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Subject: Re: [MMUSIC] Optional SRTP?
From: Cullen Jennings <fluffy@cisco.com>
To: Alan Johnston <alan@sipstation.com>, Rick Porter <rporter@covergence.com>
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Thread-Topic: [MMUSIC] Optional SRTP?
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Hi Rick - this is an important problem and it needs to be solved. There are
a lot of proposal but certainly no solutions yet.

On 3/30/06 9:23 AM, "Alan Johnston" <alan@sipstation.com> wrote:

> Another bad alternative is to offer a secure SDP offer first, and then
> if it fails, send a second insecure SDP offer.

As has been pointed out before, one of the downsides of this is, if your
voicemail system supports SRTP, and your phone does not. This guarantees
that callers that do SRTP will only be able to get voicemail.

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From: Colin Perkins <csp@csperkins.org>
Subject: Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-fec-grouping-03.txt 
Date: Tue, 4 Apr 2006 18:25:58 +0100
To: Adam Li <adamli@hyervision.com>
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On 21 Mar 2006, at 20:50, 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		: Forward Error Correction Grouping Semantics in Session  
> Description Protocol
> 	Author(s)	: A. Li
> 	Filename	: draft-ietf-mmusic-fec-grouping-03.txt
> 	Pages		: 7
> 	Date		: 2006-3-21
> 	
> This document defines the semantics that allows for grouping of
>    forward error correction (FEC) streams with the protected payload
>    streams in Session Description Protocol (SDP). The semantics  
> defined
>    in this document is to be used with Grouping of Media Lines in the
>    Session Description Protocol (RFC 3388) to group together "m" lines
>    in the same session.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-fec- 
> grouping-03.txt

Re-reading this, I noticed a bug in the SDP example (section 4.3).  
The parity FEC codes use the "application" media type, but should  
match the media they are protecting:

        v=0
        o=adam 289083124 289083124 IN IP4 host.example.com
        s=ULP FEC Seminar
        t=0 0
        c=IN IP4 224.2.17.12/127
        a=group:FEC 1 2
        a=group:FEC 3 4
        m=audio 30000 RTP/AVP 0
        a=mid:1
        m=audio 30002 RTP/AVP 100
          ^^^^^
        a=rtpmap:100 ulpfec/8000
        a=mid:2
        m=video 30004 RTP/AVP 31
        a=mid:3
        m=video 30004 RTP/AVP 101
          ^^^^^
        c=IN IP4 224.2.17.13/127
        a=rtpmap:101 ulpfec/8000
        a=mid:4

Colin

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I apologize if this has been answered previously, but is there a mailing
list for the multiple-choice SRTP (and ZRTP) discussion?  I thought I heard
there would be when I listened to the audio-cast of the Dallas meeting, but
I haven't been able to find it. 

Thanks!

-hadriel

 


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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>I apologize if this has been answered previously, but =
is
there a mailing list for the multiple-choice SRTP (and ZRTP) =
discussion?&nbsp; I
thought I heard there would be when I listened to the audio-cast of the =
Dallas
meeting, but I haven&#8217;t been able to find it. =
<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'>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'>-hadriel<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>

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From mmusic-bounces@ietf.org Wed Apr 05 09:22:11 2006
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Subject: RE: [MMUSIC] Optional SRTP?
Date: Wed, 5 Apr 2006 09:22:03 -0400
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Thread-Topic: [MMUSIC] Optional SRTP?
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From: "Rick Porter" <rporter@covergence.com>
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Thanks for all the responses. I'm fascinated by the discussion, as I =
thought I would be when I asked the question. I raised my original =
question due to a situation I encountered in the lab. I have two phones =
as follows:=20

A - secure (SRTP, possibly TLS)=20
B - non-secure (no TLS or SRTP)

When A calls B, B rejects the call because he is not configured to =
accept a secure call (this phone requires TLS signaling to do SRTP). Per =
the SDES draft (Section 5.1.2, top of page 10), this is the correct =
behavior. I found the result (B didn't even ring, and call was not =
completed) to be rather unsatisfying. This sounds like a known issue =
with the current draft.

I hate to throw more proposals in the mix, but I have a couple thoughts =
on "simple" changes to the current draft to make a more deployable =
solution. I'm trying to take a simple, pragmatic approach that will get =
this deployed ASAP. It sounds like the conditional SDP is not deployed, =
so I do not want to depend on that. Take these proposals with as many =
shakers of salt as you see appropriate.

1) Use the same media stream. No special transport. Crypto is just =
another "optional" attribute of the media description.

2) Add an "a=3Dencryption" attribute.=20

I believe those suggestions would result in the following behavior for =
the above scenario.

B would accept the call, but not provide a crypto attribute in the =
200/OK/SDP. Then, if A does not like the 200/OK/SDP without crypto he =
can choose to CANCEL the call or continue the non-secure call. Based on =
configuration, phone A may prompt the user to accept that it is a =
non-secure call. B already knows that he is not configured to do secure =
calls, so being non-secure is not news to him.=20

In an early draft by Ahmar Ghaffar at Snom, I liked the =
"a=3Dencryption:<optional|mandatory>" attribute that could be included =
as part of the media description. If the encryption attribute is =
present, you know whether you can accept the call even if you do not =
share a common set of crypto algorithms/options. It could give the end =
receiving the INVITE the chance to reject the call early (e.g. 480) =
instead of sending the 200/OK (without crypto) and letting the other end =
cancel the call upon receipt of the 200/OK.

I know the above scenario is not the only one, so I probably missed =
something...

I believe Alan and Paul are correct with respect to not being concerned =
about a bid-down attack. If you have access to the signaling stream, =
then you have access to key material necessary to decrypt each and every =
RTP packet. No need to bid it down crypto if you can decrypt the RTP =
packets anyway. Until you do some sort of end-to-end public key exchange =
(e.g. ZRTP), then you don't have end-to-end security. There are many =
technical and political reasons why SRTP is done on a hop-by-hop =
basis--please start another thread if you wish to discuss this.=20

I agree that the fa=E7ade of security is worse than having no security. =
However, I don't think that there should be requirements for human =
notification/intervention in the specification--network devices do not =
typically have a per session user interface! IMHO, a "should" would be =
reasonable, but a "must" is too strong. This also becomes sticky when =
dealing with multiple streams that don't all have the same security =
attributes. Additionally, would you display the session as secure if it =
was negotiated with UNENCRYPTED_SRTP and UNAUTHENTICATED_SRTP?

Cheers,
Rick


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Rick,

ZRTP essentially uses this approach - it uses a normal m= media line 
that any UA can accept and avoids all these failure modes.

In the past I have proposed that other key management techniques could 
benefit from this as well by permitting the crypto and key-mgt 
attributes in normal m= lines, just as you discuss.

If the SRTP session is then established, a re-INVITE could be sent to 
change the m= line to indicate that resulting session is in fact secure.

I think any device performing this kind of function needs to provide an 
indication of the secure status of the call (hence a MUST), even if it 
is just a log or other non-real time indication.

Thanks,
Alan

Rick Porter wrote:

>Thanks for all the responses. I'm fascinated by the discussion, as I thought I would be when I asked the question. I raised my original question due to a situation I encountered in the lab. I have two phones as follows: 
>
>A - secure (SRTP, possibly TLS) 
>B - non-secure (no TLS or SRTP)
>
>When A calls B, B rejects the call because he is not configured to accept a secure call (this phone requires TLS signaling to do SRTP). Per the SDES draft (Section 5.1.2, top of page 10), this is the correct behavior. I found the result (B didn't even ring, and call was not completed) to be rather unsatisfying. This sounds like a known issue with the current draft.
>
>I hate to throw more proposals in the mix, but I have a couple thoughts on "simple" changes to the current draft to make a more deployable solution. I'm trying to take a simple, pragmatic approach that will get this deployed ASAP. It sounds like the conditional SDP is not deployed, so I do not want to depend on that. Take these proposals with as many shakers of salt as you see appropriate.
>
>1) Use the same media stream. No special transport. Crypto is just another "optional" attribute of the media description.
>
>2) Add an "a=encryption" attribute. 
>
>I believe those suggestions would result in the following behavior for the above scenario.
>
>B would accept the call, but not provide a crypto attribute in the 200/OK/SDP. Then, if A does not like the 200/OK/SDP without crypto he can choose to CANCEL the call or continue the non-secure call. Based on configuration, phone A may prompt the user to accept that it is a non-secure call. B already knows that he is not configured to do secure calls, so being non-secure is not news to him. 
>
>In an early draft by Ahmar Ghaffar at Snom, I liked the "a=encryption:<optional|mandatory>" attribute that could be included as part of the media description. If the encryption attribute is present, you know whether you can accept the call even if you do not share a common set of crypto algorithms/options. It could give the end receiving the INVITE the chance to reject the call early (e.g. 480) instead of sending the 200/OK (without crypto) and letting the other end cancel the call upon receipt of the 200/OK.
>
>I know the above scenario is not the only one, so I probably missed something...
>
>I believe Alan and Paul are correct with respect to not being concerned about a bid-down attack. If you have access to the signaling stream, then you have access to key material necessary to decrypt each and every RTP packet. No need to bid it down crypto if you can decrypt the RTP packets anyway. Until you do some sort of end-to-end public key exchange (e.g. ZRTP), then you don't have end-to-end security. There are many technical and political reasons why SRTP is done on a hop-by-hop basis--please start another thread if you wish to discuss this. 
>
>I agree that the façade of security is worse than having no security. However, I don't think that there should be requirements for human notification/intervention in the specification--network devices do not typically have a per session user interface! IMHO, a "should" would be reasonable, but a "must" is too strong. This also becomes sticky when dealing with multiple streams that don't all have the same security attributes. Additionally, would you display the session as secure if it was negotiated with UNENCRYPTED_SRTP and UNAUTHENTICATED_SRTP?
>
>Cheers,
>Rick
>
>
>_______________________________________________
>mmusic mailing list
>mmusic@ietf.org
>https://www1.ietf.org/mailman/listinfo/mmusic
>
>
>  
>


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From mmusic-bounces@ietf.org Wed Apr 05 12:54:55 2006
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Subject: RE: [MMUSIC] Optional SRTP?
Date: Wed, 5 Apr 2006 12:54:51 -0400
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Hi Alan,=20

Why would you send a re-INVITE if SRTP is established? I thought =
everything you need to know is in the original INVITE/OK SDP pair.=20

I don't have any strong feelings regarding indication requirements. I =
was afraid that you wanted real-time, user-interface indication which is =
not practical for multi-user network devices (e.g. b2bua, =
media-gateway). I agree that good UA's will provide the best indication =
they can to their user. =20

Thanks,
Rick

-----Original Message-----
From: Alan Johnston [mailto:alan@sipstation.com]=20
Sent: Wednesday, April 05, 2006 9:51 AM
To: Rick Porter
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] Optional SRTP?

Rick,

ZRTP essentially uses this approach - it uses a normal m=3D media line=20
that any UA can accept and avoids all these failure modes.

In the past I have proposed that other key management techniques could=20
benefit from this as well by permitting the crypto and key-mgt=20
attributes in normal m=3D lines, just as you discuss.

If the SRTP session is then established, a re-INVITE could be sent to=20
change the m=3D line to indicate that resulting session is in fact =
secure.

I think any device performing this kind of function needs to provide an=20
indication of the secure status of the call (hence a MUST), even if it=20
is just a log or other non-real time indication.

Thanks,
Alan

Rick Porter wrote:

>Thanks for all the responses. I'm fascinated by the discussion, as I =
thought I would be when I asked the question. I raised my original =
question due to a situation I encountered in the lab. I have two phones =
as follows:=20
>
>A - secure (SRTP, possibly TLS)=20
>B - non-secure (no TLS or SRTP)
>
>When A calls B, B rejects the call because he is not configured to =
accept a secure call (this phone requires TLS signaling to do SRTP). Per =
the SDES draft (Section 5.1.2, top of page 10), this is the correct =
behavior. I found the result (B didn't even ring, and call was not =
completed) to be rather unsatisfying. This sounds like a known issue =
with the current draft.
>
>I hate to throw more proposals in the mix, but I have a couple thoughts =
on "simple" changes to the current draft to make a more deployable =
solution. I'm trying to take a simple, pragmatic approach that will get =
this deployed ASAP. It sounds like the conditional SDP is not deployed, =
so I do not want to depend on that. Take these proposals with as many =
shakers of salt as you see appropriate.
>
>1) Use the same media stream. No special transport. Crypto is just =
another "optional" attribute of the media description.
>
>2) Add an "a=3Dencryption" attribute.=20
>
>I believe those suggestions would result in the following behavior for =
the above scenario.
>
>B would accept the call, but not provide a crypto attribute in the =
200/OK/SDP. Then, if A does not like the 200/OK/SDP without crypto he =
can choose to CANCEL the call or continue the non-secure call. Based on =
configuration, phone A may prompt the user to accept that it is a =
non-secure call. B already knows that he is not configured to do secure =
calls, so being non-secure is not news to him.=20
>
>In an early draft by Ahmar Ghaffar at Snom, I liked the =
"a=3Dencryption:<optional|mandatory>" attribute that could be included =
as part of the media description. If the encryption attribute is =
present, you know whether you can accept the call even if you do not =
share a common set of crypto algorithms/options. It could give the end =
receiving the INVITE the chance to reject the call early (e.g. 480) =
instead of sending the 200/OK (without crypto) and letting the other end =
cancel the call upon receipt of the 200/OK.
>
>I know the above scenario is not the only one, so I probably missed =
something...
>
>I believe Alan and Paul are correct with respect to not being concerned =
about a bid-down attack. If you have access to the signaling stream, =
then you have access to key material necessary to decrypt each and every =
RTP packet. No need to bid it down crypto if you can decrypt the RTP =
packets anyway. Until you do some sort of end-to-end public key exchange =
(e.g. ZRTP), then you don't have end-to-end security. There are many =
technical and political reasons why SRTP is done on a hop-by-hop =
basis--please start another thread if you wish to discuss this.=20
>
>I agree that the fa=E7ade of security is worse than having no security. =
However, I don't think that there should be requirements for human =
notification/intervention in the specification--network devices do not =
typically have a per session user interface! IMHO, a "should" would be =
reasonable, but a "must" is too strong. This also becomes sticky when =
dealing with multiple streams that don't all have the same security =
attributes. Additionally, would you display the session as secure if it =
was negotiated with UNENCRYPTED_SRTP and UNAUTHENTICATED_SRTP?
>
>Cheers,
>Rick
>
>
>_______________________________________________
>mmusic mailing list
>mmusic@ietf.org
>https://www1.ietf.org/mailman/listinfo/mmusic
>
>
> =20
>


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



From mmusic-bounces@ietf.org Wed Apr 05 15:03:23 2006
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Subject: RE: [MMUSIC] Optional SRTP?
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From: "Wayne Mills \(wmills\)" <wmills@cisco.com>
To: "Rick Porter" <rporter@covergence.com>, <mmusic@ietf.org>
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If I understand the ZRTP proposal correctly ....

The initial media connection is made with RTP, then the endpoints =
"upgrade" to SRTP using the media-inband key exchange suggested by ZRTP, =
thus SRTP is negotiated outside the call signaling plane.  Re-INVITE (by =
correcting SDP after the fact) informs the signaling plane that it =
happened.

Regards,
Wayne

> -----Original Message-----
> From: Rick Porter [mailto:rporter@covergence.com]=20
> Sent: Wednesday, April 05, 2006 12:55 PM
> To: mmusic@ietf.org
> Subject: RE: [MMUSIC] Optional SRTP?
>=20
> Hi Alan,=20
>=20
> Why would you send a re-INVITE if SRTP is established? I=20
> thought everything you need to know is in the original=20
> INVITE/OK SDP pair.=20
>=20
> I don't have any strong feelings regarding indication=20
> requirements. I was afraid that you wanted real-time,=20
> user-interface indication which is not practical for=20
> multi-user network devices (e.g. b2bua, media-gateway). I=20
> agree that good UA's will provide the best indication they=20
> can to their user. =20
>=20
> Thanks,
> Rick
>=20
> -----Original Message-----
> From: Alan Johnston [mailto:alan@sipstation.com]
> Sent: Wednesday, April 05, 2006 9:51 AM
> To: Rick Porter
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] Optional SRTP?
>=20
> Rick,
>=20
> ZRTP essentially uses this approach - it uses a normal m=3D=20
> media line that any UA can accept and avoids all these failure modes.
>=20
> In the past I have proposed that other key management=20
> techniques could benefit from this as well by permitting the=20
> crypto and key-mgt attributes in normal m=3D lines, just as you =
discuss.
>=20
> If the SRTP session is then established, a re-INVITE could be=20
> sent to change the m=3D line to indicate that resulting session=20
> is in fact secure.
>=20
> I think any device performing this kind of function needs to=20
> provide an indication of the secure status of the call (hence=20
> a MUST), even if it is just a log or other non-real time indication.
>=20
> Thanks,
> Alan
>=20
> Rick Porter wrote:
>=20
> >Thanks for all the responses. I'm fascinated by the=20
> discussion, as I thought I would be when I asked the=20
> question. I raised my original question due to a situation I=20
> encountered in the lab. I have two phones as follows:=20
> >
> >A - secure (SRTP, possibly TLS)
> >B - non-secure (no TLS or SRTP)
> >
> >When A calls B, B rejects the call because he is not=20
> configured to accept a secure call (this phone requires TLS=20
> signaling to do SRTP). Per the SDES draft (Section 5.1.2, top=20
> of page 10), this is the correct behavior. I found the result=20
> (B didn't even ring, and call was not completed) to be rather=20
> unsatisfying. This sounds like a known issue with the current draft.
> >
> >I hate to throw more proposals in the mix, but I have a=20
> couple thoughts on "simple" changes to the current draft to=20
> make a more deployable solution. I'm trying to take a simple,=20
> pragmatic approach that will get this deployed ASAP. It=20
> sounds like the conditional SDP is not deployed, so I do not=20
> want to depend on that. Take these proposals with as many=20
> shakers of salt as you see appropriate.
> >
> >1) Use the same media stream. No special transport. Crypto=20
> is just another "optional" attribute of the media description.
> >
> >2) Add an "a=3Dencryption" attribute.=20
> >
> >I believe those suggestions would result in the following=20
> behavior for the above scenario.
> >
> >B would accept the call, but not provide a crypto attribute=20
> in the 200/OK/SDP. Then, if A does not like the 200/OK/SDP=20
> without crypto he can choose to CANCEL the call or continue=20
> the non-secure call. Based on configuration, phone A may=20
> prompt the user to accept that it is a non-secure call. B=20
> already knows that he is not configured to do secure calls,=20
> so being non-secure is not news to him.=20
> >
> >In an early draft by Ahmar Ghaffar at Snom, I liked the=20
> "a=3Dencryption:<optional|mandatory>" attribute that could be=20
> included as part of the media description. If the encryption=20
> attribute is present, you know whether you can accept the=20
> call even if you do not share a common set of crypto=20
> algorithms/options. It could give the end receiving the=20
> INVITE the chance to reject the call early (e.g. 480) instead=20
> of sending the 200/OK (without crypto) and letting the other=20
> end cancel the call upon receipt of the 200/OK.
> >
> >I know the above scenario is not the only one, so I probably=20
> missed something...
> >
> >I believe Alan and Paul are correct with respect to not=20
> being concerned about a bid-down attack. If you have access=20
> to the signaling stream, then you have access to key material=20
> necessary to decrypt each and every RTP packet. No need to=20
> bid it down crypto if you can decrypt the RTP packets anyway.=20
> Until you do some sort of end-to-end public key exchange=20
> (e.g. ZRTP), then you don't have end-to-end security. There=20
> are many technical and political reasons why SRTP is done on=20
> a hop-by-hop basis--please start another thread if you wish=20
> to discuss this.=20
> >
> >I agree that the fa=E7ade of security is worse than having no=20
> security. However, I don't think that there should be=20
> requirements for human notification/intervention in the=20
> specification--network devices do not typically have a per=20
> session user interface! IMHO, a "should" would be reasonable,=20
> but a "must" is too strong. This also becomes sticky when=20
> dealing with multiple streams that don't all have the same=20
> security attributes. Additionally, would you display the=20
> session as secure if it was negotiated with UNENCRYPTED_SRTP=20
> and UNAUTHENTICATED_SRTP?
> >
> >Cheers,
> >Rick
> >
> >
> >_______________________________________________
> >mmusic mailing list
> >mmusic@ietf.org
> >https://www1.ietf.org/mailman/listinfo/mmusic
> >
> >
> > =20
> >
>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
>=20

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



From mmusic-bounces@ietf.org Wed Apr 05 15:15:24 2006
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To: "'Colin Perkins'" <csp@csperkins.org>,
	"'Rick Porter'" <rporter@covergence.com>
Subject: RE: [MMUSIC] Optional SRTP?
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Hi Colin,
Was there a consensus to determine a single approach?  I didn't hear it. (I
was on audio-cast, and it was hard to hear)  

Or is the consensus to improve the SDP syntax so that a common mechanism
could be chosen more easily during offer/answer? (if a common one exists)
Assuming negotiation even occurs in SDP.

Or by choosing a "single" approach do you mean choosing signaling handshake
vs. bearer (ZRTP, RTP/DTLS).

-hadriel


> -----Original Message-----
> From: Colin Perkins [mailto:csp@csperkins.org]
> Sent: Thursday, March 30, 2006 12:56 PM
> To: Rick Porter
> Cc: Alan Johnston; IETF MMUSIC WG
> Subject: Re: [MMUSIC] Optional SRTP?
> 
> Rick,
> 
> It's important to note that there was a lot of discussion on this
> subject in the MMUSIC and AVT working groups, and the RAI area
> meeting, at the IETF last week. The zrtp draft is one of many
> proposals, and it is by no means clear that there is consensus on any
> single approach.
> 
> Colin


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Subject: Re: [MMUSIC] Optional SRTP?
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Hadriel,

It's clear that a single approach is desirable, rather than multiple  
incompatible solutions, but I don't see any consensus on how to  
proceed yet.

Colin



On 5 Apr 2006, at 20:15, Hadriel Kaplan wrote:
> Hi Colin,
> Was there a consensus to determine a single approach?  I didn't  
> hear it. (I
> was on audio-cast, and it was hard to hear)
>
> Or is the consensus to improve the SDP syntax so that a common  
> mechanism
> could be chosen more easily during offer/answer? (if a common one  
> exists)
> Assuming negotiation even occurs in SDP.
>
> Or by choosing a "single" approach do you mean choosing signaling  
> handshake
> vs. bearer (ZRTP, RTP/DTLS).
>
> -hadriel
>
>
>> -----Original Message-----
>> From: Colin Perkins [mailto:csp@csperkins.org]
>> Sent: Thursday, March 30, 2006 12:56 PM
>> To: Rick Porter
>> Cc: Alan Johnston; IETF MMUSIC WG
>> Subject: Re: [MMUSIC] Optional SRTP?
>>
>> Rick,
>>
>> It's important to note that there was a lot of discussion on this
>> subject in the MMUSIC and AVT working groups, and the RAI area
>> meeting, at the IETF last week. The zrtp draft is one of many
>> proposals, and it is by no means clear that there is consensus on any
>> single approach.
>>
>> Colin
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic


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Subject: RE: [MMUSIC] Optional SRTP?
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I could not see why a re-INVITE was necessary using the scheme outlined
in the Security Descriptions draft (with possible modification).=20

I'm still a ZRTP novice, but I can understand the re-INVITE when doing
ZRTP. However, the re-INVITE to keep signaling informed raises a red
flag about layering. My instinct tells me that if it is important to the
signaling, then the signaling should be driving the train. I still have
many ZRTP discussions to catch up on...=20

Thanks,
Rick

-----Original Message-----
From: Wayne Mills (wmills) [mailto:wmills@cisco.com]=20
Sent: Wednesday, April 05, 2006 1:04 PM
To: Rick Porter; mmusic@ietf.org
Subject: RE: [MMUSIC] Optional SRTP?

If I understand the ZRTP proposal correctly ....

The initial media connection is made with RTP, then the endpoints
"upgrade" to SRTP using the media-inband key exchange suggested by ZRTP,
thus SRTP is negotiated outside the call signaling plane.  Re-INVITE (by
correcting SDP after the fact) informs the signaling plane that it
happened.

Regards,
Wayne


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Subject: Re: [MMUSIC] Optional SRTP?
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Rick, all
   We've been promised a list to discuss these topics under the  
auspices of the RAI.  I'd prefer to have this discussion there  
because it is more appropriate than MMUSIC.  Note that zrtp and the  
other proposals were not submitted to mmusic nor discussed at the  
recent wg meeting.

Mark
On Apr 6, 2006, at 5:36 AM, Rick Porter wrote:

> I could not see why a re-INVITE was necessary using the scheme  
> outlined
> in the Security Descriptions draft (with possible modification).
>
> I'm still a ZRTP novice, but I can understand the re-INVITE when doing
> ZRTP. However, the re-INVITE to keep signaling informed raises a red
> flag about layering. My instinct tells me that if it is important  
> to the
> signaling, then the signaling should be driving the train. I still  
> have
> many ZRTP discussions to catch up on...
>
> Thanks,
> Rick
>
> -----Original Message-----
> From: Wayne Mills (wmills) [mailto:wmills@cisco.com]
> Sent: Wednesday, April 05, 2006 1:04 PM
> To: Rick Porter; mmusic@ietf.org
> Subject: RE: [MMUSIC] Optional SRTP?
>
> If I understand the ZRTP proposal correctly ....
>
> The initial media connection is made with RTP, then the endpoints
> "upgrade" to SRTP using the media-inband key exchange suggested by  
> ZRTP,
> thus SRTP is negotiated outside the call signaling plane.  Re- 
> INVITE (by
> correcting SDP after the fact) informs the signaling plane that it
> happened.
>
> Regards,
> Wayne
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic

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It has been set up but is not active yet. You can subscribe at
http://www.imc.org/ietf-rtpsec

On 4/4/06 11:05 AM, "Hadriel Kaplan" <HKaplan@acmepacket.com> wrote:

> I apologize if this has been answered previously, but is there a mailing =
list
> for the multiple-choice SRTP (and ZRTP) discussion?  I thought I heard th=
ere
> would be when I listened to the audio-cast of the Dallas meeting, but I
> haven=B9t been able to find it.
> Thanks!
> -hadriel
> =20
>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
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<BR>
</SPAN></FONT><BLOCKQUOTE><FONT SIZE=3D"4"><FONT FACE=3D"Arial"><SPAN STYLE=3D'fo=
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I thought I heard there would be when I listened to the audio-cast of the Da=
llas meeting, but I haven&#8217;t been able to find it. <BR>
Thanks!<BR>
-hadriel<BR>
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From mmusic-bounces@ietf.org Fri Apr 07 08:47:29 2006
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Subject: [MMUSIC] Re: File tranfer in SIP
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Hi Gonzalo,

moving this to MMUSIC:

> I agree that we want to provide the same information as existing file 
> description mechanisms provide. In particular, we have looked into MIME. 
> We will make sure that we go thru MIME again to see if we have forgotten 
> something (e.g., creation of the file timestamp).

Sounds good.

> I have looked into FLUTE, and it provides the same information as we 
> provide in the draft. From:
> http://www.ietf.org/internet-drafts/draft-ietf-rmt-flute-revised-01.txt
> 
>    -  Name, Identification and Location of file (specified by the URI)
>    -  MIME media type of file
>    -  Size of file
>    -  Encoding of file
>    -  Message digest of file
> 
> In any case, this information needs to be encoded in the SDP way, and 
> therefore, we should use 'a' lines for that.

I am still missing a last modification timestamp here.  Regarding the
encoding, I suppose we will have to worry about how to put this into
SDP.  This brings up -- again -- the discussion of whether to include
some "blob" of, from an SDP perspective, opaque metadata (e.g., b64
encoded zip version of the FLUTE FDT, just to give some weird example)
or whether to make these attributes explicit.

If we make the attributes explicit, this is essentially a more general
aspect of describing the purpuse of a media stream and would not be
restricted to file transfer.  We have some informal hints for audio
and video already and this appears conceptually similar in that it
identifies the purpose and helps the answerer to decide whether to
accept it or not.

Keeping this in mind, it would be extremely useful, not to invent
independent sets of attributes for file transfer, MSRP sessions,
audio, video, whiteboard, and so so.

I clearly _do_not_ want to end up in an XCON-style discussion about
metadata and the like, but it may be worthwhile given some
consideration to future evolution of this stuff and compatibility
with existing attributes.

>> I am also concerned about motivating the use of inline images for
>> icons.  I can hardly see why you would need an icon to decide
>> whether or not to accept a file transfer.
> 
> A lot of commercial IM systems support it, and users find it very 
> useful. SIP-based systems need to be able to provide a similar user 
> experience if we do not want people to start doing things is proprietary 
> ways.

It's your call in the end of the day.  Often, icons are created based
upon the filename extension -- and I suspect that the filename is going
to be much more relevant that the icon in deciding whether or not to
accept it.  Icons are not worth much and only increase volume.  But if
we decide that it is useful:

You must decide how much you worry about the SIP signaling architecture.
Given that the INFO method is "damned" (which I fully support), I do not
believe that turning parts of SDP into a transport is a particularly
smart idea.  It clearly does not ad efficiency.  If you follow
traditional SIPPING thinking, you will need to separate the desired
effect from the mechanism to achieve it.  There may be better ways.

I am simply worried to open a door that we better leave closed.
Would you want to limit the media type or the size of an icon?
If so, what would be the right value?  Would you negotiate it and
come up with an "embedded contents too large" status code?

Also, remember, that any SDP contents potentially wants to be signed.

>> Performing file selection based upon only the hash may be nice for
>> the receiver but it may be counter-intuitive and difficult to implement
>> for the sender since it would require the sender to keep a hash of
>> all its local files which if there are so many that on-the-fly
>> calculation may not be a good idea.
> 
> The user selects the file to send, the application hashes it on the fly, 
> and sends the hash to the receiver. I just calculated the MD5 and SHA1 
> hashes of a big file (165 MB) on a non-very-powerful laptop, and it took 
> 8 and 10 seconds respectively. If you send pictures, which are much 
> smaller, the hash calculation would be very fast.

I think I got somewhat confused that I constructed a close to impossible
scenario.  For the sender it is indeed trivial if the sending user
selects the file.  Of course, the filename should be included because
otherwise the receiver does not easily find out whether or not it has
a copy of the file already.  If the receiving users wants to initiate
a file transfer, of course, it does not have a checksum at hand to
include.

>> The draft refers to MSRP for connection setup and teardown but the
>> examples only show trivia.  The most interesting thing would be
>> to show how sending, say, 4 files in each direction works and
>> how the SDP descriptions grow (and if there will be a chance to
>> shrink them again).
> 
> We will include a note pointing out that 'm' lines can be reused after a 
> media stream is terminated. That is regular SDP behavior.

True.  But an example would still be nice (you know, most people code
by example rather than by spec and we don't want them to screw up :-)
After all, I would be surprised that re-INVITEs or UPDATEs with media
stream addition, removal, addition, etc. have been common practice so
far.  But I may be wrong.

>>  Also, how do simultaneous initiations in both
>> directions work (concerning counting the many m= lines)?
> 
> I do not see any issue with having simultaneous media streams. It is 
> again regular SDP behavior.

Well, I assume that if I send you a second offer within a session
and you also send me one at the same time, we will have to do some
conflict resolution since both of us will use the (n+1)st m-line
for potentially different purposes.  Since this has probably not
received much implementation attention in the past (and happened
rather rarely in practice), it have not been noticed.  But it may
be specified and just did not find this upon a quick search.

Cheers,
Joerg

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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		: Forward Error Correction Grouping Semantics in Session Description Protocol
	Author(s)	: A. Li, et al.
	Filename	: draft-ietf-mmusic-fec-grouping-04.txt
	Pages		: 7
	Date		: 2006-4-10
	
This document defines the semantics that allows for grouping of
   forward error correction (FEC) streams with the protected payload
   streams in Session Description Protocol (SDP). The semantics defined
   in this document is to be used with Grouping of Media Lines in the
   Session Description Protocol (RFC 3388) to group together "m" lines
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From mmusic-bounces@ietf.org Tue Apr 11 03:08:01 2006
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Hi,

I have incorporated all the WGLC comments to the SDP Content Attribute 
draft (given comments were mainly editorial). You can find the updated, 
and submitted, draft from:

http://users.piuha.net/jantzgu/temp/draft-ietf-mmusic-sdp-media-content-03.txt

Thanks to Tom Taylor once again for a thorough review!


-- 
Jani

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Hello, is it any work in progress regarding Video transcoding services
using SIP and SDP.

I'm looking something like RFC 4117 for video.

=20

Thanks in advance

=20

Osher

=20


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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Hello, is it any work in progress regarding Video
transcoding services using SIP and SDP.<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'>I&#8217;m looking something like RFC =
</span></font>4117 for
video.<o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><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 in advance<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'>Osher<o:p></o:p></span></font></p>

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

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From mmusic-bounces@ietf.org Tue Apr 11 10:13:26 2006
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On Tue, 11 Apr 2006, Osher Hmelnizky wrote:

> Hello, is it any work in progress regarding Video transcoding services
> using SIP and SDP.
>
> I'm looking something like RFC 4117 for video.

I am looking for the same, so far all the systems I have run into are
proprietary. In talking with vendors they are saying it is because of lack
of standard, but that could just be an easy out.

-Nathan

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Hi,

In IETF 64, we presented requirements for seamless session mobility and
one of the comments was that more use cases of session mobility would
help in understanding the motivation and importance of seamless session
mobility. So we listed a number of use cases of session mobility,
categorized according to the merits for the
users in draft-komiya-mmusic-session-mobility-usecases-00.txt.

Unfortunately our presentation on the draft in IETF65 was cancelled
because the WG meeting ran out of time.
Below are the presentation slides.
http://www3.ietf.org/proceedings/06mar/slides/mmusic-9.pdf

We would appreciate if you take a look at the slides or the draft and
provide us some comments.

Here are our questions for the group:
- What do the working group think of the use cases?
- Is there any missing use case?
- Do the group think the use cases should be supported?

Thanks.

Eunsoo



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In reading RFC 3264 it states in section 6.1:

"The connection address and port in the answer indicate the address where
the answerer wishes to receive the media ... This address and port MUST be
present even for sendonly streams..."

My interpretation of this and other text in the document is that the
address/port information in an SDP offer / answer always refers only to
where media wants to be receive, but NEVER to where media will be
sourced/sent from. 

Is my understanding correct or am I confused?

Thanks in advance.

-Steve


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Subject: RE: [MMUSIC] use cases of seamless session mobility
Date: Tue, 11 Apr 2006 13:08:02 -0400
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From: "Montpetit Marie-Jose-MGIF0300" <mmontpetit@motorola.com>
To: "Eunsoo Shim" <eunsoo@research.panasonic.com>, "mmusic" <mmusic@ietf.org>
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=20
Is this really in the realm of the IETF? Session mobility is more like a
"system capability" the type of work that is done in ETSI, CableLabs
etc. Are you suggesting current IETF protocols could not suport your use
cases?

/mjm
-----Original Message-----
From: Eunsoo Shim [mailto:eunsoo@research.panasonic.com]=20
Sent: Tuesday, April 11, 2006 10:39 AM
To: 'mmusic'
Cc: jo@netlab.tkk.fi; csp@csperkins.org
Subject: [MMUSIC] use cases of seamless session mobility

Hi,

In IETF 64, we presented requirements for seamless session mobility and
one of the comments was that more use cases of session mobility would
help in understanding the motivation and importance of seamless session
mobility. So we listed a number of use cases of session mobility,
categorized according to the merits for the users in
draft-komiya-mmusic-session-mobility-usecases-00.txt.

Unfortunately our presentation on the draft in IETF65 was cancelled
because the WG meeting ran out of time.
Below are the presentation slides.
http://www3.ietf.org/proceedings/06mar/slides/mmusic-9.pdf

We would appreciate if you take a look at the slides or the draft and
provide us some comments.

Here are our questions for the group:
- What do the working group think of the use cases?
- Is there any missing use case?
- Do the group think the use cases should be supported?

Thanks.

Eunsoo



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

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From mmusic-bounces@ietf.org Tue Apr 11 13:47:28 2006
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From: "Dan Wing" <dwing@cisco.com>
To: <steven.d.whitehead@verizon.com>
Subject: RE: [MMUSIC] SDP clarification question
Date: Tue, 11 Apr 2006 10:47:24 -0700
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Steven,

You're correct, except that in order for SBCs or ICE to work, the same UDP
port has to be used to send and to receive RTP.

See draft-wing-behave-symmetric-rtprtcp-01.txt, which is in the RFC Editor's
queue.

-d
 

> -----Original Message-----
> From: steven.d.whitehead@verizon.com 
> [mailto:steven.d.whitehead@verizon.com] 
> Sent: Tuesday, April 11, 2006 9:58 AM
> To: mmusic@ietf.org
> Subject: [MMUSIC] SDP clarification question
> 
> In reading RFC 3264 it states in section 6.1:
> 
> "The connection address and port in the answer indicate the 
> address where
> the answerer wishes to receive the media ... This address and 
> port MUST be
> present even for sendonly streams..."
> 
> My interpretation of this and other text in the document is that the
> address/port information in an SDP offer / answer always 
> refers only to
> where media wants to be receive, but NEVER to where media will be
> sourced/sent from. 
> 
> Is my understanding correct or am I confused?
> 
> Thanks in advance.
> 
> -Steve
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic

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From: Mark Baugher <mbaugher@cisco.com>
Subject: Re: [MMUSIC] SDP clarification question
Date: Tue, 11 Apr 2006 12:18:04 -0700
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Dan
   Seems that we have been mulling over this topic for years.  Let me  
add to the muddle by claiming that the use of the SDP name "sendonly"  
can be defined in different ways by different application protocols  
such as SIP or SAP.

Mark
On Apr 11, 2006, at 10:47 AM, Dan Wing wrote:

> Steven,
>
> You're correct, except that in order for SBCs or ICE to work, the  
> same UDP
> port has to be used to send and to receive RTP.
>
> See draft-wing-behave-symmetric-rtprtcp-01.txt, which is in the RFC  
> Editor's
> queue.
>
> -d
>
>
>> -----Original Message-----
>> From: steven.d.whitehead@verizon.com
>> [mailto:steven.d.whitehead@verizon.com]
>> Sent: Tuesday, April 11, 2006 9:58 AM
>> To: mmusic@ietf.org
>> Subject: [MMUSIC] SDP clarification question
>>
>> In reading RFC 3264 it states in section 6.1:
>>
>> "The connection address and port in the answer indicate the
>> address where
>> the answerer wishes to receive the media ... This address and
>> port MUST be
>> present even for sendonly streams..."
>>
>> My interpretation of this and other text in the document is that the
>> address/port information in an SDP offer / answer always
>> refers only to
>> where media wants to be receive, but NEVER to where media will be
>> sourced/sent from.
>>
>> Is my understanding correct or am I confused?
>>
>> Thanks in advance.
>>
>> -Steve
>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www1.ietf.org/mailman/listinfo/mmusic
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic

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Subject: [MMUSIC] Working group last call: draft-ietf-mmusic-ice-08.txt 
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This is to announce a working group last call on ICE <draft-ietf- 
mmusic-ice-08.txt>. Please send any final comments on this draft to  
the mailing list before 5th May 2006. If no substantive issues are  
raised by that time, we'll request the IESG consider the draft for  
publication as a proposed standard RFC.

Due to the length and complexity of the draft, we are especially  
soliciting comments and reports from those who have implemented from  
the draft (either on the list, or privately to the chairs).

Colin



On 31 Mar 2006, at 00:50, 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		: Interactive Connectivity Establishment (ICE): A  
> Methodology for Network Address Translator (NAT) Traversal for  
> Offer/Answer Protocols
> 	Author(s)	: J. Rosenberg
> 	Filename	: draft-ietf-mmusic-ice-08.txt
> 	Pages		: 96
> 	Date		: 2006-3-30
> 	
> This document describes a protocol for Network Address Translator
>    (NAT) traversal for multimedia session signaling protocols based on
>    the offer/answer model, such as the Session Initiation Protocol
>    (SIP).  This protocol is called Interactive Connectivity
>    Establishment (ICE).  ICE makes use of the Simple Traversal of UDP
>    through NAT (STUN), applying its binding discovery, connectivity
>    check and relay usages.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-ice-08.txt


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From: Colin Perkins <csp@csperkins.org>
Subject: Re: [MMUSIC] Working group last call: draft-ietf-mmusic-ice-08.txt 
Date: Tue, 11 Apr 2006 20:38:30 +0100
To: IETF MMUSIC WG <mmusic@ietf.org>
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On 11 Apr 2006, at 20:29, Colin Perkins wrote:
> This is to announce a working group last call on ICE <draft-ietf- 
> mmusic-ice-08.txt>. Please send any final comments on this draft to  
> the mailing list before 5th May 2006. If no substantive issues are  
> raised by that time, we'll request the IESG consider the draft for  
> publication as a proposed standard RFC.
>
> Due to the length and complexity of the draft, we are especially  
> soliciting comments and reports from those who have implemented  
> from the draft (either on the list, or privately to the chairs).

Note that the IETF has been notified of potential IPR relating to  
this draft and owned by Tandberg Telecom AS. The licensing  
declaration states:

	The Patent Holder is prepare to grant - on the basis of reciprocity  
for the
	above IETF specification - a licence to an unrestricted number of  
applicants
	on a worldwide, non-discriminatory basis and on reasonable term and  
conditions
	to make, use and sell implementations of the above IETF specification.
	Negotiations are left to the parties concerned, and are performed  
outside
	the IETF.

	As used herein, the word "reciprocity" means that the Patent Holder  
shall
	only be required to license any prospective licencee if such  
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For full details, see https://datatracker.ietf.org/public/ 
ipr_search.cgi?option=document_search&id_document_tag=11119

Colin

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I too am pleased to see work going on to determine how to deal with the
SIP-RTSP relationship and acknowledge the efforts of the authors to
provide multiple options for consideration.   As is evident from my
comments below I do not believe that one of the models presented can
possibly meet the needs of the application.=20
=20
I think it is very important to create a simple architectural rendering
whereby a session and a stream are viewed as different architectural
objects with different needs.  SIP was created for the puprose of
establishing a session and the session may in turn support some number
of media streams.  The streams however are transitive - they may come
and go while the session persists.  There seems to be an unstated
assumption in some of the discussion points in the document that SIP is
good for stream control.   This is not its purpose and not what it is
good for.  =20
=20
There are many network elements which may be interested in or
participate in establishing or managing a session.  SIP proxies, B2BUAs,
p-CSCF, s-CSCF, ....  For the most part streams are endpoint to endpoint
(yes, there are exceptions but the hierarchy should never approach that
of the session control network). As such, stream control should be
endpoint to endpoint.  This is well accepted for VoIP where RTCP for
example is point to point regardless as to whether SIP, H.323, H.248 or
whatever was used to establish the session.   Can we imagine replacing
RTCP with some SIP extensions? (I'm sure someone has proposed this so
forgive my ignorance)
=20
As initially defined, RTSP encompasses both the session establishment as
well as the stream management.  This is as much a violation of the
architectural principles I implied above as trying to use SIP to do
both, but it has succeeded to this point because there has not been much
of a session framework in place in the networks upon which it has been
deployed.  Clearly the industry is moving in a direction whereby a more
complex session control plane is required and SIP has become accepted as
the session control protocol of choice. =20
=20
While the document does highlight signaling latency as a key issue for
the "all SIP" approach, I don't think it truly conveys the magnitude of
the problem nor does it fully acknowledge the other performance issues
associated with using SIP for streaming control.   Consider an IMS
network scenario whereby the p-CSCF and the s-CSCF are separated by some
1000 miles.  Imagine signaling a trick play command across the access
network - through the p-CSCF - across several more network hops to the
s-CSCF - on to the video AS and finally across a few more network hops
to the video source.   Ok, latency is clearly a problem, but consider
performance also.  If 50K subscribers are viewing material, what is the
hit to the p-CSCF, s-CSCF and video AS to process trick play commands?
Suddenly, the session control plane components must scale to meet the
needs of the media control plane.  The problem can be expressed in
non-IMS terms using SIP proxies, B2BUAs, etc. as well.
=20
In summary, the SIP + RTSP coexistence model is the only scalable option
presented and maps very well to a domain architecture model whereby
sessions and media streams are related but separate objects with
different needs. =20
=20
A few other minor comments:
- The scenarios covered in section 3.4 - Use Cases should be worded more
as simple scenario descriptions rather than positioning a solution
option
- In scenario one, it should be noted that TV and VoIP are two separate
sessions and that the choice of protocols which are used to implement
them is irrelevant in defining an interaction between them.  That is,
there must be an application level entity which is aware of both
sessions and the need for the interaction between. This application
level entity performs the logic of pausing one stream in order to
present another.  There is nothing inherent to SIP which enables the
protocol layer to do this for free or enables the network to do it for
the user.=20
- Section 6 - solution options discusses a Channel ID and Opaque token
but does not define what either of these are.  Perhaps one of the
referenced documents discusses this, but I believe SDP can be used to
describe an RTSP session without requiring Opaque data of any kind.
That is, establishment of a video stream using SIP+SDP can include not
only the video stream, but also the RTSP stream.   I think this is kind
of what was intended here but it wasn't obvious.
=20
Regards,
David
=20
=20

------_=_NextPart_001_01C65DA6.B4173E03
Content-Type: text/html;
	charset="us-ascii"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<TITLE>Message</TITLE>

<META content=3D"MSHTML 6.00.2800.1528" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D859292216-11042006><FONT face=3DArial size=3D2>I too =
am pleased to=20
see work going on to determine how to deal with the SIP-RTSP =
relationship and=20
acknowledge the efforts of the authors to provide&nbsp;multiple =
options&nbsp;for=20
consideration.&nbsp;&nbsp;&nbsp;As is evident from my comments =
below&nbsp;I do=20
not believe that one of the models presented&nbsp;can possibly meet the =
needs of=20
the application. </FONT></SPAN></DIV>
<DIV><SPAN class=3D859292216-11042006><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D859292216-11042006><FONT face=3DArial size=3D2>I =
think it is very=20
important to create a simple architectural rendering whereby a session =
and a=20
stream are viewed as different architectural objects with different =
needs.&nbsp;=20
</FONT></SPAN><SPAN class=3D859292216-11042006><FONT face=3DArial =
size=3D2>SIP was=20
created for the puprose of establishing a session and the session may in =
turn=20
support some number of media streams.&nbsp; The streams however are =
transitive -=20
they may come and go while the session persists.&nbsp; There seems to be =
an=20
unstated assumption in some of the discussion points in the document =
that SIP is=20
good for stream control.&nbsp;&nbsp; This is not its purpose and not =
what it is=20
good for.&nbsp;&nbsp; </FONT></SPAN></DIV>
<DIV><SPAN class=3D859292216-11042006><FONT face=3DArial =
size=3D2></FONT></SPAN><SPAN=20
class=3D859292216-11042006><FONT face=3DArial =
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D859292216-11042006><FONT face=3DArial size=3D2>There =
are many=20
network elements which may be interested in or participate in =
establishing or=20
managing a session.&nbsp; SIP proxies, B2BUAs, p-CSCF, s-CSCF, =
....&nbsp; For=20
the most part streams are endpoint to endpoint (yes, there are =
exceptions but=20
the hierarchy should never approach that of the session control =
network). As=20
such, stream control should be endpoint to endpoint.&nbsp; This is well =
accepted=20
for VoIP where RTCP for example is point to point regardless as to =
whether SIP,=20
H.323, H.248 or whatever was used to establish the session.&nbsp;&nbsp; =
Can we=20
imagine replacing RTCP with some SIP extensions? (I'm sure someone has =
proposed=20
this so forgive my ignorance)</FONT></SPAN></DIV>
<DIV><SPAN class=3D859292216-11042006><FONT face=3DArial =
size=3D2></FONT></SPAN><SPAN=20
class=3D859292216-11042006><FONT face=3DArial =
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D859292216-11042006><FONT face=3DArial size=3D2>As =
initially=20
defined, RTSP encompasses both the session establishment as well as the =
stream=20
management.&nbsp; This is as much a violation of the architectural =
principles I=20
implied above as trying to use SIP to do both, but it has succeeded to =
this=20
point because there has not been much of a session framework in place in =
the=20
networks upon which it has been deployed.&nbsp;</FONT></SPAN><SPAN=20
class=3D859292216-11042006><FONT face=3DArial size=3D2>&nbsp;Clearly the =
industry is=20
moving in a direction whereby a more complex session control plane is =
required=20
and SIP has&nbsp;become accepted as the session control protocol of =
choice.=20
&nbsp;</FONT></SPAN></DIV>
<DIV><SPAN class=3D859292216-11042006><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D859292216-11042006><FONT face=3DArial size=3D2>While =
the document=20
does highlight signaling latency as a key issue for the "all SIP" =
approach, I=20
don't think it truly conveys the magnitude of the problem nor does it =
fully=20
acknowledge the other performance issues associated with using SIP for =
streaming=20
control.&nbsp;</FONT></SPAN><SPAN class=3D859292216-11042006><FONT =
face=3DArial=20
size=3D2>&nbsp; Consider an IMS network scenario whereby the p-CSCF and =
the s-CSCF=20
are separated by some 1000 miles.&nbsp; Imagine signaling a trick play =
command=20
across the access network - through the p-CSCF - across several more =
network=20
hops to the s-CSCF - on to the video AS&nbsp;and finally&nbsp;across a =
few more=20
network hops to the video source.&nbsp;&nbsp;&nbsp;Ok, latency is =
clearly a=20
problem, but consider performance also.&nbsp; If 50K subscribers are =
viewing=20
material, what is the hit to the p-CSCF, s-CSCF and video AS to process =
trick=20
play commands? Suddenly, the session control plane components must scale =
to meet=20
the needs of the media control plane.&nbsp; The problem can be expressed =
in=20
non-IMS terms using SIP proxies, B2BUAs, etc. as =
well.</FONT></SPAN></DIV>
<DIV><SPAN class=3D859292216-11042006><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D859292216-11042006><FONT face=3DArial size=3D2>In =
summary, the SIP=20
+ RTSP coexistence model is the only&nbsp;scalable option presented and =
maps=20
very well to a domain architecture model whereby sessions and media =
streams are=20
related but separate objects with different needs.&nbsp; =
</FONT></SPAN></DIV>
<DIV><SPAN class=3D859292216-11042006><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D859292216-11042006><FONT face=3DArial size=3D2>A few =
other minor=20
comments:</FONT></SPAN></DIV>
<DIV><SPAN class=3D859292216-11042006><FONT face=3DArial size=3D2>- The =
scenarios=20
covered in section 3.4 - Use Cases should be worded more as simple =
scenario=20
descriptions rather than positioning a solution =
option</FONT></SPAN></DIV>
<DIV><SPAN class=3D859292216-11042006><FONT face=3DArial size=3D2>- In =
scenario one,=20
it should be noted that TV and VoIP are two separate sessions and that =
the=20
choice of protocols which are used to implement them is irrelevant in =
defining=20
an interaction between them.&nbsp; That is, there must be an application =
level=20
entity which is aware of both sessions and the need for the interaction =
between.=20
This application level entity performs the logic of pausing one stream =
in order=20
to present another.&nbsp; There is nothing inherent to SIP which enables =
the=20
protocol layer to do this for free or enables the network to do it for =
the user.=20
</FONT></SPAN></DIV>
<DIV><SPAN class=3D859292216-11042006><FONT face=3DArial size=3D2>- =
Section 6 -=20
solution options discusses a Channel ID and Opaque token but does not =
define=20
what either of these are.&nbsp; Perhaps one of the referenced documents=20
discusses this, but I believe SDP can be used to describe an RTSP =
session=20
without requiring Opaque data of any kind.&nbsp; That is, establishment =
of a=20
video stream using SIP+SDP can include not only the video stream, but =
also the=20
RTSP stream.&nbsp;&nbsp; I think this is kind of what was intended here =
but it=20
wasn't obvious.</FONT></SPAN></DIV>
<DIV><SPAN class=3D859292216-11042006><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D859292216-11042006><FONT face=3DArial=20
size=3D2>Regards,</FONT></SPAN></DIV>
<DIV><SPAN class=3D859292216-11042006><FONT face=3DArial=20
size=3D2>David</FONT></SPAN></DIV>
<DIV><SPAN class=3D859292216-11042006><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><STRONG><FONT =
color=3D#800000></FONT></STRONG>&nbsp;</DIV></BODY></HTML>

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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		: The SDP (Session Description Protocol) Content Attribute
	Author(s)	: J. Hautakorpi, G. Camarillo
	Filename	: draft-ietf-mmusic-sdp-media-content-03.txt
	Pages		: 12
	Date		: 2006-4-11
	
This document defines a new Session Description Protocol (SDP) media-
level attribute, 'content'.  The 'content' attribute defines the
content of the media stream in more detailed level than the media
description line.  The sender of an SDP session description can
attach the 'content' attribute to one or more media streams.  The
receiving application can then treat each media stream differently
(e.g., show it on a big screen or small screen) based on its content.

A URL for this Internet-Draft is:
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Subject: [MMUSIC] RFC 4435 on A Framework for the Usage of Internet Media
	Guides (IMGs)
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A new Request for Comments is now available in online RFC libraries.

        
        RFC 4435

        Title:      A Framework for the Usage 
                    of Internet Media Guides (IMGs) 
        Author:     Y. Nomura, R. Walsh,
                    J-P. Luoma, H. Asaeda,
                    H. Schulzrinne
        Status:     Informational
        Date:       April 2006
        Mailbox:    nom@flab.fujitsu.co.jp, 
                    rod.walsh@nokia.com, 
                    juha-pekka.luoma@nokia.com, asaeda@wide.ad.jp, 
                    schulzrinne@cs.columbia.edu
        Pages:      22
        Characters: 51687
        Updates/Obsoletes/SeeAlso:   None

        I-D Tag:    draft-ietf-mmusic-img-framework-09.txt

        URL:        http://www.rfc-editor.org/rfc/rfc4435.txt

This document defines a framework for the delivery of Internet Media
Guides (IMGs).  An IMG is a structured collection of multimedia
session descriptions expressed using the Session Description Protocol
(SDP), SDPng, or some similar session description format.  This
document describes a generalized model for IMG delivery mechanisms,
the use of existing protocols, and the need for additional work to
create an IMG delivery infrastructure.  This memo provides information for
the Internet community.

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

INFORMATIONAL: This memo provides information for the Internet community. 
It does not specify an Internet standard of any kind. Distribution
of this memo is unlimited.

This announcement is sent to the IETF list and the RFC-DIST list.
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should be sent to IETF-REQUEST@IETF.ORG.  Requests to be
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specifically noted otherwise on the RFC itself, all RFCs are for
unlimited distribution.

Submissions for Requests for Comments should be sent to
RFC-EDITOR@RFC-EDITOR.ORG.  Please consult RFC 2223, Instructions to RFC
Authors, for further information.


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

...



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Hi,

Is there any place where I can find SIP call flow examples that contain
PRACK, UPDATE, SigComp, etc?

Is there something similar to RFC 3665, which contains any of the above?

Thanks,

Sujay.


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<p class=3DMsoNormal><font size=3D3 color=3Dblue face=3D"Times New =
Roman"><span
style=3D'font-size:12.0pt;color:blue'>Is there any place where I can =
find SIP
call flow examples that contain PRACK, UPDATE, SigComp, =
etc?</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>Is there
something similar to RFC 3665, which contains any of the =
above?</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'>Thanks,</span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
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Following our discussion in the RAI Open Area Meeting in Dallas a few =
weeks ago on the many strategies circulating for securing RTP, we've =
formed a new mailing list in order to further refine our understanding =
of the problem space and figure out a way forward.

Subscription information for the list is available here.=20

<http://www.imc.org/ietf-rtpsec/>

The agenda item from the RAI Open Meeting, including the list of =
associated drafts, is pasted below. Interested parties, please move =
discussion of these documents to this list.

Jon Peterson
NeuStar, Inc.

----

    There are many proposal discussing keying for RTP when used with =
SIP.
    These issues cross SIPPING, MMUSIC, AVT, and MSEC.
    What are the common underling requirements driving these proposals?
    At the high level what are the various architectures?
    What cross WG architectural level question do we need to answer?
    Can we come to consensus on high level approaches we would like to
    investigate?


        RFC 3830 (MIKEY)
        RFC 3711 (SRTP)
        draft-ietf-msec-mikey-rsa-r-02
        draft-ietf-mmusic-sdescriptions-12
        draft-ietf-mmusic-kmgmt-ext-15
        draft-wing-mmusic-sdes-early-media-00
        draft-mcgrew-srtp-ekt-00
        draft-baugher-mmusic-sdp-dh-00
        draft-zimmerman-avt-zrtp-00
        draft-tschofenig-avt-rtp-dtls-00
        draft-fischl-sipping-media-dtls-0
        draft-fischl-mmusic-sdp-dtls-00
        draft-rescorla-tls-partial-00
        draft-modadugu-dtls-short-00
        draft-fries-msec-mikey-applicability-00
        draft-ietf-msec-mikey-dhhmac-11
        draft-lehtovirta-srtp-rcc-01
        draft-saito-mmusic-ipsec-negotiation-req-02


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<BR>

<P><FONT SIZE=3D2 FACE=3D"Courier New">Following our discussion in the =
RAI Open Area Meeting in Dallas a few weeks ago on the many strategies =
circulating for securing RTP, we've formed a new mailing list in order =
to further refine our understanding of the problem space and figure out =
a way forward.</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">Subscription information for the =
list is available here. </FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">&lt;<A =
HREF=3D"http://www.imc.org/ietf-rtpsec/">http://www.imc.org/ietf-rtpsec/<=
/A>&gt;</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">The agenda item from the RAI Open =
Meeting, including the list of associated drafts, is pasted below. =
Interested parties, please move discussion of these documents to this =
list.</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">Jon Peterson</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">NeuStar, Inc.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">----</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp;&nbsp; There are many =
proposal discussing keying for RTP when used with SIP.</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp;&nbsp; These issues =
cross SIPPING, MMUSIC, AVT, and MSEC.</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp;&nbsp; What are the =
common underling requirements driving these proposals?</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp;&nbsp; At the high =
level what are the various architectures?</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp;&nbsp; What cross WG =
architectural level question do we need to answer?</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp;&nbsp; Can we come =
to consensus on high level approaches we would like to</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp;&nbsp; =
investigate?</FONT>
</P>
<BR>

<P><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; RFC 3830 (MIKEY)</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; RFC 3711 (SRTP)</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
draft-ietf-msec-mikey-rsa-r-02</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
draft-ietf-mmusic-sdescriptions-12</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
draft-ietf-mmusic-kmgmt-ext-15</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
draft-wing-mmusic-sdes-early-media-00</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
draft-mcgrew-srtp-ekt-00</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
draft-baugher-mmusic-sdp-dh-00</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
draft-zimmerman-avt-zrtp-00</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
draft-tschofenig-avt-rtp-dtls-00</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
draft-fischl-sipping-media-dtls-0</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
draft-fischl-mmusic-sdp-dtls-00</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
draft-rescorla-tls-partial-00</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
draft-modadugu-dtls-short-00</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
draft-fries-msec-mikey-applicability-00</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
draft-ietf-msec-mikey-dhhmac-11</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
draft-lehtovirta-srtp-rcc-01</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
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From mmusic-bounces@ietf.org Thu Apr 13 13:36:42 2006
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Subject: [MMUSIC] Request for review:
	draft-ietf-mmusic-sdp-media-content-03.txt
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The MMUSIC working group is considering draft-ietf-mmusic-sdp-media- 
content-03.txt:

    This document defines a new Session Description Protocol (SDP)  
media-
    level attribute, 'content'.  The 'content' attribute defines the
    content of the media stream in more detailed level than the media
    description line.  The sender of an SDP session description can
    attach the 'content' attribute to one or more media streams.  The
    receiving application can then treat each media stream differently
    (e.g., show it on a big screen or small screen) based on its  
content.

It has been suggested that members of the ietf-types list may have  
experience with this type of attribute in other areas. Accordingly,  
this message is to solicit cross-area review of the draft. We would  
be grateful if any comments could be provided before 1st May 2006.

Thanks,
Colin (mmusic co-chair)

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From mmusic-bounces@ietf.org Fri Apr 14 09:33:35 2006
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From: Colin Perkins <csp@csperkins.org>
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Subject: [MMUSIC] Working group last call:
	draft-ietf-mmusic-fec-grouping-04.txt 
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This is to announce a working group last call on the Forward Error  
Correction Grouping Semantics in SDP <draft-ietf-mmusic-fec- 
grouping-04.txt>. Please send any final comments on this draft to the  
mailing list by 28 April 2006. If no substantive comments are  
received by that time, we will request the IESG consider this for  
publication as a proposed standard RFC.

Colin


On 10 Apr 2006, at 23:50, 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		: Forward Error Correction Grouping Semantics in Session  
> Description Protocol
> 	Author(s)	: A. Li, et al.
> 	Filename	: draft-ietf-mmusic-fec-grouping-04.txt
> 	Pages		: 7
> 	Date		: 2006-4-10
> 	
> This document defines the semantics that allows for grouping of
>    forward error correction (FEC) streams with the protected payload
>    streams in Session Description Protocol (SDP). The semantics  
> defined
>    in this document is to be used with Grouping of Media Lines in the
>    Session Description Protocol (RFC 3388) to group together "m" lines
>    in the same session.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-fec- 
> grouping-04.txt

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From mmusic-bounces@ietf.org Fri Apr 14 19:56:38 2006
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Jonathan,

As promised in Dallas, I have done a review of =
draft-ietf-mmusic-ice-tcp-00.

Most of my comments are editorial in nature, except perhaps a minor =
comment
on section 3.


Section 1

- 2nd paragraph, last line: change "communicatoins" to "communications".

Section 2

- It would be much clearer if a reference to RFC 4145 be placed at the
  very begining of this section (after the first paragraph). It is
  very difficult to follow starting with the 2nd paragraph unless the
  concepts described in RFC 4145 on active/passive/actpass are =
understood.

- 4th paragraph, 4th line. Change "by the agent" to "by the remote =
agent" or
  "by the other agent" to make it clear it is the other side you are =
talking
  about.

- 6th paragraph (top of p. 5). It is not clear what "specifications are=20
  being developped". Perhaps an informative reference to the draft is in =

=A0 order =
(http://www.ietf.org/internet-drafts/draft-ietf-behave-tcp-00.txt).

Section 3

- Top of page 7. The sentence "Only in cases ... be omitted" probably =
should
  be removed as it seems very restrictive. I understand what you are =
trying
  to do, i.e., get people to implement ICE-TCP in UAs, which I am OK =
with.=20
  What I think is important is that UAs have the ability to have ICE-TCP
  turned on or off depending on network preferences. For example, an=20
  enterprise may decide that ICE-TCP inside the network is unacceptable
  and should be turned off: this could be communicated to the client =
through
  Config-framework (so that it would work when the client is "nomad"). =
Or
  maybe there is two configuration profiles for the client (one for =
inside
  the enterprise, one for outside). In any case, I think the =
explanation, or
  at least the last sentence, is overly restrictive and should be=20
  adjusted.

Section 5

- First paragraph, first line. Change "procedurs" to "procedures".

- Second paragraph. Change "set to 9" to "set to 9 (i.e., Discard)" for=20
  clarity.

Section 6

- First paragraph, first line. Change "procedurs" to "procedures".

Section 7

- First word of page 12. Change "Even Ta" to "Every Ta".

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

Reported by Colin Perkins

    The MMUSIC working group met once at the 65th IETF meeting (Dallas,
    March 2006). Topics under discussion included the SDP Content  
Attribute,
    ICE, secure media stream setup, SDP for file transfer, RTSP, SIP for
    streaming media, QoS identification for preconditions, codec- 
specific
    parameters in SDP, use cases for session mobility and  
requirements for
    IPsec negotiation in SDP. The meeting was chaired by Joerg Ott  
and Colin
    Perkins and attended by some 130 participants.


Introduction and progress report

    The chairs introduced the session and gave a brief status review.  
The
    group has had one RFC published since the last meeting (offer/answer
    examples -> RFC 4317) and has a further ten drafts in the RFC Editor
    queue (including, finally, draft-ietf-mmusic-sdp-new-26.txt). There
    is one open rechartering issue, on how to continue with the Internet
    Media Guide work, and discussion on that subject is ongoing with the
    authors and area directors.


SDP Content Attribute

    draft-ietf-mmusic-sdp-media-content-02.txt

    Jani Hautakorpi presented the SDP Content Attribute draft.  This  
version
    clarifies the usage with offer/answer, but otherwise has no  
changes from
    the previous version. The group seemed happy with the draft,  
which looks
    to be ready for working group last call.


ICE

    draft-ietf-mmusic-ice-07.txt
    draft-ietf-mmusic-ice-tcp-00.txt

    Jonathan Rosenberg presented ICE, summarising changes since the  
previous
    version (see slides for details). There were some questions and  
comments:

    - Flemming Andreasen asked why the "a=ice-pwd" SDP attribute is  
session
      level only, rather than both session level and media level? It  
has to
      be shared across all candidates, but that doesn't necessarily  
imply it
      is a session level attribute.

    - Derek MacDonald asked for clarification if reliable 183 can  
work with
      "m=" lines that are set to inactive in the 180? Yes.

    - Magnus Westerlund asked if there is a way of saying "don't send  
early
      media" for a particular complete ICE exchange, since this  
option might
      be useful for RTSP. Jonathan didn't want to add this for SIP,  
but it
      can be added for RTSP in a separate draft if needed.

    - Magnus Westerlund noted that the current text regarding setting  
the
      RTP marker bit is audio-codec specific; this should be clarified.

    - Rohan Mahy asked for clarification of the 183 retransmit intervals
      for STUN reliability.

    - Magnus Westerlund asked if a change in the source address could  
trigger
      the RTP SSRC collision detection algorithm? It's not clear that  
this
      is an issue, since one can tell it is a change in address  
rather than
      a collision through the signalling exchanges. Steve Casner also  
noted
      that the wording on SSRC collision detection has changed in RFC  
3550,
      and that may affect this.

    - Erkki Koivusalo reminded the group of issues with using MAPPED- 
ADDRESS
      rather than XOR-MAPPED-ADDRESS and with learning new candidates  
from
      connectivity checks, as discussed on the mailing list.

    Jonathan Rosenberg stated that he considered the draft ready for  
working
    group last call apart from minor details. It is expected that an  
updated
    draft will be submitted shortly after the meeting to address  
these, then
    a working group last call.

    Jonathan Rosenberg also presented changes to the ICE-TCP draft.  
This is
    a major update, due to the changes to STUN and also the  
recognition that
    simultaneous open broke the mechanisms in the previous draft,  
requiring
    a rework of the specification. There are no known open issues,  
and there
    was little discussion (indeed, it appears that few people have  
read the
    draft). Joerg Ott asked for volunteers to provide detailed  
review: Jason
    Fischel, Rohan Mahy and Francois Audet volunteered.


Secure media stream setup

    draft-baugher-mmusic-sdp-dh-00.txt

    Mark Baugher presented a new draft on Diffie-Hellman exchanges for
    multimedia sessions. This adds the ability for sdescriptions to do a
    Diffie-Hellman key exchange (using a new session level "a=DH:" SDP
    attribute), providing perfect forward secrecy for media keys, and
    removing the need for secure transport of the SDP by using public  
key
    cryptography. This does not solve the "early media" problem with  
SIP key
    establishment, but does reduce the potential vulnerabilities of  
sending
    plain text keys and relying on secure SDP transport.

    Alan Johnston commented that he liked the idea better than the  
standard
    sdescriptions. He noted that person to person identification is a  
MUST,
    but the draft does not specify how the fingerprint is to be  
displayed. To
    ensure interoperability, and standard mechanism should be  
defined. Erik
    Rescorla wondered how to propose two different types of DH  
crypto?  This
    can be done by putting both in the SDP.

    Cullen Jennings and Joerg Ott noted that there are numerous  
proposals
    for updating security negotiation for SRTP, and that this fits  
within
    that general space. Discussion of this in comparison with the other
    proposals should take place within the RAI area meeting.


SDP for file transfer sessions

    draft-isomaki-sipping-file-transfer-01.txt
    draft-garcia-sipping-file-transfer-mech-00.txt

    Gonzalo Camarillo presented drafts on file transfer in the offer/ 
answer
    model. The idea is to describe, using a set of new SDP  
attributes, the
    file to be transferred, and so allow MSRP sessions to be  
negotiated to
    transfer the file, e.g. as a part of a SIMPLE instant messaging  
system.

    Discussion centred on whether duplicating MIME headers in SDP is  
a good
    idea, and why content-indirection is not appropriate.  There was  
a fair
    amount of requirements discussion, to understand why this is  
useful and
    to determine the correct home for the work (SIPPING or MMUSIC).  
There
    appears to be interest in the work, and it appears to make sense  
from
    an SDP perspective, so we should figure out the right home and  
proceed.


RTSP

    draft-ietf-mmusic-rfc2326bis-12.txt
    draft-ietf-mmusic-rtsp-nat-04.txt
    draft-westerlund-mmusic-3gpp-sdp-rtsp-02.txt

    Magnus Westerlund presented RTSP/2.0, outlining open issues with the
    base specification. These include: the definition of non-interleaved
    TCP/RTP/AVP, the inclusion of 50 and 60 frames-per-second SMPTE time
    code formats, the format of error message bodies, the format of  
the URI
    list in 300 responses (multiple choices), whether the dest_addr  
should
    contain used addresses, should there be a scoped address for IPv6
    multicast, use of unregistered media types in examples, should  
Expires
    also provide cache-control instructions, and proxy handling of
    Accept-Credentials. Details are provided in the slides.

    Regarding the SMPTE 50 and 60 FPS formats, Stephan Wenger  
volunteered
    to help with up a definition for the formats.

    Regarding the format of the error message body, Colin Perkins and
    Jonathan Rosenberg suggested stating that the format is HTML by  
default,
    with an Accept header being used to signal support for alternatives.

    Regarding the 200 response URI lists, Joerg Ott wondered why  
multiple
    Location headers couldn't be used?

    Regarding cache control, Colin Perkins noted that explicit  
specification
    is usually appropriate, rather than overloading the field, but  
Magnus
    doesn't want to diverge from HTTP behaviour more than necessary.  
Joerg
    Ott asked if live streams affect the cache behaviour? Not clear.

    Magnus noted that he hopes to resolve these open issues quickly,  
and is
    working with the TLS working group to review the TLS solution.  
Review of
    the draft was solicited to get it ready for working group last call:
    Stephan Wenger and Sean Sheedy volunteered to provide reviews.

    Magnus also briefly discussed the RTSP NAT traversal draft,  
noting that
    there have been no changes since the last meeting, and asking for  
help
    to complete the draft.


SIP for Streaming Media

    draft-whitehead-mmusic-sip-for-streaming-media-00.txt

    Marie-Jose Montpetit presented a new draft on evaluating the use  
of SIP
    for streaming media applications. The aim here is to discuss  
whether it
    makes sense to use SIP as an alternative to RTSP to control  
streaming
    media, if the protocols might somehow be combined (e.g. with SIP for
    session initiation handing over to RTSP for control), or if the  
current
    separation is the best approach.

    There was considerable discussion on the requirements and use  
cases. It
    is clear that the idea of blended services is popular, but it is  
not at
    all clear that a combined is needed to achieve them. The authors  
were
    suggested to further consider use cases, looking how the existing  
pieces
    of protocol infrastructure can be combined to address those use  
cases,
    and where potential holes might lie, to come up with a better  
problem
    statement for further discussion.


QoS Identification for Preconditions

    draft-polk-mmusic-qos-mechanism-identification-01.txt

    Gonzalo Camarillo briefly presented a new draft on QoS mechanism
    identification for use with preconditions. This adds a new attribute
    to identify which QoS mechanism is to be used to attempt to  
satisfy a
    QoS precondition. Flemming Andreasen agreed that this is a good  
idea,
    but asked if things like proxy-RSVP could be supported.

    The general consensus of the room was that this is conceptually a  
good
    idea, but not yet sufficiently baked to become a working group  
draft.


Codec Specific Parameters

    draft-xu-mmusic-sdp-codec-param-01.txt

    Peili Xu updated the group of the codec specific parameters  
draft. This
    was discussed at the last meeting, where it was suggested to  
consider
    how existing mechanisms (SDP extensions, SDP grouping, SDPng)  
might be
    used. The draft now makes such a comparison.

    Adam Roach noted that it is much easier to use existing mechanisms,
    rather than defining a new SDP extension, since there is no  
deployment
    challenge. Colin Perkins supported this, noting that codec specific
    parameters could be supported using the SDP grouping mechanism.  
Colin
    also noted that implementations that require codec specific ptime  
are
    broken according to the RTP a/v profile (a point which Flemming
    Andreasen disputed somewhat, noting that there might be  
legitimate use
    cases for this).

    To summarise, it is not clear that there is a problem to solve  
here that
    cannot be solved using existing mechanisms. Further clarification  
of the
    problem is needed, before we consider defining SDP extensions.


Use Cases for Session Mobility

    draft-komiya-mmusic-session-mobility-usecases-00.txt

    This presentation was dropped, due to lack of time.


IPsec negotiation

    draft-saito-mmusic-ipsec-negotiation-req-02.txt

    This presentation was dropped, due to lack of time.


				   - + -



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The Mobile TV World Congress, will be held in Paris January 23 to 26,
together with the worldwide leader event TVoDSL 2007. 

Carriers, content providers and equipment vendors will address technical
issues, describe the standardization process and discuss business and
content delivery models. 

 

A call for proposals is online at:

 <http://www.upperside.fr/mobiletv2007/mobiletv2007intro.htm>
http://www.upperside.fr/mobiletv2007/mobiletv2007intro.htm

 


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<p class=3DMsoNormal><strong><b><font size=3D2 face=3DArial><span =
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style=3D'font-size:
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vendors
will address technical issues, describe the standardization process and =
discuss
business and content delivery models. <o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
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Jonathan:

Here are some comments. They are mostly pretty minor.
I am still re-reading the draft to more fully understand it,
so more may follow.

General comments:
- The grammar given in section 12 specifies that a colon (":") is to  
immediately follow the word "candidate"
in the "a=candidate" line. This is consistent with the SDP syntax for  
an attribute with a value as specified in
draft-ietf-mmusic-sdp-new-26.
However, none of the examples given elsewhere in the document use a  
colon.

Section 1:
- (First PP) Misspelling - "Descrption" should be "Description".

Section 2:
- I found it difficult to understand much of the terminology on my  
first pass through
   the document, even though I had a basic understanding of how ICE  
works. In the end,
   I just skipped over it and then came back later.
   Perhaps consider placing the introductory sections before the  
terminology section,
   perhaps by making sections 3 - 6 subsections of section 1.  If you  
did this,
   then the terminology section would clearly divide the Informative  
text from the Normative text.
- It would be nice if the definitions of Reflexive Transport Address  
and Relayed Transport Address
   could make it clearer that these are subclasses of Derived  
Transport Addresses
   (for example, the definition text could start "A type of Derived  
Transport Address where ...").
- I found the definitions of the various Transport Address forms  
difficult to understand
   (I was familiar with the concepts, but hadn't see the terms  
before, and was uncertain for a while
   which term matched which concept).
   I might be useful to add a diagram (perhaps somewhere in sections  
3-6) that illustrates the
   various forms  (Derived, Reflexive, etc...)

Section 3:
- "in addition to private networks connected to the public Internet  
through NAT".
   The wording here suggests  that the combination of the Internet  
plus various private networks
   connected to it via NAT boxes is one realm.
- (top of page 10) "reply from that target Section 7.13".
   There is some sort of grammatical error here.

Section 7.1:
- In the case of a candidate whose components are all server- 
reflexive or relayed, must the component
   transport addresses all be derived from the same local address?
   I suspect so, but you don't say this.
- (page 13)
       "Local candidates are obtained by binding to ephemeral ports  
on an interface ..."
    AND
      "... obtain a set of candidates (one for each interface) by  
binding to N ephemeral UDP ports ..."
   This text seems to imply that the UDP ports must be ephemeral.  
This seems overly restrictive.
   I believe that the specification will work fine if a well-known  
port is used instead.
   I suggest replacing these with the following:
       "Local candidates are obtained by binding to ports (typically  
ephemeral ports) on an interface ..."
   AND
      "... obtain a set of candidates (one for each interface) by  
binding to N UDP ports ..."
- (page 14, indented paragraph)
   "The ability to process STUN connectivity checks allows for  
clients to obtain peer reflexive transport
    addresses that can be used by the network server to reach them  
without a relay, even through NATs
    with restrictive mapping and filtering policies."
   I don't understand this sentence.
   In general, I found the wording in this indented paragraph to be  
much below the very high standard
   set by the rest of the document. Perhaps you might rewrite this  
entire paragraph.
- (page 15)
    "... where there are multiple layers of NAT,  and that layering  
is known to the provider of the client."
   Suggest adding an extra "where", so it reads
    "... where there are multiple layers of NAT,  and where that  
layering is known to the provider of the client."

Section 7.8:
-  "symmetric NAT". Replace with BEHAVE terminology?

Section 12:
- In addition to all the SDP examples in section 11, it would be nice  
to see one or two examples of SDP
   that do NOT use $L1, $L-PRIV-1.IP, $STUN-PUB-2.PORT, etc.
   This would help the reader understand the grammar rules.


- Philip


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Subject: [MMUSIC] Some more questions and comments on
	draft-ietf-mmusic-ice-08
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Jonathan:

A few questions about the specification:

1)  Can a single offer contain candidates for the same media stream  
that use different transport protocols??
      The spec seems to handle this case, but I could not find a  
statement that this is allowed.

2)  Must all components of a candidate use the same transport protocol?

3) Must the components of a candidate appear together in increasing  
order in the SDP, or can they appear
      in any order and intermixed with candidates from other components?

And here are some comments. These are mostly editorial in nature.

General comments:
-  Shouldn't this specification reference draft-ietf-mmusic-sdp-new  
instead of RFC 2327?
    (See, for example, the references in section 12)

-  RFC 3264 (offer/answer model) and draft-ietf-mmusic-sdp-new state  
that the address and port
    in the m/c line is the one used to receive media on. However,  
when using ICE in the presence of NAT,
    this must also be the address and port from which the media is  
_sent_.
    I think it would be helpful to state this explicitly somewhere  
near the  beginning of the specification.

Section 2:
-  In the definition of a Relayed Transport Address, it might be  
helpful to readers to mention TURN.


Section 7.1:
-    "... transport addresses for each component of each component  
match".
    Delete "for each component" ?

Section 7.2:

-   Must all the components of a given candidate have the same priority?
     Section 7.3 says "yes", but I feel that, to a reader looking for  
this information, section 7.2 would be the
     logical place to look. Suggest mentioning this in section 7.2 as  
well.

-   I think it would be helpful to have a clear statement in this  
section that the prioritization is between
     all the candidates for a single media stream, regardless of  
their  transport type, address type, port, etc.

-       "If the active candidate pair in the initial offer is being  
validated, media will flow to that pair once it is validated."
    Suggest replacing with
        "The active candidate pair in the initial offer is validated  
first, and if that validation succeeds,
          media will immediately begin to flow between the pair."

Section 7.3:
-        "If there is no active candidate, the agent MUST include an  
a=inactive attribute.
           The RTP address and port in the m/c-line is  
inconsequential, since it won't be used."
    In the second sentence, suggest replacing "RTP" by "media", to  
make the statement accurate
    even if something other than RTP is used.

-       "The candidate-id MUST be chosen randomly and contain at  
least 24 bits of randomness
          (this does not mean that  the candidate-id is 24 bits long;  
just that it has at least 24 bits
          of randomness)."
    I personally found this sentence very confusing the first couple  
of times I read it.
    Suggest replacing this with the following text:
         "The candidate ID MUST be chosen randomly and contain at  
least 24 bits of randomness.
          This means that a candidate ID must be at least 4  
characters long, since each character
          in the base64 alphabet used for candidate IDs contains at  
most 6 bits of randomness.
          A candidate ID MAY be longer than 4 characters, and  
different candidate IDs MAY have
          different lengths."

-       "It is needed only to prevent a possible simultaneous selection
          by two agents within a private network for the useful  
lifetime of the
          software or hardware."
     This sentence meant nothing to me until I read the example two  
pages later.
     Perhaps add a note "(For more explanation, see the example later  
in this section)"?


-       "... a unique identifier, call the tid"
   In this section, you use only the term "tid".
   By contrast, in section 7.4 you always write "transport address ID".
   Perhaps, in each section, say  "transport address ID or tid" the  
first time the term appears in that section,
   and then use tid from then on?

-     "The tid gets combined ... to form the username that is placed  
in the STUN checks ...".
    Perhaps mention the "Transport Address Pair ID"? That is, perhaps  
reword as follows:
       "The tid gets combined ... to form the Transport Address Pair  
ID (see section 7.4)
        which is used as the username in the STUN checks ...".


Section 12:

- The rules DIGIT and SP are never defined.
   Suggest stating that they are from section 9 of draft-ietf-mmusic- 
sdp-new

-      "   base64-char           = ALPHANUM / DIGIT / "+" / "/"   "
    Because of the overlap between ALPHANUM and DIGIT,  I suggest it  
would be less confusing to write this as
        "   base64-char           = ALPHA  / DIGIT / "+" / "/"   "
   where ALPHA is also defined in section 9 of draft-ietf-mmusic-sdp- 
new.
   Alternatively, a rule for "base64-char" is defined explicitly in  
section 9 of draft-ietf-mmusic-sdp-new
   (albeit under the comment "      ; sub-rules of 'k='   ").

-  The rule "token" is never defined.
    Presumably you mean "token" as defined in section 9 of draft-ietf- 
mmusic-sdp-new?

-    "                       addr SP     ;addr from RFC 3266    "
   Suggest instead using instead either "unicast-address" or  
"connection-address"
   both of which are defined in section 9 of draft-ietf-mmusic-sdp-new.


- Philip


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Hi,
Two more editorial comments:

1. In section 9 second paragraph "successul" should be "successful"

2. In section 7 " since it is the STUN requests identify the candidate =
pairs." I think that a "that" is missing after the "STUN requests"

Roni Even

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From: Kylin Wei <weiqikun@huawei.com>
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Subject: [MMUSIC] A question about draft-ietf-mmusic-rfc2326bis-12
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I'm studying draft-ietf-mmusic-rfc2326bis-12 now. I had a question when
I read an example about "SETUP" method at Page 48.
Original text:
C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/2.0
      CSeq: 302
     Transport: RTP/AVP;unicast;dest_addr=":4588"/":4589",
                 RTP/AVP/TCP;unicast;interleaved=0-1

I don't know why there isn't receiver's IP address after "dest_addr"
parameter.
It should be 'dest_addr="192.168.2.1:4588"/"192.168.2.1:4589"' mode
because of unicast transport.



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Dear mmusic experts, 

I would like to request some feedback about a question on RTSP and 3GPP PSS. I 
have been working and doing some research around streaming over wireless during 
the last years, so that’s where I am coming from. 

My understanding is that there is no real support for RTSP pipelining in 
standard RTSP. As far as I understand, this also stands for RTSP 2.0 I-D, but 
please correct me if I’m wrong. 

I was wondering that enabling RTSP pipelining is an interesting area to 
consider, particularly as we see that interactivity and fast session setup are 
key to keep user’s attention on the service. 

Additionally, enabling early activation of Secondary PDP-Context(s) in a 3GPP 
PSS environment would be a good idea (3GPP TS 26.234). All in all, IMMO I think 
some time could be saved in session setup by defining a proper standard 
extension to RTSP. Defining it under a feature tag would keep backwards 
compatibility untouched. 

Just wanted to trigger some discussion to see if anyone believes this could be 
an interesting area to think about. In case you believe it doesn’t make any 
sense, I am happy to hear your thoughts as well. 

Thanks in advance and kind regards, 

David 

PS.- I know that with WLAN and HSxPA out there, the gains of pipelining RTSP 
may be reduced. Still, in cases when signalling is carried over relatively 
narrowband bearers –leaving broadband connections for media- I believe a sort 
of extension would possibly help... 

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Please review the following statement from page 97:


         o The lack of the dest_addr parameter indicates that the server
           SHALL send media to same address for which the RTSP messages
           originates. Does not work for transports requiring explicitly
           given destination ports.

Cheers

Magnus

Kylin Wei wrote:
> I'm studying draft-ietf-mmusic-rfc2326bis-12 now. I had a question when
> I read an example about "SETUP" method at Page 48.
> Original text:
> C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/2.0
>       CSeq: 302
>      Transport: RTP/AVP;unicast;dest_addr=":4588"/":4589",
>                  RTP/AVP/TCP;unicast;interleaved=0-1
> 
> I don't know why there isn't receiver's IP address after "dest_addr"
> parameter.
> It should be 'dest_addr="192.168.2.1:4588"/"192.168.2.1:4589"' mode
> because of unicast transport.
> 
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 


-- 

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
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From mmusic-bounces@ietf.org Wed Apr 26 08:30:41 2006
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To: David Viamonte <viamonte@entel.upc.edu>
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Hi David,

Request pipelining is allowed. However it can't be applied between the=20
first and second SETUP for example. Also between the last SETUP and the=20
first PLAY there is a restriction on doing request pipelining to avoid=20
needing to deal with the unknown state. I think what you should call=20
this is session establishment optimization.

I think there is a general interest in optimizing RTSP in regards to the=20
time it takes to do session establishment. However, I think that just=20
discussing it will not make it happen, because that have been the state=20
of things for the last 6 years. If you like something to happen in this=20
area you will need to step in and start proposing real solution and=20
actually write specification proposals for how to accomplish it.

Cheers

Magnus

David Viamonte wrote:
> Dear mmusic experts,=20
>=20
> I would like to request some feedback about a question on RTSP and 3GPP=
 PSS. I=20
> have been working and doing some research around streaming over wireles=
s during=20
> the last years, so that=92s where I am coming from.=20
>=20
> My understanding is that there is no real support for RTSP pipelining i=
n=20
> standard RTSP. As far as I understand, this also stands for RTSP 2.0 I-=
D, but=20
> please correct me if I=92m wrong.=20
>=20
> I was wondering that enabling RTSP pipelining is an interesting area to=
=20
> consider, particularly as we see that interactivity and fast session se=
tup are=20
> key to keep user=92s attention on the service.=20
>=20
> Additionally, enabling early activation of Secondary PDP-Context(s) in =
a 3GPP=20
> PSS environment would be a good idea (3GPP TS 26.234). All in all, IMMO=
 I think=20
> some time could be saved in session setup by defining a proper standard=
=20
> extension to RTSP. Defining it under a feature tag would keep backwards=
=20
> compatibility untouched.=20
>=20
> Just wanted to trigger some discussion to see if anyone believes this c=
ould be=20
> an interesting area to think about. In case you believe it doesn=92t ma=
ke any=20
> sense, I am happy to hear your thoughts as well.=20
>=20
> Thanks in advance and kind regards,=20
>=20
> David=20
>=20
> PS.- I know that with WLAN and HSxPA out there, the gains of pipelining=
 RTSP=20
> may be reduced. Still, in cases when signalling is carried over relativ=
ely=20
> narrowband bearers =96leaving broadband connections for media- I believ=
e a sort=20
> of extension would possibly help...=20
>=20
> -------------------------------------------------
> This mail sent through IMP: http://horde.org/imp/
>=20
>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
>=20


--=20

Magnus Westerlund

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

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Hi,

ICE -08 still refers to offers and answers using terminology which
contradicts Section 13.2.1 of RFC3261.  By my reading, a session descriptio=
n
in an unreliable 18x is never an offer or an answer.  I often describe this
as an "early session description" or "courtesy SDP".

Several places in ICE (particularly Section 8) talk about performing ICE
operations once you receive an answer and refer to courtesy SDP as if it is
an answer.

For example, the current wording of Section 8 para 2:

   ICE requires a series of STUN-based connectivity checks to take place
   between endpoints.  These checks start from the answerer on
   generation of its answer, and start from the offerer when it receives
   the answer.  These checks can take time to complete, and as such, the
   selection of messages to use with offers and answers can effect
   perceived user latency.  Two latency of figures are of particular
   interest.  These are the post-pickup delay and the post-dial delay.
   The post-pickup delay refers to the time between when a user "answers
   the phone" and when any speech they utter can be delivered to the
   caller.  The post-dial delay refers to the time between when a user
   enters the destination address for the user, and ringback begins as a
   consequence of having succesfully started ringing the phone of the
   called party.


and my proposed rewording:

   ICE requires a series of STUN-based connectivity checks to take place
   between endpoints.  These checks start from the answerer on
   generation of its first session description sent after the offer,
   and start from the offerer when it receives
   this first session description.  If this first message arrives in
   an unreliable message, the actual answer will arrive in the first reliab=
le
   message after the offer.

   STUN connectivity checks can take time to complete, and as such, the
   number and timing of messages used for these checks can affect
   perceived user latency.  Two latency figures are of particular
   interest.  These are the post-pickup delay and the post-dial delay.
   The post-pickup delay refers to the time between when a user "answers
   the phone" and when any speech they utter can be delivered to the
   caller.  The post-dial delay refers to the time between when a user
   enters the destination address for the user, and ringback begins as a
   consequence of having succesfully started ringing the phone of the
   called party.

and also in the 5th paragraph (currently):

   For this reason, if an offer is received in an INVITE request, the
   UAS SHOULD immediately gather its candidates and then generate an
   answer in a provisional response.  When reliable provisional
   responses are not used, the SDP in the provisional response is not
   formally the answer; the value in the 200 OK is the actual answer.
   However, RFC 3261 allows for SDP to appear in an unreliable
   provisional response, in which case its value has to be identical to
   the value placed in the 200 OK.  Thus, we refer to the SDP in the
   provisional response, even when unreliable, as the answer.  To deal
   with possible losses of the provisional response, it SHOULD be
   retransmitted until some indication of receipt.  This indication can
   either be through PRACK [11], or through the receipt of a STUN
   Binding Request with a correct username and password.  Even if PRACK
   is not used, the provisional response SHOULD be retransmitted using
   the exponential backoff described in [11].  Furthermore, once the
   answer has been sent, the agent SHOULD begin its connectivity checks.
   Once a candidate reaches the Valid or Recv-Valid state, the UAS has a
   known-valid path for media packets towards the UAC.  This point is
   called the connected point in ICE.

and my proposed rewording:

   For this reason, if an offer is received in an INVITE request, the
   UAS SHOULD immediately gather its candidates and then generate a
   session descripition in a provisional response.  When reliable provision=
al
   responses are not used, the SDP in the provisional response is not
   formally the answer; the value in the 200 OK is the actual answer.
   However, RFC 3261 encourages implementors to include this "courtesy SDP"
   in an unreliable
   provisional response, in which case its value has to be identical to
   the value placed in the 200 OK.  To deal
   with possible losses, the provisional response SHOULD be
   retransmitted until the UAS receives some indication of receipt.=20
This indication can
   either be through PRACK [11], or through the receipt of a STUN
   Binding Request with a correct username and password.  Even if PRACK
   is not used, the provisional response SHOULD be retransmitted using
   the exponential backoff described in [11].  Furthermore, once the
   session description has been sent, the agent SHOULD begin its
connectivity checks.
   Once a candidate reaches the Valid or Recv-Valid state, the UAS has a
   known-valid path for media packets towards the UAC.  This point is
   called the connected point in ICE.

I can propose specific text for the other usages if other agree this needs
to be changed.

thanks,
-rohan

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Hi,<br><br>ICE -08 still refers to offers and answers using terminology whi=
ch contradicts Section 13.2.1 of RFC3261.&nbsp; By my reading, a session de=
scription in an unreliable 18x is never an offer or an answer.&nbsp; I ofte=
n describe this as an &quot;early session description&quot; or &quot;courte=
sy SDP&quot;.
<br><br>Several places in ICE (particularly Section 8) talk about performin=
g ICE operations once you receive an answer and refer to courtesy SDP as if=
 it is an answer.<br><br>For example, the current wording of Section 8 para=
 2:
<br><br><pre>   ICE requires a series of STUN-based connectivity checks to =
take place<br>   between endpoints.  These checks start from the answerer o=
n<br>   generation of its answer, and start from the offerer when it receiv=
es
<br>   the answer.  These checks can take time to complete, and as such, th=
e<br>   selection of messages to use with offers and answers can effect<br>=
   perceived user latency.  Two latency of figures are of particular<br>
   interest.  These are the post-pickup delay and the post-dial delay.<br> =
  The post-pickup delay refers to the time between when a user &quot;answer=
s<br>   the phone&quot; and when any speech they utter can be delivered to =
the
<br>   caller.  The post-dial delay refers to the time between when a user<=
br>   enters the destination address for the user, and ringback begins as a=
<br>   consequence of having succesfully started ringing the phone of the
<br>   called party.</pre><br>and my proposed rewording:<br><br><pre>   ICE=
 requires a series of STUN-based connectivity checks to take place<br>   be=
tween endpoints.  These checks start from the answerer on<br>   generation =
of its first session description sent after the offer,=20
<br>   and start from the offerer when it receives<br>   this first session=
 description.  If this first message arrives in <br>   an unreliable messag=
e, the actual answer will arrive in the first reliable<br>   message after =
the offer. =20
<br><br>   STUN connectivity checks can take time to complete, and as such,=
 the<br>   number and timing of messages used for these checks can affect<b=
r>   perceived user latency.  Two latency figures are of particular<br>
   interest.  These are the post-pickup delay and the post-dial delay.<br> =
  The post-pickup delay refers to the time between when a user &quot;answer=
s<br>   the phone&quot; and when any speech they utter can be delivered to =
the
<br>   caller.  The post-dial delay refers to the time between when a user<=
br>   enters the destination address for the user, and ringback begins as a=
<br>   consequence of having succesfully started ringing the phone of the
<br>   called party.<br><br></pre>and also in the 5th paragraph (currently)=
:<br><br><pre>   For this reason, if an offer is received in an INVITE requ=
est, the<br>   UAS SHOULD immediately gather its candidates and then genera=
te an
<br>   answer in a provisional response.  When reliable provisional<br>   r=
esponses are not used, the SDP in the provisional response is not<br>   for=
mally the answer; the value in the 200 OK is the actual answer.<br>   Howev=
er, RFC 3261 allows for SDP to appear in an unreliable
<br>   provisional response, in which case its value has to be identical to=
<br>   the value placed in the 200 OK.  Thus, we refer to the SDP in the<br=
>   provisional response, even when unreliable, as the answer.  To deal
<br>   with possible losses of the provisional response, it SHOULD be<br>  =
 retransmitted until some indication of receipt.  This indication can<br>  =
 either be through PRACK [11], or through the receipt of a STUN<br>   Bindi=
ng Request with a correct username and password.  Even if PRACK
<br>   is not used, the provisional response SHOULD be retransmitted using<=
br>   the exponential backoff described in [11].  Furthermore, once the<br>=
   answer has been sent, the agent SHOULD begin its connectivity checks.
<br>   Once a candidate reaches the Valid or Recv-Valid state, the UAS has =
a<br>   known-valid path for media packets towards the UAC.  This point is<=
br>   called the connected point in ICE.<br></pre>and my proposed rewording=
:
<br><br><pre>   For this reason, if an offer is received in an INVITE reque=
st, the<br>   UAS SHOULD immediately gather its candidates and then generat=
e a<br>   session descripition in a provisional response.  When reliable pr=
ovisional
<br>   responses are not used, the SDP in the provisional response is not<b=
r>   formally the answer; the value in the 200 OK is the actual answer.<br>=
   However, RFC 3261 encourages implementors to include this &quot;courtesy=
 SDP&quot;=20
<br>   in an unreliable<br>   provisional response, in which case its value=
 has to be identical to<br>   the value placed in the 200 OK.  To deal<br> =
  with possible losses, the provisional response SHOULD be<br>   retransmit=
ted until the UAS receives some indication of receipt.  This indication can
<br>   either be through PRACK [11], or through the receipt of a STUN<br>  =
 Binding Request with a correct username and password.  Even if PRACK<br>  =
 is not used, the provisional response SHOULD be retransmitted using<br>
   the exponential backoff described in [11].  Furthermore, once the<br>   =
session description has been sent, the agent SHOULD begin its connectivity =
checks.<br>   Once a candidate reaches the Valid or Recv-Valid state, the U=
AS has a
<br>   known-valid path for media packets towards the UAC.  This point is<b=
r>   called the connected point in ICE.<br></pre>I can propose specific tex=
t for the other usages if other agree this needs to be changed.<br><br>
thanks,<br>-rohan<br><br>

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The fourth edition of the TVoDSL International Conference and the Triple
Play Showcase will stand in Paris next January 23 to 26, 2007.

With the opening of a second exhibition floor and the launch of the Mobile
TV World Congress, TVoDSL will confirm his rank of very first worldwide
rendez-vous in the IPTV realm.

 

A call for proposals has been launched for the TVoDSL conference.

 

All details at:

 <http://www.upperside.fr/tvodsl2007/tvodsl2007intro.htm>
http://www.upperside.fr/tvodsl2007/tvodsl2007intro.htm

 


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<p class=3DMsoNormal><font size=3D1 face=3DArial><span lang=3DEN-GB =
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style=3D'font-family:Arial'>TVoDSL International =
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and the Triple Play Showcase will stand in <st1:place =
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<p class=3DMsoNormal><font size=3D1 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
9.0pt;font-family:Arial'>With the opening of a second exhibition floor =
and the
launch of the Mobile TV World Congress, TVoDSL will =
confirm&nbsp;his<span
class=3Dtextetrois>&nbsp;rank of very first worldwide rendez-vous in the =
IPTV
realm.</span><o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span lang=3DEN-GB =
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<p class=3DMsoNormal><span class=3Dtextetrois><font size=3D1 =
face=3DArial><span
lang=3DEN-GB style=3D'font-size:9.0pt;font-family:Arial'>A call for =
proposals has
been launched for the TVoDSL conference.</span></font></span><font =
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From mmusic-bounces@ietf.org Thu Apr 27 14:08:12 2006
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From: Philip Matthews <philip_matthews@magma.ca>
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Subject: [MMUSIC] Still more questions and comments on
	draft-ietf-mmusic-ice-08
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Jonathan:

Below are some more questions and comments on draft-ietf-mmusic-ice-08.

- Philip

Technical comments and questions
============================

1)  If there are three m/c lines in the SDP, does the agent construct  
three candidate pair check orderings,
      or just one ordering containing the candidates for all three  
media streams?
      Viewed another way, is there just one Ta timer, or one Ta timer  
per m/c line?
      Taking this even farther, if the agent is taking to two or more  
different remote agents (because it is
      trying to establish two different call legs, perhaps for the  
same call or perhaps for different calls),
      then is the result one or multiple candidate pair check orderings?
      (It seems to me that, to reduce the load on intervening NATs,  
there should be just one.)

2)  When an agent receives a STUN Request, section 7.8 seems to say  
that the agent validates
       the request and sends a reply based only on local information.  
In particular, the USERNAME
       is validated by checking for a match on the local tid (and not  
the remote tid).
       The reason for this seems to be the case when an the offerer  
receives a STUN Request but has
       not yet received the answer from the remote agent. (This case  
is presumably pretty common since
       STUN Requests go directly, but signaling goes indirectly).
       In this case, the text seems to say that the STUN Request is  
validated and a STUN response sent
       immediately,  and information about the STUN Request is stored  
for further processing once
       the answer is received.
       However, I can see implementations deciding instead to ignore  
STUN Requests where there is not
       an exact match for the USERNAME.  I can see them doing this  
for three reasons:
       a) It is easier. In this case, there is no need to queue up  
information about the STUN Request for
           later processing when the answer arrives.
       b) For security. The agent might be acting as a STUN server  
for more than one STUN usage, and use the
           USERNAME to differentiate the usages. For security and  
because of DoS attacks, the agent
           simply ignores STUN Requests with unknown USERNAMES.
       c) Better validity checking. If an agent just does a prefix  
match, then USERNAMES corresponding
           to illegal transport address combinations are not detected.
           For example, a request with USERNAME "Lxx:1:Ryy: 
2" (mismatched component ids) or
           a request where the candidates are from different media  
streams, or use different transport addresses, etc.
       My proposal is that:
        - An agent always does an exact match against the USERNAME  
when a STUN Request arrives.
        - The answerer waits some time (say 50 or 100 msec) before  
starting its connectivity checks,
          or waits until it receives a STUN Request from the offerer,
          or doesn't bother waiting at all (because things will work  
out anyway after retries).

3) Another reason to do an exact match against the USERNAME might be  
the following.
     Consider two agents, L and R and a candidate pair Lxx:1:Ryy:2.
     L sends a STUN Request for this pair and receives a success  
response, and thus
     moves its local state machine into the Recv-Wait state.
     R sends a STUN Request for this pair, but due to some problem,
     L does not like this request, and sends an error response,
     which causes R to give up and move into the Invalid state.
     At this point, L continues to send STUN Requests each Tr interval ,
     and R replies with a success response, and the candidate pair  
remains stuck
     in the (RECV-WAIT, INVALID) state combination.
     To avoid this problem, it might be appropriate for an agent to  
return an error response,
     when its local state machine is in the Invalid state.

4) According to section 7.10.1, when an agent receives a STUN request  
and discovers a new
     peer reflexive transport address, the agent generates a new  
candidate id from candidate IDs
     of the candidate pair specified by the USERNAME:
           "In particular, the candidate ID is constructed by
            concatenating the remote candidate ID with the native  
candidate ID (without the colon)."
     It is possible, due to routing changes or a glitch in the  
intervening NAT(s),
     that the same candidate pair might generate TWO or more peer  
reflexive transport addresses.
     For example, say L and R are the two agents. The network in  
which R resides is configured
     to send packets through NAT box N1 in normal use, but if N1  
fails, packets are sent through
     NAT box N2.
     For a given candidate pair, R sends multiple STUN Requests to L.  
It may be that the first
     STUN Request goes through N1, but the second goes through N2.  
This causes L to see
     two different peer reflexive transport addresses for the same  
USERNAME.
     These two peer reflexive transport addresses will be assigned  
the same candidate ID by L,
     violating the rule that candidate IDs must be unique.
     I can see two possible ways to handle this situation:
     a) Append a string to the candidate ID to ensure they are unique  
(e.g., "1", "2", ...),  OR
     b) Use the last (or the first) peer reflexive transport address  
seen for this USERNAME.
     Option (a) is may be tricky, especially when the remote  
(generating) candidate has multiple components,
     so I suggest option (b).


5)  How are two candidate IDs of different lengths compared? My  
assumption is that the shorter
      one is padded at the end with Null's (0x00) until the two IDs  
have the same length.
      Is this correct?
      If so, then am I just confused, or is there an error in the  
following sentence?
        "For example, the candidate with candidate ID agD is greater  
than the candidate with ID ad7,
         and both of those are greater than the candidate with ID zz."
     I would have thought that, in reverse lexicographic order, the  
IDs would have be ordered as
     zz, agD, ad7, since the "zz" is padded at the end with a Null  
character.

6) If agent L has two candidates with the same derived address (but  
different local addresses)
    and picks one of the candidates as the one to put in the m/c  
line, then as the spec points out,
    the remote agent won't know which candidate the m/c line  
corresponds to. The spec recommends
    the remote agent just guesses and picks one to move to the head  
of the check ordering.
    Wouldn't it be better to:
      - Move BOTH candidates to the head of the ordering, or
      - Somehow indicate in the SDP which candidate was chosen?
   In other words, this case breaks the requirement that both  
endpoints create the same
   candidate pair check ordering.


7)  Am I correct in saying that none of the example candidate IDs in  
the document are actually legal?
      Because they are all too short?
      If so, it might be useful to use realistic candidate IDs or at  
least say something about this.

8) In section 7.6, there is the sentence:
           "The check is started for the first transport address pair  
in the transport address
            pair check ordered list (which will be part of the active  
candidate) that is
            in the Waiting state"
     I don't understand the part in parentheses.
     Don't we check all the pairs, not just the pairs associated with  
the active candidate?


Editorial comments
===============

General:

- In a number of places in the document, you use the phrase   
"reflexive or relayed"
   where it would be more general and more accurate to say "derived".
   (For example, "reflexive or relayed candidate", rather than  
"derived candidate", etc).

- In the first part of the document, you use "NAT" to mean the  
technology (i.e., "Network Address Translation"),
   for example in "Numerous solutions have been proposed for allowing  
these protocols to
    operate through NAT" in section 1.  But later in the document,  
you start using "NAT" to
    mean a single box (i.e., "Network Address Translator"), for example
    "...when two agents are separated by a NAT ..."  in section 7.10.
   It would be nice if you were consistent.
   In my opinion, the most common usage is to use "NAT" to refer to a  
single device and "NATs" when
   referring to multiple devices. However, I would be happy with  
another solution as long as the usage
   was consistent and made explicit.


Section 7.1:

- "... transport addresses for each component of each component  
match ...".
    I suggest deleting "for each component".


Section 7.3:

-      "even if the transport address from a previous  session was  
being recycled."
   For consistency (in verb tense) with the rest of the sentence,  
suggest changing "was" to "is".

-    "In this case, thats ...".
   "thats" should be either "that's" or "that is".

-    "10.0.1.1:8866 and 8877".
   I understand what you are trying to say here,  but you never  
explain this notation,
   and this paragraph is the only place in the document where you use  
it.
   (It is not, for example, legal in the a=candidate lines).
   Furthermore, you have just been talking about using a colon to  
construct tids and usernames.
   Perhaps write this in a different way to make sure the reader  
understands?
   Note also that a few sentences later, you say
          "A will send a STUN request to 10.0.1.1:8866 and 8877"
   when what you really mean is
          "A will send one STUN request to 10.0.1.1:8866 and another  
STUN request to 10.0.1.1:8877".

-     "To fix this, tid takes on ..."
   For consistency with previous usage, I think there should be a  
"the" before "tid", so the sentence reads
        "To fix this, the tid takes on ..."

- "ususally" should be "usually"


Section 7.4:

- (first two PPs) The intermixing of the running example and the  
specification in these paragraphs
   is a bit confusing, because it is not clear which statements are  
normative and which just apply
   to the example. Suggest splitting the example out into its own  
paragraph.

-      "If an offer/answer exchange took place for a session  
comprised of an audio
          and a video stream, and each agent had two candidates per  
media
          stream, there would be 8 candidate pairs, 4 for audio and 4  
for video."
    If an offer or answer can have candidates with different  
transport protocols
    (not sure if this is allowed -- I asked this question in my last  
message),
    then this assumes all candidates have the same transport protocol.

-       "Returning to the previous example, for each of the 8  
candidate pairs,
          there would be two transport address pairs - one for RTP,  
and one for
          RTCP.  If one candidate has more components than the other,  
those
          extra components will not be part of a transport address  
pair, won't
          be validated, and will effectively be treated as if they  
weren't
          included in the candidate pair in the first place."
    The first sentence seems to assume that the audio and video  
streams will use RTCP,
    and so both ends offered two components in each candidate.
    The second sentence then confuses things further -- is this  
supposed to be a continuation
    of the example, or a normative statement?

Section 7.6:

-       "The state machine has six states - waiting, testing, Recv- 
Valid, Send-Valid, Valid and Invalid."
     I think  "waiting" and "testing" should be capitalized.

- Immediately after the state transition diagram, I suggest giving a  
brief description
   of the meaning of each state.
   Though there are some descriptions later on, I think it would be  
nice to have the descriptions
   complete and all together in one place

-  The notation in the state diagram indicates that the state machine  
moves from Waiting to Testing
    when timer Ta expires. However, this is true only if the state  
machine is the next in line to start testing.

-      "... captures the progress of connectivity checks."
    Grammar: Insert a "the" before "connectivity"?

-         "This is done by  computing the matching transport address  
pair for each Binding
             Request.  This is done by examining the USERNAME of the  
incoming  Binding Request. "
    Suggest rewording these sentences along the following lines:
          "This is done by determining the matching transport address  
pair for each incoming STUN message.
           For an incoming Binding Request message, this is done by  ...
           For an incoming response message, this is done by ...


-        "Extensions to ICE that describe other transport protocols  
SHOULD
            describe the state machine and the procedures for  
connectivity checks."
     To be clearer and more precise, perhaps reword as follows?
          "Extensions to ICE that describe other transport protocols  
SHOULD
           describe the state machine and procedures for those  
protocols."

-     "... in Figure 5"
    Missing period at end of sentence.



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[removing recipients so that the filters do not mark it as moderated]

Hi Joerg,

 > moving this to MMUSIC:

perfect.

 > Regarding the
 > encoding, I suppose we will have to worry about how to put this into
 > SDP.  This brings up -- again -- the discussion of whether to include
 > some "blob" of, from an SDP perspective, opaque metadata (e.g., b64
 > encoded zip version of the FLUTE FDT, just to give some weird example)
 > or whether to make these attributes explicit.

I would go for explicit SDP attributed (i.e., 'a' lines).

 > If we make the attributes explicit, this is essentially a more general
 > aspect of describing the purpuse of a media stream and would not be
 > restricted to file transfer.  We have some informal hints for audio
 > and video already and this appears conceptually similar in that it
 > identifies the purpose and helps the answerer to decide whether to
 > accept it or not.
 >
 > Keeping this in mind, it would be extremely useful, not to invent
 > independent sets of attributes for file transfer, MSRP sessions,
 > audio, video, whiteboard, and so so.

The attributes defined in the draft describe a file to be transfered. If 
somebody comes up with different attributes to describe the contents of 
the audio being sent over a media stream, those will not have any 
interaction with our attributes. Different types of media streams need 
different attributes to describe their contents.

 > I clearly _do_not_ want to end up in an XCON-style discussion about
 > metadata and the like, but it may be worthwhile given some
 > consideration to future evolution of this stuff and compatibility
 > with existing attributes.

We have not found any interactions with existing attributes.

 >>> I am also concerned about motivating the use of inline images for
 >>> icons.  I can hardly see why you would need an icon to decide
 >>> whether or not to accept a file transfer.
 >>
 >> A lot of commercial IM systems support it, and users find it very 
useful. SIP-based systems need to be able to provide a similar user 
experience if we do not want people to start doing things is proprietary 
ways.
 >
 > It's your call in the end of the day.  Often, icons are created based
 > upon the filename extension -- and I suspect that the filename is going
 > to be much more relevant that the icon in deciding whether or not to
 > accept it.  Icons are not worth much and only increase volume.  But if
 > we decide that it is useful:

If I want to send you a picture called PCJ00234.JPG, you will not know 
whether or not you want to receive it. However, if you see a small icon 
of the picture, you will be in a better position to make that decision. 
Many people find icons very useful, in many cases much more than the 
file name.

 > You must decide how much you worry about the SIP signaling architecture.
 > Given that the INFO method is "damned" (which I fully support), I do not
 > believe that turning parts of SDP into a transport is a particularly
 > smart idea.  It clearly does not ad efficiency.  If you follow
 > traditional SIPPING thinking, you will need to separate the desired
 > effect from the mechanism to achieve it.  There may be better ways.
 >
 > I am simply worried to open a door that we better leave closed.
 > Would you want to limit the media type or the size of an icon?
 > If so, what would be the right value?  Would you negotiate it and
 > come up with an "embedded contents too large" status code?

We have created small icons with a resolution of 32 x 24 pixels using 4 
bits for depth color whose size was around 500 bytes. In any case, we 
can add text stating the problems of sending too large SIP messages.


 > I think I got somewhat confused that I constructed a close to impossible
 > scenario.  For the sender it is indeed trivial if the sending user
 > selects the file.  Of course, the filename should be included because
 > otherwise the receiver does not easily find out whether or not it has
 > a copy of the file already.  If the receiving users wants to initiate
 > a file transfer, of course, it does not have a checksum at hand to
 > include.

If some scenarios (e.g., file sharing) you will have the hash of the 
file you want, but you will not have the file yet.

 >> I do not see any issue with having simultaneous media streams. It is 
again regular SDP behavior.
 >
 > Well, I assume that if I send you a second offer within a session
 > and you also send me one at the same time, we will have to do some
 > conflict resolution since both of us will use the (n+1)st m-line
 > for potentially different purposes.  Since this has probably not
 > received much implementation attention in the past (and happened
 > rather rarely in practice), it have not been noticed.  But it may
 > be specified and just did not find this upon a quick search.

That conflict resolution is already defined at the offer/answer level. 
Of course, we do not intend to redefine that behavior.

Thanks,

Gonzalo

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Hi James,

This is described in RFC 4091 and RFC 4092:
ftp://ftp.rfc-editor.org/in-notes/rfc4091.txt
ftp://ftp.rfc-editor.org/in-notes/rfc4092.txt

Cheers

Magnus

James M. Polk wrote:
> Hey
> 
> Can a SIP INVITE, carrying SDP have a "c=" line include both a IPv4 
> address and an IPv6 address?
> 
> The intent is for the offerer to offer both choices, and the answerer 
> answer with IPv6 if they have that stack (or both stacks), and v4 if 
> they only have this one stack.
> 
> Is this example given/rules stated in any RFC?  I don't see this usecase 
> in 3264.
> 
> Is there a play on whether the "c=" line is above the "m=" line, or 
> within one/each media description?
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 


-- 

Magnus Westerlund

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

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Hi,

note that the policy we currently use is not to cross-post to the SIP, 
SIPPING, and SIMPLE mailing lists. A particular discussion should take 
place in one and only one of those lists.

In any case, in the last IETF it was decided that this draft will be 
discussed in MMUSIC. I have cc:ed the MMUSIC mailing list in this 
message. Please, remove the SIPPING and SIMPLE mailing lists in your 
replies from now on.

Thanks,

Gonzalo
SIPPING co-chair


Amitkumar.Goel@infineon.com wrote:
> Hi Miguel,
> 
> If SDP offer/answer m-line ulr does not contain session id,as session ID
> is a optional param, Is it mean that the offerer/answerer does not
> support MSRP session multiplexing for sharing the same TCP connection? 
> 
> Regards,
> Amit 
> 
>  
> 
> -----Original Message-----
> From: Sudha [mailto:sudha.vs@globaledgesoft.com] 
> Sent: Friday, April 28, 2006 1:02 PM
> To: Miguel Garcia
> Cc: Goel Amitkumar (IFIN SW WS); sipping@ietf.org; simple@ietf.org;
> OMA-MWG-IM@mail.openmobilealliance.org
> Subject: Re: [Simple] Handling of incoming INVITE request for different
> MSRP session
> 
> Thanks Miguel.
> Inline discussion
> 
> 
> On Fri, 28 Apr 2006 11:46:42 +0530, Miguel Garcia
> <Miguel.An.Garcia@nokia.com> wrote:
> 
>> Inline discussion.
>>
>> Sudha wrote:
>>> Hi Miguel,
>>> -Does "m=" lines have different port numbers?
>>>     If so, it would result in 2 TCP Connections and hence 2 MSRP 
>>> Sessions. This would naturally result in two "a=path:" attributes 
>>> which would complicate the message parsing.
>>>     else if the media lines have the same port numbers it is like 
>>> duplication of the previous media line and it doesnt carry any
> meaning.
>> Different m= lines will have the same port number, because they will 
>> be sharing the same TCP connection. There is nothing new here, the 
>> MSRP draft already discusses the TCP sharing in Sections 5.1 and 8.1.
>>
>> What we say in our draft is that each file transfer is identified with
> 
>> a different MSRP session. We can discuss if we need to share the same 
>> session, but at the moment the draft says each m= line has a unique 
>> MSRP sessionId. Still the port number used for the TCP connection is
> the same.
> 
> With reference to sec. 8.1 of draft-ietf-simple-message-sessions-14.txt
> the format of the m= line is:
> m=<media> <port> <protocol> <format list> The format list field MUST be
> set to "*".
> The media doesn't accomadate for the MSRP sessionId.
> The SessionId is specified in the url of the "a=path" attribute. Can the
> session Id be specified in the format list?
> 
>>>  -How to distinguish between which media line is for normal MSRP Chat
> 
>>> and MSRP File Transfer.
>> MSRP for File Transfer will have a file selector in the SDP. The file 
>> selector is described in Section 6.1 of the File Transfer draft.
>>
>>>  -In the previous mail you had stated that, "Each m=message 
>>> represents an MSRP media stream. Both can use the same MSRP session 
>>> (TCP connection + session-id in MSRP), but they are different media 
>>> streams, just multiplexed over the same TCP connection and MSRP 
>>> session."
>>>
>> Note that sharing the same MSRP session is not correct, at least,  
>> according to what we have in the draft now, my apologies.
>>
>>> It is possible to mutliplex MSRP Sessions(Identified by the session
> Id)  
>>> over the same TCP Connection. But I am not very clear with what you  
>>> meant here by "MSRP media stream". Can you please give some info on
> how  
>>> to identify the media streams in the "m=" lines.
>> That is probably the main point, we need a unique identifier, and that
> 
>> is the session Id in MSRP. That is the reason why the session ID has
> to  
>> be different.
>>
>>
>> BR,
>>
>>          Miguel
>>
>>>  Regds,
>>> Sudha.
>>>   On Thu, 27 Apr 2006 19:56:01 +0530, Miguel Garcia  
>>> <Miguel.An.Garcia@nokia.com> wrote:
>>>
>>>> Hi Sudha:
>>>>
>>>> Inline discussion.
>>>>
>>>> Sudha wrote:
>>>>> Hi all,
>>>>> Is it valid to have two m= lines in a single INVITE?
>>>> Yes, of course.
>>>>
>>>>> If such is a case, Is it valid to have two MSRP sessions created
> for  
>>>>> each of the media line?
>>>> yes, why not?
>>>>
>>>>> As per the draft on File transfer using MSRP, once the File
> transfer  
>>>>> is completed, the MSRP Session associated with it will be closed.
> So  
>>>>> it should be apt to have 2 sessions for each of the media line.
>>>> I am not sure what you mean with "2 sessions fr each of the media  
>>>> line". Each m=message represents an MSRP media stream. Both can use
> 
>>>> the same MSRP session (TCP connection + session-id in MSRP), but
> they  
>>>> are different media streams, just multiplexed over the same TCP  
>>>> connection and MSRP session.
>>>>
>>>> BR,
>>>>
>>>>        Miguel
>>>>
>>>>
>>>>>  Regds,
>>>>> Sudha
>>>>>  On Thu, 27 Apr 2006 19:19:52 +0530, Miguel Garcia  
>>>>> <Miguel.An.Garcia@nokia.com> wrote:
>>>>>
>>>>>> Hi Amit:
>>>>>>
>>>>>> As you correctly mentioned, the spirit of the file transfer draft
> is  
>>>>>> to reuse existing MSRP sessions, if they are available. But we  
>>>>>> always signal them in different m= lines. Notice that in the
> example  
>>>>>> you provided below there is only one m= line (for file transfer),
> so  
>>>>>> if you wanted to have chat and file transfer, you should have two
> 
>>>>>> m=message lines, one is a regular MSRP (chat), the other specific
> 
>>>>>> for file transfer.
>>>>>>
>>>>>> So, your question... "how do we handle a case where we receive a
> SIP
>>>>>> INVITE with a SDP offer containing both types of attributes (for  
>>>>>> text messaging and file transfer) in a single m-line? " seems to
> be  
>>>>>> not applicable. We will have two m=message line, each one can have
> 
>>>>>> its own direction attribute if needed.
>>>>>>
>>>>>> BR,
>>>>>>
>>>>>>         Miguel
>>>>>>
>>>>>>
>>>>>> simple-bounces@ietf.org wrote:
>>>>>>> Hi Garcia, Ben and all,
>>>>>>>  draft-garcia-sipping-file-transfer-mech-00.txt mandates that
> there
>>>>>>> should be separate msrp session (separate m-line) for the file  
>>>>>>> transfer.
>>>>>>> But draft-ietf-simple-message-sessions-14.txt does not really
>>>>>>> differentiate between normal chat session and file transfer
> session.
>>>>>>> Also, it also does not mention the direction attribute. So User
> can  
>>>>>>> also
>>>>>>> have uni directional MSRP session.  If user wants to create a  
>>>>>>> session for chat and file transfer in the
>>>>>>> single SIP INVITE. Considering the above mentioned drafts, our
>>>>>>> understanding is that we will get two different m-lines in SDP  
>>>>>>> offer for
>>>>>>> msrp sessions.  In such a scenario, how do we handle a case where
> 
>>>>>>> we receive a SIP
>>>>>>> INVITE with a SDP offer containing both types of attributes (for
> 
>>>>>>> text
>>>>>>> messaging and file transfer) in a single m-line? For example if
> we
>>>>>>> receive an OFFER.
>>>>>>>      v=0
>>>>>>>       o=alice 2890844526 2890844526 IN IP4
> host.atlanta.example.com
>>>>>>>       s=
>>>>>>>       c=IN IP4 host.atlanta.example.com
>>>>>>>       t=0 0
>>>>>>>       m=message 7654 TCP/MSRP *
>>>>>>>     a=sendonly
>>>>>>>       a=accept-types:text/plain *
>>>>>>>       a=path:msrp://atlanta.example.com:7654/jshA7we;tcp
>>>>>>>       a=filename:My cool picture.jpg
>>>>>>>       a=filetype:image/jpeg
>>>>>>>       a=disposition:inline
>>>>>>>       a=filesize:32349
>>>>>>>       a=icon:cid:id2@alicepc.example.com
>>>>>>>       a=hash:SHA 72245fe8653ddaf371362f86d471913ee4a2ce2e
>>>>>>>  Regards,
>>>>>>> Amit
>>>>>>>  _______________________________________________
>>>>>>> Simple mailing list
>>>>>>> Simple@ietf.org
>>>>>>> https://www1.ietf.org/mailman/listinfo/simple
>>>>>>>
>>>>>    --Best Regards,
>>>>> Sudha V.S
>>>    --Best Regards,
>>> Sudha V.S
> 
> 
> 

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo

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Joerg,

regarding the issue about transferring icons in the body of a SIP 
message, note that this was already defined by RFC 3261.

Section 20.11 of RFC 3261 says:

   "The disposition type "icon" indicates that the body part contains an
    image suitable as an iconic representation of the caller or callee
    that could be rendered informationally by a user agent when a message
    has been received, or persistently while a dialog takes place."

You can find the registration of the 'icon' disposition type at:
http://www.iana.org/assignments/mail-cont-disp

Cheers,

Gonzalo

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Continuing the discussion in mmusic...

It is clear that the m= line does not contain the MSRP session-id, which 
is only visible in the a=path line. For example:

    c=IN IP4 atlanta.example.com
    m=message 7654 TCP/MSRP *
    a=accept-types:text/plain
    a=path:msrp://atlanta.example.com:7654/jshA7we;tcp

I don't really understand your question below. MSRP multiplexing over 
the same TCP connection is an essential feature of MSRP, and it is at 
the discretion of the MSRP endpoint to decide whether to multiplex or 
not (although it is recommended to multiplex, otherwise you are 
bypassing the congestion control built into TCP).

So, I don't really see your question about "support for multiplexing"... 
Perhaps you can enlight me.

/Miguel

Amitkumar.Goel@infineon.com wrote:
> Hi Miguel,
> 
> If SDP offer/answer m-line ulr does not contain session id,as session ID
> is a optional param, Is it mean that the offerer/answerer does not
> support MSRP session multiplexing for sharing the same TCP connection? 
> 
> Regards,
> Amit 
> 
>  
> 
>
-- 
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.an.garcia@openlaboratory.net
Nokia Research Center      Helsinki, Finland


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So now posting to MMUSIC and the OMA mailing list.

I have cut the discussion only to one relevant point


Sudha wrote:
> With reference to sec. 8.1 of draft-ietf-simple-message-sessions-14.txt 
> the format of the m= line is:
> m=<media> <port> <protocol> <format list>
> The format list field MUST be set to "*".
> The media doesn't accomadate for the MSRP sessionId.
> The SessionId is specified in the url of the "a=path" attribute. Can the 
> session Id be specified in the format list?
> 

So, what you are saying is that if someone wants to establish a regular 
MSRP session for chatting, and a side-by MSRP session for file transfer, 
and it wants to multiplex both MSRP sessions over the same TCP 
connection, both m=message lines will be equal. True. What is the 
problem? I can't see a problem, one m=message line will have the typical 
MSRP attributes, the other one will contain, in addition to the typical, 
those specific for the file transfer. I don't see the problem.

/Miguel

-- 
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.an.garcia@openlaboratory.net
Nokia Research Center      Helsinki, Finland


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Subject: [MMUSIC] Re: Handling of incoming INVITE request for different MSRP
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Hi all,
There wont be any problem if normal MSRP chat and Filetransfer takes place  
in 2 diffrent MSRP sessions, each of them having their own attributes.

But what I meant to say in the previous discussions is:
-File Transfer and Chat cant take place in single session.

-I mistook the flow of SDP headers with two m= lines in the following way:
     v=0
     o=alice 2890844526 2890844527 IN IP4 alice.example.com
     s= -
     c=IN IP4 alice.example.com
     t=0 0
     m=message 7394 TCP/MSRP *
     m=message 7394 TCP/MSRP *
     a=accept-types:text/plain
     a=path:msrp://alice.example.com:7394/2s93i;tcp
     a=filename:My cool picture.jpg
     a=filetype:image/jpeg
This would certainly create a problem coz, we will not be able to specify,  
the Session Ids of MSRP Sessions and their accept-types.


Sample SDP Message with 2 m= lines:
     v=0
     o=alice 2890844526 2890844527 IN IP4 alice.example.com
     s= -
     c=IN IP4 alice.example.com
     t=0 0
     m=message 7394 TCP/MSRP *
     a=accept-types:text/plain
     a=path:msrp://alice.example.com:7394/2s93i;tcp
     m=message 7394 TCP/MSRP *
     a=path:msrp://atlanta.example.com:7394/jshA7we;tcp
     i=This is my latest picture
     a=sendonly
     a=accept-types:*
     a=filename:My cool picture.jpg
     a=filetype:image/jpeg
     ....

Hope the above flow is valid, and I feel there wont be any problem.

Thanks n Regds,
Sudha.

On Fri, 28 Apr 2006 14:07:21 +0530, Miguel Garcia  
<Miguel.An.Garcia@nokia.com> wrote:

> So now posting to MMUSIC and the OMA mailing list.
>
> I have cut the discussion only to one relevant point
>
>
> Sudha wrote:
>> With reference to sec. 8.1 of draft-ietf-simple-message-sessions-14.txt  
>> the format of the m= line is:
>> m=<media> <port> <protocol> <format list>
>> The format list field MUST be set to "*".
>> The media doesn't accomadate for the MSRP sessionId.
>> The SessionId is specified in the url of the "a=path" attribute. Can  
>> the session Id be specified in the format list?
>>
>
> So, what you are saying is that if someone wants to establish a regular  
> MSRP session for chatting, and a side-by MSRP session for file transfer,  
> and it wants to multiplex both MSRP sessions over the same TCP  
> connection, both m=message lines will be equal. True. What is the  
> problem? I can't see a problem, one m=message line will have the typical  
> MSRP attributes, the other one will contain, in addition to the typical,  
> those specific for the file transfer. I don't see the problem.
>
> /Miguel
>



-- 
Best Regards,
Sudha V.S

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From mmusic-bounces@ietf.org Fri Apr 28 06:22:09 2006
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Subject: [MMUSIC] Re: Handling of incoming INVITE request for different MSRP
	session
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I guess (my guess), that the syntax allows it to identify, for example, 
an MSRP relay. Probably you would need to input the MSRP URL of your 
relay in your endpoint configuration. So from that point of view, it is 
syntactically valid to have an MSRP URL without a session-ID. But when 
you include MSRP URLs in SDP, then you really need the session-id, as 
the draft indicates.

/Miguel

Amitkumar.Goel@infineon.com wrote:
> Thanks Miguel for the clarification. 
> 
> The formal syntax  in section 9 of the MSRP draft mislead me. 
> 
> ...
> ...
> ...
>  MSRP-URL = msrp-scheme "://" [userinfo "@"] hostport
>        ["/" session-id] ";" transport *( ";" url-parameter)
>                         ; userinfo as defined in RFC3986, except
>                         ; limited to unreserved.
>                         ; hostport as defined in RFC3261
> ...
> ...
> 
> Regards,
> Amit
> 
> -----Original Message-----
> From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com] 
> Sent: Friday, April 28, 2006 3:20 PM
> To: Goel Amitkumar (IFIN SW WS)
> Cc: sudha.vs@globaledgesoft.com; mmusic@ietf.org;
> OMA-MWG-IM@mail.openmobilealliance.org
> Subject: Re: Handling of incoming INVITE request for different MSRP
> session
> 
> Amit:
> 
> You are arguing that that MSRP URLs that are included in a=path line may
> not contain session-ids, as you indicated in your example below.
> 
> I would argue that such idea is not possible. While it is true that you
> can create an MSRP URL that does not include the session-id, it is only
> valid as a way to refer to the endpoint. But here, in SDP, we are
> referring not to an endpoint, but to a media stream of MSRP traffic,
> thus, so the MSRP URL it has to be fully qualified with the session-id.
> 
> I would refer to Section 8.2 of the MSRP draft. It mandates the usage of
> session IDs:
> 
>     The rightmost URL in the path attribute MUST identify the endpoint
>     that generated the SDP document, or some other location where that
>     endpoint wishes to receive requests associated with the session.  It
>     MUST be assigned for this particular session, and MUST NOT duplicate
>     any URL in use for any other session in which the endpoint is
>     currently participating.  It SHOULD be hard to guess, and protected
>     from eavesdroppers.  This is discussed in more detail in Section 14.
> 
> So in my opinion your example below is not valid according to MSRP.
> 
> Regards,
> 
>           Miguel
> 
> Amitkumar.Goel@infineon.com wrote:
>> Hi Miguel,
>>
>> Consider the case  when we get sdp offer with two m-line (without 
>> session id)and corresponding attributes.
>>
>>     c=IN IP4 atlanta.example.com
>>     m=message 7654 TCP/MSRP *
>>     a=accept-types:text/plain
>>     a=path:msrp://atlanta.example.com:7654;tcp   
>>
>>     c=IN IP4 atlanta.example.com
>>     m=message 7654 TCP/MSRP *
>>     a=accept-types:*
>>     a=path:msrp://atlanta.example.com:7654;tcp
>>     a=filename:My cool picture.jpg
>>     a=filetype:image/jpeg
>>     a=disposition:inline
>>     a=filesize:32349
>>     a=icon:cid:id2@alicepc.example.com
>>     a=hash:SHA 72245fe8653ddaf371362f86d471913ee4a2ce2e
>>
>> Here both the path attributes are without session id.
>>  
>> Using this offer, 2 MSRP sessions are created. When MSRP data is 
>> received over the TCP connection, how will I demultiplex MSRP data? To
> 
>> identify the recipient session for the received MSRP data, I think we 
>> must add session id to the MSRP URL. Hope I am clear this time.
>>
>>
>> Thanks and Regards,
>> Amit Kumar Goel
>>
>> -----Original Message-----
>> From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com]
>> Sent: Friday, April 28, 2006 2:19 PM
>> To: Goel Amitkumar (IFIN SW WS)
>> Cc: sudha.vs@globaledgesoft.com; mmusic@ietf.org; 
>> OMA-MWG-IM@mail.openmobilealliance.org
>> Subject: Re: Handling of incoming INVITE request for different MSRP 
>> session
>>
>> Continuing the discussion in mmusic...
>>
>> It is clear that the m= line does not contain the MSRP session-id, 
>> which is only visible in the a=path line. For example:
>>
>>     c=IN IP4 atlanta.example.com
>>     m=message 7654 TCP/MSRP *
>>     a=accept-types:text/plain
>>     a=path:msrp://atlanta.example.com:7654/jshA7we;tcp
>>
>> I don't really understand your question below. MSRP multiplexing over 
>> the same TCP connection is an essential feature of MSRP, and it is at 
>> the discretion of the MSRP endpoint to decide whether to multiplex or 
>> not (although it is recommended to multiplex, otherwise you are 
>> bypassing the congestion control built into TCP).
>>
>> So, I don't really see your question about "support for
> multiplexing"...
>> Perhaps you can enlight me.
>>
>> /Miguel
>>
>> Amitkumar.Goel@infineon.com wrote:
>>> Hi Miguel,
>>>
>>> If SDP offer/answer m-line ulr does not contain session id,as session
> 
>>> ID is a optional param, Is it mean that the offerer/answerer does not
> 
>>> support MSRP session multiplexing for sharing the same TCP
> connection?
>>> Regards,
>>> Amit
>>>
>>>  
>>>
>>>
> 

-- 
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.an.garcia@openlaboratory.net
Nokia Research Center      Helsinki, Finland


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Hi Gonzali,
Refer to inline discussion.

On Fri, 28 Apr 2006 14:50:06 +0530, Gonzalo Camarillo  
<Gonzalo.Camarillo@ericsson.com> wrote:

> Hi Sudha,
>
> I do not fully understand what you mean, but note that you can transfer  
> files within an MSRP session as usual. In that case, you would only need  
> a single 'm' line, as usual.
	Did you mean to say, File transfer can take place in the same session  
which is used for chat? This is quiet impossible coz, you will not be able  
to differentiate which is the data frm the file and chat-data. Further  
MSRP Chat session will be terminated once the file transfer is complete.
>
> If you want to use another m line to transfer the file so that the other  
> party has a chance to accept or reject the transfer, you would add a  
> second 'm' line whose attributes would contain a different MSRP session  
> ID.
>
> I hope this helps,
>
> Cheers,
>
> Gonzalo
>
> Sudha wrote:
>> Hi all,
>> There wont be any problem if normal MSRP chat and Filetransfer takes  
>> place in 2 diffrent MSRP sessions, each of them having their own  
>> attributes.
>>  But what I meant to say in the previous discussions is:
>> -File Transfer and Chat cant take place in single session.
>>  -I mistook the flow of SDP headers with two m= lines in the following  
>> way:
>>     v=0
>>     o=alice 2890844526 2890844527 IN IP4 alice.example.com
>>     s= -
>>     c=IN IP4 alice.example.com
>>     t=0 0
>>     m=message 7394 TCP/MSRP *
>>     m=message 7394 TCP/MSRP *
>>     a=accept-types:text/plain
>>     a=path:msrp://alice.example.com:7394/2s93i;tcp
>>     a=filename:My cool picture.jpg
>>     a=filetype:image/jpeg
>> This would certainly create a problem coz, we will not be able to  
>> specify, the Session Ids of MSRP Sessions and their accept-types.
>>   Sample SDP Message with 2 m= lines:
>>     v=0
>>     o=alice 2890844526 2890844527 IN IP4 alice.example.com
>>     s= -
>>     c=IN IP4 alice.example.com
>>     t=0 0
>>     m=message 7394 TCP/MSRP *
>>     a=accept-types:text/plain
>>     a=path:msrp://alice.example.com:7394/2s93i;tcp
>>     m=message 7394 TCP/MSRP *
>>     a=path:msrp://atlanta.example.com:7394/jshA7we;tcp
>>     i=This is my latest picture
>>     a=sendonly
>>     a=accept-types:*
>>     a=filename:My cool picture.jpg
>>     a=filetype:image/jpeg
>>     ....
>>  Hope the above flow is valid, and I feel there wont be any problem.
>>  Thanks n Regds,
>> Sudha.
>>  On Fri, 28 Apr 2006 14:07:21 +0530, Miguel Garcia  
>> <Miguel.An.Garcia@nokia.com> wrote:
>>
>>> So now posting to MMUSIC and the OMA mailing list.
>>>
>>> I have cut the discussion only to one relevant point
>>>
>>>
>>> Sudha wrote:
>>>> With reference to sec. 8.1 of  
>>>> draft-ietf-simple-message-sessions-14.txt the format of the m= line  
>>>> is:
>>>> m=<media> <port> <protocol> <format list>
>>>> The format list field MUST be set to "*".
>>>> The media doesn't accomadate for the MSRP sessionId.
>>>> The SessionId is specified in the url of the "a=path" attribute. Can  
>>>> the session Id be specified in the format list?
>>>>
>>>
>>> So, what you are saying is that if someone wants to establish a  
>>> regular MSRP session for chatting, and a side-by MSRP session for file  
>>> transfer, and it wants to multiplex both MSRP sessions over the same  
>>> TCP connection, both m=message lines will be equal. True. What is the  
>>> problem? I can't see a problem, one m=message line will have the  
>>> typical MSRP attributes, the other one will contain, in addition to  
>>> the typical, those specific for the file transfer. I don't see the  
>>> problem.
>>>
>>> /Miguel
>>>
>>    --Best Regards,
>> Sudha V.S
>>    
>> ------------------------------------------------------------------------
>>  This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended  
>> solely for the use of the addressee(s). If you are not the intended  
>> recipient, please notify the sender by e-mail and delete the original  
>> message.Global Edge Software Ltd has taken every reasonable precaution  
>> to minimize this risk, but is not liable for any damage you may sustain  
>> as a result of any virus in this e-mail. You should carry out your own  
>> virus checks before opening the e-mail or attachment. Global Edge  
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-- 
Best Regards,
Sudha V.S

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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: [MMUSIC] Re: Handling of incoming INVITE request for different MSRP
	session
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Hi Sudha,

I do not fully understand what you mean, but note that you can transfer 
files within an MSRP session as usual. In that case, you would only need 
a single 'm' line, as usual.

If you want to use another m line to transfer the file so that the other 
party has a chance to accept or reject the transfer, you would add a 
second 'm' line whose attributes would contain a different MSRP session ID.

I hope this helps,

Cheers,

Gonzalo

Sudha wrote:
> Hi all,
> There wont be any problem if normal MSRP chat and Filetransfer takes 
> place in 2 diffrent MSRP sessions, each of them having their own 
> attributes.
> 
> But what I meant to say in the previous discussions is:
> -File Transfer and Chat cant take place in single session.
> 
> -I mistook the flow of SDP headers with two m= lines in the following way:
>     v=0
>     o=alice 2890844526 2890844527 IN IP4 alice.example.com
>     s= -
>     c=IN IP4 alice.example.com
>     t=0 0
>     m=message 7394 TCP/MSRP *
>     m=message 7394 TCP/MSRP *
>     a=accept-types:text/plain
>     a=path:msrp://alice.example.com:7394/2s93i;tcp
>     a=filename:My cool picture.jpg
>     a=filetype:image/jpeg
> This would certainly create a problem coz, we will not be able to 
> specify, the Session Ids of MSRP Sessions and their accept-types.
> 
> 
> Sample SDP Message with 2 m= lines:
>     v=0
>     o=alice 2890844526 2890844527 IN IP4 alice.example.com
>     s= -
>     c=IN IP4 alice.example.com
>     t=0 0
>     m=message 7394 TCP/MSRP *
>     a=accept-types:text/plain
>     a=path:msrp://alice.example.com:7394/2s93i;tcp
>     m=message 7394 TCP/MSRP *
>     a=path:msrp://atlanta.example.com:7394/jshA7we;tcp
>     i=This is my latest picture
>     a=sendonly
>     a=accept-types:*
>     a=filename:My cool picture.jpg
>     a=filetype:image/jpeg
>     ....
> 
> Hope the above flow is valid, and I feel there wont be any problem.
> 
> Thanks n Regds,
> Sudha.
> 
> On Fri, 28 Apr 2006 14:07:21 +0530, Miguel Garcia 
> <Miguel.An.Garcia@nokia.com> wrote:
> 
>> So now posting to MMUSIC and the OMA mailing list.
>>
>> I have cut the discussion only to one relevant point
>>
>>
>> Sudha wrote:
>>> With reference to sec. 8.1 of 
>>> draft-ietf-simple-message-sessions-14.txt the format of the m= line is:
>>> m=<media> <port> <protocol> <format list>
>>> The format list field MUST be set to "*".
>>> The media doesn't accomadate for the MSRP sessionId.
>>> The SessionId is specified in the url of the "a=path" attribute. Can 
>>> the session Id be specified in the format list?
>>>
>>
>> So, what you are saying is that if someone wants to establish a 
>> regular MSRP session for chatting, and a side-by MSRP session for file 
>> transfer, and it wants to multiplex both MSRP sessions over the same 
>> TCP connection, both m=message lines will be equal. True. What is the 
>> problem? I can't see a problem, one m=message line will have the 
>> typical MSRP attributes, the other one will contain, in addition to 
>> the typical, those specific for the file transfer. I don't see the 
>> problem.
>>
>> /Miguel
>>
> 
> 
> 
> --Best Regards,
> Sudha V.S
> 
> 
> ------------------------------------------------------------------------
> 
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Subject: [MMUSIC] RE: Handling of incoming INVITE request for different MSRP
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Hi Miguel,

Consider the case  when we get sdp offer with two m-line (without
session id)and corresponding attributes. =20

    c=3DIN IP4 atlanta.example.com
    m=3Dmessage 7654 TCP/MSRP *
    a=3Daccept-types:text/plain
    a=3Dpath:msrp://atlanta.example.com:7654;tcp  =20

    c=3DIN IP4 atlanta.example.com
    m=3Dmessage 7654 TCP/MSRP *
    a=3Daccept-types:*
    a=3Dpath:msrp://atlanta.example.com:7654;tcp
    a=3Dfilename:My cool picture.jpg
    a=3Dfiletype:image/jpeg
    a=3Ddisposition:inline
    a=3Dfilesize:32349
    a=3Dicon:cid:id2@alicepc.example.com
    a=3Dhash:SHA 72245fe8653ddaf371362f86d471913ee4a2ce2e

Here both the path attributes are without session id.
=20
Using this offer, 2 MSRP sessions are created. When MSRP data is
received over the TCP connection, how will I demultiplex MSRP data? To
identify the recipient session for the received MSRP data, I think we
must add session id to the MSRP URL. Hope I am clear this time.


Thanks and Regards,
Amit Kumar Goel

-----Original Message-----
From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com]=20
Sent: Friday, April 28, 2006 2:19 PM
To: Goel Amitkumar (IFIN SW WS)
Cc: sudha.vs@globaledgesoft.com; mmusic@ietf.org;
OMA-MWG-IM@mail.openmobilealliance.org
Subject: Re: Handling of incoming INVITE request for different MSRP
session

Continuing the discussion in mmusic...

It is clear that the m=3D line does not contain the MSRP session-id, =
which
is only visible in the a=3Dpath line. For example:

    c=3DIN IP4 atlanta.example.com
    m=3Dmessage 7654 TCP/MSRP *
    a=3Daccept-types:text/plain
    a=3Dpath:msrp://atlanta.example.com:7654/jshA7we;tcp

I don't really understand your question below. MSRP multiplexing over
the same TCP connection is an essential feature of MSRP, and it is at
the discretion of the MSRP endpoint to decide whether to multiplex or
not (although it is recommended to multiplex, otherwise you are
bypassing the congestion control built into TCP).

So, I don't really see your question about "support for multiplexing"...

Perhaps you can enlight me.

/Miguel

Amitkumar.Goel@infineon.com wrote:
> Hi Miguel,
>=20
> If SDP offer/answer m-line ulr does not contain session id,as session=20
> ID is a optional param, Is it mean that the offerer/answerer does not=20
> support MSRP session multiplexing for sharing the same TCP connection?
>=20
> Regards,
> Amit
>=20
> =20
>=20
>
--=20
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.an.garcia@openlaboratory.net
Nokia Research Center      Helsinki, Finland

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Subject: [MMUSIC] Re: Handling of incoming INVITE request for different MSRP
	session
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Amit:

You are arguing that that MSRP URLs that are included in a=path line may 
not contain session-ids, as you indicated in your example below.

I would argue that such idea is not possible. While it is true that you 
can create an MSRP URL that does not include the session-id, it is only 
valid as a way to refer to the endpoint. But here, in SDP, we are 
referring not to an endpoint, but to a media stream of MSRP traffic, 
thus, so the MSRP URL it has to be fully qualified with the session-id.

I would refer to Section 8.2 of the MSRP draft. It mandates the usage of 
session IDs:

    The rightmost URL in the path attribute MUST identify the endpoint
    that generated the SDP document, or some other location where that
    endpoint wishes to receive requests associated with the session.  It
    MUST be assigned for this particular session, and MUST NOT duplicate
    any URL in use for any other session in which the endpoint is
    currently participating.  It SHOULD be hard to guess, and protected
    from eavesdroppers.  This is discussed in more detail in Section 14.

So in my opinion your example below is not valid according to MSRP.

Regards,

          Miguel

Amitkumar.Goel@infineon.com wrote:
> Hi Miguel,
> 
> Consider the case  when we get sdp offer with two m-line (without
> session id)and corresponding attributes.  
> 
>     c=IN IP4 atlanta.example.com
>     m=message 7654 TCP/MSRP *
>     a=accept-types:text/plain
>     a=path:msrp://atlanta.example.com:7654;tcp   
> 
>     c=IN IP4 atlanta.example.com
>     m=message 7654 TCP/MSRP *
>     a=accept-types:*
>     a=path:msrp://atlanta.example.com:7654;tcp
>     a=filename:My cool picture.jpg
>     a=filetype:image/jpeg
>     a=disposition:inline
>     a=filesize:32349
>     a=icon:cid:id2@alicepc.example.com
>     a=hash:SHA 72245fe8653ddaf371362f86d471913ee4a2ce2e
> 
> Here both the path attributes are without session id.
>  
> Using this offer, 2 MSRP sessions are created. When MSRP data is
> received over the TCP connection, how will I demultiplex MSRP data? To
> identify the recipient session for the received MSRP data, I think we
> must add session id to the MSRP URL. Hope I am clear this time.
> 
> 
> Thanks and Regards,
> Amit Kumar Goel
> 
> -----Original Message-----
> From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com] 
> Sent: Friday, April 28, 2006 2:19 PM
> To: Goel Amitkumar (IFIN SW WS)
> Cc: sudha.vs@globaledgesoft.com; mmusic@ietf.org;
> OMA-MWG-IM@mail.openmobilealliance.org
> Subject: Re: Handling of incoming INVITE request for different MSRP
> session
> 
> Continuing the discussion in mmusic...
> 
> It is clear that the m= line does not contain the MSRP session-id, which
> is only visible in the a=path line. For example:
> 
>     c=IN IP4 atlanta.example.com
>     m=message 7654 TCP/MSRP *
>     a=accept-types:text/plain
>     a=path:msrp://atlanta.example.com:7654/jshA7we;tcp
> 
> I don't really understand your question below. MSRP multiplexing over
> the same TCP connection is an essential feature of MSRP, and it is at
> the discretion of the MSRP endpoint to decide whether to multiplex or
> not (although it is recommended to multiplex, otherwise you are
> bypassing the congestion control built into TCP).
> 
> So, I don't really see your question about "support for multiplexing"...
> 
> Perhaps you can enlight me.
> 
> /Miguel
> 
> Amitkumar.Goel@infineon.com wrote:
>> Hi Miguel,
>>
>> If SDP offer/answer m-line ulr does not contain session id,as session 
>> ID is a optional param, Is it mean that the offerer/answerer does not 
>> support MSRP session multiplexing for sharing the same TCP connection?
>>
>> Regards,
>> Amit
>>
>>  
>>
>>

-- 
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.an.garcia@openlaboratory.net
Nokia Research Center      Helsinki, Finland


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Subject: [MMUSIC] RE: Handling of incoming INVITE request for different MSRP
	session
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Thanks Miguel for the clarification.=20

The formal syntax  in section 9 of the MSRP draft mislead me.=20

...
...
...
 MSRP-URL =3D msrp-scheme "://" [userinfo "@"] hostport
       ["/" session-id] ";" transport *( ";" url-parameter)
                        ; userinfo as defined in RFC3986, except
                        ; limited to unreserved.
                        ; hostport as defined in RFC3261
...
...

Regards,
Amit

-----Original Message-----
From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com]=20
Sent: Friday, April 28, 2006 3:20 PM
To: Goel Amitkumar (IFIN SW WS)
Cc: sudha.vs@globaledgesoft.com; mmusic@ietf.org;
OMA-MWG-IM@mail.openmobilealliance.org
Subject: Re: Handling of incoming INVITE request for different MSRP
session

Amit:

You are arguing that that MSRP URLs that are included in a=3Dpath line =
may
not contain session-ids, as you indicated in your example below.

I would argue that such idea is not possible. While it is true that you
can create an MSRP URL that does not include the session-id, it is only
valid as a way to refer to the endpoint. But here, in SDP, we are
referring not to an endpoint, but to a media stream of MSRP traffic,
thus, so the MSRP URL it has to be fully qualified with the session-id.

I would refer to Section 8.2 of the MSRP draft. It mandates the usage of
session IDs:

    The rightmost URL in the path attribute MUST identify the endpoint
    that generated the SDP document, or some other location where that
    endpoint wishes to receive requests associated with the session.  It
    MUST be assigned for this particular session, and MUST NOT duplicate
    any URL in use for any other session in which the endpoint is
    currently participating.  It SHOULD be hard to guess, and protected
  From mmusic-bounces@ietf.org Sat Apr 29 07:51:41 2006
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Subject: [MMUSIC] RE: Handling of incoming INVITE request for different MSRP
	session
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Thanks Miguel for the clarification.=20

The formal syntax  in section 9 of the MSRP draft mislead me.=20

...
...
...
 MSRP-URL =3D msrp-scheme "://" [userinfo "@"] hostport
       ["/" session-id] ";" transport *( ";" url-parameter)
                        ; userinfo as defined in RFC3986, except
                        ; limited to unreserved.
                        ; hostport as defined in RFC3261
...
...

Regards,
Amit

-----Original Message-----
From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com]=20
Sent: Friday, April 28, 2006 3:20 PM
To: Goel Amitkumar (IFIN SW WS)
Cc: sudha.vs@globaledgesoft.com; mmusic@ietf.org;
OMA-MWG-IM@mail.openmobilealliance.org
Subject: Re: Handling of incoming INVITE request for different MSRP
session

Amit:

You are arguing that that MSRP URLs that are included in a=3Dpath line =
may
not contain session-ids, as you indicated in your example below.

I would argue that such idea is not possible. While it is true that you
can create an MSRP URL that does not include the session-id, it is only
valid as a way to refer to the endpoint. But here, in SDP, we are
referring not to an endpoint, but to a media stream of MSRP traffic,
thus, so the MSRP URL it has to be fully qualified with the session-id.

I would refer to Section 8.2 of the MSRP draft. It mandates the usage of
session IDs:

    The rightmost URL in the path attribute MUST identify the endpoint
    that generated the SDP document, or some other location where that
    endpoint wishes to receive requests associated with the session.  It
    MUST be assigned for this particular session, and MUST NOT duplicate
    any URL in use for any other session in which the endpoint is
    currently participating.  It SHOULD be hard to guess, and protected
    from eavesdroppers.  This is discussed in more detail in Section 14.

So in my opinion your example below is not valid according to MSRP.

Regards,

          Miguel

Amitkumar.Goel@infineon.com wrote:
> Hi Miguel,
>=20
> Consider the case  when we get sdp offer with two m-line (without=20
> session id)and corresponding attributes.
>=20
>     c=3DIN IP4 atlanta.example.com
>     m=3Dmessage 7654 TCP/MSRP *
>     a=3Daccept-types:text/plain
>     a=3Dpath:msrp://atlanta.example.com:7654;tcp  =20
>=20
>     c=3DIN IP4 atlanta.example.com
>     m=3Dmessage 7654 TCP/MSRP *
>     a=3Daccept-types:*
>     a=3Dpath:msrp://atlanta.example.com:7654;tcp
>     a=3Dfilename:My cool picture.jpg
>     a=3Dfiletype:image/jpeg
>     a=3Ddisposition:inline
>     a=3Dfilesize:32349
>     a=3Dicon:cid:id2@alicepc.example.com
>     a=3Dhash:SHA 72245fe8653ddaf371362f86d471913ee4a2ce2e
>=20
> Here both the path attributes are without session id.
> =20
> Using this offer, 2 MSRP sessions are created. When MSRP data is=20
> received over the TCP connection, how will I demultiplex MSRP data? To

> identify the recipient session for the received MSRP data, I think we=20
> must add session id to the MSRP URL. Hope I am clear this time.
>=20
>=20
> Thanks and Regards,
> Amit Kumar Goel
>=20
> -----Original Message-----
> From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com]
> Sent: Friday, April 28, 2006 2:19 PM
> To: Goel Amitkumar (IFIN SW WS)
> Cc: sudha.vs@globaledgesoft.com; mmusic@ietf.org;=20
> OMA-MWG-IM@mail.openmobilealliance.org
> Subject: Re: Handling of incoming INVITE request for different MSRP=20
> session
>=20
> Continuing the discussion in mmusic...
>=20
> It is clear that the m=3D line does not contain the MSRP session-id,=20
> which is only visible in the a=3Dpath line. For example:
>=20
>     c=3DIN IP4 atlanta.example.com
>     m=3Dmessage 7654 TCP/MSRP *
>     a=3Daccept-types:text/plain
>     a=3Dpath:msrp://atlanta.example.com:7654/jshA7we;tcp
>=20
> I don't really understand your question below. MSRP multiplexing over=20
> the same TCP connection is an essential feature of MSRP, and it is at=20
> the discretion of the MSRP endpoint to decide whether to multiplex or=20
> not (although it is recommended to multiplex, otherwise you are=20
> bypassing the congestion control built into TCP).
>=20
> So, I don't really see your question about "support for
multiplexing"...
>=20
> Perhaps you can enlight me.
>=20
> /Miguel
>=20
> Amitkumar.Goel@infineon.com wrote:
>> Hi Miguel,
>>
>> If SDP offer/answer m-line ulr does not contain session id,as session

>> ID is a optional param, Is it mean that the offerer/answerer does not

>> support MSRP session multiplexing for sharing the same TCP
connection?
>>
>> Regards,
>> Amit
>>
>> =20
>>
>>

--=20
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.an.garcia@openlaboratory.net
Nokia Research Center      Helsinki, Finland

_______________________________________________
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	Fri,  from eavesdroppers.  This is discussed in more detail in Section 14.

So in my opinion your example below is not valid according to MSRP.

Regards,

          Miguel

Amitkumar.Goel@infineon.com wrote:
> Hi Miguel,
>=20
> Consider the case  when we get sdp offer with two m-line (without=20
> session id)and corresponding attributes.
>=20
>     c=3DIN IP4 atlanta.example.com
>     m=3Dmessage 7654 TCP/MSRP *
>     a=3Daccept-types:text/plain
>     a=3Dpath:msrp://atlanta.example.com:7654;tcp  =20
>=20
>     c=3DIN IP4 atlanta.example.com
>     m=3Dmessage 7654 TCP/MSRP *
>     a=3Daccept-types:*
>     a=3Dpath:msrp://atlanta.example.com:7654;tcp
>     a=3Dfilename:My cool picture.jpg
>     a=3Dfiletype:image/jpeg
>     a=3Ddisposition:inline
>     a=3Dfilesize:32349
>     a=3Dicon:cid:id2@alicepc.example.com
>     a=3Dhash:SHA 72245fe8653ddaf371362f86d471913ee4a2ce2e
>=20
> Here both the path attributes are without session id.
> =20
> Using this offer, 2 MSRP sessions are created. When MSRP data is=20
> received over the TCP connection, how will I demultiplex MSRP data? To

> identify the recipient session for the received MSRP data, I think we=20
> must add session id to the MSRP URL. Hope I am clear this time.
>=20
>=20
> Thanks and Regards,
> Amit Kumar Goel
>=20
> -----Original Message-----
> From: Miguel Garcia [mailto:Miguel.An.Garcia@nokia.com]
> Sent: Friday, April 28, 2006 2:19 PM
> To: Goel Amitkumar (IFIN SW WS)
> Cc: sudha.vs@globaledgesoft.com; mmusic@ietf.org;=20
> OMA-MWG-IM@mail.openmobilealliance.org
> Subject: Re: Handling of incoming INVITE request for different MSRP=20
> session
>=20
> Continuing the discussion in mmusic...
>=20
> It is clear that the m=3D line does not contain the MSRP session-id,=20
> which is only visible in the a=3Dpath line. For example:
>=20
>     c=3DIN IP4 atlanta.example.com
>     m=3Dmessage 7654 TCP/MSRP *
>     a=3Daccept-types:text/plain
>     a=3Dpath:msrp://atlanta.example.com:7654/jshA7we;tcp
>=20
> I don't really understand your question below. MSRP multiplexing over=20
> the same TCP connection is an essential feature of MSRP, and it is at=20
> the discretion of the MSRP endpoint to decide whether to multiplex or=20
> not (although it is recommended to multiplex, otherwise you are=20
> bypassing the congestion control built into TCP).
>=20
> So, I don't really see your question about "support for
multiplexing"...
>=20
> Perhaps you can enlight me.
>=20
> /Miguel
>=20
> Amitkumar.Goel@infineon.com wrote:
>> Hi Miguel,
>>
>> If SDP offer/answer m-line ulr does not contain session id,as session

>> ID is a optional param, Is it mean that the offerer/answerer does not

>> support MSRP session multiplexing for sharing the same TCP
connection?
>>
>> Regards,
>> Amit
>>
>> =20
>>
>>

--=20
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.an.garcia@openlaboratory.net
Nokia Research Center      Helsinki, Finland

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

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Hi Joerg,

> moving this to MMUSIC:

perfect.

> Regarding the
> encoding, I suppose we will have to worry about how to put this into
> SDP.  This brings up -- again -- the discussion of whether to include
> some "blob" of, from an SDP perspective, opaque metadata (e.g., b64
> encoded zip version of the FLUTE FDT, just to give some weird example)
> or whether to make these attributes explicit.

I would go for explicit SDP attributed (i.e., 'a' lines).

> If we make the attributes explicit, this is essentially a more general
> aspect of describing the purpuse of a media stream and would not be
> restricted to file transfer.  We have some informal hints for audio
> and video already and this appears conceptually similar in that it
> identifies the purpose and helps the answerer to decide whether to
> accept it or not.
> 
> Keeping this in mind, it would be extremely useful, not to invent
> independent sets of attributes for file transfer, MSRP sessions,
> audio, video, whiteboard, and so so.

The attributes defined in the draft describe a file to be transfered. If 
somebody comes up with different attributes to describe the contents of 
the audio being sent over a media stream, those will not have any 
interaction with our attributes. Different types of media streams need 
different attributes to describe their contents.

> I clearly _do_not_ want to end up in an XCON-style discussion about
> metadata and the like, but it may be worthwhile given some
> consideration to future evolution of this stuff and compatibility
> with existing attributes.

We have not found any interactions with existing attributes.

>>> I am also concerned about motivating the use of inline images for
>>> icons.  I can hardly see why you would need an icon to decide
>>> whether or not to accept a file transfer.
>>
>> A lot of commercial IM systems support it, and users find it very 
>> useful. SIP-based systems need to be able to provide a similar user 
>> experience if we do not want people to start doing things is 
>> proprietary ways.
> 
> It's you 28 Apr 2006 09:37:14 +0200
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Hi Joerg,

> moving this to MMUSIC:

perfect.

> Regarding the
> encoding, I suppose we will have to worry about how to put this into
> SDP.  This brings up -- again -- the discussion of whether to include
> some "blob" of, from an SDP perspective, opaque metadata (e.g., b64
> encoded zip version of the FLUTE FDT, just to give some weird example)
> or whether to make these attributes explicit.

I would go for explicit SDP attributed (i.e., 'a' lines).

> If we make the attributes explicit, this is essentially a more general
> aspect of describing the purpuse of a media stream and would not be
> restricted to file transfer.  We have some informal hints for audio
> and video already and this appears conceptually similar in that it
> identifies the purpose and helps the answerer to decide whether to
> accept it or not.
> 
> Keeping this in mind, it would be extremely useful, not to invent
> independent sets of attributes for file transfer, MSRP sessions,
> audio, video, whiteboard, and so so.

The attributes defined in the draft describe a file to be transfered. If 
somebody comes up with different attributes to describe the contents of 
the audio being sent over a media stream, those will not have any 
interaction with our attributes. Different types of media streams need 
different attributes to describe their contents.

> I clearly _do_not_ want to end up in an XCON-style discussion about
> metadata and the like, but it may be worthwhile given some
> consideration to future evolution of this stuff and compatibility
> with existing attributes.

We have not found any interactions with existing attributes.

>>> I am also concerned about motivating the use of inline images for
>>> icons.  I can hardly see why you would need an icon to decide
>>> whether or not to accept a file transfer.
>>
>> A lot of commercial IM systems support it, and users find it very 
>> useful. SIP-based systems need to be able to provide a similar user 
>> experience if we do not want people to start doing things is 
>> proprietary ways.
> 
> It's your call in the end of the day.  Often, icons are created based
> upon the filename extension -- and I suspect that the filename is going
> to be much more relevant that the icon in deciding whether or not to
> accept it.  Icons are not worth much and only increase volume.  But if
> we decide that it is useful:

If I want to send you a picture called PCJ00234.JPG, you will not know 
whether or not you want to receive it. However, if you see a small icon 
of the picture, you will be in a better position to make that decision. 
Many people find icons very useful, in many cases much more than the 
file name.

> You must decide how much you worry about the SIP signaling architecture.
> Given that the INFO method is "damned" (which I fully support), I do not
> believe that turning parts of SDP into a transport is a particularly
> smart idea.  It clearly does not ad efficiency.  If you follow
> traditional SIPPING thinking, you will need to separate the desired
> effect from the mechanism to achieve it.  There may be better ways.
> 
> I am simply worried to open a door that we better leave closed.
> Would you want to limit the media type or the size of an icon?
> If so, what would be the right value?  Would you negotiate it and
> come up with an "embedded contents too large" status code?

We have created small icons with a resolution of 32 x 24 pixels using 4 
bits for depth color whose size was around 500 bytes. In any case, we 
can add text stating the problems of sending too large SIP messages.


> I think I got somewhat confused that I constructed a close to impossible
> scenario.  For the sender it is indeed trivial if the sending user
> selects the file.  Of course, the filename should be included because
> otherwise the receiver does not easily find out whether or not it has
> a copy of the file already.  If the receiving users wants to initiate
> a file transfer, of course, it does not have a checksum at hand to
> include.

If some scenarios (e.g., file sharing) you will have the hash of the 
file you want, but you will not have the file yet.

>> I do not see any issue with having simultaneous media streams. It is 
>> again regular SDP behavior.
> 
> Well, I assume that if I send you a second offer within a session
> and you also send me one at the same time, we will have to do some
> conflict resolution since both of us will use the (n+1)st m-line
> for potentially different purposes.  Since this has probably not
> received much implementation attention in the past (and happened
> rather rarely in practice), it have not been noticed.  But it may
> be specified and just did not find this upon a quick search.

That conflict resolution is already defined at the offer/answer level. 
Of course, we do not intend to redefine that behavior.

Thanks,

Gonzalo

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r call in the end of the day.  Often, icons are created based
> upon the filename extension -- and I suspect that the filename is going
> to be much more relevant that the icon in deciding whether or not to
> accept it.  Icons are not worth much and only increase volume.  But if
> we decide that it is useful:

If I want to send you a picture called PCJ00234.JPG, you will not know 
whether or not you want to receive it. However, if you see a small icon 
of the picture, you will be in a better position to make that decision. 
Many people find icons very useful, in many cases much more than the 
file name.

> You must decide how much you worry about the SIP signaling architecture.
> Given that the INFO method is "damned" (which I fully support), I do not
> believe that turning parts of SDP into a transport is a particularly
> smart idea.  It clearly does not ad efficiency.  If you follow
> traditional SIPPING thinking, you will need to separate the desired
> effect from the mechanism to achieve it.  There may be better ways.
> 
> I am simply worried to open a door that we better leave closed.
> Would you want to limit the media type or the size of an icon?
> If so, what would be the right value?  Would you negotiate it and
> come up with an "embedded contents too large" status code?

We have created small icons with a resolution of 32 x 24 pixels using 4 
bits for depth color whose size was around 500 bytes. In any case, we 
can add text stating the problems of sending too large SIP messages.


> I think I got somewhat confused that I constructed a close to impossible
> scenario.  For the sender it is indeed trivial if the sending user
> selects the file.  Of course, the filename should be included because
> otherwise the receiver does not easily find out whether or not it has
> a copy of the file already.  If the receiving users wants to initiate
> a file transfer, of course, it does not have a checksum at hand to
> include.

If some scenarios (e.g., file sharing) you will have the hash of the 
file you want, but you will not have the file yet.

>> I do not see any issue with having simultaneous media streams. It is 
>> again regular SDP behavior.
> 
> Well, I assume that if I send you a second offer within a session
> and you also send me one at the same time, we will have to do some
> conflict resolution since both of us will use the (n+1)st m-line
> for potentially different purposes.  Since this has probably not
> received much implementation attention in the past (and happened
> rather rarely in practice), it have not been noticed.  But it may
> be specified and just did not find this upon a quick search.

That conflict resolution is already defined at the offer/answer level. 
Of course, we do not intend to redefine that behavior.

Thanks,

Gonzalo

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





