sent by:         |
> |         |           <mmusic-admin@iet|
> |         |           f.org>           |
> |         |                            |
> |         |                            |
> |         |           03/31/2003 01:56 |
> |         |           PM               |
> |         |                            |
> |---------+---------------------------->
>   >
----------------------------------------------------------------------------------------------------------------|

>   |
|
>   |       To:       "'Tom Marshall'" <tmarshall@real.com>, Magnus
Westerlund <magnus.westerlund@era.ericsson.se>   |
>   |       cc:       IETF MMUSIC WG <mmusic@ietf.org>
|
>   |       Subject:  RE: [MMUSIC] RTSP teleconference, have or not?
|
>   >
----------------------------------------------------------------------------------------------------------------|

>
>
>
>
> Happy Birtheday, Tom.
>
> I don't mind skipping this Wed's telecon, either. I am behind in RTSP
> related work after SF meetings.
>
> - thomas
>
>
> -----Original Message-----
> From: Tom Marshall [mailto:tmarshall@real.com]
> Sent: Monday, March 31, 2003 10:18 AM
> To: Magnus Westerlund
> Cc: IETF MMUSIC WG
> Subject: Re: [MMUSIC] RTSP teleconference, have or not?
>
>
> I would not mind cancelling or postponing the call, as it would be nice
to
> be able to sleep in on my b'day.  :-)
>
> On Mon, Mar 31, 2003 at 01:10:49PM +0200, Magnus Westerlund wrote:
> > Hi,
> >
> > It was planned to have a RTSP telecon on Wednesday (2 April). I am
> > somewhat short of time but I are willing to have this meeting at 18.00
> > CET, which after having changed to daylight savings are GMT+2. As I
> > understand it US are changing to daylight saving next sunday and
> > therefore the conference will be one hour earlier than usual. This I
> > have heard is a bit difficult for the people on the US west cost. Due
to
> > these considerations I would like to ask your opinion on what to do.
Are
> > we going to:
> > 1. Cancel the meeting and having it next week instead?
> > 2. Have the meeting at the proposed time.
> >
> > Please send me your view before 12.00 tuesday (CET).
>
> --
> Coding is easy; All you do is sit staring at a terminal until the drops
of
> blood form on your forehead.
>         -- Anonymous
> _______________________________________________
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>
>
>
>
>
> _______________________________________________
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> https://www1.ietf.org/mailman/listinfo/mmusic
--
Rob Lanphier, Helix Community Coordinator - RealNetworks
http://helixcommunity.org http://rtsp.org http://realnetworks.com

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Okay,

Next RTSP teleconference the 16th of April at 18.00 CET (GMT+2).

Best Regards

Magnus Westerlund 

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



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Hi,

I understand that there is interest to fix RTSP's NAT traversal 
capabilities to handle cases when both parties using private address 
spaces with NATs between them. This will require work from the one 
interested in this. With these assumptions you must also look into the 
problem of how to connect with TCP to the RTSP server in the first 
place. This seem to require RTSP proxies on the borders between 
different address domains. However I assume that a lot of these 
information can be had by looking how SIP solves the problem.

I have some comments about TURN and RTSP. On the way to San Francisco I 
managed to analyze TURN and how it fits with RTSP. The major hurdle here 
is that the media destination address will be totally different from 
where the TCP connection comes from. If we are going to continue with 
our assumptions about when it is okay to send to another address we need 
the mechanism to determine that the destination accepts the data.

When it comes to a solution of the problem of having servers on NATed 
addresses the ones doing the work will need to look into how to solves 
this problem. There seem to exist a couple of different ways to solve 
the problem.
1. Incorporate ICE into RTSP SETUP procedures. Includes need for doing 
multiple ICE rounds. This is similar to the Offer Answer problem for RTSP.
2. change RTSP to use SIP style session initiation. This work I think is 
similar to actually defining RTSP 2.0 and might be a rather grand work. 
I see need for having to determine which part of SIP that is needed. 
What extensions in SIP that will be needed to allow the media control 
part of RTSP to work as before. Also the whole integration between a 
normal conversational service and media control services needs to be 
looked into. I don't think it is a simple problem. Otherwise IETF would 
not have taken this time to even come close to a working conference 
solution.
3. Other alternatives. Does it exist other possible solutions?

My conclusion is that the peer-to-peer RTSP over NATed networks will not 
be solved in a short time frame. And there is need to solve how to 
handle RTSP when only the client is in a private network. My proposal 
would be to progress the work simultaneously. I strongly believe that we 
can create something that work in time frame of the updated RTSP 
specification for public servers. There is of course important to align 
the work and ensure that either the whole public server work is a 
reusable for the peer-to-peer work, or at least does not create 
problems. I know that a single solution are normally better, but I think 
market forces will again make themselves remembered.

Best Regards

Magnus





Jonathan Rosenberg wrote:

>
>
> philippe.gentric@philips.com wrote:
> > Magnus, I concur with Thomas,
> >
> > Since we have telephone handsets that can encode video in real time,
> > Since we have PCs at home with WebCams, Since all these are behind
> > NATs, Since multicast (and SSM) will not be
> > available/desirable/permitted in many cases, We need to make sure
> > that a  RTSP server can exist in these environements, as RTSP is the
> > obvious peer-to-peer video protocol to use.
> >
> > I suspect the key issue is that we have to assume that both end
> > points (peers) are behind NATs and then I am afraid that the solution
> > requires a "public" server in the middle.
> >
> > At this moment I suspect that TURN is one possible candidate(?)
>
> NATs on both sides does not always require a relay on the public 
> Internet. It only does if both NATs are symmetric. If at least one is 
> full cone, port restricted, etc., then the relay can be avoided. ICE 
> will make this determination for you, and only use TURN if 
> connectivity cannot be established directly.

>
> >
> > Some kind of a "reversed" RTSP relay could also do the trick;
> >> From the second peer (receiver of streams) point on view that relay
> >> would be a plain RTSP server.
> > The special trick is that the first peer (origin of streams) would
> > need to contact this public gateway to "expose" that it has something
> > to stream... And here comes the "reversed" weird stuff: That first
> > peer acts as a RTSP server but it *initiates* the first TCP
> > connection to the RTSP relay and that relay acts as a RTSP client but
> > it is listening on some TCP port...
> >
> > It assumes that there is some way for the first peer to tell the
> > second peer that the streams are available at that relay server.
> > Email could do that, or a voice call, or pigeons ;-)
> >
> > Not sure if/how  this is similar/different to what is intended with
> > TURN ?
>
> TURN is similar, in that both sides need to rendezvous at a central 
> server. However, only one side needs to know about TURN. The other 
> will rendezvous by simply sending media to the TURN server.
>
>
> >
> > Jonathan can you explain what good TURN is supposed to do to RTSP ?
>
> It would support the case of the RTSP client and server both behind 
> separate, symmetric NAT.
>
> To support it, you need only make sure that RTSP allows the client to 
> specify the IP/port on which it will receive media. This is the same 
> thing you would need to do to support STUN.
>
> Getting back to this general issue of peer-to-peer RTSP, in my view 
> this is the ideal case to consider the model where SIP is used to 
> establish the session, and then RTSP is introduced for the control 
> aspects. This way, RTSP does not have to reinvent all the work that is 
> being done in SIP for nat/fw traversal, and all the applications where 
> the media streaming is integrated with person to person communications 
> is very easy.
>
> I think that there are no changes to RTSP to support this. We would 
> need some kind of SIP or SDP extensions to signal the usage of RTSP as 
> a control for another media stream, and then use SDP extensions to 
> pass any RTSP information needed to make this work (for example, the 
> session ID).
>
> Thoughts?
>
> -Jonathan R.
>

-- 

Magnus Westerlund 

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



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Hi,

And no RTSP meeting tonight/today.

Best Regards

Magnus

Magnus Westerlund wrote:

> Okay,
>
> Next RTSP teleconference the 16th of April at 18.00 CET (GMT+2).
>
> Best Regards
>
> Magnus Westerlund
> Multimedia Technologies, Ericsson Research ERA/TVA/A
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
>

-- 

Magnus Westerlund 

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



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TVoDSL 2004 CONFERENCE
Broadcasting TV content over ADSL

As ADSL access becomes more widespread, new markets for this technology =
will start to materialize.=20
TV over ADSL is one of them. Major manufacturing, economic and =
regulatory players in this area will share their experience at the =
TVoDSL Conference, to take place in Paris on January 20 to 23, 2004.=20
=20
A call for proposal is online at:
http://www.upperside.fr/tvodsl04/tvodsl04intro.htm

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ADSL</SPAN><BR><FONT class=3Dtextebold></FONT><BR></FONT><FONT =
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start to=20
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Subject: [MMUSIC] Last Call: RTCP attribute in SDP to Proposed Standard
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The IESG has received a request from the Multiparty Multimedia Session 
Control Working Group to consider RTCP attribute in SDP 
<draft-ietf-mmusic-sdp4nat-03.txt> as a Proposed Standard.  

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send any comments to the 
iesg@ietf.org or ietf@ietf.org mailing lists by 2003-4-14.

Files can be obtained via http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp4nat-03.txt



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From: Tom Marshall <tmarshall@real.com>
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Subject: [MMUSIC] RTSP ALGs, STUN, and destination parameter
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I'm writing an RTSP ALG for Linux iptables, and came across the problem of
handling the destination parameter.  There are several scenarios:

0. destination parameter not present.  Not a problem.

1. Client is using STUN.

   destination parameter will be the client's internal address.

   It must not be rewritten, or streaming will fail.

2. Client is not using STUN, just being verbose.

   destination parameter will be the client's internal address.

   It must be rewritten to the external NAT address or stripped out, or
   streaming will fail.

3. Client wants the data to be sent somewhere else.

   destination parameter will not be the client's internal address.

   Three cases here:
     3a. Client has dual nics.
     3b. Client wants to send the data to another machine behind the NAT.
     3c. Client wants to send the data to another machine on the internet.

   I don't think one can distinguish 3a and 3b so they can be considered as
   the same problem.  The best way to deal with this case is probably to let
   the administrator decide with a security setting.  Possible values are:

     - Always strip destination parameter.  Not recommended.
     - Strip destination parameter if it does not match client's internal
       address.  This is the default and recommended setting.
     - Never strip the destination parameter.  Not recommended.

This is all pretty straightforward so far.  Now comes the tricky part.  What
to do with the address in the destination parameter as it passes through the
NAT (assuming it's not stripped out)?  I need a way to distinguish between
cases 1 and 2 above to decide.  If I arbitrarily choose to rewrite the
address or not, the other case will be broken.

Does anyone have any suggestions?

--=20
Write a wise saying and your name will live forever.
        -- Anonymous

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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Subject: Re: [MMUSIC] RTSP ALGs, STUN, and destination parameter
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<rant>I can't help pointing out that this kind of problem is exactly why 
ALGs are problematic. They are attempting to interpret the intention of 
the user so that they can "fix" the message to maintain the desired 
intention of the user in the presence of a nat.</rant>

a more pragmatic comment inline:



Tom Marshall wrote:
> I'm writing an RTSP ALG for Linux iptables, and came across the problem
> of
> handling the destination parameter.  There are several scenarios:
> 
> 0. destination parameter not present.  Not a problem.
> 
> 1. Client is using STUN.
> 
>    destination parameter will be the client's internal address.
> 
>    It must not be rewritten, or streaming will fail.

If the client is using STUN, the destination parameter will not be an 
internal address, it will be an external one. If the NAT knows which 
addresses are internal and which are external, it can differentiate case 
1 from case 2 in this way.

> 
> 2. Client is not using STUN, just being verbose.
> 
>    destination parameter will be the client's internal address.
> 
>    It must be rewritten to the external NAT address or stripped out, or
>    streaming will fail.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Scientist                             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: Tom Marshall <tmarshall@real.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] RTSP ALGs, STUN, and destination parameter
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On Wed, Apr 02, 2003 at 05:12:54PM -0500, Jonathan Rosenberg wrote:
> <rant>I can't help pointing out that this kind of problem is exactly why=
=20
> ALGs are problematic. They are attempting to interpret the intention of=
=20
> the user so that they can "fix" the message to maintain the desired=20
> intention of the user in the presence of a nat.</rant>

Remember, STUN is a very new idea and it works by using some knowledge of
common NAT implementations.  Are going to volunteer to fix all the existing
clients that are out in the field to use STUN and/or Symmetric RTP?  ;-)

Additionally, even if you have the asbolute best STUN (or Symmetric RTP)
client and server imaginable, it is still possible that you won't be able to
punch a hole in a NAT firewall.  Why not?  Because they depend on the NAT
firewall setting up a "connection" when it sees outbound UDP packets.  If
the NAT firewall is sufficiently secure and it requires that UDP packets are
related to an existing TCP connection, it won't work.

> Tom Marshall wrote:
> >I'm writing an RTSP ALG for Linux iptables, and came across the problem
> >of
> >handling the destination parameter.  There are several scenarios:
> >
> >0. destination parameter not present.  Not a problem.
> >
> >1. Client is using STUN.
> >
> >   destination parameter will be the client's internal address.
> >
> >   It must not be rewritten, or streaming will fail.
>=20
> If the client is using STUN, the destination parameter will not be an=20
> internal address, it will be an external one. If the NAT knows which=20
> addresses are internal and which are external, it can differentiate case=
=20
> 1 from case 2 in this way.

Good idea.  I'll see if this is possible, thanks.

--=20
One is not superior merely because one sees the world as odious.
        -- Chateaubriand (1768-1848)

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From: Thomas Zeng <zeng@PacketVideo.COM>
To: "'Magnus Westerlund'" <magnus.westerlund@era.ericsson.se>, mmusic@ietf.org
Subject: RE: [MMUSIC] RTSP server behind NAT
Date: Thu, 3 Apr 2003 12:11:14 -0800 
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Magnus:

On the high level,  I agree that the first focus of RTSP taskforce should be
to solve the easy problem: RTSP server in the open
IP public network, clients are behind NAT.

- thomas

-----Original Message-----
From: Magnus Westerlund [mailto:magnus.westerlund@era.ericsson.se]
Sent: Tuesday, April 01, 2003 11:59 PM
To: mmusic@ietf.org
Subject: Re: [MMUSIC] RTSP server behind NAT


Hi,

I understand that there is interest to fix RTSP's NAT traversal 
capabilities to handle cases when both parties using private address 
spaces with NATs between them. This will require work from the one 
interested in this. With these assumptions you must also look into the 
problem of how to connect with TCP to the RTSP server in the first 
place. This seem to require RTSP proxies on the borders between 
different address domains. However I assume that a lot of these 
information can be had by looking how SIP solves the problem.

I have some comments about TURN and RTSP. On the way to San Francisco I 
managed to analyze TURN and how it fits with RTSP. The major hurdle here 
is that the media destination address will be totally different from 
where the TCP connection comes from. If we are going to continue with 
our assumptions about when it is okay to send to another address we need 
the mechanism to determine that the destination accepts the data.

When it comes to a solution of the problem of having servers on NATed 
addresses the ones doing the work will need to look into how to solves 
this problem. There seem to exist a couple of different ways to solve 
the problem.
1. Incorporate ICE into RTSP SETUP procedures. Includes need for doing 
multiple ICE rounds. This is similar to the Offer Answer problem for RTSP.
2. change RTSP to use SIP style session initiation. This work I think is 
similar to actually defining RTSP 2.0 and might be a rather grand work. 
I see need for having to determine which part of SIP that is needed. 
What extensions in SIP that will be needed to allow the media control 
part of RTSP to work as before. Also the whole integration between a 
normal conversational service and media control services needs to be 
looked into. I don't think it is a simple problem. Otherwise IETF would 
not have taken this time to even come close to a working conference 
solution.
3. Other alternatives. Does it exist other possible solutions?

My conclusion is that the peer-to-peer RTSP over NATed networks will not 
be solved in a short time frame. And there is need to solve how to 
handle RTSP when only the client is in a private network. My proposal 
would be to progress the work simultaneously. I strongly believe that we 
can create something that work in time frame of the updated RTSP 
specification for public servers. There is of course important to align 
the work and ensure that either the whole public server work is a 
reusable for the peer-to-peer work, or at least does not create 
problems. I know that a single solution are normally better, but I think 
market forces will again make themselves remembered.

Best Regards

Magnus





Jonathan Rosenberg wrote:

>
>
> philippe.gentric@philips.com wrote:
> > Magnus, I concur with Thomas,
> >
> > Since we have telephone handsets that can encode video in real time,
> > Since we have PCs at home with WebCams, Since all these are behind
> > NATs, Since multicast (and SSM) will not be
> > available/desirable/permitted in many cases, We need to make sure
> > that a  RTSP server can exist in these environements, as RTSP is the
> > obvious peer-to-peer video protocol to use.
> >
> > I suspect the key issue is that we have to assume that both end
> > points (peers) are behind NATs and then I am afraid that the solution
> > requires a "public" server in the middle.
> >
> > At this moment I suspect that TURN is one possible candidate(?)
>
> NATs on both sides does not always require a relay on the public 
> Internet. It only does if both NATs are symmetric. If at least one is 
> full cone, port restricted, etc., then the relay can be avoided. ICE 
> will make this determination for you, and only use TURN if 
> connectivity cannot be established directly.

>
> >
> > Some kind of a "reversed" RTSP relay could also do the trick;
> >> From the second peer (receiver of streams) point on view that relay
> >> would be a plain RTSP server.
> > The special trick is that the first peer (origin of streams) would
> > need to contact this public gateway to "expose" that it has something
> > to stream... And here comes the "reversed" weird stuff: That first
> > peer acts as a RTSP server but it *initiates* the first TCP
> > connection to the RTSP relay and that relay acts as a RTSP client but
> > it is listening on some TCP port...
> >
> > It assumes that there is some way for the first peer to tell the
> > second peer that the streams are available at that relay server.
> > Email could do that, or a voice call, or pigeons ;-)
> >
> > Not sure if/how  this is similar/different to what is intended with
> > TURN ?
>
> TURN is similar, in that both sides need to rendezvous at a central 
> server. However, only one side needs to know about TURN. The other 
> will rendezvous by simply sending media to the TURN server.
>
>
> >
> > Jonathan can you explain what good TURN is supposed to do to RTSP ?
>
> It would support the case of the RTSP client and server both behind 
> separate, symmetric NAT.
>
> To support it, you need only make sure that RTSP allows the client to 
> specify the IP/port on which it will receive media. This is the same 
> thing you would need to do to support STUN.
>
> Getting back to this general issue of peer-to-peer RTSP, in my view 
> this is the ideal case to consider the model where SIP is used to 
> establish the session, and then RTSP is introduced for the control 
> aspects. This way, RTSP does not have to reinvent all the work that is 
> being done in SIP for nat/fw traversal, and all the applications where 
> the media streaming is integrated with person to person communications 
> is very easy.
>
> I think that there are no changes to RTSP to support this. We would 
> need some kind of SIP or SDP extensions to signal the usage of RTSP as 
> a control for another media stream, and then use SDP extensions to 
> pass any RTSP information needed to make this work (for example, the 
> session ID).
>
> Thoughts?
>
> -Jonathan R.
>

-- 

Magnus Westerlund 

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



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I've written an RTSP conntrack/nat module for Linux's netfilter.  Please see
the url below for more information, including the download link:

	http://lists.netfilter.org/pipermail/netfilter-devel/2003-April/010949.html

I would appreciate as much testing and feedback as possible.  Thanks.

--=20
A candidate is a person who gets money from the rich and votes from the poor
to protect them from each other.
        -- Unknown

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--NextPart

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


	Title		: XML Schema for Media Control
	Author(s)	: O. Levin, S. Olson, R. Even
	Filename	: draft-levin-mmusic-xml-media-control-02.txt
	Pages		: 10
	Date		: 2003-4-7
	
This document defines an XML Schema for Media Control in a tightly 
controlled environment. The current version includes commands for 
managing of video streams only. Implementation of this schema for 
interactive video applications in SIP environments significantly 
improves user experience.

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rfc2326 defines the Connection header.  I have some questions about
implementation.

1. Table 3 says that the header is a general header, it is required to be
   supported, and it applies to all messages (presumedly this means both
   requests and responses).  When is the server required to include a
   Connection header in a response?  Always?  At its discretion?  When the
   client has sent one in the corresponding request?

2. Per section 12.10, the Connection header is defined in rfc2068 14.10.  It
   says this:

     The Connection header has the following grammar:

            Connection-header =3D "Connection" ":" 1#(connection-token)
            connection-token  =3D token

     HTTP/1.1 proxies MUST parse the Connection header field before a
     message is forwarded and, for each connection-token in this field,
     remove any header field(s) from the message with the same name as the
     connection-token.  Connection options are signaled by the presence of a
     connection-token in the Connection header field, not by any
     corresponding additional header field(s), since the additional header
     field may not be sent if there are no parameters associated with that
     connection option.


   What exactly does this mean?  It appears to mean that if the proxy gets a
   message with "Connection: close, foo" and "foo: bar" headers, it must
   remove the "foo: bar" header.  Is this correct?

   This says to me that we can have something like the following:

     Connection: Timed-Close
     Timed-Close: 30

   The above (fictitious) example might indicate to a server that the client
   intends to close the control channel in 30 seconds.  Is this a valid
   interpretation?  What headers are excluded from this behavior (imagine
   someone sending "Connection: Content-Length")?

--=20
You can complain because roses have thorns, or you can rejoice because
thorns have roses.
        -- Abraham Lincoln

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From: Colin Perkins <csp@csperkins.org>
Subject: [MMUSIC] Comments on draft-ietf-mmusic-kmgmt-ext-07.txt
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I've just completed a detailed read-through the the key management
extensions draft, and have a number of comments:

 - In section 3.1, first paragraph after bullet points on page 6, what
   happens if the key management protocol accepts the data in some
   attributes, and rejects the others? For example if a media level 
   key management attribute is valid but the session level isn't? 

 - In the second paragraph on page 8, does "The offer MUST then abort the
   security setup" mean that the entire session is aborted? Or does it go
   ahead without security?

 - I don't understand the discussion of resynchronization in the last
   paragraph of section 3.2. Can you clarify what is intended?

 - Given the risks identified in the third paragraph of section 5, it's not
   clear to me why the entire SDP message doesn't need to be authenticated?
   Are attacks possible by manipulating the non-authenticated parts of the
   SDP?

I also have a number of editorial suggestions, which I'll forward to the 
authors.

Colin
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From: "Fredrik Lindholm (EAB)" <Fredrik.Lindholm@era.ericsson.se>
To: "'Colin Perkins'" <csp@csperkins.org>
Cc: mmusic@ietf.org
Subject: RE: [MMUSIC] Comments on draft-ietf-mmusic-kmgmt-ext-07.txt
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Hi Colin,

see comments inline.

> -----Original Message-----
> From: Colin Perkins [mailto:csp@csperkins.org]
> Sent: den 9 april 2003 06:45
> To: mmusic@ietf.org
> Subject: [MMUSIC] Comments on draft-ietf-mmusic-kmgmt-ext-07.txt
> 
> 
> I've just completed a detailed read-through the the key management
> extensions draft, and have a number of comments:
> 
>  - In section 3.1, first paragraph after bullet points on page 6, what
>    happens if the key management protocol accepts the data in some
>    attributes, and rejects the others? For example if a media level 
>    key management attribute is valid but the session level isn't? 
> 
>  - In the second paragraph on page 8, does "The offer MUST 
> then abort the
>    security setup" mean that the entire session is aborted? 
> Or does it go
>    ahead without security?

To answer both these questions: session and media level failures 
should be treated the same way, i.e., the entire session is 
aborted (then it is up to the local policy to decide what to do). 

> 
>  - I don't understand the discussion of resynchronization in the last
>    paragraph of section 3.2. Can you clarify what is intended?
>

Basically, if you need to change keys during an ongoing session, 
we propose to send a new INVITE message (with new crypto parameters
inside).
 
>  - Given the risks identified in the third paragraph of 
> section 5, it's not
>    clear to me why the entire SDP message doesn't need to be 
> authenticated?

I have never understood why people don't apply security to things :-) 
However, when it comes to SDP, the protection of the SDP will 
very much depend on the protocol carrying SDP and what security 
features are implemented (and defined as necessary). So it is out 
of the scope of this document to specify how to protect the SDP, 
and that is also why we only give a recommendation that you should 
apply some form of protection to the SDP.

>    Are attacks possible by manipulating the non-authenticated 
> parts of the
>    SDP?

If you, by attacks, are referring to attacks towards the key management 
and the secured media, I would say no.


Fredrik

> 
> I also have a number of editorial suggestions, which I'll 
> forward to the 
> authors.
> 
> Colin
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 
_______________________________________________
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Subject: [MMUSIC] CFP IEEE/IFIP Intern. Conference on Management of Multimedia Networks & Services 2003
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Dear Colleagues,=20
Apologize for any duplicates;

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

 Please distribute to interested people to encourage contributions ans =
submissions to the 6th IFIP/IEEE International Conference on Management =
of Multimedia Networks and Services :
        7th - 10th September 2003, Queen's University of Belfast, =
Northern Ireland=20
                            http://www.ee.qub.ac.uk/mmns2003 .
-------------------------------------------------------------------

Please note the extension to the deadline for submission of papers to =
MMNS2003 is 18th April 2003. Please circulate to your colleagues to =
ensure a good turnout.

For further information about the conference please see the web site at =
http://ee-server.ee.qub.ac.uk/dsp/mmns2003/

Best Regards,

Ahmed Mehaoua
(Publicity chair)


 =
-------------------------------------------------------------------------=
--------------------------------

MMNS 2003
6th IFIP/IEEE International Conference on Management of Multimedia =
Networks and Services
        7th - 10th September 2003, Queen's University of Belfast, =
Northern Ireland=20
                            http://www.ee.qub.ac.uk/mmns2003=20


                                  Second Call For Papers

Multimedia services over IP networks are proliferating at an enormous =
speed. There=20
is also increasing demand for solutions which provide assured levels of =
service quality.=20
All of these require novel paradigms, models and architectures for =
realising integrated=20
end-to-end service management rather than managing network elements in =
isolation.=20
Providing scalable Quality of Service (QoS) while maintaining fairness, =
along with=20
secure and optimal network resource management are key challenges for =
the future=20
Internet. These challenges apply to both fixed and wireless networks.=20

The IFIP/IEEE International Conference on Management of Multimedia =
Networks and Services=20
will hold its sixth annual meeting from September 7th to September 10th, =
2003 in Belfast,
Northern Ireland. MMNS provides an intimate setting for discussion and =
debate. In just 6 years,=20
MMNS has established itself as one of the premier conferences with a =
focus on the management=20
of multimedia networks and services. The conference objective is to =
bring together researchers=20
working in all facets of network and service management as applied to =
broadband networks and=20
multimedia services. MMNS deals with all aspects of designing, =
developing and deploying=20
networked multimedia systems and it serves as a forum for the =
dissemination of state-of-the-art=20
research and development results.=20

MMNS 2003 will also include panel sessions in which experts=20
offer their observations and opinions about current hot topics. The =
keynote speaker will be=20
Professor Ian Akyildiz, Georgia Institute of Technology who will present =
a vision of future=20
interplanetary network architectures. Professor Derek McAuley, head of =
Intel's recently formed=20
laboratory at Cambridge, UK, will describe some of the new research =
being undertaken on global=20
overlay networks and applications.


The program committee is soliciting original papers describing research =
in the area of=20
management of multimedia networks and services. Topics of interest =
include, but are not=20
limited to, the following:

        * Active multimedia network management=20
        * Ad-hoc and Sensor Networks=20
        * Augmented and Virtual Reality Networks=20
        * Billing and Accounting=20
        * Cable multimedia network management=20
        * Content distribution internetworking=20
        * Deployment of multimedia services=20
        * Distributed multimedia service management=20
        * End-to-end IP multimedia network and service management=20
        * IP Video, streaming, interactive video service management=20
        * Middleware support for management=20
        * Multimedia network traffic engineering and optimization=20
        * Multimedia traffic management=20
        * Multimedia content protection=20
        * Multimedia session management=20
        * Multi-point, multicast services management=20
        * Network management models and architectures=20
        * Network programmability for multimedia services=20
        * Optical multimedia network management=20
        * Policy-based management for multi-media services=20
        * Provisioning of multimedia networks and services=20
        * QoS in WLANs=20
        * QoS management=20
        * Resource, performance and fault management=20
        * Security and Authentication=20
        * VoIP service management=20
        * Web Services=20
        * Wireless and mobile multimedia network management=20


Papers must be submitted electronically in postscript or PDF format. =
Detailed=20
instructions are provided on the conference web site, =
http://www.ee.qub.ac.uk/mmns2003.=20

Submission date:            18th April 2003
Notification of acceptance: 6th June 2003
Final version:              4th July 2003

Conference Chairs:
   Professor Alan Marshall, a.marshall@ee.qub.ac.uk, Queen's University =
of Belfast, UK
   Professor Nazim Agoulmine, nazim@rp.lip6.fr, University of Evry, =
France


Qiang Gu
Advanced Telecommunication Systems Laboratory
School of Electrical & Electronic Engineering
The Queen's University of Belfast
Belfast
Northern Ireland.
BT9 5AH
Email: qiang.gu@ee.qub.ac.uk
Telephone: +44 -2890-274142
Fax: +44 -2890-274417
---------------------------------------------------------
Ahmed Mehaoua
University of Versailles - CNRS PRISM Lab.
45 av. des etats unis 78000 Versailles - France
Email : mea@prism.uvsq.fr
Tel : +33 1 39 25 40 45



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<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>
<DIV><FONT face=3DArial size=3D2>
<DIV><FONT face=3DArial size=3D2>Dear Colleagues, </FONT></DIV>
<DIV><FONT face=3D"Times New Roman" size=3D3>Apologize&nbsp;for any=20
duplicates;</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><FONT face=3D"Times New Roman"=20
size=3D3>----------------------------------------------------------------=
---<BR><BR>&nbsp;Please=20
distribute to interested people to encourage contributions ans =
submissions to=20
the&nbsp;<FONT face=3DArial size=3D2>6th IFIP/IEEE International =
Conference on=20
Management of Multimedia Networks and Services=20
:<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 7th - 10th September =
2003,=20
Queen's University of Belfast, Northern Ireland=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;=20
<A =
href=3D"http://www.ee.qub.ac.uk/mmns2003">http://www.ee.qub.ac.uk/mmns200=
3</A>=20
</FONT>.</FONT></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><FONT face=3D"Times New Roman"=20
size=3D3>----------------------------------------------------------------=
---</FONT><BR></DIV></FONT>
<DIV><FONT face=3DArial size=3D2>Please note the extension to the =
deadline for=20
submission of papers to MMNS2003&nbsp;</FONT><FONT face=3DArial =
size=3D2>is 18th=20
April 2003. Please circulate to your colleagues to ensure a good=20
turnout.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>For further information about the =
conference please=20
see the web site at <A=20
href=3D"http://ee-server.ee.qub.ac.uk/dsp/mmns2003/">http://ee-server.ee.=
qub.ac.uk/dsp/mmns2003/</A></FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Best Regards,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Ahmed Mehaoua</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>(Publicity&nbsp;chair)</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial=20
size=3D2>&nbsp;----------------------------------------------------------=
-----------------------------------------------</FONT></DIV>
<DIV><FONT face=3DArial size=3D2><STRONG></STRONG></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><STRONG>MMNS 2003<BR></STRONG>6th =
IFIP/IEEE=20
International Conference on Management of Multimedia Networks and=20
Services<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 7th - 10th =
September=20
2003, Queen's University of Belfast, Northern Ireland=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;=20
<A =
href=3D"http://www.ee.qub.ac.uk/mmns2003">http://www.ee.qub.ac.uk/mmns200=
3</A>=20
</FONT></DIV>
<DIV>&nbsp;</DIV><FONT face=3DArial size=3D2>
<DIV><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<STRONG>Second Call For Papers</STRONG></DIV>
<DIV>&nbsp;</DIV>
<DIV>Multimedia services over IP networks are proliferating at an =
enormous=20
speed. There <BR>is also increasing demand for solutions which provide =
assured=20
levels of service quality. <BR>All of these require novel paradigms, =
models and=20
architectures for realising integrated <BR>end-to-end service management =
rather=20
than managing network elements in isolation. <BR>Providing scalable =
Quality of=20
Service (QoS) while maintaining fairness, along with <BR>secure and =
optimal=20
network resource management are key challenges for the future =
<BR>Internet.=20
These challenges apply to both fixed and wireless networks. </DIV>
<DIV>&nbsp;</DIV>
<DIV>The IFIP/IEEE International Conference on Management of Multimedia =
Networks=20
and Services <BR>will hold its sixth annual meeting from September 7th =
to=20
September 10th, 2003 in Belfast,<BR>Northern Ireland. MMNS provides an =
intimate=20
setting for discussion and debate. In just 6 years, <BR>MMNS has =
established=20
itself as one of the premier conferences with a focus on the management =
<BR>of=20
multimedia networks and services. The conference objective is to bring =
together=20
researchers <BR>working in all facets of network and service management =
as=20
applied to broadband networks and <BR>multimedia services. MMNS deals =
with all=20
aspects of designing, developing and deploying <BR>networked multimedia =
systems=20
and it serves as a forum for the dissemination of state-of-the-art =
<BR>research=20
and development results. </DIV>
<DIV>&nbsp;</DIV>
<DIV>MMNS 2003 will also include panel sessions in which experts =
<BR>offer their=20
observations and opinions about current hot topics. The keynote speaker =
will be=20
<BR>Professor Ian Akyildiz, Georgia Institute of Technology who will =
present a=20
vision of future <BR>interplanetary network architectures. Professor =
Derek=20
McAuley, head of Intel's recently formed <BR>laboratory at Cambridge, =
UK, will=20
describe some of the new research being undertaken on global <BR>overlay =

networks and applications.</DIV>
<DIV>&nbsp;</DIV>
<DIV><BR>The program committee is soliciting original papers describing =
research=20
in the area of <BR>management of multimedia networks and services. =
Topics of=20
interest include, but are not <BR>limited to, the following:</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Active multimedia =
network=20
management <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Ad-hoc and =
Sensor=20
Networks <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Augmented and =
Virtual=20
Reality Networks <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * =
Billing and=20
Accounting <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Cable =
multimedia=20
network management <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * =
Content=20
distribution internetworking =
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; *=20
Deployment of multimedia services =
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
* Distributed multimedia service management=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * End-to-end IP =
multimedia=20
network and service management =
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; *=20
IP Video, streaming, interactive video service management=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Middleware support for=20
management <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Multimedia =
network=20
traffic engineering and optimization=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Multimedia traffic =
management=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Multimedia content =
protection=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Multimedia session =
management=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Multi-point, multicast =
services=20
management <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Network =
management=20
models and architectures <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
*=20
Network programmability for multimedia services=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Optical multimedia =
network=20
management <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Policy-based =

management for multi-media services=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Provisioning of =
multimedia=20
networks and services <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * =
QoS in=20
WLANs <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * QoS management=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Resource, performance =
and fault=20
management <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Security and =

Authentication <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * VoIP =
service=20
management <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Web Services =

<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Wireless and mobile =
multimedia=20
network management </DIV>
<DIV>&nbsp;</DIV>
<DIV><BR>Papers must be submitted electronically in postscript or PDF =
format.=20
Detailed <BR>instructions are provided on the conference web site, <A=20
href=3D"http://www.ee.qub.ac.uk/mmns2003">http://www.ee.qub.ac.uk/mmns200=
3</A>.=20
</DIV>
<DIV>&nbsp;</DIV>
<DIV><STRONG>Submission=20
date:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
18th=20
April 2003<BR>Notification of acceptance: 6th June 2003<BR>Final=20
version:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;=20
4th July 2003</STRONG></DIV>
<DIV><STRONG></STRONG>&nbsp;</DIV>
<DIV>Conference Chairs:<BR>&nbsp;&nbsp; Professor Alan Marshall, <A=20
href=3D"mailto:a.marshall@ee.qub.ac.uk">a.marshall@ee.qub.ac.uk</A>, =
Queen=92s=20
University of Belfast, UK<BR>&nbsp;&nbsp; Professor Nazim Agoulmine, <A=20
href=3D"mailto:nazim@rp.lip6.fr">nazim@rp.lip6.fr</A>, University of =
Evry,=20
France<BR></FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV></FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Qiang Gu<BR>Advanced Telecommunication =
Systems=20
Laboratory<BR>School of Electrical &amp; Electronic Engineering<BR>The =
Queen's=20
University of Belfast<BR>Belfast<BR>Northern Ireland.<BR>BT9 =
5AH<BR>Email: <A=20
href=3D"mailto:qiang.gu@ee.qub.ac.uk">qiang.gu@ee.qub.ac.uk</A><BR>Teleph=
one: +44=20
-2890-274142<BR>Fax: +44 -2890-274417</FONT></DIV></FONT></DIV>
<DIV><FONT face=3DArial=20
size=3D2>---------------------------------------------------------<BR>Ahm=
ed=20
Mehaoua<BR>University of Versailles - CNRS PRISM Lab.<BR>45 av. des =
etats unis=20
78000 Versailles - France<BR>Email : <A=20
href=3D"mailto:mea@prism.uvsq.fr">mea@prism.uvsq.fr</A><BR>Tel : +33 1 =
39 25 40=20
45</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV></BODY></HTML>

------=_NextPart_000_06CD_01C2FE02.914ABFE0--

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With our apologize for any duplicates;


         **********************************************************
                                       INTERNETWORKING 2003 International 
Conference
                                   Hilton San Jose & Towers Hotel, California
                              Sunday June 22 through Tuesday June 24, 2003
                                 with the support of the IEEE and IFIP WG 6.2
         **********************************************************


Internetworking-2003 is a conference focused on internetworking
technologies in the IP, optical, and wireless domains.
Last year's event was collocated with the highly successful MPLS 2002
conference, which was held in Washington DC. Internetworking 2003 is more
expansive than the MPLS conference series and encompasses a wider range of
topics.
The Internetworking 2003 event in March will include participation from
industry, government agencies, and the academia.
Presentations are through invitation and via open call.Technical sessions
at the conference include the following tracks:

-       Routing
-       OSS and Network Management
-       Virtual Private Networks (VPNs)
-       IP Network Security
-       Voice over IP
-       IPv6

Call For Papers:
----------------

The Internetworking 2003 Program Committee cordially invites you to submit
proposals for original, unpublished presentations focusing on
internetworking technologies in the IP, optical, and wireless domains.
Summaries not exceeding 250 words can be submitted to submissions@caitr.org
for review and possible inclusion in the program, no later than April 11,
2003. Topics of interest include, but are not limited to the following:

* Voice over IP (VoIP)
* IP Video Conferencing
* Storage Area Networks (SANs)
* Unicast and Multicast Routing and Convergence
* QoS Routing
* Network Security and Service Integration
* Operational Support Systems
* Virtual Private Networks
* Internetworking Wireless LANs and 3G Wireless Networks
* IP-based Infrastructure for Wireless Networks
* Internetworking IP and Optical Networks
* Internetworking MPLS with Legacy ATM and Frame Relay Networks
* Transition from IPv4 to IPv6 and interworking
* Pervasive Computing
* High Speed Transport Layer Protocols
* Peer to Peer Networking and Grid Computing
* Video Teleconferencing (VTC)
*  802.11 Hotspots
* Ethernet First Mile networks (EPON, EDSL)
* IEEE 802.17 MAN, RPR

Conference Web_site: http://www.caitr.org/internetworking03/program.htm
--------------------

Technical Programme Committee :
--------------------------------------------

- Maurice Gagnaire (ENST- France) co-chair
- Daniel Awduche (Isocore - USA) co-chair


- Augusto Casaca (INESC- Portugal)
- Nada Golmie (NIST - USA)
- Ibrahim Habib (CUNY - USA)
- Bijan Jabbari (GMU - USA)
- Marc Lobelle (UCL - Belgium)
- Fabio Neri (Politecnico di Torino - Italy) *
- Guy Pujolle (LIP6 - France)
- James Roberts (France Telecom R&D) *
- Roberto Sabella (Erisson - Italy)
- Jochen H. Schiller (Freie Univ. Berlin, Germany)
- Vishal Sharma (Metanoia - USA)
- Samir Tohme (ENST - France)
- Stefano Trumpy (CNR - Italy)
- Peter Vetter (Alcatel Antwerpen - Belgium) *
- Parviv Yegani (Cisco - USA)
- Ferit Yegenoglu (Isocore -USA)
- Ali Zadeh (GMU - USA)
- Moshe Zukerman (Univ. of Melbourne - Australia)

(* = to be confirmed)


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To: "Fredrik Lindholm (EAB)" <Fredrik.Lindholm@era.ericsson.se>
cc: mmusic@ietf.org
Subject: Re: [MMUSIC] Comments on draft-ietf-mmusic-kmgmt-ext-07.txt 
In-Reply-To: Your message of "Wed, 09 Apr 2003 18:39:30 +0200."
             <1F55F6582266314A85A55F6241509B670575B332@Esealnt863.al.sw.ericsson.se> 
Date: Sat, 12 Apr 2003 16:16:46 -0400
From: Colin Perkins <csp@csperkins.org>
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--> "Fredrik Lindholm (EAB)" writes:
>
>Hi Colin,
>
>see comments inline.
>
>> -----Original Message-----
>> From: Colin Perkins [mailto:csp@csperkins.org]
>> Sent: den 9 april 2003 06:45
>> To: mmusic@ietf.org
>> Subject: [MMUSIC] Comments on draft-ietf-mmusic-kmgmt-ext-07.txt
>> 
>> 
>> I've just completed a detailed read-through the the key management
>> extensions draft, and have a number of comments:
>> 
>>  - In section 3.1, first paragraph after bullet points on page 6, what
>>    happens if the key management protocol accepts the data in some
>>    attributes, and rejects the others? For example if a media level 
>>    key management attribute is valid but the session level isn't? 
>> 
>>  - In the second paragraph on page 8, does "The offer MUST 
>> then abort the
>>    security setup" mean that the entire session is aborted? 
>> Or does it go
>>    ahead without security?
>
>To answer both these questions: session and media level failures 
>should be treated the same way, i.e., the entire session is 
>aborted (then it is up to the local policy to decide what to do). 

Okay. The draft needs clarification though.

>>  - I don't understand the discussion of resynchronization in the last
>>    paragraph of section 3.2. Can you clarify what is intended?
>>
>
>Basically, if you need to change keys during an ongoing session, 
>we propose to send a new INVITE message (with new crypto parameters
>inside).

Again, okay but the text needs to be clarified.

>>  - Given the risks identified in the third paragraph of 
>> section 5, it's not
>>    clear to me why the entire SDP message doesn't need to be 
>> authenticated?
>
>I have never understood why people don't apply security to things :-) 
>However, when it comes to SDP, the protection of the SDP will 
>very much depend on the protocol carrying SDP and what security 
>features are implemented (and defined as necessary). So it is out 
>of the scope of this document to specify how to protect the SDP, 
>and that is also why we only give a recommendation that you should 
>apply some form of protection to the SDP.

But if the SDP has to be transferred by a secure channel, what is the point
of MIKEY?  

>>    Are attacks possible by manipulating the non-authenticated 
>> parts of the
>>    SDP?
>
>If you, by attacks, are referring to attacks towards the key management 
>and the secured media, I would say no.

But is may be possible to redirect a media stream, for example?
Colin
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From: Colin Perkins <csp@csperkins.org>
Subject: [MMUSIC] Draft minutes from San Francisco
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Enclosed are draft minutes from MMUSIC in San Francisco. These need to be
submitted by tomorrow (Monday 14th), so please send any comments quickly. 
Copies of the minutes and presentation materials are available at
http://www.dmn.tzi.org/ietf/mmusic/

Colin

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

Multipary Multimedia Session Control (MMUSIC) Working Group Minutes

Reported by Colin Perkins

    The MMUSIC working group met once at the 56th IETF meeting in San
    Francisco. The group discussed connection oriented media, various
    SDP extensions, Offer/Answer examples, SDPng, Internet Media Guides,
    and RTSP.

Introduction and Status Update

    Joerg Ott began the meeting with an introduction and status update.
    One RFC has been published since the last meeting (RFC 3388 on Flow
    Identification), and one draft is with the RFC Editor (draft-ietf-
    mmusic-reservations-flows-01.txt). The revision to SDP (draft-ietf-
    mmusic-sdp-new-12.txt) is under AD review, as are drafts in the RTCP
    attribute in SDP (draft-ietf-mmusic-sdp4nat-03.txt) and the SDPng
    Transition Strategies (draft-ietf-mmusic-sdpng-trans-03.txt). The
    SDP bandwidth parameters (draft-ietf-mmusic-sdp-bwparam-01.txt) are
    in working group last call, and comments are solicited.

    Joerg noted that the SDP specification has been revised due to Area
    Director feedback. Colin Perkins described the changes: a number of
    minor fixes to the ABNF, and significant updates to the description
    of the "k=" field and the IANA Considerations section. The registration
    requirements for attributes and other parameters have been significantly
    tightened with this revision, and comments are solicited regarding the
    degree to which these changes are appropriate.

    The SDP source filter draft (draft-ietf-mmusic-srcfilter-03.txt) is
    essentially complete, just needing editorial work. The chairs noted
    that they expect to issue working group last call on this in early
    April, and that anyone with comments should post them shortly.

    There is a new draft on SDP implementation status and interoperability
    (draft-ietf-mmusic-sdp-implem-00.txt), which is a first cut at the
    documentation needed to advance SDP to draft standard. Please read and
    comment.

    The working group has completed its charter, and a draft for a revised
    charter was sent to the mailing list prior to the meeting. This draft
    is under discussion between the chairs and area director, and comments
    are solicited from the group. It is expected that the revision will
    include work on SDP, SDPng, RTSP and Internet Media Guides. Conference
    Control has been ruled out of scope at this time.

Connection-Oriented Media Transport in SDP

    David Yon discussed comedia (draft-ietf-mmusic-sdp-comedia-05.txt). The
    changes in -05 include removal of source address/port per discussion on
    the mailing list, and removal of the current security considerations in
    preparation for a revision in the next version (again, as discussed on
    the mailing list).

    David noted that the main security issue is session hijacking, which
    can be solved by using secure media streams. In the absence of this,
    the attack can be downgraded to a denial of service, if the endpoints
    follow the rules in Section 4(b) and (c) of the draft, using duplicate
    incoming connections as a signal of the attack (this works for TCP and
    bidirectional RTP). Jonathan Rosenberg noted that using double connect
    as an indication of an attack might fail in the presence of forking
    invites; this needs further consideration.

    Next steps are to come to closure on the RTSP issues discussed on the
    mailing list, and to fold in the expanded security considerations into
    -06. Colin Perkins confirmed that the "eid" would be implemented as a
    separate draft if needed. David expressed the hope that the -06
    version would be ready for working group last call.

SDP key management extensions

    Elisabetta Carrara discussed changes to the key management extensions
    work (draft-ietf-mmusic-kmgmt-ext-07.txt).  The changes are primarily
    for clarification, and include a more detailed description of the
    interactions between SDP and the key management module, and rekeying.
    In addition, the IANA considerations section has been updated.

    The main issue previously noted was a bidding down attack, which
    Elisabetta described, but this has been solved in -07 by authenticating
    the list of proposed key management protocols in the SDP.

    Mark Baugher noted that another way to avoid the attack is to use only a
    single key management protocol, to avoid the possibility of bidding. Ran
    Canetti described how the attack is still valid in that case, since the
    bid down can be done over multiple INVITEs with different protocols
    meaning that the addition of authentication is the correct solution
    (even though it complicates the protocol).

    It was asked if Kerberos still works with the changes? Elisabetta said
    it should, but she would check to make sure.

    Colin Perkins noted that approval is needed from the transport area
    security advisor before this draft can advance, but that the changes
    appear to address the previously open issues.

SDP Security Descriptions for Media Streams

    Mark Baugher discussed SDP security descriptions for media streams
    (draft-ietf-mmusic-sdescriptions-00.txt). The aim of this draft is to
    replace k= with an attribute, suitable for signalling SRTP parameters.
    The draft is intended for applications where the signalling channel is
    secure. If the signalling is hop-by-hop, as is common with telephony,
    the intermediate hops must be trusted to maintain security.

    Changes from (draft-baugher-mmusic-sdpmediasec-00.txt) apply parameters
    to media streams, not codecs (see discussion at the previous meeting);
    apply parameters to the SDP media level only (as a result of improving
    offer/answer); add an "application" parameter allowing separate
    descriptions for SRTP and SRTCP; define use with offer/answer and
    augment the ABNF.

    Next steps are to fix known problems, in particular the ambiguities with
    SDP direction attributes, to generalize offer/answer, and to generalize
    to transports beyond SRTP, and to get implementation experience.

    Magnus Westerlund noted that RTSP doesn't currently define a secure
    signalling channel. This will need to be fixed, if this work is to be
    used with RTSP.

    Steve Casner asked if the semantics were well defined if more than one
    of "k=", "a=keymgmt", and "a=crypto" are present? This is not legal, but
    appropriate behaviour needs to be specified. It was asked how a system
    should behave if the key is referenced by a non-secure URI? This is
    clearly not appropriate use.

SDP attribute for qualifying media formats with generic parameters

    Flemming Andreasen discussed (draft-rajeshkumar-mmusic-gpmd-02.txt).
    This is intended to provide an alternative to "a=fmtp:", so that new
    parameters can be added to existing media formats. The example being
    a voice-band data parameter, to apply to codecs such as PCMU when
    used for modem or fax relay.

    Changes since -01 are to firm up the definition of the gpmd attribute,
    clarifying that it is intended for informative hints, and that lack of
    a gpmd parameters MUST NOT render the media format useless. Use with
    offer/answer has also been specified, as have registration procedures.

    Ruediger Kreuter asked if an echo cancellor parameter has been considered?
    Flemming replied that there are many ways of specifying this - see V.150 -
    how should it be specified? Colin Perkins noted that this should be a new
    extension draft, rather than being including in the gpmd draft, to avoid
    delaying the work.

    The chairs asked for comments, noting that they intend to issue a
    working group last call on this draft shortly.

SDP Offer/Answer Examples

    Alan Johnston outlined draft-johnston-mmusic-offer-answer-examples-01.txt
    which provides examples of the use of the offer/answer model. There have
    been a number of minor corrections and clarifications since the -00 draft,
    and the plan is to continue collecting new scenarios and seek review and
    comment from implementors. If you are implementing SIP and offer/answer,
    you are urged to review the draft and send feedback.

SDPng Update

    Dirk Kutscher presented an update on the SDPng work (draft-ietf-mmusic-
    sdpng-06.txt). Changes in -06 include a new document structure, new
    requirements for capability specifications, new capability negotiation
    rules, the removal of XML-Schema-based formal definitions and text on
    session-level information.

    Capability specifications are now required to support feature independent
    negotiation and must not require access to a package definition in order
    to process capability descriptions. This requires definition of a fixed
    set of basic types, encoding the type into values. In addition, names
    must be fully qualified.

    Capability negotiation rules have been updated to describe the general
    behaviour, leaving concrete behaviour implementation specific. Two
    alternative procedures are specified: an offer/answer based scheme
    and an RFC 2533 (conneg) style approach.

    Open issues include a concrete syntax definition, session level data,
    and possible use of libraries of definitions. Next steps are to give
    details on describing actual configurations and transport parameters,
    and a formal definition mechanisms for packasges. Other work items
    include specific package definitions (e.g. audio, video) and session
    level metadata.

Internet Media Guides

    Rod Walsh outlined Internet Media Guides (IMGs), a potential new work
    item for the group. An Internet Media Guide (IMG) is a collection of
    multimedia session descriptions expressed using SDP, SDPng or some
    similar session description format. It is used to describe a collection
    of multimedia sessions (e.g.  television programme schedules).  The IMG
    must be delivered to a potentially large audience, who use it to join a
    subset of the sessions described, and who may need to be notified of
    changes to the IMG.

    This problem lends itself to solution by (extensions of) some existing
    mechanisms: SAP and SDP/SDPng. MMUSIC therefore seems the natural home
    for the work. Other groups (e.g. DVB-IPI, TV Anytime, and the IP datacast
    forum) have identified similar needs, but do not have the experience to
    solve the problem.

    The chairs presented some initial charter text discussing the problem
    (subject to modification/approval from the area directors) and asked
    if the working group considered this an appropriate work item? Strong
    support was received.

RTSP

    Magnus Westerlund discussed changes to the Real Time Streaming Protocol
    (draft-ietf-mmusic-rfc2326bis-03.txt). The goal is to update and clarify
    the RTSP specification, with the aim of moving to draft standard RFC (a
    new proposed standard RFC may be required first, depending on the scale
    of the changes). The aim is to have the core specification completed by
    October 2003. The new version will be a minimal core specification, with
    extensions documents for NAT traversal, MUTE and UNMUTE and (possibly)
    a MIB, RECORD, secure RTSP, and RTSP over unreliable transport.

    Since the previous meeting, there have been 5 teleconferences discussing
    the spec (details announce on the mailing list or see http://rtsp.org/).
    A list of known bugs in the specification is also maintained at
    http://rtspspec.sourceforge.net/.

    Changes updates discussed recently include:
    	- Non-persistent connection support is now mandatory
	- IPv6 support added
	- Accept-Ranges as proposed in the core spec
	- Byte ranges will be allowed
	- Via header will not be changed to include client address
	- Clarification of 304 "Not modified"
	- Clarification of redirection
	- Agreed feature extension mechanism
	- Strengthened requirements for Destination header
	- Added source and destination addresses to the Transport header
	- RTP-Info's base URL is a PLAY request URL
	- Clarified interleaved usage
	- A SETUP to change transport parameters leaves the previous state
	  intact if it fails
	- Clarification that RTSP URLs are case sensitive
	- Range header now required in all PLAY responses
	- Range is formulated as start point (inclusive) to stop point
	  (non-inclusive)
	- Use of PAUSE when in Ready clarified

    Open issues include how RTSP extensions registrations are to be handled,
    and what are the requirements for new extensions; can timed requests be
    removed; handling of multiple SSRCs in RTSP; and inclusion of a warning
    header.

    Input from those working with RTSP was sought, since there is a shortage
    of manpower on this effort.

RTSP and NAT

    Magnus Westerlund discussed NAT traversal for RTSP (draft-ietf-mmusic-
    rtsp-nat-00.txt). The issue is not RTSP signalling itself, but how the
    media streams traverse the NAT, and how RTSP knows the correct external
    addresses to use after NAT traversal. Will also address cooperation with
    firewalls.

    The draft currently discusses use of STUN, symmetric RTSP, tunnelled
    media in TCP and application level gateways. Open issues exist with
    the use of STUN for symmetric NATs and symmetric RTP. Mark Baugher asked
    if the authors have considered potential issues with secure RTSP? May
    not be as simple as "just use SSL"? Philippe Gentric noted that the
    attacker is often the client, so a secure channel may not help security.

    Magnus asked for input, and is seeking a co-author to help complete the
    work.

Stream Switching Control

    Philippe Gentric discussed draft-gentric-mmusic-stream-switching-00.txt,
    a new draft describing how a server can signal that it is switching from
    one version of a stream to another (for example, for coarse-grained rate
    adaptation, when the limits of adaptation of a particular codec are
    exceeded).

    Alan Duric was glad to see the work, but asked if Philippe considered
    the problem that routers may not be saturated due to bandwidth, but due
    to packet rate? Maybe we should add a feature to change packetization
    interval? Joerg Ott agreed that this is something to consider, it's just
    another change in the media format.

    Magnus Westerlund asked if you can do other things than switch streams
    for congestion control? Philippe said yes, streams can be adapted in
    various ways, but this doesn't need protocol supoprt from RTSP.

    Ross Finlayson agreed that the concept was useful, but was unsure that
    RTSP is the place to put the signalling? Will the TCP nature of RTSP
    delay the congestion control signalling? Maybe use RTCP? Colin Perkins
    noted that it is important to differentiate between indication of loss
    for a stream (RTCP) and signalling to switch to a different stream (RTSP).

    Anders Klemets also liked the idea, but not the details. We have standard
    ways of specifying alternates in SDP, maybe want to leave those details
    out of this spec?

    Given the interest, it was agreed to move forward with this discussion
    on the mailing list.

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From: "Fredrik Lindholm (EAB)" <Fredrik.Lindholm@era.ericsson.se>
To: "'Colin Perkins'" <csp@csperkins.org>
Cc: mmusic@ietf.org
Subject: RE: [MMUSIC] Comments on draft-ietf-mmusic-kmgmt-ext-07.txt 
Date: Mon, 14 Apr 2003 08:36:32 +0200
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<snip...>
> 
> 
> >>  - Given the risks identified in the third paragraph of 
> >> section 5, it's not
> >>    clear to me why the entire SDP message doesn't need to be 
> >> authenticated?
> >
> >I have never understood why people don't apply security to 
> things :-) 
> >However, when it comes to SDP, the protection of the SDP will 
> >very much depend on the protocol carrying SDP and what security 
> >features are implemented (and defined as necessary). So it is out 
> >of the scope of this document to specify how to protect the SDP, 
> >and that is also why we only give a recommendation that you should 
> >apply some form of protection to the SDP.
> 
> But if the SDP has to be transferred by a secure channel, 
> what is the point
> of MIKEY?  

The SIP/SDP may only be protected hop-by-hop (i.e. between the 
SIP proxies and SIP proxy - terminal). MIKEY is then needed to provide 
end-to-end security for the media streams. More or less, you only 
need to trust the proxies to provide a reliable transport, but 
you do not need to trust the proxies for your media.

An almost analogous example would be email. You may use SSL to your 
mail server (to secure your session and account), but if you like 
end-to-end protection, you do not rely on this but apply something 
like S/MIME... 

> 
> >>    Are attacks possible by manipulating the non-authenticated 
> >> parts of the
> >>    SDP?
> >
> >If you, by attacks, are referring to attacks towards the key 
> management 
> >and the secured media, I would say no.
> 
> But is may be possible to redirect a media stream, for example?

Of course (note though that there are plenty of other methods to 
do re-direction attacks of streams if you happen to be "on-the-path"...).  

Fredrik

> Colin
> 
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Subject: [MMUSIC] WG Last Call on draft-rajeshkumar-mmusic-gpmd-02.txt
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This is to issue WGLC on draft-rajeshkumar-mmusic-gpmd-02.txt

   "SDP attribute for Qualifying Media Formats with Generic Parameters"

for Proposed Standard.  The WGLC is to expire on April, 30th, 2003.

Please direct any technical comments to the list and the authors,
and please send purely editorial remarks only to the authors.

Thanks,
Joerg

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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Fredrik Lindholm (EAB)'" <Fredrik.Lindholm@era.ericsson.se>,
        "'Colin Perkins'" <csp@csperkins.org>
Cc: mmusic@ietf.org
Subject: RE: [MMUSIC] Comments on draft-ietf-mmusic-kmgmt-ext-07.txt 
Date: Tue, 15 Apr 2003 04:45:59 -0400
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A few notes below.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Fredrik Lindholm (EAB) [mailto:Fredrik.Lindholm@era.ericsson.se]
> Sent: Sunday, April 13, 2003 11:37 PM
> To: 'Colin Perkins'
> Cc: mmusic@ietf.org
> Subject: RE: [MMUSIC] Comments on draft-ietf-mmusic-kmgmt-ext-07.txt 
> 
[snip]
> 
> The SIP/SDP may only be protected hop-by-hop (i.e. between the 
> SIP proxies and SIP proxy - terminal). MIKEY is then needed to provide 
> end-to-end security for the media streams. More or less, you only 
> need to trust the proxies to provide a reliable transport, but 
> you do not need to trust the proxies for your media.
> 
> An almost analogous example would be email. You may use SSL to your 
> mail server (to secure your session and account), but if you like 
> end-to-end protection, you do not rely on this but apply something 
> like S/MIME... 
> 

It is worth pointing out that SIP encourages support of S/MIME for just this
reason.

> > 
> > >>    Are attacks possible by manipulating the non-authenticated parts
of the
> > >>    SDP?
> > >
> > >If you, by attacks, are referring to attacks towards the key
management 
> > >and the secured media, I would say no.
> > 
> > But is may be possible to redirect a media stream, for example?
> 
> Of course (note though that there are plenty of other methods to 
> do re-direction attacks of streams if you happen to be 
> "on-the-path"...).  
> 

Again, it is worth noting that wrapping SDP in S/MIME prevents this sort of
attack (at least redirection at the layer of SDP - SIP, for example, has
some other safeguards to prevent malicious redirection at the SIP layer). 

This of course speaks only to carrying SDP in SIP. But is it true that all
of the protocols that currently use SDP (RTSP, SAP, SIP...) carry MIME
types, and consequently could use S/MIME if needed?

> Fredrik
> 
> > Colin
> > 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 
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To: "Peterson, Jon" <jon.peterson@neustar.biz>
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        mmusic@ietf.org
Subject: Re: [MMUSIC] Comments on draft-ietf-mmusic-kmgmt-ext-07.txt 
In-Reply-To: Your message of "Tue, 15 Apr 2003 04:45:59 EDT."
             <0449D80A0E9B614A83FA9031B07E8D3B2579B9@stntexch2.va.neustar.com> 
Date: Tue, 15 Apr 2003 07:45:57 -0500
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--> "Peterson, Jon" writes:
>
>A few notes below.
>
>Jon Peterson
>NeuStar, Inc.
>
>> -----Original Message-----
>> From: Fredrik Lindholm (EAB) [mailto:Fredrik.Lindholm@era.ericsson.se]
>> Sent: Sunday, April 13, 2003 11:37 PM
>> To: 'Colin Perkins'
>> Cc: mmusic@ietf.org
>> Subject: RE: [MMUSIC] Comments on draft-ietf-mmusic-kmgmt-ext-07.txt 
>> 
>[snip]
>> 
>> The SIP/SDP may only be protected hop-by-hop (i.e. between the 
>> SIP proxies and SIP proxy - terminal). MIKEY is then needed to provide 
>> end-to-end security for the media streams. More or less, you only 
>> need to trust the proxies to provide a reliable transport, but 
>> you do not need to trust the proxies for your media.
>> 
>> An almost analogous example would be email. You may use SSL to your 
>> mail server (to secure your session and account), but if you like 
>> end-to-end protection, you do not rely on this but apply something 
>> like S/MIME... 
>> 
>
>It is worth pointing out that SIP encourages support of S/MIME for just this
>reason.

Although this requires you to trust any intermediate proxies. This need
could perhaps be avoided if MIKEY authenticated the entire SDP end2end?

>> > >>    Are attacks possible by manipulating the non-authenticated parts
>of the
>> > >>    SDP?
>> > >
>> > >If you, by attacks, are referring to attacks towards the key
>management 
>> > >and the secured media, I would say no.
>> > 
>> > But is may be possible to redirect a media stream, for example?
>> 
>> Of course (note though that there are plenty of other methods to 
>> do re-direction attacks of streams if you happen to be 
>> "on-the-path"...).  
>> 
>
>Again, it is worth noting that wrapping SDP in S/MIME prevents this sort of
>attack (at least redirection at the layer of SDP - SIP, for example, has
>some other safeguards to prevent malicious redirection at the SIP layer). 
>
>This of course speaks only to carrying SDP in SIP. But is it true that all
>of the protocols that currently use SDP (RTSP, SAP, SIP...) carry MIME
>types, and consequently could use S/MIME if needed?

RTSP certainly could use S/MIME; SAP probably can't as it's unidirectional
and one-to-many.

Colin
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Date: Tue, 15 Apr 2003 16:27:53 +0200
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Subject: [MMUSIC] RTSP meeting agenda 16 april
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Hi,

There is a RTSP telecon tomorrow wednesday, the 16th April at 18.00 CET 
(GMT +2). I have sent out the meeting details to the RTSP logistics 
lists. IF you want to participate and haven't received the details send 
me a mail.

Best Regards

Magnus

-----------------
Agenda 030416

1. Opening of meeting.
2. Since last meeting (5th of March).
    a. Bug513737 - Who will write it up?
    b. Bug 533691 &  513880 - Aravind to write up, progress?
    c. http://rtsp.org/bug556061 - Fix state machine to allow for 
REDIRECT at any time, including in INIT state.
    d. http://rtsp.org/bug556295 - Allow: needs explanation.
    e. Remove timed PLAY and PAUSE. With time I mean the specified usage 
of the time parameter in the Range header.
    f. Stream switching requirements document (Phillipe Gentric)
    g. Offer Answer for RTSP. Anybody willing to write requirements and 
use case?
    h. IANA registration comments?
    i. Ways to allow direct media destination to other than own IP 
address. (Phillipe Gentric)
    j. Multiple SSRC in RTSP. Redefine RTP-Info to solve issue?
    k. How to distinguish the updated specification.
    l. Warning header: For further discussion.
    m. RTSP servers behind NATs. Use case seems to exist. Requires 
someone to write scenario document. Decide on how to go forward compared 
with public network servers. Time line considerations.
    n. RTSP ALG for Linux netfilter. (Tom Marshall)

3. How to get things into the spec?

4. New Things
    a. Specification review. Plan to look through the latest draft and 
see what comments people has about it.
    b. http://rtsp.org/bug631148 <http://rtsp.org/bug556061> better Info 
on RTSP proxies.
    c. http://rtsp.org/bug631957 Use of # and ? in RTSP URLs?
    d. http://rtsp.org/bug633148 Use of TCP and UDP.
    e. http://rtsp.org/bug633779 SET_PARAMETER responses.
5. Any other Issues
6. Next meeting
7. End meeting

-- 

Magnus Westerlund 

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


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Colin Perkins wrote:

>>>An almost analogous example would be email. You may use SSL to your 
>>>mail server (to secure your session and account), but if you like 
>>>end-to-end protection, you do not rely on this but apply something 
>>>like S/MIME... 
>>>
>>
>>It is worth pointing out that SIP encourages support of S/MIME for just this
>>reason.
> 
> 
> Although this requires you to trust any intermediate proxies. This need
> could perhaps be avoided if MIKEY authenticated the entire SDP end2end?

I'm confused.

S/MIME does not require you to trust intermediate proxies, thats the 
whole point.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Scientist                             Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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Subject: Re: [MMUSIC] Comments on draft-ietf-mmusic-kmgmt-ext-07.txt 
In-Reply-To: Your message of "Tue, 15 Apr 2003 11:02:11 EDT."
             <3E9C1EF3.6030507@dynamicsoft.com> 
Date: Tue, 15 Apr 2003 10:49:05 -0500
From: Colin Perkins <csp@csperkins.org>
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--> Jonathan Rosenberg writes:
>
>
>Colin Perkins wrote:
>
>>>>An almost analogous example would be email. You may use SSL to your 
>>>>mail server (to secure your session and account), but if you like 
>>>>end-to-end protection, you do not rely on this but apply something 
>>>>like S/MIME... 
>>>>
>>>
>>>It is worth pointing out that SIP encourages support of S/MIME for just this
>>>reason.
>> 
>> 
>> Although this requires you to trust any intermediate proxies. This need
>> could perhaps be avoided if MIKEY authenticated the entire SDP end2end?
>
>I'm confused.
>
>S/MIME does not require you to trust intermediate proxies, thats the 
>whole point.

Duh, my bad. Confusing S/MIME and SSL. Ignore that message, not enough
coffee this morning :)

Colin
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Date: Tue, 15 Apr 2003 11:26:07 -0700
To: "Peterson, Jon" <jon.peterson@neustar.biz>
From: Mark Baugher <mbaugher@cisco.com>
Subject: RE: [MMUSIC] Comments on draft-ietf-mmusic-kmgmt-ext-07.txt 
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Jon,

At 04:45 AM 4/15/2003 -0400, Peterson, Jon wrote:
<...>

>This of course speaks only to carrying SDP in SIP. But is it true that all
>of the protocols that currently use SDP (RTSP, SAP, SIP...) carry MIME
>types, and consequently could use S/MIME if needed?

SAP is to a group so we won't get confidentiality but would get authentication.

Mark


> > Fredrik
> >
> > > Colin
> > >
> > _______________________________________________
> > 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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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Mark Baugher'" <mbaugher@cisco.com>
Cc: mmusic@ietf.org
Subject: RE: [MMUSIC] Comments on draft-ietf-mmusic-kmgmt-ext-07.txt 
Date: Tue, 15 Apr 2003 15:26:41 -0400
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[snip]
> 
> SAP is to a group so we won't get confidentiality but would 
> get authentication.
> 

I imagine that for redirection attacks targeting m= lines, and similar
mischief, the properties we really want are authentication and integrity,
which do seem achievable for SAP.

Jon Peterson
NeuStar, Inc.


> Mark
> 
> 
> > > Fredrik
> > >
> > > > Colin
> > > >
> > > _______________________________________________
> > > mmusic mailing list
> > > mmusic@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/mmusic
> > >
> >_______________________________________________
> >mmusic mailing list
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> 
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Date: Tue, 15 Apr 2003 13:26:49 -0700
To: mmusic@ietf.org
From: Ross Finlayson <finlayson@live.com>
Cc: bquinn@celoxnetworks.com
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Subject: [MMUSIC] Updated SDP "source filter" Internet Draft
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FYI, I have just submitted an updated copy of the SDP "source filter" 
draft.  You can also find a copy online at
	http://www.live.com/draft-mmusic-sdp-srcfilter-04.txt

This version contains several formatting and 'I-D nit' improvements (thanks 
to feedback from Magnus Westerlund and Colin Perkins), and clarifies how 
source filters are applied to "c=" lines that specify multiple multicast 
addresses.

	Ross.

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Cc: bquinn@celoxnetworks.com
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FYI, I have just submitted an updated copy of the SDP "source filter" 
draft.  You can also find a copy online at
	http://www.live.com/draft-mmusic-sdp-srcfilter-04.txt

This version contains several formatting and 'I-D nit' improvements (thanks 
to feedback from Magnus Westerlund and Colin Perkins), and clarifies how 
source filters are applied to "c=" lines that specify multiple multicast 
addresses.

	Ross.

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



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Subject: Re: [MMUSIC] Comments on draft-ietf-mmusic-kmgmt-ext-07.txt
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"Peterson, Jon" wrote:

>

<snip>

>
> This of course speaks only to carrying SDP in SIP. But is it true that all
> of the protocols that currently use SDP (RTSP, SAP, SIP...) carry MIME
> types, and consequently could use S/MIME if needed?
>

Megaco (and MGCP) don't

-- Flemming

>
> > Fredrik
> >
> > > Colin
> > >
> > _______________________________________________
> > 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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From: Tom Taylor <taylor@nortelnetworks.com>
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        "'Fredrik Lindholm (EAB)'" <Fredrik.Lindholm@era.ericsson.se>,
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Subject: Re: [MMUSIC] Comments on draft-ietf-mmusic-kmgmt-ext-07.txt
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The other point about these protocols is that they are not intended to be handled by 
intermediate proxies.

Flemming Andreasen wrote:
> 
> "Peterson, Jon" wrote:
> 
> 
> 
> <snip>
> 
>>This of course speaks only to carrying SDP in SIP. But is it true that all
>>of the protocols that currently use SDP (RTSP, SAP, SIP...) carry MIME
>>types, and consequently could use S/MIME if needed?
>>
> 
> 
> Megaco (and MGCP) don't
> 
> -- Flemming
> 
> 

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Subject: Re: [MMUSIC] Comments on draft-ietf-mmusic-kmgmt-ext-07.txt
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Tom Taylor wrote:

> The other point about these protocols is that they are not intended to be handled by
> intermediate proxies.
>

This is true, an although I agree than an MG needs to place some amount of trust in its
MGC, not everybody want to place 100% trust in their MGC. For example, some MGs may want
to negotiate media stream ciphers (or at least keys) in such a way that their MGC will
not be able to decipher the media stream.

-- Flemming

>
> Flemming Andreasen wrote:
> >
> > "Peterson, Jon" wrote:
> >
> >
> >
> > <snip>
> >
> >>This of course speaks only to carrying SDP in SIP. But is it true that all
> >>of the protocols that currently use SDP (RTSP, SAP, SIP...) carry MIME
> >>types, and consequently could use S/MIME if needed?
> >>
> >
> >
> > Megaco (and MGCP) don't
> >
> > -- Flemming
> >
> >

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Subject: [MMUSIC] Backwards range and negative scale clarification (bugs 533691, 513880)
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--------------040401060004040602080805
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Hi Folks,    
In my attempt to clarify the negative scale and backwards range issues 
brought up by bugs http://rtsp.org/bug533691 &  
http://rtsp.org/bug513880 some issues surfaced with regards to the 
current definition of ranges in the spec.  I wanted to discuss these 
issues before sending off my write up.

Issue 1:
---------
Range is defined in RTSP as a half open interval.  To quote from draft-03

Ranges are half-open intervals, including the lower point, but
excluding the upper point. In other words, a range of a-b starts
exactly at time a, but stops just before b.

Now the resolution that we came up with for this issue was that we would 
always go from the first number in the range to the last number in the 
range.  So for example, if the range of a PLAY request was
    Range: npt=10-15
we would go from 10 -> 15, where the play would stop just before 15.  If 
the request had  range and scale headers as the following:
    Scale: -1
    Range: npt=15-10
the play would go backwards and we would go from 15 -> 10.

Now the question is how do we interpret the half-open rule.  I.e.  Do we 
stop at 10 or just before we reach 10 (i.e. npt > 10) in the backwards 
scenario above? The latter seems like they way to interpret the half - 
open rule.  But if we interpret it that way, then how do we deal with 
the following:
    Scale: -1
    Range: npt=15-0
We would have to stop before 0 and there would be no way to get to 0 
since you cannot specify negative ranges.  Obviously the way to get 
around this is to do the following:
    Scale: -1
    Range: npt=15-
But this doesn't seem too clean to me.

The other option is to say that  backwards ranges are always closed 
ranges.  This, however, makes ranges a bit more inconsisted and harder 
to deal with.

Issue 2:
----------
The other thing that doesn't make sense with backwards ranges are the 
Clock ranges.  For example:
    Scale: -1
    Range: clock=20030416T000000.00Z-20030101T000000.00Z
makes no sense because clocks cant go backwards.

So I propose that Scale can only be negative for ranges that deal with 
relative time rather than absolute time.



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</head>
<body>
Hi Folks,&nbsp;&nbsp; &nbsp;<br>
In my attempt to clarify the negative scale and backwards range issues
brought up by bugs <a class="moz-txt-link-freetext" href="http://rtsp.org/bug533691">http://rtsp.org/bug533691</a> &amp;&nbsp;
<a class="moz-txt-link-freetext" href="http://rtsp.org/bug513880">http://rtsp.org/bug513880</a> some issues surfaced with regards to the
current definition of ranges in the spec. &nbsp;I wanted to discuss these
issues before sending off my write up.<br>
<br>
Issue 1:<br>
---------<br>
Range is defined in RTSP as a half open interval. &nbsp;To quote from
draft-03<br>
<blockquote>
  <pre>Ranges are half-open intervals, including the lower point, but
excluding the upper point. In other words, a range of a-b starts
exactly at time a, but stops just before b.</pre>
</blockquote>
Now the resolution that we came up with for this issue was that we
would always go from the first number in the range to the last number
in the range. &nbsp;So for example, if the range of a PLAY request was <br>
<tt>&nbsp;&nbsp; &nbsp;Range: npt=10-15</tt> <br>
we would go from 10 -&gt; 15, where the play would stop just before 15.
&nbsp;If the request had&nbsp; range and scale headers as the following:<tt><br>
&nbsp; &nbsp; Scale: -1</tt><br>
<tt>&nbsp; &nbsp; Range: npt=15-10</tt> <tt><br>
</tt>the play would go backwards and we would go from 15 -&gt; 10.<br>
<br>
Now the question is how do we interpret the half-open rule. &nbsp;I.e. &nbsp;Do
we stop at 10 or just before we reach 10 (i.e. npt &gt; 10) in the
backwards scenario above? The latter seems like they way to interpret
the half - open rule. &nbsp;But if we interpret it that way, then how do we
deal with the following:<br>
<tt>&nbsp;&nbsp;&nbsp; Scale: -1<br>
&nbsp;&nbsp;&nbsp; Range: npt=15-0</tt><br>
We would have to stop before 0 and there would be no way to get to 0
since you cannot specify negative ranges. &nbsp;Obviously the way to get
around this is to do the following:<br>
<tt>&nbsp;&nbsp;&nbsp; Scale: -1<br>
&nbsp; &nbsp; Range: npt=15-<br>
</tt>But this doesn't seem too clean to me.<br>
<br>
The other option is to say that &nbsp;backwards ranges are always closed
ranges. &nbsp;This, however, makes ranges a bit more inconsisted and harder
to deal with.<br>
<br>
Issue 2:<br>
----------<br>
The other thing that doesn't make sense with backwards ranges are the
Clock ranges. &nbsp;For example:<br>
<tt>&nbsp; &nbsp; Scale: -1<br>
&nbsp;&nbsp;&nbsp; Range: clock=20030416T000000.00Z-20030101T000000.00Z<br>
</tt>makes no sense because clocks cant go backwards.<br>
<br>
So I propose that Scale can only be negative for ranges that deal with
relative time rather than absolute time.<br>
<br>
<br>
</body>
</html>

--------------040401060004040602080805--

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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 Protocol (SDP) Source Filters
	Author(s)	: B. Quinn, R. Finlayson
	Filename	: draft-ietf-mmusic-sdp-srcfilter-04.txt
	Pages		: 10
	Date		: 2003-4-16
	
This document describes how to adapt the Session Description Protocol
(SDP) to express one or more source addresses as a source filter for
one or more destination 'connection' addresses.  It defines the
syntax and semantics for an SDP 'source-filter' attribute that may
reference either IPv4 or IPv6 address(es) as either an inclusive or
exclusive source list for either multicast or unicast destinations.
In particular, an inclusive source-filter can be used to specify a
Source-Specific Multicast (SSM) session.

A URL for this Internet-Draft is:
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From mailnull@www1.ietf.org  Thu Apr 17 18:05:44 2003
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Date: Fri, 18 Apr 2003 00:01:05 +0200
From: Kostas Katrinis <katrinis@ceid.upatras.gr>
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Subject: [MMUSIC] Comments on draft-levin-sipping-conferencing-requirements-03.txt
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    Dear all,

although the draft-levin-sipping-conferencing-requirements-03.txt 
expired last March, I wanted to ask some guideline questions on this:

- according to the functionality offered by SIP as mentioned in RFC 
3261, SIP is responsible for creation/termination and management of 
sessions. As such, is it in the scope of SIP (or of services implemented
via SIP messages) to facilitate also conference control?
As an example,in
draft-levin-sipping-conferencing-requirements-03.txt the following
functionality is posed as a required service(citing the draft):
"   REQ -2: Given a conference identifier, a means MUST be defined for a
   user agent to remove a participant from the conference."

Given that participant 1 (that is authorized to remove a participant) 
does not have the dialog information in order to remove participant 2
from the conference, participant 1 should first signal the conference
server the command to remove participant 2 from the conference and then
the conference focus should take over the removal of participant 2.
Now, should the instruction from the removal issued from the participant 
1 to the focus be implemented with SIP means or with another
proprietary protocol?

- regarding the conference policy control protocol, is there any 
provision to standardize it? Will it be implemented in SIP scope 
(additional SIP messages) or by defining a new protocol?

Thank you in advance,
Kostas Katrinis


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Date: Thu, 17 Apr 2003 17:49:59 -0700
From: Tom Marshall <tmarshall@real.com>
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Subject: [MMUSIC] What is "current"?
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Here's an interesting issue.  RFC2326 says this:

   Since many of the definitions and syntax are identical to HTTP/1.1, =20
   this specification only points to the section where they are defined
   rather than copying it. For brevity, [HX.Y] is to be taken to refer
   to Section X.Y of the current HTTP/1.1 specification (RFC 2068 [2]).

"current" can be taken two ways: current as of the writing, or current as of
the reading.

Further, it seems the meaning of an empty "Accept-Encoding:" header has been
reversed between 2068 and 2616.  The former says explicitly that it means
none are acceptable and the latter says that the "identity" encoding is
always acceptable unless explicitly refused.

It seems that there is a bug in the HTTP spec and this makes it a bug in the
RTSP spec.

What is a well-behaved server to do in this case???

--=20
A revised schedule is to business what a new season is to an athlete or a
new canvas to an artist.
        -- Norman Augustine

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From: Aaron Colwell <acolwell@real.com>
To: Tom Marshall <tmarshall@real.com>
cc: mmusic@ietf.org
Subject: Re: [MMUSIC] What is "current"?
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Looks like you are our server guy looking into this bug. :)

Here is my interpretation of the HTTP spec. In RFC 2068 it says "An empty
Accept-Encoding value indicates none are acceptable". I interpret that to
mean that none of the encoding types (ie. compress or gzip) are
acceptable. Since not specifying the Accept-Encoding field means that the
server can choose any encoding it wants, sending the header with an empty
value seems to me to be the only way to tell the server you don't want it
to use any encoding.

I think that RFC 2616 tries to clarify this condition, but makes things a
little confusing by introducing the "identity" encoding. RFC 2616 says "If
the Accept-Encoding field-value is empty, then only the "identity"
encoding is acceptable." If you use my interpretation of RFC 2068, then
the 2 RFCs are consistent. If my interpretation is wrong then I agree with
you that they are inconsistent.

Aaron

On Thu, 17 Apr 2003, Tom Marshall wrote:

> Here's an interesting issue.  RFC2326 says this:
>
>    Since many of the definitions and syntax are identical to HTTP/1.1,
>    this specification only points to the section where they are defined
>    rather than copying it. For brevity, [HX.Y] is to be taken to refer
>    to Section X.Y of the current HTTP/1.1 specification (RFC 2068 [2]).
>
> "current" can be taken two ways: current as of the writing, or current as of
> the reading.
>
> Further, it seems the meaning of an empty "Accept-Encoding:" header has been
> reversed between 2068 and 2616.  The former says explicitly that it means
> none are acceptable and the latter says that the "identity" encoding is
> always acceptable unless explicitly refused.
>
> It seems that there is a bug in the HTTP spec and this makes it a bug in the
> RTSP spec.
>
> What is a well-behaved server to do in this case???
>
> --
> A revised schedule is to business what a new season is to an athlete or a
> new canvas to an artist.
>         -- Norman Augustine
>
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Subject: Re: [MMUSIC] Comments on draft-levin-sipping-conferencing-requirements-03.txt
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inline.

Kostas Katrinis wrote:
>    Dear all,
> 
> although the draft-levin-sipping-conferencing-requirements-03.txt 
> expired last March, I wanted to ask some guideline questions on this:
> 
> - according to the functionality offered by SIP as mentioned in RFC 
> 3261, SIP is responsible for creation/termination and management of 
> sessions. As such, is it in the scope of SIP (or of services implemented
> via SIP messages) to facilitate also conference control?

No.

> As an example,in
> draft-levin-sipping-conferencing-requirements-03.txt the following
> functionality is posed as a required service(citing the draft):
> "   REQ -2: Given a conference identifier, a means MUST be defined for a
>   user agent to remove a participant from the conference."
> 
> Given that participant 1 (that is authorized to remove a participant) 
> does not have the dialog information in order to remove participant 2
> from the conference, participant 1 should first signal the conference
> server the command to remove participant 2 from the conference and then
> the conference focus should take over the removal of participant 2.
> Now, should the instruction from the removal issued from the participant 
> 1 to the focus be implemented with SIP means or with another
> proprietary protocol?

A different protocol will be used for conference control. The 
conferencing design team in sipping is working on it. There is no 
proposed protocol right now, though SOAP is a likely candidate. We do 
have requirements for this protocol however:

http://www.ietf.org/internet-drafts/draft-koskelainen-sipping-conf-policy-req-00.txt

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Scientist                             Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
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From: Rob Lanphier <robla@real.com>
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Subject: [MMUSIC] Minutes for RTSP telecon April 16
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Hi all,

Below are the minutes from the last RTSP conf call.  They are also
online here:

http://rtsp.org/telecon/

Rob
---
Rob Lanphier, Helix Community Coordinator - RealNetworks
http://helixcommunity.org http://rtsp.org http://realnetworks.com

= Attendees =
 * Magnus Westerlund
 * Rob Lanphier
 * Aravind Narasimhan
 * Sean Sheedy
 * Andrea Basso
 * Thomas Zeng
 * Anders Klemets

= Bug discussion =
   
== Bug 513737 ==

 * http://rtsp.org/bug513737
 * Magnus: Who will write it up?
 * Aravind: if I get time
 * Aravind: in RTSP, range is defined as half-open interval
 * Aravind: with negative ranges, there's no way to get to zero
 * Magnus: is this really a problem?
 * Thomas: most video file have a frame at time=0
 * Magnus: we could define a corner case for 0.  I.e. the range would
stay a half-open interval with the first number closed and the last
number open, except in the case of 0.
 * Aravind: What about reversing the half-open interval
 * Rob: I like this
 * Andrea: what about I-frames
 * Thomas: it works the same as forwards
 * Andrea: it seems that consistency is more expressive
 * Aravind: now there's no way to start at a given time. Also the range
header becomes harder to deal with because which interval is open
depends on the Scale header.
 * Sean: define range to always mean that smallest range that is
possible.
 * Thomas: changes existing implementations
 * Magnus: I'm don't think it does
 * Thomas: I'm still concerned
 * Magnus: we need a writeup
 * Sean: we could make range closed on both ends, rather than having it
as half open. this definition change probably won't change current
implementations (at least, not by enough to break things).
 * Thomas: why is 2326 the way it is
 * Rob: thought was about stringing together multiple ranges
 * conclusion: need more discussion on email

== Bug 533691 and Bug 513880 ==
 * http://rtsp.org/bug533691 &  http://rtsp.org/bug513880
 * Aravind to write up, progress?
 * Magnus: need more discussion on email

== Issue #2 in Aravind's email ==
 * See
[http://www1.ietf.org/mail-archive/working-groups/mmusic/current/msg01388.html Aravind's email]
 * Magnus: it is ok to have backwards clock ranges because you may have
film index by clock ranges and you may want to go backwards in time.
 * Aravind:  It seems to me that clock ranges refer to absolute time and
that they shouldn't be allowed to go backwards.
 * magnus: Don't see a problem with clock ranges going backwards.

== Bug 556061 ==
 * http://rtsp.org/bug556061
 * Fix state machine to allow for REDIRECT at any time, including in
INIT state.
 * Magnus: needs writeup

== Bug 556295 ==
  
 * http://rtsp.org/bug556295 
 * Client issues PAUSE while server ready
 * Allow: needs explanation.
 * Magnus: been discussed...needs writeup
 * Thomas: I can write this up

== Remove timed PLAY and PAUSE? ==
 * Discussed at San Francisco IETF
 * Proposal: Remove timed PLAY and PAUSE. 
 * Magnus: With time I mean the specified usage of the time parameter in
the Range header.

== Stream switching requirements document (Phillipe Gentric) ==
 * Waiting for writeup

== Offer Answer for RTSP ==
 *  Anybody willing to write requirements and use case?
 *  Rob: no one really wanted to commit to this
 *  Thomas: Phillipe may be interested
 *  Andrea: May also be interested
 *  Andrea will check into thsi

== IANA registration comments? ==
 * Magnus: send comments to the list

== Ways to allow direct media destination to other than own IP address
==
 * Magnus: Phillipe Gentric volunteered to look into this

== Multiple SSRC in RTSP ==
 * Magnus: Redefine RTP-Info to solve issue?
 * Thomas: this is because of stream switching
 * Magnus: no, relaying different sources
 * Magnus: e.g. relaying MBONE sessions
 * Thomas: our server mixes
 * Aravind: what about multiple participants?
 * Thomas: those should be different streams
 * Rob: that's inconsisent with standard use of RTP
 * Rob: we can solve this for the single SSRC case in RTP-Info, and then
 * Aravind: are we requiring SSRC in RTP-Info now?
 * Magnus: no...just explaining the limitation
 * Aravind: what about SSRC collision resolution?
 * Thomas: doesn't need to involve RTSP
 * Magnus: client is responsible for resolving
 * Rob: RTP-Info should be honored as initialization info only...RTCP
takes over after that
 * Magnus: problem is when server is invited to existing RTP session
 * Thomas: can't really think about this problem without someone who's
done it
 * Magnus: need more discussion
 * Thomas: I think Rob's proposal is sound
 * Aravind: what happens when client requests SSRC?
 * Magnus: comparing 2326 to current version.  Non-normative suggestion
to use SSRC from client.
 * Thomas: current 2326bis SSRC request is for unicast only
 * Aravind: multicast source gateway to unicast, the client may request
SSRC...how does server honor?
 * Magnus: need to whether reconsider SSRC requests are useful or even
valid
 * Aravind will write up

== How to distinguish the updated specification ==
 * Magnus: Need to consider bumping current version number to RTSP/1.1
 * Thomas: we should increase version number
 * Magnus: if we work quickly, we can pull this off
 * Thomas: I can write this up

== Next meeting ==

 * Monday, April 28, at 9am PDT (GMT-7)/18:00 CMT (GMT+2)

== Postponed items ==
 * Bugs
   * Warning header: For further discussion.
   *  RTSP servers behind NATs. Use case seems to exist. Requires
someone to write scenario document. Decide on how to go forward compared
with public network servers. Time line considerations.
   * RTSP ALG for Linux netfilter. (Tom Marshall)
 * How to get things into the spec?
 * New Things
   *  Specification review. Plan to look through the latest draft and
see what comments people has about it.
   * http://rtsp.org/bug631148 <http://rtsp.org/bug556061> better Info
on RTSP proxies.
   *  http://rtsp.org/bug631957 Use of # and ? in RTSP URLs?
   *  http://rtsp.org/bug633148 Use of TCP and UDP.
   * http://rtsp.org/bug633779 SET_PARAMETER responses.


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Date: Thu, 17 Apr 2003 19:16:27 +0200
From: Kostas Katrinis <katrinis@tik.ee.ethz.ch>
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Organization: TIK - ETH Zurich
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Subject: [MMUSIC] Comments on draft-levin-sipping-conferencing-requirements-03.txt
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	Dear all,

although the draft-levin-sipping-conferencing-requirements-03.txt 
expired last March, I wanted to ask some guideline questions on this:

- according to the functionality offered by SIP as mentioned in RFC 
3261, SIP is responsible for creation/termination and management of 
sessions. As such, is it in the scope of SIP (or of services implemented
via SIP messages) to facilitate also conference control?
As an example,in
draft-levin-sipping-conferencing-requirements-03.txt the following
functionality is posed as a required service(citing the draft):
"   REQ -2: Given a conference identifier, a means MUST be defined for a
    user agent to remove a participant from the conference."

Given that participant 1 (that is authorized to remove a participant) 
does not have the dialog information in order to remove participant 2
from the conference, participant 1 should first signal the conference
server the command to remove participant 2 from the conference and then
the conference focus should take over the removal of participant 2.
Now, should the instruction from the removal issued from the participant 
1 to the focus be implemented with SIP means or with another
proprietary protocol?

- regarding the conference policy control protocol, is there any 
provision to standardize it? Will it be implemented in SIP scope 
(additional SIP messages) or by defining a new protocol?

Thank you in advance,
Kostas Katrinis

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/* PLEASE APOLOGIZE IF YOU RECEIVE MULTIPLE COPIES OF THIS ANNOUNCEMENT =
* /
/*  LAST DAYS FOR ARTICLE SUBMISSIONS TO IFIP/IEEE MMNS 2003
/* THE SUBMISSION DEADLINE IS 18th APRIL 2003 */
Best Regards
A. Mehaoua (Univ. of Versailles)

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

 Please distribute to interested people to encourage contributions and =
submissions to MMNS 2003
6th IFIP/IEEE International Conference on Management of Multimedia =
Networks and Services
        7th - 10th September 2003, Queen's University of Belfast, =
Northern Ireland=20
                            http://www.ee.qub.ac.uk/mmns2003=20


                                  Second Call For Papers

Multimedia services over IP networks are proliferating at an enormous =
speed. There=20
is also increasing demand for solutions which provide assured levels of =
service quality.=20
All of these require novel paradigms, models and architectures for =
realising integrated=20
end-to-end service management rather than managing network elements in =
isolation.=20
Providing scalable Quality of Service (QoS) while maintaining fairness, =
along with=20
secure and optimal network resource management are key challenges for =
the future=20
Internet. These challenges apply to both fixed and wireless networks.=20

The IFIP/IEEE International Conference on Management of Multimedia =
Networks and Services=20
will hold its sixth annual meeting from September 7th to September 10th, =
2003 in Belfast,
Northern Ireland. MMNS provides an intimate setting for discussion and =
debate. In just 6 years,=20
MMNS has established itself as one of the premier conferences with a =
focus on the management=20
of multimedia networks and services. The conference objective is to =
bring together researchers=20
working in all facets of network and service management as applied to =
broadband networks and=20
multimedia services. MMNS deals with all aspects of designing, =
developing and deploying=20
networked multimedia systems and it serves as a forum for the =
dissemination of state-of-the-art=20
research and development results.=20

MMNS 2003 will also include panel sessions in which experts=20
offer their observations and opinions about current hot topics. The =
keynote speaker will be=20
Professor Ian Akyildiz, Georgia Institute of Technology who will present =
a vision of future=20
interplanetary network architectures. Professor Derek McAuley, head of =
Intel's recently formed=20
laboratory at Cambridge, UK, will describe some of the new research =
being undertaken on global=20
overlay networks and applications.


The program committee is soliciting original papers describing research =
in the area of=20
management of multimedia networks and services. Topics of interest =
include, but are not=20
limited to, the following:

        * Active multimedia network management=20
        * Ad-hoc and Sensor Networks=20
        * Augmented and Virtual Reality Networks=20
        * Billing and Accounting=20
        * Cable multimedia network management=20
        * Content distribution internetworking=20
        * Deployment of multimedia services=20
        * Distributed multimedia service management=20
        * End-to-end IP multimedia network and service management=20
        * IP Video, streaming, interactive video service management=20
        * Middleware support for management=20
        * Multimedia network traffic engineering and optimization=20
        * Multimedia traffic management=20
        * Multimedia content protection=20
        * Multimedia session management=20
        * Multi-point, multicast services management=20
        * Network management models and architectures=20
        * Network programmability for multimedia services=20
        * Optical multimedia network management=20
        * Policy-based management for multi-media services=20
        * Provisioning of multimedia networks and services=20
        * QoS in WLANs=20
        * QoS management=20
        * Resource, performance and fault management=20
        * Security and Authentication=20
        * VoIP service management=20
        * Web Services=20
        * Wireless and mobile multimedia network management=20


Papers must be submitted electronically in postscript or PDF format. =
Detailed=20
instructions are provided on the conference web site, =
http://www.ee.qub.ac.uk/mmns2003.=20

Submission date:            18th April 2003
Notification of acceptance: 6th June 2003
Final version:              4th July 2003

Conference Chairs:
   Professor Alan Marshall, a.marshall@ee.qub.ac.uk, Queen's University =
of Belfast, UK
   Professor Nazim Agoulmine, nazim@rp.lip6.fr, University of Evry, =
France



------=_NextPart_000_1029_01C305A7.66886D00
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	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
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<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Dear Colleagues, </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>/* PLEASE APOLOGIZE IF YOU RECEIVE =
MULTIPLE COPIES=20
OF THIS ANNOUNCEMENT * /</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>/*&nbsp; LAST DAYS FOR ARTICLE =
SUBMISSIONS&nbsp;TO=20
IFIP/IEEE MMNS 2003</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>/*&nbsp;THE SUBMISSION DEADLINE&nbsp;IS =
18th APRIL=20
2003 */</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Best Regards</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>A. Mehaoua (Univ.&nbsp;of=20
Versailles)</FONT></DIV><FONT face=3DArial size=3D2></FONT>
<DIV><BR><FONT face=3DArial size=3D2><FONT face=3DArial size=3D2><FONT =
face=3DArial=20
size=3D2><FONT face=3D"Times New Roman"=20
size=3D3>----------------------------------------------------------------=
---<BR><BR>&nbsp;Please=20
distribute to interested people to encourage contributions and =
submissions to=20
</FONT></FONT></FONT></FONT><FONT size=3D2><FONT =
face=3DArial><STRONG>MMNS=20
2003<BR></STRONG>6th IFIP/IEEE International Conference on Management of =

Multimedia Networks and =
Services<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
7th - 10th September 2003, Queen's University of Belfast, Northern =
Ireland=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;=20
</FONT></FONT><A href=3D"http://www.ee.qub.ac.uk/mmns2003"><FONT =
face=3DArial=20
size=3D2>http://www.ee.qub.ac.uk/mmns2003</FONT></A><FONT face=3DArial =
size=3D2>=20
</FONT></DIV>
<DIV>
<DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><BR><FONT size=3D2><FONT=20
face=3DArial>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<STRONG>Second Call For Papers</STRONG></FONT></FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Multimedia services over IP networks =
are=20
proliferating at an enormous speed. There <BR>is also increasing demand =
for=20
solutions which provide assured levels of service quality. <BR>All of =
these=20
require novel paradigms, models and architectures for realising =
integrated=20
<BR>end-to-end service management rather than managing network elements =
in=20
isolation. <BR>Providing scalable Quality of Service (QoS) while =
maintaining=20
fairness, along with <BR>secure and optimal network resource management =
are key=20
challenges for the future <BR>Internet. These challenges apply to both =
fixed and=20
wireless networks. </FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>The IFIP/IEEE International Conference =
on=20
Management of Multimedia Networks and Services <BR>will hold its sixth =
annual=20
meeting from September 7th to September 10th, 2003 in =
Belfast,<BR>Northern=20
Ireland. MMNS provides an intimate setting for discussion and debate. In =
just 6=20
years, <BR>MMNS has established itself as one of the premier conferences =
with a=20
focus on the management <BR>of multimedia networks and services. The =
conference=20
objective is to bring together researchers <BR>working in all facets of =
network=20
and service management as applied to broadband networks and =
<BR>multimedia=20
services. MMNS deals with all aspects of designing, developing and =
deploying=20
<BR>networked multimedia systems and it serves as a forum for the =
dissemination=20
of state-of-the-art <BR>research and development results. </FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>MMNS 2003 will also include panel =
sessions in which=20
experts <BR>offer their observations and opinions about current hot =
topics. The=20
keynote speaker will be <BR>Professor Ian Akyildiz, Georgia Institute of =

Technology who will present a vision of future <BR>interplanetary =
network=20
architectures. Professor Derek McAuley, head of Intel's recently formed=20
<BR>laboratory at Cambridge, UK, will describe some of the new research =
being=20
undertaken on global <BR>overlay networks and applications.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><BR><FONT face=3DArial size=3D2>The program committee is soliciting =
original=20
papers describing research in the area of <BR>management of multimedia =
networks=20
and services. Topics of interest include, but are not <BR>limited to, =
the=20
following:</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial =
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Active=20
multimedia network management =
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; *=20
Ad-hoc and Sensor Networks =
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; *=20
Augmented and Virtual Reality Networks=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Billing and Accounting=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Cable multimedia =
network=20
management <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Content =
distribution=20
internetworking <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * =
Deployment of=20
multimedia services <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * =
Distributed=20
multimedia service management =
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; *=20
End-to-end IP multimedia network and service management=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * IP Video, streaming,=20
interactive video service management=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Middleware support for=20
management <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Multimedia =
network=20
traffic engineering and optimization=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Multimedia traffic =
management=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Multimedia content =
protection=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Multimedia session =
management=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Multi-point, multicast =
services=20
management <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Network =
management=20
models and architectures <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
*=20
Network programmability for multimedia services=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Optical multimedia =
network=20
management <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Policy-based =

management for multi-media services=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Provisioning of =
multimedia=20
networks and services <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * =
QoS in=20
WLANs <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * QoS management=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Resource, performance =
and fault=20
management <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Security and =

Authentication <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * VoIP =
service=20
management <BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Web Services =

<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Wireless and mobile =
multimedia=20
network management </FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><BR><FONT face=3DArial size=3D2>Papers must be submitted =
electronically in=20
postscript or PDF format. Detailed <BR>instructions are provided on the=20
conference web site, </FONT><A =
href=3D"http://www.ee.qub.ac.uk/mmns2003"><FONT=20
face=3DArial size=3D2>http://www.ee.qub.ac.uk/mmns2003</FONT></A><FONT =
face=3DArial=20
size=3D2>. </FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><STRONG><FONT face=3DArial size=3D2>Submission=20
date:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<F=
ONT=20
color=3D#ff0000> 18th April 2003<BR></FONT>Notification of acceptance: =
6th June=20
2003<BR>Final=20
version:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;=20
4th July 2003</FONT></STRONG></DIV>
<DIV><STRONG><FONT face=3DArial size=3D2></FONT></STRONG>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Conference Chairs:<BR>&nbsp;&nbsp; =
Professor Alan=20
Marshall, </FONT><A href=3D"mailto:a.marshall@ee.qub.ac.uk"><FONT =
face=3DArial=20
size=3D2>a.marshall@ee.qub.ac.uk</FONT></A><FONT face=3DArial size=3D2>, =
Queen=92s=20
University of Belfast, UK<BR>&nbsp;&nbsp; Professor Nazim Agoulmine, =
</FONT><A=20
href=3D"mailto:nazim@rp.lip6.fr"><FONT face=3DArial=20
size=3D2>nazim@rp.lip6.fr</FONT></A><FONT face=3DArial size=3D2>, =
University of Evry,=20
France<BR></FONT></DIV>
<DIV><FONT face=3DArial =
size=3D2></FONT>&nbsp;</DIV></DIV></DIV></BODY></HTML>

------=_NextPart_000_1029_01C305A7.66886D00--

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Date: Tue, 22 Apr 2003 20:52:49 -0700
From: Tom Marshall <tmarshall@real.com>
To: mmusic@ietf.org
Message-ID: <20030423035249.GA22348@real.com>
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Subject: [MMUSIC] Secure RTSP thoughts
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Secure RTSP has been the topic of several conversations recently, both in
the telecons and internally at work.  I got to thinking about the problem
and came up with a possible framework.  It's by no means complete, it's just
a start.

I briefly read the SIP spec recently.  SIP has the capability to encrypt
portions of the messages.  What if we applied that capability to RTSP?  We
might be able to come up with a backward compatible, firewall friendly,
encryption capable protocol.  Most of the things that one would want
encrypted are in the headers or entity already: the WWW-Authenticate header
and the DESCRIBE response entity.  Here's an example:

  DESCRIBE rtsp://media.example.org/video.mpeg RTSP/1.0
  CSeq: 1
  Content-Type: application/encrypted
  Content-Length: 27

  WWW-Authenticate: foo:bar

The WWW-Authenticate header would be sent encrypted.  The response might be
like this:

  RTSP/1.0 200 OK
  CSeq: 1
  Date: Wed, 23 Apr 2003 04:36:22 GMT
  Content-Type: application/encrypted
  Content-Length: 1344

  Content-Base: rtsp://media.example.org/video.mpeg/
  Content-Type: application/sdp
  Content-Length: 1234

  v=3D0
  ...

Here the last three headers and the SDP are encrypted.  Note that there is
no use in encrypting the Content-Base when the request-uri is sent in the
clear.  The request-uri could be sent as "*" and a new header in the
encrypted section could carry the actual request-uri.  We could call this
header something tricky like Request-URI.

Some headers obviously should never be encrypted.  Examples include CSeq,
Transport, and Via.  A list of such headers would need to be made and
included in the spec.

Proxies might need special treatment.  If one would want to encrypt some
data to the proxy and other data to the server, a double encryption might be
useful.

And finally, we would need to come up with a tag for this feature in the new
spec.

Thoughts?

--=20
Windows 2000: Designed for the Internet. The Internet: Designed for UNIX.

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You are basically talking about using S/MIME.

S/MIME is useful in SIP because of its peer-to-peer nature. But, it 
comes at a high cost - the requirement to have a PKI. Especially for 
encryption, where you need the key of the recipient.

If RTSP is focusing now on the client/server model exclusively (per our 
nat debate), I would think that TLS is a better match. Doing the 
equivalent of sips - rtsps - would be much more deployable, since it 
would not require some additional key distribution mechanism.

-Jonathan R.

Tom Marshall wrote:
> Secure RTSP has been the topic of several conversations recently, both
> in
> the telecons and internally at work.  I got to thinking about the
> problem
> and came up with a possible framework.  It's by no means complete, it's
> just
> a start.
> 
> I briefly read the SIP spec recently.  SIP has the capability to encrypt
> portions of the messages.  What if we applied that capability to RTSP?
> We
> might be able to come up with a backward compatible, firewall friendly,
> encryption capable protocol.  Most of the things that one would want
> encrypted are in the headers or entity already: the WWW-Authenticate
> header
> and the DESCRIBE response entity.  Here's an example:
> 
>   DESCRIBE rtsp://media.example.org/video.mpeg RTSP/1.0
>   CSeq: 1
>   Content-Type: application/encrypted
>   Content-Length: 27
> 
>   WWW-Authenticate: foo:bar
> 
> The WWW-Authenticate header would be sent encrypted.  The response might
> be
> like this:
> 
>   RTSP/1.0 200 OK
>   CSeq: 1
>   Date: Wed, 23 Apr 2003 04:36:22 GMT
>   Content-Type: application/encrypted
>   Content-Length: 1344
> 
>   Content-Base: rtsp://media.example.org/video.mpeg/
>   Content-Type: application/sdp
>   Content-Length: 1234
> 
>   v=0
>   ...
> 
> Here the last three headers and the SDP are encrypted.  Note that there
> is
> no use in encrypting the Content-Base when the request-uri is sent in
> the
> clear.  The request-uri could be sent as "*" and a new header in the
> encrypted section could carry the actual request-uri.  We could call
> this
> header something tricky like Request-URI.
> 
> Some headers obviously should never be encrypted.  Examples include
> CSeq,
> Transport, and Via.  A list of such headers would need to be made and
> included in the spec.
> 
> Proxies might need special treatment.  If one would want to encrypt some
> data to the proxy and other data to the server, a double encryption
> might be
> useful.
> 
> And finally, we would need to come up with a tag for this feature in the
> new
> spec.
> 
> Thoughts?
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Scientist                             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: Tue, 22 Apr 2003 21:57:39 -0700
From: Tom Marshall <tmarshall@real.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] Secure RTSP thoughts
Message-ID: <20030423045739.GA22517@real.com>
References: <20030423035249.GA22348@real.com> <3EA61886.9090605@dynamicsoft.com>
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On Wed, Apr 23, 2003 at 12:37:26AM -0400, Jonathan Rosenberg wrote:
> You are basically talking about using S/MIME.
>=20
> S/MIME is useful in SIP because of its peer-to-peer nature. But, it=20
> comes at a high cost - the requirement to have a PKI. Especially for=20
> encryption, where you need the key of the recipient.

There must be some way to make it work reasonably well.  I am not an
encryption expert, but I can start reading up on it.

> If RTSP is focusing now on the client/server model exclusively (per our=
=20
> nat debate), I would think that TLS is a better match. Doing the=20
> equivalent of sips - rtsps - would be much more deployable, since it=20
> would not require some additional key distribution mechanism.

We may focus on client/server but RTSP does have peer-to-peer uses.

Doesn't TLS encrypt all of the data?  I think that is quite undesirable.=20
You lose the ability to use ALGs and existing RTSP proxies.

--=20
The problem with most conspiracy theories is that they seem to believe that
for a group of people to behave in a way detrimental to the common good
requires intent.

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Hi Tom,

It seems evident that we must start discussing what we want to 
accomplish with securing the RTSP messages.

Some requirements on what I see as necessary for transport secured RTSP:

+ Message authentication to prevent tampering. In my opinion we must 
authenticate the whole message to provide any benefit. This will create 
problems with ALGs and Proxies. The Transport header is a vital header 
for RTSP functionality and it is desired to have it secured, and at the 
same time it is the header that an proxy or ALG needs to both read and 
possibly change.

+ Message privacy, I don't know what parts we actually need to have 
encrypted. There is several issues here, user privacy, preventing 
disclosure of information that an attacker could use.

What should the security architecture be? Do we need two levels, one 
between each entity, i.e. between client and proxy and then proxy to 
server. Then a second level that is protected end to end, i.e. client 
<-> server.

I think we should get a security advisor for this. One knowledgeable 
about these issues would be very good.

Colin and Joerg: Who is our security advisor?

>>If RTSP is focusing now on the client/server model exclusively (per
>>    
>>
>our 
>  
>
>>nat debate), I would think that TLS is a better match. Doing the 
>>equivalent of sips - rtsps - would be much more deployable, since it 
>>would not require some additional key distribution mechanism.
>>    
>>
>
>We may focus on client/server but RTSP does have peer-to-peer uses.
>
>Doesn't TLS encrypt all of the data?  I think that is quite undesirable.
>
>You lose the ability to use ALGs and existing RTSP proxies.
>  
>
As pointed out above, I think we need to think about what we would like 
to accomplish. We need to review what security we need and what is 
possible to fit into this. I think we will have to face some hard 
decisions before the end of this work. Would a proxy really be able to 
handle such a message you propose. One bug I see is that the proxy needs 
at least the host part of the request URI to be able to send the request 
to the correct entity. And if you need trusted proxies, I think one 
actually needs consider that these proxies must be updated and security 
audited.

I get the feeling that using TLS is rather simple but requires a limited 
usage model or trusted proxies. And as proxies can't really be trusted, 
what do we do?


Best Regards

Magnus Westerlund 

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



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From: "Amedeo Michelin-Salomon" <Michelin@ingegneria.unime.it>
To: <mmusic@ietf.org>
Date: Tue, 22 Apr 2003 14:08:00 +0200
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Subject: [MMUSIC] Help rtp Timestamp
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 Hi,I have read your post on ucb.digital-video. I'm a Italian Student
 of Eletronical Engineering....I have develop a application for VOD on
 Wireless System.....I have a problem to calculate the NTP of the clip
 for a fix Timestamp.....do you have a solution for this problem? i
 have read that NTP = (LastTimestam-FirstTimestam)/frequency but i
 don't know how get this frequency....  thanx for help
 AMS
Michelin@ingegneria.unime.it 

P.s. my application in over RTSP on RTP
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Subject: [MMUSIC] RTSP telecon 28 of Apri - Note date.
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Hi,

Here is the agenda for MONDAY's RTSP telecon at 18.00 CET (GMT+2). 
Details of the meeting is sent on the RTSP logistics list. If you are 
not part of the list and want to participate, please send me a mail.

Regards

Magnus

----------------------------
Agenda 030428

1. Opening of meeting.
2. Since last meeting (16th of April).
   a. http://rtsp.org/bug513737 - Waiting for write up.
   b. Bug 533691 &  513880 - Aravind to write up, progress?
      - Issue with range header in Scale=-1
   c. http://rtsp.org/bug556295 - Thomas to write text.
   d. Stream switching requirements document (Phillipe Gentric)
   e. Offer Answer for RTSP. Anybody willing to write requirements and 
use case?
      - Andrea Basso or Phillipe Gentric?
   f. Ways to allow direct media destination to other than own IP 
address. (Phillipe Gentric)
   g. Multiple SSRC in RTSP.
      - Conclusion explain what is supported today. That given SSRC in 
transport header is connected to RTP-Info. Use RTCP for other media 
streams or develop a extension
      - Aravind to write updated text.
   h. How to distinguish the updated specification. Please read up on 
how versions are handled in the HTTP specification.
   i. Warning header: For further discussion.
   j. RTSP servers behind NATs. Use case seems to exist. Requires 
someone to write scenario document. Decide on how to go forward compared 
with public network servers. Time line considerations.
   k. RTSP ALG for Linux netfilter. (Tom Marshall)

3. New Things
   a. http://rtsp.org/bug631148 better Info on RTSP proxies.
   b. http://rtsp.org/bug631957 Use of # and ? in RTSP URLs?
   c. http://rtsp.org/bug633148 Use of TCP and UDP.
   d. http://rtsp.org/bug633779 SET_PARAMETER responses. 
  
5. Any other Issues
6. Next meeting
7. End meeting

-- 

Magnus Westerlund 

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


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cc: mmusic@ietf.org
Subject: Re: [MMUSIC] Secure RTSP thoughts 
In-Reply-To: Your message of "Wed, 23 Apr 2003 09:48:50 +0200."
             <3EA64562.2070608@era.ericsson.se> 
Date: Wed, 23 Apr 2003 09:54:26 -0400
From: Colin Perkins <csp@csperkins.org>
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--> Magnus Westerlund writes:
>Hi Tom,
>
>It seems evident that we must start discussing what we want to 
>accomplish with securing the RTSP messages.
>
>Some requirements on what I see as necessary for transport secured RTSP:
>
>+ Message authentication to prevent tampering. In my opinion we must 
>authenticate the whole message to provide any benefit. This will create 
>problems with ALGs and Proxies. The Transport header is a vital header 
>for RTSP functionality and it is desired to have it secured, and at the 
>same time it is the header that an proxy or ALG needs to both read and 
>possibly change.
>
>+ Message privacy, I don't know what parts we actually need to have 
>encrypted. There is several issues here, user privacy, preventing 
>disclosure of information that an attacker could use.
>
>What should the security architecture be? Do we need two levels, one 
>between each entity, i.e. between client and proxy and then proxy to 
>server. Then a second level that is protected end to end, i.e. client 
><-> server.
>
>I think we should get a security advisor for this. One knowledgeable 
>about these issues would be very good.
>
>Colin and Joerg: Who is our security advisor?

Eric Rescorla.

Colin
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Date: Wed, 23 Apr 2003 13:54:05 -0700
To: mmusic@ietf.org
From: Ross Finlayson <finlayson@live.com>
Subject: Re: [MMUSIC] Secure RTSP thoughts
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At 12:48 AM 4/23/03, Magnus Westerlund wrote:
>What should the security architecture be? Do we need two levels, one 
>between each entity, i.e. between client and proxy and then proxy to 
>server. Then a second level that is protected end to end, i.e. client <-> 
>server.

One possible approach would be for us to *first* specify an end-to-end 
security solution - e.g., using TLS ("rtsps://").  Although this would not 
work with ALGs and proxies, it would be relatively easy to do (building on 
existing work), and be very useful.

In the longer term, we could also look at the more complex problem of 
defining an alternative, multi-layered solution that would work with 
ALGs/proxies.

>+ Message privacy, I don't know what parts we actually need to have 
>encrypted.

One clear requirement is to encrypt the SDP description that gets returned 
in response to a "DESCRIBE", because that will contain decryption keys (if 
SRTP is to be used for media transport).

I think that a TLS-based solution - even though it would work only 
end-to-end - would be a good starting point, and it would help make 
RTSP/SRTP streaming a reality fairly quickly.

>I think we should get a security advisor for this. One knowledgeable about 
>these issues would be very good.

Yes - definitely!

         Ross.

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Date: Thu, 24 Apr 2003 15:20:45 -0700
To: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
From: Mark Baugher <mbaugher@cisco.com>
Subject: Re: [MMUSIC] Secure RTSP thoughts
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Magnus
   This reminds me of a topic that you raised at the last meeting: We need 
to define RTSP-over-SSL/TLS.  I don't understand what needs to be defined 
outside of ALGs/proxies, which break the end-to-end security model and are 
a special problem for other protocols - notably SIP.

   Are your concerns limited to ALGs/proxies?

Mark
At 09:48 AM 4/23/2003 +0200, Magnus Westerlund wrote:
>Hi Tom,
>
>It seems evident that we must start discussing what we want to accomplish 
>with securing the RTSP messages.
>
>Some requirements on what I see as necessary for transport secured RTSP:
>
>+ Message authentication to prevent tampering. In my opinion we must 
>authenticate the whole message to provide any benefit. This will create 
>problems with ALGs and Proxies. The Transport header is a vital header for 
>RTSP functionality and it is desired to have it secured, and at the same 
>time it is the header that an proxy or ALG needs to both read and possibly 
>change.
>
>+ Message privacy, I don't know what parts we actually need to have 
>encrypted. There is several issues here, user privacy, preventing 
>disclosure of information that an attacker could use.
>
>What should the security architecture be? Do we need two levels, one 
>between each entity, i.e. between client and proxy and then proxy to 
>server. Then a second level that is protected end to end, i.e. client <-> 
>server.
>
>I think we should get a security advisor for this. One knowledgeable about 
>these issues would be very good.
>
>Colin and Joerg: Who is our security advisor?
>
>>>If RTSP is focusing now on the client/server model exclusively (per
>>>
>>our
>>>nat debate), I would think that TLS is a better match. Doing the 
>>>equivalent of sips - rtsps - would be much more deployable, since it 
>>>would not require some additional key distribution mechanism.
>>>
>>
>>We may focus on client/server but RTSP does have peer-to-peer uses.
>>
>>Doesn't TLS encrypt all of the data?  I think that is quite undesirable.
>>
>>You lose the ability to use ALGs and existing RTSP proxies.
>>
>As pointed out above, I think we need to think about what we would like to 
>accomplish. We need to review what security we need and what is possible 
>to fit into this. I think we will have to face some hard decisions before 
>the end of this work. Would a proxy really be able to handle such a 
>message you propose. One bug I see is that the proxy needs at least the 
>host part of the request URI to be able to send the request to the correct 
>entity. And if you need trusted proxies, I think one actually needs 
>consider that these proxies must be updated and security audited.
>
>I get the feeling that using TLS is rather simple but requires a limited 
>usage model or trusted proxies. And as proxies can't really be trusted, 
>what do we do?
>
>
>Best Regards
>
>Magnus Westerlund
>Multimedia Technologies, Ericsson Research ERA/TVA/A
>----------------------------------------------------------------------
>Ericsson AB                | Phone +46 8 4048287
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From: Thomas Zeng <zeng@PacketVideo.COM>
To: "'mmusic@ietf.org'" <mmusic@ietf.org>
Date: Thu, 24 Apr 2003 18:20:44 -0700
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Subject: [MMUSIC] RTSP client issues PAUSE while server ready
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Hello, MMUSIC:

During last weeks RTSP teleconf, I took the action to report a tentative
resolution to everyone on MMUSIC mailing list.
This issue is reported in:
http://sourceforge.net/tracker/index.php?func=detail&group_id=23194&atid=377
744&aid=556295
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? 



The tentative agreement is to have RTSP server send back 200 OK in this
case, even though no action is taken by the server.

Are there any objections from anyone?

Thanks,
- thomas
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From: Thomas Zeng <zeng@PacketVideo.COM>
To: mmusic@ietf.org
Subject: RE: [MMUSIC] Secure RTSP thoughts
Date: Thu, 24 Apr 2003 18:28:45 -0700
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Mark:

So far concerns seem to be limited to ALGs/proxies.

Having said that, in the case of HTTP, there is the informative RFC2818
"HTTP Over TLS" which describes how to use HTTP over TLS. See:

I wonder whether similar RFC  on "RTSP over TLS" is needed?

- thomas


-----Original Message-----
From: Mark Baugher [mailto:mbaugher@cisco.com]
Sent: Thursday, April 24, 2003 3:21 PM
To: Magnus Westerlund
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] Secure RTSP thoughts


Magnus
   This reminds me of a topic that you raised at the last meeting: We need 
to define RTSP-over-SSL/TLS.  I don't understand what needs to be defined 
outside of ALGs/proxies, which break the end-to-end security model and are 
a special problem for other protocols - notably SIP.

   Are your concerns limited to ALGs/proxies?

Mark
At 09:48 AM 4/23/2003 +0200, Magnus Westerlund wrote:
>Hi Tom,
>
>It seems evident that we must start discussing what we want to accomplish 
>with securing the RTSP messages.
>
>Some requirements on what I see as necessary for transport secured RTSP:
>
>+ Message authentication to prevent tampering. In my opinion we must 
>authenticate the whole message to provide any benefit. This will create 
>problems with ALGs and Proxies. The Transport header is a vital header for 
>RTSP functionality and it is desired to have it secured, and at the same 
>time it is the header that an proxy or ALG needs to both read and possibly 
>change.
>
>+ Message privacy, I don't know what parts we actually need to have 
>encrypted. There is several issues here, user privacy, preventing 
>disclosure of information that an attacker could use.
>
>What should the security architecture be? Do we need two levels, one 
>between each entity, i.e. between client and proxy and then proxy to 
>server. Then a second level that is protected end to end, i.e. client <-> 
>server.
>
>I think we should get a security advisor for this. One knowledgeable about 
>these issues would be very good.
>
>Colin and Joerg: Who is our security advisor?
>
>>>If RTSP is focusing now on the client/server model exclusively (per
>>>
>>our
>>>nat debate), I would think that TLS is a better match. Doing the 
>>>equivalent of sips - rtsps - would be much more deployable, since it 
>>>would not require some additional key distribution mechanism.
>>>
>>
>>We may focus on client/server but RTSP does have peer-to-peer uses.
>>
>>Doesn't TLS encrypt all of the data?  I think that is quite undesirable.
>>
>>You lose the ability to use ALGs and existing RTSP proxies.
>>
>As pointed out above, I think we need to think about what we would like to 
>accomplish. We need to review what security we need and what is possible 
>to fit into this. I think we will have to face some hard decisions before 
>the end of this work. Would a proxy really be able to handle such a 
>message you propose. One bug I see is that the proxy needs at least the 
>host part of the request URI to be able to send the request to the correct 
>entity. And if you need trusted proxies, I think one actually needs 
>consider that these proxies must be updated and security audited.
>
>I get the feeling that using TLS is rather simple but requires a limited 
>usage model or trusted proxies. And as proxies can't really be trusted, 
>what do we do?
>
>
>Best Regards
>
>Magnus Westerlund
>Multimedia Technologies, Ericsson Research ERA/TVA/A
>----------------------------------------------------------------------
>Ericsson AB                | Phone +46 8 4048287
>Torshamsgatan 23           | Fax   +46 8 7575550
>S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se
>
>
>
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Subject: [MMUSIC] How to identify the updated RTSP specification ?
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Dear MMUSIC Mailing List:

During last week's RTSP teleconference, we had a discussion on how to
distinguish the updated RTSP (i.e., the ongoing work
around draft-ietf-mmusic-rfc2326bis-03.txt, see
http://rtsp.org/2003/drafts/draft03/draft-ietf-mmusic-rfc2326bis-03.pdf).

There is enough change on the table that we feel version number should be
increased to avoid confusion when RTSP server and client might be
implementing the old (rfc2326) and the new RTSP spec (will be a new RFC).

Should we increase the minor version to say RTSP/1.1, or should we up the
major version (to say RTSP/2.0)?

Your comments are welcome.

Thanks,
thomas


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Date: Thu, 24 Apr 2003 20:00:17 -0700
To: Thomas Zeng <zeng@packetvideo.com>
From: Mark Baugher <mbaugher@cisco.com>
Subject: RE: [MMUSIC] Secure RTSP thoughts
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Thomas,

At 06:28 PM 4/24/2003 -0700, Thomas Zeng wrote:
>Mark:
>
>So far concerns seem to be limited to ALGs/proxies.
>
>Having said that, in the case of HTTP, there is the informative RFC2818
>"HTTP Over TLS" which describes how to use HTTP over TLS. See:
>
>I wonder whether similar RFC  on "RTSP over TLS" is needed?

Perhaps an informational or BCP document would be useful or even 
necessary.  But I expect that it would recommend that we don't use TLS/SSL 
to proxies (or whenever the endpoint of the SSL connection isn't coincident 
with the endpoint of the RTSP connection).  Seems to me that the proxy 
security problem is a different barrel of monkeys.

Mark


>- thomas
>
>
>-----Original Message-----
>From: Mark Baugher [mailto:mbaugher@cisco.com]
>Sent: Thursday, April 24, 2003 3:21 PM
>To: Magnus Westerlund
>Cc: mmusic@ietf.org
>Subject: Re: [MMUSIC] Secure RTSP thoughts
>
>
>Magnus
>    This reminds me of a topic that you raised at the last meeting: We need
>to define RTSP-over-SSL/TLS.  I don't understand what needs to be defined
>outside of ALGs/proxies, which break the end-to-end security model and are
>a special problem for other protocols - notably SIP.
>
>    Are your concerns limited to ALGs/proxies?
>
>Mark
>At 09:48 AM 4/23/2003 +0200, Magnus Westerlund wrote:
> >Hi Tom,
> >
> >It seems evident that we must start discussing what we want to accomplish
> >with securing the RTSP messages.
> >
> >Some requirements on what I see as necessary for transport secured RTSP:
> >
> >+ Message authentication to prevent tampering. In my opinion we must
> >authenticate the whole message to provide any benefit. This will create
> >problems with ALGs and Proxies. The Transport header is a vital header for
> >RTSP functionality and it is desired to have it secured, and at the same
> >time it is the header that an proxy or ALG needs to both read and possibly
> >change.
> >
> >+ Message privacy, I don't know what parts we actually need to have
> >encrypted. There is several issues here, user privacy, preventing
> >disclosure of information that an attacker could use.
> >
> >What should the security architecture be? Do we need two levels, one
> >between each entity, i.e. between client and proxy and then proxy to
> >server. Then a second level that is protected end to end, i.e. client <->
> >server.
> >
> >I think we should get a security advisor for this. One knowledgeable about
> >these issues would be very good.
> >
> >Colin and Joerg: Who is our security advisor?
> >
> >>>If RTSP is focusing now on the client/server model exclusively (per
> >>>
> >>our
> >>>nat debate), I would think that TLS is a better match. Doing the
> >>>equivalent of sips - rtsps - would be much more deployable, since it
> >>>would not require some additional key distribution mechanism.
> >>>
> >>
> >>We may focus on client/server but RTSP does have peer-to-peer uses.
> >>
> >>Doesn't TLS encrypt all of the data?  I think that is quite undesirable.
> >>
> >>You lose the ability to use ALGs and existing RTSP proxies.
> >>
> >As pointed out above, I think we need to think about what we would like to
> >accomplish. We need to review what security we need and what is possible
> >to fit into this. I think we will have to face some hard decisions before
> >the end of this work. Would a proxy really be able to handle such a
> >message you propose. One bug I see is that the proxy needs at least the
> >host part of the request URI to be able to send the request to the correct
> >entity. And if you need trusted proxies, I think one actually needs
> >consider that these proxies must be updated and security audited.
> >
> >I get the feeling that using TLS is rather simple but requires a limited
> >usage model or trusted proxies. And as proxies can't really be trusted,
> >what do we do?
> >
> >
> >Best Regards
> >
> >Magnus Westerlund
> >Multimedia Technologies, Ericsson Research ERA/TVA/A
> >----------------------------------------------------------------------
> >Ericsson AB                | Phone +46 8 4048287
> >Torshamsgatan 23           | Fax   +46 8 7575550
> >S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se
> >
> >
> >
> >_______________________________________________
> >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
>_______________________________________________
>mmusic mailing list
>mmusic@ietf.org
>https://www1.ietf.org/mailman/listinfo/mmusic


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Subject: RE: [MMUSIC] Secure RTSP thoughts
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Hi,

 > -----Original Message-----
 > From: ext Mark Baugher [mailto:mbaugher@cisco.com]
 > Sent: 25 April, 2003 06:00
 > To: Thomas Zeng
 > Cc: mmusic@ietf.org
 > Subject: RE: [MMUSIC] Secure RTSP thoughts
 > 
 > 
 > Thomas,
 > 
 > At 06:28 PM 4/24/2003 -0700, Thomas Zeng wrote:
 > >Mark:
 > >
 > >So far concerns seem to be limited to ALGs/proxies.
 > >
 > >Having said that, in the case of HTTP, there is the 
 > informative RFC2818
 > >"HTTP Over TLS" which describes how to use HTTP over TLS. See:
 > >
 > >I wonder whether similar RFC  on "RTSP over TLS" is needed?
 > 
 > Perhaps an informational or BCP document would be useful or even 
 > necessary.  But I expect that it would recommend that we 
 > don't use TLS/SSL 
 > to proxies (or whenever the endpoint of the SSL connection 
 > isn't coincident 
 > with the endpoint of the RTSP connection).  Seems to me that 
 > the proxy 
 > security problem is a different barrel of monkeys.

I think you're right. For HTTP, RFC 2818 solves a separate problem from RFC 2817, which actually describes how TLS can be tunneled through proxies and how a TCP connection can be upgraded to TLS.

Cheers,
Aki

 > 
 > Mark
 > 
 > 
 > >- thomas
 > >
 > >
 > >-----Original Message-----
 > >From: Mark Baugher [mailto:mbaugher@cisco.com]
 > >Sent: Thursday, April 24, 2003 3:21 PM
 > >To: Magnus Westerlund
 > >Cc: mmusic@ietf.org
 > >Subject: Re: [MMUSIC] Secure RTSP thoughts
 > >
 > >
 > >Magnus
 > >    This reminds me of a topic that you raised at the last 
 > meeting: We need
 > >to define RTSP-over-SSL/TLS.  I don't understand what needs 
 > to be defined
 > >outside of ALGs/proxies, which break the end-to-end 
 > security model and are
 > >a special problem for other protocols - notably SIP.
 > >
 > >    Are your concerns limited to ALGs/proxies?
 > >
 > >Mark
 > >At 09:48 AM 4/23/2003 +0200, Magnus Westerlund wrote:
 > > >Hi Tom,
 > > >
 > > >It seems evident that we must start discussing what we 
 > want to accomplish
 > > >with securing the RTSP messages.
 > > >
 > > >Some requirements on what I see as necessary for 
 > transport secured RTSP:
 > > >
 > > >+ Message authentication to prevent tampering. In my 
 > opinion we must
 > > >authenticate the whole message to provide any benefit. 
 > This will create
 > > >problems with ALGs and Proxies. The Transport header is a 
 > vital header for
 > > >RTSP functionality and it is desired to have it secured, 
 > and at the same
 > > >time it is the header that an proxy or ALG needs to both 
 > read and possibly
 > > >change.
 > > >
 > > >+ Message privacy, I don't know what parts we actually 
 > need to have
 > > >encrypted. There is several issues here, user privacy, preventing
 > > >disclosure of information that an attacker could use.
 > > >
 > > >What should the security architecture be? Do we need two 
 > levels, one
 > > >between each entity, i.e. between client and proxy and 
 > then proxy to
 > > >server. Then a second level that is protected end to end, 
 > i.e. client <->
 > > >server.
 > > >
 > > >I think we should get a security advisor for this. One 
 > knowledgeable about
 > > >these issues would be very good.
 > > >
 > > >Colin and Joerg: Who is our security advisor?
 > > >
 > > >>>If RTSP is focusing now on the client/server model 
 > exclusively (per
 > > >>>
 > > >>our
 > > >>>nat debate), I would think that TLS is a better match. Doing the
 > > >>>equivalent of sips - rtsps - would be much more 
 > deployable, since it
 > > >>>would not require some additional key distribution mechanism.
 > > >>>
 > > >>
 > > >>We may focus on client/server but RTSP does have 
 > peer-to-peer uses.
 > > >>
 > > >>Doesn't TLS encrypt all of the data?  I think that is 
 > quite undesirable.
 > > >>
 > > >>You lose the ability to use ALGs and existing RTSP proxies.
 > > >>
 > > >As pointed out above, I think we need to think about what 
 > we would like to
 > > >accomplish. We need to review what security we need and 
 > what is possible
 > > >to fit into this. I think we will have to face some hard 
 > decisions before
 > > >the end of this work. Would a proxy really be able to 
 > handle such a
 > > >message you propose. One bug I see is that the proxy 
 > needs at least the
 > > >host part of the request URI to be able to send the 
 > request to the correct
 > > >entity. And if you need trusted proxies, I think one 
 > actually needs
 > > >consider that these proxies must be updated and security audited.
 > > >
 > > >I get the feeling that using TLS is rather simple but 
 > requires a limited
 > > >usage model or trusted proxies. And as proxies can't 
 > really be trusted,
 > > >what do we do?
 > > >
 > > >
 > > >Best Regards
 > > >
 > > >Magnus Westerlund
 > > >Multimedia Technologies, Ericsson Research ERA/TVA/A
 > > 
 > >------------------------------------------------------------
 > ----------
 > > >Ericsson AB                | Phone +46 8 4048287
 > > >Torshamsgatan 23           | Fax   +46 8 7575550
 > > >S-164 80 Stockholm, Sweden | mailto: 
 > magnus.westerlund@era.ericsson.se
 > > >
 > > >
 > > >
 > > >_______________________________________________
 > > >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
 > >_______________________________________________
 > >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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--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 Protocol Offer Answer Examples
	Author(s)	: A. Johnston, R. Sparks
	Filename	: draft-ietf-mmusic-offer-answer-examples-00.txt
	Pages		: 24
	Date		: 2003-4-24
	
This document gives examples of Session Description Protocol (SDP)
offer answer exchanges.  Examples include codec negotiation and
selection, hold and resume, and addition and deletion of media
streams.  The examples show multiple media types, bidirectional,
unidirectional, inactive streams and dynamic payload types. Common
Third Party Call Control (3pcc) examples are also given.

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To: Thomas Zeng <zeng@PacketVideo.COM>
cc: "'mmusic@ietf.org'" <mmusic@ietf.org>
Subject: Re: [MMUSIC] How to identify the updated RTSP specification ? 
In-Reply-To: Your message of "Thu, 24 Apr 2003 18:39:02 PDT."
             <72263E8E8622D611975C0002B32C19D84A79F3@MISTY> 
Date: Fri, 25 Apr 2003 09:23:09 -0400
From: Colin Perkins <csp@csperkins.org>
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--> Thomas Zeng writes:
>Dear MMUSIC Mailing List:
>
>During last week's RTSP teleconference, we had a discussion on how to
>distinguish the updated RTSP (i.e., the ongoing work
>around draft-ietf-mmusic-rfc2326bis-03.txt, see
>http://rtsp.org/2003/drafts/draft03/draft-ietf-mmusic-rfc2326bis-03.pdf).
>
>There is enough change on the table that we feel version number should be
>increased to avoid confusion when RTSP server and client might be
>implementing the old (rfc2326) and the new RTSP spec (will be a new RFC).

Okay, that makes sense.

>Should we increase the minor version to say RTSP/1.1, or should we up the
>major version (to say RTSP/2.0)?

If implementations of the two specifications can still interwork, albet
with reduced functionality, then I'd go for RTSP/1.1.  

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



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Date: Fri, 25 Apr 2003 09:28:56 -0400
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To: Colin Perkins <csp@csperkins.org>
CC: Thomas Zeng <zeng@PacketVideo.COM>, "'mmusic@ietf.org'" <mmusic@ietf.org>
Subject: Re: [MMUSIC] How to identify the updated RTSP specification ?
References: <200304251323.h3PDN9Mj034446@purple.nge.isi.edu>
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Unless there is a real and significant backwards-compatibility issue, I 
would strongly recommend against this. It gratuitously breaks existing 
implementations and greatly increases the deployment pain and confusion.

Colin Perkins wrote:
> --> Thomas Zeng writes:
> 
>>Dear MMUSIC Mailing List:
>>
>>During last week's RTSP teleconference, we had a discussion on how to
>>distinguish the updated RTSP (i.e., the ongoing work
>>around draft-ietf-mmusic-rfc2326bis-03.txt, see
>>http://rtsp.org/2003/drafts/draft03/draft-ietf-mmusic-rfc2326bis-03.pdf).
>>
>>There is enough change on the table that we feel version number should be
>>increased to avoid confusion when RTSP server and client might be
>>implementing the old (rfc2326) and the new RTSP spec (will be a new RFC).
> 
> 
> Okay, that makes sense.
> 
> 
>>Should we increase the minor version to say RTSP/1.1, or should we up the
>>major version (to say RTSP/2.0)?
> 
> 
> If implementations of the two specifications can still interwork, albet
> with reduced functionality, then I'd go for RTSP/1.1.  
> 
> Colin
> _______________________________________________
> 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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To: Henning Schulzrinne <hgs@cs.columbia.edu>
cc: Thomas Zeng <zeng@PacketVideo.COM>, "'mmusic@ietf.org'" <mmusic@ietf.org>
Subject: Re: [MMUSIC] How to identify the updated RTSP specification ? 
In-Reply-To: Your message of "Fri, 25 Apr 2003 09:28:56 EDT."
             <3EA93818.5080505@cs.columbia.edu> 
Date: Fri, 25 Apr 2003 09:35:50 -0400
From: Colin Perkins <csp@csperkins.org>
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It may be less painful in the long run than a long list of supported
features tags, etc.  

Colin


--> Henning Schulzrinne writes:
>Unless there is a real and significant backwards-compatibility issue, I 
>would strongly recommend against this. It gratuitously breaks existing 
>implementations and greatly increases the deployment pain and confusion.
>
>Colin Perkins wrote:
>> --> Thomas Zeng writes:
>> 
>>>Dear MMUSIC Mailing List:
>>>
>>>During last week's RTSP teleconference, we had a discussion on how to
>>>distinguish the updated RTSP (i.e., the ongoing work
>>>around draft-ietf-mmusic-rfc2326bis-03.txt, see
>>>http://rtsp.org/2003/drafts/draft03/draft-ietf-mmusic-rfc2326bis-03.pdf).
>>>
>>>There is enough change on the table that we feel version number should be
>>>increased to avoid confusion when RTSP server and client might be
>>>implementing the old (rfc2326) and the new RTSP spec (will be a new RFC).
>> 
>> 
>> Okay, that makes sense.
>> 
>> 
>>>Should we increase the minor version to say RTSP/1.1, or should we up the
>>>major version (to say RTSP/2.0)?
>> 
>> 
>> If implementations of the two specifications can still interwork, albet
>> with reduced functionality, then I'd go for RTSP/1.1.  
>> 
>> Colin
>> _______________________________________________
>> 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
_______________________________________________
mmusic mailing list
mmusic@ietf.org
https://www1.ietf.org/mailman/listinfo/mmusic



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Subject: Re: [MMUSIC] How to identify the updated RTSP specification ?
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To make a more informed communications, I would like to have a clear 
list of issues that would interfere with backward-compability.

Colin Perkins wrote:

> It may be less painful in the long run than a long list of supported
> features tags, etc.  


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



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To: Colin Perkins <csp@isi.edu>
From: Rajesh Kumar <rkumar@cisco.com>
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Colin, Steve,
Is the use of multiple 'm=' lines with the same port number within an SDP 
descriptor permitted?

In SDP, as it is defined, all the media formats on an 'm=' line are of the 
same media type (e.g. audio, image, text etc.).

Now consider the case in which two RTP media formats can use a destination 
port but are registered under different MIME types. For example, text/t140 
and audio/PCMU. In this case, there is no recourse except to use multiple 
media lines with the same port number. For example:

m=audio 49232 RTP/AVP 0
m=text 49232 RTP/AVP 100
a=rtpmap:100 t140

My thought is that although this is unsual, it is not proscribed by SDP 
syntax in RFC 2327 or in its latest internet draft.

Thanks,

Rajesh


--=====================_1614110297==_.ALT
Content-Type: text/html; charset="us-ascii"

<html>
<font size=3>Colin, Steve,<br>
Is the use of multiple 'm=' lines with the same port number within an SDP
descriptor permitted?<br>
<br>
In SDP, as it is defined, all the media formats on an 'm=' line are of
the same media type (e.g. audio, image, text etc.).<br>
<br>
Now consider the case in which two RTP media formats can use a
destination port but are registered under different MIME types. For
example, text/t140 and audio/PCMU. In this case, there is no recourse
except to use multiple media lines with the same port number. For
example:<br>
<br>
m=audio 49232 RTP/AVP 0<br>
m=text 49232 RTP/AVP 100<br>
a=rtpmap:100 t140<br>
<br>
My thought is that although this is unsual, it is not proscribed by SDP
syntax in RFC 2327 or in its latest internet draft.<br>
<br>
Thanks,<br>
<br>
Rajesh<br>
<br>
</font></html>
--=====================_1614110297==_.ALT--

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From: Stephen Casner <casner@acm.org>
To: Rajesh Kumar <rkumar@cisco.com>
cc: Colin Perkins <csp@isi.edu>, <mmusic@ietf.org>,
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Rajesh,

> Is the use of multiple 'm=' lines with the same port number within an SDP
> descriptor permitted?

Presuming that the multiple m= lines are all RTP/AVP, then doing so
causes all of the payload formats listed on the m= lines to become
part of a single RTP session (unless they are distinguished by some
other aspect of addressing).  Section 5.2 of the RTP spec addresses
NOT multiplexing different media in the same RTP session.

> In SDP, as it is defined, all the media formats on an 'm=' line are of the
> same media type (e.g. audio, image, text etc.).
>
> Now consider the case in which two RTP media formats can use a destination
> port but are registered under different MIME types. For example, text/t140
> and audio/PCMU. In this case, there is no recourse except to use multiple
> media lines with the same port number. For example:

The "can use a destination port" is the tricky part.  We may not agree.

> m=audio 49232 RTP/AVP 0
> m=text 49232 RTP/AVP 100
> a=rtpmap:100 t140
>
> My thought is that although this is unsual, it is not proscribed by SDP
> syntax in RFC 2327 or in its latest internet draft.

I believe this is proscribed by section 5.2 of the RTP spec because
these two media should not be multiplexed in one RTP session.

My bet is that the real example you are thinking about is fax or modem
over IP.  We've been there before.

                                                        -- Steve

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To: Stephen Casner <casner@acm.org>
From: Rajesh Kumar <rkumar@cisco.com>
Subject: Re: [MMUSIC] Re: Multiple 'm' lines with the same port number
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--=====================_1619836802==_.ALT
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Steve,
The real examples I am thinking of are all RTP-encapsulated payloads such 
as text/t140 and the new t38-facsimile-over-RTP work that I am involved in 
with Mario Garakani and Paul Jones and others. In all these cases, the 
payload is encaspulated in RTP and uses the RTP/AVP profile. So this is not 
a repeat of an old issue.

So if facsimile were encapsulated in RTP, you would need a new MIME 
audio/t38 to multiplex it with PCMU into the same UDP port? The current 
MIME for T.38 facsimile over IP is image/t38. Or does the standard 
proscribe the multiplexing into a single port of  PCMU and 
RTP-encapsulate-facsimile altogether? I do not think so.

Thanks,

Rajesh

At 11:57 AM 4/25/2003 -0700, Stephen Casner wrote:
>Rajesh,
>
> > Is the use of multiple 'm=' lines with the same port number within an SDP
> > descriptor permitted?
>
>Presuming that the multiple m= lines are all RTP/AVP, then doing so
>causes all of the payload formats listed on the m= lines to become
>part of a single RTP session (unless they are distinguished by some
>other aspect of addressing).  Section 5.2 of the RTP spec addresses
>NOT multiplexing different media in the same RTP session.
>
> > In SDP, as it is defined, all the media formats on an 'm=' line are of the
> > same media type (e.g. audio, image, text etc.).
> >
> > Now consider the case in which two RTP media formats can use a destination
> > port but are registered under different MIME types. For example, text/t140
> > and audio/PCMU. In this case, there is no recourse except to use multiple
> > media lines with the same port number. For example:
>
>The "can use a destination port" is the tricky part.  We may not agree.
>
> > m=audio 49232 RTP/AVP 0
> > m=text 49232 RTP/AVP 100
> > a=rtpmap:100 t140
> >
> > My thought is that although this is unsual, it is not proscribed by SDP
> > syntax in RFC 2327 or in its latest internet draft.
>
>I believe this is proscribed by section 5.2 of the RTP spec because
>these two media should not be multiplexed in one RTP session.
>
>My bet is that the real example you are thinking about is fax or modem
>over IP.  We've been there before.
>
>                                                         -- Steve
>
>_______________________________________________
>mmusic mailing list
>mmusic@ietf.org
>https://www1.ietf.org/mailman/listinfo/mmusic

--=====================_1619836802==_.ALT
Content-Type: text/html; charset="us-ascii"

<html>
<font size=3>Steve,<br>
The real examples I am thinking of are all RTP-encapsulated payloads such
as text/t140 and the new t38-facsimile-over-RTP work that I am involved
in with Mario Garakani and Paul Jones and others. In all these cases, the
payload is encaspulated in RTP and uses the RTP/AVP profile. So this is
not a repeat of an old issue.&nbsp; <br>
<br>
So if facsimile were encapsulated in RTP, you would need a new MIME
audio/t38 to multiplex it with PCMU into the same UDP port? The current
MIME for T.38 facsimile over IP is image/t38. Or does the standard
proscribe the multiplexing into a single port of&nbsp; PCMU and
RTP-encapsulate-facsimile altogether? I do not think so.<br>
<br>
Thanks,<br>
<br>
Rajesh<br>
<br>
At 11:57 AM 4/25/2003 -0700, Stephen Casner wrote:<br>
<blockquote type=cite cite>Rajesh,<br>
<br>
&gt; Is the use of multiple 'm=' lines with the same port number within
an SDP<br>
&gt; descriptor permitted?<br>
<br>
Presuming that the multiple m= lines are all RTP/AVP, then doing so<br>
causes all of the payload formats listed on the m= lines to become<br>
part of a single RTP session (unless they are distinguished by some<br>
other aspect of addressing).&nbsp; Section 5.2 of the RTP spec
addresses<br>
NOT multiplexing different media in the same RTP session.<br>
<br>
&gt; In SDP, as it is defined, all the media formats on an 'm=' line are
of the<br>
&gt; same media type (e.g. audio, image, text etc.).<br>
&gt;<br>
&gt; Now consider the case in which two RTP media formats can use a
destination<br>
&gt; port but are registered under different MIME types. For example,
text/t140<br>
&gt; and audio/PCMU. In this case, there is no recourse except to use
multiple<br>
&gt; media lines with the same port number. For example:<br>
<br>
The &quot;can use a destination port&quot; is the tricky part.&nbsp; We
may not agree.<br>
<br>
&gt; m=audio 49232 RTP/AVP 0<br>
&gt; m=text 49232 RTP/AVP 100<br>
&gt; a=rtpmap:100 t140<br>
&gt;<br>
&gt; My thought is that although this is unsual, it is not proscribed by
SDP<br>
&gt; syntax in RFC 2327 or in its latest internet draft.<br>
<br>
I believe this is proscribed by section 5.2 of the RTP spec because<br>
these two media should not be multiplexed in one RTP session.<br>
<br>
My bet is that the real example you are thinking about is fax or
modem<br>
over IP.&nbsp; We've been there before.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
-- Steve<br>
<br>
_______________________________________________<br>
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mmusic@ietf.org<br>
<a href="https://www1.ietf.org/mailman/listinfo/mmusic" eudora="autourl">https://www1.ietf.org/mailman/listinfo/mmusic</a>
</font></blockquote></html>

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To: Stephen Casner <casner@acm.org>
CC: Rajesh Kumar <rkumar@cisco.com>, Colin Perkins <csp@isi.edu>,
        mmusic@ietf.org, mmostafa@cisco.com, hisham@cisco.com,
        mgarakan@cisco.com, paulej@cisco.com
Subject: Re: [MMUSIC] Re: Multiple 'm' lines with the same port number
References: <20030425114849.X77830-100000@ash.packetdesign.com>
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Stephen Casner wrote:

> Rajesh,
>
> > Is the use of multiple 'm=' lines with the same port number within an SDP
> > descriptor permitted?
>
> Presuming that the multiple m= lines are all RTP/AVP, then doing so
> causes all of the payload formats listed on the m= lines to become
> part of a single RTP session (unless they are distinguished by some
> other aspect of addressing).  Section 5.2 of the RTP spec addresses
> NOT multiplexing different media in the same RTP session.
>
> > In SDP, as it is defined, all the media formats on an 'm=' line are of the
> > same media type (e.g. audio, image, text etc.).
> >
> > Now consider the case in which two RTP media formats can use a destination
> > port but are registered under different MIME types. For example, text/t140
> > and audio/PCMU. In this case, there is no recourse except to use multiple
> > media lines with the same port number. For example:
>
> The "can use a destination port" is the tricky part.  We may not agree.
>

We also seem to be entering a bit of a gray area here in terms of offer/answer
(RFC 3264). It's not clear that the definition of "m=" lines and media streams
allow for something like this; for sure the parties would have to recognize that
the two "m=" lines in fact are one media stream and hence should not be sending
to "both" at the same time.  However, I'm not aware of such behavior being
specified anywhere.

-- Flemming


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Subject: Re: [MMUSIC] How to identify the updated RTSP specification ?
From: Rob Lanphier <robla@real.com>
To: Henning Schulzrinne <hgs@cs.columbia.edu>
Cc: Colin Perkins <csp@csperkins.org>, Thomas Zeng <zeng@PacketVideo.COM>,
        "'mmusic@ietf.org'" <mmusic@ietf.org>
In-Reply-To: <3EA93A93.4030906@cs.columbia.edu>
References: <200304251335.h3PDZoMj034767@purple.nge.isi.edu>
	 <3EA93A93.4030906@cs.columbia.edu>
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Read the draft.  You are a co-author after all  ;-)

Rob


On Fri, 2003-04-25 at 06:39, Henning Schulzrinne wrote:
> To make a more informed communications, I would like to have a clear 
> list of issues that would interfere with backward-compability.
> 
> Colin Perkins wrote:
> 
> > It may be less painful in the long run than a long list of supported
> > features tags, etc.  
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
-- 
Rob Lanphier, Helix Community Coordinator - RealNetworks
http://helixcommunity.org http://rtsp.org http://realnetworks.com

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From: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
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To: Henning Schulzrinne <hgs@cs.columbia.edu>
CC: "'mmusic@ietf.org'" <mmusic@ietf.org>
Subject: Re: [MMUSIC] How to identify the updated RTSP specification ?
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Hi Henning,

The major incompatibilities are:

1. State machine change. A old client that want to play a media again 
after reaching the end of the media may not be able to restart.

2. Transport header changes:
    - Defined extensible
    - New source and destination header types that handle lists and IPv6 
addresses
 
Plus all the small things that really are quite a number of. Depending 
on the use case you might be able to avoid most of them.

The three options available to us in this question are:

1. Stay with version 1.0 and use feature tags to allow clients and 
server to know that the other party supports the functionality. Feature 
tags will still be need within a version to detect functionality support.

2. Go to version 1.1, i.e. a minor version upgrade. By looking into RFC 
2616 and RFC 2145 the accepted changes shall not effect the general 
parsing mechanism. And from RFC 2145 the following quote can be made:

       It is, and has always been, the explicit intent of the
      HTTP specification that the interpretation of an HTTP message
      header does not change between minor versions of the same major
      version.

Another interesting quote is:

   One consequence of these rules is that an HTTP/1.1 message sent to an
   HTTP/1.0 recipient (or a recipient whose version is unknown) MUST be
   constructed so that it remains a valid HTTP/1.0 message when all
   headers not defined in the HTTP/1.0 specification [1] are removed.


3. Change the major version number, i.e. version 2.0.

The big questions to consider is:
- What effect on deployment will a change of the version number?
- What is acceptable from the protocol side?

To the second question the quotes seems to indicate that we would be 
forced to a version 2.0 for what we have done. We have after all changed 
a rather large number of header syntax's. I don't know it this is a good 
thing for deployment.

More input into this question is encouraged.

Best Regards

Magnus


Henning Schulzrinne wrote:

> To make a more informed communications, I would like to have a clear 
> list of issues that would interfere with backward-compability.
>
> Colin Perkins wrote:
>
>> It may be less painful in the long run than a long list of supported
>> features tags, etc.  
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
>

-- 

Magnus Westerlund 

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



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From: Aravind Narasimhan <aravind.narasimhan@sun.com>
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Subject: [MMUSIC] Clarification of Re-Setup handling in RTSP
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Hey Folks,

I have come up with the proposed wording for the resolution of bug 
513737 (http://rtsp.org/bug513737).  First let me cite the resolution 
that was reached in conference call on March 5th:

Resolution: go with alternative #1 from bug report (1. Keep the old 
setting and states as if the second SETUP never happend.)

Here my attempt to clarify the wording in the spec according to the 
resolution of the concall:

Wording from draft-ietf-mmusic-rfc2326bis-03.txt

11.3 SETUP

   The SETUP request for a URI specifies the transport mechanism to be
   used for the streamed media. A client can 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).

   A server MAY allow a client to do SETUP while in playing state to add
   additional media streams. If not supported the server shall responde
   with error 455 (Method Not Allowed In This State). If supported the
   added media shall then start to play in sync with the already playing
   media. To be able to sync the media with the already playing streams
   the SETUP response MUST include a RTP-Info header with the timestamp
   value, and a Range header with the corresponding normal play time. To
   indicate support for this optional feature the feature-tag:
   "setup.playing" is defined.


New proposed wording:

11.3 SETUP

   The SETUP request for a URI specifies the transport mechanism to be
   used for the streamed media. A client can 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).

   A server MAY allow a client to do SETUP while in playing state to add
   additional media streams. If not supported, the server shall respond    |
   with error 455 (Method Not Valid in This State). If supported the       |
   added media shall then start to play in sync with the already playing
   media. To be able to sync the media with the already playing streams
   the SETUP response MUST include a RTP-Info header with the timestamp
   value, and a Range header with the corresponding normal play time. To
   indicate support for this optional feature the feature-tag:
   "setup.playing" is defined.

   If a SETUP request on a session fails for any reason, the session       |
   state, as well as transport and other parameters for associated         |
   streams will remain unchanged from their values as if the SETUP         |
   request had never been received by the server.                          |

Please send your comments.

Thanks

Aravind

 


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Date: Mon, 28 Apr 2003 11:56:50 -0400
From: Aravind Narasimhan <aravind.narasimhan@sun.com>
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Subject: Re: [MMUSIC] Backwards range and negative scale clarification (bugs
 533691, 513880)
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Hey Magnus,
Thanks for the excellent summary of our concall. 

Another potential way to resolve the situation with the closed/openeness 
of ranges headers that we haven't discussed before is to include another 
field, "type=" (similar to the "time=" field) in the range header which 
explicitly defines the exact nature of the range header.  For example:

PLAY rtsp://foo RTSP/1.0
Session: bar
Range: ntp=5-10; type=1

where we have a predefined set of types:

    0 = default RFC 2326 behavior
    1 = closed on both ends
    2 = open on both ends
    3 = start open, end closed

Obviously, we dont need to specify more types than absolutely necessary.

Now if the type is not specified in a play request, we can specify that 
the default behavior as specified in RFC 2326 with halfopen interval 
will be followed.  However, for more precise handling of the Range, the 
type would have to be specified.. .  

I think this is a good compromise without having to make corner cases as 
is option A below, or changing the semantic of Range completely as in 
option C below.

more comments inline:

Magnus Westerlund wrote:

> Hi,
>
> This was discussed at the last telecon. There was no conclusions on 
> how to solve it. There where three proposals. I also give some of my 
> personal views on the matter.
>
> A. Make an exception at 0. So that any half open end point that 
> includes 0 actually is closed at 0. The only real problem with having 
> a open end interval when playing backwards is the end point a 0, as we 
> can't give negative numbers in npt scale. One can of course leave the 
> interval without a second value. However if a server in a response 
> would like to tell that it is playing to 0 it is still not good.
>
>
> B. Make the range header interpretation depend on scale header sign. 
> For negative scale the range header would go from a open to a closed 
> point. Example Range: Scale=-1 npt=10-5 would mean: play from just 
> before 10 to and including 5. However I see some problem with trying 
> to start playback with an open interval. Where is the differnece 
> between a point and very, very close to it. Also how does one handle 
> RTP-Info if the synchronization point is not defined.

I dont think that the RTP-Info causes a problem.  The RTP-Info is 
returned for the actual range this is going to be played back.  This 
range needs to go in PLAY response and that RTP-Info will then 
correspond to that range start time.

> C. Make the Range header interval closed in both ends. The argument 
> was that we don't need the distinction provided by an half open 
> interval and by closing it in both ends one forces the server to play 
> the whole given interval and the client trims any excess. There are 
> some backwards considerations in this case.
>
>
> To the second issue: I think there is reasons to play media with real 
> time scale backwards. My favourite example is the Surveilance video 
> storage server. This media is of course indexed using real world 
> clock. So if you desire to view the surveilance tape backwards then 
> you need it.

Ok I can kind of buy this argument.  However, I think it is important to 
call this out in the spec. or in an appendix to the spec, because I'm 
sure it will be many people's first assumption to assume that time 
cannot travel backwards.

>
> Any more comments?
>
> Best  Regards
>
> Magnus
>
>
> Aravind Narasimhan wrote:
>
>> Hi Folks,    In my attempt to clarify the negative scale and 
>> backwards range issues brought up by bugs http://rtsp.org/bug533691 
>> &  http://rtsp.org/bug513880 some issues surfaced with regards to the 
>> current definition of ranges in the spec.  I wanted to discuss these 
>> issues before sending off my write up.
>>
>> Issue 1:
>> ---------
>> Range is defined in RTSP as a half open interval.  To quote from 
>> draft-03
>>
>> Ranges are half-open intervals, including the lower point, but
>> excluding the upper point. In other words, a range of a-b starts
>> exactly at time a, but stops just before b.
>>
>> Now the resolution that we came up with for this issue was that we 
>> would always go from the first number in the range to the last number 
>> in the range.  So for example, if the range of a PLAY request was
>>     Range: npt=10-15
>> we would go from 10 -> 15, where the play would stop just before 15. 
>>  If the request had  range and scale headers as the following:
>>     Scale: -1
>>     Range: npt=15-10
>> the play would go backwards and we would go from 15 -> 10.
>>
>> Now the question is how do we interpret the half-open rule.  I.e.  Do 
>> we stop at 10 or just before we reach 10 (i.e. npt > 10) in the 
>> backwards scenario above? The latter seems like they way to interpret 
>> the half - open rule.  But if we interpret it that way, then how do 
>> we deal with the following:
>>     Scale: -1
>>     Range: npt=15-0
>> We would have to stop before 0 and there would be no way to get to 0 
>> since you cannot specify negative ranges.  Obviously the way to get 
>> around this is to do the following:
>>     Scale: -1
>>     Range: npt=15-
>> But this doesn't seem too clean to me.
>>
>> The other option is to say that  backwards ranges are always closed 
>> ranges.  This, however, makes ranges a bit more inconsisted and 
>> harder to deal with.
>>
>> Issue 2:
>> ----------
>> The other thing that doesn't make sense with backwards ranges are the 
>> Clock ranges.  For example:
>>     Scale: -1
>>     Range: clock=20030416T000000.00Z-20030101T000000.00Z
>> makes no sense because clocks cant go backwards.
>>
>> So I propose that Scale can only be negative for ranges that deal 
>> with relative time rather than absolute time.
>>
>>
>


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Subject: [MMUSIC] RFC 3524 on Mapping of Media Streams to Resource Reservation Flows
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--NextPart


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


        RFC 3324

        Title:      Mapping of Media Streams to Resource Reservation
                    Flows
        Author(s):  G. Camarillo, A. Monrad
        Status:     Standards Track
        Date:       April 2003
        Mailbox:    Gonzalo.Camarillo@ericsson.com,
                    atle.monrad@ericsson.com
        Pages:      6
        Characters: 11249
        Updates/Obsoletes/SeeAlso:  None

        I-D Tag:    draft-ietf-mmusic-reservation-flows-01.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3524.txt


This document defines an extension to the Session Description Protocol
(SDP) grouping framework.  It allows requesting a group of media
streams to be mapped into a single resource reservation flow.  The SDP
syntax needed is defined, as well as a new "semantics" attribute
called Single Reservation Flow (SRF).

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

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

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Date: Mon, 28 Apr 2003 19:10:53 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Subject: Re: [MMUSIC] Re: Multiple 'm' lines with the same port number
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inline.

Flemming Andreasen wrote:
> 
> Stephen Casner wrote:
> 
> 
>>Rajesh,
>>
>>
>>>Is the use of multiple 'm=' lines with the same port number within an SDP
>>>descriptor permitted?
>>
>>Presuming that the multiple m= lines are all RTP/AVP, then doing so
>>causes all of the payload formats listed on the m= lines to become
>>part of a single RTP session (unless they are distinguished by some
>>other aspect of addressing).  Section 5.2 of the RTP spec addresses
>>NOT multiplexing different media in the same RTP session.
>>
>>
>>>In SDP, as it is defined, all the media formats on an 'm=' line are of the
>>>same media type (e.g. audio, image, text etc.).
>>>
>>>Now consider the case in which two RTP media formats can use a destination
>>>port but are registered under different MIME types. For example, text/t140
>>>and audio/PCMU. In this case, there is no recourse except to use multiple
>>>media lines with the same port number. For example:
>>
>>The "can use a destination port" is the tricky part.  We may not agree.
>>
> 
> 
> We also seem to be entering a bit of a gray area here in terms of offer/answer
> (RFC 3264). It's not clear that the definition of "m=" lines and media streams
> allow for something like this; for sure the parties would have to recognize that
> the two "m=" lines in fact are one media stream and hence should not be sending
> to "both" at the same time.  However, I'm not aware of such behavior being
> specified anywhere.

Using two m lines to represent alternate encodings for a stream is not 
consistent with the offer/answer model, IMHO. RFC 3264 has mechanisms 
for transitioning from audio to fax, for example, but these require a 
re-invite to do:

8.3.3 Changing Media Types

    The media type (audio, video, etc.) for a stream MAY be changed.  It
    is RECOMMENDED that the media type be changed (as opposed to adding a
    new stream), when the same logical data is being conveyed, but just
    in a different media format.  This is particularly useful for
    changing between voiceband fax and fax in a single stream, which are
    both separate media types.  To do this, the offerer creates a new
    media description, with a new media type, in place of the description
    in the previous SDP which is to be changed.


-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Scientist                             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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--NextPart


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


        RFC 3524

        Title:      Mapping of Media Streams to Resource Reservation
                    Flows
        Author(s):  G. Camarillo, A. Monrad
        Status:     Standards Track
        Date:       April 2003
        Mailbox:    Gonzalo.Camarillo@ericsson.com,
                    atle.monrad@ericsson.com
        Pages:      6
        Characters: 11249
        Updates/Obsoletes/SeeAlso:  None

        I-D Tag:    draft-ietf-mmusic-reservation-flows-01.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3524.txt


This document defines an extension to the Session Description Protocol
(SDP) grouping framework.  It allows requesting a group of media
streams to be mapped into a single resource reservation flow.  The SDP
syntax needed is defined, as well as a new "semantics" attribute
called Single Reservation Flow (SRF).

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
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USC/Information Sciences Institute

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From: Aaron Colwell <acolwell@real.com>
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I have a few questions about section 3.2.3 list item 8. It says
"For RTCP it starts transmitting RTCP packets according to the rules."

1. Does this mean that it starts sending  compound SR and SDES packets?
2. Is the client supposed to created it's own CNAME for this stream of
   packets or does the CNAME need to be sent from the server to the client?
3. Should the client and server both send RR info to eachother about the
   RTP packets they are sending?
4. Are there any requirements for the RTP packet timestamps other than
   being monotonically increasing?

Aaron
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To: Aaron Colwell <acolwell@real.com>, IETF MMUSIC WG <mmusic@ietf.org>
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Hi Aaron,

See my answer below inline.

Aaron Colwell wrote:

 >I have a few questions about section 3.2.3 list item 8. It says
 >"For RTCP it starts transmitting RTCP packets according to the rules."
 >
 >1. Does this mean that it starts sending  compound SR and SDES packets?
 >

Depends on the RTP session state. Unless the server has not sent any RTP
packets yet it must send RR and SDES. If it has sent RTP packets then it
need to send SR and SDES.

 >2. Is the client supposed to created it's own CNAME for this stream of
 >   packets or does the CNAME need to be sent from the server to the 
client?
 >
The client must set a CNAME of its own. That it must always do when
sending RTCP in a RTP session.

 >3. Should the client and server both send RR info to eachother about the
 >   RTP packets they are sending?
 >

Yes, It should report on these packets.

 >4. Are there any requirements for the RTP packet timestamps other than
 >   being monotonically increasing?
 >

No, I need to think over what would be the best strategy here. One would
be to use the RTP send time as TS basis.


In general I think we need to further look over this section. Is it a
good thing to use PT=0, i.e. PCM audio with no samples for these
packets? Please comment.

Best Regards

Magnus Westerlund

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




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To: "'Magnus Westerlund'" <magnus.westerlund@era.ericsson.se>,
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Subject: RE: [MMUSIC] draft-ietf-mmusic-rtsp-nat-00.txt questions
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>  -----Original Message-----
> From: 	Magnus Westerlund [mailto:magnus.westerlund@era.ericsson.se]
> 
> Sent:	Tuesday, April 29, 2003 11:04 AM
> To:	Aaron Colwell; IETF MMUSIC WG
> Subject:	Re: [MMUSIC] draft-ietf-mmusic-rtsp-nat-00.txt questions
> 
> Hi Aaron,
> 
> See my answer below inline.
> 
> Aaron Colwell wrote:
> 
>  >I have a few questions about section 3.2.3 list item 8. It says
>  >"For RTCP it starts transmitting RTCP packets according to the rules."
>  >
>  >1. Does this mean that it starts sending  compound SR and SDES packets?
>  >
> 
> Depends on the RTP session state. Unless the server has not sent any RTP
> packets yet it must send RR and SDES. If it has sent RTP packets then it
> need to send SR and SDES.
> 
>  >2. Is the client supposed to created it's own CNAME for this stream of
>  >   packets or does the CNAME need to be sent from the server to the 
> client?
>  >
> The client must set a CNAME of its own. That it must always do when
> sending RTCP in a RTP session.
> 
>  >3. Should the client and server both send RR info to eachother about the
>  >   RTP packets they are sending?
>  >
> 
> Yes, It should report on these packets.
> 
>  >4. Are there any requirements for the RTP packet timestamps other than
>  >   being monotonically increasing?
>  >
> 
> No, I need to think over what would be the best strategy here. One would
> be to use the RTP send time as TS basis.
	[Thomas Zeng]  This is the reason that makes it hard to treat
ip/port binding packets as RTP packets. These binding packets are not really
media packets, yet according to RTP rules, they would have to be accounted
for in RR and SRs. All of a sudden RTSP clients would now need to send SRs
back to  RTSP servers. That probably won't please implementors of thin RTSP
clients.


> In general I think we need to further look over this section. Is it a
> good thing to use PT=0, i.e. PCM audio with no samples for these
> packets? Please comment.
	[Thomas Zeng]  How about not treating binding packets as RTP packets
at all? My suggestion would be to use a simple UDP packet that contains an
SSRC (the server SSRC as indicated in SETUP response, so as to  make it
possible to uniquely identify RTP session in the event server is sending RTP
packets from a shared UDP port in order to conserve server resources -- a
common practice among implementors of large scale servers) ? If you want to
indicate that re-transmission of binding packets are taking place, we can
put a 4- byte counter in this binding packets as well. That is, the first
binding packet is try number 1, the second is try number 2, etc...  The fact
that this binding packet is smaller than RTP / RTCP header makes it real
easy for the server to distinguish between RTP and RTCP packets and the port
binding packets. Any other packets would be disgarded. A binding packet with
incorrect SSRC or counter are also discarded.

> Best Regards
> 
> Magnus Westerlund
> 
> Multimedia Technologies, Ericsson Research ERA/TVA/A
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se
> 
> 
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
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From: Tom Marshall <tmarshall@real.com>
To: Thomas Zeng <zeng@packetvideo.com>
Cc: "'Magnus Westerlund'" <magnus.westerlund@era.ericsson.se>,
        Aaron Colwell <acolwell@real.com>, IETF MMUSIC WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] draft-ietf-mmusic-rtsp-nat-00.txt questions
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> > No, I need to think over what would be the best strategy here. One would
> > be to use the RTP send time as TS basis.
> 	[Thomas Zeng]  This is the reason that makes it hard to treat
> ip/port binding packets as RTP packets. These binding packets are not rea=
lly
> media packets, yet according to RTP rules, they would have to be accounted
> for in RR and SRs. All of a sudden RTSP clients would now need to send SRs
> back to  RTSP servers. That probably won't please implementors of thin RT=
SP
> clients.
>=20
>=20
> > In general I think we need to further look over this section. Is it a
> > good thing to use PT=3D0, i.e. PCM audio with no samples for these
> > packets? Please comment.
> 	[Thomas Zeng]  How about not treating binding packets as RTP packets
> at all? My suggestion would be to use a simple UDP packet that contains an
> SSRC (the server SSRC as indicated in SETUP response, so as to  make it
> possible to uniquely identify RTP session in the event server is sending =
RTP
> packets from a shared UDP port in order to conserve server resources -- a
> common practice among implementors of large scale servers) ? If you want =
to
> indicate that re-transmission of binding packets are taking place, we can
> put a 4- byte counter in this binding packets as well. That is, the first
> binding packet is try number 1, the second is try number 2, etc...  The f=
act
> that this binding packet is smaller than RTP / RTCP header makes it real
> easy for the server to distinguish between RTP and RTCP packets and the p=
ort
> binding packets. Any other packets would be disgarded. A binding packet w=
ith
> incorrect SSRC or counter are also discarded.

I have mixed feelings on this issue.  On the one hand, it isn't really an
RTP data packet.  It's just a binding packet that happens to require a
couple of the same values that are in an RTP packet.  On the other hand, the
RTP header is very well defined, it's pretty short (12 octets), and the PT
field is really the only troublesome one.  I think the other fields have
pretty obvious values.

Why can't we say that the binding packet is not an RTP packet but it is
formatted to look like one?  There may be advantages to masquerading as an
RTP packet, especially with firewalls.  We can choose some arbitrary value
for PT (I think 0 or 127 is logical) and we get the version, extension
mechanism, and sequence number for free.  What a deal.

--=20
Mountain Dew and doughnuts...  because breakfast is the most important meal
of the day.

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Hi Aaron

I hope you don't mind that I again include the MMUSIC list. I think the 
question and replies should go there.

Aaron Colwell wrote:

>Hi Magnus,
>
>Thanks for you response. My comments are inline
>
>On Tue, 29 Apr 2003, Magnus Westerlund wrote:
>
>  
>
>>Hi Aaron,
>>
>>See my answer below inline.
>>
>>Aaron Colwell wrote:
>>
>>    
>>
>>>I have a few questions about section 3.2.3 list item 8. It says
>>>"For RTCP it starts transmitting RTCP packets according to the rules."
>>>
>>>1. Does this mean that it starts sending  compound SR and SDES packets?
>>>
>>>      
>>>
>>Depends on the RTP session state. Unless the server has not sent any RTP
>>packets yet it must send RR and SDES. If it has sent RTP packets then it
>>need to send SR and SDES.
>>    
>>
>
>I guess I was assuming that if the client is behind a NAT that it won't
>get any packets from the server until it sends packets to the server
>through the NAT. In this case it seems to me that the only valid thing
>for the client to do is send a compound SR and SDES packets initially.
>  
>

Yes, If the client is behind a NAT it will require that the client sends 
its RTCP packets to the server to bind them. This packet shall be a 
normal RTCP packet. So if RTCP is sent before any RTP packet has been 
sent, then the client SSRC has status as receiver only and sends RR and 
SDES. If the RTP packet(s) has been sent then it will have status of 
active sender and send SR and SDES. This doesn't matter for the binding 
mechanism. As this is based on SSRC the server shall be able to send 
replies as soon as one of the client RTCP packets reach the server.

>  
>
>>>2. Is the client supposed to created it's own CNAME for this stream of
>>>  packets or does the CNAME need to be sent from the server to the client?
>>>
>>>      
>>>
>>The client must set a CNAME of its own. That it must always do when
>>sending RTCP in a RTP session.
>>    
>>
>
>This comment is actually a surprise to me. I always thought that the CNAME
>was always specified by an RTP sender. I've always thought that the client
>just used the CNAME in the SDES packets sent by the server. Is that not
>the case? Perhaps this is something that needs to be clarified in the RTP
>spec. It doesn't seem to say much about who generates the CNAMEs and which
>CNAMEs are used where. It just says that all compound RTCP packets must
>have an SDES packet with CNAME field.
>

All RTP session participants MUST have its own CNAME independent if he 
is a active RTP sender or not. I think you should re-read section 6.5.1 
of draft-ietf-avt-rtp-new-12.txt. I think this is rather clear there. 
The perhaps only somewhat confusing think is the use of the word source. 
It is in some cases meaning participant due to the fact that all 
participant get a SSRC

>
>  
>
>>>3. Should the client and server both send RR info to eachother about the
>>>  RTP packets they are sending?
>>>
>>>      
>>>
>>Yes, It should report on these packets.
>>
>>    
>>
>>>4. Are there any requirements for the RTP packet timestamps other than
>>>  being monotonically increasing?
>>>
>>>      
>>>
>>No, I need to think over what would be the best strategy here. One would
>>be to use the RTP send time as TS basis.
>>    
>>
>
>I like the idea of using the RTP send time here. Since we are using PT=0
>are the units still in 1/8000 of a second?
>  
>

Yes.

>  
>
>>In general I think we need to further look over this section. Is it a
>>good thing to use PT=0, i.e. PCM audio with no samples for these
>>packets? Please comment.
>>
>>    
>>
>
>I don't know if using PT=0 is a good idea or not. I'm struggling a little
>bit with what it means to get a packet without any payload data. In the
>case of PCM audio it doesn't seem to really matter, but in other payload
>formats it might. We might want to have a reserved payload type for this
>type of packet. That way we could also include some sort of challenge like
>you mention later in the document. Whatever challenge mechanism we devise,
>we should have something similar for RTCP as well. In the RTCP case we
>could probably just add a new SDES item type and do the challenge in
>there.
>  
>

Yes, this is something that needs more considerations. I think Thomas 
has some other ideas on how to accomplish this. I would also like a 
binding payload format with its own static payload type. However I don't 
know what possibilities exist to get such developed in a reasonable time 
frame.

>I agree that we should look over this section more. A few weeks ago I had
>to solve this NAT problem for streaming to a mobile phone. I didn't
>really have any specs to reference so I made up my own solution. I like
>this solution much better and I'd like to see this developed more so that
>we'll have an interoperable solution.
>  
>

Yes, there is a total lack of specified solutions for RTSP. The RTSP NAT draft is our attempt to solve this. 


Best Regards

Magnus Westerlund 

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



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Subject: RE: [MMUSIC] draft-ietf-mmusic-rtsp-nat-00.txt questions
To: Thomas Zeng <zeng@packetvideo.com>
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all,

This seems to be getting pretty complicated and tricky....

Actually in the ligth of this discussion the usage of STUN looks
so much more straigthforward no ?

regards


Philippe Gentric
Software Architect
Philips MP4Net
philippe.gentric@philips.com
http://www.platform4.philips.com


                                                                                                                                       
                                                                                                                                       
                                                   To:   "'Magnus Westerlund'" <magnus.westerlund@era.ericsson.se>                     
                                                    Aaron Colwell <acolwell@real.com>                                                  
                                                    IETF MMUSIC WG <mmusic@ietf.org>                                                   
                                                   cc:   (bcc: Philippe Gentric/MP4-SUR/CE/PHILIPS)                                    
               Thomas Zeng                         Subject:    RE: [MMUSIC] draft-ietf-mmusic-rtsp-nat-00.txt questions                
               <zeng@packetvideo.com>                                                                                                  
                                                   Classification:                                                                     
               Sent by:                                                                                                                
               mmusic-admin@ietf.org                                                                                                   
                                                                                                                                       
               29/04/2003 23:24                                                                                                        
                                                                                                                                       
                                                                                                                                       






>  -----Original Message-----
> From:            Magnus Westerlund [mailto:magnus.westerlund@era.ericsson.se]
>
> Sent:            Tuesday, April 29, 2003 11:04 AM
> To:        Aaron Colwell; IETF MMUSIC WG
> Subject:         Re: [MMUSIC] draft-ietf-mmusic-rtsp-nat-00.txt questions
>
> Hi Aaron,
>
> See my answer below inline.
>
> Aaron Colwell wrote:
>
>  >I have a few questions about section 3.2.3 list item 8. It says
>  >"For RTCP it starts transmitting RTCP packets according to the rules."
>  >
>  >1. Does this mean that it starts sending  compound SR and SDES packets?
>  >
>
> Depends on the RTP session state. Unless the server has not sent any RTP
> packets yet it must send RR and SDES. If it has sent RTP packets then it
> need to send SR and SDES.
>
>  >2. Is the client supposed to created it's own CNAME for this stream of
>  >   packets or does the CNAME need to be sent from the server to the
> client?
>  >
> The client must set a CNAME of its own. That it must always do when
> sending RTCP in a RTP session.
>
>  >3. Should the client and server both send RR info to eachother about the
>  >   RTP packets they are sending?
>  >
>
> Yes, It should report on these packets.
>
>  >4. Are there any requirements for the RTP packet timestamps other than
>  >   being monotonically increasing?
>  >
>
> No, I need to think over what would be the best strategy here. One would
> be to use the RTP send time as TS basis.
             [Thomas Zeng]  This is the reason that makes it hard to treat
ip/port binding packets as RTP packets. These binding packets are not really
media packets, yet according to RTP rules, they would have to be accounted
for in RR and SRs. All of a sudden RTSP clients would now need to send SRs
back to  RTSP servers. That probably won't please implementors of thin RTSP
clients.


> In general I think we need to further look over this section. Is it a
> good thing to use PT=0, i.e. PCM audio with no samples for these
> packets? Please comment.
             [Thomas Zeng]  How about not treating binding packets as RTP packets
at all? My suggestion would be to use a simple UDP packet that contains an
SSRC (the server SSRC as indicated in SETUP response, so as to  make it
possible to uniquely identify RTP session in the event server is sending RTP
packets from a shared UDP port in order to conserve server resources -- a
common practice among implementors of large scale servers) ? If you want to
indicate that re-transmission of binding packets are taking place, we can
put a 4- byte counter in this binding packets as well. That is, the first
binding packet is try number 1, the second is try number 2, etc...  The fact
that this binding packet is smaller than RTP / RTCP header makes it real
easy for the server to distinguish between RTP and RTCP packets and the port
binding packets. Any other packets would be disgarded. A binding packet with
incorrect SSRC or counter are also discarded.

> Best Regards
>
> Magnus Westerlund
>
> Multimedia Technologies, Ericsson Research ERA/TVA/A
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se
>
>
>
>
> _______________________________________________
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From: Kreuter Ruediger <ruediger.kreuter@siemens.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Cc: Flemming Andreasen <fandreas@cisco.com>, Stephen Casner
	 <casner@acm.org>,
        Rajesh Kumar <rkumar@cisco.com>, Colin Perkins
	 <csp@isi.edu>,
        mmusic@ietf.org, mmostafa@cisco.com, hisham@cisco.com,
        mgarakan@cisco.com, paulej@cisco.com
Subject: [MMUSIC] Re: Multiple 'm' lines with the same port number
Date: Wed, 30 Apr 2003 11:34:30 +0200
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Sending a reinvite for a switchover from voice 
to voiceband data (incl. fax) is applicable.
But Rajesh's example below is text/t140, that 
is, we are "talking" about TTY/TDD. This is 
completely different because the transmission 
of TTY characters and voice can be mixed in
any way on a TDM line, and therefore must be
transferable in a VoIP connection too.
That is, it can be that during a connection
a sender types some characters on his TTY,
then he speeks some sentences, then he types
some characters ...
You don't really want to send re-invites
every couple of sentences, do you ?
The only practicable way is to mix text/t140 
and audio/G7xx the same way as audio/RFC2833 
and audio/G7xx.

Another good reason to mix these media types
is performance. Depending on the TTY terminal
type, a TTY transmission can start without a
startup sequence as it is used by fax or data
modem transmissions. Therefore the switch over
to text/t140 must be very fast otherwise you
loose characters. And the sending person doesn't
know if his characters were lost or not.
Now imagine a deaf person calling an emergancy
line ...
Sending a re-invite to switch over the connection
may use up a second in a heavy loaded network.

Regards,
Ruediger Kreuter
Ruediger.Kreuter@siemens.com



> -----Ursprüngliche Nachricht-----
> Von: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Gesendet: Dienstag, 29. April 2003 01:11
> An: Flemming Andreasen
> Cc: Stephen Casner; Rajesh Kumar; Colin Perkins; mmusic@ietf.org;
> mmostafa@cisco.com; hisham@cisco.com; mgarakan@cisco.com;
> paulej@cisco.com
> Betreff: Re: [MMUSIC] Re: Multiple 'm' lines with the same port number
> 
> 
> inline.
> 
> Flemming Andreasen wrote:
> > 
> > Stephen Casner wrote:
> > 
> > 
> >>Rajesh,
> >>
> >>
> >>>Is the use of multiple 'm=' lines with the same port 
> number within an SDP
> >>>descriptor permitted?
> >>
> >>Presuming that the multiple m= lines are all RTP/AVP, then doing so
> >>causes all of the payload formats listed on the m= lines to become
> >>part of a single RTP session (unless they are distinguished by some
> >>other aspect of addressing).  Section 5.2 of the RTP spec addresses
> >>NOT multiplexing different media in the same RTP session.
> >>
> >>
> >>>In SDP, as it is defined, all the media formats on an 'm=' 
> line are of the
> >>>same media type (e.g. audio, image, text etc.).
> >>>
> >>>Now consider the case in which two RTP media formats can 
> use a destination
> >>>port but are registered under different MIME types. For 
> example, text/t140
> >>>and audio/PCMU. In this case, there is no recourse except 
> to use multiple
> >>>media lines with the same port number. For example:
> >>
> >>The "can use a destination port" is the tricky part.  We 
> may not agree.
> >>
> > 
> > 
> > We also seem to be entering a bit of a gray area here in 
> terms of offer/answer
> > (RFC 3264). It's not clear that the definition of "m=" 
> lines and media streams
> > allow for something like this; for sure the parties would 
> have to recognize that
> > the two "m=" lines in fact are one media stream and hence 
> should not be sending
> > to "both" at the same time.  However, I'm not aware of such 
> behavior being
> > specified anywhere.
> 
> Using two m lines to represent alternate encodings for a 
> stream is not 
> consistent with the offer/answer model, IMHO. RFC 3264 has mechanisms 
> for transitioning from audio to fax, for example, but these require a 
> re-invite to do:
> 
> 8.3.3 Changing Media Types
> 
>     The media type (audio, video, etc.) for a stream MAY be 
> changed.  It
>     is RECOMMENDED that the media type be changed (as opposed 
> to adding a
>     new stream), when the same logical data is being 
> conveyed, but just
>     in a different media format.  This is particularly useful for
>     changing between voiceband fax and fax in a single 
> stream, which are
>     both separate media types.  To do this, the offerer creates a new
>     media description, with a new media type, in place of the 
> description
>     in the previous SDP which is to be changed.
> 
> 
> -Jonathan R.
> -- 
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Scientist                             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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> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 
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Date: Wed, 30 Apr 2003 07:43:43 -0700
From: Tom Marshall <tmarshall@real.com>
To: philippe.gentric@philips.com
Cc: Thomas Zeng <zeng@packetvideo.com>, Aaron Colwell <acolwell@real.com>,
        "'Magnus Westerlund'" <magnus.westerlund@era.ericsson.se>,
        IETF MMUSIC WG <mmusic@ietf.org>, mmusic-admin@ietf.org
Subject: Re: [MMUSIC] draft-ietf-mmusic-rtsp-nat-00.txt questions
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On Wed, Apr 30, 2003 at 10:22:35AM +0200, philippe.gentric@philips.com wrot=
e:
> all,
>=20
> This seems to be getting pretty complicated and tricky....

Unfortunately, that is the nature of NAT.

> Actually in the ligth of this discussion the usage of STUN looks
> so much more straigthforward no ?

Well, the case of a STUN server that is a third party to the client and
server is quite straightforward.  But that doesn't work with 5-tuple NAT
setups, so we have the RTP/STUN multiplexing case.  And that, really, is no
more or less complicated than "symmetric rtp".  They are mostly equivalent
solutions.  It only looks less complicated at this point in time because the
format for STUN packets has already been agreed upon, and we are still
looking at the format for the "symmetric rtp" packets.

--=20
Did you ever notice that fat chance and slim chance mean the same thing?

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Subject: Re: [MMUSIC] draft-ietf-mmusic-rtsp-nat-00.txt questions
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Tom Marshall wrote:

> 
>> Actually in the ligth of this discussion the usage of STUN looks so
>> much more straigthforward no ?

I was thinking the same thing.

> 
> 
> Well, the case of a STUN server that is a third party to the client
> and server is quite straightforward.  But that doesn't work with
> 5-tuple NAT setups, so we have the RTP/STUN multiplexing case.  And
> that, really, is no more or less complicated than "symmetric rtp".
> They are mostly equivalent solutions.  It only looks less complicated
> at this point in time because the format for STUN packets has already
> been agreed upon, and we are still looking at the format for the
> "symmetric rtp" packets.

Its an issue of - why are you inventing yet another thing when we 
already have something on the table?

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Scientist                             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: Re: [MMUSIC] draft-ietf-mmusic-rtsp-nat-00.txt questions
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Hi Jonathan,

Jonathan Rosenberg wrote:

> Its an issue of - why are you inventing yet another thing when we 
> already have something on the table?


Well, I think I might have to crawl to the cross. I don't like the 
multiplexing of RTP & RTCP with STUN. However it seems that symmetric 
RTP will probably be only slightly less evil. It is the RTP binding part 
that I am worried over. If we had a binding payload with a static PT I 
would not hesitate a minute. However without it the implications is a 
bit more problematic. RTCP is obviously not a problem as the binding 
RTCP packets are normal and can be forwarded to the normal processing 
without any consideration.

However one issue that remains is that a server will most likely need to 
have announced ports for RTP and RTCP which can't be changed and for 
which all RTP/RTCP and STUN traffic is directed too. From minimizing 
setup time a STUN solution for symmetric NATs should probably get the 
SDP, Do all STUN request in parallel for the media streams to the in SDP 
announce ports. Then do SETUP request with the learned destination 
addresses. The other alternative sequence where a client learns the 
servers STUN port numbers through RTSP would require two SETUP requests. 
Doesn't this make the server vulnerable?

Best Regards

Magnus Westerlund 

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



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From: Aaron Colwell <acolwell@real.com>
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Subject: Re: [MMUSIC] draft-ietf-mmusic-rtsp-nat-00.txt questions
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In my opinion requiring an implementation to use STUN to do this is a
pretty large requirement. It means the client will have to have a TLS
implementation right? The cost in code space or CPU usage of the TLS
implementation may be too high for some embedded applications like 3GPP
devices. I think demultiplexing the STUN and RTP/RTCP messages could get
tricky. The STUN solution also won't work behind a Symmetric NAT unless
the server implements STUN as well. I just feel that the use of STUN puts
too many requirements on the client and server.

I agree with Thomas though that by making these binding packets RTP data
packets, the client and server end up having to do much more RTCP protocol
then we really need. We really just need a packet that looks like an RTP
packet like Tom Marshall said. We could just get one of the reserved
payload types assigned for binding packets and specify that these packets
do not require the normal RTCP chat associated with a data stream. Old
implementations will keep state for these packets and send RTCP chat for
the data, but newer impelementations will know that they don't have to
care about these packets.

What do people think?

Aaron

On Wed, 30 Apr 2003, Jonathan Rosenberg wrote:

>
>
> Tom Marshall wrote:
>
> >
> >> Actually in the ligth of this discussion the usage of STUN looks so
> >> much more straigthforward no ?
>
> I was thinking the same thing.
>
> >
> >
> > Well, the case of a STUN server that is a third party to the client
> > and server is quite straightforward.  But that doesn't work with
> > 5-tuple NAT setups, so we have the RTP/STUN multiplexing case.  And
> > that, really, is no more or less complicated than "symmetric rtp".
> > They are mostly equivalent solutions.  It only looks less complicated
> > at this point in time because the format for STUN packets has already
> > been agreed upon, and we are still looking at the format for the
> > "symmetric rtp" packets.
>
> Its an issue of - why are you inventing yet another thing when we
> already have something on the table?
>
> -Jonathan R.
>
>
> --
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Scientist                             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: Tom Marshall <tmarshall@real.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: philippe.gentric@philips.com, Thomas Zeng <zeng@packetvideo.com>,
        Aaron Colwell <acolwell@real.com>,
        "'Magnus Westerlund'" <magnus.westerlund@era.ericsson.se>,
        IETF MMUSIC WG <mmusic@ietf.org>, mmusic-admin@ietf.org
Subject: Re: [MMUSIC] draft-ietf-mmusic-rtsp-nat-00.txt questions
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> >Well, the case of a STUN server that is a third party to the client
> >and server is quite straightforward.  But that doesn't work with
> >5-tuple NAT setups, so we have the RTP/STUN multiplexing case.  And
> >that, really, is no more or less complicated than "symmetric rtp".
> >They are mostly equivalent solutions.  It only looks less complicated
> >at this point in time because the format for STUN packets has already
> >been agreed upon, and we are still looking at the format for the
> >"symmetric rtp" packets.
>=20
> Its an issue of - why are you inventing yet another thing when we=20
> already have something on the table?

I can think of two reasons:

1. The technique is already in use, and has been proven successful.  We are
   trying to document that technique.

2. It has advantages over STUN.  It is more successful than nonmultiplexed
   STUN (eg. with 5-tuple NATs) and it does not require any protocol
   multiplexing to achieve maximum usefulness (it's a cleaner solution).

I can see where STUN may have uses for protocols other than RTP, and that is
fine.  But to imply that STUN is the one grand solution and we shouldn't
pursue symmetric rtp is, in my opinion, a mistake.

--=20
There is nothing new under the sun, but there are lots of old things we
don't know yet.
        -- Ambrose Bierce

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Subject: Re: [MMUSIC] draft-ietf-mmusic-rtsp-nat-00.txt questions
To: Aaron Colwell <acolwell@real.com>
Cc: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        "'Magnus Westerlund'" <magnus.westerlund@era.ericsson.se>,
        IETF MMUSIC WG <mmusic@ietf.org>, mmusic-admin@ietf.org,
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TLS ?

but there is no TLS involved in the type of STUN usage we want
right ? it is just sending a few UDP packets with the right (STUN) syntax,
nothing more


Philippe Gentric
Software Architect
Philips MP4Net
philippe.gentric@philips.com
http://www.platform4.philips.com


                                                                                                                                       
                                                                                                                                       
                                                   To:   Jonathan Rosenberg <jdrosen@dynamicsoft.com>                                  
                                                   cc:   Tom Marshall <tmarshall@real.com>                                             
                                                    Philippe Gentric/MP4-SUR/CE/PHILIPS@EMEA1                                          
                                                    Thomas Zeng <zeng@packetvideo.com>                                                 
               Aaron Colwell                        "'Magnus Westerlund'" <magnus.westerlund@era.ericsson.se>                          
               <acolwell@real.com>                  IETF MMUSIC WG <mmusic@ietf.org>                                                   
                                                    mmusic-admin@ietf.org                                                              
               30/04/2003 18:00                    Subject:    Re: [MMUSIC] draft-ietf-mmusic-rtsp-nat-00.txt questions                
                                                                                                                                       
                                                   Classification:                                                                     
                                                                                                                                       
                                                                                                                                       




In my opinion requiring an implementation to use STUN to do this is a
pretty large requirement. It means the client will have to have a TLS
implementation right? The cost in code space or CPU usage of the TLS
implementation may be too high for some embedded applications like 3GPP
devices. I think demultiplexing the STUN and RTP/RTCP messages could get
tricky. The STUN solution also won't work behind a Symmetric NAT unless
the server implements STUN as well. I just feel that the use of STUN puts
too many requirements on the client and server.

I agree with Thomas though that by making these binding packets RTP data
packets, the client and server end up having to do much more RTCP protocol
then we really need. We really just need a packet that looks like an RTP
packet like Tom Marshall said. We could just get one of the reserved
payload types assigned for binding packets and specify that these packets
do not require the normal RTCP chat associated with a data stream. Old
implementations will keep state for these packets and send RTCP chat for
the data, but newer impelementations will know that they don't have to
care about these packets.

What do people think?

Aaron

On Wed, 30 Apr 2003, Jonathan Rosenberg wrote:

>
>
> Tom Marshall wrote:
>
> >
> >> Actually in the ligth of this discussion the usage of STUN looks so
> >> much more straigthforward no ?
>
> I was thinking the same thing.
>
> >
> >
> > Well, the case of a STUN server that is a third party to the client
> > and server is quite straightforward.  But that doesn't work with
> > 5-tuple NAT setups, so we have the RTP/STUN multiplexing case.  And
> > that, really, is no more or less complicated than "symmetric rtp".
> > They are mostly equivalent solutions.  It only looks less complicated
> > at this point in time because the format for STUN packets has already
> > been agreed upon, and we are still looking at the format for the
> > "symmetric rtp" packets.
>
> Its an issue of - why are you inventing yet another thing when we
> already have something on the table?
>
> -Jonathan R.
>
>
> --
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Scientist                             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: Wed, 30 Apr 2003 10:51:55 -0700 (PDT)
From: Aaron Colwell <acolwell@real.com>
To: philippe.gentric@philips.com
cc: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        "'Magnus Westerlund'" <magnus.westerlund@era.ericsson.se>,
        IETF MMUSIC WG <mmusic@ietf.org>, mmusic-admin@ietf.org,
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Subject: Re: [MMUSIC] draft-ietf-mmusic-rtsp-nat-00.txt questions
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Are we intending to skip over the Shared Secret part of the STUN protocol?
On pages 7 and 24 of RFC 3489 it talks about RTP use scenarios
specifically and it says the first step is to send a shared secret
request. On pages 6 and 8 the RFC says that shared secret requests are
always sent over TLS.

Aaron

On Wed, 30 Apr 2003 philippe.gentric@philips.com wrote:

> TLS ?
>
> but there is no TLS involved in the type of STUN usage we want
> right ? it is just sending a few UDP packets with the right (STUN) syntax,
> nothing more
>
>
> Philippe Gentric
> Software Architect
> Philips MP4Net
> philippe.gentric@philips.com
> http://www.platform4.philips.com
>
>
>
>
>                                                    To:   Jonathan Rosenberg <jdrosen@dynamicsoft.com>
>                                                    cc:   Tom Marshall <tmarshall@real.com>
>                                                     Philippe Gentric/MP4-SUR/CE/PHILIPS@EMEA1
>                                                     Thomas Zeng <zeng@packetvideo.com>
>                Aaron Colwell                        "'Magnus Westerlund'" <magnus.westerlund@era.ericsson.se>
>                <acolwell@real.com>                  IETF MMUSIC WG <mmusic@ietf.org>
>                                                     mmusic-admin@ietf.org
>                30/04/2003 18:00                    Subject:    Re: [MMUSIC] draft-ietf-mmusic-rtsp-nat-00.txt questions
>
>                                                    Classification:
>
>
>
>
>
>
> In my opinion requiring an implementation to use STUN to do this is a
> pretty large requirement. It means the client will have to have a TLS
> implementation right? The cost in code space or CPU usage of the TLS
> implementation may be too high for some embedded applications like 3GPP
> devices. I think demultiplexing the STUN and RTP/RTCP messages could get
> tricky. The STUN solution also won't work behind a Symmetric NAT unless
> the server implements STUN as well. I just feel that the use of STUN puts
> too many requirements on the client and server.
>
> I agree with Thomas though that by making these binding packets RTP data
> packets, the client and server end up having to do much more RTCP protocol
> then we really need. We really just need a packet that looks like an RTP
> packet like Tom Marshall said. We could just get one of the reserved
> payload types assigned for binding packets and specify that these packets
> do not require the normal RTCP chat associated with a data stream. Old
> implementations will keep state for these packets and send RTCP chat for
> the data, but newer impelementations will know that they don't have to
> care about these packets.
>
> What do people think?
>
> Aaron
>
> On Wed, 30 Apr 2003, Jonathan Rosenberg wrote:
>
> >
> >
> > Tom Marshall wrote:
> >
> > >
> > >> Actually in the ligth of this discussion the usage of STUN looks so
> > >> much more straigthforward no ?
> >
> > I was thinking the same thing.
> >
> > >
> > >
> > > Well, the case of a STUN server that is a third party to the client
> > > and server is quite straightforward.  But that doesn't work with
> > > 5-tuple NAT setups, so we have the RTP/STUN multiplexing case.  And
> > > that, really, is no more or less complicated than "symmetric rtp".
> > > They are mostly equivalent solutions.  It only looks less complicated
> > > at this point in time because the format for STUN packets has already
> > > been agreed upon, and we are still looking at the format for the
> > > "symmetric rtp" packets.
> >
> > Its an issue of - why are you inventing yet another thing when we
> > already have something on the table?
> >
> > -Jonathan R.
> >
> >
> > --
> > Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> > Chief Scientist                             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: Wed, 30 Apr 2003 16:43:21 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: Aaron Colwell <acolwell@real.com>
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        Thomas Zeng <zeng@packetvideo.com>,
        "'Magnus Westerlund'" <magnus.westerlund@era.ericsson.se>,
        IETF MMUSIC WG <mmusic@ietf.org>, mmusic-admin@ietf.org
Subject: Re: [MMUSIC] draft-ietf-mmusic-rtsp-nat-00.txt questions
References: <20030430144343.GA5170@real.com> <3EAFE52A.8080805@dynamicsoft.com> <Pine.LNX.4.51.0304300827550.17855@raven.dev.prognet.com>
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I think there are some misconceptions here. More inline.

Aaron Colwell wrote:
> In my opinion requiring an implementation to use STUN to do this is a
> pretty large requirement. It means the client will have to have a TLS
> implementation right?

No. In the usage of STUN here, which is peer-to-peer, you would not use 
TLS. You would not need most of the features of STUN. Just the HMAC 
authentication, and the baseline case of sending a response to the place 
the request came from (i.e., no change flags, no response address, 
etc.). Really just a "ping" plus an HMAC over the address that gets 
returned.


  The cost in code space or CPU usage of the TLS
> implementation may be too high for some embedded applications like 3GPP
> devices. I think demultiplexing the STUN and RTP/RTCP messages could get
> tricky. The STUN solution also won't work behind a Symmetric NAT unless
> the server implements STUN as well. I just feel that the use of STUN puts
> too many requirements on the client and server.

The idea is that the RTSP server implements STUN, yes. Muxing the RTP 
and STUN is how you deal with symmetric RTP. How is that different from 
having the RTSP server implement the new "fake RTP", and muxing it with 
regular RTP?

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Scientist                             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: Wed, 30 Apr 2003 16:46:46 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: Tom Marshall <tmarshall@real.com>
CC: philippe.gentric@philips.com, Thomas Zeng <zeng@packetvideo.com>,
        Aaron Colwell <acolwell@real.com>,
        "'Magnus Westerlund'" <magnus.westerlund@era.ericsson.se>,
        IETF MMUSIC WG <mmusic@ietf.org>, mmusic-admin@ietf.org
Subject: Re: [MMUSIC] draft-ietf-mmusic-rtsp-nat-00.txt questions
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Tom Marshall wrote:
>>>Well, the case of a STUN server that is a third party to the client
>>>and server is quite straightforward.  But that doesn't work with
>>>5-tuple NAT setups, so we have the RTP/STUN multiplexing case.  And
>>>that, really, is no more or less complicated than "symmetric rtp".
>>>They are mostly equivalent solutions.  It only looks less complicated
>>>at this point in time because the format for STUN packets has already
>>>been agreed upon, and we are still looking at the format for the
>>>"symmetric rtp" packets.
>>
>>Its an issue of - why are you inventing yet another thing when we 
>>already have something on the table?
> 
> 
> I can think of two reasons:
> 
> 1. The technique is already in use, and has been proven successful.  We
> are
>    trying to document that technique.

IETF proposed standards are not about documenting existing technique. 
They are about proposing the right solution for all to implement.

> 
> 2. It has advantages over STUN.  It is more successful than
> nonmultiplexed
>    STUN (eg. with 5-tuple NATs) and it does not require any protocol
>    multiplexing to achieve maximum usefulness (it's a cleaner solution).

The only argument you are making here is that multiplexing is ugly. To 
that I agree. But, if you want these fake RTP packets to not affect 
RTCP, you are already basically saying that you are multiplexing a 
non-RTP protocol on the RTP port. So I dont see the difference really.

> 
> I can see where STUN may have uses for protocols other than RTP, and
> that is
> fine.  But to imply that STUN is the one grand solution and we shouldn't
> pursue symmetric rtp is, in my opinion, a mistake.
> 

Well, as is pointed out in my ICE proposal:
http://www.ietf.org/internet-drafts/draft-rosenberg-sipping-ice-00.txt

The general approach solves a ton more cases than symmetric RTP alone 
will. If you ever think that you will want RTSP in peer-to-peer cases, 
symmetric RTP isnt going to cut it.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Scientist                             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: Re: [MMUSIC] Re: Multiple 'm' lines with the same port number
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Kreuter Ruediger wrote:

> Sending a reinvite for a switchover from voice
> to voiceband data (incl. fax) is applicable.
> But Rajesh's example below is text/t140, that
> is, we are "talking" about TTY/TDD. This is
> completely different because the transmission
> of TTY characters and voice can be mixed in
> any way on a TDM line, and therefore must be
> transferable in a VoIP connection too.
> That is, it can be that during a connection
> a sender types some characters on his TTY,
> then he speeks some sentences, then he types
> some characters ...
> You don't really want to send re-invites
> every couple of sentences, do you ?
> The only practicable way is to mix text/t140
> and audio/G7xx the same way as audio/RFC2833
> and audio/G7xx.

Right - as long as both the transport and the underlying "type" of media
(not to be confused with MIME type which I think ends up being a bit too
restrictive) is the same, this should be just fine. For example, let's say
we defined a T.38 RTP payload format, but we were still using the existing
"image/t38" MIME type, then it should be perfectly fine to switch between
T.38 RTP packets and say PCMU. However, mixing different types of media,
such as audio and video, or different types of transport, such as RTP/AVP
and image/udptl for non-RTP T.38 fax is clearly not a good idea.

>
> Another good reason to mix these media types
> is performance. Depending on the TTY terminal
> type, a TTY transmission can start without a
> startup sequence as it is used by fax or data
> modem transmissions. Therefore the switch over
> to text/t140 must be very fast otherwise you
> loose characters. And the sending person doesn't
> know if his characters were lost or not.
> Now imagine a deaf person calling an emergancy
> line ...
> Sending a re-invite to switch over the connection
> may use up a second in a heavy loaded network.

Agreed - you really need a solution that enables you to switch on the fly,
however currently we don't have that when the top-level MIME types differ
(note that FID and ALT don't give you these capabilities either).  The
problem is, that SDP currently lacks a mechanism that would enable us to
negotiate different top-level MIME types in a single media stream. This can
however be addressed fairly easily by writing an extension to offer/answer
that uses SIMCAP (RFC 3407) capabilities during negotiation. If there is
interest, I can write something up.

-- Flemming



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Subject: [MMUSIC] A proposal for comedia-06 security section (repeated)
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At the risk of being rude, I'm reposting my analysis and request-for-input 
email that I submitted way back in March.

Feedback would be appreciated.

The sooner the MMUSIC group can get engaged on this, the sooner we can get 
it off our collective plate, and the sooner it can be used.

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

After talking this over with some of the folks here, it's clear that 
perhaps that the three-way handshake is the Grand Simplifying 
Notion(tm).  SIP has this property, and to date it is SIP that is the only 
protocol causing grief with the forking-INVITE.

So, my assertion is that forking-INVITE ceases to be a problem if the 
condition described in section 4 of comedia is defined to be the ACK 
message from Alice rather than the 200 response from Bob.  Taking my 
earlier forking INVITE example, here's what happens:

Alice           Proxy       Bob            Dan

Offer   ------->
                  Offer ---->
                        |
                        +------------------>
                        <----Answer
        <---------Answer
                        <------------------Answer
                  Cancel------------------->

CONNECT ------------------->
ACK     ------------------->

By delaying the initiation of the connect-then-answer listed in section 
4(b/c) to the transmission of the ACK, we've allowed the fork to collapse 
back to a pair of endpoints before any connections are initiated.  So 
instead of dealing with the large permutations caused by the 1-to-n fan-out 
of Alice to possible destinations, we're back to just Alice and Bob (Bob 
being defined to always be the "winner" of the forked INVITE).

With this in mind, there are now five unique scenarios of Alice, Bob, and 
Carol:

1) FULL CONNECTIVITY (OR BOB CANNOT CONNECT TO ALICE):

Alice           Carol          Bob

Offer ------------------------>
       <-----------------------Answer
                CONNECT------->

CONNECT----------------------> ??
ACK   ----------------------->
       <--------CONNECT

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

2) ALICE CANNOT CONNECT TO BOB:

Alice           Carol          Bob

Offer ------------------------>
       <-----------------------Answer
                CONNECT------->

CONNECT----X
ACK   ----------------------->
       <--------CONNECT
                Hijack!

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

3) ALICE/CAROL CANNOT CONNECT TO BOB:

Alice           Carol          Bob

Offer ------------------------>
       <-----------------------Answer
                CONNECT--X

CONNECT----X
ACK   ----------------------->
       <--------CONNECT
??    <-----------------------CONNECT

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

4) CAROL CANNOT CONNECT TO ALICE (OR CAROL/BOB CANNOT CONNECT TO ALICE):

Alice           Carol          Bob

Offer ------------------------>
       <-----------------------Answer
                CONNECT------->

CONNECT----------------------> ??
ACK   ----------------------->
             X--CONNECT

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

5) CAROL CANNOT CONNECT TO BOB:

Alice           Carol          Bob

Offer ------------------------>
       <-----------------------Answer
                CONNECT--X

CONNECT---------------------->
ACK   ----------------------->
       <--------CONNECT

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

So there are two possible problems.  The first is #2, where the connection 
can be hijacked.  The second is #5, where the stakes are raised for Alice 
to choose correctly between the connection to Bob and the connection to Carol.

#2 has the same mitigation strategy as before---if Alice is on the public 
Internet and Bob is not, then Alice should use direction:passive.

#5 is trickier, any thoughts are welcomed.

Jonathan, you mentioned that there were some permutations of Offer/Answer 
that may upset this particular apple cart.  Now that this has been all laid 
out, what say you on this topic?

At 01:14 PM 3/17/2003, David Yon wrote:

>Temporarily flipping to a SIP-centric worldview, wouldn't one answer be to 
>map comedia section 4 (b/c) to the ACK message from Alice rather than the 
>OK reply from Bob?  This doesn't answer the question for two-way 
>handshakes, but it seems to for three-way.

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