
From nobody Mon Mar  3 03:42:54 2014
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From: Yana Stamcheva <yana@jitsi.org>
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Subject: [Stox] Status and agenda update
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Hi all,

We're not currently meeting at IETF 89, but here's a quick status and =
milestones update.=20

Goals and milestones:
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D
Done:
- Feb 12, 2014: draft-ietf-stox-presence in RFC-Editor's Queue =
(http://www.rfc-editor.org/queue2.html#draft-ietf-stox-presence)

- Feb 26, 2014: draft-ietf-stox-core in RFC-Editor's Queue =
(http://www.rfc-editor.org/queue2.html#draft-ietf-stox-core)

Future:=20
- March 10, 2014: WG chairs send to ADs shepherd write-ups for documents =
draft-ietf-stox-chat and draft-ietf-stox-im

- March 14, 2014: Start WGLC on draft-ietf-stox-groupchat

- End of April, 2014: Virtual Interim Meeting focused on =
draft-ietf-stox-media - This means the document should be updated and =
should be addressing the issues raised by Jonathan Lennox and Paul =
Kyzivat by mid April.

- Beginning of June 2014: Start WGLC on draft-ietf-stox-media

- End of June 2014: Submit last document to the IESG

Let the chairs know if you have any comments or suggestions on the =
agenda.

Best regards,
Yana=


From nobody Tue Mar  4 17:35:35 2014
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Date: Wed, 05 Mar 2014 01:35:24 +0000
From: Peter Saint-Andre <stpeter@stpeter.im>
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That all looks good to me. I'll commit to incorporating feedback as 
quickly as it comes in (and to working on the -media document with Emil 
and Saúl in a timely manner, too).

Peter

On 3/3/14, 11:42 AM, Yana Stamcheva wrote:
> Hi all,
>
> We're not currently meeting at IETF 89, but here's a quick status and milestones update.
>
> Goals and milestones:
> ============================
> Done:
> - Feb 12, 2014: draft-ietf-stox-presence in RFC-Editor's Queue (http://www.rfc-editor.org/queue2.html#draft-ietf-stox-presence)
>
> - Feb 26, 2014: draft-ietf-stox-core in RFC-Editor's Queue (http://www.rfc-editor.org/queue2.html#draft-ietf-stox-core)
>
> Future:
> - March 10, 2014: WG chairs send to ADs shepherd write-ups for documents draft-ietf-stox-chat and draft-ietf-stox-im
>
> - March 14, 2014: Start WGLC on draft-ietf-stox-groupchat
>
> - End of April, 2014: Virtual Interim Meeting focused on draft-ietf-stox-media - This means the document should be updated and should be addressing the issues raised by Jonathan Lennox and Paul Kyzivat by mid April.
>
> - Beginning of June 2014: Start WGLC on draft-ietf-stox-media
>
> - End of June 2014: Submit last document to the IESG
>
> Let the chairs know if you have any comments or suggestions on the agenda.
>
> Best regards,
> Yana
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox
>


From nobody Tue Mar 11 15:10:17 2014
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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 SIP-TO-XMPP Working Group of the IETF.

        Title           : Interworking between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP): One-to-One Text Chat Sessions
        Authors         : Peter Saint-Andre
                          Salvatore Loreto
	Filename        : draft-ietf-stox-chat-06.txt
	Pages           : 18
	Date            : 2014-03-11

Abstract:
   This document defines a bidirectional protocol mapping for the
   exchange of instant messages in the context of a one-to-one chat
   session between a user of the Session Initiation Protocol (SIP) and a
   user of the Extensible Messaging and Presence Protocol (XMPP).
   Specifically for SIP text chat, this document specifies a mapping to
   the Message Session Relay Protocol (MSRP).


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-stox-chat/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-stox-chat-06

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-stox-chat-06


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


From nobody Tue Mar 11 15:10:37 2014
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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 SIP-TO-XMPP Working Group of the IETF.

        Title           : Interworking between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP): Instant Messaging
        Authors         : Peter Saint-Andre
                          Avshalom Houri
                          Joe Hildebrand
	Filename        : draft-ietf-stox-im-08.txt
	Pages           : 12
	Date            : 2014-03-11

Abstract:
   This document defines a bidirectional protocol mapping for the
   exchange of single instant messages between the Session Initiation
   Protocol (SIP) and the Extensible Messaging and Presence Protocol
   (XMPP).


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-stox-im/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-stox-im-08

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-stox-im-08


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From nobody Sun Mar 16 14:40:32 2014
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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 SIP-TO-XMPP Working Group of the IETF.

        Title           : Interworking between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP): Groupchat
        Authors         : Peter Saint-Andre
                          Saul Ibarra
                          Salvatore Loreto
	Filename        : draft-ietf-stox-groupchat-03.txt
	Pages           : 35
	Date            : 2014-03-16

Abstract:
   This document defines a bidirectional protocol mapping for the
   exchange of instant messages in the context of a multiparty chat
   session among users of the Session Initiation Protocol (SIP) and
   users of the Extensible Messaging and Presence Protocol (XMPP).
   Specifically, this document defines a mapping between the SIP-based
   Message Session Relay Protocol (MSRP) and the XMPP Multi-User Chat
   (MUC) extension.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-stox-groupchat/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-stox-groupchat-03

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-stox-groupchat-03


Please note that it may take a couple of minutes from the time of submission
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Internet-Drafts are also available by anonymous FTP at:
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From nobody Wed Mar 19 20:39:17 2014
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Paul, thanks for the feedback and my apologies for the extremely slow reply.

On 1/9/14, 10:41 AM, Paul Kyzivat wrote:
> As requested, I have taken a look at treatment of Hold in section 6 of
> draft-ietf-stox-media-02.
>
> The draft says:
>
>     The same semantics are also
>     supported by Jingle through the "senders" element and its "initiator"
>     and "responder" values.
>
>                   [example to follow]
>
>     In addition to these semantics however Jingle also defines a more
>     concise way for achieving the same, which consists in sending a
>     "hold" command within a "session-info" action:
>
> Before I can make any useful comments I need to get educated on some
> aspects of xmpp. (I *browsed* XEP 0166, but couldn't quickly figure this
> out.) Hopefully somebody can clue me in on the following:
>
> 1) I see that the "senders" element has values that correspond 1:1 with
> SDP sendrecv/sendonly/recvonly/inactive. Are there equivalent
> constraints on O/A behavior? Specifically that:
>
>      Offer    Permissible Answer
>      -----    ------------------
>      sendrecv sendrecv/sendonly/recvonly/inactive
>      sendonly recvonly/inactive
>      recvonly sendonly/inactive
>      inactive inactive

Although those constraints look reasonable, I think the original Jingle 
authors were not aware of them when defining the protocol. In general, 
Jingle implementations are simpler than SIP implementations, and might 
tend to assume senders='both' / a=sendrecv. Also, not all uses of the 
'senders' attribute quite follow the offer/answer model - see, for 
instance, the content-modify action:

###

The content-modify action is used to change the direction of an existing 
content definition through modification of the 'senders' attribute. If 
the recipient deems the directionality of a content-modify action to be 
unacceptable, it MAY reply with a contrary content-modify action, 
terminate the session, or simply refuse to send or accept application 
data in the new direction. In any case, the recipient MUST NOT send a 
content-accept action in response to the content-modify.

###

> 2) Are the "senders" element values semantically equivalent to the
> corresponding SDP attributes in their implications to how the
> corresponding RTP stream is used?

As far as I understand those implications, yes.

> 3)) What are the precise semantics of the "hold" command within a
> "session-info" action?

Those are defined in XEP-0167 for RTP sessions (not in the core Jingle 
spec):

###

The <hold/> payload indicates that the principal is temporarily not 
listening for media from the other party. It is RECOMMENDED for the 
parties to handle informational <hold/> messages as follows (where the 
holdee is the party that receives the hold message and the holder is the 
party that sends the hold message):

   *  The holdee SHOULD stop sending media.
   *  The holdee MUST keep accepting media (this ensures that the holder 
can immediately start sending media again when switching back from hold 
to active, or can send hold music or other media).
   *  The holder MAY continue to send media (e.g. hold music).
   *  The holder MAY silently drop all media that it receives from the 
holdee.

###

- http://xmpp.org/extensions/xep-0167.html#info-hold

> Is it formally equivalent to "initiator" on every
> media session?

Hmm, not quite, because either party can send a hold payload, and the 
semantics of the hold depend on who is sending it.

In any case, as I recall from the discussion in Vancouver, we were 
trying to formulate a recommended approach based on the behavior of 
existing Jingle implementations. I think we still need to do a bit of 
field research to figure out what's most prevalent (e.g., using 
'senders' or the informational hold message).

Emil, what does Jitsi do? Are you aware of how other implementations behave?

Peter


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Subject: Re: [Stox] Stox-media: Should XEP-176 translations have Require: ice?
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Once again, my apologies for the slow reply.

On 1/9/14, 10:31 AM, Jonathan Lennox wrote:
> I was thinking about the issue that unlike RFC 5245, XEP-176's
> definition of ICE doesn't support fallback to a non-ICE mode.

Correct.

> It occurred to me that SIP has a way of saying "do ICE, or fail the
> call": putting a "Require: ice" SIP option tag (from RFC 5768) in the
> SIP INVITE.
>
> Should we recommend this?  It clearly has the right semantics, and
> will prevent interop failure when a non-ICE SIP endpoint answers a
> XEP-176 Jingle call.

Theoretically that makes sense.

> My concern, though, is whether there are a) endpoints that implement
> RFC 5245 but not RFC 5768, or b) non-media-terminating B2BUAs that
> will pass ICE parameters through, but will reject calls with Require
> headers they don't know.  In either of these cases, adding this
> Require header would cause a call that would otherwise have worked to
> fail.

That would be bad.

> Any thoughts, especially from folks who know the state of deployed
> SIP better than I do?

Inquiring minds want to know. :-)

Peter


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Subject: Re: [Stox] stox-media: format parameter translation
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On 1/9/14, 10:22 AM, Jonathan Lennox wrote:
> This is the issue I brought up at the Interim -- I'll try to put it
> in writing.
>
> The point I raised is that the stox-media draft's section on format
> parameter translation should try to focus on media types which are
> actually used by existing, deployed Jingle clients, since this is
> probably a relatively small set.  The important question is which
> payload formats in use by existing Jingle clients have SDP fmtp
> formats that don't follow the normal semicolon-separated parameter
> model.
>
> I of course haven't done a full inventory of such payload formats,
> but two that seem reasonably likely and I think should be
> particularly called out are:
>
> audio/telephone-event: RFC 4733: the fmtp contains the value of an
> implicit "events" parameter.

That one seems relevant.

> audio/red: RFC 2198, with its media type registration in RFC 3555:
> the fmtp is the payload types of the encompassed formats.  The RFC
> 3555 definition of the actual media type parameters is weird (in an
> attempt to make it self-contained), and I doubt that any Jingle
> client that does RED would actually use it that way. So we need to
> figure out what any actual Jingle implementations do.
>
> Does anyone have a reasonably-authoritative list of which RTP payload
> formats are supported by existing Jingle clients?

I do not, but it's a small enough universe that we can poll the 
developers. (I expect it's mostly limited to common audio and video 
payloads - Speex, Opus, etc. - but we'll find out.)

> Does anyone know
> of any other real-world usage of codecs that have unusual fmtp
> encodings?

I think audio/telephone-event is a possibility.

> Someone on the call mentioned that they thought that
> Speex's was weird, but as far as I can tell, it's a normal
> semicolon-separated encoding.

Agreed.

Peter


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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 SIP-TO-XMPP Working Group of the IETF.

        Title           : Interworking between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP): Media Sessions
        Authors         : Peter Saint-Andre
                          Saul Ibarra
                          Emil Ivov
	Filename        : draft-ietf-stox-media-04.txt
	Pages           : 25
	Date            : 2014-03-20

Abstract:
   This document defines a bidirectional protocol mapping for use by
   gateways that enable the exchange of media signalling messages
   between systems that implement the Session Initiation Protocol (SIP)
   and systems that implement the Jingle extensions to the Extensible
   Messaging and Presence Protocol (XMPP).


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-stox-media/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-stox-media-04

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-stox-media-04


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


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A bit of a general cleanup and editorial review...

On 3/20/14, 8:56 PM, 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 SIP-TO-XMPP Working Group of the IETF.
>
>          Title           : Interworking between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP): Media Sessions
>          Authors         : Peter Saint-Andre
>                            Saul Ibarra
>                            Emil Ivov
> 	Filename        : draft-ietf-stox-media-04.txt
> 	Pages           : 25
> 	Date            : 2014-03-20
>
> Abstract:
>     This document defines a bidirectional protocol mapping for use by
>     gateways that enable the exchange of media signalling messages
>     between systems that implement the Session Initiation Protocol (SIP)
>     and systems that implement the Jingle extensions to the Extensible
>     Messaging and Presence Protocol (XMPP).
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-stox-media/
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-stox-media-04
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-stox-media-04
>
>
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox
>


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I checked the diff and all is good for me, thanks Peter!

I have no further comments on this document.


On Mar 16, 2014, at 10:40 PM, internet-drafts@ietf.org wrote:

>=20
> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
> This draft is a work item of the SIP-TO-XMPP Working Group of the =
IETF.
>=20
>        Title           : Interworking between the Session Initiation =
Protocol (SIP) and the Extensible Messaging and Presence Protocol =
(XMPP): Groupchat
>        Authors         : Peter Saint-Andre
>                          Saul Ibarra
>                          Salvatore Loreto
> 	Filename        : draft-ietf-stox-groupchat-03.txt
> 	Pages           : 35
> 	Date            : 2014-03-16
>=20
> Abstract:
>   This document defines a bidirectional protocol mapping for the
>   exchange of instant messages in the context of a multiparty chat
>   session among users of the Session Initiation Protocol (SIP) and
>   users of the Extensible Messaging and Presence Protocol (XMPP).
>   Specifically, this document defines a mapping between the SIP-based
>   Message Session Relay Protocol (MSRP) and the XMPP Multi-User Chat
>   (MUC) extension.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-stox-groupchat/
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-stox-groupchat-03
>=20
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-stox-groupchat-03
>=20
>=20
> Please note that it may take a couple of minutes from the time of =
submission
> until the htmlized version and diff are available at tools.ietf.org.
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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Subject: Re: [Stox] STOX Virtual Interim Meeting 2013-12-17 Minutes
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Hi,

On Mar 20, 2014, at 4:39 AM, Peter Saint-Andre wrote:

> Paul, thanks for the feedback and my apologies for the extremely slow =
reply.
>=20
> On 1/9/14, 10:41 AM, Paul Kyzivat wrote:
>> As requested, I have taken a look at treatment of Hold in section 6 =
of
>> draft-ietf-stox-media-02.
>>=20
>> The draft says:
>>=20
>>    The same semantics are also
>>    supported by Jingle through the "senders" element and its =
"initiator"
>>    and "responder" values.
>>=20
>>                  [example to follow]
>>=20
>>    In addition to these semantics however Jingle also defines a more
>>    concise way for achieving the same, which consists in sending a
>>    "hold" command within a "session-info" action:
>>=20
>> Before I can make any useful comments I need to get educated on some
>> aspects of xmpp. (I *browsed* XEP 0166, but couldn't quickly figure =
this
>> out.) Hopefully somebody can clue me in on the following:
>>=20
>> 1) I see that the "senders" element has values that correspond 1:1 =
with
>> SDP sendrecv/sendonly/recvonly/inactive. Are there equivalent
>> constraints on O/A behavior? Specifically that:
>>=20
>>     Offer    Permissible Answer
>>     -----    ------------------
>>     sendrecv sendrecv/sendonly/recvonly/inactive
>>     sendonly recvonly/inactive
>>     recvonly sendonly/inactive
>>     inactive inactive
>=20
> Although those constraints look reasonable, I think the original =
Jingle authors were not aware of them when defining the protocol. In =
general, Jingle implementations are simpler than SIP implementations, =
and might tend to assume senders=3D'both' / a=3Dsendrecv. Also, not all =
uses of the 'senders' attribute quite follow the offer/answer model - =
see, for instance, the content-modify action:
>=20

Hum. Actually the XEP does mention compatibility with SDP O/A model:=20

"Which parties in the session will be generating content (i.e., the =
direction in which a Jingle session is active); the allowable values are =
"both", "initiator", "none", and "responder" (where the default is =
"both"). Note that the defined values of the 'senders' attribute in =
Jingle correspond to the SDP attributes of "sendrecv", "sendonly", =
"inactive", and "recvonly" defined in RFC 4566 [30] and used in the =
offer-answer model RFC 3264."

However that element is *optional* except for content-modify. IIRC we =
did discuss about recommended practices, since this didn't seem to be =
very used out there. FWIW, when I worked on the SIP-XMPP gateway =
functionality for SylkServer I relied solely on session-info with =
'hold'. This does of course limit the possible translation scenarios =
(ie, starting a session with a=3Dinactive) though.

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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Subject: Re: [Stox] Stox-media: Should XEP-176 translations have Require: ice?
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On Mar 20, 2014, at 4:45 AM, Peter Saint-Andre wrote:

> Once again, my apologies for the slow reply.
>=20
> On 1/9/14, 10:31 AM, Jonathan Lennox wrote:
>> I was thinking about the issue that unlike RFC 5245, XEP-176's
>> definition of ICE doesn't support fallback to a non-ICE mode.
>=20
> Correct.
>=20
>> It occurred to me that SIP has a way of saying "do ICE, or fail the
>> call": putting a "Require: ice" SIP option tag (from RFC 5768) in the
>> SIP INVITE.
>>=20
>> Should we recommend this?  It clearly has the right semantics, and
>> will prevent interop failure when a non-ICE SIP endpoint answers a
>> XEP-176 Jingle call.
>=20
> Theoretically that makes sense.

Agreed. Maybe a gateway could do ICE on behalf of the non-ICE capable =
endpoint, so can we say this is implementation specific and thus add a =
MAY to it?

>=20
>> My concern, though, is whether there are a) endpoints that implement
>> RFC 5245 but not RFC 5768, or b) non-media-terminating B2BUAs that
>> will pass ICE parameters through, but will reject calls with Require
>> headers they don't know.  In either of these cases, adding this
>> Require header would cause a call that would otherwise have worked to
>> fail.
>=20
> That would be bad.
>=20
>> Any thoughts, especially from folks who know the state of deployed
>> SIP better than I do?
>=20
> Inquiring minds want to know. :-)
>=20

FWIW, in Blink we do use the option tag in the Contact header but we =
don't add the Requires header, since we can work fine without ICE. Can't =
say what SBCs do though...

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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Subject: Re: [Stox] stox-media: format parameter translation
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On Mar 20, 2014, at 4:50 AM, Peter Saint-Andre wrote:

> On 1/9/14, 10:22 AM, Jonathan Lennox wrote:
>> This is the issue I brought up at the Interim -- I'll try to put it
>> in writing.
>>=20
>> The point I raised is that the stox-media draft's section on format
>> parameter translation should try to focus on media types which are
>> actually used by existing, deployed Jingle clients, since this is
>> probably a relatively small set.  The important question is which
>> payload formats in use by existing Jingle clients have SDP fmtp
>> formats that don't follow the normal semicolon-separated parameter
>> model.
>>=20
>> I of course haven't done a full inventory of such payload formats,
>> but two that seem reasonably likely and I think should be
>> particularly called out are:
>>=20
>> audio/telephone-event: RFC 4733: the fmtp contains the value of an
>> implicit "events" parameter.
>=20
> That one seems relevant.
>=20
>> audio/red: RFC 2198, with its media type registration in RFC 3555:
>> the fmtp is the payload types of the encompassed formats.  The RFC
>> 3555 definition of the actual media type parameters is weird (in an
>> attempt to make it self-contained), and I doubt that any Jingle
>> client that does RED would actually use it that way. So we need to
>> figure out what any actual Jingle implementations do.
>>=20
>> Does anyone have a reasonably-authoritative list of which RTP payload
>> formats are supported by existing Jingle clients?
>=20
> I do not, but it's a small enough universe that we can poll the =
developers. (I expect it's mostly limited to common audio and video =
payloads - Speex, Opus, etc. - but we'll find out.)
>=20
>> Does anyone know
>> of any other real-world usage of codecs that have unusual fmtp
>> encodings?
>=20
> I think audio/telephone-event is a possibility.
>=20
>> Someone on the call mentioned that they thought that
>> Speex's was weird, but as far as I can tell, it's a normal
>> semicolon-separated encoding.
>=20
> Agreed.
>=20

Same goes for H264 format AFAIS. It has many parameters, but in the end =
it's just a list of attribute=3Dvalue separated by semicolons.

--
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AG Projects




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On 3/25/14, 2:44 AM, Saúl Ibarra Corretgé wrote:
> I checked the diff and all is good for me, thanks Peter!

Thanks for checking. The one thing I still need to do is make the 
protocol flow diagrams consistent with the other documents in the 
series. I'll try to do that today or tonight.

Peter



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From: Jonathan Lennox <jonathan@vidyo.com>
To: =?Windows-1252?Q?Sa=FAl_Ibarra_Corretg=E9?= <saul@ag-projects.com>
Thread-Topic: [Stox] Stox-media: Should XEP-176 translations have Require: ice?
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Subject: Re: [Stox] Stox-media: Should XEP-176 translations have Require: ice?
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On Mar 25, 2014, at 4:55 AM, Sa=FAl Ibarra Corretg=E9 <saul@ag-projects.com=
> wrote:
>>=20
>>=20
>>> It occurred to me that SIP has a way of saying "do ICE, or fail the
>>> call": putting a "Require: ice" SIP option tag (from RFC 5768) in the
>>> SIP INVITE.
>>>=20
>>> Should we recommend this?  It clearly has the right semantics, and
>>> will prevent interop failure when a non-ICE SIP endpoint answers a
>>> XEP-176 Jingle call.
>>=20
>> Theoretically that makes sense.
>=20
> Agreed. Maybe a gateway could do ICE on behalf of the non-ICE capable end=
point, so can we say this is implementation specific and thus add a MAY to =
it?

Right, of course.  I was talking about how you=92d want to handle a signali=
ng-only gateway.

>>> My concern, though, is whether there are a) endpoints that implement
>>> RFC 5245 but not RFC 5768, or b) non-media-terminating B2BUAs that
>>> will pass ICE parameters through, but will reject calls with Require
>>> headers they don't know.  In either of these cases, adding this
>>> Require header would cause a call that would otherwise have worked to
>>> fail.
>>=20
>> That would be bad.
>>=20
>>> Any thoughts, especially from folks who know the state of deployed
>>> SIP better than I do?
>>=20
>> Inquiring minds want to know. :-)
>>=20
>=20
> FWIW, in Blink we do use the option tag in the Contact header but we don'=
t add the Requires header, since we can work fine without ICE. Can't say wh=
at SBCs do though=85

When you say =93can work fine without ICE=94, do you mean you support full =
RFC 5245 fallback to the use of the m/c line for media?  That does indeed s=
ound like the case when you should have the Contact tag but not the require=
s header.


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Date: Tue, 25 Mar 2014 18:37:21 +0100
From: Emil Ivov <emcho@jitsi.org>
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Subject: Re: [Stox] Stox-media: Should XEP-176 translations have Require: ice?
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On 25.03.14, 15:15, Jonathan Lennox wrote:
>
> On Mar 25, 2014, at 4:55 AM, Saúl Ibarra Corretgé <saul@ag-projects.com> wrote:
>>>
>>>
>>>> It occurred to me that SIP has a way of saying "do ICE, or fail the
>>>> call": putting a "Require: ice" SIP option tag (from RFC 5768) in the
>>>> SIP INVITE.
>>>>
>>>> Should we recommend this?  It clearly has the right semantics, and
>>>> will prevent interop failure when a non-ICE SIP endpoint answers a
>>>> XEP-176 Jingle call.
>>>
>>> Theoretically that makes sense.
>>
>> Agreed. Maybe a gateway could do ICE on behalf of the non-ICE capable endpoint, so can we say this is implementation specific and thus add a MAY to it?
>
> Right, of course.  I was talking about how you’d want to handle a signaling-only gateway.

There's also "XEP-0177: Jingle Raw UDP Transport Method" which is 
basically equivalent to ICE-less SIP.

That said, I don't mind mandating ICE at all.

Emil

-- 
https://jitsi.org


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From: Jonathan Lennox <jonathan@vidyo.com>
To: Emil Ivov <emcho@jitsi.org>
Thread-Topic: [Stox] Stox-media: Should XEP-176 translations have Require: ice?
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Subject: Re: [Stox] Stox-media: Should XEP-176 translations have Require: ice?
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On Mar 25, 2014, at 1:37 PM, Emil Ivov <emcho@jitsi.org> wrote:

> On 25.03.14, 15:15, Jonathan Lennox wrote:
>>=20
>> On Mar 25, 2014, at 4:55 AM, Sa=FAl Ibarra Corretg=E9 <saul@ag-projects.=
com> wrote:
>>>>=20
>>>>=20
>>>>> It occurred to me that SIP has a way of saying "do ICE, or fail the
>>>>> call": putting a "Require: ice" SIP option tag (from RFC 5768) in the
>>>>> SIP INVITE.
>>>>>=20
>>>>> Should we recommend this?  It clearly has the right semantics, and
>>>>> will prevent interop failure when a non-ICE SIP endpoint answers a
>>>>> XEP-176 Jingle call.
>>>>=20
>>>> Theoretically that makes sense.
>>>=20
>>> Agreed. Maybe a gateway could do ICE on behalf of the non-ICE capable e=
ndpoint, so can we say this is implementation specific and thus add a MAY t=
o it?
>>=20
>> Right, of course.  I was talking about how you=92d want to handle a sign=
aling-only gateway.
>=20
> There's also "XEP-0177: Jingle Raw UDP Transport Method" which is basical=
ly equivalent to ICE-less SIP.
>=20
> That said, I don't mind mandating ICE at all.

Right, but there=92s no way for a Jingle client to offer a choice between X=
EP-176 and XEP-177, whereas RFC 5245 requires that both ICE and fallback no=
n-ICE be supported.

RFC 5768 allows a signaling-only XMPP-to-SIP gateway to (mostly) express th=
e XMPP semantic that ICE is required, with no fallback supported.  However,=
 using it would mean failures if you hit a device that supports (or passes)=
 RFC 5245 but not RFC 5768.=


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Subject: Re: [Stox] Stox-media: Should XEP-176 translations have Require: ice?
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On 25.03.14, 19:03, Jonathan Lennox wrote:
>
> On Mar 25, 2014, at 1:37 PM, Emil Ivov <emcho@jitsi.org> wrote:
>
>> On 25.03.14, 15:15, Jonathan Lennox wrote:
>>>
>>> On Mar 25, 2014, at 4:55 AM, Saúl Ibarra Corretgé <saul@ag-projects.com> wrote:
>>>>>
>>>>>
>>>>>> It occurred to me that SIP has a way of saying "do ICE, or fail the
>>>>>> call": putting a "Require: ice" SIP option tag (from RFC 5768) in the
>>>>>> SIP INVITE.
>>>>>>
>>>>>> Should we recommend this?  It clearly has the right semantics, and
>>>>>> will prevent interop failure when a non-ICE SIP endpoint answers a
>>>>>> XEP-176 Jingle call.
>>>>>
>>>>> Theoretically that makes sense.
>>>>
>>>> Agreed. Maybe a gateway could do ICE on behalf of the non-ICE capable endpoint, so can we say this is implementation specific and thus add a MAY to it?
>>>
>>> Right, of course.  I was talking about how you’d want to handle a signaling-only gateway.
>>
>> There's also "XEP-0177: Jingle Raw UDP Transport Method" which is basically equivalent to ICE-less SIP.
>>
>> That said, I don't mind mandating ICE at all.
>
> Right, but there’s no way for a Jingle client to offer a choice between XEP-176 and XEP-177, whereas RFC 5245 requires that both ICE and fallback non-ICE be supported.
>
> RFC 5768 allows a signaling-only XMPP-to-SIP gateway to (mostly) express the XMPP semantic that ICE is required, with no fallback supported.  However, using it would mean failures if you hit a device that supports (or passes) RFC 5245 but not RFC 5768.

Agreed!

-- 
https://jitsi.org


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Subject: Re: [Stox] Stox-media: Should XEP-176 translations have Require: ice?
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On 3/25/14, 12:03 PM, Jonathan Lennox wrote:
>
> On Mar 25, 2014, at 1:37 PM, Emil Ivov <emcho@jitsi.org> wrote:
>
>> On 25.03.14, 15:15, Jonathan Lennox wrote:
>>>
>>> On Mar 25, 2014, at 4:55 AM, Saúl Ibarra Corretgé
>>> <saul@ag-projects.com> wrote:
>>>>>
>>>>>
>>>>>> It occurred to me that SIP has a way of saying "do ICE, or
>>>>>> fail the call": putting a "Require: ice" SIP option tag
>>>>>> (from RFC 5768) in the SIP INVITE.
>>>>>>
>>>>>> Should we recommend this?  It clearly has the right
>>>>>> semantics, and will prevent interop failure when a non-ICE
>>>>>> SIP endpoint answers a XEP-176 Jingle call.
>>>>>
>>>>> Theoretically that makes sense.
>>>>
>>>> Agreed. Maybe a gateway could do ICE on behalf of the non-ICE
>>>> capable endpoint, so can we say this is implementation specific
>>>> and thus add a MAY to it?
>>>
>>> Right, of course.  I was talking about how you’d want to handle a
>>> signaling-only gateway.
>>
>> There's also "XEP-0177: Jingle Raw UDP Transport Method" which is
>> basically equivalent to ICE-less SIP.
>>
>> That said, I don't mind mandating ICE at all.
>
> Right, but there’s no way for a Jingle client to offer a choice
> between XEP-176 and XEP-177, whereas RFC 5245 requires that both ICE
> and fallback non-ICE be supported.

Well, an XMPP initiator would not offer the ice-udp transport method if 
the responder didn't advertise support via XMPP service discovery (i.e., 
only advertised support for raw-udp). So, in general, fallback shouldn't 
really be necessary. However, we have defined a protocol flow for it 
just in case:

http://xmpp.org/extensions/xep-0176.html#fallback

Peter




From nobody Tue Mar 25 20:03:54 2014
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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 SIP-TO-XMPP Working Group of the IETF.

        Title           : Interworking between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP): Groupchat
        Authors         : Peter Saint-Andre
                          Saul Ibarra
                          Salvatore Loreto
	Filename        : draft-ietf-stox-groupchat-04.txt
	Pages           : 38
	Date            : 2014-03-25

Abstract:
   This document defines a bidirectional protocol mapping for the
   exchange of instant messages in the context of a multiparty chat
   session among users of the Session Initiation Protocol (SIP) and
   users of the Extensible Messaging and Presence Protocol (XMPP).
   Specifically, this document defines a mapping between the SIP-based
   Message Session Relay Protocol (MSRP) and the XMPP Multi-User Chat
   (MUC) extension.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-stox-groupchat/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-stox-groupchat-04

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-stox-groupchat-04


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


From nobody Tue Mar 25 20:05:00 2014
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As promised, I've made the protocol flow diagrams consistent with the 
other STOX documents.

Peter

On 3/25/14, 9:03 PM, 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 SIP-TO-XMPP Working Group of the IETF.
>
>          Title           : Interworking between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP): Groupchat
>          Authors         : Peter Saint-Andre
>                            Saul Ibarra
>                            Salvatore Loreto
> 	Filename        : draft-ietf-stox-groupchat-04.txt
> 	Pages           : 38
> 	Date            : 2014-03-25
>
> Abstract:
>     This document defines a bidirectional protocol mapping for the
>     exchange of instant messages in the context of a multiparty chat
>     session among users of the Session Initiation Protocol (SIP) and
>     users of the Extensible Messaging and Presence Protocol (XMPP).
>     Specifically, this document defines a mapping between the SIP-based
>     Message Session Relay Protocol (MSRP) and the XMPP Multi-User Chat
>     (MUC) extension.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-stox-groupchat/
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-stox-groupchat-04
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-stox-groupchat-04
>
>
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox
>


From nobody Wed Mar 26 04:09:32 2014
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Subject: Re: [Stox] Stox-media: Should XEP-176 translations have Require: ice?
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On Mar 25, 2014, at 3:15 PM, Jonathan Lennox wrote:

>=20
> On Mar 25, 2014, at 4:55 AM, Sa=FAl Ibarra Corretg=E9 =
<saul@ag-projects.com> wrote:
>>>=20
>>>=20
>>>> It occurred to me that SIP has a way of saying "do ICE, or fail the
>>>> call": putting a "Require: ice" SIP option tag (from RFC 5768) in =
the
>>>> SIP INVITE.
>>>>=20
>>>> Should we recommend this?  It clearly has the right semantics, and
>>>> will prevent interop failure when a non-ICE SIP endpoint answers a
>>>> XEP-176 Jingle call.
>>>=20
>>> Theoretically that makes sense.
>>=20
>> Agreed. Maybe a gateway could do ICE on behalf of the non-ICE capable =
endpoint, so can we say this is implementation specific and thus add a =
MAY to it?
>=20
> Right, of course.  I was talking about how you=92d want to handle a =
signaling-only gateway.

Ah, by bad, sorry. As Peter pointed out there is a fallback mechanism, =
so if the Jingle endpoint advertised support for raw UDP it should be =
possible to fallback. In case it didn't there is nothing that can be =
done.

>=20
>>>> My concern, though, is whether there are a) endpoints that =
implement
>>>> RFC 5245 but not RFC 5768, or b) non-media-terminating B2BUAs that
>>>> will pass ICE parameters through, but will reject calls with =
Require
>>>> headers they don't know.  In either of these cases, adding this
>>>> Require header would cause a call that would otherwise have worked =
to
>>>> fail.
>>>=20
>>> That would be bad.
>>>=20
>>>> Any thoughts, especially from folks who know the state of deployed
>>>> SIP better than I do?
>>>=20
>>> Inquiring minds want to know. :-)
>>>=20
>>=20
>> FWIW, in Blink we do use the option tag in the Contact header but we =
don't add the Requires header, since we can work fine without ICE. Can't =
say what SBCs do though=85
>=20
> When you say =93can work fine without ICE=94, do you mean you support =
full RFC 5245 fallback to the use of the m/c line for media?  That does =
indeed sound like the case when you should have the Contact tag but not =
the requires header.
>=20

Yep.

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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Thread-Topic: [Stox] Stox-media: Should XEP-176 translations have Require: ice?
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Date: Wed, 26 Mar 2014 15:50:16 +0000
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Subject: Re: [Stox] Stox-media: Should XEP-176 translations have Require: ice?
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On Mar 25, 2014, at 6:12 PM, Peter Saint-Andre <stpeter@stpeter.im<mailto:s=
tpeter@stpeter.im>> wrote:

On 3/25/14, 12:03 PM, Jonathan Lennox wrote:

On Mar 25, 2014, at 1:37 PM, Emil Ivov <emcho@jitsi.org<mailto:emcho@jitsi.=
org>> wrote:

On 25.03.14, 15:15, Jonathan Lennox wrote:

On Mar 25, 2014, at 4:55 AM, Sa=FAl Ibarra Corretg=E9
<saul@ag-projects.com<mailto:saul@ag-projects.com>> wrote:


It occurred to me that SIP has a way of saying "do ICE, or
fail the call": putting a "Require: ice" SIP option tag
(from RFC 5768) in the SIP INVITE.

Should we recommend this?  It clearly has the right
semantics, and will prevent interop failure when a non-ICE
SIP endpoint answers a XEP-176 Jingle call.

Theoretically that makes sense.

Agreed. Maybe a gateway could do ICE on behalf of the non-ICE
capable endpoint, so can we say this is implementation specific
and thus add a MAY to it?

Right, of course.  I was talking about how you=92d want to handle a
signaling-only gateway.

There's also "XEP-0177: Jingle Raw UDP Transport Method" which is
basically equivalent to ICE-less SIP.

That said, I don't mind mandating ICE at all.

Right, but there=92s no way for a Jingle client to offer a choice
between XEP-176 and XEP-177, whereas RFC 5245 requires that both ICE
and fallback non-ICE be supported.

Well, an XMPP initiator would not offer the ice-udp transport method if the=
 responder didn't advertise support via XMPP service discovery (i.e., only =
advertised support for raw-udp). So, in general, fallback shouldn't really =
be necessary. However, we have defined a protocol flow for it just in case:

http://xmpp.org/extensions/xep-0176.html#fallback

The problem with that flow is the statement "Based on capabilities informat=
ion, the gateway knows that the SIP endpoint does not support ICE=94.  Unfo=
rtunately, SIP doesn=92t really have any workable way of providing capabili=
ties information, which is why RFC 5245 (and so much of the rest of SIP) is=
 single-exchange fallback.

An alternative flow is possible, where you only send the transport-replace =
on the Jingle side if the SIP answer indicates no ICE support, and then sen=
d a SIP re-INVITE with the addresses in the transport-accept.  It=92ll resu=
lt a fairly substantial post-answer delay before the media will get through=
, though.

Here=92s a sketch:


Romeo                    Gateway                    Juliet
  |                         |                         |
  |   session-initiate      |                         |
  |   (audio definition)    |                         |
  |------------------------>|                         |
  |   ack                   |                         |
  |<------------------------|   SIP INVITE (ICE)      |
  |                         |------------------------>|
  |   transport-replace     |   200 OK (Raw UDP)      |
  |   (Raw UDP)             |<------------------------|
  |<------------------------|                         |
  |   ack                   |                         |
  |------------------------>|                         |
  |   transport-accept      |                         |
  |------------------------>|                         |
  |   ack                   |                         |
  |<------------------------|   SIP INVITE (Raw UDP)  |
  |                         |------------------------>|
  |                         |   200 OK (Raw UDP)      |
  |   session-accept        |<------------------------|
  |<------------------------|                         |
  |   ack                   |                         |
  |------------------------>|                         |
  |                    MEDIA SESSION                  |
  |<=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D>|
  |                         |                         |



This flow also relies on the SIP side not changing its local IP address and=
 port in response to a re-INVITE with a changed address, or the XMPP side n=
ot changing its in response to a transport-replace maintaining raw UDP; oth=
erwise, the gateway can get into an infinite re-INVITE/transport-replace lo=
op.

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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DWindows-1=
252">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space;" class=3D"">
<br class=3D"">
<div>
<div class=3D"">On Mar 25, 2014, at 6:12 PM, Peter Saint-Andre &lt;<a href=
=3D"mailto:stpeter@stpeter.im">stpeter@stpeter.im</a>&gt; wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">On 3/25/14, 12:03 PM, Jonathan Lennox wrote:<br c=
lass=3D"">
<blockquote type=3D"cite"><br class=3D"">
On Mar 25, 2014, at 1:37 PM, Emil Ivov &lt;<a href=3D"mailto:emcho@jitsi.or=
g">emcho@jitsi.org</a>&gt; wrote:<br class=3D"">
<br class=3D"">
<blockquote type=3D"cite">On 25.03.14, 15:15, Jonathan Lennox wrote:<br cla=
ss=3D"">
<blockquote type=3D"cite"><br class=3D"">
On Mar 25, 2014, at 4:55 AM, Sa=FAl Ibarra Corretg=E9<br class=3D"">
&lt;<a href=3D"mailto:saul@ag-projects.com">saul@ag-projects.com</a>&gt; wr=
ote:<br class=3D"">
<blockquote type=3D"cite">
<blockquote type=3D"cite"><br class=3D"">
<br class=3D"">
<blockquote type=3D"cite">It occurred to me that SIP has a way of saying &q=
uot;do ICE, or<br class=3D"">
fail the call&quot;: putting a &quot;Require: ice&quot; SIP option tag<br c=
lass=3D"">
(from RFC 5768) in the SIP INVITE.<br class=3D"">
<br class=3D"">
Should we recommend this? &nbsp;It clearly has the right<br class=3D"">
semantics, and will prevent interop failure when a non-ICE<br class=3D"">
SIP endpoint answers a XEP-176 Jingle call.<br class=3D"">
</blockquote>
<br class=3D"">
Theoretically that makes sense.<br class=3D"">
</blockquote>
<br class=3D"">
Agreed. Maybe a gateway could do ICE on behalf of the non-ICE<br class=3D""=
>
capable endpoint, so can we say this is implementation specific<br class=3D=
"">
and thus add a MAY to it?<br class=3D"">
</blockquote>
<br class=3D"">
Right, of course. &nbsp;I was talking about how you=92d want to handle a<br=
 class=3D"">
signaling-only gateway.<br class=3D"">
</blockquote>
<br class=3D"">
There's also &quot;XEP-0177: Jingle Raw UDP Transport Method&quot; which is=
<br class=3D"">
basically equivalent to ICE-less SIP.<br class=3D"">
<br class=3D"">
That said, I don't mind mandating ICE at all.<br class=3D"">
</blockquote>
<br class=3D"">
Right, but there=92s no way for a Jingle client to offer a choice<br class=
=3D"">
between XEP-176 and XEP-177, whereas RFC 5245 requires that both ICE<br cla=
ss=3D"">
and fallback non-ICE be supported.<br class=3D"">
</blockquote>
<br class=3D"">
Well, an XMPP initiator would not offer the ice-udp transport method if the=
 responder didn't advertise support via XMPP service discovery (i.e., only =
advertised support for raw-udp). So, in general, fallback shouldn't really =
be necessary. However, we have defined
 a protocol flow for it just in case:<br class=3D"">
<br class=3D"">
<a href=3D"http://xmpp.org/extensions/xep-0176.html#fallback">http://xmpp.o=
rg/extensions/xep-0176.html#fallback</a></blockquote>
<br class=3D"">
</div>
<div>The problem with that flow is the statement &quot;Based on capabilitie=
s information, the gateway knows that the SIP endpoint does not support ICE=
=94. &nbsp;Unfortunately, SIP doesn=92t really have any workable way of pro=
viding capabilities information, which is why
 RFC 5245 (and so much of the rest of SIP) is single-exchange fallback.</di=
v>
<div><br class=3D"">
</div>
<div>An alternative flow is possible, where you only send the transport-rep=
lace on the Jingle side if the SIP answer indicates no ICE support, and the=
n send a SIP re-INVITE with the addresses in the transport-accept. &nbsp;It=
=92ll result a fairly substantial post-answer
 delay before the media will get through, though.</div>
<div><br class=3D"">
</div>
<div>Here=92s a sketch:</div>
<div><br class=3D"">
</div>
<div>
<div class=3D"indent">
<pre class=3D"prettyprint"><div style=3D"margin: 0px; font-size: 11px; font=
-family: Menlo;" class=3D"">Romeo &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=
 &nbsp; &nbsp; &nbsp; &nbsp;Gateway &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp;Juliet</div><div style=3D"margin: 0px; font-s=
ize: 11px; font-family: Menlo;" class=3D"">&nbsp; | &nbsp; &nbsp; &nbsp; &n=
bsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | &nbsp; &nbsp=
; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</=
div><div style=3D"margin: 0px; font-size: 11px; font-family: Menlo;" class=
=3D"">&nbsp; | &nbsp; session-initiate &nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</di=
v><div style=3D"margin: 0px; font-size: 11px; font-family: Menlo;" class=3D=
"">&nbsp; | &nbsp; (audio definition) &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</div><div s=
tyle=3D"margin: 0px; font-size: 11px; font-family: Menlo;" class=3D"">&nbsp=
; |------------------------&gt;| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</div><div style=3D"margin: 0px;=
 font-size: 11px; font-family: Menlo;" class=3D"">&nbsp; | &nbsp; ack &nbsp=
; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</div=
><div style=3D"margin: 0px; font-size: 11px; font-family: Menlo;" class=3D"=
">&nbsp; |&lt;------------------------| &nbsp; SIP INVITE (ICE)&nbsp; &nbsp=
; &nbsp; |</div><div style=3D"margin: 0px; font-size: 11px; font-family: Me=
nlo;" class=3D"">&nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |------------------------&gt;|</div><div=
 style=3D"margin: 0px; font-size: 11px; font-family: Menlo;" class=3D"">&nb=
sp; | &nbsp; transport-replace &nbsp; &nbsp; | &nbsp; 200 OK (Raw UDP)&nbsp=
; &nbsp; &nbsp; |</div><div style=3D"margin: 0px; font-size: 11px; font-fam=
ily: Menlo;" class=3D"">&nbsp; | &nbsp; (Raw UDP) &nbsp; &nbsp; &nbsp; &nbs=
p; &nbsp; &nbsp; |&lt;------------------------|</div><div style=3D"margin: =
0px; font-size: 11px; font-family: Menlo;" class=3D"">&nbsp; |&lt;---------=
---------------| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &n=
bsp; &nbsp; &nbsp; &nbsp; |</div><div style=3D"margin: 0px; font-size: 11px=
; font-family: Menlo;" class=3D"">&nbsp; | &nbsp; ack &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nb=
sp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</div><div style=3D"m=
argin: 0px; font-size: 11px; font-family: Menlo;" class=3D"">&nbsp; |------=
------------------&gt;| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &n=
bsp; &nbsp; &nbsp; &nbsp; &nbsp; |</div><div style=3D"margin: 0px; font-siz=
e: 11px; font-family: Menlo;" class=3D"">&nbsp; | &nbsp; transport-accept &=
nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
; &nbsp; &nbsp; &nbsp; &nbsp; |</div><div style=3D"margin: 0px; font-size: =
11px; font-family: Menlo;" class=3D"">&nbsp; |------------------------&gt;|=
 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbs=
p; &nbsp; |</div><div style=3D"margin: 0px; font-size: 11px; font-family: M=
enlo;" class=3D"">&nbsp; | &nbsp; ack &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &n=
bsp; &nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</div><div style=3D"margin: 0px; font=
-size: 11px; font-family: Menlo;" class=3D"">&nbsp; |&lt;------------------=
------| &nbsp; SIP INVITE (Raw UDP)&nbsp; |</div><div style=3D"margin: 0px;=
 font-size: 11px; font-family: Menlo;" class=3D"">&nbsp; | &nbsp; &nbsp; &n=
bsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |------=
------------------&gt;|</div><div style=3D"margin: 0px; font-size: 11px; fo=
nt-family: Menlo;" class=3D"">&nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | &nbsp; 200 OK (Raw UDP)&n=
bsp; &nbsp; &nbsp; |</div><div style=3D"margin: 0px; font-size: 11px; font-=
family: Menlo;" class=3D"">&nbsp; | &nbsp; session-accept &nbsp; &nbsp; &nb=
sp; &nbsp;|&lt;------------------------|</div><div style=3D"margin: 0px; fo=
nt-size: 11px; font-family: Menlo;" class=3D"">&nbsp; |&lt;----------------=
--------| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &n=
bsp; &nbsp; &nbsp; |</div><div style=3D"margin: 0px; font-size: 11px; font-=
family: Menlo;" class=3D"">&nbsp; | &nbsp; ack &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nb=
sp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</div><div style=3D"margin: =
0px; font-size: 11px; font-family: Menlo;" class=3D"">&nbsp; |-------------=
-----------&gt;| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &n=
bsp; &nbsp; &nbsp; &nbsp; |</div><div style=3D"margin: 0px; font-size: 11px=
; font-family: Menlo;" class=3D"">&nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;MEDIA SESSION &nbsp; &nbsp; &nbsp; &nb=
sp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|</div><div style=3D"margin: 0px; fon=
t-size: 11px; font-family: Menlo;" class=3D"">&nbsp; |&lt;=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D&gt;|</div><div st=
yle=3D"margin: 0px; font-size: 11px; font-family: Menlo;" class=3D"">&nbsp;=
 | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &n=
bsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
; &nbsp; &nbsp; &nbsp; |</div>
</pre>
</div>
<div class=3D"">This flow also relies on the SIP side not changing its loca=
l IP address and port in response to a re-INVITE with a changed address, or=
 the XMPP side not changing its in response to a transport-replace maintain=
ing raw UDP; otherwise, the gateway
 can get into an infinite re-INVITE/transport-replace loop.</div>
</div>
</body>
</html>

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