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From: Flemming Andreasen <fandreas@cisco.com>
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Hi Joerg and Colin

We have updates to the following two drafts, which we presented in Seoul:
1. Security preconditions 
(http://www.ietf.org/internet-drafts/draft-andreasen-mmusic-securityprecondition-02.txt)
2. Connectivity preconditions 
(http://www.ietf.org/internet-drafts/draft-andreasen-mmusic-connectivityprecondition-01.txt)

I'm not aware of any open technical issues with either draft, but can 
present them both again if you like. Also, I'd appreciate your thoughts 
on what the next steps for each of these drafts should be.

Thanks

-- Flemming



Joerg Ott wrote:

> Folks,
>
> MMUSIC is scheduled to meet once in Washington DC, currently
>
>     on Tuesday, 9 Nov 2004, 9:00-11:30
>
> This mail is to solicit input on the agenda.  We have quite a
> few work items to finish up, so clear preference will be given
> to discuss and resolve issues with our deliverables.
>
> Please send requests for agenda time to the chairs (if you have not
> already done so) and please refer to the Internet Draft you will
> be discussing (and to issues to be discussed).
>
> Thanks,
> Joerg
>
>
>
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
>

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A draft agenda is attached. Comments, corrections and additions to the  
chairs please.

Colin


======================================================================== 
===
Multiparty Multimedia Session Control (MMUSIC) Working Group

CHAIRS: Joerg Ott     <jo@acm.org>
         Colin Perkins <csp@csperkins.org>


Monday, 8th November 2004 at 09:00-11:30
========================================

Introduction and Status Update                          (Chairs)         
10
         Remote Camera Control -> AVT
         draft-ietf-mmusic-anat-02.txt
         draft-ietf-mmusic-offer-answer-examples-05.txt
         draft-ietf-mmusic-rfc2326bis-08.txt
         draft-ietf-mmusic-rtsp-announce-00.txt
         draft-ietf-mmusic-sdp-comedia-09.txt
         draft-ietf-mmusic-comedia-tls-02.txt
         draft-wing-mmusic-symmetric-rtprtcp-01.txt
         draft-ekim-mmusic-sdp-membership-01.txt

Application Sharing                                     (Schulzrinne)    
10
         draft-schulzrinne-mmusic-sharing-00.txt

Interactive Connectivity Establishment (ICE)            (Rosenberg)      
30
         draft-ietf-mmusic-ice-03.txt

Internet Media Guides                                   (Nomura)         
20
         draft-nomura-mmusic-img-notify-00.txt
         draft-walsh-mmusic-img-envelope-01.txt

SDP                                                     (Perkins)        
10
	draft-ietf-mmusic-sdp-new-21.txt

SDP Extension for Media Loopback                        (Hedayat)        
10
         draft-hedayat-media-loopback-01.txt

SDP Format for Binary Floor Control Protocol (BFCP)     (Camarillo)      
10
         draft-camarillo-mmusic-sdp-bfcp-00.txt

SDP Label Attribute                                     (Camarillo)      
5
         draft-ietf-mmusic-sdp-media-label-00.txt

SDP Connection-Establishment Precondition               (Camarillo)      
5
         draft-ietf-mmusic-connection-precon-00.txt

SDP Security Precondition                               (Andreasen)      
5
         draft-andreasen-mmusic-securityprecondition-02.txt

SDP Connectivity Precondition                           (Andreasen)      
5
         draft-andreasen-mmusic-connectivityprecondition-01.txt


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Hi,

I was just reading over the discussion regarding the bidding 
down attacks with respect to draft-ietf-mmusic-kmgmt-ext-11.txt 
and the DHHMAC draft (extension to MIKEY).

I understand the argumentation that just checking the the SDP 
part may not help. I have a scenario in mind which I don't see 
covered by the current discussion:
Let's assume somebody includes two MIKEY offers in an INVITE, 
one for PW based encryption and one for public key encryption. 
The SDP payload for this would look like 

   a=key-mgmt:mikey dhsoKkdOokdo7eWsnDSJDuiSDF9sdhs727ghsd/..
   a=key-mgmt:mikey uiSDF9sdhs727ghsd/dhsoKkdOokdo7eWsnDSJD..

were the first line is bound to the password based encryption 
method and the second to the public key encryption method. Now, 
the protocol list, which goes to MIKEY to be protected would be 
"mikey;mikey". 

Assuming that an attacker may have gotten my password (the 
assumption is not a weak MIKEY method), he might be interested 
in replacing the second line by the first line. The integrity 
protection of "mikey;mikey" is still vaild, but the responder 
doesn't really have a choice since both offers are equeal. I'm 
not sure if this example is somewhat constructed, but I think 
the responder wouldn't even notice that a MITM changed the offer 
(unless protected by other means like TLS or S/MIME). Taking the 
approach from Martin would at least enable the receiver that 
something changed, even if he may not be able 
to process the other MIKEY method.

How does the current draft deal with such a situation?

Regards
	Steffen

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<div align="left"><font face="Courier New"><span style="font-size:10pt">Hi Mats,</span></font></div>
<div align="left"><br/>
</div>
<div align="left"><font face="Courier New"><span style="font-size:10pt">i put my comments also inline:</span></font></div>
<div align="left"><br/></div>
<div align="left"><br/></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; Hi Steffen,</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; Thanks for the additional comments. Our reply is inline.</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; Steffen Fries wrote:</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; Hi,</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; I was just reading over the discussion regarding the bidding</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; down attacks with respect to draft-ietf-mmusic-kmgmt-ext-11.txt and</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; the DHHMAC draft (extension to MIKEY).</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; I understand the argumentation that just checking the the SDP</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; part may not help. I have a scenario in mind which I don't see</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; covered by the current discussion:</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; Let's assume somebody includes two MIKEY offers in an INVITE,</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; one for PW based encryption and one for public key encryption.</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; The SDP payload for this would look like</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt;&#160;&#160;&#160; a=key-mgmt:mikey dhsoKkdOokdo7eWsnDSJDuiSDF9sdhs727ghsd/..</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt;&#160;&#160;&#160; a=key-mgmt:mikey uiSDF9sdhs727ghsd/dhsoKkdOokdo7eWsnDSJD..</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; were the first line is bound to the password based encryption</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; method and the second to the public key encryption method. Now, the</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; protocol list, which goes to MIKEY to be protected would be</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; &quot;mikey;mikey&quot;.</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; Assuming that an attacker may have gotten my password (the</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; assumption is not a weak MIKEY method), he might be interested</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; in replacing the second line by the first line. The integrity</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; protection of &quot;mikey;mikey&quot; is still vaild, but the responder</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; doesn't really have a choice since both offers are equeal. I'm</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; not sure if this example is somewhat constructed, but I think</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; Well, let's make another construction. Suppose we do introduce</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; Martin's suggestion and that all else is as you assume above.</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; The initiator sends something like</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt;&#160;&#160;&#160;&#160;&#160; a=key-mgmt:mikey dhsoKkdOokdo7eWsnDSJDuiSDF9sdhs727ghsd/..</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt;&#160;&#160;&#160;&#160;&#160; a=key-mgmt:mikey uiSDF9sdhs727ghsd/dhsoKkdOokdo7eWsnDSJD..</span></font></div>
<div align="left"><br/></div>
<div align="left"><font face="Courier New"><span style="font-size:10pt">I actually thought here on something like </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&#160;&#160;&#160;&#160; a=key-mgmt:mikeyPW dhsoKkdOokdo7eWsnDSJDuiSDF9sdhs727ghsd/..</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&#160;&#160;&#160;&#160; a=key-mgmt:mikeyPS uiSDF9sdhs727ghsd/dhsoKkdOokdo7eWsnDSJD..</span></font></div>
<div align="left"><font face="Courier New"><span style="font-size:10pt">which would enable an easier processing as Matin suggested.</span></font></div>
<div align="left"><br/>
</div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; where the both lines use the protocol list &quot;mikeyPK; mikeyPW&quot;.</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; Since the attacker knows the password, he could just as easily</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; delete the first offer altogether and fake the (single) offer</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt;&#160;&#160;&#160;&#160;&#160; a=key-mgmt:mikey &lt;something&gt;</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; which is a mikeyPW offer using the protocol list &quot;mikeyPW&quot;</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; (Guess this is equivalent to your attack but with a different</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &quot;flavor&quot;). This would also be a bidding down attack, but</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; the extended list format does not help against this.</span></font></div>
<div align="left"><font face="Courier New"><span style="font-size:10pt">Agreed</span></font></div>
<div align="left"><br/>
</div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; In fact, if the Initiator is not even on line, the fact</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; that the attacker knows the password enables him to generate</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; arbitrary (single) mikeyPW offers at any time, regardless</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; of the format of the protocol list.</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; the responder wouldn't even notice that a MITM changed the offer</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; (unless protected by other means like TLS or S/MIME). Taking the</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; Same in our example.</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; approach from Martin would at least enable the receiver that</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; something changed, even if he may not be able</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; to process the other MIKEY method.</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; Our view is that to demonstrate the need for the extended identifiers</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; you need to make the assumption of an existing serious security hole</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; (a VERY weak MIKEY version, the knowledge of a key/password, etc). But</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; in that case there are much worse things to worry about than the</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; bidding-down issue. And, even *with* the extended identifiers the</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; bidding down problem remains in that case as we hope we showed above.</span></font></div>
<div align="left"><font face="Courier New"><span style="font-size:10pt">Okay, I got your point. Just for my curiosity, with this 
argument, what is the point to protect the KMGT IDs by the key 
management protocol at all? If some property like the password 
is known, a MITM may replace the complete key management offer 
with something else (like in your example). If a weak protocol 
is used than you have the chance that the attacker is not able 
to break it in the normal call setup time. Is that the goal?</span></font></div>
<div align="left"><br/>
</div>
<div align="left"><br/></div>
<div align="left"><font face="Courier New"><span style="font-size:10pt">regards</span></font></div>
<div align="left"><font face="Courier New"><span style="font-size:10pt">&#160;&#160;&#160;&#160;&#160;&#160;Steffen</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; We thus do not believe the &quot;real&quot; security issue is solved</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; by the extension of the protocol IDs. Why would an attacker wait</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; to see &quot;real&quot; offers flying by and then change them, when he</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; can attack himself at any time?</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; How does the current draft deal with such a situation?</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; Regards</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt;&#160;&#160;&#160;&#160;&#160;&#160;&#160;&#160; Steffen</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; &gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; It is not addressed specifically.</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; Regardless of the above, we again emphasize that the negotiation</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; mechanism of key management protocols is intended for different</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; protocols, not negotiating different types of the same protocol.</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; Best,</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; /Mats and MIKEY team</span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; </span></font></div>
<div align="left"><font face="Courier New" color="#7f0000"><span style="font-size:10pt">&gt; </span></font></div>
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Subject: [MMUSIC] Re: draft-ietf-mmusic-kmgmt-ext-11.txt: KMID? IANA
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Hi again Steffen,

Steffen Fries wrote:

 > Hi Mats,
 >
 > i put my comments also inline:
 >
 >
 >>  Hi Steffen,
 >>
 >>  Thanks for the additional comments. Our reply is inline.
 >>
 >>  Steffen Fries wrote:
 >>
 >>  > Hi,
 >>  >
 >>  > I was just reading over the discussion regarding the bidding
 >>  > down attacks with respect to draft-ietf-mmusic-kmgmt-ext-11.txt and
 >>  > the DHHMAC draft (extension to MIKEY).
 >>  >
 >>  > I understand the argumentation that just checking the the SDP
 >>  > part may not help. I have a scenario in mind which I don't see
 >>  > covered by the current discussion:
 >>  > Let's assume somebody includes two MIKEY offers in an INVITE,
 >>  > one for PW based encryption and one for public key encryption.
 >>  > The SDP payload for this would look like
 >>  >
 >>  >    a=key-mgmt:mikey dhsoKkdOokdo7eWsnDSJDuiSDF9sdhs727ghsd/..
 >>  >    a=key-mgmt:mikey uiSDF9sdhs727ghsd/dhsoKkdOokdo7eWsnDSJD..
 >>  >
 >>  > were the first line is bound to the password based encryption
 >>  > method and the second to the public key encryption method. Now, the
 >>  > protocol list, which goes to MIKEY to be protected would be
 >>  > "mikey;mikey".
 >>  >
 >>  > Assuming that an attacker may have gotten my password (the
 >>  > assumption is not a weak MIKEY method), he might be interested
 >>  > in replacing the second line by the first line. The integrity
 >>  > protection of "mikey;mikey" is still vaild, but the responder
 >>  > doesn't really have a choice since both offers are equeal. I'm
 >>  > not sure if this example is somewhat constructed, but I think
 >>
 >>  Well, let's make another construction. Suppose we do introduce
 >>  Martin's suggestion and that all else is as you assume above.
 >>
 >>  The initiator sends something like
 >>
 >>      a=key-mgmt:mikey dhsoKkdOokdo7eWsnDSJDuiSDF9sdhs727ghsd/..
 >>      a=key-mgmt:mikey uiSDF9sdhs727ghsd/dhsoKkdOokdo7eWsnDSJD..
 >
 > I actually thought here on something like
 >      a=key-mgmt:mikeyPW dhsoKkdOokdo7eWsnDSJDuiSDF9sdhs727ghsd/..
 >      a=key-mgmt:mikeyPS uiSDF9sdhs727ghsd/dhsoKkdOokdo7eWsnDSJD..
 > which would enable an easier processing as Matin suggested.

Oh, sure, this was a typo by me. Nevertheless the issues remain
as we seem to agree (and just "seeing" mikeyX at this point doesn't
mean you can trust it until after the line has been parsed and
authenticated.)

 >
 >>  where the both lines use the protocol list "mikeyPK; mikeyPW".
 >>  Since the attacker knows the password, he could just as easily
 >>  delete the first offer altogether and fake the (single) offer
 >>
 >>      a=key-mgmt:mikey <something>
 >>
 >>  which is a mikeyPW offer using the protocol list "mikeyPW"
 >>  (Guess this is equivalent to your attack but with a different
 >>  "flavor"). This would also be a bidding down attack, but
 >>  the extended list format does not help against this.
 > Agreed
 >
 >>  In fact, if the Initiator is not even on line, the fact
 >>  that the attacker knows the password enables him to generate
 >>  arbitrary (single) mikeyPW offers at any time, regardless
 >>  of the format of the protocol list.
 >>
 >>
 >>  > the responder wouldn't even notice that a MITM changed the offer
 >>  > (unless protected by other means like TLS or S/MIME). Taking the
 >>
 >>  Same in our example.
 >>
 >>  > approach from Martin would at least enable the receiver that
 >>  > something changed, even if he may not be able
 >>  > to process the other MIKEY method.
 >>
 >>  Our view is that to demonstrate the need for the extended identifiers
 >>  you need to make the assumption of an existing serious security hole
 >>  (a VERY weak MIKEY version, the knowledge of a key/password, etc). But
 >>  in that case there are much worse things to worry about than the
 >>  bidding-down issue. And, even *with* the extended identifiers the
 >>  bidding down problem remains in that case as we hope we showed above.
 > Okay, I got your point. Just for my curiosity, with this argument, what
 > is the point to protect the KMGT IDs by the key management protocol at
 > all? If some property like the password is known, a MITM may replace the
 > complete key management offer with something else (like in your
 > example). If a weak protocol is used than you have the chance that the
 > attacker is not able to break it in the normal call setup time. Is that
 > the goal?


The inclusion of the (authenticated) protocol list is mainly to protect
against somebody *deleting* an entire a-line which (had it been
received) would have been the Responder's preferred choice.

For example, say I offer you
one a-line based on MIKEY and one based on ProtocolZ (maybe because
I have your public key, but I am not sure which protocol support).
Now suppose the MITM just deletes the MIKEY line but lets ProtocolZ
pass. Assuming you support ProtcolZ, you will probably accept that,
but you  might actually have preferred MIKEY (for security or other
reasons). The protocol list gives a way for you to check that
*something* was lost along the way.

Similarly, if I offer you three MIKEY options (though we do not see
that as realistic) the Responder can detect if one or two were
deleted by an attacker. He cannot detect *what* the other offers were,
but he does detect the attack.

Hope this clarifies the rationale.

/Mats and MIKEY team


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From: "Steffen Fries" <steffen.fries@siemens.com>
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Hi Mats,

I've a question to your last point:
 
> Similarly, if I offer you three MIKEY options (though we do not see
> that as realistic) the Responder can detect if one or two were deleted
> by an attacker. He cannot detect *what* the other offers were, but he
> does detect the attack.

Given the example we used before:

   a=key-mgmt:mikey dhsoKkdOokdo7eWsnDSJDuiSDF9sdhs727ghsd/..
   a=key-mgmt:mikey uiSDF9sdhs727ghsd/dhsoKkdOokdo7eWsnDSJD..

were the first line is bound to the password based encryption 
method and the second to the public key encryption method. Now, 
the protocol list, which goes to MIKEY to be protected would be 
"mikey;mikey". By replacing line 2 with line 1 you force the 
receipient to go with MIKEY PW, even he would have preferred 
MIKEY PK. And the integrity check here should be fine.

Is it worth to mention this attack somewere in the security 
considerations?

Regards
	Steffen



-----------------------------------------------------------
    Steffen Fries,     Siemens AG, CT IC 3	
    Otto-Hahn-Ring 6,  D-81730 Munich, Germany 
    Phone:  (+49) 89 / 636-53403,    
    Fax  :  (+49) 89 / 636-48000
    Email:  Steffen.Fries@siemens.com
-----------------------------------------------------------



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Ciao Steffen,

Steffen Fries wrote:

> Hi Mats,
> 
> I've a question to your last point:
>  
>  > Similarly, if I offer you three MIKEY options (though we do not see
>  > that as realistic) the Responder can detect if one or two were deleted
>  > by an attacker. He cannot detect *what* the other offers were, but he
>  > does detect the attack.
> 
> Given the example we used before:
> 
>    a=key-mgmt:mikey dhsoKkdOokdo7eWsnDSJDuiSDF9sdhs727ghsd/..
>    a=key-mgmt:mikey uiSDF9sdhs727ghsd/dhsoKkdOokdo7eWsnDSJD..
> 
> were the first line is bound to the password based encryption
> method and the second to the public key encryption method. Now,
> the protocol list, which goes to MIKEY to be protected would be
> "mikey;mikey". By replacing line 2 with line 1 you force the
> receipient to go with MIKEY PW, even he would have preferred
> MIKEY PK. And the integrity check here should be fine.

Yes, this is possible if you (still contrary to intent) negotiate
within the same protocol. The duplication is still in principle
detectable though (by pure semantics), since it makes no sense to
offer *exactly* the same twice.

In fact, even technically, how could the Responder choose between
two apparently identical offers without parsing them? According
to MIKEY RFC, any MIKEY message that has come in and been authenticated
should be in the replay cache. Thus, a compliant MIKEY implementation
should in fact automatically catch the second line as a replay.

> 
> Is it worth to mention this attack somewere in the security
> considerations?

Not sure, but with reference to this (long) discussion leading
(in our opinion) to the conclusion that not even misuse of the
functionality has any real security implication, we believe that
perhaps it is the intended use of multi-offer that should be
elaborated?

Best,

/Mats and MIKEY team


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Hi Mats,
> 
> Yes, this is possible if you (still contrary to intent) negotiate
> within the same protocol. The duplication is still in principle
> detectable though (by pure semantics), since it makes no sense to
> offer *exactly* the same twice.
correct, it makes not sense, it was just a constructed scenario

> 
> In fact, even technically, how could the Responder choose between two
> apparently identical offers without parsing them? According to MIKEY
> RFC, any MIKEY message that has come in and been authenticated should
> be in the replay cache. Thus, a compliant MIKEY implementation should
> in fact automatically catch the second line as a replay.
good point, I did not think on the replay cache as both offers 
are transmitted in the same message. But that would be detected.
> 
> > 
> > Is it worth to mention this attack somewere in the security
> > considerations?
> 
> Not sure, but with reference to this (long) discussion leading
> (in our opinion) to the conclusion that not even misuse of the
> functionality has any real security implication, we believe that
> perhaps it is the intended use of multi-offer that should be
> elaborated?
yes, I think it would be worth to provide more infos here. Maybe 
once in a while people having the same thoughts I had, then it 
is good to reflect the result of the discussion within the 
draft.

Regards
	Steffen---------------------------------------------------
--------
    Steffen Fries,     Siemens AG, CT IC 3	
    Otto-Hahn-Ring 6,  D-81730 Munich, Germany 
    Phone:  (+49) 89 / 636-53403,    
    Fax  :  (+49) 89 / 636-48000
    Email:  Steffen.Fries@siemens.com
-----------------------------------------------------------



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Magnus,

Sorry for the late response...

Comments inline..

At 01:07 AM 10/19/2004, Magnus Westerlund wrote:
>Hi Mathew
>
>See below
>
>Rishi Mathew wrote:
>
>>Magnus/All,
>>This is regarding the Transport header defined in Chapter 14.43 of the 
>>RTSP spec. Right now, it says:
>>    The usage of any parameter that was not defined in RFC 2326 or in an
>>    extended way requires that request or response contains a Require
>>    header with the "play.basic" feature tag.
>
>What I refer to as parameter is the addition of what in the ABNF is called 
>parameter, i.e. what comes after the protocol specifier. For example a new 
>parameter would be the "new_parm" below:
>
>Transport: x-real-rdt/udp;client_port=6970;server_port=17738; new_parm=XYZ

Ok..

>>It is claimed that this was done to prevent backwards compatibility 
>>problems. However, in our field implementation, this parameter has always 
>>had a different value than what was defined in the original RTSP spec. It 
>>looks like this:
>>         Transport: x-real-rdt/udp;client_port=6970;server_port=17738\r\n
>>This was legal in the original spec. Obviously, any such stipulation is 
>>going to create more problems than it solves on the field. Our suggestion 
>>is that you remove this stipulation.
>
>"x-real-rdt/udp" is protocol specifier. However that you claim that it is 
>legal in RFC 2326 is not correct. The ABNF for the transport header looks 
>like this:
>
>    Transport           =    "Transport" ":"
>                             1\#transport-spec
>    transport-spec      =    transport-protocol/profile[/lower-transport]
>                             *parameter
>    transport-protocol  =    "RTP"
>    profile             =    "AVP"
>    lower-transport     =    "TCP" | "UDP"
>
>Which means that no extension allowance was made in the ABNF. Thus it is 
>unclear that a parser will correctly handle any extension. I do agree that 
>the intention was to allow for this. However due to the shortcoming of the 
>RFC 2326 it is a dangerous proposition to use extension protocol 
>specifiers or parameters. That is the reason the text is present in the draft.
>
>If you compare the above ABNF with the new one present in the draft:
>
>Transport       =  "Transport" ":" transport-spec
>                       *("," LWS transport-spec)
>    transport-spec  =  transport-id *parameter
>    transport-id    =  transport-prot "/" profile ["/" lower-transport]
>                       ; no LWS is allowed inside transport-id
>    transport-prot    =  "RTP" / token
>    profile           =  "AVP" / token
>    lower-transport   =  "TCP" / "UDP" / token
>
>You see that there now is specified that all these types can be extended.
>
>The draft do also define IANA registries for these points of extension, 
>see 20.5.
>
>So I don't think it is appropriate to remove this, as it can't be 
>guaranteed that your proprietary extensions would correctly work with 
>another implementation following RFC 2326. When clients support the 
>updated specification and indicates this correctly, then you can be 
>certain that any correct implementation can safely ignore your proprietary 
>transport protocol.
>
>So I think I will need to clarify the note to apply to any extensions. I 
>also should change the wording to not use the word "require". My proposed 
>change is:
>
>"The usage of any parameter or protocol specifier not defined in RFC 2326 
>or which is extended, SHALL be indicated by the use of the Require header 
>and a feature tag(s) to ensure support by the receiver. This
>is to protect against failures with uncertain consequences, as the a 
>receiver of a message may do anything between ignoring a parameter, to 
>complete refuse the message."
>
>Any Comments?

The current wording as I understand it, precludes efficient negotiation of 
transport.  For example, let's say that RDT is preferred, but RTP is
acceptable.   This would be expressed by the requestor as:

Transport: 
x-real-rdt/udp;client_port=6970;mode=play,RTP/AVP/UDP;client_port=6970;mode=play

In the (intended) RFC 2326 world, a server that doesn't understand RDT
would return:
Transport: RTP/AVP/UDP;client_port=6970;server_port=17738

However, in the current specification, this would result in another round 
trip, as the SETUP would fail and would need to
be reissued.  Worse, in many current RFC 2326 implementations, the 
connection would be torn down.

Dont you think it is worth REQUIRING new server implementations to ignore a 
particular transport and go on to the next one if there is something they 
cannot understand and parse?

Thanks,
Rishi.


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


Rishi Mathew
Software Development Engineer
Helix Development Support Group
RealNetworks, Inc.
rmathew@real.com
206.892.6126
http://www.helixcommunity.org
http://www.realnetworks.com/products/support/devsupport.html


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From: "Steffen Fries" <steffen.fries@siemens.com>
Organization: Siemens AG
To: "=?ISO-8859-1?Q?Mats_N=E4slund?=" <mats.naslund@ericsson.com>,
        elisabetta.carrara@ericsson.com
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Cc: mmusic@ietf.org
Subject: [MMUSIC] Re: draft-ietf-mmusic-kmgmt-ext-11.txt: KMID? IANA
	considerations section
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Hi Mats,

yes, I will talk with Elisabetta in Washington. Anyway, I think 
we cleared the points mostly and I think whats left is only the 
reflection of the result in the draft in a way that everybody 
understands it.

Regards
	Steffen

Date sent:      	Wed, 03 Nov 2004 19:08:03 +0100
From:           	Mats N=E4slund <mats.naslund@ericsson.com>
To:             	steffen.fries@siemens.com,
       	"Elisabetta Carrara (KI/EAB)" <elisabetta.carrara@ericsson.com>
Subject:        	Re: draft-ietf-mmusic-kmgmt-ext-11.txt: KMID? IANA consid=
erations
	section

> 
> Hi again Steffen,
> 
> I saw no reason to go "public" with this:
> 
> Steffen Fries wrote:
> 
> > yes, I think it would be worth to provide more infos here. Maybe
> > once in a while people having the same thoughts I had, then it is
> > good to reflect the result of the discussion within the draft.
> 
> It was very good that you (and Martin) brought this up. One would
> indeed wish that this whole email exchange could be included in the
> RFC so as to settle it once and for all and so that future worries of
> this sort could avoided. However, that would make a rather long RFC so
> if we do anything about it, it needs to be rather compact text.
> 
> As I understood, you and Elisabetta would probably meet in Washington,
> where it may be more efficient with a f2f discussion.
> 
> Thanks,
> 
> /Mats
> 

-----------------------------------------------------------
    Steffen Fries,     Siemens AG, CT IC 3	
    Otto-Hahn-Ring 6,  D-81730 Munich, Germany 
    Phone:  (+49) 89 / 636-53403,    
    Fax  :  (+49) 89 / 636-48000
    Email:  Steffen.Fries@siemens.com
-----------------------------------------------------------



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Ping...

On Wednesday 27 October 2004 03:42 pm, Jon Ringle wrote:
> The -03 ICE draft in Figure 2, shows the Accept message occuring before the
> STUN checks are performed. I believe that this is because the STUN checks
> can't be performed until both the Initiator and Responder have created
> their connection tables. However the Accept message seems to imply that the
> user has accepted the call. The media won't flow until the after the
> connectivity checks has selected a usable transport address. This gives me
> concern in that normally when a user Accepts a call, the user speaks a
> greeting (ie. "Hello"). The greeting could be dropped because media from
> the Responder hasn't begun yet when it is spoken.
>
> Might I suggest that the Accept message shown as step (4) be replaced with
> a Provisional message (which for SIP could map to a 183 Early Media
> response). The connectivity checks could then be done before the user
> Accepts the call.
>
> Jon
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic

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. What is the business model for WiMAX? 
. What do we learn from earlier deployments? 
. What about the future extensions of the standard? 
. How is addressed the interoperability challenge? 

These questions, among others, will be addressed during the second edition
of the WiMAX Summit, next April 5-8 2005, by distinguished experts and key
players in the field.

 

The call for proposal dead line has been extended to November 30.

 

Details at:

 <http://www.upperside.fr/wimax05/wimax2005intro.htm>
http://www.upperside.fr/wimax05/wimax2005intro.htm

 


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<p class=3DMsoNormal><strong><b><font size=3D2 face=3DArial><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial'>&#8226; =
</span></font></b></strong><font
size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:10.0pt;font-family:Arial'>What
is the <strong><b><font face=3DArial><span =
style=3D'font-family:Arial'>business
model</span></font></b></strong> for WiMAX? <br>
<strong><b><font face=3DArial><span style=3D'font-family:Arial'>&#8226; =
</span></font></b></strong>What
do we learn from <strong><b><font face=3DArial><span =
style=3D'font-family:Arial'>earlier
deployments</span></font></b></strong>? <br>
<strong><b><font face=3DArial><span style=3D'font-family:Arial'>&#8226; =
</span></font></b></strong>What
about the <strong><b><font face=3DArial><span =
style=3D'font-family:Arial'>future
extensions</span></font></b></strong> of the standard? <br>
<strong><b><font face=3DArial><span style=3D'font-family:Arial'>&#8226; =
</span></font></b></strong>How
is addressed the <strong><b><font face=3DArial><span =
style=3D'font-family:Arial'>interoperability</span></font></b></strong>
challenge? <br>
<br>
These questions, among others, will be addressed during the second =
edition of
the <strong><b><font face=3DArial><span =
style=3D'font-family:Arial'>WiMAX Summit,
next April 5-8 2005</span></font></b></strong>,&nbsp;by distinguished =
experts
and key players in the field.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
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<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial'>The call for proposal dead line has been =
extended to
November 30.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial'>Details at:<o:p></o:p></span></font></p>

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font-family:Arial'><a =
href=3D"http://www.upperside.fr/wimax05/wimax2005intro.htm"
title=3D"http://www.upperside.fr/wimax05/wimax2005intro.htm"><span =
lang=3DEN-GB>http://www.upperside.fr/wimax05/wimax2005intro.htm</span></a=
></span></font><font
size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:10.0pt;font-family:Arial'><o:p></o:p></span></font></p=
>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

</div>

</body>

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To: Rishi Mathew <rmathew@real.com>
Subject: Re: [MMUSIC] Transport Header Stipulation
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Hi Rishi,

Yes, you are right the mechanism to negotiate the transport protocol is
needed. See below proposal.

Rishi Mathew wrote:
> 
> The current wording as I understand it, precludes efficient negotiation of
> transport.  For example, let's say that RDT is preferred, but RTP is
> acceptable.   This would be expressed by the requestor as:
> 
> Transport:
> x-real-rdt/udp;client_port=6970;mode=play,RTP/AVP/UDP;client_port=6970;mode=play 
> 
> 
> In the (intended) RFC 2326 world, a server that doesn't understand RDT
> would return:
> Transport: RTP/AVP/UDP;client_port=6970;server_port=17738
> 
> However, in the current specification, this would result in another round
> trip, as the SETUP would fail and would need to
> be reissued.  Worse, in many current RFC 2326 implementations, the
> connection would be torn down.
> 
> Dont you think it is worth REQUIRING new server implementations to ignore a
> particular transport and go on to the next one if there is something they
> cannot understand and parse?
> 

Already today there is a sentence in the transport header section:

    Unknown transport parameters SHALL be ignored.

I would propose that one clarifies that this also applies to complete
transport specification, and especially the transport protocol
identifier. Thus one can include a list of potentially understood
transport specifications.

However it still leaves the problem of how one indicates that a
parameter within a specific transport spec must be understood. Maybe one
should create a new transport parameter that takes parameter names that
are mandatory for this specification to be understood. That way, one
can for each transport protocol define the parameters that must be
understood. If some special case requires extension parameters, or
normally optional parameters to be understood, then one includes the
parameter names in the mandatory parameter.

Example:

Transport: RTP/AVP;multicast;mode="PLAY",
RTP/AVP;unicast;spec_par="xyz";client_port=3456-3457;mode="PLAY";mand="spec_par"

Further suggestions or comments?

Cheers

Magnus Westerlund

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




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From mmusic-bounces@ietf.org  Fri Nov  5 16:09:10 2004
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To: Magnus Westerlund <magnus.westerlund@ericsson.com>
From: Rishi Mathew <rmathew@real.com>
Subject: Re: [MMUSIC] Transport Header Stipulation
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Magnus,

Comments inline...

At 05:14 AM 11/5/2004, Magnus Westerlund wrote:
>Hi Rishi,
>
>Yes, you are right the mechanism to negotiate the transport protocol is
>needed. See below proposal.
>
>Rishi Mathew wrote:
>>The current wording as I understand it, precludes efficient negotiation of
>>transport.  For example, let's say that RDT is preferred, but RTP is
>>acceptable.   This would be expressed by the requestor as:
>>Transport:
>>x-real-rdt/udp;client_port=6970;mode=play,RTP/AVP/UDP;client_port=6970;mode=play 
>>
>>In the (intended) RFC 2326 world, a server that doesn't understand RDT
>>would return:
>>Transport: RTP/AVP/UDP;client_port=6970;server_port=17738
>>However, in the current specification, this would result in another round
>>trip, as the SETUP would fail and would need to
>>be reissued.  Worse, in many current RFC 2326 implementations, the
>>connection would be torn down.
>>Dont you think it is worth REQUIRING new server implementations to ignore a
>>particular transport and go on to the next one if there is something they
>>cannot understand and parse?
>
>Already today there is a sentence in the transport header section:
>
>    Unknown transport parameters SHALL be ignored.
>
>I would propose that one clarifies that this also applies to complete
>transport specification, and especially the transport protocol
>identifier. Thus one can include a list of potentially understood
>transport specifications.
>
>However it still leaves the problem of how one indicates that a
>parameter within a specific transport spec must be understood. Maybe one
>should create a new transport parameter that takes parameter names that
>are mandatory for this specification to be understood. That way, one
>can for each transport protocol define the parameters that must be
>understood. If some special case requires extension parameters, or
>normally optional parameters to be understood, then one includes the
>parameter names in the mandatory parameter.
>
>Example:
>
>Transport: RTP/AVP;multicast;mode="PLAY",
>RTP/AVP;unicast;spec_par="xyz";client_port=3456-3457;mode="PLAY";mand="spec_par"

This looks right... but does it mean that if there is a "MUST understand" 
parameter in a transport that I want to ignore, I have to send back an error?

For example,
Transport: 
x-real-rdt/udp;client_port=6970;mode=play;spec_par="xyz";mand="spec_par",RTP/AVP/UDP;client_port=6970;mode=play 


Lets assume a server does not understand RDT. Preferably, it should just 
ignore everything until the next "transport" and start from there. But, 
since there is a mandatory parameter in a transport that it does not 
understand, will it be forced to return an error?

>Further suggestions or comments?

I would also suggest that we modify the BNF to reflect the fact that the 
"profile" makes sense with RTP and not necessarily with other proprietary 
transport protocols.

Current syntax:
    transport-id    =  transport-prot "/" profile ["/" lower-transport]
                       ; no LWS is allowed inside transport-id
    transport-prot    =  "RTP" / token
    profile           =  "AVP" / token
    lower-transport   =  "TCP" / "UDP" / token

We could modify the BNF so that everything remains the same for RTP, but 
for other transport protocols, we could make the "profile" optional.

Proposed:
    transport-id    =  [ rtp-trans | other-trans ]
                       ; no LWS is allowed inside transport-id
    rtp-trans       =  "RTP" "/" profile ["/" lower-transport]
                       ; no LWS is allowed inside transport-id
    profile           =  "AVP" / token
    lower-transport   =  "TCP" / "UDP" / token

    other-trans     = token [ "/" token ]*

Cheers,
Rishi.

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


Rishi Mathew
Software Development Engineer
Helix Development Support Group
RealNetworks, Inc.
rmathew@real.com
206.892.6126
http://www.helixcommunity.org
http://www.realnetworks.com/products/support/devsupport.html


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Subject: [MMUSIC] Slides for tomorrow
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Folks,

please send a copy of your presentation slides for the Monday
MMUSIC session to Colin and myself by 8:00am EST tomorrow
so that we can do all the presentations from a single laptop.

Thanks,
Joerg




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Cc: mmusic@ietf.org, csp@csperkins.org, mboned@lists.uoregon.edu
Subject: [MMUSIC] Re: mboned: updated MBONED agenda for IETF 61
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Hello,

I am happy to see that multi-source SSM session layering  is still a=20
potential working topic at the IETF.
I continued my work, but mainly on the implementation (the API changed)=20
and finally named the very simple protocol that I presented here  more =20
than one year ago  : Source Specific Multicast Source Discovery Protocol=20
(SSMSDP). It still uses a shared SSM tree to signal new channels=20
presents in the session.
Source announcements toward the root of the tree can use TCP or UDP (see=20
the API doc),
but are not MSDP formated as in Rami lethonen work.

Well known applications (vic/rat/beacon)  has been launched without=20
recompilation over SSM with a tool called "ssmsdpifier", it simply trap=20
MROUTING setsockopt with ssmsdp calls. On a 10Mb/s with a single linux=20
PC with
64Mb of ram and 350Mhz the implementation is able to sustain about 1500=20
actives sources signaling
on the same process (with an announcement period of 5 sec)

Documentation, implementation, Faqs, Status are all there :

http://clarinet.u-strasbg.fr/~hoerdt/libssmsdp/

Cheers,

Hoerdt Micka=EBl


David Meyer a =E9crit :

>MBONE Deployment WG (mboned)
>
>MONDAY, November 8, 2004 (1530-1730)
>=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>
>CHAIR: David Meyer <dmm@1-4-5.net>
>
>AGENDA
>
> o Administriva
>
>   - Mailing list: majordomo@lists.uoregon.edu
>     subscribe mboned
>
>   - Scribe?
>
>   - Blue Sheets
>
> o Agenda Bashing                                               2 minute=
s
>   Meyer                                          =20
>
>
> o Review and status of work items                     =20
>
>   Active Drafts
>   -------------
>   draft-ietf-mboned-ssm232-08.txt                               1 minut=
e
>     Shepherd, et al
>     RFC Ed Queue (BCP)
>   draft-ietf-mboned-embeddedrp-06.txt                           1 minut=
e
>     Savola/Haberman
>     RFC Ed Queue (PS)
>   draft-ietf-mboned-mroutesec-04.txt                            3 minut=
es
>     Savola et al (Point Raised - writeup needed)
>   draft-ietf-mboned-msdp-deploy-06.txt                          1 minut=
e
>     McBride, et al
>     RFC Ed Queue (BCP)
>   draft-ietf-mboned-rfc3171bis-02.txt                           2 minut=
es
>     Albanna, et al
>     (Revised ID Needed)
>   draft-ietf-mboned-ipv6-multicast-issues-01.txt                5 minut=
es
>     Savola
>   draft-ietf-mboned-msdp-mib-00.txt                             5 minut=
es
>     Fenner/Thaler
>   draft-ietf-mboned-auto-multicast-03.txt                      10 minut=
es
>     Thaler, et al
>   draft-ietf-mboned-ipv4-uni-based-mcast-02.txt                 5 minut=
es
>     Thaler
>
>   Potental WG Drafts
>   ------------------
>   - Overview of the Internet Multicast Addressing Architecture 10 minut=
es
>      draft-savola-mboned-addrarch-00.txt
>      Savola
>
>   - Dynamic Multi-Source Discovery for SSM using MSDP          10 minut=
es
>      draft-lehtonen-mboned-multissm-01.txt
>      Lehtonen
>
>   - Accouting Issues In Well Managed IP Multicasting Services  10 minut=
es
>      draft-hayashi-maccnt-01.txt
>      Hayashi
>
> o IMDOC=20
>
>   - MP-LSPs (Requirements/overview of current status)         =20
>      Introduction                                               2 minut=
es
>       Meyer
>      A Chair's view on what MPLS is doing                       5 minut=
es
>       Swallow
>      Summary of Requirements                                    5 minut=
es
>       Yasukawa Seisho
>      Proposed Solutions                                         5 minut=
es
>       Papadimitriou
>      Discussion                                                13 minut=
es
>       All
>
>   - p2mp ping and traceroute                                   10 minut=
es
>      draft-yasukawa-mpls-p2mp-lsp-ping-00.txt
>      Farrel, et. al.
>          =20
>   - On-link IGMP/MLD/PIM threats and problems		        10 minutes
>      Savola
>
>
>_______________________________________________________________
>user interface: http://darkwing.uoregon.edu/~llynch/mboned.html
>web archive:  http://darkwing.uoregon.edu/~llynch/mboned/
> =20
>


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To: Awais Ali <aali@i2telecom.com>
Subject: Re: [MMUSIC] Clarification about ICE Management of Resources
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inline.

Awais Ali wrote:

> I would like to verify if my understanding of the Management of 
> Resources section in the ICE draft is correct.
> 
> Let us assume device A and device B want to establish an RTP/RTCP 
> session. A and B have 3 addresses each, as follows:
> 
> AL: local address of A
> AS: Stun derived address for A
> AT: Turn derived address for A
> 
> BL: local address of B
> BS: Stun derived address for B
> BT: Turn derived address for B
> 
> After A completes its connectivity checks, A's connection table for RTP 
> will look like this
> 
> 1. AS     BS   ... ... ... 0.2    GOOD
> 2. AS     BT   ... ... ... 0.1    GOOD
> 3. AS     BL   ... ... ... 0.4    BAD
> 4. AT     BS   ... ... ... 0.2    GOOD
> 5. AT     BT   ... ... ... 0.1    GOOD
> 6. AT     BL   ... ... ... 0.4    BAD
> 7. AL     BS   ... ... ... 0.2    GOOD
> 8. AL     BT   ... ... ... 0.1    GOOD
> 9. AL     BL   ... ... ... 0.4    BAD
> 

Right so far.

> Lets say A selects Line 7 to send RTP packets, then A will keep the 
> other five connections open by sending keep-alive packets on them. 

No. In the current draft, it would only keep alive the one its using, 
since keepalives are done at the application layer and therefore imply 
usage of the address.

> Similarly there will be 5 RTCP channels. So in total A has to keep-alive 
> 10 connections.

No, just two (RTP and RTCP).

> 
> B will also have a simillar connnection table. If B also selects the BL 
> AS line for sending Media packets, then in the above exampe, we will 
> have the situation where A is sending from a different port than it is 
> receiving on. 

Hmm, in -03, yes it would appear that this would happen.

In this case then A must send keep alive packets from AS
> to keep the NAT binding open, even if it decides not to keep the other 
> connections as backup.

This is an excellent point, thank you for pointing it out. It is a 
definite problem in the specification, since right now those keepalives 
are not happening.

What needs to happen is that you end up with a symmetric relationship, 
where both sides send to, and receive from, the same address. As such, 
you need to add an algorithm to the spec to select the source address in 
a much more deterministic way. I'd suggest that the following algorithm 
works:

A client MUST send from the transport address on which it most recently 
received an RTP packet. If there is no such address (as no RTP packets 
are being received yet), it SHOULD send from the transport address of 
highest local priority on which a STUN connectivity check has been 
received. If there is no such address (as no connectivity checks have 
been received yet), it SHOULD send from the local transport address of 
highest priority.



The first sentence of this paragraph achieves the desired effect - it 
will force media to be sent from the same port on which it is being 
received. The remaining sentences help avoid flaps. That is, they help 
ensure that you start sending RTP packets initially from the place where 
you are likely to receive them. There is no guarantee, so its really an 
optimization of sorts, which is why I put it at SHOULD strength.

Comments on this algorithm?

Thanks,
Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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Subject: [MMUSIC] ASM-like SSM [Re: mboned: updated MBONED agenda for IETF
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On Mon, 8 Nov 2004, Hoerdt Mickael wrote:
> I am happy to see that multi-source SSM session layering is still a 
> potential working topic at the IETF. I continued my work, but mainly 
> on the implementation (the API changed) and finally named the very 
> simple protocol that I presented here more than one year ago : 
> Source Specific Multicast Source Discovery Protocol (SSMSDP). It 
> still uses a shared SSM tree to signal new channels presents in the 
> session. Source announcements toward the root of the tree can use 
> TCP or UDP (see the API doc), but are not MSDP formated as in Rami 
> lethonen work.

What's more, using SSMSDP approach likely avoids the IPR concerns 
about a patent application on mroutessm-01.

The problem here is about where the work can move forward.  multissm 
could be done here, which is why some might see it more feasible; an 
application effort might get more resistance from app folks, while it 
might be in some other ways a better approach.

>>   - Dynamic Multi-Source Discovery for SSM using MSDP          10 minutes
>>      draft-lehtonen-mboned-multissm-01.txt
>>      Lehtonen

-- 
Pekka Savola                 "You each name yourselves king, yet the
Netcore Oy                    kingdom bleeds."
Systems. Networks. Security. -- George R.R. Martin: A Clash of Kings

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Sorry for the delay, Jon. Inline.

Jon Ringle wrote:

> The -03 ICE draft in Figure 2, shows the Accept message occuring before the 
> STUN checks are performed. 

The responder can perform them immediately (i.e. 4 and 5 can happen in 
parallel). However, Message 4 has to arrive before message 5 can be 
successful. Obviously, message 6 has to happen after message 4.

> I believe that this is because the STUN checks 
> can't be performed until both the Initiator and Responder have created their 
> connection tables.

Right.


  However the Accept message seems to imply that the user
> has accepted the call. 

Accept corresponds to an answer as defined in RFC 3264. That doesn't 
have to come in a SIP 200 OK; it could come in a 183.

The media won't flow until the after the connectivity
> checks has selected a usable transport address. This gives me concern in that 
> normally when a user Accepts a call, the user speaks a greeting (ie. 
> "Hello"). The greeting could be dropped because media from the Responder 
> hasn't begun yet when it is spoken.

This is called post-pickup delay. If you do send the SDP answer in the 
200 OK, the post-pickup delay is increased by a STUN RTT. That is, after 
the 200 OK, it has to wait for a STUN request and response before it can 
send media. Those RTT are e2e, unlike the signaling. Thus, it is not too 
bad.

Sending the SDP in the 183 would make the post-pickup delay zero.

> 
> Might I suggest that the Accept message shown as step (4) be replaced with a 
> Provisional message (which for SIP could map to a 183 Early Media response). 
> The connectivity checks could then be done before the user Accepts the call.

What needs to change is that the section on mapping ICE into SIP/SDP has 
to clarify that the answer can come in a 18x, and that doing so is a 
good idea from a post-pickup perspective. I'll make that change.

Thanks,
Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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Shouldn't the example in draft-ietf-mmusic-connection-precon-00.txt be 
taken through the message flow point that the attribute lines are:

a=curr e2e sendrecv
a=des e2e mandatory sendrecv

for clarity?


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


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Subject: [MMUSIC] RE: ASM-like SSM [Re: mboned: updated MBONED agenda for
	IETF 61]
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> -----Original Message-----
> From: owner-mboned@lists.uoregon.edu
> [mailto:owner-mboned@lists.uoregon.edu]On Behalf Of Pekka Savola
>
> On Mon, 8 Nov 2004, Hoerdt Mickael wrote:
> > I am happy to see that multi-source SSM session layering is still a=20
> > potential working topic at the IETF. I continued my work,=20
> but mainly=20
> > on the implementation (the API changed) and finally named the very=20
> > simple protocol that I presented here more than one year ago :=20
> > Source Specific Multicast Source Discovery Protocol (SSMSDP). It=20
> > still uses a shared SSM tree to signal new channels presents in the=20
> > session. Source announcements toward the root of the tree can use=20
> > TCP or UDP (see the API doc), but are not MSDP formated as in Rami=20
> > lethonen work.
>=20
> What's more, using SSMSDP approach likely avoids the IPR concerns=20
> about a patent application on mroutessm-01.
>=20
> The problem here is about where the work can move forward.  multissm=20
> could be done here, which is why some might see it more feasible; an=20
> application effort might get more resistance from app folks, while it=20
> might be in some other ways a better approach.

While the multissm work can be done also totally on the application =
layer, it is important to notice that network level (RP like) support is =
needed in some cases to guarantee availability and redundancy for the =
source discovery. This makes the standardisation of host level source =
discovery important.

Still there are many common goals/ideas with Mickael's source discovery =
and multissm, so perhaps we should look for one common solution for the =
end-host source discovery.

- Rami

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From mmusic-bounces@ietf.org  Mon Nov  8 15:27:16 2004
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Have we considered making the use of symmetric RTP mandatory or 
suggested with ICE? Wouldn't that solve numerous of these corner case 
issues? Aside from that, the algorithm suggested below works quite well 
to fix the keep alive problem mentioned.

-Aron Rosenberg

>This is an excellent point, thank you for pointing it out. It is a 
>definite problem in the specification, since right now those keepalives 
>are not happening.
>
>What needs to happen is that you end up with a symmetric relationship, 
>where both sides send to, and receive from, the same address. As such, 
>you need to add an algorithm to the spec to select the source address in 
>a much more deterministic way. I'd suggest that the following algorithm 
>works:
>
>A client MUST send from the transport address on which it most recently 
>received an RTP packet. If there is no such address (as no RTP packets 
>are being received yet), it SHOULD send from the transport address of 
>highest local priority on which a STUN connectivity check has been 
>received. If there is no such address (as no connectivity checks have 
>been received yet), it SHOULD send from the local transport address of 
>highest priority.
>
>
>
>The first sentence of this paragraph achieves the desired effect - it 
>will force media to be sent from the same port on which it is being 
>received. The remaining sentences help avoid flaps. That is, they help 
>ensure that you start sending RTP packets initially from the place where 
>you are likely to receive them. There is no guarantee, so its really an 
>optimization of sorts, which is why I put it at SHOULD strength.
>Comments on this algorithm?
>
>Thanks,
>Jonathan R.
>  
>

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Subject: [MMUSIC] Transport header protocol/profile/lower-layer ABNF
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Hi,

 >
 > I would also suggest that we modify the BNF to reflect the fact that the
 > "profile" makes sense with RTP and not necessarily with other proprietary
 > transport protocols.
 >
 > Current syntax:
 >     transport-id    =  transport-prot "/" profile ["/" lower-transport]
 >                        ; no LWS is allowed inside transport-id
 >     transport-prot    =  "RTP" / token
 >     profile           =  "AVP" / token
 >     lower-transport   =  "TCP" / "UDP" / token
 >
 > We could modify the BNF so that everything remains the same for RTP, but
 > for other transport protocols, we could make the "profile" optional.
 >
 > Proposed:
 >     transport-id    =  [ rtp-trans | other-trans ]
 >                        ; no LWS is allowed inside transport-id
 >     rtp-trans       =  "RTP" "/" profile ["/" lower-transport]
 >                        ; no LWS is allowed inside transport-id
 >     profile           =  "AVP" / token
 >     lower-transport   =  "TCP" / "UDP" / token
 >
 >     other-trans     = token [ "/" token ]*
 >

I assume that you mean to have
other-trans = token *("/" token)

I think that this is maybe not a bad idea as it seems more general. It 
however seem to have an  interoperability issue as a old receiver could 
throw up over the lack of a "/" after the first token.

More input on this question is appropriate.

Cheers

Magnus Westerlund

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

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Rishi Mathew wrote:

> 
> This looks right... but does it mean that if there is a "MUST understand"
> parameter in a transport that I want to ignore, I have to send back an 
> error?
> 
> For example,
> Transport:
> x-real-rdt/udp;client_port=6970;mode=play;spec_par="xyz";mand="spec_par",RTP/AVP/UDP;client_port=6970;mode=play 
> 
> 
> 
> Lets assume a server does not understand RDT. Preferably, it should just
> ignore everything until the next "transport" and start from there. But,
> since there is a mandatory parameter in a transport that it does not
> understand, will it be forced to return an error?
> 
>  >Further suggestions or comments?

Sean Sheedy made a very good suggestion to me in a hallway discussion. A 
simpler rule that makes this easier to handle would be to change the 
rule to:

A receiver MAY ignore any transport specification, but MUST understand 
all parameters within any specification it selects to use.

That way one has a very simple rule, and in cases where one has 
parameter that could be ignored, one can simply offer one transport spec 
with the parameter and another without. I don't think this will result 
in a huge number of specifications.

What do people think about it?

Cheers

Magnus Westerlund

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

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From mmusic-bounces@ietf.org  Thu Nov 11 21:51:25 2004
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From: "Fan, Chen (Chen)" <cfan@lucent.com>
To: mmusic@ietf.org
Date: Fri, 12 Nov 2004 10:46:29 +0800
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Subject: [MMUSIC] What is the necessity for using RTP/UDP instead of
	HTTP/TCP to tr ansmit video content?
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Folks,

Maybe this is a very naive question, but when I was asked this question, I
don't think I have many strong reasons.

One benefit of using RTP/UDP is the synchronization capability, but we can
also insert timestamps into the HTTP packets to realize the synchronization.


The other  benefit of using RTP/UDP is that UDP does not impose any repeat
transmission, which can reduce the network overhead. However, to guarantee
the end2end video transmission quality, usually the application layer will
use the repeat transmission mechanism in some cases. 

In addition, to penetrate the firewalls, HTTP tunnel is usually used to
encapsulate the RTP packets, so, why don't we just use HTTP for video
streaming? I don't see any stop signs on this way...

Best regards
Chen

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From mmusic-bounces@ietf.org  Fri Nov 12 10:13:45 2004
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From: Colin Perkins <csp@csperkins.org>
Subject: Re: [MMUSIC] What is the necessity for using RTP/UDP instead of
	HTTP/TCP to tr ansmit video content?
Date: Fri, 12 Nov 2004 10:04:49 -0500
To: "Fan, Chen (Chen)" <cfan@lucent.com>
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[This is really a question for the AVT working group]

On 11 Nov 2004, at 21:46, Fan, Chen (Chen) wrote:
> Maybe this is a very naive question, but when I was asked this 
> question, I
> don't think I have many strong reasons.
>
> One benefit of using RTP/UDP is the synchronization capability, but we 
> can
> also insert timestamps into the HTTP packets to realize the 
> synchronization.
>
>
> The other  benefit of using RTP/UDP is that UDP does not impose any 
> repeat
> transmission, which can reduce the network overhead. However, to 
> guarantee
> the end2end video transmission quality, usually the application layer 
> will
> use the repeat transmission mechanism in some cases.
>
> In addition, to penetrate the firewalls, HTTP tunnel is usually used to
> encapsulate the RTP packets, so, why don't we just use HTTP for video
> streaming? I don't see any stop signs on this way...

Inappropriate congestion response and retransmission behaviour are the 
most obvious reasons, especially for applications with a requirement 
for low end to end latency.

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


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This is a multi-part message in MIME format.

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Hello,

=20

I have a question regarding RTSP session IDs that perhaps someone would
answer.

=20

RFC 2326 indicates that an RTSP session identifier is not tied to a
single TCP connection or some well-defined set

of UDP packets (1.1, 1.7, 12.37, 16 session hijacking).

=20

I am curious as to what a use-case or two might be where a session ID
spans multiple TCP connections to the server? I'm guessing here, but
would they revolve around conferencing?

=20

For the generic video/audio user on the internet I assume that when they
exit out of something like real player or windows media player, that the
server stops streaming data to them, and the chances of them
reconnecting to the server and resuming where they left off are slim.
But perhaps that is only because the clients don't support that mode of
operation, i.e., saving RTSP session IDs, reconnecting to a server and
resuming the data stream.

=20

Do most media clients and servers out there support session management
across multiple control channels?

=20

Thanks,

Mark Crosland


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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Hello,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>I have a question regarding RTSP session IDs that =
perhaps
someone would answer.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>RFC 2326 indicates that an RTSP session identifier is =
not
tied to a single TCP connection or some well-defined =
set<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>of UDP packets (1.1, 1.7, 12.37, 16 session =
hijacking).<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>I am curious as to what a use-case or two might be =
where a
session ID spans multiple TCP connections to the server? I&#8217;m =
guessing
here, but would they revolve around =
conferencing?<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>For the generic video/audio user on the internet I =
assume
that when they exit out of something like real player or windows media =
player, that
the server stops streaming data to them, and the chances of them =
reconnecting
to the server and resuming where they left off are slim. But perhaps =
that is
only because the clients don&#8217;t support that mode of operation, =
i.e.,
saving RTSP session IDs, reconnecting to a server and resuming the data =
stream.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Do most media clients and servers out there support =
session
management across multiple control =
channels?<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Thanks,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Mark Crosland<o:p></o:p></span></font></p>

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Magnus,

Comment inline...

At 01:46 PM 11/10/2004, Magnus Westerlund wrote:
>Hi,
>
> >
> > I would also suggest that we modify the BNF to reflect the fact that the
> > "profile" makes sense with RTP and not necessarily with other proprietary
> > transport protocols.
> >
> > Current syntax:
> >     transport-id    =  transport-prot "/" profile ["/" lower-transport]
> >                        ; no LWS is allowed inside transport-id
> >     transport-prot    =  "RTP" / token
> >     profile           =  "AVP" / token
> >     lower-transport   =  "TCP" / "UDP" / token
> >
> > We could modify the BNF so that everything remains the same for RTP, but
> > for other transport protocols, we could make the "profile" optional.
> >
> > Proposed:
> >     transport-id    =  [ rtp-trans | other-trans ]
> >                        ; no LWS is allowed inside transport-id
> >     rtp-trans       =  "RTP" "/" profile ["/" lower-transport]
> >                        ; no LWS is allowed inside transport-id
> >     profile           =  "AVP" / token
> >     lower-transport   =  "TCP" / "UDP" / token
> >
> >     other-trans     = token [ "/" token ]*
> >
>
>I assume that you mean to have
>other-trans = token *("/" token)

Yes. Basically, any number of tokens separated by a "/"

Thanks,
Rishi.

>I think that this is maybe not a bad idea as it seems more general. It 
>however seem to have an  interoperability issue as a old receiver could 
>throw up over the lack of a "/" after the first token.
>
>More input on this question is appropriate.

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


Rishi Mathew
Software Development Engineer
Helix Development Support Group
RealNetworks, Inc.
rmathew@real.com
206.892.6126
http://www.helixcommunity.org
http://www.realnetworks.com/products/support/devsupport.html


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From mmusic-bounces@ietf.org  Fri Nov 12 19:54:28 2004
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From: Rishi Mathew <rmathew@real.com>
Subject: Re: [MMUSIC] Transport Header Stipulation
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Magnus,

Comments inline..

At 01:48 PM 11/10/2004, Magnus Westerlund wrote:
>Rishi Mathew wrote:
>
>>This looks right... but does it mean that if there is a "MUST understand"
>>parameter in a transport that I want to ignore, I have to send back an error?
>>For example,
>>Transport:
>>x-real-rdt/udp;client_port=6970;mode=play;spec_par="xyz";mand="spec_par",RTP/AVP/UDP;client_port=6970;mode=play 
>>
>>
>>Lets assume a server does not understand RDT. Preferably, it should just
>>ignore everything until the next "transport" and start from there. But,
>>since there is a mandatory parameter in a transport that it does not
>>understand, will it be forced to return an error?
>>  >Further suggestions or comments?
>
>Sean Sheedy made a very good suggestion to me in a hallway discussion. A 
>simpler rule that makes this easier to handle would be to change the rule to:
>
>A receiver MAY ignore any transport specification, but MUST understand all 
>parameters within any specification it selects to use.
>
>That way one has a very simple rule, and in cases where one has parameter 
>that could be ignored, one can simply offer one transport spec with the 
>parameter and another without. I don't think this will result in a huge 
>number of specifications.
>
>What do people think about it?

I think this is a good idea and simplifies the spec to a great extent.

Thanks,
Rishi.

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


Rishi Mathew
Software Development Engineer
Helix Development Support Group
RealNetworks, Inc.
rmathew@real.com
206.892.6126
http://www.helixcommunity.org
http://www.realnetworks.com/products/support/devsupport.html


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From mmusic-bounces@ietf.org  Tue Nov 16 10:40:41 2004
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Subject: [MMUSIC] Key mgmt extensions
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Dear all,

Roughly 1-2 weeks ago there was a discussion thread on this list
concerning possible security problems with offering several key
mgmt options in draft-ietf-mmusic-kmgmt-ext-11.txt

We have since had some discussion with Martin and Steffen, and we
have come to an agreement that we do not see any security issues,
but would anyway like to add the following text to clarify the
intention of the mechanism:

"Note that the multiple offer option has been added to optimize
signaling overhead in case the Initiator knows some key (e.g. a
public key) that the Responder has, but is unsure of what protocol
the Responder supports. The mechanism is not intended to negotiate
options within one and the same protocol."

Even if there are no security problems, we want to avoid
overly complex uses of the mechanism, e.g. it could be used
to duplicate functionality that is anyway already present
in some key mgmt protocol, and create multiple ways of doing
the same thing.

A new draft will submitted with this text included.

Best,

/Mats


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Symmetric RTP will basically be what you get with the behavior below. As 
such, it is effectively mandatory as an implicit by-product of the 
algorithm below.

-Jonathan R.

Aron Rosenberg wrote:

> Have we considered making the use of symmetric RTP mandatory or 
> suggested with ICE? Wouldn't that solve numerous of these corner case 
> issues? Aside from that, the algorithm suggested below works quite well 
> to fix the keep alive problem mentioned.
> 
> -Aron Rosenberg
> 
>> This is an excellent point, thank you for pointing it out. It is a 
>> definite problem in the specification, since right now those 
>> keepalives are not happening.
>>
>> What needs to happen is that you end up with a symmetric relationship, 
>> where both sides send to, and receive from, the same address. As such, 
>> you need to add an algorithm to the spec to select the source address 
>> in a much more deterministic way. I'd suggest that the following 
>> algorithm works:
>>
>> A client MUST send from the transport address on which it most 
>> recently received an RTP packet. If there is no such address (as no 
>> RTP packets are being received yet), it SHOULD send from the transport 
>> address of highest local priority on which a STUN connectivity check 
>> has been received. If there is no such address (as no connectivity 
>> checks have been received yet), it SHOULD send from the local 
>> transport address of highest priority.
>>
>>
>>
>> The first sentence of this paragraph achieves the desired effect - it 
>> will force media to be sent from the same port on which it is being 
>> received. The remaining sentences help avoid flaps. That is, they help 
>> ensure that you start sending RTP packets initially from the place 
>> where you are likely to receive them. There is no guarantee, so its 
>> really an optimization of sorts, which is why I put it at SHOULD 
>> strength.
>> Comments on this algorithm?
>>
>> Thanks,
>> Jonathan R.
>>  
>>
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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inline.

Awais Ali wrote:


>> Lets say I have a local socket S1 listening for UDP packets. I use 
>> STUN and TURN to derive two additional address/port pairs for that 
>> socket. I send them out in an invite. THere is one stun server 
>> listening on S1. My peer in the call will perform connectivity checks 
>> to *all three* address/port pairs, all three of which will be stun 
>> requests against the single server on S1. Now, S1 gets an RTP packet. 
>> Presumably, this is because at least one connectivity check has 
>> succeeded for the peer. However, it has three connectivity checks to 
>> do, all three of which are against the same socket, all three of 
>> whicih, if successful, could result in an RTP packet being received on 
>> that socket. Unfortunately, I cannot tell from the RTP packet alone 
>> *which* of these three checks has succeeded and thus generated an RTP 
>> packet. As a result, there is no way for me to know if I may still get 
>> valid STUn requests which need to be checked in order for connectivity 
>> checks to continue.
>>
> Ok I understand that it is not possible to determine from the RTP packet 
> which connection was used. Why can we not use the Giveup timer to 
> determine when the client has stopped its connectivity checks?. If a 
> maximum Giveup-time is specified, a peer can stop its STUN server, 
> Giveup-time after receiving the first RTP packet.

This was discussed at the IETF meeting last week.

The conclusion was that there remained sufficient compelling reasons to 
avoid the usage of so many sockets. As such, the next revision of the 
draft (which I promised to complete this week) will allow for the 
currently documented 1-1 case, and a 1-many case. In the 1-many case, 
the usage of a timer as you suggest can be used. It will be an 
implementation choice as to which approach is used; both interoperate.

-Jonathan R.



-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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Hi,


I have been trawling the web to see if there is an official way to 
describe a RTSP/HTTP (URI) stream using SDP.

I have seen some SDP descriptions that put 'http' into the transport 
in m=, the server in c=, and the path as an attribute.


Has there been any Internet draft to standardize this ?


I have mixed feelings about it - half of me thinks it is 'wrong' and 
there are better ways of describing these types of (evil ;-) 
shoutcast/icecast streams as URIs (M3U, PLS).

But the other half of me thinks it would be nice to be able to 
describe all types of media streams using SDP and announce them over 
SAP. And store them in a common format in directories of streams.



nick.



PS. Advertisement for my two perl modules:

http://search.cpan.org/~njh/Net-SAP/
http://search.cpan.org/~njh/Net-SDP/


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Subject: Re: [MMUSIC] SDP description for HTTP/RTSP streams
From: Rob Lanphier <robla@real.com>
To: Nicholas Humfrey <njh@ecs.soton.ac.uk>
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Hi Nick,

I'll leave it to others to describe the current status for various
specifications for this, since I'm very rusty on things.  The RTSP
specification itself describes how to do this for RTSP.  Additionally, I
wrote a draft for how to do this with HTTP which was fairly well
received, but the motivation for pushing it forward was fleeting.

Here's the spec:
http://www.dmn.tzi.org/ietf/mmusic/53/id/draft-lanphier-radplay-00.txt

If you'd like to pick up the torch and carry it forward, modifying it as
appropriate, I'd be happy to hand it off to you.

Rob

On Tue, 2004-11-23 at 18:28 +0000, Nicholas Humfrey wrote:
> Hi,
> 
> 
> I have been trawling the web to see if there is an official way to 
> describe a RTSP/HTTP (URI) stream using SDP.
> 
> I have seen some SDP descriptions that put 'http' into the transport 
> in m=, the server in c=, and the path as an attribute.
> 
> 
> Has there been any Internet draft to standardize this ?
> 
> 
> I have mixed feelings about it - half of me thinks it is 'wrong' and 
> there are better ways of describing these types of (evil ;-) 
> shoutcast/icecast streams as URIs (M3U, PLS).
> 
> But the other half of me thinks it would be nice to be able to 
> describe all types of media streams using SDP and announce them over 
> SAP. And store them in a common format in directories of streams.
> 
> 
> 
> nick.
> 
> 
> 
> PS. Advertisement for my two perl modules:
> 
> http://search.cpan.org/~njh/Net-SAP/
> http://search.cpan.org/~njh/Net-SDP/
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
-- 
Rob Lanphier, Development Support Manager - RealNetworks
Helix Community: http://helixcommunity.org 
Development Support:
http://www.realnetworks.com/products/support/devsupport


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From mmusic-bounces@ietf.org  Tue Nov 23 14:37:07 2004
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Subject: RE: [MMUSIC] SDP description for HTTP/RTSP streams
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Well there's descriptive language in the RTSP document itself. There's
an open source implementation at vovida.org you may use as reference.
Here's a link to the class information
http://www.vovida.org/document/rtsp/rtspstack/RtspSdp.html.

-Tom=20

> -----Original Message-----
> From: Nicholas Humfrey [mailto:njh@ecs.soton.ac.uk]=20
> Sent: Tuesday, November 23, 2004 10:28 AM
> To: mmusic@ietf.org
> Subject: [MMUSIC] SDP description for HTTP/RTSP streams
>=20
>=20
> Hi,
>=20
>=20
> I have been trawling the web to see if there is an official=20
> way to describe a RTSP/HTTP (URI) stream using SDP.
>=20
> I have seen some SDP descriptions that put 'http' into the=20
> transport in m=3D, the server in c=3D, and the path as an attribute.
>=20
>=20
> Has there been any Internet draft to standardize this ?
>=20
>=20
> I have mixed feelings about it - half of me thinks it is 'wrong' and=20
> there are better ways of describing these types of (evil ;-)=20
> shoutcast/icecast streams as URIs (M3U, PLS).
>=20
> But the other half of me thinks it would be nice to be able to=20
> describe all types of media streams using SDP and announce them over=20
> SAP. And store them in a common format in directories of streams.
>=20
>=20
>=20
> nick.
>=20
>=20
>=20
> PS. Advertisement for my two perl modules:
>=20
> http://search.cpan.org/~njh/Net-SAP/
> http://search.cpan.org/~njh/Net-SDP/
>=20
>=20
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.

	Title		: Key Management Extensions for Session Description
                          Protocol (SDP) and Real Time Streaming Protocol (RTSP)
	Author(s)	: J. Arkko, et al.
	Filename	: draft-ietf-mmusic-kmgmt-ext-12.txt
	Pages		: 26
	Date		: 2004-11-23
	
This document defines general extensions for SDP and RTSP to carry
messages, as specified by a key management protocol, in order to
secure the media. These extensions are presented as a framework, to
be used by one or more key management protocols. As such, their use
is meaningful only when complemented by an appropriate key management
protocol.

General guidelines are also given on how the framework should be used
together with SIP and RTSP. The usage with the MIKEY key management
protocol is also defined.

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From: Ross Finlayson <finlayson@live.com>
Subject: RE: [MMUSIC] SDP description for HTTP/RTSP streams
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>Well there's descriptive language in the RTSP document itself.

I don't think so.  The RTSP specification notes that a SDP description for 
a stream *could* be received using a HTTP "GET", instead of using a RTSP 
"DESCRIBE".  But that's not what Nicholas Humfrey was asking about.  I 
believe he was asking how to describe a HTTP (as opposed to RTP) media 
stream within SDP.  I.e., his question really didn't have anything to do 
with RTSP.

         Ross.



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Subject: [MMUSIC] A comment re. draft-ietf-mmusic-rfc2326bis-06.txt
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In section 16.4 ("Live Media Presentation Using Multicast") there's the 
following "Editor's Note"):

Is this example really valid? In what situations does it make sense to do a 
setup to a multicast distribution channel, and also issue PLAY requests?

Please remove this "Editor's Note".  As I noted in a message to this 
mailing list back in August:

------------------------
Why did this even arise as an issue?  RTSP *is definitely* useful for 
playing multicast sessions.  In particular, the RTSP "DESCRIBE" command is 
an excellent way for a client to get the SDP description for a multicast 
session (even though, in this case, the subsequent "SETUP" and "PLAY" 
commands turn out to be effectively no-ops).

A media player client can play a stream using a "rtsp://" URL, without 
having to know in advance whether the stream is multicast or unicast.  It 
would be silly to require that a media player use one mechanism for playing 
multicast streams, and another for playing unicast streams.

It baffles me that so many people seem to think that RTSP is for unicast 
streams only.  Please don't perpetuate this myth by removing the multicast 
example from the spec.

         Ross (who has implemented and deplayed several multicast streaming 
applications with built-in RTSP servers)



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From: Nicholas J Humfrey <njh@ecs.soton.ac.uk>
Subject: Re: [MMUSIC] SDP description for HTTP streams
Date: Wed, 24 Nov 2004 03:25:19 +0000
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Yes Ross is right. Sorry I didn't make it clear what I meant before. 
Here is an example of a URI to a Icecast stream:

http://152.78.104.83:8000/surge-live-high-mp3


And here is a proposed SDP description of that URI:

v=0
o=njh 3310243893 3310243893 IN IN4 hwickabab.ecs.soton.ac.uk
s=SURGE Live!
i=Southampton University's Radio from Glen Eyre
t=0 0
a=type:broadcast
m=audio 8000 http mpeg
c=IN IP4 152.78.104.83
a=http-path:/surge-live-high-mp3


This is modeled on a SDP description I saw generated by 
VLC/miniSAPserver - I don't think it is based on any Internet Draft.

By combining the 'audio' and the 'mpeg' in the m= line, you get the 
MIME type of the HTTP stream...


nick.



On 23 Nov 2004, at 21:32, Ross Finlayson wrote:

>
>> Well there's descriptive language in the RTSP document itself.
>
> I don't think so.  The RTSP specification notes that a SDP description 
> for a stream *could* be received using a HTTP "GET", instead of using 
> a RTSP "DESCRIBE".  But that's not what Nicholas Humfrey was asking 
> about.  I believe he was asking how to describe a HTTP (as opposed to 
> RTP) media stream within SDP.  I.e., his question really didn't have 
> anything to do with RTSP.
>
>         Ross.
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic


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Hi Mark,

Sorry for not responding before, see below for my comments.

Mark Crosland wrote:

> I have a question regarding RTSP session IDs that perhaps someone would=
=20
> answer.
>=20
> RFC 2326 indicates that an RTSP session identifier is not tied to a=20
> single TCP connection or some well-defined set
> of UDP packets (1.1, 1.7, 12.37, 16 session hijacking).
>=20
> I am curious as to what a use-case or two might be where a session ID=20
> spans multiple TCP connections to the server? I=92m guessing here, but=20
> would they revolve around conferencing?
>=20

The reasons that RTSP has session ID is not tied to a specific transport=20
is many:
- Support of UDP / non-connection oriented transport
- The possibility to have one transport connection that controls=20
multiple sessions.
- Fault tolerance, against the loss of a transport connection when in a=20
session.

There are also has been considered to use the session ID as a=20
possibility to pass the control between multiple users in a conference=20
that share the streaming servers output.
> =20
>=20
> For the generic video/audio user on the internet I assume that when the=
y=20
> exit out of something like real player or windows media player, that th=
e=20
> server stops streaming data to them, and the chances of them=20
> reconnecting to the server and resuming where they left off are slim.=20
> But perhaps that is only because the clients don=92t support that mode =
of=20
> operation, i.e., saving RTSP session IDs, reconnecting to a server and=20
> resuming the data stream.
>=20

A session and its ID is not a long lived entity normally, the default=20
session timeout is 60 seconds. Thus it can't normally be used to save=20
the state and later return to the session. What has been discussed has=20
been application layer mobility, where after a change of IP address the=20
client reconnects to the session and do a new SETUP to move to streams=20
to the new address. Another case has been to move the session between=20
devices, for example between the living room and the bedroom displays.

> =20
>=20
> Do most media clients and servers out there support session management=20
> across multiple control channels?
>=20

There are some servers that support it, some other that do not. In the=20
discussion on the topic the conclusion is to mandate servers to shall be=20
required to support non-permanent transport connection. The main=20
motivation being fault tolerance.

Cheers

Magnus Westerlund

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

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Hi Ross,

See below:

Ross Finlayson wrote:

> In section 16.4 ("Live Media Presentation Using Multicast") there's the
> following "Editor's Note"):
> 
> Is this example really valid? In what situations does it make sense to do a
> setup to a multicast distribution channel, and also issue PLAY requests?
> 
> Please remove this "Editor's Note".  As I noted in a message to this
> mailing list back in August:
> 
Yes, I don't see a big problem with removing the editors note. However 
the question was also on a more technical level rather than on the 
principle level.

So have you looked at the actual example and think it is correct?

I am also sorry that I didn't manage to edit the draft to the shape I 
have desired before the meeting. There are a number of open issues not 
edited in, and this editorial note, is clearly one things that needs to 
be concluded.

Cheers

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
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Subject: [MMUSIC] RTSP: Text proposal for new error code 463 (Destination
	Prohibited)
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Hi,

Sean has provided text for the 463 error code. I think it looks fine, 
and would like check if there is any further feedback before including 
it in the specification.

Cheers

Magnus

---- Proposal begins  -----

For Table 4:
    463   Destination Prohibited     SETUP


Add new section:

13.4.15 463 Destination Prohibited

The data transmission channel was not established because the server
prohibited access to the client address.  This error is most likely the
result of a client attempt to redirect media traffic to another
destination with a either a Destination or a dest_addr parameter in the
Transport field.


Add to section 14.45 Transport, "destination" subsection:

If the server does not allow the destination field to be set, it SHALL 
return
463 (Destination Prohibited).

Add to section 14.45 Transport, "dest_addr" subsection:

If the server does not allow the dest_addr field to be set, it SHALL return
463 (Destination Prohibited).

------- end of proposal -----------

--

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
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Subject: RE: [MMUSIC] SDP description for HTTP/RTSP streams
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An HTTP/TCP protocol ID would be useful for this (although this is =
clearly an evil thing that you are doing, suggesting this in AVT might =
get you burnt at the stake :)

(I haven't yet worked out the process for registering the much more =
angelical FLUTE/UDP protocol ID but I am told it is straightforward - so =
I live in blissful and unhelpful ignorance for now)

Otherwise I guess SDP already has all you need. It's worth taking a look =
at the mandatory fields to see which are helpful (e.g. do you want to =
time bound or is the tear-down of the TCP enough), as well as the =
multitude of options (e.g. do you expect to specify the bandwidth in =
detail, any capabilities requirements need making explicit?).

If your use case if "first try RTP/UDP" and then "something/TCP" you =
might be using a similar "alternative media" semantic to ANAT. ANAT =
provides a way to swap to an alternative IP addressing media (e.g. one =
media using v4 and one using v6). Might be worth a look (and sorry if it =
turns out to be a wild goose chase).

Cheers, Rod.


> -----Original Message-----
> From: mmusic-bounces@ietf.org=20
> [mailto:mmusic-bounces@ietf.org]On Behalf
> Of ext Nicholas Humfrey
> Sent: Tuesday, November 23, 2004 8:28 PM
> To: mmusic@ietf.org
> Subject: [MMUSIC] SDP description for HTTP/RTSP streams
>=20
>=20
>=20
> Hi,
>=20
>=20
> I have been trawling the web to see if there is an official way to=20
> describe a RTSP/HTTP (URI) stream using SDP.
>=20
> I have seen some SDP descriptions that put 'http' into the transport=20
> in m=3D, the server in c=3D, and the path as an attribute.
>=20
>=20
> Has there been any Internet draft to standardize this ?
>=20
>=20
> I have mixed feelings about it - half of me thinks it is 'wrong' and=20
> there are better ways of describing these types of (evil ;-)=20
> shoutcast/icecast streams as URIs (M3U, PLS).
>=20
> But the other half of me thinks it would be nice to be able to=20
> describe all types of media streams using SDP and announce them over=20
> SAP. And store them in a common format in directories of streams.
>=20
>=20
>=20
> nick.
>=20
>=20
>=20
> PS. Advertisement for my two perl modules:
>=20
> http://search.cpan.org/~njh/Net-SAP/
> http://search.cpan.org/~njh/Net-SDP/
>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
>=20

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James,

> Shouldn't the example in draft-ietf-mmusic-connection-precon-00.txt be 
> taken through the message flow point that the attribute lines are:
> 
> a=curr e2e sendrecv
> a=des e2e mandatory sendrecv
> 
> for clarity?

The example is complete. The thing is that, since the callee is the last 
one setting up his radio bearer, it does not need to generate any 
further offer informing the caller. It just knows that the caller is 
ready (because of the offer received previously) and that it itself (the 
callee) is ready as well. So, it simply sends a TCP SYN and, when the 
connection is ready, it starts alerting the user.

Thanks,

Gonzalo


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This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.

	Title		: SDP: Session Description Protocol
	Author(s)	: M. Handley, et al.
	Filename	: draft-ietf-mmusic-sdp-new-22.txt
	Pages		: 47
	Date		: 2004-11-29
	
This memo defines the Session Description Protocol (SDP). SDP is
intended for describing multimedia sessions for the purposes of
session announcement, session invitation, and other forms of
multimedia session initiation.

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From mmusic-bounces@ietf.org  Tue Nov 30 04:45:40 2004
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From: Colin Perkins <csp@csperkins.org>
Subject: Re: [MMUSIC] SDP description for HTTP streams
Date: Mon, 29 Nov 2004 16:29:11 +0000
To: Nicholas J Humfrey <njh@ecs.soton.ac.uk>
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Nick,

On 24 Nov 2004, at 03:25, Nicholas J Humfrey wrote:
> Yes Ross is right. Sorry I didn't make it clear what I meant before. 
> Here is an example of a URI to a Icecast stream:
>
> http://152.78.104.83:8000/surge-live-high-mp3
>
> And here is a proposed SDP description of that URI:
>
> v=0
> o=njh 3310243893 3310243893 IN IN4 hwickabab.ecs.soton.ac.uk
> s=SURGE Live!
> i=Southampton University's Radio from Glen Eyre
> t=0 0
> a=type:broadcast
> m=audio 8000 http mpeg
> c=IN IP4 152.78.104.83
> a=http-path:/surge-live-high-mp3
>
> This is modeled on a SDP description I saw generated by 
> VLC/miniSAPserver - I don't think it is based on any Internet Draft.

Looks reasonable, although I don't think it's documented anywhere. If 
there's a future option of running HTTP over something other than TCP, 
then it might make sense to use "m=audio 8000 TCP/HTTP mpeg" to align 
with the comedia spec (draft-ietf-mmusic-sdp-comedia-09.txt).

Would you be willing to write up a draft documenting this?

Colin


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Subject: Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-new-22.txt
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On 29 Nov 2004, at 20:16, Internet-Drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts 
> directories.
> This draft is a work item of the Multiparty Multimedia Session Control 
> Working Group of the IETF.
>
> 	Title		: SDP: Session Description Protocol
> 	Author(s)	: M. Handley, et al.
> 	Filename	: draft-ietf-mmusic-sdp-new-22.txt
> 	Pages		: 47
> 	Date		: 2004-11-29
> 	
> This memo defines the Session Description Protocol (SDP). SDP is
> intended for describing multimedia sessions for the purposes of
> session announcement, session invitation, and other forms of
> multimedia session initiation.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-new-22.txt

Changes in this version are:
- Fix typo in section 5.14
- Allow "message" as a media type in "m=" lines
- Update references
- Note that internationalized domain names MUST use the ACE format 
defined
   in RFC 3490, rather than being directly represented in UTF-8.
- Note that the "control" and "data" media types are defined in RFC 3840
   but not in this version of SDP or as top-level MIME types.
- Add summary of changes from RFC 2327

I believe this is now ready for working group last call.
Colin


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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.

	Title		: Connection-Oriented Media Transport in the Session
                          Description Protocol (SDP)
	Author(s)	: D. Yon
	Filename	: draft-ietf-mmusic-sdp-comedia-10.txt
	Pages		: 15
	Date		: 2004-11-30
	
This document describes how to express media transport over TCP using
   the Session Description Protocol (SDP).  It defines the SDP 'TCP'
   protocol identifier, the SDP 'setup' attribute, which describes the
   connection setup procedure, and the SDP 'connection' attribute, which
   handles connection reestablishment.

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Cc: mmusic@ietf.org
Subject: [MMUSIC] Re: [AVT] RFC 3711 and RTCP
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Alan
   I am copying MMUSIC (and not cross-posting to AVT) to further discuss  
a possible change to SDP security descriptions, which has completed WG  
last call.  The change is needed for RFC 3611, and we also discovered  
that  
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdescriptions 
-07.txt does not address the SRTCP E bit properly. I offered the  
following text in a posting to AVT:
"As the default behavior when UNENCRYPTED_SRTCP is signaled, all SRTCP  
packets MUST have the E bit set to zero and MUST NOT be encrypted.   
Otherwise, all SRTP packets MUST have the E bit set to one and MUST be  
encrypted by default.  Individual applications MAY override these  
default behaviors in their specifications.  In a session where SRTCP  
packets are encrypted, therefore, applications that use RFC 3611 MAY  
set the E bit to 0 for XR packets as recommended by RFC 3611."

One of my co-authors pointed out that the last sentence is more than an  
editorial change.  We also don't like the fact that an offerer or  
answerer cannot explicitly signal when it wants the XR packet to be  
encrypted or unencrypted.  In other words, if ENCRYPTED_SRTCP is the  
default setting and RFC 3611 is also signaled, it is ambiguous as to  
whether the SRTCP sender will encrypt the XR packet or not.  It's  
solely up to the sender.  If the offerer or answerer want the XR packet  
to always be encrypted, there is no way to signal that.  Ditto for the  
unencrypted case.

I think we need to put the first three sentences of the above-quoted  
paragraph into SDP security descriptions as an editorial change.  I'm  
proposing this to the working group.  I also propose not to include the  
last sentence.  Instead, we could add a new signaling parameter,  
UNENCRYPTED_XR, which implies that all SRTCP messages are to be  
encrypted except for the XR packets, which will have their E bit clear,  
of course.

This new parameter could be placed in an RFC 3611bis or in a revision  
of draft-ietf-mmusic-sdescriptions-07.txt, which will then need to  
undergo another WG last call.  What do you (all) think about this?

Mark

On Nov 24, 2004, at 4:18 PM, Mark Baugher wrote:

> Alan
>   As Dave Oran has pointed out, the definition that you want is not  
> supported in the signaling, at present.  The current definition of  
> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdescriptions 
> -07.txt treats SRTCP encryption as either on or off and does not  
> address the problem of setting the E bit in SRTCP packets.  This is a  
> bug that we need to fix.
>
>   MMUSIC has finished last call on  
> draft-ietf-mmusic-sdescriptions-07.txt, and the transport ADs have  
> reviewed it.  We'll be making some changes at Jon Peterson's request  
> and post another version in about a week.  This is an opportunity to  
> address the E bit, but the MMUSIC chairs or transport ADs may want  
> another WG last call.  Off the top of my head, I would offer something  
> like the following.
>
> "As the default behavior when UNENCRYPTED_SRTCP is signaled, all SRTCP  
> packets MUST have the E bit set to zero and MUST NOT be encrypted.   
> Otherwise, all SRTP packets MUST have the E bit set to one and MUST be  
> encrypted by default.  Individual applications MAY override these  
> default behaviors in their specifications.  In a session where SRTCP  
> packets are encrypted, therefore, applications that use RFC 3611 MAY  
> set the E bit to 0 for XR packets as recommended by RFC 3611."
>
> Mark
>
> On Nov 24, 2004, at 2:33 PM, Alan Clark wrote:
>
>>
>>
>> Mark
>>
>> We tried to flag this in RFC3611 with the following text in the  
>> "security
>> considerations" section
>>
>>    Any encryption or filtering of XR report blocks entails a loss of
>>    monitoring information to third parties.  For example, a network  
>> that
>>    establishes a tunnel to encrypt VoIP Report Blocks denies that
>>    information to the service providers traversed by the tunnel.  The
>>    service providers cannot then monitor or respond to the quality of
>>    the VoIP calls that they carry, potentially creating problems for  
>> the
>>    network's users.  As a default, XR packets should not be encrypted  
>> or
>>    filtered.
>>
>> Could we potentially interpret this as being the definition of what to
>> selectively encrypt in the case of the RFC3611 specification?
>>
>> Regards
>>
>> Alan
>>
>>
>> -----Original Message-----
>> From: Mark Baugher [mailto:mbaugher@cisco.com]
>> Sent: Wednesday, November 24, 2004 2:42 PM
>> To: Alan Clark
>> Cc: Oren Peleg; Cullen Jennings; avt@ietf.org; David R Oran; Michael  
>> A.
>> Ramalho
>> Subject: Re: [AVT] RFC 3711 and RTCP
>>
>>
>> So, when the sender and receiver agree to encrypt SRTCP, this would  
>> mean
>> only that some SRTCP messages would be encrypted and the sender would
>> decide which ones to encrypt?  My personal opinion is that this is not
>> secure because it is unpredictable to the receiver, which may have a
>> policy that all messages related to its session be encrypted.
>>
>> I can think of a few potential approaches to this problem.
>>
>> 1. Let individual SRTCP messages or a bundle of SRTCP messages
>> be signaled to be unencrypted when SRTCP encryption is used.
>>
>> 2. Make a blanket policy that the sender determine what
>> to selectively encrypt, as you suggest.
>>
>> 3. Define the policy of what to selectively encrypt in
>> the particular specification.  In other words, a document could
>> re-define the signaling of SRTCP encryption according to its
>> needs.  The specification of foo, therefore, could define a new
>> parameter such as foo_ENCRYPTED_SRTCP to be used in place of
>> ENCRYPTED_SRTCP.  The foo document would define what gets
>> encrypted and what does not get encrypted as per its needs.
>>
>> I commented on #2 already.  I don't like #1 because of its signaling
>> complexity.  I like #3 because it limits the complexity to the
>> specific needs of an application.  #1 would be more efficient than
>> #3 if lots of applications had their own specific needs for
>> selective encryption of SRTCP message.  But I don't think this is
>> the case.
>>
>> Mark
>>
>> On Nov 24, 2004, at 10:58 AM, Alan Clark wrote:
>>
>>>
>>> If the use of encryption in SRTCP is signalled by an explicit (E) bit
>>> in the
>>> SRTCP header then perhaps the sender could autonomously decide  
>>> whether
>>> it
>>> wants to encrypt?  Presumably since the sender is in the best  
>>> position
>>> to
>>> decide if the data is sensitive then this would be ok - and the
>>> receiver is
>>> able to just look at the E bit and hence recognize that decryption is
>>> needed.
>>>
>>> Alan
>>>
>>> -----Original Message-----
>>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org]On Behalf Of
>>> Mark Baugher
>>> Sent: Wednesday, November 24, 2004 1:33 PM
>>> To: David R Oran
>>> Cc: Cullen Jennings; Oren Peleg; Michael A. Ramalho; Alan Clark;
>>> avt@ietf.org
>>> Subject: Re: [AVT] RFC 3711 and RTCP
>>>
>>>
>>>
>>> On Nov 24, 2004, at 10:05 AM, David R Oran wrote:
>>>
>>>>
>>>> On Nov 24, 2004, at 12:23 PM, Alan Clark wrote:
>>>>
>>>>>
>>>>> Michael
>>>>>
>>>>> I don't see why this would be a problem unless you included SDES  
>>>>> and
>>>>> (SR or
>>>>> RR or XR) in the same RTCP packet.  SRTP (at least the protocol) is
>>>>> apparently able to decided on a packet by packet basis whether to
>>>>> encrypt
>>>>> RTCP - hopefully SRTP implementations also permit this.
>>>>>
>>>> Sure, they can do this. But remember offer-answer. How does a  
>>>> receiver
>>>> communicate such a granular policy to the transmitter? Right now if
>>>> can say whether to auth or auth+encrypt RTP, and auth or  
>>>> auth+encrypt
>>>> RTCP. There's no way to say auth+encrypt SDES but only auth SR/XR  
>>>> etc.
>>>
>>> Seems to me that this is a problem with SDP security descriptions.
>>> That problem might be there because we forgot about the E bit.  This
>>> seems very messy for the reason you mention above - granularity.  I
>>> don't think that we want to negotiate the encryption policy for each
>>> type SDP message, which is a problem with something like the E bit.
>>>
>>> Mark
>>>>
>>>>
>>>>> Logically I'd send SDES on its own at the start of a call to  
>>>>> provide
>>>>> some
>>>>> user info, but would not send SR/RR/XR until at least 8-10 seconds
>>>>> into the
>>>>> call as I need some time to gather statistics to report.
>>>> That's not exactly what the RTP specs say. RTCP timers don't have  
>>>> this
>>>> kind of policy today. In fact, there are porposals on the table to  
>>>> use
>>>> RTCP for early feedback to assess reception quality on some initial
>>>> packets - e.g. the RTP-noop proposal. In any event that has no  
>>>> bearing
>>>> on the present discussion.
>>>>
>>>>> After the
>>>>> start-of-call exchange there would appear to be minimal need to
>>>>> include
>>>>> SDES. I'm sure we could construct some scenarios in which you "may"
>>>>> need to
>>>>> include SDES with SR/RR/XR however the vast majority of cases would
>>>>> either
>>>>> need only a single initial exchange or may not need SDES at all.
>>>>>
>>>> Any time you change SSRC you potentially need to send SDES.
>>>> That said, I still don't see what bearing this has on the discussion
>>>> about whether we need the complexity of encrypting some RTCP packets
>>>> and not others for a single session.
>>>>
>>>> It's a tradeoff between extra complexity+less privacy versus  
>>>> allowing
>>>> snoopers to gather RTCP stats.
>>>>
>>>> Dave.
>>>>
>>>>> Regards
>>>>>
>>>>> Alan
>>>>>
>>>>>
>>>>>
>>>>> -----Original Message-----
>>>>> From: Michael A. Ramalho [mailto:mramalho@cisco.com]
>>>>> Sent: Wednesday, November 24, 2004 9:39 AM
>>>>> To: Alan Clark
>>>>> Cc: Oren Peleg; avt@ietf.org
>>>>> Subject: RE: [AVT] RFC 3711 and RTCP
>>>>>
>>>>>
>>>>> At 07:54 AM 11/24/2004 -0500, Alan Clark wrote:
>>>>>> Hi Oren
>>>>>>
>>>>>> Yes - encryption is intended to provide privacy however this  
>>>>>> relates
>>>>>> primarily to the voice payloads and to information within RTCP  
>>>>>> that
>>>>>> may be
>>>>>> associated with individual users (e.g. SDES).
>>>>>>
>>>>>> SR, RR and XR contain only peformance statistics and there is no
>>>>>> privacy
>>>>>> consideration that I can think of related to this type of data.
>>>>>>
>>>>>> The message is - encrypt RTP, optionally encrypt RTCP SDES but  
>>>>>> don't
>>>>> encrypt
>>>>>> RTCP SR/RR/XR
>>>>>
>>>>> Hi Alan,
>>>>>
>>>>> I am wondering about implementations utilizing your advise
>>>>> "optionally
>>>>> encrypt
>>>>> RTCP SDES but don't encrypt RTCP SR/RR/XR".
>>>>>
>>>>> Is there an easy way to tell SRTP to encrypt RTCP SDES, but not  
>>>>> RTCP
>>>>> SR/RR/XR? It seems that all the RTCP blocks would otherwise be  
>>>>> highly
>>>>> related to each other. For example (quote out of RFC 3611):
>>>>>
>>>>> "XR packets supplement the existing RTCP packets, and may be  
>>>>> stacked
>>>>> with
>>>>> other RTCP packets to form compound RTCP packets [9, Section 6]."
>>>>>
>>>>> Thus, RFC3611 explicitly admits the possibility of (but does not
>>>>> mandate)
>>>>> the piggybacking all the RTCP-XR block types.
>>>>>
>>>>> RTCP implementations would have to know the SRTP encryption rules  
>>>>> to
>>>>> make sure they don't perform piggybacking with SDES ... but could
>>>>> allow
>>>>> (the more bandwidth efficient) piggybacking on the other RTCP-XR
>>>>> blocks.
>>>>>
>>>>> This RTCP/SRTP interaction/dependency seems to add a burden to both
>>>>> RTCP and SRTP implementations ... let alone the expressiveness
>>>>> required
>>>>> to signal your advise in existing SDP/SRTP mechanisms.
>>>>>
>>>>> Are you aware of any implementations that have actually followed  
>>>>> your
>>>>> advise above? I rather go into this one with our eyes open ...
>>>>> developers
>>>>> highly dislike this cross-functional entity dependency (RTCP & SRTP
>>>>> along
>>>>> with expressing it in SDP/SRTP mechanisms).
>>>>>
>>>>> Regards,
>>>>>
>>>>> Michael Ramalho
>>>>>
>>>>>
>>>>>
>>>>>> Regards
>>>>>>
>>>>>> Alan
>>>>>>
>>>>>> -----Original Message-----
>>>>>> From: Oren Peleg [mailto:OrenP@audiocodes.com]
>>>>>> Sent: Wednesday, November 24, 2004 7:50 AM
>>>>>> To: Alan Clark; avt@ietf.org
>>>>>> Subject: RE: [AVT] RFC 3711 and RTCP
>>>>>>
>>>>>>
>>>>>>
>>>>>> Isnt encryption is to prevent monitoring from the first place?
>>>>>>
>>>>>> -----Original Message-----
>>>>>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf
>>>>>> Of
>>>>>> Alan Clark
>>>>>> Sent: Tuesday, November 23, 2004 5:42 PM
>>>>>> To: avt@ietf.org
>>>>>> Subject: [AVT] RFC 3711 and RTCP
>>>>>>
>>>>>>
>>>>>>> From the flurry of discussion on RFC3711 it looks as if there is
>>>>>> considerable interest in SRTP implementation.  I'd like to add a
>>>>>> request
>>>>>> to
>>>>>> SRTP implementors with regard to manageability.
>>>>>>
>>>>>> RTCP SR, RR and more particularly XR (e.g. VoIP Metrics report
>>>>>> block)
>>>>>> contain useful information for network monitoring, and are often
>>>>>> used by
>>>>>> probes and analyzers to support problem diagnosis.  RFC3711 allows
>>>>>> the
>>>>>> selective encryption of RTCP payloads and hence I would strongly
>>>>>> recomment
>>>>>> that RFC3711 implementors DO NOT encrypt RTCP SR, RR and XR  
>>>>>> payloads
>>>>>> as
>>>>>> this
>>>>>> adversely impacts manageability.
>>>>>>
>>>>>> Regards
>>>>>>
>>>>>> Alan Clark
>>>>>> Telchemy
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Audio/Video Transport Working Group
>>>>>> avt@ietf.org
>>>>>> https://www1.ietf.org/mailman/listinfo/avt
>>>>>>
>>>>>> ****************************************************************** 
>>>>>> **
>>>>>> *
>>>>>> ******
>>>>> *
>>>>>> ********************************
>>>>>> This email and any files transmitted with it are confidential
>>>>>> material.
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>>>>>> or
>>>>>> entity
>>>>>> to whom they are addressed.
>>>>>> If the reader of this message is not the intended recipient,
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>>>>> may
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>>>>>> ****************************************************************** 
>>>>>> **
>>>>>> *
>>>>>> ******
>>>>> *
>>>>>> *******************************
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Audio/Video Transport Working Group
>>>>>> avt@ietf.org
>>>>>> https://www1.ietf.org/mailman/listinfo/avt
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Audio/Video Transport Working Group
>>>>> avt@ietf.org
>>>>> https://www1.ietf.org/mailman/listinfo/avt
>>>>>
>>>> David R. Oran
>>>> Cisco Fellow
>>>> Cisco Systems
>>>> 7 Ladyslipper Lane
>>>> Acton, MA 01720 USA
>>>> Tel: +1 978 264 2048
>>>> Email: oran@cisco.com
>>>>
>>>>
>>>> _______________________________________________
>>>> Audio/Video Transport Working Group
>>>> avt@ietf.org
>>>> https://www1.ietf.org/mailman/listinfo/avt
>>>>
>>>
>>>
>>> _______________________________________________
>>> Audio/Video Transport Working Group
>>> avt@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/avt
>>>
>>
>


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