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Subject: Re: [Stox] stox-presence-03: a couple of comments
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On 8/26/13 4:49 PM, Peter Saint-Andre wrote:
> On 8/26/13 6:34 AM, Markus.Isomaki@nokia.com wrote:
>> Hi,
> 
> Hi Markus, thanks for the feedback.
> 
>> I have a couple of comments on the presence-03 draft:
>> 
>> * Some asymmetry between XMPP-to-SIP (3.2) and SIP-to-XMPP (3.3) 
>> directions
>> 
>> - 3.2.1 explains the DNS operations an XMPP domain uses to
>> determine that the foreign domain does not support XMPP but
>> actually supports SIP/SIMPLE. Presumably something similar is
>> needed in the reverse direction, but this is not explained in
>> 3.3.1. - The case where the subscription is rejected is covered
>> in 3.3.1 but not in 3.2.1.
> 
> Good point. We'll add something about that.

Actually, I think draft-ietf-stox-core already covers that:

   When an XMPP service receives an XMPP stanza whose 'to' address
   specifies or includes a domain other than the domain of the XMPP
   service, it needs to determine whether the destination domain offers
   an XMPP service or a SIP service.  To do so, it performs one or more
   DNS SRV lookups [RFC2782] for "_xmpp-server" records as specified in
   [RFC6120].  If the response returns a hostname, the service can
   attempt XMPP communication.  If not, the service can attempt to
   locate a SIP service for that domain using the procedures specified
   in [RFC3263].

   Similarly, when a SIP service receives a SIP message whose Request-
   URI specifies or includes a domain other than the domain of the SIP
   service, it needs to determine whether the destination domain offers
   a SIP service or an XMPP service.  To do so, it uses the procedures
   specified in [RFC3263].  If that response returns a hostname, the
   service can attempt SIP communication.  If not, the service can
   perform one or more DNS SRV lookups [RFC2782] for "_xmpp-server"
   records as specified in [RFC6120].

IIRC, we planned to remove such text from all but the core document,
since it applies to presence, im, chat, groupchat, media, etc.

Peter

- -- 
Peter Saint-Andre
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Subject: [Stox] I-D Action: draft-ietf-stox-presence-04.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): Presence
	Author(s)       : Peter Saint-Andre
                          Avshalom Houri
                          Joe Hildebrand
	Filename        : draft-ietf-stox-presence-04.txt
	Pages           : 24
	Date            : 2013-09-05

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-04

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-stox-presence-04


Please note that it may take a couple of minutes from the time of submission
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On 8/26/13 8:59 PM, Peter Saint-Andre wrote:
> Hej Olle, thanks for the review.

Thanks again. I have one follow-up comment inline.

> On 8/22/13 2:10 AM, Olle E. Johansson wrote:

<snip/>

>> I would also like to mention that Contacts can include display
>> names, like "Olles superduperphone" or "Conference Room Jupiter
>> Phone". I don't know if this can be translated into XMPP.
> 
> XMPP doesn't have display names in addresses (as email and SIP
> do), although those could be translated into a nickname / handle
> using XEP-0172:
> 
> http://xmpp.org/extensions/xep-0172.html
> 
> So adding a note about that seems reasonable.

I've looked into this further. The Contact header is specified on SIP
INVITES, whereas the Nickname extension is specified on XMPP messages
and presence. I don't think there's much if any overlap there, so it
doesn't seem useful to specify a mapping in draft-ietf-stox-core.
However, I'm open to further discussion on this matter.

I've incoporated all the rest of your feedback and will submit a
revised I-D now.

Peter

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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): Architecture,=
 Addresses, and Error Handling
	Author(s)       : Peter Saint-Andre
                          Avshalom Houri
                          Joe Hildebrand
	Filename        : draft-ietf-stox-core-04.txt
	Pages           : 18
	Date            : 2013-09-05

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-04

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-stox-core-04


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until the htmlized version and diff are available at tools.ietf.org.

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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-02.txt
	Pages           : 11
	Date            : 2013-09-05

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

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http://tools.ietf.org/html/draft-ietf-stox-im-02

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From stpeter@stpeter.im  Thu Sep  5 19:59:48 2013
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Folks, as far as I can tell I've addressed all feedback sent so far on
the -core, -presence, and -im documents. I'll try to do the same for
the -chat document by the end of next week. I'll also talk with Saúl
about setting a schedule for the -groupchat document (perhaps also by
the end of next week).

Peter

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


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From Markus.Isomaki@nokia.com  Fri Sep  6 06:35:38 2013
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From: <Markus.Isomaki@nokia.com>
To: <stpeter@stpeter.im>, <stox@ietf.org>
Thread-Topic: [Stox] current status
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Hi,

There have been a fair number of comments on -core, -im and -presence since=
 the Berlin IETF. Please take a look at the new versions
http://tools.ietf.org/id/draft-ietf-stox-core-04.txt
http://tools.ietf.org/id/draft-ietf-stox-im-02.txt
http://tools.ietf.org/id/draft-ietf-stox-presence-04.txt
and see if your issues have been properly addressed, and/or whether you wan=
t to still raise any new issues.

If nothing specific comes up, the chairs will consider starting WG LC on th=
ese documents in the near future.=20

One thing that I wonder about -core is whether it is a good idea to stabili=
ze or last call it yet at this point, since there have been some issues rai=
sed about -media that might have impact to -core, such as dealing with SIP =
side forking or loop detection. So is it better to wait with -core until -m=
edia is more stable?

But let's move forward with -im and -presence unless there are issues left.=
=20

Markus=20

> -----Original Message-----
> From: stox-bounces@ietf.org [mailto:stox-bounces@ietf.org] On Behalf Of
> ext Peter Saint-Andre
> Sent: 06 September, 2013 06:00
> To: stox@ietf.org
> Subject: [Stox] current status
>=20
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>=20
> Folks, as far as I can tell I've addressed all feedback sent so far on th=
e -core, -
> presence, and -im documents. I'll try to do the same for the -chat docume=
nt
> by the end of next week. I'll also talk with Sa=FAl about setting a sched=
ule for
> the -groupchat document (perhaps also by the end of next week).
>=20
> Peter
>=20
> - --
> Peter Saint-Andre
> https://stpeter.im/
>=20
>=20
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From Markus.Isomaki@nokia.com  Fri Sep  6 06:36:53 2013
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From: <Markus.Isomaki@nokia.com>
To: <stpeter@stpeter.im>
Thread-Topic: [Stox] stox-presence-03: a couple of comments
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Subject: Re: [Stox] stox-presence-03: a couple of comments
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Hi,

OK. I do recall too that the intent was to put all DNS handling to -core an=
d that has now happened. Thanks, it makes things clearer as there is less o=
verlap between the documents.

Markus=20

> -----Original Message-----
> From: ext Peter Saint-Andre [mailto:stpeter@stpeter.im]
> Sent: 06 September, 2013 04:12
> To: Isomaki Markus (Nokia-CIC/Espoo)
> Cc: stox@ietf.org
> Subject: Re: [Stox] stox-presence-03: a couple of comments
>=20
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>=20
> On 8/26/13 4:49 PM, Peter Saint-Andre wrote:
> > On 8/26/13 6:34 AM, Markus.Isomaki@nokia.com wrote:
> >> Hi,
> >
> > Hi Markus, thanks for the feedback.
> >
> >> I have a couple of comments on the presence-03 draft:
> >>
> >> * Some asymmetry between XMPP-to-SIP (3.2) and SIP-to-XMPP (3.3)
> >> directions
> >>
> >> - 3.2.1 explains the DNS operations an XMPP domain uses to determine
> >> that the foreign domain does not support XMPP but actually supports
> >> SIP/SIMPLE. Presumably something similar is needed in the reverse
> >> direction, but this is not explained in 3.3.1. - The case where the
> >> subscription is rejected is covered in 3.3.1 but not in 3.2.1.
> >
> > Good point. We'll add something about that.
>=20
> Actually, I think draft-ietf-stox-core already covers that:
>=20
>    When an XMPP service receives an XMPP stanza whose 'to' address
>    specifies or includes a domain other than the domain of the XMPP
>    service, it needs to determine whether the destination domain offers
>    an XMPP service or a SIP service.  To do so, it performs one or more
>    DNS SRV lookups [RFC2782] for "_xmpp-server" records as specified in
>    [RFC6120].  If the response returns a hostname, the service can
>    attempt XMPP communication.  If not, the service can attempt to
>    locate a SIP service for that domain using the procedures specified
>    in [RFC3263].
>=20
>    Similarly, when a SIP service receives a SIP message whose Request-
>    URI specifies or includes a domain other than the domain of the SIP
>    service, it needs to determine whether the destination domain offers
>    a SIP service or an XMPP service.  To do so, it uses the procedures
>    specified in [RFC3263].  If that response returns a hostname, the
>    service can attempt SIP communication.  If not, the service can
>    perform one or more DNS SRV lookups [RFC2782] for "_xmpp-server"
>    records as specified in [RFC6120].
>=20
> IIRC, we planned to remove such text from all but the core document, sinc=
e
> it applies to presence, im, chat, groupchat, media, etc.
>=20
> Peter
>=20
> - --
> Peter Saint-Andre
> https://stpeter.im/
>=20
>=20
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Subject: Re: [Stox] stox-presence-03: a couple of comments
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On 9/6/13 7:26 AM, Markus.Isomaki@nokia.com wrote:
> Hi,
> 
> OK. I do recall too that the intent was to put all DNS handling to
> -core and that has now happened. Thanks, it makes things clearer as
> there is less overlap between the documents.

I just realized that last night I pushed out a corrected version of
the -im spec with DNS text still in there. I'll submit a revised I-D.

Peter

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


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From saul@ag-projects.com  Fri Sep  6 09:17:00 2013
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On Sep 6, 2013, at 3:25 PM, <Markus.Isomaki@nokia.com> =
<Markus.Isomaki@nokia.com> wrote:

> Hi,
>=20
> There have been a fair number of comments on -core, -im and -presence =
since the Berlin IETF. Please take a look at the new versions
> http://tools.ietf.org/id/draft-ietf-stox-core-04.txt
> http://tools.ietf.org/id/draft-ietf-stox-im-02.txt
> http://tools.ietf.org/id/draft-ietf-stox-presence-04.txt
> and see if your issues have been properly addressed, and/or whether =
you want to still raise any new issues.
>=20
> If nothing specific comes up, the chairs will consider starting WG LC =
on these documents in the near future.=20
>=20
> One thing that I wonder about -core is whether it is a good idea to =
stabilize or last call it yet at this point, since there have been some =
issues raised about -media that might have impact to -core, such as =
dealing with SIP side forking or loop detection. So is it better to wait =
with -core until -media is more stable?
>=20

At some point we agreed that those issues would be handled in -media, =
because in other cases (chat, presence, im) the gateway is itself the =
endpoint and handles all the "media", so those didn't seem to apply.

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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On 9/6/13 10:16 AM, Saúl Ibarra Corretgé wrote:
> 
> On Sep 6, 2013, at 3:25 PM, <Markus.Isomaki@nokia.com> 
> <Markus.Isomaki@nokia.com> wrote:
> 
>> Hi,
>> 
>> There have been a fair number of comments on -core, -im and 
>> -presence since the Berlin IETF. Please take a look at the new 
>> versions http://tools.ietf.org/id/draft-ietf-stox-core-04.txt 
>> http://tools.ietf.org/id/draft-ietf-stox-im-02.txt 
>> http://tools.ietf.org/id/draft-ietf-stox-presence-04.txt and see
>> if your issues have been properly addressed, and/or whether you
>> want to still raise any new issues.
>> 
>> If nothing specific comes up, the chairs will consider starting
>> WG LC on these documents in the near future.
>> 
>> One thing that I wonder about -core is whether it is a good idea
>> to stabilize or last call it yet at this point, since there have
>> been some issues raised about -media that might have impact to
>> -core, such as dealing with SIP side forking or loop detection.
>> So is it better to wait with -core until -media is more stable?
>> 
> 
> At some point we agreed that those issues would be handled in
> -media, because in other cases (chat, presence, im) the gateway is
> itself the endpoint and handles all the "media", so those didn't
> seem to apply.

That is my recollection, too.

Thus I'd be in favor of stabilizing the -core spec, and hanlding the
forking issues in the -media spec.

Peter

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


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On Fri, Sep 6, 2013 at 11:35 AM, Peter Saint-Andre <stpeter@stpeter.im>wrot=
e:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> On 9/6/13 10:16 AM, Sa=FAl Ibarra Corretg=E9 wrote:
> >
> > On Sep 6, 2013, at 3:25 PM, <Markus.Isomaki@nokia.com>
> > <Markus.Isomaki@nokia.com> wrote:
> >
> >> Hi,
> >>
> >> There have been a fair number of comments on -core, -im and
> >> -presence since the Berlin IETF. Please take a look at the new
> >> versions http://tools.ietf.org/id/draft-ietf-stox-core-04.txt
> >> http://tools.ietf.org/id/draft-ietf-stox-im-02.txt
> >> http://tools.ietf.org/id/draft-ietf-stox-presence-04.txt and see
> >> if your issues have been properly addressed, and/or whether you
> >> want to still raise any new issues.
> >>
> >> If nothing specific comes up, the chairs will consider starting
> >> WG LC on these documents in the near future.
> >>
> >> One thing that I wonder about -core is whether it is a good idea
> >> to stabilize or last call it yet at this point, since there have
> >> been some issues raised about -media that might have impact to
> >> -core, such as dealing with SIP side forking or loop detection.
> >> So is it better to wait with -core until -media is more stable?
> >>
> >
> > At some point we agreed that those issues would be handled in
> > -media, because in other cases (chat, presence, im) the gateway is
> > itself the endpoint and handles all the "media", so those didn't
> > seem to apply.
>
> That is my recollection, too.
>
 [MB] That was also what I thought when I reviewed -core. [/MB]

Thus I'd be in favor of stabilizing the -core spec, and hanlding the
> forking issues in the -media spec.
>
[MB] I agree. [/MB]

>
> Peter
>
> - --
> Peter Saint-Andre
> https://stpeter.im/
>
>
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Sep 6, 2013 at 11:35 AM, Peter Saint-Andre <span dir=3D"ltr=
">&lt;<a href=3D"mailto:stpeter@stpeter.im" target=3D"_blank">stpeter@stpet=
er.im</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im">-----BEGIN PGP SIGNED MESS=
AGE-----<br>
Hash: SHA1<br>
<br>
</div><div class=3D"im">On 9/6/13 10:16 AM, Sa=FAl Ibarra Corretg=E9 wrote:=
<br>
&gt;<br>
&gt; On Sep 6, 2013, at 3:25 PM, &lt;<a href=3D"mailto:Markus.Isomaki@nokia=
.com">Markus.Isomaki@nokia.com</a>&gt;<br>
&gt; &lt;<a href=3D"mailto:Markus.Isomaki@nokia.com">Markus.Isomaki@nokia.c=
om</a>&gt; wrote:<br>
&gt;<br>
&gt;&gt; Hi,<br>
&gt;&gt;<br>
&gt;&gt; There have been a fair number of comments on -core, -im and<br>
&gt;&gt; -presence since the Berlin IETF. Please take a look at the new<br>
&gt;&gt; versions <a href=3D"http://tools.ietf.org/id/draft-ietf-stox-core-=
04.txt" target=3D"_blank">http://tools.ietf.org/id/draft-ietf-stox-core-04.=
txt</a><br>
&gt;&gt; <a href=3D"http://tools.ietf.org/id/draft-ietf-stox-im-02.txt" tar=
get=3D"_blank">http://tools.ietf.org/id/draft-ietf-stox-im-02.txt</a><br>
&gt;&gt; <a href=3D"http://tools.ietf.org/id/draft-ietf-stox-presence-04.tx=
t" target=3D"_blank">http://tools.ietf.org/id/draft-ietf-stox-presence-04.t=
xt</a> and see<br>
&gt;&gt; if your issues have been properly addressed, and/or whether you<br=
>
&gt;&gt; want to still raise any new issues.<br>
&gt;&gt;<br>
&gt;&gt; If nothing specific comes up, the chairs will consider starting<br=
>
&gt;&gt; WG LC on these documents in the near future.<br>
&gt;&gt;<br>
&gt;&gt; One thing that I wonder about -core is whether it is a good idea<b=
r>
&gt;&gt; to stabilize or last call it yet at this point, since there have<b=
r>
&gt;&gt; been some issues raised about -media that might have impact to<br>
&gt;&gt; -core, such as dealing with SIP side forking or loop detection.<br=
>
&gt;&gt; So is it better to wait with -core until -media is more stable?<br=
>
&gt;&gt;<br>
&gt;<br>
&gt; At some point we agreed that those issues would be handled in<br>
&gt; -media, because in other cases (chat, presence, im) the gateway is<br>
&gt; itself the endpoint and handles all the &quot;media&quot;, so those di=
dn&#39;t<br>
&gt; seem to apply.<br>
<br>
</div>That is my recollection, too.<br></blockquote><div>=A0[MB] That was a=
lso what I thought when I reviewed -core. [/MB]</div><div><br></div><blockq=
uote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc =
solid;padding-left:1ex">

Thus I&#39;d be in favor of stabilizing the -core spec, and hanlding the<br=
>
forking issues in the -media spec.<br></blockquote><div>[MB] I agree. [/MB]=
</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-l=
eft:1px #ccc solid;padding-left:1ex">
<div class=3D"im"><br>
Peter<br>
<br>
- --<br>
Peter Saint-Andre<br>
<a href=3D"https://stpeter.im/" target=3D"_blank">https://stpeter.im/</a><b=
r>
<br>
<br>
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-----END PGP SIGNATURE-----<br>
<div class=3D"HOEnZb"><div class=3D"h5">___________________________________=
____________<br>
stox mailing list<br>
<a href=3D"mailto:stox@ietf.org">stox@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/stox" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/stox</a><br>
</div></div></blockquote></div><br></div></div>

--001a11c2d6e6ac5dfb04e5b9b9e0--

From internet-drafts@ietf.org  Fri Sep  6 11:23:59 2013
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Subject: [Stox] I-D Action: draft-ietf-stox-im-03.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): Instant Messa=
ging
	Author(s)       : Peter Saint-Andre
                          Avshalom Houri
                          Joe Hildebrand
	Filename        : draft-ietf-stox-im-03.txt
	Pages           : 10
	Date            : 2013-09-06

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-03

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


From ag@ag-projects.com  Fri Sep 13 05:42:38 2013
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From: Adrian Georgescu <ag@ag-projects.com>
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To: "stox@ietf.org" <stox@ietf.org>
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Subject: [Stox] OTR encryption between SIP and XMPP
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Hello,

I am not sure if this is relevant to any of the deliverables of the =
working group, but just in case it is, we were able to create end-to-end =
encryption for chat sessions using OTR protocol =
http://www.cypherpunks.ca/otr/  between a SIP client using MSRP (that is =
Blink) and XMPP client (Jitsi).

The implementation is described here:

http://projects.ag-projects.com/projects/blinkc/wiki/OTR

Regards,
Adrian




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From Markus.Isomaki@nokia.com  Wed Sep 18 04:17:43 2013
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Hi,

There have been no further comments on this, so I think we can declare it d=
ecided that we won't expand the scope of -core with topics like forking, bu=
t leave that to the -media to handle.

My proposed plan for the next steps of the WG is:

-          Before Vancouver: Issue WG Last Calls on -core, -im, and -presen=
ce and see if they indeed are mature enough to be delivered forward.

-          For Vancouver: Get updated versions of especially -media, and po=
ssibly -chat and -groupchat, and collect the open issues about them.

-          In Vancouver: Discuss the open issues about -media, and if neede=
d -chat and -groupchat.

-          After Vancouver: Depending on the maturity and issues raised, sc=
hedule WG Last Calls for the remaining deliverables.

Let the chairs (Yana and me) know if you have issues with this plan or even=
 better, suggestions how to improve it.

Markus


From: ext Mary Barnes [mailto:mary.ietf.barnes@gmail.com]
Sent: 06 September, 2013 19:43
To: Peter Saint-Andre
Cc: Sa=FAl Ibarra Corretg=E9; stox@ietf.org; Isomaki Markus (Nokia-CIC/Espo=
o)
Subject: Re: [Stox] current status



On Fri, Sep 6, 2013 at 11:35 AM, Peter Saint-Andre <stpeter@stpeter.im<mail=
to:stpeter@stpeter.im>> wrote:
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
On 9/6/13 10:16 AM, Sa=FAl Ibarra Corretg=E9 wrote:
>
> On Sep 6, 2013, at 3:25 PM, <Markus.Isomaki@nokia.com<mailto:Markus.Isoma=
ki@nokia.com>>
> <Markus.Isomaki@nokia.com<mailto:Markus.Isomaki@nokia.com>> wrote:
>
>> Hi,
>>
>> There have been a fair number of comments on -core, -im and
>> -presence since the Berlin IETF. Please take a look at the new
>> versions http://tools.ietf.org/id/draft-ietf-stox-core-04.txt
>> http://tools.ietf.org/id/draft-ietf-stox-im-02.txt
>> http://tools.ietf.org/id/draft-ietf-stox-presence-04.txt and see
>> if your issues have been properly addressed, and/or whether you
>> want to still raise any new issues.
>>
>> If nothing specific comes up, the chairs will consider starting
>> WG LC on these documents in the near future.
>>
>> One thing that I wonder about -core is whether it is a good idea
>> to stabilize or last call it yet at this point, since there have
>> been some issues raised about -media that might have impact to
>> -core, such as dealing with SIP side forking or loop detection.
>> So is it better to wait with -core until -media is more stable?
>>
>
> At some point we agreed that those issues would be handled in
> -media, because in other cases (chat, presence, im) the gateway is
> itself the endpoint and handles all the "media", so those didn't
> seem to apply.
That is my recollection, too.
 [MB] That was also what I thought when I reviewed -core. [/MB]

Thus I'd be in favor of stabilizing the -core spec, and hanlding the
forking issues in the -media spec.
[MB] I agree. [/MB]

Peter

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


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<body lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">There have been no furthe=
r comments on this, so I think we can declare it decided that we won&#8217;=
t expand the scope of &#8211;core with topics like forking, but leave
 that to the &#8211;media to handle. <o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">My proposed plan for the =
next steps of the WG is:<o:p></o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-18.0pt;mso-list:l0 leve=
l1 lfo1"><![if !supportLists]><span style=3D"font-size:11.0pt;font-family:&=
quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=3D"mso=
-list:Ignore">-<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Before Vancouver:=
 Issue WG Last Calls on &#8211;core, -im, and &#8211;presence and see if th=
ey indeed are mature enough to be delivered forward.<o:p></o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-18.0pt;mso-list:l0 leve=
l1 lfo1"><![if !supportLists]><span style=3D"font-size:11.0pt;font-family:&=
quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=3D"mso=
-list:Ignore">-<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">For Vancouver: Ge=
t updated versions of especially &#8211;media, and possibly &#8211;chat and=
 &#8211;groupchat, and collect the open issues about them.<o:p></o:p></span=
></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-18.0pt;mso-list:l0 leve=
l1 lfo1"><![if !supportLists]><span style=3D"font-size:11.0pt;font-family:&=
quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=3D"mso=
-list:Ignore">-<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">In Vancouver: Dis=
cuss the open issues about &#8211;media, and if needed &#8211;chat and &#82=
11;groupchat.<o:p></o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-18.0pt;mso-list:l0 leve=
l1 lfo1"><![if !supportLists]><span style=3D"font-size:11.0pt;font-family:&=
quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=3D"mso=
-list:Ignore">-<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">After Vancouver: =
Depending on the maturity and issues raised, schedule WG Last Calls for the=
 remaining deliverables.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Let the chairs (Yana and =
me) know if you have issues with this plan or even better, suggestions how =
to improve it.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Markus<o:p></o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm 0cm 0cm =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><a name=3D"_____replyseparator"></a><b><span style=
=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"=
>From:</span></b><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&q=
uot;,&quot;sans-serif&quot;"> ext Mary Barnes [mailto:mary.ietf.barnes@gmai=
l.com]
<br>
<b>Sent:</b> 06 September, 2013 19:43<br>
<b>To:</b> Peter Saint-Andre<br>
<b>Cc:</b> Sa=FAl Ibarra Corretg=E9; stox@ietf.org; Isomaki Markus (Nokia-C=
IC/Espoo)<br>
<b>Subject:</b> Re: [Stox] current status<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On Fri, Sep 6, 2013 at 11:35 AM, Peter Saint-Andre &=
lt;<a href=3D"mailto:stpeter@stpeter.im" target=3D"_blank">stpeter@stpeter.=
im</a>&gt; wrote:<o:p></o:p></p>
<div>
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<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">On 9/6/13 10:16 AM, S=
a=FAl Ibarra Corretg=E9 wrote:<br>
&gt;<br>
&gt; On Sep 6, 2013, at 3:25 PM, &lt;<a href=3D"mailto:Markus.Isomaki@nokia=
.com">Markus.Isomaki@nokia.com</a>&gt;<br>
&gt; &lt;<a href=3D"mailto:Markus.Isomaki@nokia.com">Markus.Isomaki@nokia.c=
om</a>&gt; wrote:<br>
&gt;<br>
&gt;&gt; Hi,<br>
&gt;&gt;<br>
&gt;&gt; There have been a fair number of comments on -core, -im and<br>
&gt;&gt; -presence since the Berlin IETF. Please take a look at the new<br>
&gt;&gt; versions <a href=3D"http://tools.ietf.org/id/draft-ietf-stox-core-=
04.txt" target=3D"_blank">
http://tools.ietf.org/id/draft-ietf-stox-core-04.txt</a><br>
&gt;&gt; <a href=3D"http://tools.ietf.org/id/draft-ietf-stox-im-02.txt" tar=
get=3D"_blank">http://tools.ietf.org/id/draft-ietf-stox-im-02.txt</a><br>
&gt;&gt; <a href=3D"http://tools.ietf.org/id/draft-ietf-stox-presence-04.tx=
t" target=3D"_blank">
http://tools.ietf.org/id/draft-ietf-stox-presence-04.txt</a> and see<br>
&gt;&gt; if your issues have been properly addressed, and/or whether you<br=
>
&gt;&gt; want to still raise any new issues.<br>
&gt;&gt;<br>
&gt;&gt; If nothing specific comes up, the chairs will consider starting<br=
>
&gt;&gt; WG LC on these documents in the near future.<br>
&gt;&gt;<br>
&gt;&gt; One thing that I wonder about -core is whether it is a good idea<b=
r>
&gt;&gt; to stabilize or last call it yet at this point, since there have<b=
r>
&gt;&gt; been some issues raised about -media that might have impact to<br>
&gt;&gt; -core, such as dealing with SIP side forking or loop detection.<br=
>
&gt;&gt; So is it better to wait with -core until -media is more stable?<br=
>
&gt;&gt;<br>
&gt;<br>
&gt; At some point we agreed that those issues would be handled in<br>
&gt; -media, because in other cases (chat, presence, im) the gateway is<br>
&gt; itself the endpoint and handles all the &quot;media&quot;, so those di=
dn't<br>
&gt; seem to apply.<o:p></o:p></p>
</div>
<p class=3D"MsoNormal">That is my recollection, too.<o:p></o:p></p>
<div>
<p class=3D"MsoNormal">&nbsp;[MB] That was also what I thought when I revie=
wed -core. [/MB]<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<p class=3D"MsoNormal">Thus I'd be in favor of stabilizing the -core spec, =
and hanlding the<br>
forking issues in the -media spec.<o:p></o:p></p>
</blockquote>
<div>
<p class=3D"MsoNormal">[MB] I agree. [/MB]<o:p></o:p></p>
</div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
Peter<br>
<br>
- --<br>
Peter Saint-Andre<br>
<a href=3D"https://stpeter.im/" target=3D"_blank">https://stpeter.im/</a><b=
r>
<br>
<br>
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<div>
<div>
<p class=3D"MsoNormal">_______________________________________________<br>
stox mailing list<br>
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<a href=3D"https://www.ietf.org/mailman/listinfo/stox" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/stox</a><o:p></o:p></p>
</div>
</div>
</blockquote>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</div>
</div>
</body>
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From: <Markus.Isomaki@nokia.com>
To: <stox@ietf.org>
Thread-Topic: Meeting slot in Vancouver
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Date: Wed, 18 Sep 2013 11:21:16 +0000
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Hi,

Following the plan below means also that STOX would need a meeting slot in =
Vancouver meeting. The main topic would be the -media draft. So the chairs =
are planning to submit a request for a 1 hour slot. Peter, Emil, S=E1ul, as=
 co-authors of the -media draft we would need at minimum one of you to be p=
resent to lead that discussion.

Markus


From: Isomaki Markus (Nokia-CIC/Espoo)
Sent: 18 September, 2013 14:16
To: 'ext Mary Barnes'; Peter Saint-Andre
Cc: Sa=FAl Ibarra Corretg=E9; stox@ietf.org
Subject: RE: [Stox] current status

Hi,

There have been no further comments on this, so I think we can declare it d=
ecided that we won't expand the scope of -core with topics like forking, bu=
t leave that to the -media to handle.

My proposed plan for the next steps of the WG is:

-          Before Vancouver: Issue WG Last Calls on -core, -im, and -presen=
ce and see if they indeed are mature enough to be delivered forward.

-          For Vancouver: Get updated versions of especially -media, and po=
ssibly -chat and -groupchat, and collect the open issues about them.

-          In Vancouver: Discuss the open issues about -media, and if neede=
d -chat and -groupchat.

-          After Vancouver: Depending on the maturity and issues raised, sc=
hedule WG Last Calls for the remaining deliverables.

Let the chairs (Yana and me) know if you have issues with this plan or even=
 better, suggestions how to improve it.

Markus


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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Following the plan below =
means also that STOX would need a meeting slot in Vancouver meeting. The ma=
in topic would be the &#8211;media draft. So the chairs are planning
 to submit a request for a 1 hour slot. Peter, Emil, S=E1ul, as co-authors =
of the &#8211;media draft we would need at minimum one of you to be present=
 to lead that discussion.
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Markus
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm 0cm 0cm =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><a name=3D"_____replyseparator"></a><b><span style=
=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"=
>From:</span></b><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&q=
uot;,&quot;sans-serif&quot;"> Isomaki Markus (Nokia-CIC/Espoo)
<br>
<b>Sent:</b> 18 September, 2013 14:16<br>
<b>To:</b> 'ext Mary Barnes'; Peter Saint-Andre<br>
<b>Cc:</b> Sa=FAl Ibarra Corretg=E9; stox@ietf.org<br>
<b>Subject:</b> RE: [Stox] current status<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">There have been no furthe=
r comments on this, so I think we can declare it decided that we won&#8217;=
t expand the scope of &#8211;core with topics like forking, but leave
 that to the &#8211;media to handle. <o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">My proposed plan for the =
next steps of the WG is:<o:p></o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-18.0pt;mso-list:l0 leve=
l1 lfo2"><![if !supportLists]><span style=3D"font-size:11.0pt;font-family:&=
quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=3D"mso=
-list:Ignore">-<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Before Vancouver:=
 Issue WG Last Calls on &#8211;core, -im, and &#8211;presence and see if th=
ey indeed are mature enough to be delivered forward.<o:p></o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-18.0pt;mso-list:l0 leve=
l1 lfo2"><![if !supportLists]><span style=3D"font-size:11.0pt;font-family:&=
quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=3D"mso=
-list:Ignore">-<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">For Vancouver: Ge=
t updated versions of especially &#8211;media, and possibly &#8211;chat and=
 &#8211;groupchat, and collect the open issues about them.<o:p></o:p></span=
></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-18.0pt;mso-list:l0 leve=
l1 lfo2"><![if !supportLists]><span style=3D"font-size:11.0pt;font-family:&=
quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=3D"mso=
-list:Ignore">-<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">In Vancouver: Dis=
cuss the open issues about &#8211;media, and if needed &#8211;chat and &#82=
11;groupchat.<o:p></o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-18.0pt;mso-list:l0 leve=
l1 lfo2"><![if !supportLists]><span style=3D"font-size:11.0pt;font-family:&=
quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=3D"mso=
-list:Ignore">-<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">After Vancouver: =
Depending on the maturity and issues raised, schedule WG Last Calls for the=
 remaining deliverables.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Let the chairs (Yana and =
me) know if you have issues with this plan or even better, suggestions how =
to improve it.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Markus<o:p></o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
</div>
</div>
</body>
</html>

--_000_E44893DD4E290745BB608EB23FDDB7620A0B3FA2008AM1MPN1043mg_--

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To: <Markus.Isomaki@nokia.com> <Markus.Isomaki@nokia.com>
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Subject: Re: [Stox] Meeting slot in Vancouver
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On Sep 18, 2013, at 1:21 PM, <Markus.Isomaki@nokia.com> =
<Markus.Isomaki@nokia.com> wrote:

> Hi,
> =20
> Following the plan below means also that STOX would need a meeting =
slot in Vancouver meeting. The main topic would be the =96media draft. =
So the chairs are planning to submit a request for a 1 hour slot. Peter, =
Emil, S=E1ul, as co-authors of the =96media draft we would need at =
minimum one of you to be present to lead that discussion.
> =20

Hi Markus,

Unfortunately I won't be present in Vancouver, I'll participate remotely =
though.

--
Sa=FAl Ibarra Corretg=E9
AG Projects




From stpeter@stpeter.im  Wed Sep 18 05:55:09 2013
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Cc: stox@ietf.org, "<Markus.Isomaki@nokia.com>" <Markus.Isomaki@nokia.com>
Subject: Re: [Stox] Meeting slot in Vancouver
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-----BEGIN PGP SIGNED MESSAGE-----
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On 9/18/13 6:01 AM, Saúl Ibarra Corretgé wrote:
> 
> On Sep 18, 2013, at 1:21 PM, <Markus.Isomaki@nokia.com> 
> <Markus.Isomaki@nokia.com> wrote:
> 
>> Hi,
>> 
>> Following the plan below means also that STOX would need a
>> meeting slot in Vancouver meeting. The main topic would be the
>> –media draft. So the chairs are planning to submit a request for
>> a 1 hour slot. Peter, Emil, Sául, as co-authors of the –media
>> draft we would need at minimum one of you to be present to lead
>> that discussion.

I will almost certainly be there.

> Unfortunately I won't be present in Vancouver, I'll participate 
> remotely though.

We might want to make sure our session is earlier in the day so that
Saúl (and other folks in Europe) aren't joining at 2 AM.

Peter

- -- 
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From stpeter@stpeter.im  Wed Sep 18 05:56:42 2013
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Subject: Re: [Stox] current status
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On 9/18/13 5:16 AM, Markus.Isomaki@nokia.com wrote:
> Hi,
> 
> 
> 
> There have been no further comments on this, so I think we can
> declare it decided that we won?t expand the scope of ?core with
> topics like forking, but leave that to the ?media to handle.
> 
> 
> 
> My proposed plan for the next steps of the WG is:
> 
> -          Before Vancouver: Issue WG Last Calls on ?core, -im,
> and ?presence and see if they indeed are mature enough to be
> delivered forward.
> 
> -          For Vancouver: Get updated versions of especially
> ?media, and possibly ?chat and ?groupchat, and collect the open
> issues about them.
> 
> -          In Vancouver: Discuss the open issues about ?media, and
> if needed ?chat and ?groupchat.
> 
> -          After Vancouver: Depending on the maturity and issues
> raised, schedule WG Last Calls for the remaining deliverables.

That all sounds good. I will work to update at least the -chat draft
very soon, and work with Saúl on the groupchat draft too.

Peter

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


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From fippo@goodadvice.pages.de  Wed Sep 18 06:25:50 2013
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Subject: [Stox] media: rejecting an m-line in jingle
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I just stumbled over a case where I have to map a rejected m-line over 
jingle... i.e. the port in the m-line is set to 0.

Jingle doesn't have a mapping for the port in the m-line. So how can one 
reject a single m-line (content) in jingle?

While jingle has a content-reject option that only happens in response to 
a content-add. A XEP-0177 (raw udp) <transport/> element with port set to
zero might be an option... but that seems onerous and I doubt that anything
which currently implement 0177 handles it correctly.



From stpeter@stpeter.im  Wed Sep 18 07:15:25 2013
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On 9/13/13 6:42 AM, Adrian Georgescu wrote:
> Hello,
> 
> I am not sure if this is relevant to any of the deliverables of the
> working group, but just in case it is, we were able to create
> end-to-end encryption for chat sessions using OTR protocol
> http://www.cypherpunks.ca/otr/  between a SIP client using MSRP
> (that is Blink) and XMPP client (Jitsi).
> 
> The implementation is described here:
> 
> http://projects.ag-projects.com/projects/blinkc/wiki/OTR

Cool!

Given that OTR is an "over-the-top" encryption method that's agnostic
about the underlying messaging transport, it does seem like a
reasonable approach. I'm not sure whether we need or want to say
anything about that in draft-ietf-stox-chat.

Peter

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


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Hash: SHA1

I've taken a look at the mapping between XMPP chat states (XEP-0085)
and the isComposing event (RFC 3994).

RFC 3994 has two states: idle and active. RFC 3994 has only two states
because it's really just about composing (in messaging, whether or not
the sender is actively typing or otherwise creating text for sending
to the recipient -- simply toggling between the two, changing to
active if the user is composing a message and changing to idle when
not composing a message).

XEP-0085 has five states: active, inactive, gone, composing, and
paused. XEP-0085 has more states than RFC 3994 because it's not just
about composing, but about the sender's involvement with the chat
session. Thus in XEP-0085, "active" means "paying attention to the
conversation", which ironically maps best to RFC 3994 "idle", I think.

It might be best if we don't map XEP-0085 chat states other than
composing and paused to RFC 3994 isComposing events, since RFC 3994 is
just about composing. However, in XEP-0085 it is possible to have
state transitions such as composing to active (not paused).

Thus the safest mapping might be

isComposing => chat states
  idle => active
  active => composing

chat states => isComposing
  active => idle
  inactive => idle
  gone => ???
  composing => active
  paused => idle

This mapping loses some of the granularity on the XMPP side (i.e., the
paused and inactive states), but that might be fine in this context.

I know Saúl has looked at this mapping, so I would appreciate his
feedback (and input from other folks, too).

Peter

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


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From internet-drafts@ietf.org  Wed Sep 18 20:22:17 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): One-to-One Te=
xt Chat
	Author(s)       : Peter Saint-Andre
                          Salvatore Loreto
                          Eddy Gavita
                          Nazin Hossain
	Filename        : draft-ietf-stox-chat-01.txt
	Pages           : 16
	Date            : 2013-09-18

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-01

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


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

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


From stpeter@stpeter.im  Wed Sep 18 20:23:35 2013
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A first attempt at including composing notifications and delivery
receipts...

Peter

On 9/18/13 9:22 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): One-to-One Text Chat
> 	Author(s)       : Peter Saint-Andre
>                           Salvatore Loreto
>                           Eddy Gavita
>                           Nazin Hossain
> 	Filename        : draft-ietf-stox-chat-01.txt
> 	Pages           : 16
> 	Date            : 2013-09-18
> 
> 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-01
> 
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-stox-chat-01
> 
> 
> 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/



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Subject: Re: [Stox] review of core, chat, groupchat and presence
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

On 8/8/13 12:33 PM, Philipp Hancke wrote:

> chat: example f2: call-id: contain jid? how about uniqueness?. Is
> there any text on the use of <thread/>? in -im @example.com callid
> is used a=lang could be taken xml:lang? before example F4: the
> gateway acknowledges on behalf of juliet?
> 
> example f5: the initial message in F1 should have an id attribute
> which gets mapped in f5. Does a786hjs2 map to thread/call id?
> 
> example f6: shouldn't message-id should be different from f5? 
> mapping of message-id to id attribute, should be randomly generated
> if not present
> 
> section 4: sessinos -> sessions;

Sorry, I neglected to incorporate this feedback into version -01. I'll
do that for -02.

Peter

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


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From saul@ag-projects.com  Thu Sep 19 00:44:22 2013
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Subject: Re: [Stox] chat states <-> isComposing
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Hi Peter!

Thanks for looking into this, I think it's really useful.

On Sep 18, 2013, at 5:10 PM, Peter Saint-Andre wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>=20
> I've taken a look at the mapping between XMPP chat states (XEP-0085)
> and the isComposing event (RFC 3994).
>=20
> RFC 3994 has two states: idle and active. RFC 3994 has only two states
> because it's really just about composing (in messaging, whether or not
> the sender is actively typing or otherwise creating text for sending
> to the recipient -- simply toggling between the two, changing to
> active if the user is composing a message and changing to idle when
> not composing a message).
>=20
> XEP-0085 has five states: active, inactive, gone, composing, and
> paused. XEP-0085 has more states than RFC 3994 because it's not just
> about composing, but about the sender's involvement with the chat
> session. Thus in XEP-0085, "active" means "paying attention to the
> conversation", which ironically maps best to RFC 3994 "idle", I think.
>=20
> It might be best if we don't map XEP-0085 chat states other than
> composing and paused to RFC 3994 isComposing events, since RFC 3994 is
> just about composing. However, in XEP-0085 it is possible to have
> state transitions such as composing to active (not paused).
>=20
> Thus the safest mapping might be
>=20
> isComposing =3D> chat states
>  idle =3D> active
>  active =3D> composing
>=20

Yes, I implemented this mapping following my intuition and feels ok in =
actual usage.

> chat states =3D> isComposing
>  active =3D> idle
>  inactive =3D> idle
>  gone =3D> ???
>  composing =3D> active
>  paused =3D> idle
>=20

Here comes the dilemma: that gone state. Not many clients implement the =
'gone' state, but those who do tend to render a message such as "Peter =
left the conversation" if they get one, and likewise, send it if you =
close the conversation tab. Based on this non-scientific testing I =
mapped it to a SIP BYE, both ways. That is, if we are having a chat =
session (I am on MSRP and you on XMPP) if I end the SIP session, the =
gateway would send a gone chatstate, and if your client would send it, =
the gateway would terminate the SIP session.

I know this looks weird, since we are mixing payload translation with =
session management, but since the gateway needs to handle all chat media =
I don't see a problem with that.

Thoughts?

> This mapping loses some of the granularity on the XMPP side (i.e., the
> paused and inactive states), but that might be fine in this context.
>=20

We'll need to adjust to the minimum common denominator, like we do in =
groupchat :-)


Cheers,

--
Sa=FAl Ibarra Corretg=E9
AG Projects




From fippo@goodadvice.pages.de  Thu Sep 19 00:53:34 2013
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From: Philipp Hancke <fippo@goodadvice.pages.de>
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Subject: Re: [Stox] [Jingle] media: rejecting an m-line in jingle
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On Wed, 18 Sep 2013, Olivier CrÃªte wrote:

> On Wed, 2013-09-18 at 15:25 +0200, Philipp Hancke wrote:
>> Jingle doesn't have a mapping for the port in the m-line. So how can one
>> reject a single m-line (content) in jingle?
>
> You just send a "content-remove" stanza. The jingle design tries to
> avoid nasty hack such as the port=0 SIP/SDP thing.

So you'd send a content-remove before sending the session-accept?

FWIW, it's not actually clear to me whether jingle imposes a 
"session-accept must have the same number of contents as session-initiate" 
constraint (3264 is only used for the inclusion of payloads).
If it doesn't then not sending rejected contents in the answer would be 
valid, but I doubt gateways will appreciate that more than the 
content-remove way :-)
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From stpeter@stpeter.im  Thu Sep 19 05:50:32 2013
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Subject: Re: [Stox] chat states <-> isComposing
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[removing areas of agreement]

On 9/19/13 1:44 AM, Saúl Ibarra Corretgé wrote:

> On Sep 18, 2013, at 5:10 PM, Peter Saint-Andre wrote:
> 
>> gone => ???
>> 
> 
> Here comes the dilemma: that gone state. Not many clients
> implement the 'gone' state, but those who do tend to render a
> message such as "Peter left the conversation" if they get one, and
> likewise, send it if you close the conversation tab. Based on this
> non-scientific testing I mapped it to a SIP BYE, both ways. That
> is, if we are having a chat session (I am on MSRP and you on XMPP)
> if I end the SIP session, the gateway would send a gone chatstate,
> and if your client would send it, the gateway would terminate the
> SIP session.
> 
> I know this looks weird, since we are mixing payload translation
> with session management, but since the gateway needs to handle all
> chat media I don't see a problem with that.

That makes a lot of sense to me, given that the gone chat state is
kind of like in-session unavailable presence.

Peter

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


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From stpeter@stpeter.im  Thu Sep 19 09:47:20 2013
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Philipp, thanks for the review. Comments inline.

On 9/18/13 9:53 PM, Peter Saint-Andre wrote:
> On 8/8/13 12:33 PM, Philipp Hancke wrote:
> 
>> chat: example f2: call-id: contain jid? how about uniqueness?.

Yes, RFC 3261 says they need to be unique:

   In a new request created by a UAC outside of any dialog, the Call-ID
   header field MUST be selected by the UAC as a globally unique
   identifier over space and time unless overridden by method-specific
   behavior.  All SIP UAs must have a means to guarantee that the Call-
   ID header fields they produce will not be inadvertently generated by
   any other UA.

I was unthinkingly following the examples in specs like RFC 4975. I'll
change them to be UUIDs.

>> Is there any text on the use of <thread/>?

There is text about this in draft-ietf-stox-im, which says to map
Call-ID to <thread/> and vice-versa. However, that probably belongs in
the -chat document, since thread isn't really used in pager-mode
single messages.

>> in -im @example.com callid is used

As mentioned, I'll make them UUIDs.

>> a=lang could be taken xml:lang?

In the -im I-D we map xml:lang to the Content-Language header. I think
that's more appropriate than a=lang, which has some subtleties I'd
prefer to avoid -- see draft-gellens-mmusic-negotiating-human-language
for some related considerations.

>> before example F4: the gateway acknowledges on behalf of juliet?

Yours was a very brief comment and I'm not sure I understand the force
of your question. Are you wondering why there's an ACK at this point
in the negotiation? That's what I see in RFC 4975:

           Alice                     Bob
             |                        |
             |                        |
             |(1) (SIP) INVITE        |
             |----------------------->|
             |(2) (SIP) 200 OK        |
             |<-----------------------|
             |(3) (SIP) ACK           |
             |----------------------->|
             |(4) (MSRP) SEND         |
             |----------------------->|

>> example f5: the initial message in F1 should have an id
>> attribute

Agreed.

>> which gets mapped in f5. Does a786hjs2 map to thread/call id?

That's an MSRP transaction identifier, which IMHO maps to a stanza
'id' attribute in XMPP. This too needs to be documented in the -chat I-D.

>> example f6: shouldn't message-id should be different from f5?

Yes, I think so, because that's a different transaction.

>> mapping of message-id to id attribute, should be randomly
>> generated if not present

Hmm. In MSRP, the message-id can be used to identify an overall
message, which might be chunked into separate transactions. XMPP
doesn't have the concept of a message in the MSRP sense when we're
talking about IM. In XMPP, In-Band Bytestreams (XEP-0047) enables you
to chunk a large piece of data into smaller pieces, but that's used
only for file transfer, not for IM. We could add a note about this in
the -chat I-D, but for IM purposes I think we don't want to map the
MSRP message-id to anything on the XMPP side.

>> section 4: sessinos -> sessions;

Yes, my fingers often make that mistake while typing. :-)

Peter

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


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From fippo@goodadvice.pages.de  Thu Sep 19 10:13:46 2013
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Am 19.09.2013 18:46, schrieb Peter Saint-Andre:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Philipp, thanks for the review. Comments inline.

Stuff that is done removed. Thanks for addressing the issues.

[...]
>>> chat: example f2: call-id: contain jid? how about uniqueness?.
>
> Yes, RFC 3261 says they need to be unique:
>
>     In a new request created by a UAC outside of any dialog, the Call-ID
>     header field MUST be selected by the UAC as a globally unique
>     identifier over space and time unless overridden by method-specific
>     behavior.  All SIP UAs must have a means to guarantee that the Call-
>     ID header fields they produce will not be inadvertently generated by
>     any other UA.
>
> I was unthinkingly following the examples in specs like RFC 4975. I'll
> change them to be UUIDs.

I think what I mean was "the call id should be something along the lines 
of 711609sa.juliet.example.com". The intent was the same however.

[...]
>>> before example F4: the gateway acknowledges on behalf of juliet?
>
> Yours was a very brief comment and I'm not sure I understand the force
> of your question. Are you wondering why there's an ACK at this point
> in the negotiation? That's what I see in RFC 4975:

What I meant (and i needed to parse this again, too) was that I think 
the gateway acknowledges on behalf of Juliet, not Romeo as the text 
says. The example itself is ok.

philipp


From internet-drafts@ietf.org  Thu Sep 19 20:39:16 2013
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Subject: [Stox] I-D Action: draft-ietf-stox-chat-02.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 Sessions
	Author(s)       : Peter Saint-Andre
                          Salvatore Loreto
                          Eddy Gavita
                          Nazin Hossain
	Filename        : draft-ietf-stox-chat-02.txt
	Pages           : 17
	Date            : 2013-09-19

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-02

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-stox-chat-02


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

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


From salvatore.loreto@ericsson.com  Fri Sep 20 02:39:17 2013
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Date: Fri, 20 Sep 2013 12:39:05 +0300
From: Salvatore Loreto <salvatore.loreto@ericsson.com>
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Subject: [Stox] review: stox-core-04
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I have reviewed the core-04 draft, and I think that is almost ready for 
the WGLC

I have one general comment
(btw I apologize in advance if this has already been discussed and I 
have overlooked or forgot it)
and it is about the fact that draft does not talk at all about the inter 
working when secure URIs are involved.
I think that at least we should say something about in the Security 
consideration.

* Editorial nits

-In section 6.2, the following sentence

    The codes listed below are limited to those defined in the core SIP
    specification [RFC3261  <http://tools.ietf.org/html/rfc3261>] and in RFCs that formally update [RFC3261  <http://tools.ietf.org/html/rfc3261>].

It suggests, at least to me, that in the list below are listed all the 
codes defined
in 3261 and in RFC that formally update 3261, but this is not the case.

-first paragraph section 5.1
     s/spports/supports

Cheers
Salvatore

-- 
Salvatore Loreto, PhD
www.sloreto.com


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    I have reviewed the core-04 draft, and I think that is almost ready
    for the WGLC<br>
    <br>
    I have one general comment <br>
    (btw I apologize in advance if this has already been discussed and I
    have overlooked or forgot it)<br>
    and it is about the fact that draft does not talk at all about the
    inter working when secure URIs are involved.<br>
    I think that at least we should say something about in the Security
    consideration.<br>
    <br>
    * Editorial nits<br>
    <br>
    -In section 6.2, the following sentence<br>
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    <pre class="newpage">   The codes listed below are limited to those defined in the core SIP
   specification [<a href="http://tools.ietf.org/html/rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>] and in RFCs that formally update [<a href="http://tools.ietf.org/html/rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>].</pre>
    It suggests, at least to me, that in the list below are listed all
    the codes defined<br>
    in 3261 and in RFC that formally update 3261, but this is not the
    case. <br>
    <br>
    -first paragraph section 5.1<br>
    &nbsp;&nbsp;&nbsp; s/spports/supports<br>
    <br>
    Cheers<br>
    Salvatore<br>
    <pre class="moz-signature" cols="72">-- 
Salvatore Loreto, PhD
<a class="moz-txt-link-abbreviated" href="http://www.sloreto.com">www.sloreto.com</a></pre>
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Subject: [Stox] review: stox-im-03
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I have reviewed the core-04 draft, and I think that is almost ready for 
the WGLC

my only comments is about the fact that section 3. XMPP to SIP
implies that the MESSAGE is always (and maybe can only be) delivered 
over a TCP connection;
RFC3428 states

    Whenever possible, MESSAGE requests SHOULD be sent over transports
    that implement end-to-end congestion control, such as TCP or SCTP.
    However, SIP does not provide a mechanism to prevent a downstream hop
    from sending a request over UDP.

so you can delivery it over UDP as well and that has also being raised 
in a previous thread
while talking on the Call-ID mapping to <thread/>.
Actually you will be forced to deliver over UDP if the SIMPLE server 
only supports UDP.

So I would suggest to change

    Here we assume that the XMPP server has
    determined the foreign domain is serviced by a SIMPLE server,

in

    Here we assume that the XMPP server has
    determined the foreign domain is serviced by a SIMPLE TCP server,


best regards
Salvatore

-- 
Salvatore Loreto, PhD
www.sloreto.com


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    <div class="moz-cite-prefix">I have reviewed the core-04 draft, and
      I think that is almost ready for the WGLC<br>
      <br>
      my only comments is about the fact that section 3. XMPP to SIP <br>
      implies that the MESSAGE is always (and maybe can only be)
      delivered over a TCP connection;<br>
      RFC3428 states<br>
      <meta http-equiv="content-type" content="text/html;
        charset=ISO-8859-1">
      <pre class="newpage">   Whenever possible, MESSAGE requests SHOULD be sent over transports
   that implement end-to-end congestion control, such as TCP or SCTP.
   However, SIP does not provide a mechanism to prevent a downstream hop
   from sending a request over UDP.</pre>
      so you can delivery it over UDP as well and that has also being
      raised in a previous thread<br>
      while talking on the Call-ID mapping to &lt;thread/&gt;.<br>
      Actually you will be forced to deliver over UDP if the SIMPLE
      server only supports UDP.<br>
      <br>
      So I would suggest to change <br>
      <br>
      <meta http-equiv="content-type" content="text/html;
        charset=ISO-8859-1">
      <pre class="newpage">   Here we assume that the XMPP server has
   determined the foreign domain is serviced by a SIMPLE server,</pre>
      in<br>
      <br>
      <meta http-equiv="content-type" content="text/html;
        charset=ISO-8859-1">
      <pre class="newpage">   Here we assume that the XMPP server has
   determined the foreign domain is serviced by a SIMPLE TCP server,</pre>
      <br>
      best regards<br>
      Salvatore<br>
    </div>
    <pre class="moz-signature" cols="72">-- 
Salvatore Loreto, PhD
<a class="moz-txt-link-abbreviated" href="http://www.sloreto.com">www.sloreto.com</a></pre>
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

On 8/14/13 1:19 PM, Markus.Isomaki@nokia.com wrote:

> * Groupchat
> 
> - Issue: Need to define what to do in case of multiple clients. Use
> and mapping of JID, AoR, contact.

I think I'll need to listen to the audio recording to remember exactly
what the issue is here. Will do and report further.

Peter

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


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Thanks again for the review. Comments inline.

On 8/8/13 12:33 PM, Philipp Hancke wrote:

> groupchat: 2: at least in RFC 6120, jid stands for jabber id, not
> jabber identifier.

Correct. Will fix.

> 3.1: Room Nickname -> lowercase?

Sure.

> 3.1: example f2: conversion->conversation?

More like conversion->mapping :-)

BTW in the -chat spec I numbered all the examples sequentially, and
I'll start doing that here too.

> table 1: thread<->call id

Yes.

> 3.2: example f7: possibly use by= on the error element?

Sure, by='verona@chat.example.org' seems good.

> xep 0045 has the conflicting resource in the from

Correct. Will fix.

> 3.4: "Occupant JID in another room" -- i don't think mediated 
> invites work across different rooms.

Section 7.8.2 of XEP-0045 states in part:

###

It may be wondered why the invitee does not send the decline message
directly to the inviter. The main reason is that certain
implementations might choose to base invitations on occupant JIDs
rather than bare JIDs (so that, for example, an occupant could invite
someone from one room to another without knowing that person's bare
JID). Thus the service needs to handle both the invites and declines.

###

However, I'm not sure how widely that feature is implemented.

> example f1: hecate is benvolio? c&p error from 0045

Will fix.

> example f2: contact juliet@juliet.example.com has a wrong host?

You are right.

> reeived -> received

Yes.

> example f3 is not shown

I don't feel that we need to show every 200 OK. Yes, we put the
example in example.com, but sometimes you really can have too many
examples. :-)

> below f4 there is a sip/2.0 200 ok on its own?

Will fix.

> 3.5: table 2: xmpp From/To -> from/to

Will fix.

> ex f9: where is juliets own presence in msrp?

Good catch. Will fix.

> ex f11: where is juliets own presence with code 110?

Will fix.

> ex f14: mapping <message id/> to Message-ID does msrp have a
> concept of room history?

I'm not sure.

> 3.6.2: do we have an equivalent disco feature for muc?

No, it's assumed by support for 'http://jabber.org/protocol/muc'.

> 3.7: does xmpp still support the <status/> when in muc? In IRC
> this has caused PART spam

Yes:

http://xmpp.org/extensions/xep-0045.html#changepres

> 4.1: ex f5: lack of error until when? must wait until receiving
> 110 status?

We might way "within a reasonable amount of time, such as 5 seconds".

> 4.4: presence broadcast before ack? must buffer until receiving
> 110 ack?

Your comment is a bit cryptic. Would you please expand it?

> 4.5.1: use message id instead of guessing?

Good idea.

Thanks for the feedback!

Peter

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 > Thanks again for the review. Comments inline.

As before, removing items done.

new nit: [length] looks like it should be replaced with the examples 
actual length in three places. Only seen in -groupchat.

[...]
>> example f3 is not shown
>
> I don't feel that we need to show every 200 OK. Yes, we put the
> example in example.com, but sometimes you really can have too many
> examples. :-)

Right. But then it might make sense to either add a "not shown" marker 
in the ascii art or not number them. I'll check if that happens often 
enough to make sense.

[...]
>> 4.4: presence broadcast before ack? must buffer until receiving
>> 110 ack?
>
> Your comment is a bit cryptic. Would you please expand it?

Hah, that was the only comment you found cryptic? :-)

btw: it might make sense to reuse the caption from XEP-0045 (7.2.3 
Presence Broadcast) for the caption of F7 (which is raw, unbridled xmpp 
and not translated; current text seems copy-paste from F11 in sect 3.5):
  "Service Sends Presence from Existing Occupants to New Occupant"

Now the attempt to decipher my comments:
First part:
In MUC, the participant lists is sent to the new occupant (muc example 
21) before the new occupants own presence (muc  example 24) and 
therefore potentially before any acknowledgements that might be required 
on the MSRP side. Is that a problem?


Second part:
In example F10, two (or more) stanzas are translated into a single 
event. Unless you can do partial updates (Emil? I think you mentioned 
such a thing for COIN) here, the gateway needs to know when it can 
dispatch example 4.4 -- upon receiving the participants own presence, 
which might come with a 110 status code. Since I don't think all xmpp 
serveers do 110, i think there is some text required that the gateway 
needs to be able to recognize the self-presence and not mistake it as 
part of the participants list.
(should it be participant_s_ presence in the caption?)

Is that clearer?

From stpeter@stpeter.im  Mon Sep 23 16:45:56 2013
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Hi Sal, thanks for the review.

On 9/20/13 4:42 AM, Salvatore Loreto wrote:

> my only comments is about the fact that section 3. XMPP to SIP 
> implies that the MESSAGE is always (and maybe can only be)
> delivered over a TCP connection; RFC3428 states
> 
> Whenever possible, MESSAGE requests SHOULD be sent over transports 
> that implement end-to-end congestion control, such as TCP or SCTP. 
> However, SIP does not provide a mechanism to prevent a downstream
> hop from sending a request over UDP.
> 
> so you can delivery it over UDP as well and that has also being
> raised in a previous thread while talking on the Call-ID mapping to
> <thread/>. Actually you will be forced to deliver over UDP if the
> SIMPLE server only supports UDP.

It's not clear to me why the XMPP-to-SIP gateway couldn't communicate
with the SIMPLE server over UDP. Which text in Section 3 do you think
implies that only TCP connections are supported?

Peter

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On 9/20/13 3:39 AM, Salvatore Loreto wrote:
> I have reviewed the core-04 draft, and I think that is almost ready
> for the WGLC
> 
> I have one general comment (btw I apologize in advance if this has
> already been discussed and I have overlooked or forgot it) and it
> is about the fact that draft does not talk at all about the inter 
> working when secure URIs are involved. I think that at least we
> should say something about in the Security consideration.

Good catch. After reading and re-reading Section 26.4.4 of RFC 3261, I
have to admit that I'm not sure exactly how SIP entities are supposed
to handle SIPS URIs. Even if we understand those requirements, I'm
also not sure how they would apply to the XMPP side of the
communications path. At the least, it seems we'd want to specify that
if the To header or Request-URI is a SIPS URI, then the SIP-to-XMPP
gateway needs to connect to the XMPP server over a TLS-protected
stream. However, do we also need to stipulate that the XMPP
server-to-client connection is TLS-protected? If so, how would the
gateway associated with the XMPP server ensure that? (The gateway
might be an external component of the XMPP server, without control
over how the XMPP server communicates with XMPP clients.)

Peter

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On 9/22/13 11:57 PM, Philipp Hancke wrote:
>> Thanks again for the review. Comments inline.
> 
> As before, removing items done.
> 
> new nit: [length] looks like it should be replaced with the
> examples actual length in three places. Only seen in -groupchat.

Will fix.

> [...]
>>> example f3 is not shown
>> 
>> I don't feel that we need to show every 200 OK. Yes, we put the 
>> example in example.com, but sometimes you really can have too
>> many examples. :-)
> 
> Right. But then it might make sense to either add a "not shown"
> marker in the ascii art or not number them. I'll check if that
> happens often enough to make sense.

Well, maybe we'll just add the examples for the sake of completeness.

> [...]
>>> 4.4: presence broadcast before ack? must buffer until
>>> receiving 110 ack?
>> 
>> Your comment is a bit cryptic. Would you please expand it?
> 
> Hah, that was the only comment you found cryptic? :-)

I think that I managed to decryptify the others. ;-)

> btw: it might make sense to reuse the caption from XEP-0045 (7.2.3 
> Presence Broadcast) for the caption of F7 (which is raw, unbridled
> xmpp and not translated; current text seems copy-paste from F11 in
> sect 3.5): "Service Sends Presence from Existing Occupants to New
> Occupant"

WFM.

> Now the attempt to decipher my comments: First part: In MUC, the
> participant lists is sent to the new occupant (muc example 21)
> before the new occupants own presence (muc  example 24) and 
> therefore potentially before any acknowledgements that might be
> required on the MSRP side. Is that a problem?

Well, neither RFC 4975 nor draft-ietf-simple-chat really specifies the
handling of participant lists. There is a passing mention in the
latter document, but it's not very detailed:

   Typically the conference focus acts as a notifier of the conference
   event package, RFC 4575 [RFC4575].  It is RECOMMENDED that conference
   foci and endpoints support RFC 6502 [RFC6502] for providing
   information regarding the conference, and in particular, supplying
   information of the roaster of the conference.  It is also RECOMMENDED
   that conference foci and endpoints support RFC 6501 [RFC6501], which
   extends the <user> element originally specified in RFC 4575 [RFC4575]
   with a new 'nickname' attribute.  This allows endpoints to learn the
   nicknames of participants of the chat room.

However, in draft-ietf-stox-groupchat we assume that the conference
focus does support the conference-info package for the room roster.

[ Hopefully the RFC Editor will correct "roaster" to "roster" :-) ]

> Second part: In example F10, two (or more) stanzas are translated
> into a single event. Unless you can do partial updates (Emil? I
> think you mentioned such a thing for COIN) here, the gateway needs
> to know when it can dispatch example 4.4 -- upon receiving the
> participants own presence, which might come with a 110 status code.
> 
That's all correct.

> Since I don't think all xmpp serveers do 110, i think there is some
> text required that the gateway needs to be able to recognize the
> self-presence and not mistake it as part of the participants list. 
> (should it be participant_s_ presence in the caption?)

The SIP-to-XMPP gateway will need to maintain state about the MSRP
session that the SIP user agent has requested. This implies that the
gateway will need to know the user's SIP URI and the mapping of that
URI into an XMPP address, then based on that mapping determine when it
has received a complete room roster from the MUC room (with or without
a 110 status code), i.e., when it has received the in-room presence of
the SIP user (which according to XEP-0045 is the last presence stanza
received in the initial batch sent after joining). Once that happens,
the SIP-to-XMPP gateway can construct the conference-info document
containg the complete room roster and send that to the SIP user.

And yes, we need to make that clear in the spec.

> Is that clearer?

Yes, thanks.

Peter

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


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On 9/24/13 2:45 AM, Peter Saint-Andre wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi Sal, thanks for the review.
>
> On 9/20/13 4:42 AM, Salvatore Loreto wrote:
>
>> my only comments is about the fact that section 3. XMPP to SIP
>> implies that the MESSAGE is always (and maybe can only be)
>> delivered over a TCP connection; RFC3428 states
>>
>> Whenever possible, MESSAGE requests SHOULD be sent over transports
>> that implement end-to-end congestion control, such as TCP or SCTP.
>> However, SIP does not provide a mechanism to prevent a downstream
>> hop from sending a request over UDP.
>>
>> so you can delivery it over UDP as well and that has also being
>> raised in a previous thread while talking on the Call-ID mapping to
>> <thread/>. Actually you will be forced to deliver over UDP if the
>> SIMPLE server only supports UDP.
> It's not clear to me why the XMPP-to-SIP gateway couldn't communicate
> with the SIMPLE server over UDP. Which text in Section 3 do you think
> implies that only TCP connections are supported?
there is not text at all, and that is the problem.
When I read the Section 3 and the example I got the impression that
I can only use TCP to forward the message on the SIP side
but maybe it is only me!

cheers
Salvatore

>
> Peter
>
> - -- 
> Peter Saint-Andre
> https://stpeter.im/
>
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On 9/24/13 12:02 AM, Salvatore Loreto wrote:
> On 9/24/13 2:45 AM, Peter Saint-Andre wrote: Hi Sal, thanks for the
> review.
> 
> On 9/20/13 4:42 AM, Salvatore Loreto wrote:
> 
>>>> my only comments is about the fact that section 3. XMPP to
>>>> SIP implies that the MESSAGE is always (and maybe can only
>>>> be) delivered over a TCP connection; RFC3428 states
>>>> 
>>>> Whenever possible, MESSAGE requests SHOULD be sent over
>>>> transports that implement end-to-end congestion control, such
>>>> as TCP or SCTP. However, SIP does not provide a mechanism to
>>>> prevent a downstream hop from sending a request over UDP.
>>>> 
>>>> so you can delivery it over UDP as well and that has also
>>>> being raised in a previous thread while talking on the
>>>> Call-ID mapping to <thread/>. Actually you will be forced to
>>>> deliver over UDP if the SIMPLE server only supports UDP.
> It's not clear to me why the XMPP-to-SIP gateway couldn't
> communicate with the SIMPLE server over UDP. Which text in Section
> 3 do you think implies that only TCP connections are supported?
>> there is not text at all, and that is the problem. When I read
>> the Section 3 and the example I got the impression that I can
>> only use TCP to forward the message on the SIP side but maybe it
>> is only me!

I doubt it is only you. :-)

Because there is no such thing as XMPP over UDP, the gateway from SIP
to XMPP will need to communicate over TCP. However, I see no reason
why the gateway from XMPP to SIP could not be "smart" and send SIP
messages over TCP or UDP depending on what the far SIP server
supports. I'll work to add some clarifying text on these points.

Peter

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


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From: Salvatore Loreto <salvatore.loreto@ericsson.com>
To: "stpeter@stpeter.im" <stpeter@stpeter.im>
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=0A=
Thanks Peter=0A=
=0A=
=0A=
--Original message---=0A=
Sender: "Peter Saint-Andre" <stpeter@stpeter.im>=0A=
Time: Tue Sep 24 15:35:00 EEST 2013=0A=
Cc: stox@ietf.org, =0A=
Subject: Re: [Stox] review: stox-im-03=0A=
=0A=
-----BEGIN PGP SIGNED MESSAGE-----=0A=
Hash: SHA1=0A=
=0A=
On 9/24/13 12:02 AM, Salvatore Loreto wrote:=0A=
> On 9/24/13 2:45 AM, Peter Saint-Andre wrote: Hi Sal, thanks for the=0A=
> review.=0A=
> =0A=
> On 9/20/13 4:42 AM, Salvatore Loreto wrote:=0A=
> =0A=
>>>> my only comments is about the fact that section 3. XMPP to=0A=
>>>> SIP implies that the MESSAGE is always (and maybe can only=0A=
>>>> be) delivered over a TCP connection; RFC3428 states=0A=
>>>> =0A=
>>>> Whenever possible, MESSAGE requests SHOULD be sent over=0A=
>>>> transports that implement end-to-end congestion control, such=0A=
>>>> as TCP or SCTP. However, SIP does not provide a mechanism to=0A=
>>>> prevent a downstream hop from sending a request over UDP.=0A=
>>>> =0A=
>>>> so you can delivery it over UDP as well and that has also=0A=
>>>> being raised in a previous thread while talking on the=0A=
>>>> Call-ID mapping to <thread/>. Actually you will be forced to=0A=
>>>> deliver over UDP if the SIMPLE server only supports UDP.=0A=
> It's not clear to me why the XMPP-to-SIP gateway couldn't=0A=
> communicate with the SIMPLE server over UDP. Which text in Section=0A=
> 3 do you think implies that only TCP connections are supported?=0A=
>> there is not text at all, and that is the problem. When I read=0A=
>> the Section 3 and the example I got the impression that I can=0A=
>> only use TCP to forward the message on the SIP side but maybe it=0A=
>> is only me!=0A=
=0A=
I doubt it is only you. :-)=0A=
=0A=
Because there is no such thing as XMPP over UDP, the gateway from SIP=0A=
to XMPP will need to communicate over TCP. However, I see no reason=0A=
why the gateway from XMPP to SIP could not be "smart" and send SIP=0A=
messages over TCP or UDP depending on what the far SIP server=0A=
supports. I'll work to add some clarifying text on these points.=0A=
=0A=
Peter=0A=
=0A=
- -- =0A=
Peter Saint-Andre=0A=
https://stpeter.im/=0A=
=0A=
=0A=
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From stpeter@stpeter.im  Tue Sep 24 05:40:17 2013
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Oh, and this text probably belongs in the -core document, I think...

On 9/24/13 6:37 AM, Salvatore Loreto wrote:
> 
> Thanks Peter
> 
> 
> --Original message---
> Sender: "Peter Saint-Andre" <stpeter@stpeter.im>
> Time: Tue Sep 24 15:35:00 EEST 2013
> Cc: stox@ietf.org, 
> Subject: Re: [Stox] review: stox-im-03
> 
> On 9/24/13 12:02 AM, Salvatore Loreto wrote:
>> On 9/24/13 2:45 AM, Peter Saint-Andre wrote: Hi Sal, thanks for the
>> review.
> 
>> On 9/20/13 4:42 AM, Salvatore Loreto wrote:
> 
>>>>> my only comments is about the fact that section 3. XMPP to
>>>>> SIP implies that the MESSAGE is always (and maybe can only
>>>>> be) delivered over a TCP connection; RFC3428 states
>>>>>
>>>>> Whenever possible, MESSAGE requests SHOULD be sent over
>>>>> transports that implement end-to-end congestion control, such
>>>>> as TCP or SCTP. However, SIP does not provide a mechanism to
>>>>> prevent a downstream hop from sending a request over UDP.
>>>>>
>>>>> so you can delivery it over UDP as well and that has also
>>>>> being raised in a previous thread while talking on the
>>>>> Call-ID mapping to <thread/>. Actually you will be forced to
>>>>> deliver over UDP if the SIMPLE server only supports UDP.
>> It's not clear to me why the XMPP-to-SIP gateway couldn't
>> communicate with the SIMPLE server over UDP. Which text in Section
>> 3 do you think implies that only TCP connections are supported?
>>> there is not text at all, and that is the problem. When I read
>>> the Section 3 and the example I got the impression that I can
>>> only use TCP to forward the message on the SIP side but maybe it
>>> is only me!
> 
> I doubt it is only you. :-)
> 
> Because there is no such thing as XMPP over UDP, the gateway from SIP
> to XMPP will need to communicate over TCP. However, I see no reason
> why the gateway from XMPP to SIP could not be "smart" and send SIP
> messages over TCP or UDP depending on what the far SIP server
> supports. I'll work to add some clarifying text on these points.
> 
> Peter
> 
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox
> 

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



From stpeter@stpeter.im  Tue Sep 24 19:18:09 2013
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Here is proposed text for placement at the end of Section 4
("Interdomain Federation") in the -core document:

   Because [RFC6120] specifies a binding of XMPP to TCP, a gateway from
   SIP to XMPP will need to support TCP as the underlying transport
   protocol.  By contrast, as specified in [RFC3261], either TCP or UDP
   can be used as the underlying transport for SIP messages, and a given
   SIP deployment might support only UDP; therefore, a gateway from XMPP
   to SIP might need to communicate with a SIP service using either TCP
   or UDP.

Peter

On 9/24/13 6:40 AM, Peter Saint-Andre wrote:
> Oh, and this text probably belongs in the -core document, I
> think...
> 
> On 9/24/13 6:37 AM, Salvatore Loreto wrote:
>> 
>> Thanks Peter
>> 
>> 
>> --Original message--- Sender: "Peter Saint-Andre"
>> <stpeter@stpeter.im> Time: Tue Sep 24 15:35:00 EEST 2013 Cc:
>> stox@ietf.org, Subject: Re: [Stox] review: stox-im-03
>> 
>> On 9/24/13 12:02 AM, Salvatore Loreto wrote:
>>> On 9/24/13 2:45 AM, Peter Saint-Andre wrote: Hi Sal, thanks for
>>> the review.
>> 
>>> On 9/20/13 4:42 AM, Salvatore Loreto wrote:
>> 
>>>>>> my only comments is about the fact that section 3. XMPP
>>>>>> to SIP implies that the MESSAGE is always (and maybe can
>>>>>> only be) delivered over a TCP connection; RFC3428 states
>>>>>> 
>>>>>> Whenever possible, MESSAGE requests SHOULD be sent over 
>>>>>> transports that implement end-to-end congestion control,
>>>>>> such as TCP or SCTP. However, SIP does not provide a
>>>>>> mechanism to prevent a downstream hop from sending a
>>>>>> request over UDP.
>>>>>> 
>>>>>> so you can delivery it over UDP as well and that has
>>>>>> also being raised in a previous thread while talking on
>>>>>> the Call-ID mapping to <thread/>. Actually you will be
>>>>>> forced to deliver over UDP if the SIMPLE server only
>>>>>> supports UDP.
>>> It's not clear to me why the XMPP-to-SIP gateway couldn't 
>>> communicate with the SIMPLE server over UDP. Which text in
>>> Section 3 do you think implies that only TCP connections are
>>> supported?
>>>> there is not text at all, and that is the problem. When I
>>>> read the Section 3 and the example I got the impression that
>>>> I can only use TCP to forward the message on the SIP side but
>>>> maybe it is only me!
>> 
>> I doubt it is only you. :-)
>> 
>> Because there is no such thing as XMPP over UDP, the gateway from
>> SIP to XMPP will need to communicate over TCP. However, I see no
>> reason why the gateway from XMPP to SIP could not be "smart" and
>> send SIP messages over TCP or UDP depending on what the far SIP
>> server supports. I'll work to add some clarifying text on these
>> points.
>> 
>> Peter
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On 9/23/13 6:25 PM, Peter Saint-Andre wrote:
> On 9/22/13 11:57 PM, Philipp Hancke wrote:
>>> Thanks again for the review. Comments inline.
> 
>> As before, removing items done.
> 
>> new nit: [length] looks like it should be replaced with the 
>> examples actual length in three places. Only seen in -groupchat.
> 
> Will fix.
> 
>> [...]
>>>> example f3 is not shown
>>> 
>>> I don't feel that we need to show every 200 OK. Yes, we put the
>>>  example in example.com, but sometimes you really can have too 
>>> many examples. :-)
> 
>> Right. But then it might make sense to either add a "not shown" 
>> marker in the ascii art or not number them. I'll check if that 
>> happens often enough to make sense.
> 
> Well, maybe we'll just add the examples for the sake of
> completeness.
> 
>> [...]
>>>> 4.4: presence broadcast before ack? must buffer until 
>>>> receiving 110 ack?
>>> 
>>> Your comment is a bit cryptic. Would you please expand it?
> 
>> Hah, that was the only comment you found cryptic? :-)
> 
> I think that I managed to decryptify the others. ;-)
> 
>> btw: it might make sense to reuse the caption from XEP-0045
>> (7.2.3 Presence Broadcast) for the caption of F7 (which is raw,
>> unbridled xmpp and not translated; current text seems copy-paste
>> from F11 in sect 3.5): "Service Sends Presence from Existing
>> Occupants to New Occupant"
> 
> WFM.
> 
>> Now the attempt to decipher my comments: First part: In MUC, the 
>> participant lists is sent to the new occupant (muc example 21) 
>> before the new occupants own presence (muc  example 24) and 
>> therefore potentially before any acknowledgements that might be 
>> required on the MSRP side. Is that a problem?
> 
> Well, neither RFC 4975 nor draft-ietf-simple-chat really specifies
> the handling of participant lists. There is a passing mention in
> the latter document, but it's not very detailed:
> 
> Typically the conference focus acts as a notifier of the
> conference event package, RFC 4575 [RFC4575].  It is RECOMMENDED
> that conference foci and endpoints support RFC 6502 [RFC6502] for
> providing information regarding the conference, and in particular,
> supplying information of the roaster of the conference.  It is also
> RECOMMENDED that conference foci and endpoints support RFC 6501
> [RFC6501], which extends the <user> element originally specified in
> RFC 4575 [RFC4575] with a new 'nickname' attribute.  This allows
> endpoints to learn the nicknames of participants of the chat room.
> 
> However, in draft-ietf-stox-groupchat we assume that the
> conference focus does support the conference-info package for the
> room roster.
> 
> [ Hopefully the RFC Editor will correct "roaster" to "roster" :-)
> ]
> 
>> Second part: In example F10, two (or more) stanzas are
>> translated into a single event. Unless you can do partial updates
>> (Emil? I think you mentioned such a thing for COIN) here, the
>> gateway needs to know when it can dispatch example 4.4 -- upon
>> receiving the participants own presence, which might come with a
>> 110 status code.
> 
> That's all correct.
> 
>> Since I don't think all xmpp serveers do 110, i think there is
>> some text required that the gateway needs to be able to recognize
>> the self-presence and not mistake it as part of the participants
>> list. (should it be participant_s_ presence in the caption?)
> 
> The SIP-to-XMPP gateway will need to maintain state about the MSRP 
> session that the SIP user agent has requested. This implies that
> the gateway will need to know the user's SIP URI and the mapping of
> that URI into an XMPP address, then based on that mapping determine
> when it has received a complete room roster from the MUC room (with
> or without a 110 status code), i.e., when it has received the
> in-room presence of the SIP user (which according to XEP-0045 is
> the last presence stanza received in the initial batch sent after
> joining). Once that happens, the SIP-to-XMPP gateway can construct
> the conference-info document containg the complete room roster and
> send that to the SIP user.
> 
> And yes, we need to make that clear in the spec.

Here is proposed text:

   Because the "room roster" is communicated in XMPP by means of
   multiple presence stanzas (one for each participant) whereas the
   participant list is communicated in SIP by means of a single
   conference-info document, the SIP-to-XMPP gateway will need to keep
   track of the user's SIP URI and the mapping of that URI into an XMPP
   address; then, based on that mapping, it will need to determine when
   it has received a complete room roster from the MUC room, i.e., when
   it has received the in-room presence of the SIP user (which according
   to XEP-0045 is the last presence stanza received in the initial batch
   sent after joining).  Once that happens, the SIP-to-XMPP gateway can
   construct the conference-info document containing the complete room
   roster and send that to the SIP user.

Peter

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


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Subject: [Stox] I-D Action: draft-ietf-stox-core-05.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): Architecture,=
 Addresses, and Error Handling
	Author(s)       : Peter Saint-Andre
                          Avshalom Houri
                          Joe Hildebrand
	Filename        : draft-ietf-stox-core-05.txt
	Pages           : 18
	Date            : 2013-09-25

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-05

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-stox-core-05


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 Markus.Isomaki@nokia.com  Fri Sep 27 04:54:16 2013
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From: <Markus.Isomaki@nokia.com>
To: <stpeter@stpeter.im>, <salvatore.loreto@ericsson.com>, <rjsparks@nostrum.com>, <fluffy@cisco.com>
Thread-Topic: SIPS URIs and SIP/XMPP gateways - WAS: review: stox-core-04
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Subject: [Stox] SIPS URIs and SIP/XMPP gateways - WAS: review: stox-core-04
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Hi,

There was many years ago a lot confusion and discussion about the semantics=
 of the SIPS URIs.

Robert, Cullen: I recall you were there :-) Would you have guidance to the =
STOX WG how SIPS URIs should be dealt with when SIP/XMPP gateways are invol=
ved. Please check Peter's exact question from below.

Regards,
	Markus


> -----Original Message-----
> From: stox-bounces@ietf.org [mailto:stox-bounces@ietf.org] On Behalf Of
> ext Peter Saint-Andre
> Sent: 24 September, 2013 02:55
> To: Salvatore Loreto
> Cc: stox@ietf.org
> Subject: Re: [Stox] review: stox-core-04
>=20
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>=20
> On 9/20/13 3:39 AM, Salvatore Loreto wrote:
> > I have reviewed the core-04 draft, and I think that is almost ready
> > for the WGLC
> >
> > I have one general comment (btw I apologize in advance if this has
> > already been discussed and I have overlooked or forgot it) and it is
> > about the fact that draft does not talk at all about the inter working
> > when secure URIs are involved. I think that at least we should say
> > something about in the Security consideration.
>=20
> Good catch. After reading and re-reading Section 26.4.4 of RFC 3261, I ha=
ve to
> admit that I'm not sure exactly how SIP entities are supposed to handle S=
IPS
> URIs. Even if we understand those requirements, I'm also not sure how the=
y
> would apply to the XMPP side of the communications path. At the least, it
> seems we'd want to specify that if the To header or Request-URI is a SIPS
> URI, then the SIP-to-XMPP gateway needs to connect to the XMPP server
> over a TLS-protected stream. However, do we also need to stipulate that t=
he
> XMPP server-to-client connection is TLS-protected? If so, how would the
> gateway associated with the XMPP server ensure that? (The gateway might
> be an external component of the XMPP server, without control over how the
> XMPP server communicates with XMPP clients.)
>=20
> Peter
>=20
> - --
> Peter Saint-Andre
> https://stpeter.im/
>=20
>=20
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Subject: Re: [Stox] SIPS URIs and SIP/XMPP gateways - WAS: review: stox-core-04
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(Adding Jon)

Peter - is there nothing in XMPP that lets a client say "I want this to 
use secure transports only - have it fail rather than use an insecure 
transport anywhere along its delivery path?"

That's the primary property you should discuss. Without putting a lot of 
thinking into it, I suspect that if you _don't_ have a way to express 
that available (which is what I'm taking away from your last sentence), 
the right guidance in the document is to refuse to gateway a SIP request 
that expresses that requirement.

Maybe Jon has a different opinion?

We can dive into the details of where to look for a sips: URI once we 
think we're on the same page on that high level principle.

RjS



On 9/27/13 6:43 AM, Markus.Isomaki@nokia.com wrote:
> Hi,
>
> There was many years ago a lot confusion and discussion about the semantics of the SIPS URIs.
>
> Robert, Cullen: I recall you were there :-) Would you have guidance to the STOX WG how SIPS URIs should be dealt with when SIP/XMPP gateways are involved. Please check Peter's exact question from below.
>
> Regards,
> 	Markus
>
>
>> -----Original Message-----
>> From: stox-bounces@ietf.org [mailto:stox-bounces@ietf.org] On Behalf Of
>> ext Peter Saint-Andre
>> Sent: 24 September, 2013 02:55
>> To: Salvatore Loreto
>> Cc: stox@ietf.org
>> Subject: Re: [Stox] review: stox-core-04
>>
>> -----BEGIN PGP SIGNED MESSAGE-----
>> Hash: SHA1
>>
>> On 9/20/13 3:39 AM, Salvatore Loreto wrote:
>>> I have reviewed the core-04 draft, and I think that is almost ready
>>> for the WGLC
>>>
>>> I have one general comment (btw I apologize in advance if this has
>>> already been discussed and I have overlooked or forgot it) and it is
>>> about the fact that draft does not talk at all about the inter working
>>> when secure URIs are involved. I think that at least we should say
>>> something about in the Security consideration.
>> Good catch. After reading and re-reading Section 26.4.4 of RFC 3261, I have to
>> admit that I'm not sure exactly how SIP entities are supposed to handle SIPS
>> URIs. Even if we understand those requirements, I'm also not sure how they
>> would apply to the XMPP side of the communications path. At the least, it
>> seems we'd want to specify that if the To header or Request-URI is a SIPS
>> URI, then the SIP-to-XMPP gateway needs to connect to the XMPP server
>> over a TLS-protected stream. However, do we also need to stipulate that the
>> XMPP server-to-client connection is TLS-protected? If so, how would the
>> gateway associated with the XMPP server ensure that? (The gateway might
>> be an external component of the XMPP server, without control over how the
>> XMPP server communicates with XMPP clients.)
>>
>> Peter
>>
>> - --
>> Peter Saint-Andre
>> https://stpeter.im/
>>
>>
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Cc: salvatore.loreto@ericsson.com, fluffy@cisco.com, stox@ietf.org, Jon Peterson <jon.peterson@neustar.biz>, Markus.Isomaki@nokia.com
Subject: Re: [Stox] SIPS URIs and SIP/XMPP gateways - WAS: review: stox-core-04
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

On 9/27/13 7:46 AM, Robert Sparks wrote:
> (Adding Jon)
> 
> Peter - is there nothing in XMPP that lets a client say "I want
> this to use secure transports only - have it fail rather than use
> an insecure transport anywhere along its delivery path?"

No. That doesn't mean we don't need it (although in general people
have thought we *wouldn't* need it if we could just define an
end-to-end encryption method that solve all the relevant use cases).

> That's the primary property you should discuss. Without putting a
> lot of thinking into it, I suspect that if you _don't_ have a way
> to express that available (which is what I'm taking away from your
> last sentence), the right guidance in the document is to refuse to
> gateway a SIP request that expresses that requirement.

Indeed, that seems correct.

Thanks for the guidance.

Peter

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


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From internet-drafts@ietf.org  Sat Sep 28 20:40:54 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): Groupchat
	Author(s)       : Peter Saint-Andre
                          Saul Ibarra
                          Salvatore Loreto
	Filename        : draft-ietf-stox-groupchat-01.txt
	Pages           : 34
	Date            : 2013-09-28

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-01

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


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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Subject: Re: [Stox] SIPS URIs and SIP/XMPP gateways - WAS: review: stox-core-04
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

On 9/27/13 10:14 AM, Peter Saint-Andre wrote:
> On 9/27/13 7:46 AM, Robert Sparks wrote:
>> (Adding Jon)
> 
>> Peter - is there nothing in XMPP that lets a client say "I want 
>> this to use secure transports only - have it fail rather than
>> use an insecure transport anywhere along its delivery path?"
> 
> No. That doesn't mean we don't need it (although in general people 
> have thought we *wouldn't* need it if we could just define an 
> end-to-end encryption method that solve all the relevant use
> cases).
> 
>> That's the primary property you should discuss. Without putting
>> a lot of thinking into it, I suspect that if you _don't_ have a
>> way to express that available (which is what I'm taking away from
>> your last sentence), the right guidance in the document is to
>> refuse to gateway a SIP request that expresses that requirement.
> 
> Indeed, that seems correct.
> 
> Thanks for the guidance.

Here is proposed text:

   As specified in Section 26.4.4 of [RFC3261], a To header or a
   Request-URI containing a SIPS URI is used to indicate that all hops
   in a communication path need to be protected using Transport Layer
   Security [RFC5246].  Because XMPP lacks a way to signal that all hops
   need to be encrypted, if the To header or Request-URI of a SIP
   message is a SIPS URI then the SIP-to-XMPP gateway MUST NOT translate
   the SIP message into an XMPP stanza and MUST NOT route it to the
   destination XMPP server.

Peter

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


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All of the specs *should* now reference the -core I-D when describing
how a SIP server or an XMPP server routes data to a remote domain that
supports the other protocol. However, I've noticed that not all of the
specs do this (or do so consistently). I'll make an effort to clean up
the specs and submit revised I-Ds.

Peter

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


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From Markus.Isomaki@nokia.com  Mon Sep 30 04:08:25 2013
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From: <Markus.Isomaki@nokia.com>
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Subject: Re: [Stox] inconsistent text
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Hi Peter,

OK. After you have updated both the routing related part and the sips URI h=
andling in -core and submitted the updated versions, we can initiate WG las=
t calls on -core, -im and -presence with those updated versions.

Markus=20

> -----Original Message-----
> From: stox-bounces@ietf.org [mailto:stox-bounces@ietf.org] On Behalf Of
> ext Peter Saint-Andre
> Sent: 30 September, 2013 05:22
> To: stox@ietf.org
> Subject: [Stox] inconsistent text
>=20
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>=20
> All of the specs *should* now reference the -core I-D when describing how=
 a
> SIP server or an XMPP server routes data to a remote domain that supports
> the other protocol. However, I've noticed that not all of the specs do th=
is (or
> do so consistently). I'll make an effort to clean up the specs and submit
> revised I-Ds.
>=20
> Peter
>=20
> - --
> Peter Saint-Andre
> https://stpeter.im/
>=20
>=20
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From stpeter@stpeter.im  Mon Sep 30 05:48:52 2013
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On 9/30/13 5:01 AM, Markus.Isomaki@nokia.com wrote:
> Hi Peter,
> 
> OK. After you have updated both the routing related part and the sips URI handling in -core and submitted the updated versions, we can initiate WG last calls on -core, -im and -presence with those updated versions.

Sounds good. I made those fixes last night, so I can submit the I-Ds
this morning.

Peter

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



From rjsparks@nostrum.com  Mon Sep 30 06:37:28 2013
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Cc: salvatore.loreto@ericsson.com, fluffy@cisco.com, Markus.Isomaki@nokia.com, Jon Peterson <jon.peterson@neustar.biz>, stox@ietf.org
Subject: Re: [Stox] SIPS URIs and SIP/XMPP gateways - WAS: review: stox-core-04
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On 9/29/13 8:55 PM, Peter Saint-Andre wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> On 9/27/13 10:14 AM, Peter Saint-Andre wrote:
>> On 9/27/13 7:46 AM, Robert Sparks wrote:
>>> (Adding Jon)
>>> Peter - is there nothing in XMPP that lets a client say "I want
>>> this to use secure transports only - have it fail rather than
>>> use an insecure transport anywhere along its delivery path?"
>> No. That doesn't mean we don't need it (although in general people
>> have thought we *wouldn't* need it if we could just define an
>> end-to-end encryption method that solve all the relevant use
>> cases).
>>
>>> That's the primary property you should discuss. Without putting
>>> a lot of thinking into it, I suspect that if you _don't_ have a
>>> way to express that available (which is what I'm taking away from
>>> your last sentence), the right guidance in the document is to
>>> refuse to gateway a SIP request that expresses that requirement.
>> Indeed, that seems correct.
>>
>> Thanks for the guidance.
> Here is proposed text:
>
>     As specified in Section 26.4.4 of [RFC3261], a To header or a
>     Request-URI containing a SIPS URI is used to indicate that all hops
>     in a communication path need to be protected using Transport Layer
>     Security [RFC5246].  Because XMPP lacks a way to signal that all hops
>     need to be encrypted, if the To header or Request-URI of a SIP
>     message is a SIPS URI then the SIP-to-XMPP gateway MUST NOT translate
>     the SIP message into an XMPP stanza and MUST NOT route it to the
>     destination XMPP server.
wfm. You might also talk about not using sips when going XMPP->SIP.
If you haven't found it already, see also RFC5630.

RjS
>
> Peter
>
> - -- 
> Peter Saint-Andre
> https://stpeter.im/
>
>
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To: Robert Sparks <rjsparks@nostrum.com>
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Cc: fluffy@cisco.com, Jon Peterson <jon.peterson@neustar.biz>, stox@ietf.org, "Olle E. Johansson" <oej@edvina.net>, Markus.Isomaki@nokia.com, Peter Saint-Andre <stpeter@stpeter.im>, salvatore.loreto@ericsson.com
Subject: Re: [Stox] SIPS URIs and SIP/XMPP gateways - WAS: review: stox-core-04
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30 sep 2013 kl. 15:37 skrev Robert Sparks <rjsparks@nostrum.com>:

> On 9/29/13 8:55 PM, Peter Saint-Andre wrote:
>> -----BEGIN PGP SIGNED MESSAGE-----
>> Hash: SHA1
>>=20
>> On 9/27/13 10:14 AM, Peter Saint-Andre wrote:
>>> On 9/27/13 7:46 AM, Robert Sparks wrote:
>>>> (Adding Jon)
>>>> Peter - is there nothing in XMPP that lets a client say "I want
>>>> this to use secure transports only - have it fail rather than
>>>> use an insecure transport anywhere along its delivery path?"
>>> No. That doesn't mean we don't need it (although in general people
>>> have thought we *wouldn't* need it if we could just define an
>>> end-to-end encryption method that solve all the relevant use
>>> cases).
>>>=20
>>>> That's the primary property you should discuss. Without putting
>>>> a lot of thinking into it, I suspect that if you _don't_ have a
>>>> way to express that available (which is what I'm taking away from
>>>> your last sentence), the right guidance in the document is to
>>>> refuse to gateway a SIP request that expresses that requirement.
>>> Indeed, that seems correct.
>>>=20
>>> Thanks for the guidance.
>> Here is proposed text:
>>=20
>>    As specified in Section 26.4.4 of [RFC3261], a To header or a
>>    Request-URI containing a SIPS URI is used to indicate that all =
hops
>>    in a communication path need to be protected using Transport Layer
>>    Security [RFC5246].  Because XMPP lacks a way to signal that all =
hops
>>    need to be encrypted, if the To header or Request-URI of a SIP
>>    message is a SIPS URI then the SIP-to-XMPP gateway MUST NOT =
translate
>>    the SIP message into an XMPP stanza and MUST NOT route it to the
>>    destination XMPP server.
> wfm. You might also talk about not using sips when going XMPP->SIP.
> If you haven't found it already, see also RFC5630.

Agree. It will help a lot of developers to explain that even if you find =
NAPTR
records indicating that a destination ONLY supports TLS connections,
you will look up _sips._tcp in SRV but this does NOT mean that you are =
going
to use a SIPS: uri. The _sips SRV tag and the SIPS: uri are =
unfortunately
very similar but not directly related in a confusing way :-)

/O=

From fippo@goodadvice.pages.de  Mon Sep 30 07:12:33 2013
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Subject: Re: [Stox] review of core, chat, groupchat and presence
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>>> I don't feel that we need to show every 200 OK. Yes, we put the
>>> example in example.com, but sometimes you really can have too
>>> many examples. :-)
>>
>> Right. But then it might make sense to either add a "not shown"
>> marker in the ascii art or not number them. I'll check if that
>> happens often enough to make sense.
>
> Well, maybe we'll just add the examples for the sake of completeness.

humm, just noticed that -groupchat is odd here while checking this.
X4 is called ACK in both the diagram and example 4 (and is a sip ack)
whereas the S3 ACK in the diagram is a 200 in example 28.
I think it should be ACK in example 28.

I think not numbering things that are not shown (e.g. S2) would be 
consistent.

From stpeter@stpeter.im  Mon Sep 30 07:56:53 2013
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Cc: fluffy@cisco.com, Jon Peterson <jon.peterson@neustar.biz>, stox@ietf.org, Markus.Isomaki@nokia.com, salvatore.loreto@ericsson.com, Robert Sparks <rjsparks@nostrum.com>
Subject: Re: [Stox] SIPS URIs and SIP/XMPP gateways - WAS: review: stox-core-04
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On 9/30/13 7:43 AM, Olle E. Johansson wrote:
> 
> 30 sep 2013 kl. 15:37 skrev Robert Sparks <rjsparks@nostrum.com>:
> 
>> On 9/29/13 8:55 PM, Peter Saint-Andre wrote:
>>> -----BEGIN PGP SIGNED MESSAGE-----
>>> Hash: SHA1
>>>
>>> On 9/27/13 10:14 AM, Peter Saint-Andre wrote:
>>>> On 9/27/13 7:46 AM, Robert Sparks wrote:
>>>>> (Adding Jon)
>>>>> Peter - is there nothing in XMPP that lets a client say "I want
>>>>> this to use secure transports only - have it fail rather than
>>>>> use an insecure transport anywhere along its delivery path?"
>>>> No. That doesn't mean we don't need it (although in general people
>>>> have thought we *wouldn't* need it if we could just define an
>>>> end-to-end encryption method that solve all the relevant use
>>>> cases).
>>>>
>>>>> That's the primary property you should discuss. Without putting
>>>>> a lot of thinking into it, I suspect that if you _don't_ have a
>>>>> way to express that available (which is what I'm taking away from
>>>>> your last sentence), the right guidance in the document is to
>>>>> refuse to gateway a SIP request that expresses that requirement.
>>>> Indeed, that seems correct.
>>>>
>>>> Thanks for the guidance.
>>> Here is proposed text:
>>>
>>>    As specified in Section 26.4.4 of [RFC3261], a To header or a
>>>    Request-URI containing a SIPS URI is used to indicate that all hops
>>>    in a communication path need to be protected using Transport Layer
>>>    Security [RFC5246].  Because XMPP lacks a way to signal that all hops
>>>    need to be encrypted, if the To header or Request-URI of a SIP
>>>    message is a SIPS URI then the SIP-to-XMPP gateway MUST NOT translate
>>>    the SIP message into an XMPP stanza and MUST NOT route it to the
>>>    destination XMPP server.
>> wfm. You might also talk about not using sips when going XMPP->SIP.
>> If you haven't found it already, see also RFC5630.
> 
> Agree. It will help a lot of developers to explain that even if you find NAPTR
> records indicating that a destination ONLY supports TLS connections,
> you will look up _sips._tcp in SRV but this does NOT mean that you are going
> to use a SIPS: uri. The _sips SRV tag and the SIPS: uri are unfortunately
> very similar but not directly related in a confusing way :-)

Thanks for the notes. I'll review those today.

Peter

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



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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): Architecture,=
 Addresses, and Error Handling
	Author(s)       : Peter Saint-Andre
                          Avshalom Houri
                          Joe Hildebrand
	Filename        : draft-ietf-stox-core-06.txt
	Pages           : 19
	Date            : 2013-09-30

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

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Subject: [Stox] I-D Action: draft-ietf-stox-presence-05.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
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	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-05.txt
	Pages           : 23
	Date            : 2013-09-30

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).


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	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-04.txt
	Pages           : 10
	Date            : 2013-09-30

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).


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Date: Tue, 01 Oct 2013 08:48:54 +0200
From: Daniel-Constantin Mierla <miconda@gmail.com>
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Subject: Re: [Stox] I-D Action: draft-ietf-stox-im-04.txt
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Hello,

the SIP message example in chapter 4. SIP to XMPP has the From header 
without tag parameter. Should be like:

From: <sip:romeo@example.net>;tag=xyz

Daniel

On 9/30/13 11:56 PM, internet-drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>   This draft is a work item of the SIP-TO-XMPP Working Group of the IETF.
>
> 	Title           : Interworking between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP): Instant Messaging
> 	Author(s)       : Peter Saint-Andre
>                            Avshalom Houri
>                            Joe Hildebrand
> 	Filename        : draft-ietf-stox-im-04.txt
> 	Pages           : 10
> 	Date            : 2013-09-30
>
> 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-04
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-stox-im-04
>
>
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
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-- 
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