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Date: Tue, 04 Feb 2014 21:56:38 +0100
From: Philipp Hancke <fippo@goodadvice.pages.de>
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Subject: Re: [Stox] I-D Action: draft-ietf-stox-media-02.txt
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Am 31.01.2014 23:25, schrieb Peter Saint-Andre:
> On 12/17/13, 10:55 AM, Philipp Hancke wrote:
>>> 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).
>>>
>>
>> A couple of comments, I just had some time (90 minutes :-)) to review
>> this:
>
> Thanks, Philipp. Sorry about the delay in replying - as you might be
> aware I've had quite a bit happening since the interim meeting. ;-)

heh :-)

[... snipping where +1]


> As you know, that's used by the recipient to decline the addition of a
> new content type to the session (e.g., the other party wants to add
> video to an audio-only session). Potentially we could map this to a SIP
> 415 Unsupported Media Type response. However, this wouldn't round-trip
> very well because the stox-core document says to map SIP 415 to XMPP
> <not-acceptable/> whereas Jingle content-reject is not an stanza-level
> error but a Jingle-level response.

Tricky... the xmppish way to do this would be to discover support via 
disco/caps.
[...]
>> Actually, Jingle has a cool message for hold which is content-modify
>> (which only modifies senders). Why bother with two hold mechanisms when
>> you can have three :-)
>
> Sure, the more the merrier. :P
>
> Is senders='none' implemented for hold functionality? We've been trying
> to define only what is in general use.

Emil?

[...]
>> XEP-0177 doesn't define <format/>. 0167 defines <parameter/> which is
>> what you describe (and it does so in a wrong way, spec bug).
>
> Are you saying there is a spec bug in XEP-0167 or in the stox-media I-D?

stox-media should use <parameter/>. We need to fix the spec bug in 0167 
at some point, but that's an XSF activity.

> Naturally, this <parameter/> element works only for RTP communications,
> not other Jingle applications (say, file transfer). So we'll mention
> that in the text.

well, the general pattern is reused in some specs (0293 + 0294 IIRC), 
but those are RTP-related, so yeah.

>> I think DTMF is the relevant example for bullet #3 (example from RFC
>> 4733)
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> Saul : I think I disagree with your comment about name being required by
>> the xep's schema (which isn't normative anyway), both are required.
>> My take (after some discussions in jingle@) is that 'name' should be
>> left empty and you should put the stuff to 'value'.
>
> Yes, that seems better. Notice how RFC 4733 says the following things
> represent the same information:
>
> (1)
>
>        audio/telephone-event;events="0-15,66,70";rate="8000"
>
> and
>
> (2)
>
>        m=audio 12346 RTP/AVP 100
>        a=rtpmap:100 telephone-event/8000
>        a=fmtp:100 0-15,66,70
>
> Although (1) says events='', the events name doesn't appear in (2). So
> putting the stuff in the <parameter value=''/> (and empty name) seems
> reasonable.

uhm... no. This would be
<parameter name='' value='0-15,66,70'/>
Just show me your draft before submitting ;-)

>> Section 10:
>> This section actually concerns me. Jingle is based on full jids
>> typically. So in order to forward a call from a SIP entity to an xmpp
>> entity, the gateway needs to determine which full jid to send to.
>> So it either has to have a presence subscription or can use presence
>> decloaking (XEP-0276; deferred because of inactivity but works nicely).
>> If there are multiple possible resources, the gateway needs to decide,
>> where to route the call.
>
> The text here talks about the XMPP 'from' address that the gateway will
> place on provisional responses received from the SIP side. Since there
> might be multiple responses, we're saying to ignore the GRUU and make
> believe that the JID-equivalent of the SIP sender is a bare JID, not a
> full JID.
>
> I'm not yet convinced that this is the best we can do in the SIP to XMPP
> direction, but it might be...

Some things I discussed with Lance Stout at the recent XMPP summit about 
jingle+Forking+Push might be useful here. But those are probably too 
vague and untested yet, so we might need to invest some work into 
getting 0276 into a formally better state. It works quite well in 
practice, so...

[...]
>> Section 11.1:
>> the candidate in the example should probably have an id (at least where
>> the 0177 schema is concerned which is not normative).
>
> It needs to have a 'component' attribute too, per XEP-0177.

good catch.

[...]
>> other issues:
>> routing of <iq/> (for the jingle->sip call-a-telephone-number usecase)
>> that was raised at the end of the meeting:
>> this should go to the bare JID. Semantically, this means "don't route it
>> to a client, server handles it". Here, the 'server' is handling it.
>
> I'll need to listen to the audio recording in order to know the context
> here.

the discussion starts around 01:16, specifically Emil's comment at 
01:17:30 about sending from bare jids. IMO it's semantically ok to use 
bare jids on the XMPP side. But a hardcoded resource like "foo" doesn't 
hurt either and is likely to break less assumptions from client developers.

>> Peter: during the call I made a note "more disco#info examples", I think
>> this was based on a comment by Markus about capability negotiation. I
>> hope I remember the actual context when seeing the minutes.
>
> As in all of these mapping specs, we try not to reply too much on
> discovering the capabilities of other entities, since SIP doesn't have
> any reliable facilities for that. But if you remember the context, let
> us know. :-)

need to listen to the full recording again... more soon.

thanks peter!

From stpeter@stpeter.im  Tue Feb  4 18:42:56 2014
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Subject: Re: [Stox] I-D Action: draft-ietf-stox-media-02.txt
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On 2/4/14, 1:56 PM, Philipp Hancke wrote:
> Am 31.01.2014 23:25, schrieb Peter Saint-Andre:
>> On 12/17/13, 10:55 AM, Philipp Hancke wrote:

> [... snipping where +1]

Same.

>> As you know, that's used by the recipient to decline the addition of a
>> new content type to the session (e.g., the other party wants to add
>> video to an audio-only session). Potentially we could map this to a SIP
>> 415 Unsupported Media Type response. However, this wouldn't round-trip
>> very well because the stox-core document says to map SIP 415 to XMPP
>> <not-acceptable/> whereas Jingle content-reject is not an stanza-level
>> error but a Jingle-level response.
>
> Tricky... the xmppish way to do this would be to discover support via
> disco/caps.

Right. Given that we don't have such methods on the SIP side, I don't 
think we can make a strong recommendation here.

>>> XEP-0177 doesn't define <format/>. 0167 defines <parameter/> which is
>>> what you describe (and it does so in a wrong way, spec bug).
>>
>> Are you saying there is a spec bug in XEP-0167 or in the stox-media I-D?
>
> stox-media should use <parameter/>. We need to fix the spec bug in 0167
> at some point, but that's an XSF activity.

Hmm, XEP-0167 has lots of things like this:

<parameter name='foo' value='bar'/>

But if there's a spec bug in XEP-0167, let's discuss it on the 
jingle@xmpp.org list. :-)

>>> I think DTMF is the relevant example for bullet #3 (example from RFC
>>> 4733)
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-15
>>> Saul : I think I disagree with your comment about name being required by
>>> the xep's schema (which isn't normative anyway), both are required.
>>> My take (after some discussions in jingle@) is that 'name' should be
>>> left empty and you should put the stuff to 'value'.
>>
>> Yes, that seems better. Notice how RFC 4733 says the following things
>> represent the same information:
>>
>> (1)
>>
>>        audio/telephone-event;events="0-15,66,70";rate="8000"
>>
>> and
>>
>> (2)
>>
>>        m=audio 12346 RTP/AVP 100
>>        a=rtpmap:100 telephone-event/8000
>>        a=fmtp:100 0-15,66,70
>>
>> Although (1) says events='', the events name doesn't appear in (2). So
>> putting the stuff in the <parameter value=''/> (and empty name) seems
>> reasonable.
>
> uhm... no. This would be
> <parameter name='' value='0-15,66,70'/>

Right, exactly. I was just being too terse in what I wrote.

> Just show me your draft before submitting ;-)

Keeping it updated in SVN for sure:

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

>>> Section 10:
>>> This section actually concerns me. Jingle is based on full jids
>>> typically. So in order to forward a call from a SIP entity to an xmpp
>>> entity, the gateway needs to determine which full jid to send to.
>>> So it either has to have a presence subscription or can use presence
>>> decloaking (XEP-0276; deferred because of inactivity but works nicely).
>>> If there are multiple possible resources, the gateway needs to decide,
>>> where to route the call.
>>
>> The text here talks about the XMPP 'from' address that the gateway will
>> place on provisional responses received from the SIP side. Since there
>> might be multiple responses, we're saying to ignore the GRUU and make
>> believe that the JID-equivalent of the SIP sender is a bare JID, not a
>> full JID.
>>
>> I'm not yet convinced that this is the best we can do in the SIP to XMPP
>> direction, but it might be...
>
> Some things I discussed with Lance Stout at the recent XMPP summit about
> jingle+Forking+Push might be useful here. But those are probably too
> vague and untested yet, so we might need to invest some work into
> getting 0276 into a formally better state. It works quite well in
> practice, so...

I've always liked that Presence Decloaking stuff in XEP-0276, but it's 
not implemented widely if at all, so I don't think it's correct for us 
to recommend it in draft-ietf-stox-media.

>>> other issues:
>>> routing of <iq/> (for the jingle->sip call-a-telephone-number usecase)
>>> that was raised at the end of the meeting:
>>> this should go to the bare JID. Semantically, this means "don't route it
>>> to a client, server handles it". Here, the 'server' is handling it.
>>
>> I'll need to listen to the audio recording in order to know the context
>> here.
>
> the discussion starts around 01:16, specifically Emil's comment at
> 01:17:30 about sending from bare jids. IMO it's semantically ok to use
> bare jids on the XMPP side. But a hardcoded resource like "foo" doesn't
> hurt either and is likely to break less assumptions from client developers.

Ick, hardcoding. :(

But I'll listen to the recording soon and think about the issue more fully.

Peter

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

From internet-drafts@ietf.org  Fri Feb  7 08:20:35 2014
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the SIP-TO-XMPP Working Group of the IETF.

        Title           : Interworking between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP): Presence
        Authors         : Peter Saint-Andre
                          Avshalom Houri
                          Joe Hildebrand
	Filename        : draft-ietf-stox-presence-08.txt
	Pages           : 28
	Date            : 2014-02-07

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-08

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


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

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


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Changes to address IESG and SecDir feedback...

On 2/7/14, 9:20 AM, internet-drafts@ietf.org wrote:
>
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>   This draft is a work item of the SIP-TO-XMPP Working Group of the IETF.
>
>          Title           : Interworking between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP): Presence
>          Authors         : Peter Saint-Andre
>                            Avshalom Houri
>                            Joe Hildebrand
> 	Filename        : draft-ietf-stox-presence-08.txt
> 	Pages           : 28
> 	Date            : 2014-02-07
>
> 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-08
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-stox-presence-08
>
>
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
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> stox@ietf.org
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Subject: [Stox] I-D Action: draft-ietf-stox-presence-09.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the SIP-TO-XMPP Working Group of the IETF.

        Title           : Interworking between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP): Presence
        Authors         : Peter Saint-Andre
                          Avshalom Houri
                          Joe Hildebrand
	Filename        : draft-ietf-stox-presence-09.txt
	Pages           : 28
	Date            : 2014-02-09

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-09

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-stox-presence-09


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

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From stpeter@stpeter.im  Sun Feb  9 18:39:47 2014
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Pete Resnick caught several more errors, which are corrected in this 
version.

On 2/9/14, 7:38 PM, internet-drafts@ietf.org wrote:
>
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>   This draft is a work item of the SIP-TO-XMPP Working Group of the IETF.
>
>          Title           : Interworking between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP): Presence
>          Authors         : Peter Saint-Andre
>                            Avshalom Houri
>                            Joe Hildebrand
> 	Filename        : draft-ietf-stox-presence-09.txt
> 	Pages           : 28
> 	Date            : 2014-02-09
>
> 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-09
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-stox-presence-09
>
>
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox
>



From stpeter@stpeter.im  Mon Feb 10 10:24:22 2014
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( + STOX WG since one of the issues might deserve broader discussion )

Hi Stephen, thanks for your review:

https://datatracker.ietf.org/doc/draft-ietf-stox-core/ballot/

You wrote in part:

    - I wondered why you didn't just say that (D)TLS SHOULD
    be used/supported between gateways. Given that all the
    relevant bits of code are likely to support that, wouldn't
    it be a good thing?

Yes, that seems eminently reasonable.

    - Has anyone thought about confusability in the name
    mappings? I expected to see a bit of text in the
    security considerations but didn't see it.

Confusion is always possible. :-) Were you thinking about confusable 
characters from Unicode, or something else?

    - It seems a shame to not be able to gateway when the To is
    a sips URI at all but I understand that some loss of
    security is inevitable for cases like this. Is there any
    work planned for an update that would allow gatewaying for
    such cases, e.g. if the 1st XMPP server is the one to which
    the user is connected and the user is connected using
    XMPP/TLS?

Hmm. I cannot say that I am aware of planning for updates to provide 
more secure gatewaying, although folks active in the STOX WG might be 
thinking along those lines.

Depending on the deployment architecture, I think there are cases where 
it is *possible* to TLS-protect all the hops. For instance, if 
sip.example.org has a direct server-to-server connection to 
xmpp.example.com (no intermediate hops) and both organizations agree to 
force the use of TLS for client connections (e.g., via SLA), then I 
suppose that sip.example.com could honor 'sips' URIs when sending 
traffic to xmpp.example.com. However, such an arrangement is rare enough 
right now that I don't know if it is worth mentioning.

Peter

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

From stpeter@stpeter.im  Mon Feb 10 14:08:52 2014
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Pete, you wrote:

   Section 4 is (nicely) clear on the document architecturally
   describing a gateway. However, traditionally a gateway is
   transparent to the entity that communicates with it: When we
   had SMTP-to-X.400 gateways, the gateway appeared as just another
   SMTP system that noticed special qualities of the address and
   then routed the messages appropriately.

Your "traditionally" sounds like an SMTP phenomenon. In XMPP, we don't 
have intermediate transfer agents, and XMPP servers are designed to use 
add-on modules for additional functionality. So in the XMPP universe it 
makes sense for the operator of an XMPP service to install a module (in 
addition to the XMPP server) that performs the gateway function (we 
often call this a connection manager).

   Section 5 describes
   something a bit different. It's not clear that what section 5
   describes actually is part of the gateway, but rather sounds
   like a combined server which does failover between the
   protocols. I don't think this is a showstopper, but it might
   help implementers significantly if you described in section 5
   *where* in the model this function occurs. Right now, it
   sounds like the server itself does the addressing failover,
   not the gateway.

Yes, I see your point. This kind of thing is quite likely implementation 
specific (e.g., when the add-on XMPP-to-SIP gateway module gets 
configured into and trusted by the core XMPP server, it might get added 
into an event listener for core stanza delivery if an XMPP lookup fails 
for the remote domain). Let me look at what text might be useful here.

Peter

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


From stephen.farrell@cs.tcd.ie  Mon Feb 10 14:20:43 2014
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Hi Peter,

On 02/10/2014 06:24 PM, Peter Saint-Andre wrote:
> ( + STOX WG since one of the issues might deserve broader discussion )
> 
> Hi Stephen, thanks for your review:
> 
> https://datatracker.ietf.org/doc/draft-ietf-stox-core/ballot/
> 
> You wrote in part:
> 
>    - I wondered why you didn't just say that (D)TLS SHOULD
>    be used/supported between gateways. Given that all the
>    relevant bits of code are likely to support that, wouldn't
>    it be a good thing?
> 
> Yes, that seems eminently reasonable.

Just to be clear - the above is the only DISCUSS point.

> 
>    - Has anyone thought about confusability in the name
>    mappings? I expected to see a bit of text in the
>    security considerations but didn't see it.
> 
> Confusion is always possible. :-) Were you thinking about confusable
> characters from Unicode, or something else?

Right. Maybe its all there already and I just didn't see
if in the mapping stuff, or maybe not. I'm not sure.

> 
>    - It seems a shame to not be able to gateway when the To is
>    a sips URI at all but I understand that some loss of
>    security is inevitable for cases like this. Is there any
>    work planned for an update that would allow gatewaying for
>    such cases, e.g. if the 1st XMPP server is the one to which
>    the user is connected and the user is connected using
>    XMPP/TLS?
> 
> Hmm. I cannot say that I am aware of planning for updates to provide
> more secure gatewaying, although folks active in the STOX WG might be
> thinking along those lines.
> 
> Depending on the deployment architecture, I think there are cases where
> it is *possible* to TLS-protect all the hops. For instance, if
> sip.example.org has a direct server-to-server connection to
> xmpp.example.com (no intermediate hops) and both organizations agree to
> force the use of TLS for client connections (e.g., via SLA), then I
> suppose that sip.example.com could honor 'sips' URIs when sending
> traffic to xmpp.example.com. However, such an arrangement is rare enough
> right now that I don't know if it is worth mentioning.

Fair enough. I can see that detecting that you (probably) have
all (D)TLS hops might be tricky and error prone.

Cheers,
S.


> 
> Peter
> 


From stpeter@stpeter.im  Mon Feb 10 14:52:34 2014
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On 2/10/14, 3:20 PM, Stephen Farrell wrote:
>
> Hi Peter,
>
> On 02/10/2014 06:24 PM, Peter Saint-Andre wrote:
>> ( + STOX WG since one of the issues might deserve broader discussion )
>>
>> Hi Stephen, thanks for your review:
>>
>> https://datatracker.ietf.org/doc/draft-ietf-stox-core/ballot/
>>
>> You wrote in part:
>>
>>     - I wondered why you didn't just say that (D)TLS SHOULD
>>     be used/supported between gateways. Given that all the
>>     relevant bits of code are likely to support that, wouldn't
>>     it be a good thing?
>>
>> Yes, that seems eminently reasonable.
>
> Just to be clear - the above is the only DISCUSS point.

Yep.

>>     - Has anyone thought about confusability in the name
>>     mappings? I expected to see a bit of text in the
>>     security considerations but didn't see it.
>>
>> Confusion is always possible. :-) Were you thinking about confusable
>> characters from Unicode, or something else?
>
> Right. Maybe its all there already and I just didn't see
> if in the mapping stuff, or maybe not. I'm not sure.

Section 10.5 of draft-ietf-precis-framework has some very nice text 
about visually similar characters. We could borrow from that.

>>     - It seems a shame to not be able to gateway when the To is
>>     a sips URI at all but I understand that some loss of
>>     security is inevitable for cases like this. Is there any
>>     work planned for an update that would allow gatewaying for
>>     such cases, e.g. if the 1st XMPP server is the one to which
>>     the user is connected and the user is connected using
>>     XMPP/TLS?
>>
>> Hmm. I cannot say that I am aware of planning for updates to provide
>> more secure gatewaying, although folks active in the STOX WG might be
>> thinking along those lines.
>>
>> Depending on the deployment architecture, I think there are cases where
>> it is *possible* to TLS-protect all the hops. For instance, if
>> sip.example.org has a direct server-to-server connection to
>> xmpp.example.com (no intermediate hops) and both organizations agree to
>> force the use of TLS for client connections (e.g., via SLA), then I
>> suppose that sip.example.com could honor 'sips' URIs when sending
>> traffic to xmpp.example.com. However, such an arrangement is rare enough
>> right now that I don't know if it is worth mentioning.
>
> Fair enough. I can see that detecting that you (probably) have
> all (D)TLS hops might be tricky and error prone.

That's how I see it.

Oh, and we at least need to mention DTLS for the UDP cases, though (see 
Section 5 of draft-ietf-stox-core).

Peter

-- 
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From stpeter@stpeter.im  Mon Feb 10 16:11:30 2014
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Subject: Re: [Stox] Pete Resnick's IESG feedback on draft-ietf-stox-core
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On 2/10/14, 3:08 PM, Peter Saint-Andre wrote:
> Pete, you wrote:
>
>    Section 4 is (nicely) clear on the document architecturally
>    describing a gateway. However, traditionally a gateway is
>    transparent to the entity that communicates with it: When we
>    had SMTP-to-X.400 gateways, the gateway appeared as just another
>    SMTP system that noticed special qualities of the address and
>    then routed the messages appropriately.
>
> Your "traditionally" sounds like an SMTP phenomenon. In XMPP, we don't
> have intermediate transfer agents, and XMPP servers are designed to use
> add-on modules for additional functionality. So in the XMPP universe it
> makes sense for the operator of an XMPP service to install a module (in
> addition to the XMPP server) that performs the gateway function (we
> often call this a connection manager).
>
>    Section 5 describes
>    something a bit different. It's not clear that what section 5
>    describes actually is part of the gateway, but rather sounds
>    like a combined server which does failover between the
>    protocols. I don't think this is a showstopper, but it might
>    help implementers significantly if you described in section 5
>    *where* in the model this function occurs. Right now, it
>    sounds like the server itself does the addressing failover,
>    not the gateway.
>
> Yes, I see your point. This kind of thing is quite likely implementation
> specific (e.g., when the add-on XMPP-to-SIP gateway module gets
> configured into and trusted by the core XMPP server, it might get added
> into an event listener for core stanza delivery if an XMPP lookup fails
> for the remote domain). Let me look at what text might be useful here.

How is this for proposed text?

    Existing SIP and XMPP server implementations do not typically include
    the ability to communicate using the other technology (XMPP for SIP
    implementations, SIP for XMPP implementations).  One common
    architectural pattern is to associate a gateway with the core server
    implementation (e.g., in XMPP such a gateway might be called a
    "connection manager").  How exactly such a gateway interacts with the
    core server to complete tasks such as address lookups and
    communication with systems that use the other technology is a matter
    of implementation (e.g., the gateway might be an add-on module that
    is trusted by the core server to act as a fallback delivery mechanism
    if the remote domain does not support the server's native
    communication technology).

Peter

-- 
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From internet-drafts@ietf.org  Mon Feb 10 20:48:51 2014
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the SIP-TO-XMPP Working Group of the IETF.

        Title           : Interworking between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP): Architecture, Addresses, and Error Handling
        Authors         : Peter Saint-Andre
                          Avshalom Houri
                          Joe Hildebrand
	Filename        : draft-ietf-stox-core-10.txt
	Pages           : 23
	Date            : 2014-02-10

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-10

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http://www.ietf.org/rfcdiff?url2=draft-ietf-stox-core-10


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Some adjustments to address IESG feedback.

On 2/10/14, 9:48 PM, internet-drafts@ietf.org wrote:
>
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>   This draft is a work item of the SIP-TO-XMPP Working Group of the IETF.
>
>          Title           : Interworking between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP): Architecture, Addresses, and Error Handling
>          Authors         : Peter Saint-Andre
>                            Avshalom Houri
>                            Joe Hildebrand
> 	Filename        : draft-ietf-stox-core-10.txt
> 	Pages           : 23
> 	Date            : 2014-02-10
>
> 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-10
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-stox-core-10
>
>
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox
>


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Subject: [Stox] Protocol Action: 'Interworking between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP): Presence' to Proposed Standard (draft-ietf-stox-presence-09.txt)
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The IESG has approved the following document:
- 'Interworking between the Session Initiation Protocol (SIP) and the
   Extensible Messaging and Presence Protocol (XMPP): Presence'
  (draft-ietf-stox-presence-09.txt) as Proposed Standard

This document is the product of the SIP-TO-XMPP Working Group.

The IESG contact persons are Gonzalo Camarillo and Richard Barnes.

A URL of this Internet Draft is:
http://datatracker.ietf.org/doc/draft-ietf-stox-presence/




Technical Summary

  Relevant content can frequently be found in the abstract 
  and/or introduction of the document. If not, this may be 
  an indication that there are deficiencies in the abstract 
  or introduction.

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).

Working Group Summary

  Was there anything in WG process that is worth noting? For 
  example, was there controversy about particular points or 
  were there decisions where the consensus was particularly 
  rough?

This document has been initially reviewed in the DISPATCH WG as per the RAI area process for new work. The STOX WG has been chartered as a result of the submission of this and other documents that aim to define specifications for interworking between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP) based systems. This document has been subjected to a number of thorough reviews initially in the DISPATCH WG and in the STOX WG after its creation. There were no controversial points regarding it.

Document Quality

  Are there existing implementations of the protocol? Have a 
  significant number of vendors indicated their plan to 
  implement the specification? Are there any reviewers that 
  merit special mention as having done a thorough review, 
  e.g., one that resulted in important changes or a 
  conclusion that the document had no substantive issues? If 
  there was a MIB Doctor, Media Type or other expert review, 
  what was its course (briefly)? In the case of a Media Type 
  review, on what date was the request posted?

There are already several vendors having full or partial implementations of the specification, among which Jabber/Cisco, Kamailio (OpenSER), AG Projects (Silk Server). Some of the people behind these implementations have actively participated in the discussions in the WG.

Personnel

  Who is the Document Shepherd? Who is the Responsible Area
  Director?

Yana Stamcheva is the document shepherd. The responsible area director is Gonzalo Camarillo.


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On 2/10/14, 3:52 PM, Peter Saint-Andre wrote:
> On 2/10/14, 3:20 PM, Stephen Farrell wrote:
>>
>> Hi Peter,
>>
>> On 02/10/2014 06:24 PM, Peter Saint-Andre wrote:
>>> ( + STOX WG since one of the issues might deserve broader discussion )
>>>
>>> Hi Stephen, thanks for your review:
>>>
>>> https://datatracker.ietf.org/doc/draft-ietf-stox-core/ballot/
>>>
>>> You wrote in part:
>>>
>>>     - I wondered why you didn't just say that (D)TLS SHOULD
>>>     be used/supported between gateways. Given that all the
>>>     relevant bits of code are likely to support that, wouldn't
>>>     it be a good thing?
>>>
>>> Yes, that seems eminently reasonable.
>>
>> Just to be clear - the above is the only DISCUSS point.
>
> Yep.
>
>>>     - Has anyone thought about confusability in the name
>>>     mappings? I expected to see a bit of text in the
>>>     security considerations but didn't see it.
>>>
>>> Confusion is always possible. :-) Were you thinking about confusable
>>> characters from Unicode, or something else?
>>
>> Right. Maybe its all there already and I just didn't see
>> if in the mapping stuff, or maybe not. I'm not sure.
>
> Section 10.5 of draft-ietf-precis-framework has some very nice text
> about visually similar characters. We could borrow from that.
>
>>>     - It seems a shame to not be able to gateway when the To is
>>>     a sips URI at all but I understand that some loss of
>>>     security is inevitable for cases like this. Is there any
>>>     work planned for an update that would allow gatewaying for
>>>     such cases, e.g. if the 1st XMPP server is the one to which
>>>     the user is connected and the user is connected using
>>>     XMPP/TLS?
>>>
>>> Hmm. I cannot say that I am aware of planning for updates to provide
>>> more secure gatewaying, although folks active in the STOX WG might be
>>> thinking along those lines.
>>>
>>> Depending on the deployment architecture, I think there are cases where
>>> it is *possible* to TLS-protect all the hops. For instance, if
>>> sip.example.org has a direct server-to-server connection to
>>> xmpp.example.com (no intermediate hops) and both organizations agree to
>>> force the use of TLS for client connections (e.g., via SLA), then I
>>> suppose that sip.example.com could honor 'sips' URIs when sending
>>> traffic to xmpp.example.com. However, such an arrangement is rare enough
>>> right now that I don't know if it is worth mentioning.
>>
>> Fair enough. I can see that detecting that you (probably) have
>> all (D)TLS hops might be tricky and error prone.
>
> That's how I see it.
>
> Oh, and we at least need to mention DTLS for the UDP cases, though (see
> Section 5 of draft-ietf-stox-core).

Hi Stephen,

We've posted an updated version:

https://datatracker.ietf.org/doc/draft-ietf-stox-core/

http://tools.ietf.org/rfcdiff?url2=draft-ietf-stox-core-10.txt

We haven't yet added any text about visually similar characters, though. 
Let me know what you think of the rest.

Peter

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


From stpeter@stpeter.im  Tue Feb 11 07:51:39 2014
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On 2/10/14, 5:11 PM, Peter Saint-Andre wrote:
> On 2/10/14, 3:08 PM, Peter Saint-Andre wrote:
>> Pete, you wrote:
>>
>>    Section 4 is (nicely) clear on the document architecturally
>>    describing a gateway. However, traditionally a gateway is
>>    transparent to the entity that communicates with it: When we
>>    had SMTP-to-X.400 gateways, the gateway appeared as just another
>>    SMTP system that noticed special qualities of the address and
>>    then routed the messages appropriately.
>>
>> Your "traditionally" sounds like an SMTP phenomenon. In XMPP, we don't
>> have intermediate transfer agents, and XMPP servers are designed to use
>> add-on modules for additional functionality. So in the XMPP universe it
>> makes sense for the operator of an XMPP service to install a module (in
>> addition to the XMPP server) that performs the gateway function (we
>> often call this a connection manager).
>>
>>    Section 5 describes
>>    something a bit different. It's not clear that what section 5
>>    describes actually is part of the gateway, but rather sounds
>>    like a combined server which does failover between the
>>    protocols. I don't think this is a showstopper, but it might
>>    help implementers significantly if you described in section 5
>>    *where* in the model this function occurs. Right now, it
>>    sounds like the server itself does the addressing failover,
>>    not the gateway.
>>
>> Yes, I see your point. This kind of thing is quite likely implementation
>> specific (e.g., when the add-on XMPP-to-SIP gateway module gets
>> configured into and trusted by the core XMPP server, it might get added
>> into an event listener for core stanza delivery if an XMPP lookup fails
>> for the remote domain). Let me look at what text might be useful here.
>
> How is this for proposed text?
>
>     Existing SIP and XMPP server implementations do not typically include
>     the ability to communicate using the other technology (XMPP for SIP
>     implementations, SIP for XMPP implementations).  One common
>     architectural pattern is to associate a gateway with the core server
>     implementation (e.g., in XMPP such a gateway might be called a
>     "connection manager").  How exactly such a gateway interacts with the
>     core server to complete tasks such as address lookups and
>     communication with systems that use the other technology is a matter
>     of implementation (e.g., the gateway might be an add-on module that
>     is trusted by the core server to act as a fallback delivery mechanism
>     if the remote domain does not support the server's native
>     communication technology).

Pete, that text is now in draft-ietf-stox-core-10.

Peter

-- 
Peter Saint-Andre
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From stephen.farrell@cs.tcd.ie  Tue Feb 11 07:54:57 2014
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On 02/11/2014 03:49 PM, Peter Saint-Andre wrote:
> 
> Hi Stephen,
> 
> We've posted an updated version:
> 
> https://datatracker.ietf.org/doc/draft-ietf-stox-core/
> 
> http://tools.ietf.org/rfcdiff?url2=draft-ietf-stox-core-10.txt
> 
> We haven't yet added any text about visually similar characters, though.
> Let me know what you think of the rest.

Its lovely:-)

I've cleared the discuss, happy to chat more about
confusability stuff though I'm pretty sure you and
the WG know lots more about that than I do so I'll
be almost certainly be fine with whatever you think
is right. (Incl. if doing nothing is right.)

Thanks,
S.


From stpeter@stpeter.im  Tue Feb 11 08:05:10 2014
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On 2/11/14, 8:54 AM, Stephen Farrell wrote:
>
>
> On 02/11/2014 03:49 PM, Peter Saint-Andre wrote:
>>
>> Hi Stephen,
>>
>> We've posted an updated version:
>>
>> https://datatracker.ietf.org/doc/draft-ietf-stox-core/
>>
>> http://tools.ietf.org/rfcdiff?url2=draft-ietf-stox-core-10.txt
>>
>> We haven't yet added any text about visually similar characters, though.
>> Let me know what you think of the rest.
>
> Its lovely:-)
>
> I've cleared the discuss, happy to chat more about
> confusability stuff though I'm pretty sure you and
> the WG know lots more about that than I do so I'll
> be almost certainly be fine with whatever you think
> is right. (Incl. if doing nothing is right.)

Having just re-read Section 10.5 of draft-ietf-precis-framework, I think 
we might point to that for a helpful treatment of the issues.

Peter

-- 
Peter Saint-Andre
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From stephen.farrell@cs.tcd.ie  Tue Feb 11 08:06:18 2014
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On 02/11/2014 04:05 PM, Peter Saint-Andre wrote:
>>
> 
> Having just re-read Section 10.5 of draft-ietf-precis-framework, I think
> we might point to that for a helpful treatment of the issues.

WFM,
Thanks,
S.


From presnick@qti.qualcomm.com  Tue Feb 11 08:43:41 2014
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Subject: Re: [Stox] Pete Resnick's IESG feedback on draft-ietf-stox-core
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That's fine.

pr

On 2/11/14 9:51 AM, Peter Saint-Andre wrote:
> On 2/10/14, 5:11 PM, Peter Saint-Andre wrote:
>> On 2/10/14, 3:08 PM, Peter Saint-Andre wrote:
>>> Pete, you wrote:
>>>
>>>    Section 4 is (nicely) clear on the document architecturally
>>>    describing a gateway. However, traditionally a gateway is
>>>    transparent to the entity that communicates with it: When we
>>>    had SMTP-to-X.400 gateways, the gateway appeared as just another
>>>    SMTP system that noticed special qualities of the address and
>>>    then routed the messages appropriately.
>>>
>>> Your "traditionally" sounds like an SMTP phenomenon. In XMPP, we don't
>>> have intermediate transfer agents, and XMPP servers are designed to use
>>> add-on modules for additional functionality. So in the XMPP universe it
>>> makes sense for the operator of an XMPP service to install a module (in
>>> addition to the XMPP server) that performs the gateway function (we
>>> often call this a connection manager).
>>>
>>>    Section 5 describes
>>>    something a bit different. It's not clear that what section 5
>>>    describes actually is part of the gateway, but rather sounds
>>>    like a combined server which does failover between the
>>>    protocols. I don't think this is a showstopper, but it might
>>>    help implementers significantly if you described in section 5
>>>    *where* in the model this function occurs. Right now, it
>>>    sounds like the server itself does the addressing failover,
>>>    not the gateway.
>>>
>>> Yes, I see your point. This kind of thing is quite likely 
>>> implementation
>>> specific (e.g., when the add-on XMPP-to-SIP gateway module gets
>>> configured into and trusted by the core XMPP server, it might get added
>>> into an event listener for core stanza delivery if an XMPP lookup fails
>>> for the remote domain). Let me look at what text might be useful here.
>>
>> How is this for proposed text?
>>
>>     Existing SIP and XMPP server implementations do not typically 
>> include
>>     the ability to communicate using the other technology (XMPP for SIP
>>     implementations, SIP for XMPP implementations).  One common
>>     architectural pattern is to associate a gateway with the core server
>>     implementation (e.g., in XMPP such a gateway might be called a
>>     "connection manager").  How exactly such a gateway interacts with 
>> the
>>     core server to complete tasks such as address lookups and
>>     communication with systems that use the other technology is a matter
>>     of implementation (e.g., the gateway might be an add-on module that
>>     is trusted by the core server to act as a fallback delivery 
>> mechanism
>>     if the remote domain does not support the server's native
>>     communication technology).
>
> Pete, that text is now in draft-ietf-stox-core-10.
>
> Peter
>

-- 
Pete Resnick<http://www.qualcomm.com/~presnick/>
Qualcomm Technologies, Inc. - +1 (858)651-4478


From stpeter@stpeter.im  Tue Feb 11 10:33:18 2014
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Subject: Re: [Stox] Stephen Farrell's IESG feedback on draft-ietf-stox-core
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On 2/11/14, 9:06 AM, Stephen Farrell wrote:
>
>
> On 02/11/2014 04:05 PM, Peter Saint-Andre wrote:
>>>
>>
>> Having just re-read Section 10.5 of draft-ietf-precis-framework, I think
>> we might point to that for a helpful treatment of the issues.
>
> WFM,
> Thanks,
> S.

OK, I'll quickly spin a new version that contains the pointer.

Peter

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


From internet-drafts@ietf.org  Tue Feb 11 10:38:35 2014
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Subject: [Stox] I-D Action: draft-ietf-stox-core-11.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the SIP-TO-XMPP Working Group of the IETF.

        Title           : Interworking between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP): Architecture, Addresses, and Error Handling
        Authors         : Peter Saint-Andre
                          Avshalom Houri
                          Joe Hildebrand
	Filename        : draft-ietf-stox-core-11.txt
	Pages           : 23
	Date            : 2014-02-11

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-11

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-stox-core-11


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From internet-drafts@ietf.org  Thu Feb 13 06:32:30 2014
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Subject: [Stox] I-D Action: draft-ietf-stox-media-03.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the SIP-TO-XMPP Working Group of the IETF.

        Title           : Interworking between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP): Media Sessions
        Authors         : Peter Saint-Andre
                          Saul Ibarra
                          Emil Ivov
	Filename        : draft-ietf-stox-media-03.txt
	Pages           : 22
	Date            : 2014-02-13

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/

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From stpeter@stpeter.im  Thu Feb 13 07:37:53 2014
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This version addresses the feedback from Philipp Hancke. We still need 
to follow up on the feedback from Jonathan Lennox and Paul Kyzivat, add 
more examples, etc. We will do that over the next few weeks, and submit 
the next version soon after IETF 89.

Peter

On 2/13/14, 7:32 AM, internet-drafts@ietf.org wrote:
>
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>   This draft is a work item of the SIP-TO-XMPP Working Group of the IETF.
>
>          Title           : Interworking between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP): Media Sessions
>          Authors         : Peter Saint-Andre
>                            Saul Ibarra
>                            Emil Ivov
> 	Filename        : draft-ietf-stox-media-03.txt
> 	Pages           : 22
> 	Date            : 2014-02-13
>
> 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-03
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-stox-media-03
>
>
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox
>


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


From nobody Wed Feb 26 03:59:18 2014
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Subject: [Stox] Protocol Action: 'Interworking between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP): Architecture, Addresses, and Error Handling' to Proposed Standard (draft-ietf-stox-core-11.txt)
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The IESG has approved the following document:
- 'Interworking between the Session Initiation Protocol (SIP) and the
   Extensible Messaging and Presence Protocol (XMPP): Architecture,
   Addresses, and Error Handling'
  (draft-ietf-stox-core-11.txt) as Proposed Standard

This document is the product of the SIP-TO-XMPP Working Group.

The IESG contact persons are Gonzalo Camarillo and Richard Barnes.

A URL of this Internet Draft is:
http://datatracker.ietf.org/doc/draft-ietf-stox-core/




Technical Summary

  Relevant content can frequently be found in the abstract 
  and/or introduction of the document. If not, this may be 
  an indication that there are deficiencies in the abstract 
  or introduction.

This document specifies the fondation underlying the bidirectional protocol mappings between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP). Because of the wide deployment of both technologies it is important to clearly define mappings between them for the sake of interworking. This document inaugurates a series of SIP-XMPP interworking specifications by defining the architectural assumptions underlying such mappings as well as the mapping of addresses and error conditions.

Working Group Summary

  Was there anything in WG process that is worth noting? For 
  example, was there controversy about particular points or 
  were there decisions where the consensus was particularly 
  rough?

This document has been initially reviewed in the DISPATCH WG as per the RAI area process for new work. The STOX WG has been chartered as a result of the submission of this and other documents that aim to define specifications for interworking between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP) based systems. This document has been subjected to a number of thorough reviews initially in the DISPATCH WG and in the STOX WG after its creation. There were no controversial points regarding it.

Document Quality

  Are there existing implementations of the protocol? Have a 
  significant number of vendors indicated their plan to 
  implement the specification? Are there any reviewers that 
  merit special mention as having done a thorough review, 
  e.g., one that resulted in important changes or a 
  conclusion that the document had no substantive issues? If 
  there was a MIB Doctor, Media Type or other expert review, 
  what was its course (briefly)? In the case of a Media Type 
  review, on what date was the request posted?

There are already several vendors having full or partial implementations of the specification, among which Jabber/Cisco, Kamailio (OpenSER), AG Projects (Silk Server). Some of the people behind these implementations have actively participated in the discussions in the WG.

Personnel

  Who is the Document Shepherd? Who is the Responsible Area
  Director?

Yana Stamcheva is the document shepherd. The responsible area director is Gonzalo Camarillo.

