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From: Yana Stamcheva <yana@jitsi.org>
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Subject: Re: [Stox] Consensus call: Adopting the "current" drafts as WG documents to match the planned deliverables
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Hi,

On June 19 the chairs have started a consensus call on adopting the =
following drafts as WG documents:

draft-saintandre-sip-xmpp-core (Peter Saint-Andre, Avshalom Houri, Joe =
Hildebrand)
draft-saintandre-sip-xmpp-presence (Peter Saint-Andre, Avshalom Houri, =
Joe Hildebrand)
draft-saintandre-sip-xmpp-im (Peter Saint-Andre, Avshalom Houri, Joe =
Hildebrand)
draft-saintandre-sip-xmpp-chat (Peter Saint-Andre, Salvatore Loreto, =
Eddy Gavita, Nazin Hossain)
draft-saintandre-sip-xmpp-groupchat (Peter Saint-Andre, Salvatore =
Loreto, Saul Ibarra Corretge, Fabio Forno)
draft-saintandre-sip-xmpp-media (Peter Saint-Andre) - as I understand =
Emil and Saul will also be co-authors on this one

Since no objections were made and no alternative proposals were =
submitted we hereby conclude that the WG accepts the adoption of the =
drafts.

We would like to ask the co-authors of the drafts to proceed with =
submitting the -00 WG documents, so that they can be submitted as such =
by the IETF 87 -00 cut-off date (July 8).

Regards,
Yana & Markus

On Jun 19, 2013, at 12:30 PM, Markus.Isomaki@nokia.com wrote:

> Hi,
>=20
> The chartered deliverables of STOX are:
>=20
> 1. Address mapping and error handling
> 2. Presence mapping
> 3. Mapping for single instant messages
> 4. Mapping for one-to-one text chat sessions
> 5. Mapping for multi-user text chat sessions
> 6. Mapping for media signaling
>=20
> The plan is that a separate RFC will be published for each of these.=20=

>=20
> As the SIP-to-XMPP mapping effort has been going on for some years =
already even before the WG for it has been setup, we already have a set =
of relatively mature (individual) drafts to match these deliverables:
>=20
> draft-saintandre-sip-xmpp-core (Peter Saint-Andre, Avshalom Houri, Joe =
Hildebrand)
> draft-saintandre-sip-xmpp-presence (Peter Saint-Andre, Avshalom Houri, =
Joe Hildebrand)
> draft-saintandre-sip-xmpp-im (Peter Saint-Andre, Avshalom Houri, Joe =
Hildebrand)
> draft-saintandre-sip-xmpp-chat (Peter Saint-Andre, Salvatore Loreto, =
Eddy Gavita, Nazin Hossain)
> draft-saintandre-sip-xmpp-groupchat (Peter Saint-Andre, Salvatore =
Loreto, Saul Ibarra Corretge, Fabio Forno)
> draft-saintandre-sip-xmpp-media (Peter Saint-Andre)
>=20
> The first milestone of our charter is:
>=20
> Jun 2013  Accept starting-point mapping specifications as WG items
>=20
> To meet this on time, we are hereby starting a consensus call on =
adopting these drafts as WG items, so that they can be submitted as such =
by the IETF 87 -00 cut-off date (July 8). So, please send your comments =
on this to the STOX list by the end of next week (June 30), after which =
the chairs determine if there is a consensus for the WG adoption of =
these documents. Please send also simple "yes" or "+1" statements to the =
list, since we want to also see how many active participants we =
currently have. If you have issues about why the current drafts should =
not be adopted, please raise them ASAP.=20
>=20
> The other question is about the editors/co-authors of the WG =
documents. By default we will continue with the same list of co-authors =
as in the individual documents. So, we would like all the current =
co-authors to send a note to the chairs whether they are willing to =
continue (and let us know about your realistic time commitment too). One =
change has already been agreed: sip-xmpp-media draft will be taken over =
by Saul Ibarra Corretge and Emil Ivov. In case someone else is willing =
to volunteers as a co-chair for one or more of the drafts, let the =
chairs know.
>=20
> The 'sip-xmpp-media' draft (or the to-be RFC) will still also need a =
decision about its scope. We'll send a separate mail on that.
>=20
> Regards,
> 	Markus & Yana
>=20
>=20
>=20
>=20
>=20
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox


From stpeter@stpeter.im  Mon Jul  1 10:29:30 2013
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On 7/1/13 7:17 AM, Yana Stamcheva wrote:

> We would like to ask the co-authors of the drafts to proceed with
> submitting the -00 WG documents, so that they can be submitted as
> such by the IETF 87 -00 cut-off date (July 8).

Done!

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



From stpeter@stpeter.im  Mon Jul  1 10:36:29 2013
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On 7/1/13 11:29 AM, Peter Saint-Andre wrote:
> On 7/1/13 7:17 AM, Yana Stamcheva wrote:
> 
>> We would like to ask the co-authors of the drafts to proceed with
>> submitting the -00 WG documents, so that they can be submitted as
>> such by the IETF 87 -00 cut-off date (July 8).
> 
> Done!

Also checked into source control:

http://svn.tools.ietf.org/svn/wg/stox/

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



From internet-drafts@ietf.org  Mon Jul  1 14:49:13 2013
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the SIP-TO-XMPP Working Group of the IETF.

	Title           : Interworking between the Session Initiation Protocol (SI=
P) and the Extensible Messaging and Presence Protocol (XMPP): Addresses and=
 Error Conditions
	Author(s)       : Peter Saint-Andre
                          Avshalom Houri
                          Joe Hildebrand
	Filename        : draft-ietf-stox-core-00.txt
	Pages           : 14
	Date            : 2013-07-01

Abstract:
   As a foundation for the definition of bidirectional protocol mappings
   between the Session Initiation Protocol (SIP) and the Extensible
   Messaging and Presence Protocol (XMPP), this document specifies the
   architectural assumptions underlying such mappings as well as the
   mapping of addresses and error conditions.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-stox-core-00


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


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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the SIP-TO-XMPP Working Group of the IETF.

	Title           : Interworking between the Session Initiation Protocol (SI=
P) and the Extensible Messaging and Presence Protocol (XMPP): Instant Messa=
ging
	Author(s)       : Peter Saint-Andre
                          Avshalom Houri
                          Joe Hildebrand
	Filename        : draft-ietf-stox-im-00.txt
	Pages           : 11
	Date            : 2013-07-01

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-00


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ftp://ftp.ietf.org/internet-drafts/


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Subject: [Stox] I-D Action: draft-ietf-stox-chat-00.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the SIP-TO-XMPP Working Group of the IETF.

	Title           : Interworking between the Session Initiation Protocol (SI=
P) and the Extensible Messaging and Presence Protocol (XMPP): One-to-One Te=
xt Chat
	Author(s)       : Peter Saint-Andre
                          Salvatore Loreto
                          Eddy Gavita
                          Nazin Hossain
	Filename        : draft-ietf-stox-chat-00.txt
	Pages           : 13
	Date            : 2013-07-01

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-00


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


From internet-drafts@ietf.org  Mon Jul  1 14:50:59 2013
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Subject: [Stox] I-D Action: draft-ietf-stox-groupchat-00.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the SIP-TO-XMPP Working Group of the IETF.

	Title           : Interworking between the Session Initiation Protocol (SI=
P) and the Extensible Messaging and Presence Protocol (XMPP): Groupchat
	Author(s)       : Peter Saint-Andre
                          Saul Ibarra
                          Salvatore Loreto
	Filename        : draft-ietf-stox-groupchat-00.txt
	Pages           : 32
	Date            : 2013-07-01

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-00


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


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Subject: [Stox] I-D Action: draft-ietf-stox-media-00.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the SIP-TO-XMPP Working Group of the IETF.

	Title           : Interworking between the Session Initiation Protocol (SI=
P) and the Extensible Messaging and Presence Protocol (XMPP): Media Sessions
	Author(s)       : Peter Saint-Andre
                          Saul Ibarra
                          Emil Ivov
	Filename        : draft-ietf-stox-media-00.txt
	Pages           : 17
	Date            : 2013-07-01

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


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-00


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


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Subject: Re: [Stox] Consensus call: Adopting the "current" drafts as WG documents to match the planned deliverables
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Hi Peter,

On Jul 1, 2013, at 7:36 PM, Peter Saint-Andre <stpeter@stpeter.im> =
wrote:

> On 7/1/13 11:29 AM, Peter Saint-Andre wrote:
>> On 7/1/13 7:17 AM, Yana Stamcheva wrote:
>>=20
>>> We would like to ask the co-authors of the drafts to proceed with
>>> submitting the -00 WG documents, so that they can be submitted as
>>> such by the IETF 87 -00 cut-off date (July 8).
>>=20
>> Done!
>=20
> Also checked into source control:
>=20
> http://svn.tools.ietf.org/svn/wg/stox/

Thanks!

They're now approved as WG documents.

However I didn't see a submission for draft-ietf-stox-presence-00. Is =
this one coming later?

Cheers,
Yana

>=20
> Peter
>=20
> --=20
> Peter Saint-Andre
> https://stpeter.im/
>=20
>=20


From stpeter@stpeter.im  Mon Jul  1 15:35:26 2013
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From: Peter Saint-Andre <stpeter@stpeter.im>
Date: Mon, 1 Jul 2013 16:35:06 -0600
To: Yana Stamcheva <yana@jitsi.org>
Cc: "stox@ietf.org" <stox@ietf.org>
Subject: Re: [Stox] Consensus call: Adopting the "current" drafts as WG documents to match the planned deliverables
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That was an oversight. Will fix tonight.=20

Sent from mobile, might be terse=20

On Jul 1, 2013, at 4:04 PM, Yana Stamcheva <yana@jitsi.org> wrote:

> Hi Peter,
>=20
> On Jul 1, 2013, at 7:36 PM, Peter Saint-Andre <stpeter@stpeter.im> wrote:
>=20
>> On 7/1/13 11:29 AM, Peter Saint-Andre wrote:
>>> On 7/1/13 7:17 AM, Yana Stamcheva wrote:
>>>=20
>>>> We would like to ask the co-authors of the drafts to proceed with
>>>> submitting the -00 WG documents, so that they can be submitted as
>>>> such by the IETF 87 -00 cut-off date (July 8).
>>>=20
>>> Done!
>>=20
>> Also checked into source control:
>>=20
>> http://svn.tools.ietf.org/svn/wg/stox/
>=20
> Thanks!
>=20
> They're now approved as WG documents.
>=20
> However I didn't see a submission for draft-ietf-stox-presence-00. Is this=
 one coming later?
>=20
> Cheers,
> Yana
>=20
>>=20
>> Peter
>>=20
>> --=20
>> Peter Saint-Andre
>> https://stpeter.im/
>=20

From internet-drafts@ietf.org  Mon Jul  1 23:31:56 2013
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the SIP-TO-XMPP Working Group of the IETF.

	Title           : Interworking between the Session Initiation Protocol (SI=
P) and the Extensible Messaging and Presence Protocol (XMPP): Presence
	Author(s)       : Peter Saint-Andre
                          Avshalom Houri
                          Joe Hildebrand
	Filename        : draft-ietf-stox-presence-00.txt
	Pages           : 20
	Date            : 2013-07-01

Abstract:
   This document defines a bi-directional protocol mapping for the
   exchange of presence information 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-presence

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-stox-presence-00


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From yana@sip-communicator.org  Tue Jul  9 14:38:26 2013
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From: Yana Stamcheva <yana@jitsi.org>
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Date: Tue, 9 Jul 2013 23:38:08 +0200
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Subject: [Stox] Preliminary Agenda for IETF 87 STOX Meeting
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Hello Everyone,

Here's the preliminary agenda for the IETF 87 STOX meeting =
(http://tools.ietf.org/agenda/87/#THURSDAY):

* 5 min: Introduction: Note well, Administrativia, Agenda overview: =
Chairs

* 5 min: Status update and forthcoming milestones: Chairs

* 10 min: Peter Saint-Andre
	draft-ietf-stox-core
	draft-ietf-stox-presence
	draft-ietf-stox-im
	draft-ietf-stox-chat

* 10 min: Sa=FAl Ibarra Corretg=E9
	draft-ietf-stox-groupchat

* 40 min: Sa=FAl Ibarra Corretg=E9, Emil Ivov
	draft-ietf-stox-media

* 20 min: Summary and next steps: Chairs

In the above agenda we have tried to devote as much time as possible to =
the draft with the most open issues.

Please let us know if you have any suggestions and/or comments.

Cheers,
Yana & Markus









From yana@sip-communicator.org  Wed Jul 10 07:37:22 2013
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From: Yana Stamcheva <yana@jitsi.org>
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Date: Wed, 10 Jul 2013 16:37:11 +0200
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Subject: [Stox] Preliminary timetable for draft reviews
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Hello Everyone,

In order to make sure we stay on track and respect our milestones we =
would like to propose a preliminary timeline for the draft review =
deadlines and at the same time ask for volunteers to review!

Here's the preliminary timeline for the draft review deadlines:

August 16:
draft-ietf-stox-core
draft-ietf-stox-presence

August 30:
draft-ietf-stox-im
draft-ietf-stox-chat

September 13:
draft-ietf-stox-groupchat

September 27 **:
draft-ietf-stox-media

** We still have to see if the media draft would be mature enough at =
this point, but we expect that to be the case.

Note that these will also be the WG last call deadlines for each of =
these drafts. The final timeline would be determined at the meeting in =
Berlin.

Please contact the chairs if you're ready to volunteer as a reviewer on =
any of the above drafts.

Cheers,
Yana & Markus=

From prvs=9011ec0d2=devries_michael@bah.com  Mon Jul 15 09:36:40 2013
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From: "De Vries, Michael [USA]" <devries_michael@bah.com>
To: "stox@ietf.org" <stox@ietf.org>
Thread-Topic: STOX Error Condition/Response code translation
Thread-Index: Ac6BeW3GprYlO9XDTaec9Xmhavny5g==
Date: Mon, 15 Jul 2013 16:36:17 +0000
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Subject: [Stox] STOX Error Condition/Response code translation
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Hello Everyone,=0A=
=0A=
The draft-ietf-stox-core-00 document specifies mappings between XMPP error =
conditions and SIP response codes, but does not specify any requirements/re=
commendations for handling the SIP "Reason-Phrase" in the "Status-Line" hea=
der or the XMPP <text/> entry in the <error/> block.  According to the SIP =
and XMPP RFCs, the Reason-Phrase and <error/> block appear to effectively s=
erve the same purpose in conveying additional error information intended fo=
r the human user, so a direct translation between the two (when present) se=
ems reasonable to me.  =0A=
=0A=
Additionally, this could potentially help to distinguish errors which are c=
urrently mapped down from many possibilities into a single entry, such as S=
IP response codes 413-423 (and others) all being mapped to the <bad-request=
/> XMPP Error Condition, assuming that the SIP responses provide a Reason-P=
hrase.  Additionally, it may even be possible to provide the additional dis=
tinguishing information in the XMPP <text/> block even if the SIP response =
does not contain a Reason-Phrase for direct translation.=0A=
=0A=
For example:  Assume the XMPP-SIP gateway receives a SIP 415 response code =
(Unsupported Media Type) in response to some gateway-mediated action betwee=
n an XMPP user and SIP user. As per the draft-ieftf-stox-core-00 document, =
this response code should be translated and sent to the XMPP user effective=
ly as:=0A=
=0A=
  <error type=3D'modify'>=0A=
    <bad-request xmlns=3D'urn:ietf:params:xml:ns:xmpp-stanzas'/>=0A=
  </error>=0A=
=0A=
=0A=
If this 415 response contains a Reason-Phrase (though it does not appear to=
 be mandated as per RFC3261), then the gateway could instead send to the XM=
PP user:=0A=
=0A=
  <error type=3D'modify'>=0A=
    <bad-request xmlns=3D'urn:ietf:params:xml:ns:xmpp-stanzas'/>=0A=
    <text xml:lang=3D'en' xmlns=3D'urn:ietf:params:xml:ns:xmpp-stanzas'>=0A=
      [ ... Reason-Phrase translated here ... ]=0A=
    </text>=0A=
  </error>=0A=
=0A=
=0A=
Additionally, if this 415 response does not contain a Reason-Phrase, then t=
he gateway could send the following to the XMPP user as a means of distingu=
ishing from other <bad-request/> errors:=0A=
=0A=
  <error type=3D'modify'>=0A=
    <bad-request xmlns=3D'urn:ietf:params:xml:ns:xmpp-stanzas'/>=0A=
    <text xml:lang=3D'en' xmlns=3D'urn:ietf:params:xml:ns:xmpp-stanzas'>=0A=
      Unsupported Media Type=0A=
    </text>=0A=
  </error>=0A=
=0A=
=0A=
Similarly, a 416 response (Unsupported URI Scheme) without a Reason-Phrase =
could be translated into:=0A=
=0A=
  <error type=3D'modify'>=0A=
    <bad-request xmlns=3D'urn:ietf:params:xml:ns:xmpp-stanzas'/>=0A=
    <text xml:lang=3D'en' xmlns=3D'urn:ietf:params:xml:ns:xmpp-stanzas'>=0A=
      Unsupported URI Scheme=0A=
    </text>=0A=
  </error>=0A=
=0A=
=0A=
thus distinguishing between the 415 and 416 responses for the XMPP human us=
er, but effectively requiring some textual injection into the message strea=
m by the gateway which would need to be specified in the document.  That sa=
id, as these fields are all optional and not related to actual SIP/XMPP app=
lication-level handling of messages, there is no reason that the XMPP-SIP g=
ateway must perform such a translation, but I figured it might be worth exp=
loring in this discussion group.=0A=
=0A=
=0A=
Thanks,=0A=
--Mike DeVries=

From internet-drafts@ietf.org  Mon Jul 15 13:37:37 2013
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Subject: [Stox] I-D Action: draft-ietf-stox-media-01.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the SIP-TO-XMPP Working Group of the IETF.

	Title           : Interworking between the Session Initiation Protocol (SI=
P) and the Extensible Messaging and Presence Protocol (XMPP): Media Sessions
	Author(s)       : Peter Saint-Andre
                          Saul Ibarra
                          Emil Ivov
	Filename        : draft-ietf-stox-media-01.txt
	Pages           : 15
	Date            : 2013-07-15

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


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-01

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-stox-media-01


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


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Subject: [Stox] Fwd:  I-D Action: draft-ietf-stox-media-01.txt
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I uploaded a (very) raw -01 version os the media draft, looking forward =
to working on this in Berlin :-)


Begin forwarded message:

> From: internet-drafts@ietf.org
> Date: July 15, 2013 10:37:37 PM GMT+02:00
> To: i-d-announce@ietf.org
> Cc: stox@ietf.org
> Subject: [Stox] I-D Action: draft-ietf-stox-media-01.txt
>=20
>=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): Media Sessions
> 	Author(s)       : Peter Saint-Andre
>                          Saul Ibarra
>                          Emil Ivov
> 	Filename        : draft-ietf-stox-media-01.txt
> 	Pages           : 15
> 	Date            : 2013-07-15
>=20
> Abstract:
>   This document defines a bi-directional protocol mapping for use by
>   gateways that enable the exchange of media signalling messages
>   between systems that implement the Jingle extensions to the
>   Extensible Messaging and Presence Protocol (XMPP) and those that
>   implement the Session Initiation Protocol (SIP).
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-stox-media
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-stox-media-01
>=20
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-stox-media-01
>=20
>=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




From stpeter@stpeter.im  Wed Jul 17 20:23:33 2013
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From: Peter Saint-Andre <stpeter@stpeter.im>
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Subject: Re: [Stox] STOX Error Condition/Response code translation
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Hi Mike, thank you very much for the feedback!

On 7/15/13 10:36 AM, De Vries, Michael [USA] wrote:
> Hello Everyone,
> 
> The draft-ietf-stox-core-00 document specifies mappings between XMPP
> error conditions and SIP response codes, but does not specify any
> requirements/recommendations for handling the SIP "Reason-Phrase" in
> the "Status-Line" header or the XMPP <text/> entry in the <error/>
> block.  According to the SIP and XMPP RFCs, the Reason-Phrase and
> <error/> block appear to effectively serve the same purpose in
> conveying additional error information intended for the human user,
> so a direct translation between the two (when present) seems
> reasonable to me.

RFC 3261 says (Section 7.2):

   The Status-Code is a 3-digit integer result code that indicates the
   outcome of an attempt to understand and satisfy a request.  The
   Reason-Phrase is intended to give a short textual description of the
   Status-Code.  The Status-Code is intended for use by automata,
   whereas the Reason-Phrase is intended for the human user.  A client
   is not required to examine or display the Reason-Phrase.

And RFC 6120 says (Section 8.3.2):

   The <text/> element is OPTIONAL.  If included, it is to be used only
   to provide descriptive or diagnostic information that supplements the
   meaning of a defined condition or application-specific condition.  It
   MUST NOT be interpreted programmatically by an application.  It
   SHOULD NOT be used as the error message presented to a human user,
   but MAY be shown in addition to the error message associated with the
   defined condition element (and, optionally, the application-specific
   condition element).

Despite the caveat about clients not directly copying the <text/>
element and using it as the error message shown to a user, I tend to
agree that they are mostly equivalent and can reasonably be mapped to
each other for interworking purposes.

> Additionally, this could potentially help to distinguish errors which
> are currently mapped down from many possibilities into a single
> entry, such as SIP response codes 413-423 (and others) all being
> mapped to the <bad-request/> XMPP Error Condition, assuming that the
> SIP responses provide a Reason-Phrase.  Additionally, it may even be
> possible to provide the additional distinguishing information in the
> XMPP <text/> block even if the SIP response does not contain a
> Reason-Phrase for direct translation.
> 
> For example:  Assume the XMPP-SIP gateway receives a SIP 415 response
> code (Unsupported Media Type) in response to some gateway-mediated
> action between an XMPP user and SIP user. As per the
> draft-ieftf-stox-core-00 document, this response code should be
> translated and sent to the XMPP user effectively as:
> 
> <error type='modify'> <bad-request
> xmlns='urn:ietf:params:xml:ns:xmpp-stanzas'/> </error>
> 
> 
> If this 415 response contains a Reason-Phrase (though it does not
> appear to be mandated as per RFC3261), then the gateway could instead
> send to the XMPP user:
> 
> <error type='modify'> <bad-request
> xmlns='urn:ietf:params:xml:ns:xmpp-stanzas'/> <text xml:lang='en'
> xmlns='urn:ietf:params:xml:ns:xmpp-stanzas'> [ ... Reason-Phrase
> translated here ... ] </text> </error>
> 
> 
> Additionally, if this 415 response does not contain a Reason-Phrase,
> then the gateway could send the following to the XMPP user as a means
> of distinguishing from other <bad-request/> errors:
> 
> <error type='modify'> <bad-request
> xmlns='urn:ietf:params:xml:ns:xmpp-stanzas'/> <text xml:lang='en'
> xmlns='urn:ietf:params:xml:ns:xmpp-stanzas'> Unsupported Media Type 
> </text> </error>
> 
> 
> Similarly, a 416 response (Unsupported URI Scheme) without a
> Reason-Phrase could be translated into:
> 
> <error type='modify'> <bad-request
> xmlns='urn:ietf:params:xml:ns:xmpp-stanzas'/> <text xml:lang='en'
> xmlns='urn:ietf:params:xml:ns:xmpp-stanzas'> Unsupported URI Scheme 
> </text> </error>
> 
> 
> thus distinguishing between the 415 and 416 responses for the XMPP
> human user, but effectively requiring some textual injection into the
> message stream by the gateway which would need to be specified in the
> document.  That said, as these fields are all optional and not
> related to actual SIP/XMPP application-level handling of messages,
> there is no reason that the XMPP-SIP gateway must perform such a
> translation, but I figured it might be worth exploring in this
> discussion group.

IMHO your proposal is reasonable as suggested behavior, although I don't
think it rises to the level of a MUST or SHOULD requirement.

I will look into it further and propose some text on the list.

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



From yana@sip-communicator.org  Thu Jul 18 07:09:00 2013
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Subject: [Stox] Agenda for IETF 87 STOX Meeting
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Hello Everyone,

We have submitted the agenda for the IETF 87 STOX meeting =
(http://tools.ietf.org/agenda/87/#THURSDAY): =
http://www.ietf.org/proceedings/87/agenda/agenda-87-stox .

Quick summary:

17:00-17:10 Introduction and Status Update, Forthcoming milestones =
(Chairs, 10 mins)
         =20
17:10-17:20 	  draft-ietf-stox-core-00 (Peter Saint-Andre, 10 mins)
                          draft-ietf-stox-presence-00
			  draft-ietf-stox-im-00
			  draft-ietf-stox-chat-00

17:20-17:40	draft-ietf-stox-groupchat-00 (Sa=FAl Ibarra Corretg=E9, =
20 mins)

17:40-18:10	draft-ietf-stox-media-00 (Sa=FAl Ibarra Corretg=E9, Emil =
Ivov, 30 mins)
         =20
18:10-18:30, Summary and Next Steps (Chairs, 20 mins)

As we have mentioned earlier, in the above agenda we have tried to =
devote as much time as possible to the drafts with most open issues.

Thanks to all of you who have sent us feedback.

Cheers,
Yana & Markus=

From michaellundberg.ietf@gmail.com  Fri Jul 19 05:01:08 2013
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Subject: [Stox] Comments on draft-ietf-stox-presence-00
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--089e0149331eeb3e1704e1dc14d1
Content-Type: text/plain; charset=ISO-8859-1

Peter,
I think draft-ietf-stox-presence-00 looks good.  I had a previous comment
to an earlier version that was not addressed that I'm incuding it again as
I think it is important for the document.

I think it would be good to add the <show/> element to Table 7, with a note
similar to the one provided in Table 6.  This will provide a method for
mapping 'away' and 'dnd' information in the opposite direction.  Similar to
the note in Table 6, this would require the SIP implementation to support
the 'jabber:client' namespace.  If the SIP implementation supports the
namespace, the gateway can then map the values directly into the <show/>
element of the XMPP presence messages.

The only other thing I saw was that the below sentence in Section 3.2.3 is
missing the word "send" after SHOULD.

Current:

Upon sending the transformed unsubscribe, the XMPP-SIMPLE gateway SHOULD a
presence stanza of type "unsubscribed" to the XMPP user:

New:

Upon sending the transformed unsubscribe, the XMPP-SIMPLE gateway SHOULD
send a presence stanza of type "unsubscribed" to the XMPP user:

Regards,
Michael

--089e0149331eeb3e1704e1dc14d1
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><p>Peter,</p><div>I think draft-ietf-stox-presence-00 look=
s good.=A0 I had a previous comment to an earlier version that was not addr=
essed that I&#39;m incuding it again as I think it is important for the doc=
ument. </div>
<div>=A0</div><div>I think it would be good to add the &lt;show/&gt; elemen=
t to Table 7, with a note similar to the one provided in Table 6.=A0 This w=
ill provide a method for mapping &#39;away&#39; and &#39;dnd&#39; informati=
on in the opposite direction.=A0 Similar to the note in Table 6, this would=
 require the SIP implementation to support the &#39;jabber:client&#39; name=
space.=A0 If the SIP implementation supports the namespace, the gateway can=
 then map the values directly into the &lt;show/&gt; element of the XMPP pr=
esence messages.</div>
<p>The only other thing I saw was that the below sentence in Section 3.2.3 =
is missing the word &quot;send&quot; after SHOULD.</p><p>Current: </p><p>Up=
on sending the transformed unsubscribe, the XMPP-SIMPLE gateway SHOULD a pr=
esence stanza of type &quot;unsubscribed&quot; to the XMPP user:</p>
<p>New: </p><p>Upon sending the transformed unsubscribe, the XMPP-SIMPLE ga=
teway SHOULD send a presence stanza of type &quot;unsubscribed&quot; to the=
 XMPP user: </p><p>Regards,<br>Michael</p></div>

--089e0149331eeb3e1704e1dc14d1--

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From: "De Vries, Michael [USA]" <devries_michael@bah.com>
To: "stox@ietf.org" <stox@ietf.org>
Thread-Topic: Nickname Usage Conflict Resolution
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Date: Fri, 19 Jul 2013 23:07:23 +0000
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Subject: [Stox] Nickname Usage Conflict Resolution
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Hello everyone,

I have a concern with the suggested possible resolutions for a Nickname Usa=
ge Conflict in the draft-ietf-stox-groupchat-00 document.  In section 3.2, =
the document says

|   Alternatively [to translating the MSRP 425 message to a <conflict/> err=
or],
|   the gateway might generate a new nickname request on
|   behalf of the XMPP user, thus shielding the XMPP client from handling
|   the conflict error.

Additionally, section 5 says

|   If there is a conflict between the SIP nickname and the XMPP
|   nickname, the SIP-to-XMPP or XMPP-to-SIP gateway is responsible for
|   adjusting the nickname to avoid the conflict and for informing the
|   SIP or XMPP client of the unique nickname used to join the chatroom.

First, section 3.2 suggests that the gateway may choose to generate a new n=
ickname, but can instead merely forward on the translated MSRP 425 or <conf=
lict/> error, whereas section 5 says that the gateway is responsible for (a=
ka MUST) adjusting the nickname in the event of a conflict.  Both of these =
sections suggest that in the case of a nickname conflict the gateway may ac=
t independently of the client to change the client-generated nickname to on=
e which is unique within the context of the chatroom.  I'm not certain this=
 is the most appropriate approach.  Nicknames are important in that they ar=
e needed to facilitate private messaging and other features, as well as bei=
ng an important part of user identification.  It seems risky to automatical=
ly generate a new nickname when a conflict is found, instead of merely pres=
enting the error to the user (via MSRP 425 message or <conflict/> error) so=
 that they can choose a new nickname via their client.  Additionally, a cur=
sory examination of the XMPP, MSRP, and MUC RFCs/XEPs does not appear to sh=
ow any recommendation that nicknames be automatically generated in the even=
t of a conflict.

I believe that it would be better to require that the gateway merely forwar=
d on the translated MSRP 425 or <conflict/> error instead of attempting to =
address the conflict automatically, but I may not sufficiently understand t=
he reasoning behind the automated nickname resolution.

Thanks,
--Mike DeVries

From saul@ag-projects.com  Sat Jul 20 06:48:32 2013
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From: =?iso-8859-1?Q?Sa=FAl_Ibarra_Corretg=E9?= <saul@ag-projects.com>
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To: "De Vries, Michael [USA]" <devries_michael@bah.com>
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Subject: Re: [Stox] Nickname Usage Conflict Resolution
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Hi Michael,

On Jul 20, 2013, at 1:07 AM, De Vries, Michael [USA] wrote:

> Hello everyone,
>=20
> I have a concern with the suggested possible resolutions for a =
Nickname Usage Conflict in the draft-ietf-stox-groupchat-00 document.  =
In section 3.2, the document says
>=20
> |   Alternatively [to translating the MSRP 425 message to a =
<conflict/> error],
> |   the gateway might generate a new nickname request on
> |   behalf of the XMPP user, thus shielding the XMPP client from =
handling
> |   the conflict error.
>=20
> Additionally, section 5 says
>=20
> |   If there is a conflict between the SIP nickname and the XMPP
> |   nickname, the SIP-to-XMPP or XMPP-to-SIP gateway is responsible =
for
> |   adjusting the nickname to avoid the conflict and for informing the
> |   SIP or XMPP client of the unique nickname used to join the =
chatroom.
>=20
> First, section 3.2 suggests that the gateway may choose to generate a =
new nickname, but can instead merely forward on the translated MSRP 425 =
or <conflict/> error, whereas section 5 says that the gateway is =
responsible for (aka MUST) adjusting the nickname in the event of a =
conflict.  Both of these sections suggest that in the case of a nickname =
conflict the gateway may act independently of the client to change the =
client-generated nickname to one which is unique within the context of =
the chatroom.  I'm not certain this is the most appropriate approach.  =
Nicknames are important in that they are needed to facilitate private =
messaging and other features, as well as being an important part of user =
identification.  It seems risky to automatically generate a new nickname =
when a conflict is found, instead of merely presenting the error to the =
user (via MSRP 425 message or <conflict/> error) so that they can choose =
a new nickname via their client.  Additionally, a cursory examinatio
> n of the XMPP, MSRP, and MUC RFCs/XEPs does not appear to show any =
recommendation that nicknames be automatically generated in the event of =
a conflict.
>=20
> I believe that it would be better to require that the gateway merely =
forward on the translated MSRP 425 or <conflict/> error instead of =
attempting to address the conflict automatically, but I may not =
sufficiently understand the reasoning behind the automated nickname =
resolution.
>=20

I agree. This is how things are usually handled in MSRP. When testing =
things in XMPP however, not every client is able to properly deal with =
nickname conflicts though. Peter, maybe this was the reason why that =
text was added?


Regards,

--
Sa=FAl Ibarra Corretg=E9
AG Projects




From ralphm@ik.nu  Mon Jul 22 08:06:13 2013
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From: Ralph Meijer <ralphm@ik.nu>
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Subject: [Stox] Proposed message to send to the IETF rtcweb and W3C WebRTC working groups.
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Hi all,

I would like to inform the group of the recent formation [1] of the 
Jingle Special Interest Group (SIG) at the XMPP Standards Foundation 
(XSF). The recent increase of activity in the WebRTC and rtcweb working 
groups and related high-profile product developments and announcements 
were reasons for the XMPP Council to decide to concentrate efforts 
around Jingle in a SIG.

Jingle [2] is a general framework for managing media sessions between 
XMPP Sessions, including, but not limited to, audio/video streams, file 
transfer and application sharing. There are several documents describing 
applications of Jingle and the used transports, most linked from the 
overall framework specification [3].

The specification of Jingle RTP Sessions [4], most relevant to these 
working groups, defines a Jingle application type for negotiating RTP 
sessions. It has been designed such that interoperability with SIP-based 
systems is possible. This includes mapping negotiation parameters to and 
from SDP, while remaining a signaling protocol in its own right (not 
merely SDP in angle brackets).

The following work items were defined in the kick-off meeting last 
Wednesday, July 17 [5, raw log 6]:

  * Re-examining the state of the various Jingle proposals.
  * Polishing Jingle File Transfer.
  * Updating the SDP mapping in [4], including BUNDLE and Trickle-ICE
    improvements.
  * Documenting and communicating the value proposition of Jingle/XMPP.

This SIG already includes a number of people participating in 
discussions on the WebRTC and rtcweb mailing lists and is lead by Dave 
Cridland (chair), Philipp Hancke, Lance Stout and myself. It is open to 
anyone, and we are looking forward to cooperate with the WebRTC and 
rtcweb working groups to improve both WebRTC and Jingle.

The discussion venues are the Jingle mailing list [7] and the Jingle 
XMPP multi-user chat room [8]. Our next meeting in the MUC room is 
Wednesday July 24 at 15:30 UTC. This group's experience and input would 
be highly appreciated, and your participation in both the meeting and 
the on-going discussion would be most welcome.

Thanks,

Ralph Meijer

[1] <http://mail.jabber.org/pipermail/jingle/2013-June/001933.html>
[2] <http://xmpp.org/about-xmpp/technology-overview/jingle/>
[3] <http://xmpp.org/extensions/xep-0166.html>
[4] <http://xmpp.org/extensions/xep-0167.html>
[5] <http://mail.jabber.org/pipermail/jingle/2013-July/001956.html>
[6] <http://logs.xmpp.org/jingle/130717/>
[7] <http://mail.jabber.org/mailman/listinfo/jingle>
[8] <xmpp:jingle@muc.xmpp.org?join>

From ralphm@ik.nu  Mon Jul 22 08:47:26 2013
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Subject: [Stox] XSF Jingle Special Interest Group.
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On 2013-07-22 17:14, I=C3=B1aki Baz Castillo wrote:
> Great. First thing you should complain about is the fact that current
> WebRTC specification makes unfeasible for a browser to use SDP-XML as
> defined by XEP-0167. So if you have a SIP server you will be able to
> directly connect from the browser, but if you have a Jingle server you
> will need a gateway.


Hi I=C3=B1aki,

I have been following the recent discussions for a while now, and think=20
I understand the various stand points, including yours.

As I mentioned in my announcement, mapping SDP to Jingle's negotiation=20
parameters is indeed one of the concerns we are going to look into.=20
Philipp's work on his WebRTC plugin for Strophe.js [1], which uses=20
Jingle signalling, has shown that it indeed takes quite some effort to=20
map and mangle SDP to get things going. The spreadsheet questionnaire=20
that Peter Thatcher asked people to fill in (thanks!) also has several=20
comments to that effect. During our discussions on the Jingle mailing=20
list and the meeting last week, more of such comments were made.

I want to stress, again, that Jingle's negotiation parameters should not=20
be referred to as a flavor of SDP expressed in XML. This is misleading,=20
as it is not managed by MMUSIC, and actually weakens your point of not=20
wanting SDP as an API surface. I can understand that looking at the=20
current specification could give that impression, though. The Jingle=20
examples are currently interleaved by SDP examples to show the mapping,=20
and we are looking into making the distinction clearer, maybe even as a=20
document separate from this one.

[1] https://github.com/ESTOS/strophe.jingle

PS. Yes, I messed up the subject header of the original announcement.

--=20
ralphm

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [Stox] [rtcweb] Proposed message to send to the IETF rtcweb and W3C WebRTC working groups.
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Great. First thing you should complain about is the fact that current
WebRTC specification makes unfeasible for a browser to use SDP-XML as
defined by XEP-0167. So if you have a SIP server you will be able to
directly connect from the browser, but if you have a Jingle server you
will need a gateway.

2013/7/22 Ralph Meijer <ralphm@ik.nu>:
> Hi all,
>
> I would like to inform the group of the recent formation [1] of the Jingl=
e
> Special Interest Group (SIG) at the XMPP Standards Foundation (XSF). The
> recent increase of activity in the WebRTC and rtcweb working groups and
> related high-profile product developments and announcements were reasons =
for
> the XMPP Council to decide to concentrate efforts around Jingle in a SIG.
>
> Jingle [2] is a general framework for managing media sessions between XMP=
P
> Sessions, including, but not limited to, audio/video streams, file transf=
er
> and application sharing. There are several documents describing applicati=
ons
> of Jingle and the used transports, most linked from the overall framework
> specification [3].
>
> The specification of Jingle RTP Sessions [4], most relevant to these work=
ing
> groups, defines a Jingle application type for negotiating RTP sessions. I=
t
> has been designed such that interoperability with SIP-based systems is
> possible. This includes mapping negotiation parameters to and from SDP,
> while remaining a signaling protocol in its own right (not merely SDP in
> angle brackets).
>
> The following work items were defined in the kick-off meeting last
> Wednesday, July 17 [5, raw log 6]:
>
>  * Re-examining the state of the various Jingle proposals.
>  * Polishing Jingle File Transfer.
>  * Updating the SDP mapping in [4], including BUNDLE and Trickle-ICE
>    improvements.
>  * Documenting and communicating the value proposition of Jingle/XMPP.
>
> This SIG already includes a number of people participating in discussions=
 on
> the WebRTC and rtcweb mailing lists and is lead by Dave Cridland (chair),
> Philipp Hancke, Lance Stout and myself. It is open to anyone, and we are
> looking forward to cooperate with the WebRTC and rtcweb working groups to
> improve both WebRTC and Jingle.
>
> The discussion venues are the Jingle mailing list [7] and the Jingle XMPP
> multi-user chat room [8]. Our next meeting in the MUC room is Wednesday J=
uly
> 24 at 15:30 UTC. This group's experience and input would be highly
> appreciated, and your participation in both the meeting and the on-going
> discussion would be most welcome.
>
> Thanks,
>
> Ralph Meijer
>
> [1] <http://mail.jabber.org/pipermail/jingle/2013-June/001933.html>
> [2] <http://xmpp.org/about-xmpp/technology-overview/jingle/>
> [3] <http://xmpp.org/extensions/xep-0166.html>
> [4] <http://xmpp.org/extensions/xep-0167.html>
> [5] <http://mail.jabber.org/pipermail/jingle/2013-July/001956.html>
> [6] <http://logs.xmpp.org/jingle/130717/>
> [7] <http://mail.jabber.org/mailman/listinfo/jingle>
> [8] <xmpp:jingle@muc.xmpp.org?join>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 22 Jul 2013 17:44:54 +0200
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Subject: Re: [Stox] [rtcweb] Proposed message to send to the IETF rtcweb and W3C WebRTC working groups.
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2013/7/22 Daniel-Constantin Mierla <miconda@gmail.com>:
> On 7/22/13 5:14 PM, I=C3=B1aki Baz Castillo wrote:
>>
>> Great. First thing you should complain about is the fact that current
>> WebRTC specification makes unfeasible for a browser to use SDP-XML as
>> defined by XEP-0167. So if you have a SIP server you will be able to
>> directly connect from the browser, but if you have a Jingle server you
>> will need a gateway.
>
> You are obviously misinforming here. SIP is the signaling protocol and a =
SIP
> server has really little to deal with SDP -- I'm sure you know that.

I was talking about a SIP device also implementing WebRTC in the media
plane. Current WebRTC spec mandates plain-SDP usage in the wire to
signal your media description and transport/ICE information to the
peer. So if you want to communicate with a XEP-0167 compliant
server/endpoint, then you need a gateway to convert the plain-SDP
generated by the browser into the SDP-XML version defined by XEP-0167.



> And one
> cannot call directly a SIP endpoint from the browser, as SIP is not a
> mandatory signaling protocol, so there is extensive need of coding a
> javascript SIP stack (or reusing an existing one).

Reusing existing JavaScript SIP stacks is something good, don't you agree?


Please, let's focus:


Today from a browser you can speak SIP over WebSocket and connect to a
SIP media server/gateway understanding the SDP of WebRTC. So yes, you
can talk SIP fom a browser.

Today fom a browser you cannot speak XMPP/Jingle (XEP-0167) over
WebSocket (or over AJAX) because the browser produces a plain SDP
blob, while you need a XML based SDP as XEP-1067 states. You can parse
such a SDP blob string in JavaScript and map it into a XML body,
but... good luck with that...


Hope it is clear now.


Regards.




--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [Stox] XSF Jingle Special Interest Group.
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2013/7/22 Ralph Meijer <ralphm@ik.nu>:
> Hi I=C3=B1aki,
>
> I have been following the recent discussions for a while now, and think I
> understand the various stand points, including yours.
>
> As I mentioned in my announcement, mapping SDP to Jingle's negotiation
> parameters is indeed one of the concerns we are going to look into.
> Philipp's work on his WebRTC plugin for Strophe.js [1], which uses Jingle
> signalling, has shown that it indeed takes quite some effort to map and
> mangle SDP to get things going. The spreadsheet questionnaire that Peter
> Thatcher asked people to fill in (thanks!) also has several comments to t=
hat
> effect. During our discussions on the Jingle mailing list and the meeting
> last week, more of such comments were made.
>
> I want to stress, again, that Jingle's negotiation parameters should not =
be
> referred to as a flavor of SDP expressed in XML. This is misleading, as i=
t
> is not managed by MMUSIC, and actually weakens your point of not wanting =
SDP
> as an API surface. I can understand that looking at the current
> specification could give that impression, though. The Jingle examples are
> currently interleaved by SDP examples to show the mapping, and we are
> looking into making the distinction clearer, maybe even as a document
> separate from this one.

Thanks for the clarification. So Jingle's session description is not
just a plain-to-XML mapping of a common SDP. Good to know.

However, wouldn't you prefer to have a pure JS Object with all the
media and transport information you require to build your Jingle
XML-SDP instead of having to parse a plain SDP string blob?

Thanks a lot.


--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [Stox] [rtcweb] Proposed message to send to the IETF rtcweb and W3C WebRTC working groups.
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On 7/22/13 5:14 PM, IÃ±aki Baz Castillo wrote:
> Great. First thing you should complain about is the fact that current
> WebRTC specification makes unfeasible for a browser to use SDP-XML as
> defined by XEP-0167. So if you have a SIP server you will be able to
> directly connect from the browser, but if you have a Jingle server you
> will need a gateway.
You are obviously misinforming here. SIP is the signaling protocol and a 
SIP server has really little to deal with SDP -- I'm sure you know that. 
And one cannot call directly a SIP endpoint from the browser, as SIP is 
not a mandatory signaling protocol, so there is extensive need of coding 
a javascript SIP stack (or reusing an existing one).

Cheers,
Daniel

>
> 2013/7/22 Ralph Meijer <ralphm@ik.nu>:
>> Hi all,
>>
>> I would like to inform the group of the recent formation [1] of the Jingle
>> Special Interest Group (SIG) at the XMPP Standards Foundation (XSF). The
>> recent increase of activity in the WebRTC and rtcweb working groups and
>> related high-profile product developments and announcements were reasons for
>> the XMPP Council to decide to concentrate efforts around Jingle in a SIG.
>>
>> Jingle [2] is a general framework for managing media sessions between XMPP
>> Sessions, including, but not limited to, audio/video streams, file transfer
>> and application sharing. There are several documents describing applications
>> of Jingle and the used transports, most linked from the overall framework
>> specification [3].
>>
>> The specification of Jingle RTP Sessions [4], most relevant to these working
>> groups, defines a Jingle application type for negotiating RTP sessions. It
>> has been designed such that interoperability with SIP-based systems is
>> possible. This includes mapping negotiation parameters to and from SDP,
>> while remaining a signaling protocol in its own right (not merely SDP in
>> angle brackets).
>>
>> The following work items were defined in the kick-off meeting last
>> Wednesday, July 17 [5, raw log 6]:
>>
>>   * Re-examining the state of the various Jingle proposals.
>>   * Polishing Jingle File Transfer.
>>   * Updating the SDP mapping in [4], including BUNDLE and Trickle-ICE
>>     improvements.
>>   * Documenting and communicating the value proposition of Jingle/XMPP.
>>
>> This SIG already includes a number of people participating in discussions on
>> the WebRTC and rtcweb mailing lists and is lead by Dave Cridland (chair),
>> Philipp Hancke, Lance Stout and myself. It is open to anyone, and we are
>> looking forward to cooperate with the WebRTC and rtcweb working groups to
>> improve both WebRTC and Jingle.
>>
>> The discussion venues are the Jingle mailing list [7] and the Jingle XMPP
>> multi-user chat room [8]. Our next meeting in the MUC room is Wednesday July
>> 24 at 15:30 UTC. This group's experience and input would be highly
>> appreciated, and your participation in both the meeting and the on-going
>> discussion would be most welcome.
>>
>> Thanks,
>>
>> Ralph Meijer
>>
>> [1] <http://mail.jabber.org/pipermail/jingle/2013-June/001933.html>
>> [2] <http://xmpp.org/about-xmpp/technology-overview/jingle/>
>> [3] <http://xmpp.org/extensions/xep-0166.html>
>> [4] <http://xmpp.org/extensions/xep-0167.html>
>> [5] <http://mail.jabber.org/pipermail/jingle/2013-July/001956.html>
>> [6] <http://logs.xmpp.org/jingle/130717/>
>> [7] <http://mail.jabber.org/mailman/listinfo/jingle>
>> [8] <xmpp:jingle@muc.xmpp.org?join>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda


From miconda@gmail.com  Mon Jul 22 09:22:39 2013
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Subject: Re: [Stox] [rtcweb] Proposed message to send to the IETF rtcweb and W3C WebRTC working groups.
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On 7/22/13 5:44 PM, IÃ±aki Baz Castillo wrote:
> 2013/7/22 Daniel-Constantin Mierla <miconda@gmail.com>:
>> On 7/22/13 5:14 PM, IÃ±aki Baz Castillo wrote:
>>> Great. First thing you should complain about is the fact that current
>>> WebRTC specification makes unfeasible for a browser to use SDP-XML as
>>> defined by XEP-0167. So if you have a SIP server you will be able to
>>> directly connect from the browser, but if you have a Jingle server you
>>> will need a gateway.
>> You are obviously misinforming here. SIP is the signaling protocol and a SIP
>> server has really little to deal with SDP -- I'm sure you know that.
> I was talking about a SIP device also implementing WebRTC in the media
> plane.
You wrote a SIP server, just read above.

And producing a xml blob instead of text plain blob does not make much 
difference from the architecture point of view, if that was your 
concern, nor simplifies things.

Cheers,
Daniel

>   Current WebRTC spec mandates plain-SDP usage in the wire to
> signal your media description and transport/ICE information to the
> peer. So if you want to communicate with a XEP-0167 compliant
> server/endpoint, then you need a gateway to convert the plain-SDP
> generated by the browser into the SDP-XML version defined by XEP-0167.
>
>
>
>> And one
>> cannot call directly a SIP endpoint from the browser, as SIP is not a
>> mandatory signaling protocol, so there is extensive need of coding a
>> javascript SIP stack (or reusing an existing one).
> Reusing existing JavaScript SIP stacks is something good, don't you agree?
>
>
> Please, let's focus:
>
>
> Today from a browser you can speak SIP over WebSocket and connect to a
> SIP media server/gateway understanding the SDP of WebRTC. So yes, you
> can talk SIP fom a browser.
>
> Today fom a browser you cannot speak XMPP/Jingle (XEP-0167) over
> WebSocket (or over AJAX) because the browser produces a plain SDP
> blob, while you need a XML based SDP as XEP-1067 states. You can parse
> such a SDP blob string in JavaScript and map it into a XML body,
> but... good luck with that...
>
>
> Hope it is clear now.
>
>
> Regards.
>
>
>
>
> --
> IÃ±aki Baz Castillo
> <ibc@aliax.net>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda


From emil@sip-communicator.org  Mon Jul 22 09:28:06 2013
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Subject: Re: [Stox] [rtcweb] Proposed message to send to the IETF rtcweb and W3C WebRTC working groups.
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On 22.07.13, 17:44, I=C3=B1aki Baz Castillo wrote:
> 2013/7/22 Daniel-Constantin Mierla <miconda@gmail.com>:
>> On 7/22/13 5:14 PM, I=C3=B1aki Baz Castillo wrote:
>>>
>>> Great. First thing you should complain about is the fact that current=

>>> WebRTC specification makes unfeasible for a browser to use SDP-XML as=

>>> defined by XEP-0167. So if you have a SIP server you will be able to
>>> directly connect from the browser, but if you have a Jingle server yo=
u
>>> will need a gateway.
>>
>> You are obviously misinforming here. SIP is the signaling protocol and=
 a SIP
>> server has really little to deal with SDP -- I'm sure you know that.
>
> I was talking about a SIP device also implementing WebRTC in the media
> plane. Current WebRTC spec mandates plain-SDP usage in the wire to
> signal your media description and transport/ICE information to the
> peer.

I don't think this is true. You can very well translate the SDP to=20
Jingle in the browser and then only send XML on the wire.

I completely agree that this would be a pain, but there is nothing that=20
"mandates" against it.

Emil

--=20
https://jitsi.org


From thp@westhawk.co.uk  Mon Jul 22 09:29:51 2013
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Subject: Re: [Stox] [rtcweb] Proposed message to send to the IETF rtcweb and W3C WebRTC working groups.
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On 22 Jul 2013, at 17:26, Emil Ivov <emcho@jitsi.org> wrote:

>=20
>=20
> On 22.07.13, 17:44, I=F1aki Baz Castillo wrote:
>> 2013/7/22 Daniel-Constantin Mierla <miconda@gmail.com>:
>>> On 7/22/13 5:14 PM, I=F1aki Baz Castillo wrote:
>>>>=20
>>>> Great. First thing you should complain about is the fact that =
current
>>>> WebRTC specification makes unfeasible for a browser to use SDP-XML =
as
>>>> defined by XEP-0167. So if you have a SIP server you will be able =
to
>>>> directly connect from the browser, but if you have a Jingle server =
you
>>>> will need a gateway.
>>>=20
>>> You are obviously misinforming here. SIP is the signaling protocol =
and a SIP
>>> server has really little to deal with SDP -- I'm sure you know that.
>>=20
>> I was talking about a SIP device also implementing WebRTC in the =
media
>> plane. Current WebRTC spec mandates plain-SDP usage in the wire to
>> signal your media description and transport/ICE information to the
>> peer.
>=20
> I don't think this is true. You can very well translate the SDP to =
Jingle in the browser and then only send XML on the wire.
>=20
> I completely agree that this would be a pain, but there is nothing =
that "mandates" against it.

It is a pain - but it is do-able - take a look at PhonoSDK on github.

T.

>=20
> Emil
>=20
> --=20
> https://jitsi.org
>=20
>=20


From fippo@goodadvice.pages.de  Mon Jul 22 10:37:18 2013
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Subject: Re: [Stox] Proposed message to send to the IETF rtcweb and W3C WebRTC working groups.
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Am 22.07.2013 17:05, schrieb Ralph Meijer:
> Hi all,
>
> I would like to inform the group of the recent formation [1] of the
> Jingle Special Interest Group (SIG) at the XMPP Standards Foundation
> (XSF). The recent increase of activity in the WebRTC and rtcweb working
> groups and related high-profile product developments and announcements
> were reasons for the XMPP Council to decide to concentrate efforts
> around Jingle in a SIG.

The point where this is interesting for STOX is mostly the -media draft. 
For example we can fix some of the bugs (like a=fmtp) in XSF territory.

Potentially, the -media draft is even the place where to define the 
mapping between jingles xmlish session description and SDP. Even though 
I would attempt to clearly separate that from jingle signalling 
conceptually and in the document.

From stpeter@stpeter.im  Tue Jul 23 19:57:41 2013
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Subject: Re: [Stox] Nickname Usage Conflict Resolution
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On 7/20/13 7:48 AM, Saúl Ibarra Corretgé wrote:
> Hi Michael,
> 
> On Jul 20, 2013, at 1:07 AM, De Vries, Michael [USA] wrote:
> 
>> Hello everyone,
>> 
>> I have a concern with the suggested possible resolutions for a 
>> Nickname Usage Conflict in the draft-ietf-stox-groupchat-00 
>> document.  In section 3.2, the document says
>> 
>> |   Alternatively [to translating the MSRP 425 message to a 
>> <conflict/> error], |   the gateway might generate a new
>> nickname request on |   behalf of the XMPP user, thus shielding
>> the XMPP client from handling |   the conflict error.
>> 
>> Additionally, section 5 says
>> 
>> |   If there is a conflict between the SIP nickname and the XMPP
>> | nickname, the SIP-to-XMPP or XMPP-to-SIP gateway is
>> responsible for |   adjusting the nickname to avoid the conflict
>> and for informing the |   SIP or XMPP client of the unique
>> nickname used to join the chatroom.
>> 
>> First, section 3.2 suggests that the gateway may choose to
>> generate a new nickname, but can instead merely forward on the
>> translated MSRP 425 or <conflict/> error, whereas section 5 says
>> that the gateway is responsible for (aka MUST) adjusting the
>> nickname in the event of a conflict.  Both of these sections
>> suggest that in the case of a nickname conflict the gateway may
>> act independently of the client to change the client-generated
>> nickname to one which is unique within the context of the
>> chatroom.  I'm not certain this is the most appropriate approach.
>> Nicknames are important in that they are needed to facilitate
>> private messaging and other features, as well as being an
>> important part of user identification.  It seems risky to
>> automatically generate a new nickname when a conflict is found,
>> instead of merely presenting the error to the user (via MSRP 425
>> message or <conflict/> error) so that they can choose a new
>> nickname via their client.  Additionally, a cursory examinatio n
>> of the XMPP, MSRP, and MUC RFCs/XEPs does not appear to show any
>> recommendation that nicknames be automatically generated in the
>> event of a conflict.
>> 
>> I believe that it would be better to require that the gateway 
>> merely forward on the translated MSRP 425 or <conflict/> error 
>> instead of attempting to address the conflict automatically, but
>> I may not sufficiently understand the reasoning behind the
>> automated nickname resolution.
>> 
> 
> I agree. This is how things are usually handled in MSRP. When
> testing things in XMPP however, not every client is able to
> properly deal with nickname conflicts though. Peter, maybe this was
> the reason why that text was added?

I doubt that was the reason, since I don't believe in designing
protocols around buggy software. :-)

Many XMPP clients will indeed shield the user from making a decision
about the nickname (e.g., if "stpeter" is taken they will cycle
through nicks like "stpeter1" or "stpeter_" or whatever). So pushing
an error back to the originating client seems fine to me in all cases.

Question: if the gateway were to choose a nick on behalf of the
client, are we sure that the chosen nick would get communicated back
to the originating client?

In any case, I think that what Mike proposes is sensible.

Peter

- -- 
Peter Saint-Andre
https://stpeter.im/


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From stpeter@stpeter.im  Tue Jul 23 20:08:41 2013
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Hi Mike, thanks for the feedback.

On 7/19/13 6:01 AM, Michael Lundberg wrote:
> Peter,
> 
> I think draft-ietf-stox-presence-00 looks good.  I had a previous 
> comment to an earlier version that was not addressed that I'm
> incuding it again as I think it is important for the document.

Sorry I missed that.

> I think it would be good to add the <show/> element to Table 7,
> with a note similar to the one provided in Table 6. This will
> provide a method for mapping 'away' and 'dnd' information in the
> opposite direction.

Agreed.

(BTW, these are actually Table 1 and Table 2 -- we need to fix the
numbering.)

> Similar to the note in Table 6, this would require the SIP 
> implementation to support the 'jabber:client' namespace.  If the
> SIP implementation supports the namespace, the gateway can then map
> the values directly into the <show/> element of the XMPP presence
> messages.

True.

One clarifying question: when you say "if the SIP implementation
supports the namespace", do you mean the XMPP-to-SIP gateway or the
SIP user agent?

> The only other thing I saw was that the below sentence in Section
> 3.2.3 is missing the word "send" after SHOULD.
> 
> Current:
> 
> Upon sending the transformed unsubscribe, the XMPP-SIMPLE gateway
> SHOULD a presence stanza of type "unsubscribed" to the XMPP user:
> 
> New:
> 
> Upon sending the transformed unsubscribe, the XMPP-SIMPLE gateway
> SHOULD send a presence stanza of type "unsubscribed" to the XMPP
> user:

Noted. :-)

Peter

- -- 
Peter Saint-Andre
https://stpeter.im/


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From stpeter@stpeter.im  Wed Jul 24 11:09:38 2013
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Draft slides for discussion of the -core, -presence, -im, and -chat
specfications are here:

https://stpeter.im/files/ietf87-stox-core-presence.pdf

As you can see, there are very few open issues. If folks provide
further feedback about these specs in the next few days, I'll update
the slides accordingly. If not, mine will be a very short presentation
and we'll have more time to discuss the -groupchat and -media I-Ds. :-)

Peter

- -- 
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https://stpeter.im/


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Hi,

Here are draft slides for the groupchat and media documents: =
https://dl.dropboxusercontent.com/u/2243619/stox-ietf87.pdf

I didn't have time to process Philipp's feedback, I'll incorporate it =
ASAP. Also, I'll be AFK until sunday, but if further feedback is =
provided in the meantime I'll have the slides modified on monday.


Cheers,

--
Sa=FAl Ibarra Corretg=E9
AG Projects




From michaellundberg.ietf@gmail.com  Fri Jul 26 07:50:40 2013
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From: Michael Lundberg <michaellundberg.ietf@gmail.com>
To: Peter Saint-Andre <stpeter@stpeter.im>
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Subject: Re: [Stox] Comments on draft-ietf-stox-presence-00
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Peter,

Thanks.  See below for a response to your question.

Thanks,
Michael

On Tue, Jul 23, 2013 at 11:08 PM, Peter Saint-Andre <stpeter@stpeter.im> wrote:
>
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi Mike, thanks for the feedback.
>
> On 7/19/13 6:01 AM, Michael Lundberg wrote:
> > Peter,
> >
> > I think draft-ietf-stox-presence-00 looks good.  I had a previous
> > comment to an earlier version that was not addressed that I'm
> > incuding it again as I think it is important for the document.
>
> Sorry I missed that.
>
> > I think it would be good to add the <show/> element to Table 7,
> > with a note similar to the one provided in Table 6. This will
> > provide a method for mapping 'away' and 'dnd' information in the
> > opposite direction.
>
> Agreed.
>
> (BTW, these are actually Table 1 and Table 2 -- we need to fix the
> numbering.)
>
> > Similar to the note in Table 6, this would require the SIP
> > implementation to support the 'jabber:client' namespace.  If the
> > SIP implementation supports the namespace, the gateway can then map
> > the values directly into the <show/> element of the XMPP presence
> > messages.
>
> True.
>
> One clarifying question: when you say "if the SIP implementation
> supports the namespace", do you mean the XMPP-to-SIP gateway or the
> SIP user agent?

Good question.  It would be benificial if both supported the
namespace, but only the gateway is probably required to.  If the
client supports the namespace, then the gateway would just need to map
between the elements described in this document.

If the client doesn't support the namespace, the gateway would most
likely need to do an additional translation into a namespace the
client does understand.  In this case, the values might not be the
same between the two namespaces, and therefore things are 'lost' in
translation. This is one of the big issues with presence mapping today
as many implementations have thier own implementation specific
namespace, which makes it hard to map between different
implementations.  Both the implementation specific and common
namespaces could coexist, where the implementation specific namespace
is used for internal communication and a common, standard namespace
(e.g., 'jabber:client' ) is used when communicating between different
implementations.

> > The only other thing I saw was that the below sentence in Section
> > 3.2.3 is missing the word "send" after SHOULD.
> >
> > Current:
> >
> > Upon sending the transformed unsubscribe, the XMPP-SIMPLE gateway
> > SHOULD a presence stanza of type "unsubscribed" to the XMPP user:
> >
> > New:
> >
> > Upon sending the transformed unsubscribe, the XMPP-SIMPLE gateway
> > SHOULD send a presence stanza of type "unsubscribed" to the XMPP
> > user:
>
> Noted. :-)
>
> Peter
>
> - --
> Peter Saint-Andre
> https://stpeter.im/
>
>
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From stpeter@stpeter.im  Fri Jul 26 08:55:46 2013
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On 7/26/13 4:50 PM, Michael Lundberg wrote:

> One clarifying question: when you say "if the SIP implementation
> supports the namespace", do you mean the XMPP-to-SIP gateway or the
> SIP user agent?
> 
>> Good question.  It would be benificial if both supported the
>> namespace, but only the gateway is probably required to.  If the
>> client supports the namespace, then the gateway would just need to map
>> between the elements described in this document.
> 
>> If the client doesn't support the namespace, the gateway would most
>> likely need to do an additional translation into a namespace the
>> client does understand.  In this case, the values might not be the
>> same between the two namespaces, and therefore things are 'lost' in
>> translation. This is one of the big issues with presence mapping today
>> as many implementations have thier own implementation specific
>> namespace, which makes it hard to map between different
>> implementations.  Both the implementation specific and common
>> namespaces could coexist, where the implementation specific namespace
>> is used for internal communication and a common, standard namespace
>> (e.g., 'jabber:client' ) is used when communicating between different
>> implementations.

Yes, I think that makes sense. I'm not sure exactly how to translate
that into text (especially the part about proprietary / non-standard
namespaces), but I'll work something up for consideration by the list.

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



From fippo@goodadvice.pages.de  Sat Jul 27 13:14:53 2013
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> I didn't have time to process Philipp's feedback, I'll incorporate it ASAP. Also, I'll be AFK until sunday, but if further feedback is provided in the meantime I'll have the slides modified on monday.

Those were minor inconsistencies if I recall correctly. The examples in 
the draft and some text is based on an old version of the jingle specs.
We can sit down some time next week and sort this out.

From stpeter@stpeter.im  Sun Jul 28 03:20:37 2013
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Subject: Re: [Stox] Mapping for media signaling: still about the scope
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On 6/20/13 1:12 PM, Markus.Isomaki@nokia.com wrote:
> Hi Emil,

Hi Markus, at the XMPP hackfest yesterday I chatted about related issues
with Philipp Hancke, which triggered a few thoughts...

> See inline.
> 
> Emil Ivov [mailto:emcho@jitsi.org] wrote:
> 
>> On 19.06.13, 13:56, Markus.Isomaki@nokia.com wrote:
>>> Hi Emil, Saul, Peter, and others,
>>> 
>>> Recently I raised some questions about the scope of the
>>> 'sip-xmpp-media' and the related deliverable on the 'mapping for
>>> media signaling': 
>>> http://www.ietf.org/mail-archive/web/stox/current/msg00025.html
>>> 
>>> Now that STOX is officially up and running I'd like us to make
>>> an actual plan on what the scope of this document should be, i.e.
>>> what features are explicitly going to be part of this mapping
>>> spec. Although people have successfully implemented SIP-to-XMPP
>>> media signaling gateways, I at least see this topic to have some
>>> complexity due to:
>>> 
>>> -SDP offer/answer is a moving target as new features are being
>>> added as we speak,
>> 
>> Well ... it is and it isn't. 3264 isn't really changing and all
>> extensions are supposed to keep backward compatibility with it.
>> 
> 
> Right. That's good.
> 
>>> including "bundle" and "trickle-ICE". We can't know about what 
>>> features will be added later on.
>> 
>> No, but we could include some that we know of ... like "bundle" and
>> "trickle- ICE" :).
>> 
>> Trickle ICE would be easier because we already have it in Jingle
>> and the SIP version shouldn't take long.
>> 
> 
> That's one of the "administrative" decisions we should make about the
> "first release" of the media signaling mapping *RFC*. Do we only take
> in features/extensions that are completed RFCs or XSF XEPs, so that
> we for quite big certainty can get a mature document to IESG in
> October. And then extend it with new features once they complete,
> perhaps one or two years later, perhaps in multiple small steps. Or,
> do we already take features like "bundle" and "trickle-ICE" in, and
> aim to get the document to IESG at some later point, for instance
> March 2014. Trickle-ICE for SIP may not take long, but I bet it will
> at least take longer than this WG is supposed to live ;-)
> 
> I don't have an opinion about this as a co-chair. This is a question
> to people implementing and deploying the gateways or relying on them.
> What is the most useful initial scope for you? How important it is to
> have an RFC out this year vs. covering as many features as possible?
> I mean technically a draft might be as useful to an implementer as an
> RFC on those parts that are stable while others are developed, but
> would there be some
> interoperability/technical/political/marketing/getting-actually-something-completed
> reason to get an RFC out ASAP?
> 
> The only guidance we have at this point is that our charter expects
> us to complete in October, and in this case, unlike the normal IETF
> WG process, I at least as a co-chair intend to take that seriously
> :-)
> 
>>> -There are features currently supported in XMPP that are not
>>> currently supported in SIP (and perhaps vice versa?), such as the
>>> trickle-ICE support in Jingle.
>> 
>> As mentioned. I don't think trickle-ICE would be a problem. I
>> believe separating transport and encoding information would be a
>> bigger hurdle but if we accept that urn:ietf:rfc:3264 means one
>> shouldn't do this.
>> 
>>> -Regardless of what is supported in each protocol in general,
>>> the actual SIP and XMPP clients will support different sets of
>>> features.
>> 
>> Right.
>> 
>>> Ideally, some things might be mapped transparently in the gateway
>>> in an "algorithmic" manner.
>> 
>> I don't believe this would be possible.
>> 
> 
> Good, then my understanding seems to be correct.
> 
>>> And ideally, the clients supporting different feature sets can 
>>> negotiate end-to-end without needing for the gateway to mediate. 
>>> However, I'm not sure how easy this will be in practice.
>>> 
>>> Actually one reason why I see these complications may be that I
>>> don't have a very good overall understanding about how SDP O/A
>>> (as in RFC 3264 syntax and semantics) is mapped to XMPP/Jingle
>>> (as in its XML based syntax) are in general kept "in synch". For
>>> instance, if IETF extends SDP O/A in some way, will that always
>>> require an explicit corresponding extension XEP by the XSF, or is
>>> there some more straightforward "algorithmic" way to convey
>>> something new on the SIP/SDP side to the XMPP/Jingle side.
>> 
>> Well, the good thing about O/A extensions is that they generally
>> preserve backward compatibility. This is valid for things like ICE,
>> trickle ICE, rtcp-mux, bundle. So as long as we support 3264 we
>> should be OK. I think :)
>> 
> 
> Yes, let's stick to that theory ;-)
> 
> ...
> 
>> 
>>> -What will happen to features supported on one or either side of
>>> the gateway that the gateway is not aware of- can some things be 
>>> "transparently" mapped or will the gateway become the "blocking
>>> factor" in the middle. (I.e., will the session setup be limited
>>> to the features the gateway explicitly needs to "understand".)
>> 
>> I don't think the gateway could just forward anything because this
>> would imply a straightforward mapping from any SDP attribute to a
>> Jingle element.
>> 
>> Maybe gateways could behave as regular SIP B2BUAs: they forward
>> the things that they understand and they remove those that they
>> don't understand or support.
>> 
> 
> OK, that simplifies things as well. (Again was just checking I'm not
> missing any end-to-end magic that might exist. But apparently not
> ;-)
> 
>> Thoughts?
>> 
>>> -How will we extend the mapping spec in the future and how should
>>> the initial spec take this into account.
>> 
>> As in administratively?
>> 
> 
> In some sense yes. I would expect perhaps some brief section that
> gives guidelines to the authors who in the future want to extend this
> mapping spec with some new feature.
> 
> And would be rather do a full-fledged '-bis' spec at some point or a
> bunch or incremental extension drafts on top of the baseline. As an
> example "An extension to SIP-XMPP media signaling mapping to support
> trickle-ICE".

I like that approach. The Jingle XEPs already define SDP mappings for
the base use cases, and that content could be moved to the stox-media
I-D. If additional features are defined in SIP, SDP, or Jingle
(trickle-ICE, BUNDLE, multistream, etc.) then we can specify those in
incremental extensions.

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



From fippo@goodadvice.pages.de  Sun Jul 28 03:56:49 2013
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Subject: Re: [Stox] Mapping for media signaling: still about the scope
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Am 28.07.2013 12:20, schrieb Peter Saint-Andre:
[...]
> I like that approach. The Jingle XEPs already define SDP mappings for
> the base use cases, and that content could be moved to the stox-media
> I-D. If additional features are defined in SIP, SDP, or Jingle
> (trickle-ICE, BUNDLE, multistream, etc.) then we can specify those in
> incremental extensions.

Right. The problem on the XSF side is that this mapping is currently 
spread over (from the top of my head) XEPs 0166, 0167, 0176, (0177 
doesn't bother mapping), 0293, 0294 and 0320. This is something that 
needs to be more centralized anyway and the stox-media draft might be 
the normative place to do this.

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Thread-Topic: [Stox] Mapping for media signaling: still about the scope
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Subject: Re: [Stox] Mapping for media signaling: still about the scope
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Hi Peter, Philipp,

Philipp Hancke wrote,
>=20
> Am 28.07.2013 12:20, schrieb Peter Saint-Andre:
> [...]
> > I like that approach. The Jingle XEPs already define SDP mappings for
> > the base use cases, and that content could be moved to the stox-media
> > I-D. If additional features are defined in SIP, SDP, or Jingle
> > (trickle-ICE, BUNDLE, multistream, etc.) then we can specify those in
> > incremental extensions.
>=20
> Right. The problem on the XSF side is that this mapping is currently spre=
ad
> over (from the top of my head) XEPs 0166, 0167, 0176, (0177 doesn't bothe=
r
> mapping), 0293, 0294 and 0320. This is something that needs to be more
> centralized anyway and the stox-media draft might be the normative place =
to
> do this.
>

It sounds to me then that the STOX WG should:
- Consolidate the mappings for all stable specs and features into the curre=
nt -media document and aim to publish it as an RFC according to the current=
 charter, i.e. deliver it to the IESG (hopefully) already before IETF 88. T=
he value of stabilizing and publishing that document would be that the mapp=
ings would be found in one place.
- Start, first as individual (non-WG) drafts, incremental extensions for th=
e relevant new features (mentioned above). Perhaps one draft is enough for =
all of the listed ones if their timeline is roughly identical and there is =
interest to cover all of them. That draft would evolve as the actual extens=
ions in IETF and XSF develop, and would get published as an RFC around the =
same time as them (next year?). If things progress well, we could adopt tha=
t draft as a STOX WG item around IETF 88. I believe it does fit content-wis=
e ("mapping for media signaling") to our charter, we just need to create an=
 additional milestone for it. I don't know if we should keep the WG active =
for just one document, but given the overhead of creating new WGs etc., I t=
hink that would make sense.

Saul's slides for next Thursday (https://dl.dropboxusercontent.com/u/224361=
9/stox-ietf87.pdf) seem to assume that approach. The chairs can perhaps cre=
ate one-slider to be more explicit about the plan and scope.=20

Markus



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Subject: Re: [Stox] Mapping for media signaling: still about the scope
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On 7/28/13 4:15 PM, Markus.Isomaki@nokia.com wrote:
> Hi Peter, Philipp,
> 
> Philipp Hancke wrote,
>> 
>> Am 28.07.2013 12:20, schrieb Peter Saint-Andre: [...]
>>> I like that approach. The Jingle XEPs already define SDP mappings
>>> for the base use cases, and that content could be moved to the
>>> stox-media I-D. If additional features are defined in SIP, SDP,
>>> or Jingle (trickle-ICE, BUNDLE, multistream, etc.) then we can
>>> specify those in incremental extensions.
>> 
>> Right. The problem on the XSF side is that this mapping is
>> currently spread over (from the top of my head) XEPs 0166, 0167,
>> 0176, (0177 doesn't bother mapping), 0293, 0294 and 0320. This is
>> something that needs to be more centralized anyway and the
>> stox-media draft might be the normative place to do this.
>> 
> 
> It sounds to me then that the STOX WG should: - Consolidate the
> mappings for all stable specs and features into the current -media
> document and aim to publish it as an RFC according to the current
> charter, i.e. deliver it to the IESG (hopefully) already before IETF
> 88. The value of stabilizing and publishing that document would be
> that the mappings would be found in one place. - Start, first as
> individual (non-WG) drafts, incremental extensions for the relevant
> new features (mentioned above). Perhaps one draft is enough for all
> of the listed ones if their timeline is roughly identical and there
> is interest to cover all of them. That draft would evolve as the
> actual extensions in IETF and XSF develop, and would get published as
> an RFC around the same time as them (next year?). 

I think that is a reasonable approach.

> If things progress
> well, we could adopt that draft as a STOX WG item around IETF 88. I
> believe it does fit content-wise ("mapping for media signaling") to
> our charter, we just need to create an additional milestone for it. I
> don't know if we should keep the WG active for just one document, but
> given the overhead of creating new WGs etc., I think that would make
> sense.

I am not fully sure about keeping the WG around, but I leave such
decisions up to the chairs and ADs. ;-)

> Saul's slides for next Thursday
> (https://dl.dropboxusercontent.com/u/2243619/stox-ietf87.pdf) seem to
> assume that approach. The chairs can perhaps create one-slider to be
> more explicit about the plan and scope.

That sounds helpful!

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

On 7/27/13 10:14 PM, Philipp Hancke wrote:
>> I didn't have time to process Philipp's feedback, I'll
>> incorporate it ASAP. Also, I'll be AFK until sunday, but if
>> further feedback is provided in the meantime I'll have the slides
>> modified on monday.
> 
> Those were minor inconsistencies if I recall correctly. The
> examples in the draft and some text is based on an old version of
> the jingle specs. We can sit down some time next week and sort this
> out.

That sounds productive.

Peter

- -- 
Peter Saint-Andre
https://stpeter.im/


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Subject: [Stox] STOX @ IETF 87 Note takers
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Hi all,

We are looking for note takers and a jabber scribe for the STOX session =
on Thursday in Berlin (http://tools.ietf.org/agenda/87/#THURSDAY).

If you are attending and would be able to take notes, please contact the =
chairs.


Thanks,
Yana & Markus=

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Subject: Re: [Stox] Mapping for media signaling: still about the scope
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On Jul 28, 2013, at 4:15 PM, <Markus.Isomaki@nokia.com> =
<Markus.Isomaki@nokia.com> wrote:

> Hi Peter, Philipp,
>=20
> Philipp Hancke wrote,
>>=20
>> Am 28.07.2013 12:20, schrieb Peter Saint-Andre:
>> [...]
>>> I like that approach. The Jingle XEPs already define SDP mappings =
for
>>> the base use cases, and that content could be moved to the =
stox-media
>>> I-D. If additional features are defined in SIP, SDP, or Jingle
>>> (trickle-ICE, BUNDLE, multistream, etc.) then we can specify those =
in
>>> incremental extensions.
>>=20
>> Right. The problem on the XSF side is that this mapping is currently =
spread
>> over (from the top of my head) XEPs 0166, 0167, 0176, (0177 doesn't =
bother
>> mapping), 0293, 0294 and 0320. This is something that needs to be =
more
>> centralized anyway and the stox-media draft might be the normative =
place to
>> do this.
>>=20
>=20
> It sounds to me then that the STOX WG should:
> - Consolidate the mappings for all stable specs and features into the =
current -media document and aim to publish it as an RFC according to the =
current charter, i.e. deliver it to the IESG (hopefully) already before =
IETF 88. The value of stabilizing and publishing that document would be =
that the mappings would be found in one place.
> - Start, first as individual (non-WG) drafts, incremental extensions =
for the relevant new features (mentioned above). Perhaps one draft is =
enough for all of the listed ones if their timeline is roughly identical =
and there is interest to cover all of them. That draft would evolve as =
the actual extensions in IETF and XSF develop, and would get published =
as an RFC around the same time as them (next year?). If things progress =
well, we could adopt that draft as a STOX WG item around IETF 88. I =
believe it does fit content-wise ("mapping for media signaling") to our =
charter, we just need to create an additional milestone for it. I don't =
know if we should keep the WG active for just one document, but given =
the overhead of creating new WGs etc., I think that would make sense.
>=20

That sounds good to me!

> Saul's slides for next Thursday =
(https://dl.dropboxusercontent.com/u/2243619/stox-ietf87.pdf) seem to =
assume that approach. The chairs can perhaps create one-slider to be =
more explicit about the plan and scope.=20
>=20

Yeah, I somewhat assumed that,but it's definitely up for discussion. =
Since I'm not all that familiar with all the process, please do let me =
know if you rather have some extra items added to the slides and I'll =
gladly do so.


Cheers,

--
Sa=FAl Ibarra Corretg=E9
AG Projects




From stpeter@stpeter.im  Mon Jul 29 03:49:47 2013
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On 7/26/13 5:55 PM, Peter Saint-Andre wrote:
> On 7/26/13 4:50 PM, Michael Lundberg wrote:
> 
>> One clarifying question: when you say "if the SIP implementation
>> supports the namespace", do you mean the XMPP-to-SIP gateway or the
>> SIP user agent?
>>
>>> Good question.  It would be benificial if both supported the
>>> namespace, but only the gateway is probably required to.  If the
>>> client supports the namespace, then the gateway would just need to map
>>> between the elements described in this document.
>>
>>> If the client doesn't support the namespace, the gateway would most
>>> likely need to do an additional translation into a namespace the
>>> client does understand.  In this case, the values might not be the
>>> same between the two namespaces, and therefore things are 'lost' in
>>> translation. This is one of the big issues with presence mapping today
>>> as many implementations have thier own implementation specific
>>> namespace, which makes it hard to map between different
>>> implementations.  Both the implementation specific and common
>>> namespaces could coexist, where the implementation specific namespace
>>> is used for internal communication and a common, standard namespace
>>> (e.g., 'jabber:client' ) is used when communicating between different
>>> implementations.
> 
> Yes, I think that makes sense. I'm not sure exactly how to translate
> that into text (especially the part about proprietary / non-standard
> namespaces), but I'll work something up for consideration by the list.

Here is some proposed text:

OLD
   5.  Some implementations support custom extensions to encapsulate
       this information; however, there is no need to standardize a PIDF
       extension for this purpose, since PIDF is already extensible and
       thus the <show/> element can be included directly, qualified by
       the 'jabber:client' namespace in the PIDF XML.  The examples in
       this document illustrate this usage, which is RECOMMENDED.  The
       most useful values are likely "away" and "dnd", although note
       that the latter value merely means "busy" and does not imply that
       a server or client ought to block incoming traffic while the user
       is in that state.

NEW
   5.  Some implementations support custom extensions to encapsulate
       detailed information about availability; however, there is no
       need to standardize a PIDF extension for this purpose, since
       PIDF is already extensible and thus the <show/> element
       (qualified by the 'jabber:client' namespace) can be included
       directly in the PIDF XML.  The examples in this document
       illustrate this usage, which is RECOMMENDED.  The most useful
       values are likely "away" and "dnd", although note that the
       latter value merely means "busy" and does not imply that a
       server or client ought to block incoming traffic while the user
       is in that state.  Naturally, a gateway can choose to translate
       a custom extension into an established value of the <show/>
       element [RFC6121], or translate a <show/> element into a custom
       extension that the gateway knows is supported by the user agent
       of the intended recipient.  Unfortunately, this behavior does
       not guarantee that information will not be lost; to help prevent
       information loss, a gateway ought to include both the <show/>
       element and the custom extension if the gateway cannot suitably
       translate the custom value into a <show/> value.

Mike, does that text address your concern? Do we need to say more (or
less) than that?

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



From stpeter@stpeter.im  Tue Jul 30 05:15:11 2013
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After a hallway discussion with Robert Sparks, I have updated these
slides. I will post to the list about the issue he raised.

On 7/24/13 8:09 PM, Peter Saint-Andre wrote:
> Draft slides for discussion of the -core, -presence, -im, and -chat
> specfications are here:
> 
> https://stpeter.im/files/ietf87-stox-core-presence.pdf
> 
> As you can see, there are very few open issues. If folks provide
> further feedback about these specs in the next few days, I'll update
> the slides accordingly. If not, mine will be a very short presentation
> and we'll have more time to discuss the -groupchat and -media I-Ds. :-)
> 
> Peter
> 
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox
> 

From stpeter@stpeter.im  Tue Jul 30 05:24:33 2013
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Robert Sparks pointed out to me in person today that the usage of SIP
error 503 is broken in stox-core.

RFC 3261 says:

21.5.4 503 Service Unavailable

   The server is temporarily unable to process the request due to a
   temporary overloading or maintenance of the server.  The server MAY
   indicate when the client should retry the request in a Retry-After
   header field.  If no Retry-After is given, the client MUST act as if
   it had received a 500 (Server Internal Error) response.

   A client (proxy or UAC) receiving a 503 (Service Unavailable) SHOULD
   attempt to forward the request to an alternate server.  It SHOULD NOT
   forward any other requests to that server for the duration specified
   in the Retry-After header field, if present.

   Servers MAY refuse the connection or drop the request instead of
   responding with 503 (Service Unavailable).

Thus the applicability of 503 in SIP is quite different from that of
<service-unavailable/> in XMPP. In XMPP, a client (or, more often, a
server on its behalf) can return <service-unavailable/> to mean
"sorry, I can't provide the service you've requested for the user@host
you've tried to contact, but that applies only to that user@host". If
the XMPP-to-SIP gateway returns <service-unavailable/> for a single
account in that way, the SIP-to-XMPP gateway on the other side won't
be able to send *anything* to the peer domain. This breaks all
server-to-server communication. Thus Robert suggested that we take a
very close look at our mapping into SIP 503, and very probably never
use it (or only in very rare cases). I haven't had time yet to look at
this in detail, but I will do so before the STOX session on Thursday.

Peter

- -- 
Peter Saint-Andre
https://stpeter.im/


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From yana@sip-communicator.org  Tue Jul 30 05:59:27 2013
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Subject: Re: [Stox] draft slides for core, presence, im, chat
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Hi Peter,

Thanks! Your new slides are uploaded.

Yana

On Jul 30, 2013, at 2:15 PM, Peter Saint-Andre <stpeter@stpeter.im> wrote:

> After a hallway discussion with Robert Sparks, I have updated these
> slides. I will post to the list about the issue he raised.
> 
> On 7/24/13 8:09 PM, Peter Saint-Andre wrote:
>> Draft slides for discussion of the -core, -presence, -im, and -chat
>> specfications are here:
>> 
>> https://stpeter.im/files/ietf87-stox-core-presence.pdf
>> 
>> As you can see, there are very few open issues. If folks provide
>> further feedback about these specs in the next few days, I'll update
>> the slides accordingly. If not, mine will be a very short presentation
>> and we'll have more time to discuss the -groupchat and -media I-Ds. :-)
>> 
>> Peter
>> 
>> _______________________________________________
>> stox mailing list
>> stox@ietf.org
>> https://www.ietf.org/mailman/listinfo/stox
>> 
> _______________________________________________
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> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox


From saul@ag-projects.com  Tue Jul 30 06:41:54 2013
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Hi,

As promised, here is my review of the -core document:

- Sec 4.3 defines instant specific mapping, which maps a public GRUU to =
a full JID and vice versa, but doesn't contain any text on temporary =
GRUUs. They take the following form: =
sip:yyd894uue094ur4@1.2.3.4:5060;gr. I don't think we can work with =
those, since there is nothing we can map to the resource part of the =
full JID, so I think we need some text to clarify this.

- Sec 5.1 maps the XMPP unexpected-request error to SIP code 491, which =
is very tied to the dialog semantics IMHO. I tried to find a better =
matching code but I couldn't though. Maybe a 500 with a reasonable =
Retry-After header value?

- Sec 5.1 should me map the SIP outbound (RFC5626) error codes? Those =
are 430, and 439 which could be translated to bad-request.


* Editorial nits:

- Sec 3: "This assumption not meant" -> "This assumption is not meant"


Cheers,

--
Sa=FAl Ibarra Corretg=E9
AG Projects




From saul@ag-projects.com  Tue Jul 30 07:02:00 2013
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Hi,

Here is my review of the -chat document.


- What should a gateway do when it gets a message with XHTML-IM body? I =
guess send it with content-type text/html, but is there any translation =
required?

- Section 4 mentions what should happen if no formal sessions are used, =
but doesn't say anything of the opposite. Should we say that those are =
out of the scope of this specification?

- I think we should define a mapping between XMPP chatstates (XEP-0085) =
and the iscomosing indication (RFC 3994) particularly, the "gone" state =
in XMPP should trigger a BYE on the SIP side, and vice versa.

- Should we add text to map MSRP reports to XMPP receipts (XEP-0184)? =
They do map very well semantically. Now, XEP-0184 says that the ack may =
not be sent if "The recipient simply does not wish to return a receipt =
for the content message.", which pretty much renders it useless. Anyway, =
I think we do need some text explaining how MSRP success and failure =
ports should be dealt with.

- In general: we don't have any example using CPIM, not sure if we =
should say anything about it.


* Editorial nits

- Examples (multiple): the gr parameter is a URI parameter, not a header =
parameter, so: "To: <sip:juliet@example.com>;gr=3Dbalcony" should be =
"To: <sip:juliet@example.com;gr=3Dbalcony>". This applies to both To and =
Contact headers.

- Sec 1: "using the SIP INVITE and SEND request" -> "using the SIP =
INVITE and MSRP SEND request"


Cheers,

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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One more thing: the -chat document doesn't have the detailed DNS lookup =
mechanism description that the IM document does, I guess it should also =
contain it, right?


On Jul 30, 2013, at 4:01 PM, Sa=FAl Ibarra Corretg=E9 =
<saul@ag-projects.com> wrote:

> Hi,
>=20
> Here is my review of the -chat document.
>=20
>=20
> - What should a gateway do when it gets a message with XHTML-IM body? =
I guess send it with content-type text/html, but is there any =
translation required?
>=20
> - Section 4 mentions what should happen if no formal sessions are =
used, but doesn't say anything of the opposite. Should we say that those =
are out of the scope of this specification?
>=20
> - I think we should define a mapping between XMPP chatstates =
(XEP-0085) and the iscomosing indication (RFC 3994) particularly, the =
"gone" state in XMPP should trigger a BYE on the SIP side, and vice =
versa.
>=20
> - Should we add text to map MSRP reports to XMPP receipts (XEP-0184)? =
They do map very well semantically. Now, XEP-0184 says that the ack may =
not be sent if "The recipient simply does not wish to return a receipt =
for the content message.", which pretty much renders it useless. Anyway, =
I think we do need some text explaining how MSRP success and failure =
ports should be dealt with.
>=20
> - In general: we don't have any example using CPIM, not sure if we =
should say anything about it.
>=20
>=20
> * Editorial nits
>=20
> - Examples (multiple): the gr parameter is a URI parameter, not a =
header parameter, so: "To: <sip:juliet@example.com>;gr=3Dbalcony" should =
be "To: <sip:juliet@example.com;gr=3Dbalcony>". This applies to both To =
and Contact headers.
>=20
> - Sec 1: "using the SIP INVITE and SEND request" -> "using the SIP =
INVITE and MSRP SEND request"
>=20
>=20
> Cheers,
>=20
> --
> Sa=FAl Ibarra Corretg=E9
> AG Projects
>=20
>=20
>=20
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox

--
Sa=FAl Ibarra Corretg=E9
AG Projects




From salvatore.loreto@ericsson.com  Tue Jul 30 07:09:51 2013
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From: Salvatore Loreto <salvatore.loreto@ericsson.com>
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Subject: Re: [Stox] Review of -chat
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On 7/30/13 4:06 PM, Saúl Ibarra Corretgé wrote:
> One more thing: the -chat document doesn't have the detailed DNS lookup mechanism description that the IM document does, I guess it should also contain it, right?
I would prefer not to duplicate the same text among the different documents
unless there is not some specific DNS lookup detail that is specific for 
chat. Is this the case?

we can maybe think if the right place for the DNS lookup mechanism is IM 
document or we want to have it
in the Core draft

/Salvatore

>
>
> On Jul 30, 2013, at 4:01 PM, Saúl Ibarra Corretgé <saul@ag-projects.com> wrote:
>
>> Hi,
>>
>> Here is my review of the -chat document.
>>
>>
>> - What should a gateway do when it gets a message with XHTML-IM body? I guess send it with content-type text/html, but is there any translation required?
>>
>> - Section 4 mentions what should happen if no formal sessions are used, but doesn't say anything of the opposite. Should we say that those are out of the scope of this specification?
>>
>> - I think we should define a mapping between XMPP chatstates (XEP-0085) and the iscomosing indication (RFC 3994) particularly, the "gone" state in XMPP should trigger a BYE on the SIP side, and vice versa.
>>
>> - Should we add text to map MSRP reports to XMPP receipts (XEP-0184)? They do map very well semantically. Now, XEP-0184 says that the ack may not be sent if "The recipient simply does not wish to return a receipt for the content message.", which pretty much renders it useless. Anyway, I think we do need some text explaining how MSRP success and failure ports should be dealt with.
>>
>> - In general: we don't have any example using CPIM, not sure if we should say anything about it.
>>
>>
>> * Editorial nits
>>
>> - Examples (multiple): the gr parameter is a URI parameter, not a header parameter, so: "To: <sip:juliet@example.com>;gr=balcony" should be "To: <sip:juliet@example.com;gr=balcony>". This applies to both To and Contact headers.
>>
>> - Sec 1: "using the SIP INVITE and SEND request" -> "using the SIP INVITE and MSRP SEND request"
>>
>>
>> Cheers,
>>
>> --
>> Saúl Ibarra Corretgé
>> AG Projects
>>
>>
>>
>> _______________________________________________
>> stox mailing list
>> stox@ietf.org
>> https://www.ietf.org/mailman/listinfo/stox
> --
> Saúl Ibarra Corretgé
> AG Projects
>
>
>
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox


-- 
Salvatore Loreto, PhD
www.sloreto.com


From saul@ag-projects.com  Tue Jul 30 07:21:08 2013
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Hi,

Here is my review on the -im document:

- Section 3 suggests that the thread element in XMPP should be mapped to =
the Call-ID header. I see why this was done, but it looks a bit like =
abusing it. I would probably take it out or define it as a CPIM header, =
in case CPIM is used. Thoughts?

- Section 5 only mentions text/plain and text/html, I think we should =
add message/cpim in there.

- Similarly to chat, I think we should define a mapping between XMPP =
chatstates (XEP-0085) and the iscomosing indication (RFC 3994), but in =
this case there would be no mapping for the 'gone' state.


* Editorial nits:

- In the examples: From, To and Contact header URIs are not enclosed in =
angle brackets
- In the example in section 4, the RURI should have the gr parameter as =
well,as the To header is not used for routing


Cheers,

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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On 7/30/13 4:09 PM, Salvatore Loreto wrote:
> On 7/30/13 4:06 PM, Saúl Ibarra Corretgé wrote:
>> One more thing: the -chat document doesn't have the detailed DNS
>> lookup mechanism description that the IM document does, I guess it
>> should also contain it, right?
> I would prefer not to duplicate the same text among the different documents
> unless there is not some specific DNS lookup detail that is specific for
> chat. Is this the case?
> 
> we can maybe think if the right place for the DNS lookup mechanism is IM
> document or we want to have it
> in the Core draft

As I recall, we already discussed the topic, starting here:

http://www.ietf.org/mail-archive/web/stox/current/msg00069.html

I think we *might* be able to provide some informational guidance, but I
don't know how strong it can be.

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



From stpeter@stpeter.im  Tue Jul 30 08:06:44 2013
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On 7/30/13 5:03 PM, Peter Saint-Andre wrote:
> On 7/30/13 4:09 PM, Salvatore Loreto wrote:
>> On 7/30/13 4:06 PM, Saúl Ibarra Corretgé wrote:
>>> One more thing: the -chat document doesn't have the detailed DNS
>>> lookup mechanism description that the IM document does, I guess it
>>> should also contain it, right?
>> I would prefer not to duplicate the same text among the different documents
>> unless there is not some specific DNS lookup detail that is specific for
>> chat. Is this the case?
>>
>> we can maybe think if the right place for the DNS lookup mechanism is IM
>> document or we want to have it
>> in the Core draft
> 
> As I recall, we already discussed the topic, starting here:
> 
> http://www.ietf.org/mail-archive/web/stox/current/msg00069.html
> 
> I think we *might* be able to provide some informational guidance, but I
> don't know how strong it can be.

But, if we do have text, I think it belongs in stox-core.

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



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Subject: [Stox] Review on -presence
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Hi,

Here is my review of the -presence document:

- Sec 3.3.2 suggests that if the subscription is maintained but we have =
no presence state, a PIDF would be generated with a basic status of =
closed, but what would be the tuple ID, if we don't know any resource =
for this user anymore? We could also send an empty NOTIFY, which would =
achieve the same goal.

- Sec 3.3.2 specifies how to handle the case when a gateway translated a =
SIP subscription into an XMPP subscription and then the SIP side decided =
to end it, but doesn't mention what to do in case the XMPP side decides =
to end it. It should send a NOTIFY with Subscription-State: =
terminated;reason=3Drejected.

- Sec 4.1, "Instant Message URI of the form <pres:user@domain>" - =
shouldn't this be "Presence URI of the form=85" ?

- Using ID-123kdklejd doesn't seem to work as a valid xs:ID, TID-1234 =
does work though, so we could use TID- as the prefix for tuple =
identifiers in examples and such.

- Table 6 suggests that the Call-ID header should be matched to the =
presence stanza id, but the called won't change throughout the whole =
dialog, so we should use something else, which changes for each SIP =
transaction, what about the Via branch? Table 7 uses CSeq, FWIW.

- Table 6 suggests that priority is translated as the PIDF priority for =
a tuple, but that's an attribute of the contact element, which we don't =
mention, but I guess we should. Also, the priority in PIDF is a float =
between 0 and 1, is it the same for XMPP?

- Sec 4.3, the translated XMPP stanza contains the full JID of Juliet, =
but where did we get the resource from? Shouldn't it go to the bare JID?


* Editorial nits:

- Many example requests lack a To header
- Sec 3.2.3: "gateway SHOULD a presence stanza" -> "gateway SHOULD send =
a presence stanza"


Cheers,

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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One more thing: IMHO we should also mention that "headline" XMPP =
messages should be translated like the normal ones in this spec. Not in =
-chat though.

On Jul 30, 2013, at 4:19 PM, Sa=FAl Ibarra Corretg=E9 =
<saul@ag-projects.com> wrote:

> Hi,
>=20
> Here is my review on the -im document:
>=20
> - Section 3 suggests that the thread element in XMPP should be mapped =
to the Call-ID header. I see why this was done, but it looks a bit like =
abusing it. I would probably take it out or define it as a CPIM header, =
in case CPIM is used. Thoughts?
>=20
> - Section 5 only mentions text/plain and text/html, I think we should =
add message/cpim in there.
>=20
> - Similarly to chat, I think we should define a mapping between XMPP =
chatstates (XEP-0085) and the iscomosing indication (RFC 3994), but in =
this case there would be no mapping for the 'gone' state.
>=20
>=20
> * Editorial nits:
>=20
> - In the examples: From, To and Contact header URIs are not enclosed =
in angle brackets
> - In the example in section 4, the RURI should have the gr parameter =
as well,as the To header is not used for routing
>=20
>=20
> Cheers,
>=20
> --
> Sa=FAl Ibarra Corretg=E9
> AG Projects
>=20
>=20
>=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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On 7/30/13 5:54 PM, Saúl Ibarra Corretgé wrote:
> One more thing: IMHO we should also mention that "headline" XMPP 
> messages should be translated like the normal ones in this spec.
> Not in -chat though.

Agreed.

Peter

- -- 
Peter Saint-Andre
https://stpeter.im/


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This is just a warning that I'll be providing updated slides to address
some points that Saúl made in his reviews. I'll try to do that as early
today as possible.

On 7/30/13 2:59 PM, Yana Stamcheva wrote:
> Hi Peter,
> 
> Thanks! Your new slides are uploaded.
> 
> Yana
> 
> On Jul 30, 2013, at 2:15 PM, Peter Saint-Andre <stpeter@stpeter.im> wrote:
> 
>> After a hallway discussion with Robert Sparks, I have updated these
>> slides. I will post to the list about the issue he raised.
>>
>> On 7/24/13 8:09 PM, Peter Saint-Andre wrote:
>>> Draft slides for discussion of the -core, -presence, -im, and -chat
>>> specfications are here:
>>>
>>> https://stpeter.im/files/ietf87-stox-core-presence.pdf
>>>
>>> As you can see, there are very few open issues. If folks provide
>>> further feedback about these specs in the next few days, I'll update
>>> the slides accordingly. If not, mine will be a very short presentation
>>> and we'll have more time to discuss the -groupchat and -media I-Ds. :-)
>>>
>>> Peter
>>>

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Hi,

I just updated my slides after I reviewed the group chat draft and =
incorporated some feedback I got from discussions with Phillip. I don't =
expect to change them again unless there is a world-ending-problem that =
would need to be addressed :-)



On Jul 26, 2013, at 3:43 PM, Sa=FAl Ibarra Corretg=E9 =
<saul@ag-projects.com> wrote:

> Hi,
>=20
> Here are draft slides for the groupchat and media documents: =
https://dl.dropboxusercontent.com/u/2243619/stox-ietf87.pdf
>=20
> I didn't have time to process Philipp's feedback, I'll incorporate it =
ASAP. Also, I'll be AFK until sunday, but if further feedback is =
provided in the meantime I'll have the slides modified on monday.
>=20
>=20
> Cheers,
>=20
> --
> Sa=FAl Ibarra Corretg=E9
> AG Projects
>=20
>=20
>=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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On 7/30/13 3:41 PM, Saúl Ibarra Corretgé wrote:
> Hi,
> 
> As promised, here is my review of the -core document:

Many thanks for the review.

> - Sec 4.3 defines instant specific mapping, which maps a public GRUU
> to a full JID and vice versa, but doesn't contain any text on
> temporary GRUUs. They take the following form:
> sip:yyd894uue094ur4@1.2.3.4:5060;gr. I don't think we can work with
> those, since there is nothing we can map to the resource part of the
> full JID, so I think we need some text to clarify this.

Agreed on temporary GRUUs. IMHO we'll need to add a note about that (but
I haven't yet had time to propose text).

> - Sec 5.1 maps the XMPP unexpected-request error to SIP code 491,
> which is very tied to the dialog semantics IMHO. I tried to find a
> better matching code but I couldn't though. Maybe a 500 with a
> reasonable Retry-After header value?

It seems to me that we'd want it to be a 4xx series error, but I don't
see a good one in RFC 3261.

> - Sec 5.1 should me map the SIP outbound (RFC5626) error codes? Those
> are 430, and 439 which could be translated to bad-request.

Could you explain the scenarios where outbound-related error codes might
be used? Feel free to say "just read RFC 5626". :-)

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



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On 7/30/13 4:01 PM, Saúl Ibarra Corretgé wrote:
> Hi,
> 
> Here is my review of the -chat document.
> 
> - What should a gateway do when it gets a message with XHTML-IM body?
> I guess send it with content-type text/html, but is there any
> translation required?

I know about the use of XHTML in XMPP, but I don't know much about the
use in SIP. Is there a profile of (X)HTML that most SIP entities implement?

> - Section 4 mentions what should happen if no formal sessions are
> used, but doesn't say anything of the opposite. Should we say that
> those are out of the scope of this specification?

Precursors to this document talked about formal chat sessions in XMPP
(XEP-0155), however no clients support that model so I remove all of
that text. The first paragraph of Section 4 reflects the earlier specs.
I think we just need to remove a few words from Section 4...

OLD
   When an MSRP client sends messages through a gateway to an XMPP
   client that does not support formal sessinos, the order of events is
   as follows.

NEW
   When an MSRP client sends messages through a gateway to an XMPP
   client, the order of events is as follows.

> - I think we should define a mapping between XMPP chatstates
> (XEP-0085) and the iscomosing indication (RFC 3994) particularly, the
> "gone" state in XMPP should trigger a BYE on the SIP side, and vice
> versa.

Could you describe a bit why you think that would be useful? (You
explained it to me in person, but I think it would be helpful to discuss
it on the list so that it's in the records of the WG. And for the record
I am pretty sure I agree with you.)

> - Should we add text to map MSRP reports to XMPP receipts (XEP-0184)?
> They do map very well semantically. Now, XEP-0184 says that the ack
> may not be sent if "The recipient simply does not wish to return a
> receipt for the content message.", which pretty much renders it
> useless. Anyway, I think we do need some text explaining how MSRP
> success and failure ports should be dealt with.

I wonder if this belongs in the stox-chat spec or in an extension document.

(And BTW XMPP message receipts can be used outside of chat sessions, so
I think it applies more broadly than the stox-chat spec.)

> - In general: we don't have any example using CPIM, not sure if we
> should say anything about it.

What do you have in mind for CPIM examples? Do you mean examples other
than MSRP, or examples with CPIM payloads?

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



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>=20
>=20
>> - Sec 5.1 maps the XMPP unexpected-request error to SIP code 491,
>> which is very tied to the dialog semantics IMHO. I tried to find a
>> better matching code but I couldn't though. Maybe a 500 with a
>> reasonable Retry-After header value?
>=20
> It seems to me that we'd want it to be a 4xx series error, but I don't
> see a good one in RFC 3261.

Exactly my thought.

>=20
>> - Sec 5.1 should me map the SIP outbound (RFC5626) error codes? Those
>> are 430, and 439 which could be translated to bad-request.
>=20
> Could you explain the scenarios where outbound-related error codes =
might
> be used? Feel free to say "just read RFC 5626". :-)
>=20

A 430 may occur if the gateway sends a SIP INVITE to the target domain =
and the edge proxy which handles the INVITE gets a flow error from the =
authoritative proxy behind him. Forget about 439, that's for REGISTERs, =
sorry.

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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On 7/30/13 4:19 PM, Saúl Ibarra Corretgé wrote:
> Hi,
> 
> Here is my review on the -im document:
> 
> - Section 3 suggests that the thread element in XMPP should be mapped
> to the Call-ID header. I see why this was done, but it looks a bit
> like abusing it. 

What are your specific concerns? Would something break? Is it insecure?
Does it horribly violate the semantics defined in RFC 3261? I thought it
was a pretty good match:

   The Call-ID header field acts as a unique identifier to group
   together a series of messages.

But maybe I don't understand the "oral tradition" of how Call-ID has
actually be used.

> I would probably take it out or define it as a CPIM
> header, in case CPIM is used. Thoughts?

I don't see a header in RFC 3862 that we could map to. Our charter says
we're not allowed to define new protocols. Someone could define a new
CPIM header, but I don't think that would be widely adopted anytime soon.

> - Section 5 only mentions text/plain and text/html, I think we should
> add message/cpim in there.

No objections here.

> - Similarly to chat, I think we should define a mapping between XMPP
> chatstates (XEP-0085) and the iscomosing indication (RFC 3994), but
> in this case there would be no mapping for the 'gone' state.

The stox-im spec is only for single messages, so I don't see how chat
states apply.

(This document might be the place to talk about message receipts, however.)

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



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31 jul 2013 kl. 16:42 skrev Sa=FAl Ibarra Corretg=E9 =
<saul@ag-projects.com>:

>>=20
>>=20
>>> - Sec 5.1 maps the XMPP unexpected-request error to SIP code 491,
>>> which is very tied to the dialog semantics IMHO. I tried to find a
>>> better matching code but I couldn't though. Maybe a 500 with a
>>> reasonable Retry-After header value?
>>=20
>> It seems to me that we'd want it to be a 4xx series error, but I =
don't
>> see a good one in RFC 3261.
>=20
> Exactly my thought.
"unexpected-request error" seems more like a server error - or does it =
point to the actual request?

/O
>=20
>>=20
>>> - Sec 5.1 should me map the SIP outbound (RFC5626) error codes? =
Those
>>> are 430, and 439 which could be translated to bad-request.
>>=20
>> Could you explain the scenarios where outbound-related error codes =
might
>> be used? Feel free to say "just read RFC 5626". :-)
>>=20
>=20
> A 430 may occur if the gateway sends a SIP INVITE to the target domain =
and the edge proxy which handles the INVITE gets a flow error from the =
authoritative proxy behind him. Forget about 439, that's for REGISTERs, =
sorry.
>=20
> --
> Sa=FAl Ibarra Corretg=E9
> AG Projects
>=20
>=20
>=20
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox

---
* Olle E Johansson - oej@edvina.net
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden




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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Diso-8859-1"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><br><div><div>31 jul 2013 kl. 16:42 skrev Sa=FAl Ibarra Corretg=E9 =
&lt;<a =
href=3D"mailto:saul@ag-projects.com">saul@ag-projects.com</a>&gt;:</div><b=
r class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><blockquote type=3D"cite"><br><br><blockquote =
type=3D"cite">- Sec 5.1 maps the XMPP unexpected-request error to SIP =
code 491,<br>which is very tied to the dialog semantics IMHO. I tried to =
find a<br>better matching code but I couldn't though. Maybe a 500 with =
a<br>reasonable Retry-After header value?<br></blockquote><br>It seems =
to me that we'd want it to be a 4xx series error, but I don't<br>see a =
good one in RFC 3261.<br></blockquote><br>Exactly my =
thought.<br></blockquote>"unexpected-request error" seems more like a =
server error - or does it point to the actual =
request?</div><div><br></div><div>/O<br><blockquote =
type=3D"cite"><br><blockquote type=3D"cite"><br><blockquote =
type=3D"cite">- Sec 5.1 should me map the SIP outbound (RFC5626) error =
codes? Those<br>are 430, and 439 which could be translated to =
bad-request.<br></blockquote><br>Could you explain the scenarios where =
outbound-related error codes might<br>be used? Feel free to say "just =
read RFC 5626". :-)<br><br></blockquote><br>A 430 may occur if the =
gateway sends a SIP INVITE to the target domain and the edge proxy which =
handles the INVITE gets a flow error from the authoritative proxy behind =
him. Forget about 439, that's for REGISTERs, sorry.<br><br>--<br>Sa=FAl =
Ibarra Corretg=E9<br>AG =
Projects<br><br><br><br>_______________________________________________<br=
>stox mailing list<br><a =
href=3D"mailto:stox@ietf.org">stox@ietf.org</a><br>https://www.ietf.org/ma=
ilman/listinfo/stox<br></blockquote></div><br><div =
apple-content-edited=3D"true">
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; =
font-style: normal; font-variant: normal; font-weight: normal; =
letter-spacing: normal; line-height: normal; orphans: 2; text-align: =
auto; text-indent: 0px; text-transform: none; white-space: normal; =
widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0; "><div>---</div><div>* Olle E =
Johansson -&nbsp;<a =
href=3D"mailto:oej@edvina.net">oej@edvina.net</a></div><div>* Cell phone =
+46 70 593 68 51, Office +46 8 96 40 20, Sweden</div><div><br =
class=3D"webkit-block-placeholder"></div></span><br =
class=3D"Apple-interchange-newline">

</div>
<br></body></html>=

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On 7/31/13 4:51 PM, Olle E. Johansson wrote:
> 
> 31 jul 2013 kl. 16:42 skrev Saúl Ibarra Corretgé <saul@ag-projects.com
> <mailto:saul@ag-projects.com>>:
> 
>>>
>>>
>>>> - Sec 5.1 maps the XMPP unexpected-request error to SIP code 491,
>>>> which is very tied to the dialog semantics IMHO. I tried to find a
>>>> better matching code but I couldn't though. Maybe a 500 with a
>>>> reasonable Retry-After header value?
>>>
>>> It seems to me that we'd want it to be a 4xx series error, but I don't
>>> see a good one in RFC 3261.
>>
>> Exactly my thought.
> "unexpected-request error" seems more like a server error - or does it
> point to the actual request?

Not necessarily a server thing. E.g., you send me <iq type='result'/>
but I never sent you <iq type='get|set'/> (where "you" and "me" are
clients).

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



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>>=20
>> - What should a gateway do when it gets a message with XHTML-IM body?
>> I guess send it with content-type text/html, but is there any
>> translation required?
>=20
> I know about the use of XHTML in XMPP, but I don't know much about the
> use in SIP. Is there a profile of (X)HTML that most SIP entities =
implement?
>=20

SIP entities will implement text/html, can we use that?

>> - Section 4 mentions what should happen if no formal sessions are
>> used, but doesn't say anything of the opposite. Should we say that
>> those are out of the scope of this specification?
>=20
> Precursors to this document talked about formal chat sessions in XMPP
> (XEP-0155), however no clients support that model so I remove all of
> that text. The first paragraph of Section 4 reflects the earlier =
specs.
> I think we just need to remove a few words from Section 4...
>=20
> OLD
>   When an MSRP client sends messages through a gateway to an XMPP
>   client that does not support formal sessinos, the order of events is
>   as follows.
>=20
> NEW
>   When an MSRP client sends messages through a gateway to an XMPP
>   client, the order of events is as follows.
>=20

Sounds good to me.

>> - I think we should define a mapping between XMPP chatstates
>> (XEP-0085) and the iscomosing indication (RFC 3994) particularly, the
>> "gone" state in XMPP should trigger a BYE on the SIP side, and vice
>> versa.
>=20
> Could you describe a bit why you think that would be useful? (You
> explained it to me in person, but I think it would be helpful to =
discuss
> it on the list so that it's in the records of the WG. And for the =
record
> I am pretty sure I agree with you.)
>=20

For one, both SIP and XMPP clients usually implement typing indications =
so translating the semantics between the two is useful to get that =
information through. Now, as for the 'gone' state case, it's used by =
XMPP clients when the conversation is closed, so the other party can =
know that the window was closed on the other side and conversation has =
ended. Ending a conversation in SIP would mean sending a BYE, and this =
is particularly useful, because otherwise the SIP / MSRP session would =
stay up forever since there is no ending condition.

>> - Should we add text to map MSRP reports to XMPP receipts (XEP-0184)?
>> They do map very well semantically. Now, XEP-0184 says that the ack
>> may not be sent if "The recipient simply does not wish to return a
>> receipt for the content message.", which pretty much renders it
>> useless. Anyway, I think we do need some text explaining how MSRP
>> success and failure ports should be dealt with.
>=20
> I wonder if this belongs in the stox-chat spec or in an extension =
document.
>=20
> (And BTW XMPP message receipts can be used outside of chat sessions, =
so
> I think it applies more broadly than the stox-chat spec.)
>=20

I didn't know they could be used for other things, but I do think that =
their use in the context of a chat session should be translated since =
the meaning they have is basically the same as the MSRP REPORT chunk.

>> - In general: we don't have any example using CPIM, not sure if we
>> should say anything about it.
>=20
> What do you have in mind for CPIM examples? Do you mean examples other
> than MSRP, or examples with CPIM payloads?
>=20

Examples with CPIM payloads. But I see we have them in the group chat =
document, so you can skip this point.

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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On Jul 31, 2013, at 4:47 PM, Peter Saint-Andre <stpeter@stpeter.im> =
wrote:

> On 7/30/13 4:19 PM, Sa=FAl Ibarra Corretg=E9 wrote:
>> Hi,
>>=20
>> Here is my review on the -im document:
>>=20
>> - Section 3 suggests that the thread element in XMPP should be mapped
>> to the Call-ID header. I see why this was done, but it looks a bit
>> like abusing it.=20
>=20
> What are your specific concerns? Would something break? Is it =
insecure?
> Does it horribly violate the semantics defined in RFC 3261? I thought =
it
> was a pretty good match:
>=20
>   The Call-ID header field acts as a unique identifier to group
>   together a series of messages.
>=20
> But maybe I don't understand the "oral tradition" of how Call-ID has
> actually be used.
>=20

It doesn't horribly violate it, but it's use has been to correlate =
messages within a dialog, together with the tags. Since MESSAGE requests =
could come retransmitted when using UDP, duplicated Call-ID detection is =
a way to avoid it.

>> I would probably take it out or define it as a CPIM
>> header, in case CPIM is used. Thoughts?
>=20
> I don't see a header in RFC 3862 that we could map to. Our charter =
says
> we're not allowed to define new protocols. Someone could define a new
> CPIM header, but I don't think that would be widely adopted anytime =
soon.
>=20

I see. In that case I'd go for not really mapping it, but I won't fight =
to death if you leave it in ;-) However, since this document is for =
single messages ('normal' or 'headline' in XMPP), not sure if the thread =
is all that useful.=20

>> - Section 5 only mentions text/plain and text/html, I think we should
>> add message/cpim in there.
>=20
> No objections here.
>=20
>> - Similarly to chat, I think we should define a mapping between XMPP
>> chatstates (XEP-0085) and the iscomosing indication (RFC 3994), but
>> in this case there would be no mapping for the 'gone' state.
>=20
> The stox-im spec is only for single messages, so I don't see how chat
> states apply.
>=20

Hum, I think you are right, that would apply if we created a -chat =
equivalent document but mapping chat sessions to SIP MESSAGE it would be =
useful, but not here, forget it.

> (This document might be the place to talk about message receipts, =
however.)
>=20

Why? These are just one-off messages, aren't they?

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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31 jul 2013 kl. 16:52 skrev Sa=FAl Ibarra Corretg=E9 =
<saul@ag-projects.com>:

>>>=20
>>> - What should a gateway do when it gets a message with XHTML-IM =
body?
>>> I guess send it with content-type text/html, but is there any
>>> translation required?
>>=20
>> I know about the use of XHTML in XMPP, but I don't know much about =
the
>> use in SIP. Is there a profile of (X)HTML that most SIP entities =
implement?
>>=20
>=20
> SIP entities will implement text/html, can we use that?
Or recommend multipart/alternative if it's a downgrade?
Which document says that SIP Uas will support text/html?

>=20
>>> - Section 4 mentions what should happen if no formal sessions are
>>> used, but doesn't say anything of the opposite. Should we say that
>>> those are out of the scope of this specification?
>>=20
>> Precursors to this document talked about formal chat sessions in XMPP
>> (XEP-0155), however no clients support that model so I remove all of
>> that text. The first paragraph of Section 4 reflects the earlier =
specs.
>> I think we just need to remove a few words from Section 4...
>>=20
>> OLD
>>  When an MSRP client sends messages through a gateway to an XMPP
>>  client that does not support formal sessinos, the order of events is
>>  as follows.
>>=20
>> NEW
>>  When an MSRP client sends messages through a gateway to an XMPP
>>  client, the order of events is as follows.
>>=20
>=20
> Sounds good to me.
>=20
>>> - I think we should define a mapping between XMPP chatstates
>>> (XEP-0085) and the iscomosing indication (RFC 3994) particularly, =
the
>>> "gone" state in XMPP should trigger a BYE on the SIP side, and vice
>>> versa.
>>=20
>> Could you describe a bit why you think that would be useful? (You
>> explained it to me in person, but I think it would be helpful to =
discuss
>> it on the list so that it's in the records of the WG. And for the =
record
>> I am pretty sure I agree with you.)
>>=20
>=20
> For one, both SIP and XMPP clients usually implement typing =
indications so translating the semantics between the two is useful to =
get that information through. Now, as for the 'gone' state case, it's =
used by XMPP clients when the conversation is closed, so the other party =
can know that the window was closed on the other side and conversation =
has ended. Ending a conversation in SIP would mean sending a BYE, and =
this is particularly useful, because otherwise the SIP / MSRP session =
would stay up forever since there is no ending condition.
Is the closing window notification more than just closing window? Is the =
chat dead, really?

/O=

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On 7/31/13 5:07 PM, Saúl Ibarra Corretgé wrote:
> 
> On Jul 31, 2013, at 4:47 PM, Peter Saint-Andre <stpeter@stpeter.im>
> wrote:
> 
>> On 7/30/13 4:19 PM, Saúl Ibarra Corretgé wrote:
>>> Hi,
>>> 
>>> Here is my review on the -im document:
>>> 
>>> - Section 3 suggests that the thread element in XMPP should be
>>> mapped to the Call-ID header. I see why this was done, but it
>>> looks a bit like abusing it.
>> 
>> What are your specific concerns? Would something break? Is it
>> insecure? Does it horribly violate the semantics defined in RFC
>> 3261? I thought it was a pretty good match:
>> 
>> The Call-ID header field acts as a unique identifier to group 
>> together a series of messages.
>> 
>> But maybe I don't understand the "oral tradition" of how Call-ID
>> has actually be used.
>> 
> 
> It doesn't horribly violate it, but it's use has been to correlate
> messages within a dialog, together with the tags. Since MESSAGE
> requests could come retransmitted when using UDP, duplicated Call-ID
> detection is a way to avoid it.

Ah, I see.

>>> I would probably take it out or define it as a CPIM header, in
>>> case CPIM is used. Thoughts?
>> 
>> I don't see a header in RFC 3862 that we could map to. Our charter
>> says we're not allowed to define new protocols. Someone could
>> define a new CPIM header, but I don't think that would be widely
>> adopted anytime soon.
>> 
> 
> I see. In that case I'd go for not really mapping it, but I won't
> fight to death if you leave it in ;-) However, since this document is
> for single messages ('normal' or 'headline' in XMPP), not sure if the
> thread is all that useful.

I see your point.

>>> - Section 5 only mentions text/plain and text/html, I think we
>>> should add message/cpim in there.
>> 
>> No objections here.
>> 
>>> - Similarly to chat, I think we should define a mapping between
>>> XMPP chatstates (XEP-0085) and the iscomosing indication (RFC
>>> 3994), but in this case there would be no mapping for the 'gone'
>>> state.
>> 
>> The stox-im spec is only for single messages, so I don't see how
>> chat states apply.
>> 
> 
> Hum, I think you are right, that would apply if we created a -chat
> equivalent document but mapping chat sessions to SIP MESSAGE it would
> be useful, but not here, forget it.

Agreed.

>> (This document might be the place to talk about message receipts,
>> however.)
>> 
> 
> Why? These are just one-off messages, aren't they?

Well, the sender might really really want to know if the message was
delivered to the recipient. :-)

My feeling about message receipts is that we leave them out for now and
maybe define a mapping for them in another specification, but I am open
to persuasion on the topic.

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



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>>=20
>> SIP entities will implement text/html, can we use that?
> Or recommend multipart/alternative if it's a downgrade?

I have never seen multipart used in MSRP and I'd rather not open that =
door ;-)

> Which document says that SIP Uas will support text/html?
>=20

Well, nobody says UAs will support it, it's negotiated with =
a=3Daccept-types in SDP, but an MSRP client is very likely to implement =
it, directly or wrapped in CPIM.

>> For one, both SIP and XMPP clients usually implement typing =
indications so translating the semantics between the two is useful to =
get that information through. Now, as for the 'gone' state case, it's =
used by XMPP clients when the conversation is closed, so the other party =
can know that the window was closed on the other side and conversation =
has ended. Ending a conversation in SIP would mean sending a BYE, and =
this is particularly useful, because otherwise the SIP / MSRP session =
would stay up forever since there is no ending condition.
> Is the closing window notification more than just closing window? Is =
the chat dead, really?
>=20

My understanding is that chat "sessions" in XMPP is a very loose thing, =
so you could actually resume it later, but as a concept, if you close =
the window and you signal that you are "gone" I wouldn't expect further =
conversation from him. XMPP clients usually say "Olle has left the =
conversation".

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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>>=20
>=20
> Well, the sender might really really want to know if the message was
> delivered to the recipient. :-)
>=20

Unfortunately there is no way to do that with a SIP MESSAGE. Unless we =
invent a special payload for acknowledging reception of a given SIP =
MESSAGE, but I think that is really out of the scope of what we are =
doing in the IM document.

> My feeling about message receipts is that we leave them out for now =
and
> maybe define a mapping for them in another specification, but I am =
open
> to persuasion on the topic.
>=20

As I said above we have no way to use them in the -im document, but =
there is a clear equivalence between the receipt and the MSRP REPORT =
chunk which we use in -chat. If the SIP endpoint requests success and =
failure reports, we'd need to fool it by always sending a report, =
regardless of the acknowledgement of the message (in case receipts are =
supported). I can live with that if we add some text clarifying she a =
REPORT should be sent.

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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>>=20
>> It doesn't horribly violate it, but it's use has been to correlate =
messages within a dialog, together with the tags. Since MESSAGE requests =
could come retransmitted when using UDP, duplicated Call-ID detection is =
a way to avoid it.
> Which you should not do. You should use all of the SIP transaction =
identifiers - cseq included.
>=20

I know. Anyway, as I said, given that this is -im, not -chat, I don't =
think we should translate it.

>>=20
>> Why? These are just one-off messages, aren't they?
> Page mode messages is proper terminology I believe ;-)
>=20

Yeah, that!

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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Hi Saul, Peter,

Sa=FAl Ibarra Corretg=E9 wrote:
> >
> > Well, the sender might really really want to know if the message was
> > delivered to the recipient. :-)
> >
>=20
> Unfortunately there is no way to do that with a SIP MESSAGE. Unless we
> invent a special payload for acknowledging reception of a given SIP
> MESSAGE, but I think that is really out of the scope of what we are doing=
 in
> the IM document.
>=20

Please take a look at RFC 5438 (http://datatracker.ietf.org/doc/rfc5438/) a=
nd see if it applies to what would be required.

Markus

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Subject: Re: [Stox] Comments on draft-ietf-stox-presence-00
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 Peter,

The suggest text below makes sense to me.  In my opinion, this helps
more clearly defining the presence mapping from XMPP to SIP for values
such as 'away'.

I think there needs to be similar text for the second table (currently
labeled Table 6), which states that the gateway SHOULD map the <show
xmlns='jabber:client'> element in the PIDF XML to the <show/> element
in the XMPP presence stanza.  This would only be for SIP clients that
support that particular namespace, but it at least provides a
recommended mapping in the opposite direction for clients that do
support that namespace.

~Michael

On Mon, Jul 29, 2013 at 6:49 AM, Peter Saint-Andre <stpeter@stpeter.im> wrote:
> On 7/26/13 5:55 PM, Peter Saint-Andre wrote:
>> On 7/26/13 4:50 PM, Michael Lundberg wrote:
>>
>>> One clarifying question: when you say "if the SIP implementation
>>> supports the namespace", do you mean the XMPP-to-SIP gateway or the
>>> SIP user agent?
>>>
>>>> Good question.  It would be benificial if both supported the
>>>> namespace, but only the gateway is probably required to.  If the
>>>> client supports the namespace, then the gateway would just need to map
>>>> between the elements described in this document.
>>>
>>>> If the client doesn't support the namespace, the gateway would most
>>>> likely need to do an additional translation into a namespace the
>>>> client does understand.  In this case, the values might not be the
>>>> same between the two namespaces, and therefore things are 'lost' in
>>>> translation. This is one of the big issues with presence mapping today
>>>> as many implementations have thier own implementation specific
>>>> namespace, which makes it hard to map between different
>>>> implementations.  Both the implementation specific and common
>>>> namespaces could coexist, where the implementation specific namespace
>>>> is used for internal communication and a common, standard namespace
>>>> (e.g., 'jabber:client' ) is used when communicating between different
>>>> implementations.
>>
>> Yes, I think that makes sense. I'm not sure exactly how to translate
>> that into text (especially the part about proprietary / non-standard
>> namespaces), but I'll work something up for consideration by the list.
>
> Here is some proposed text:
>
> OLD
>    5.  Some implementations support custom extensions to encapsulate
>        this information; however, there is no need to standardize a PIDF
>        extension for this purpose, since PIDF is already extensible and
>        thus the <show/> element can be included directly, qualified by
>        the 'jabber:client' namespace in the PIDF XML.  The examples in
>        this document illustrate this usage, which is RECOMMENDED.  The
>        most useful values are likely "away" and "dnd", although note
>        that the latter value merely means "busy" and does not imply that
>        a server or client ought to block incoming traffic while the user
>        is in that state.
>
> NEW
>    5.  Some implementations support custom extensions to encapsulate
>        detailed information about availability; however, there is no
>        need to standardize a PIDF extension for this purpose, since
>        PIDF is already extensible and thus the <show/> element
>        (qualified by the 'jabber:client' namespace) can be included
>        directly in the PIDF XML.  The examples in this document
>        illustrate this usage, which is RECOMMENDED.  The most useful
>        values are likely "away" and "dnd", although note that the
>        latter value merely means "busy" and does not imply that a
>        server or client ought to block incoming traffic while the user
>        is in that state.  Naturally, a gateway can choose to translate
>        a custom extension into an established value of the <show/>
>        element [RFC6121], or translate a <show/> element into a custom
>        extension that the gateway knows is supported by the user agent
>        of the intended recipient.  Unfortunately, this behavior does
>        not guarantee that information will not be lost; to help prevent
>        information loss, a gateway ought to include both the <show/>
>        element and the custom extension if the gateway cannot suitably
>        translate the custom value into a <show/> value.
>
> Mike, does that text address your concern? Do we need to say more (or
> less) than that?
>
> Peter
>
> --
> Peter Saint-Andre
> https://stpeter.im/
>
>

From stpeter@stpeter.im  Wed Jul 31 23:38:50 2013
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Cc: "stox@ietf.org" <stox@ietf.org>
Subject: Re: [Stox] Review on -presence
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On 7/30/13 5:33 PM, Saúl Ibarra Corretgé wrote:
> Hi,
> 
> Here is my review of the -presence document:
> 
> - Sec 3.3.2 suggests that if the subscription is maintained but we
> have no presence state, a PIDF would be generated with a basic status
> of closed, but what would be the tuple ID, if we don't know any
> resource for this user anymore? We could also send an empty NOTIFY,
> which would achieve the same goal.

The empty NOTIFY sounds more consistent with the spirit of SIP presence.

> - Sec 3.3.2 specifies how to handle the case when a gateway
> translated a SIP subscription into an XMPP subscription and then the
> SIP side decided to end it, but doesn't mention what to do in case
> the XMPP side decides to end it. It should send a NOTIFY with
> Subscription-State: terminated;reason=rejected.

Agreed.

> - Sec 4.1, "Instant Message URI of the form <pres:user@domain>" -
> shouldn't this be "Presence URI of the form…" ?

Looks like a copy-and-paste error.

> - Using ID-123kdklejd doesn't seem to work as a valid xs:ID, TID-1234
> does work though, so we could use TID- as the prefix for tuple
> identifiers in examples and such.

I will double-check that against the XML specification.

> - Table 6 suggests that the Call-ID header should be matched to the
> presence stanza id, but the called won't change throughout the whole
> dialog, so we should use something else, which changes for each SIP
> transaction, what about the Via branch? Table 7 uses CSeq, FWIW.

You're right -- Call-ID is not right here. CSeq is better, and I think
we changed it in Table 7 but not Table 6.

> - Table 6 suggests that priority is translated as the PIDF priority
> for a tuple, but that's an attribute of the contact element, which we
> don't mention, but I guess we should. 

Good catch -- we don't seem to include the contact element in any of the
examples. Will fix.

> Also, the priority in PIDF is a
> float between 0 and 1, is it the same for XMPP?

Hmm. We had text about that in RFC 3922, and I suggest we copy that to
this draft (adjusting as necessary)...

   An XMPP presence stanza MAY contain a <priority/> child element whose
   value is an integer between -128 and +127.  The value of this element
   MAY be mapped to the 'priority' attribute of the <contact/> child of
   the PIDF <tuple/> element.  If the value of the XMPP <priority/>
   element is negative, an XMPP-CPIM gateway MUST NOT map the value. The
   range of allowable values for the PIDF 'priority' attribute is any
   decimal number from zero to one inclusive, with a maximum of three
   decimal places.  If an XMPP-CPIM gateway maps these values, it SHOULD
   treat XMPP <priority>0</priority> as PIDF priority='0' and XMPP
   <priority>127</priority> as PIDF priority='1', mapping intermediate
   values appropriately so that they are unique (e.g., XMPP priority 1
   to PIDF priority 0.007, XMPP priority 2 to PIDF priority 0.015, and
   so on up through mapping XMPP priority 126 to PIDF priority 0.992;
   note that this is an example only, and that the exact mapping shall
   be determined by the XMPP-CPIM gateway).

That's twice in one day I have looked at RFC 3922. ;-)

> - Sec 4.3, the translated XMPP stanza contains the full JID of
> Juliet, but where did we get the resource from? Shouldn't it go to
> the bare JID?

Another copy-and-paste error, I think.

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



From stpeter@stpeter.im  Wed Jul 31 23:50:09 2013
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Cc: "stox@ietf.org" <stox@ietf.org>
Subject: Re: [Stox] draft slides for core, presence, im, chat
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OK, I have updated my slides:

https://stpeter.im/files/ietf87-stox-core-presence.pdf

Sorry it took me a bit longer than promised -- I wanted to incorporate
some results of the list traffic yesterday.

Note that I have not included all issues in the slides, only the issues
that in my judgment merit a bit of face-to-face discussion.

Peter

On 7/31/13 6:39 AM, Peter Saint-Andre wrote:
> This is just a warning that I'll be providing updated slides to address
> some points that Saúl made in his reviews. I'll try to do that as early
> today as possible.
> 
> On 7/30/13 2:59 PM, Yana Stamcheva wrote:
>> Hi Peter,
>>
>> Thanks! Your new slides are uploaded.
>>
>> Yana
>>
>> On Jul 30, 2013, at 2:15 PM, Peter Saint-Andre <stpeter@stpeter.im> wrote:
>>
>>> After a hallway discussion with Robert Sparks, I have updated these
>>> slides. I will post to the list about the issue he raised.
>>>
>>> On 7/24/13 8:09 PM, Peter Saint-Andre wrote:
>>>> Draft slides for discussion of the -core, -presence, -im, and -chat
>>>> specfications are here:
>>>>
>>>> https://stpeter.im/files/ietf87-stox-core-presence.pdf
>>>>
>>>> As you can see, there are very few open issues. If folks provide
>>>> further feedback about these specs in the next few days, I'll update
>>>> the slides accordingly. If not, mine will be a very short presentation
>>>> and we'll have more time to discuss the -groupchat and -media I-Ds. :-)
>>>>
>>>> Peter
>>>>
