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From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
To: The IESG <iesg@ietf.org>
Thread-Topic: WG Review: SIP-TO-XMPP (stox)
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Hi,

I have two comments concerning this charter.=20

1. The way the Objectives section is written does not make clear whether th=
e current scope of stox is dully defined as making out of the draft-saintan=
dre-sip-xmpp-* Proposed Standard RFCs, or whether other proposals for the s=
ame mapping functionality (if submitted) will be considered.=20

2. It is very unusual that there are no proposed milestones for a document =
submitted for public review. I am pretty sure that the IESG will not approv=
ed the charter without seeing the milestones scheduled, so I suggest that t=
hey are made public before the IESG discussion.=20

Thanks and Regards,

Dan




Subject: WG Review: SIP-TO-XMPP (stox)

A new IETF working group has been proposed in the Real-time Applications an=
d Infrastructure Area. The IESG has not made any determination yet. The fol=
lowing draft charter was submitted, and is provided for informational purpo=
ses only. Please send your comments to the IESG mailing list (iesg at ietf.=
org) by 2013-06-10.

SIP-TO-XMPP (stox)
------------------------------------------------
Current Status: Proposed WG

Chairs:
  Markus Isomaki <markus.isomaki@nokia.com>
  Yana Stamcheva <yana@jitsi.org>

Assigned Area Director:
  Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>

Mailing list
  Address: stox@ietf.org
  To Subscribe: https://www.ietf.org/mailman/listinfo/stox
  Archive: http://www.ietf.org/mail-archive/web/stox/

Charter:

Problem Statement

The IETF has defined two signalling technologies that can be used for multi=
media session negotiation, instant messaging, presence, file transfer, capa=
bilities discovery, notifications, and other types of real-time functionali=
ty:

o  The Session Initiation Protocol (SIP), along with various SIP
   extensions developed within the SIP for Instant Messaging and
   Presence Leveraging Extensions (SIMPLE) Working Group.

o  The Extensible Messaging and Presence Protocol (XMPP), along
   with various XMPP extensions developed by the IETF as well as by
   the XMPP Standards Foundation.

SIP has been focused primarily on media session negotiation (e.g., audio an=
d video), whereas XMPP has been focused primarily on messaging and presence=
.  As a result, the technologies are mostly complementary.
However, there is also some overlap between SIP and XMPP, since there are S=
IP extensions for messaging, presence, groupchat, file transfer, etc., and =
there are XMPP extensions for multimedia session negotiation.
This overlap has practical implications, since some deployed services use S=
IP for both media and messaging/presence, whereas other deployed services u=
se XMPP for both messaging/presence and media.  When such services wish to =
exchange information, they often need to translate their native protocol (e=
ither SIP or XMPP) to the other protocol (either XMPP or SIP).

Implementers needing to perform such protocol mappings have often worked ou=
t their own heuristics for doing so.  Unfortunately, these heuristics are n=
ot always consistent, which can lead to interoperability problems.

Objectives

To make it easier for implementers to enable interworking between SIP-based=
 systems and XMPP-based systems, several Internet-Drafts have defined guide=
lines for protocol mapping between SIP and XMPP, starting with draft-sainta=
ndre-xmpp-simple-00 in early 2004.  The current documents are:

-core
draft-saintandre-sip-xmpp-presence
draft-saintandre-sip-xmpp-im
draft-saintandre-sip-xmpp-chat
draft-saintandre-sip-xmpp-groupchat
draft-saintandre-sip-xmpp-media

These documents are fairly stable and the authors have received feedback fr=
om a number of implementers over the years.  However, implementers do not a=
lways know about these documents because they are Internet-Drafts and somet=
imes they have become expired due to inactivity.  Thus it would be helpful =
to polish them off and publish them as RFCs.

It might also be helpful to at some point publish additional documents in

the same series, covering topics like capabilities discovery and file trans=
fer.  However, any such work would require a recharter.

The group shall not be tasked with defining any new protocols, only with sp=
ecifying mappings between existing protocols that have been defined for SIP=
 and XMPP.

Deliverables

1. Address mapping and error handling
2. Presence mapping
3. Mapping for single instant messages
4. Mapping for one-to-one text chat sessions 5. Mapping for multi-user text=
 chat sessions 6. Mapping for media signaling

All of the foregoing deliverables are standards track, since they are subje=
ct to interoperability testing.

Milestones:
TBD


From stpeter@stpeter.im  Mon Jun  3 15:38:14 2013
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Subject: Re: [Stox] WG Review: SIP-TO-XMPP (stox)
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On 6/3/13 4:58 AM, Romascanu, Dan (Dan) wrote:
> Hi,
> 
> I have two comments concerning this charter.
> 
> 1. The way the Objectives section is written does not make clear
> whether the current scope of stox is dully defined as making out of
> the draft-saintandre-sip-xmpp-* Proposed Standard RFCs, or whether
> other proposals for the same mapping functionality (if submitted)
> will be considered.

Your phrase "dully defined" is spot on: this topic is so dull that I
very much doubt that other proposals will be forthcoming! However, you
are right that the charter should probably say that those I-D are likely
(but not certain) starting points.

How about this?

OLD
These documents are fairly stable and the authors have received feedback
from a number of implementers over the years.  However, implementers do
not always know about these documents because they are Internet-Drafts
and sometimes they have become expired due to inactivity.  Thus it would
be helpful to polish them off and publish them as RFCs.

NEW
Those documents are fairly stable and the authors have received feedback
from a number of implementers over the years.  However, implementers do
not always know that such mapping specifications exist, because they are
Internet-Drafts and sometimes they have become expired due to
inactivity.  Thus it would be helpful to publish a set of mapping
specifications as RFCs; the foregoing Internet-Drafts provide likely
starting points, but other proposals are welcome as per normal IETF
working group processes.

> 2. It is very unusual that there are no proposed milestones for a
> document submitted for public review. I am pretty sure that the IESG
> will not approved the charter without seeing the milestones
> scheduled, so I suggest that they are made public before the IESG
> discussion.

Personally I think that they can all be submitted for IESG review before
IETF 88, and perhaps even well before that.

Peter

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



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From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
To: Peter Saint-Andre <stpeter@stpeter.im>
Thread-Topic: [Stox] WG Review: SIP-TO-XMPP (stox)
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Hi Peter,=20

Thank you for your answers.=20

Please see in-line.=20

Regards,

Dan




> -----Original Message-----
> From: Peter Saint-Andre [mailto:stpeter@stpeter.im]
> Sent: Tuesday, June 04, 2013 1:33 AM
> To: Romascanu, Dan (Dan)
> Cc: The IESG; stox@ietf.org
> Subject: Re: [Stox] WG Review: SIP-TO-XMPP (stox)
>=20
> On 6/3/13 4:58 AM, Romascanu, Dan (Dan) wrote:
> > Hi,
> >
> > I have two comments concerning this charter.
> >
> > 1. The way the Objectives section is written does not make clear
> > whether the current scope of stox is dully defined as making out of
> > the draft-saintandre-sip-xmpp-* Proposed Standard RFCs, or whether
> > other proposals for the same mapping functionality (if submitted) will
> > be considered.
>=20
> Your phrase "dully defined" is spot on: this topic is so dull that I
> very much doubt that other proposals will be forthcoming! However, you
> are right that the charter should probably say that those I-D are likely
> (but not certain) starting points.
>=20
> How about this?
>=20
> OLD
> These documents are fairly stable and the authors have received feedback
> from a number of implementers over the years.  However, implementers do
> not always know about these documents because they are Internet-Drafts
> and sometimes they have become expired due to inactivity.  Thus it would
> be helpful to polish them off and publish them as RFCs.
>=20
> NEW
> Those documents are fairly stable and the authors have received feedback
> from a number of implementers over the years.  However, implementers do
> not always know that such mapping specifications exist, because they are
> Internet-Drafts and sometimes they have become expired due to
> inactivity.  Thus it would be helpful to publish a set of mapping
> specifications as RFCs; the foregoing Internet-Drafts provide likely
> starting points, but other proposals are welcome as per normal IETF
> working group processes.
>=20
[[DR]] looks good

> > 2. It is very unusual that there are no proposed milestones for a
> > document submitted for public review. I am pretty sure that the IESG
> > will not approved the charter without seeing the milestones scheduled,
> > so I suggest that they are made public before the IESG discussion.
>=20
> Personally I think that they can all be submitted for IESG review before
> IETF 88, and perhaps even well before that.
>=20
[[DR]] so October 2013 as submission date for all documents? I suggest to a=
dd these to the proposed charter (unless somebody disagrees)

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


From Markus.Isomaki@nokia.com  Tue Jun  4 03:47:55 2013
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From: <Markus.Isomaki@nokia.com>
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Hi Dan, Peter,

I agree with Peter on that there are not that many different, yet meaningfu=
l, ways to do protocol mappings between SIP and XMPP. The current drafts ha=
ve been around for a while and the all comments so far have been generally =
positive that they are a good way to go. So, it is unlikely that we would r=
eceive radically different and "competing" proposals. Having said that, I a=
gree we should leave the opportunity for anyone to submit such proposals fo=
r WG's consideration. I believe Peter's new charter text proposal covers th=
at.=20

Markus =20

-----Original Message-----
From: stox-bounces@ietf.org [mailto:stox-bounces@ietf.org] On Behalf Of ext=
 Peter Saint-Andre
Sent: 04 June, 2013 01:33
To: Romascanu, Dan (Dan)
Cc: stox@ietf.org; The IESG
Subject: Re: [Stox] WG Review: SIP-TO-XMPP (stox)

On 6/3/13 4:58 AM, Romascanu, Dan (Dan) wrote:
> Hi,
>=20
> I have two comments concerning this charter.
>=20
> 1. The way the Objectives section is written does not make clear=20
> whether the current scope of stox is dully defined as making out of=20
> the draft-saintandre-sip-xmpp-* Proposed Standard RFCs, or whether=20
> other proposals for the same mapping functionality (if submitted) will=20
> be considered.

Your phrase "dully defined" is spot on: this topic is so dull that I very m=
uch doubt that other proposals will be forthcoming! However, you are right =
that the charter should probably say that those I-D are likely (but not cer=
tain) starting points.

How about this?

OLD
These documents are fairly stable and the authors have received feedback fr=
om a number of implementers over the years.  However, implementers do not a=
lways know about these documents because they are Internet-Drafts and somet=
imes they have become expired due to inactivity.  Thus it would be helpful =
to polish them off and publish them as RFCs.

NEW
Those documents are fairly stable and the authors have received feedback fr=
om a number of implementers over the years.  However, implementers do not a=
lways know that such mapping specifications exist, because they are Interne=
t-Drafts and sometimes they have become expired due to inactivity.  Thus it=
 would be helpful to publish a set of mapping specifications as RFCs; the f=
oregoing Internet-Drafts provide likely starting points, but other proposal=
s are welcome as per normal IETF working group processes.

> 2. It is very unusual that there are no proposed milestones for a=20
> document submitted for public review. I am pretty sure that the IESG=20
> will not approved the charter without seeing the milestones scheduled,=20
> so I suggest that they are made public before the IESG discussion.

Personally I think that they can all be submitted for IESG review before IE=
TF 88, and perhaps even well before that.

Peter

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


_______________________________________________
stox mailing list
stox@ietf.org
https://www.ietf.org/mailman/listinfo/stox

From Markus.Isomaki@nokia.com  Tue Jun  4 05:08:24 2013
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From: <Markus.Isomaki@nokia.com>
To: <stox@ietf.org>
Thread-Topic: sip-xmpp-core: gateway in source or destination domain?
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Hi,

The sip-xmpp-core draft [1] says:

 "We further assume that protocol translation
   will occur within a gateway in the source domain, so that information
   generated by the user of an XMPP service will be translated by a
   gateway within the trust domain of that XMPP service, and information
   generated by the user of a SIP service will be translated by a
   gateway within the trust domain of that SIP service."

In the context of STOX, I have a few questions about this:
- Is this how the current deployments in general work? Is that enough?
- Should we keep this assumption or also explicitly consider the case where=
 the gateway would be in the destination domain?
- If we wanted to take destination-domain gateways into consideration, what=
 would the implications be to the mapping specs?=20

Regards,
	Markus=20

[1] http://datatracker.ietf.org/doc/draft-saintandre-sip-xmpp-core/





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Hi,

A high-level comment about sip-xmpp-media [1]:

SDP/Jingle Offer/Answer and ICE are moving targets by themselves. For insta=
nce right now MMUSIC WG is working on new features such as "trickle-ICE" (a=
dding new candidates on the fly - I know this is already supported in Jingl=
e) and "bundle" (multiplexing several RTP sessions/streams into a single tr=
ansport-level packet flow).

We need to ensure that the sip-xmpp-media RFC-to-become is generic enough t=
o stay valid even when such extensions are developed, and we have a story o=
n how and where the SIP-XMPP mappings to various extensions can be develope=
d if needed.

For instance "bundle", AFAIK, is going to be entirely SDP/Jingle level issu=
e, so presumably it will be enough that after the MMUSIC SDP spec is done, =
a mapping spec to Jingle is written in XSF? On the other hand, trickle-ICE,=
 AFAIK, will require the use of SIP INFO (gasp! ;-) method to carry the new=
 candidates. That seems like something that would require an extension (" a=
 new RFC") to sip-xmpp-media in the IETF, explaining how Jingle's trickle i=
s mapped to SIP's trickle.

Regards,
                Markus

[1] http://datatracker.ietf.org/doc/draft-saintandre-sip-xmpp-media/



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<p class=3D"MsoNormal">Hi,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">A high-level comment about sip-xmpp-media [1]:<o:p><=
/o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">SDP/Jingle Offer/Answer and ICE are moving targets b=
y themselves. For instance right now MMUSIC WG is working on new features s=
uch as &#8220;trickle-ICE&#8221; (adding new candidates on the fly &#8211; =
I know this is already supported in Jingle) and &#8220;bundle&#8221;
 (multiplexing several RTP sessions/streams into a single transport-level p=
acket flow).
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">We need to ensure that the sip-xmpp-media RFC-to-bec=
ome is generic enough to stay valid even when such extensions are developed=
, and we have a story on how and where the SIP-XMPP mappings to various ext=
ensions can be developed if needed.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">For instance &#8220;bundle&#8221;, AFAIK, is going t=
o be entirely SDP/Jingle level issue, so presumably it will be enough that =
after the MMUSIC SDP spec is done, a mapping spec to Jingle is written in X=
SF? On the other hand, trickle-ICE, AFAIK, will
 require the use of SIP INFO (gasp! ;-) method to carry the new candidates.=
 That seems like something that would require an extension (&#8220; a new R=
FC&#8221;) to sip-xmpp-media in the IETF, explaining how Jingle&#8217;s tri=
ckle is mapped to SIP&#8217;s trickle. &nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Markus <o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">[1] <a href=3D"http://datatracker.ietf.org/doc/draft=
-saintandre-sip-xmpp-media/">
http://datatracker.ietf.org/doc/draft-saintandre-sip-xmpp-media/</a><o:p></=
o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</body>
</html>

--_000_E44893DD4E290745BB608EB23FDDB7620A00461C008AM1MPN1043mg_--

From ag@ag-projects.com  Tue Jun  4 07:04:09 2013
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To: <Markus.Isomaki@nokia.com> <Markus.Isomaki@nokia.com>
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Subject: Re: [Stox] sip-xmpp-core: gateway in source or destination domain?
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On Jun 4, 2013, at 12:46 PM, <Markus.Isomaki@nokia.com> =
<Markus.Isomaki@nokia.com> wrote:

> Hi,
>=20
> The sip-xmpp-core draft [1] says:
>=20
> "We further assume that protocol translation
>   will occur within a gateway in the source domain, so that =
information
>   generated by the user of an XMPP service will be translated by a
>   gateway within the trust domain of that XMPP service, and =
information
>   generated by the user of a SIP service will be translated by a
>   gateway within the trust domain of that SIP service."
>=20
> In the context of STOX, I have a few questions about this:
> - Is this how the current deployments in general work? Is that enough?

I deployed this service and and this is how it works.  It is enough in =
my experience and doe snot require any new addressing scheme or location =
service.=20

> - Should we keep this assumption or also explicitly consider the case =
where the gateway would be in the destination domain?

Not sure how one could know what gateways the foreign domain has and for =
what purpose. One can only lookup DNS records for specific services like =
SIP or XMPP. You do not know if they are gateways or something else. The =
local domain policy can detect a remote domain capability and route =
based its own policy.

> - If we wanted to take destination-domain gateways into consideration, =
what would the implications be to the mapping specs?=20

One can only decide what to do for its own domain and advertise it into =
the DNS. One cannot guess or know about topology of a foreign domain, =
can't he?

Adrian

> Regards,
> 	Markus=20
>=20
> [1] http://datatracker.ietf.org/doc/draft-saintandre-sip-xmpp-core/
>=20
>=20
>=20
>=20
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox
>=20


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From: <Markus.Isomaki@nokia.com>
To: <ag@ag-projects.com>
Thread-Topic: [Stox] sip-xmpp-core: gateway in source or destination domain?
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Subject: Re: [Stox] sip-xmpp-core: gateway in source or destination domain?
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Hi Adrian,

Thanks. One clarification about your deployment:

Adrian Georgescu [mailto:ag@ag-projects.com] wrote:

>> - Is this how the current deployments in general work? Is that enough?

> I deployed this service and and this is how it works.  It is enough in my=
 experience and doe snot require any new addressing scheme or location serv=
ice.

So, when one domain runs SIP and the other domain runs XMPP, in order to ta=
lk to each other, each domain is responsible for deploying a gateway that t=
ranslates outgoing messages from their "native" protocol to the other proto=
col? It is not enough to just deploy a gateway in one of the domains (and h=
ave that domain advertise both SIP and XMPP in DNS)?

Current "sip-xmpp-core" draft explicitly assumes that SIP messages are tran=
slated to XMPP in the SIP domain, while XMPP messages are translated to SIP=
 in the XMPP domain.=20

What I was wondering is how that assumption in practice affects the draft-s=
aintandre-sip-xmpp-* specs in general. To my understanding the mappings are=
 the same no matter in which trust domain the gateway is located. The sip-x=
mpp-core does talk about the DNS operations which are certainly affected de=
pending on where the translation happens.=20

Markus=20


-----Original Message-----
From: ext Adrian Georgescu [mailto:ag@ag-projects.com]=20
Sent: 04 June, 2013 15:30
To: Isomaki Markus (Nokia-CIC/Espoo)
Cc: stox@ietf.org
Subject: Re: [Stox] sip-xmpp-core: gateway in source or destination domain?


On Jun 4, 2013, at 12:46 PM, <Markus.Isomaki@nokia.com> <Markus.Isomaki@nok=
ia.com> wrote:

> Hi,
>=20
> The sip-xmpp-core draft [1] says:
>=20
> "We further assume that protocol translation
>   will occur within a gateway in the source domain, so that information
>   generated by the user of an XMPP service will be translated by a
>   gateway within the trust domain of that XMPP service, and information
>   generated by the user of a SIP service will be translated by a
>   gateway within the trust domain of that SIP service."
>=20
> In the context of STOX, I have a few questions about this:
> - Is this how the current deployments in general work? Is that enough?

I deployed this service and and this is how it works.  It is enough in my e=
xperience and doe snot require any new addressing scheme or location servic=
e.=20

> - Should we keep this assumption or also explicitly consider the case whe=
re the gateway would be in the destination domain?

Not sure how one could know what gateways the foreign domain has and for wh=
at purpose. One can only lookup DNS records for specific services like SIP =
or XMPP. You do not know if they are gateways or something else. The local =
domain policy can detect a remote domain capability and route based its own=
 policy.

> - If we wanted to take destination-domain gateways into consideration, wh=
at would the implications be to the mapping specs?=20

One can only decide what to do for its own domain and advertise it into the=
 DNS. One cannot guess or know about topology of a foreign domain, can't he=
?

Adrian

> Regards,
> 	Markus=20
>=20
> [1] http://datatracker.ietf.org/doc/draft-saintandre-sip-xmpp-core/
>=20
>=20
>=20
>=20
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox
>=20


From ag@ag-projects.com  Tue Jun  4 08:30:04 2013
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Subject: Re: [Stox] sip-xmpp-core: gateway in source or destination domain?
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On Jun 4, 2013, at 3:25 PM, <Markus.Isomaki@nokia.com> wrote:

> Hi Adrian,
>=20
> Thanks. One clarification about your deployment:
>=20
> Adrian Georgescu [mailto:ag@ag-projects.com] wrote:
>=20
>>> - Is this how the current deployments in general work? Is that =
enough?
>=20
>> I deployed this service and and this is how it works.  It is enough =
in my experience and doe snot require any new addressing scheme or =
location service.
>=20
> So, when one domain runs SIP and the other domain runs XMPP, in order =
to talk to each other, each domain is responsible for deploying a =
gateway that translates outgoing messages from their "native" protocol =
to the other protocol? It is not enough to just deploy a gateway in one =
of the domains (and have that domain advertise both SIP and XMPP in =
DNS)?
>=20
> Current "sip-xmpp-core" draft explicitly assumes that SIP messages are =
translated to XMPP in the SIP domain, while XMPP messages are translated =
to SIP in the XMPP domain.=20

This is most likely not possible in real life because it would require =
that both domains have such gateway in place and in practice the chance =
of this happening is close to zero (unless adoption of such solution =
goes through the roof, unlikely). Most likely the SIP/XMPP gateway is =
deployed only by those who care about such interoperability, the remote =
domains may not care and they should not know or be required to do =
anything about this, it should just work.

My use case is the following.=20

I serve SIP users and I want to provide my users with the possibility to =
interact with remote XMPP domains without having to know or configure =
anything in their SIP clients. For this purpose, I have configured my =
SIP server to route outgoing requests for that XMPP remote domain =
through the gateway I control. For receiving incoming requests from any =
remote XMPP domains, I configured the proper XMPP related  DNS records =
to point to the same gateway (its XMPP server listening interface). This =
was it works both ways and the remote XMPP domain does not need to have =
a gateway not does he  need to know anything about my topology or the =
fact that I have a gateway, it just talks native XMPP as it would =
federate with any other XMPP foreign domain.


>=20
> What I was wondering is how that assumption in practice affects the =
draft-saintandre-sip-xmpp-* specs in general. To my understanding the =
mappings are the same no matter in which trust domain the gateway is =
located. The sip-xmpp-core does talk about the DNS operations which are =
certainly affected depending on where the translation happens.=20

Again, requiring both domains to have such gateway is both unnecessary =
and unrealistic. Nobody can be forced to deploy a gateway just because a =
foreign domain would lkike or require so. If the remote domain cares =
about talking my protocol, they should deploy a gateway on their own and =
it should just talk to my service natively.

Adrian


> Markus=20
>=20
>=20
> -----Original Message-----
> From: ext Adrian Georgescu [mailto:ag@ag-projects.com]=20
> Sent: 04 June, 2013 15:30
> To: Isomaki Markus (Nokia-CIC/Espoo)
> Cc: stox@ietf.org
> Subject: Re: [Stox] sip-xmpp-core: gateway in source or destination =
domain?
>=20
>=20
> On Jun 4, 2013, at 12:46 PM, <Markus.Isomaki@nokia.com> =
<Markus.Isomaki@nokia.com> wrote:
>=20
>> Hi,
>>=20
>> The sip-xmpp-core draft [1] says:
>>=20
>> "We further assume that protocol translation
>>  will occur within a gateway in the source domain, so that =
information
>>  generated by the user of an XMPP service will be translated by a
>>  gateway within the trust domain of that XMPP service, and =
information
>>  generated by the user of a SIP service will be translated by a
>>  gateway within the trust domain of that SIP service."
>>=20
>> In the context of STOX, I have a few questions about this:
>> - Is this how the current deployments in general work? Is that =
enough?
>=20
> I deployed this service and and this is how it works.  It is enough in =
my experience and doe snot require any new addressing scheme or location =
service.=20
>=20
>> - Should we keep this assumption or also explicitly consider the case =
where the gateway would be in the destination domain?
>=20
> Not sure how one could know what gateways the foreign domain has and =
for what purpose. One can only lookup DNS records for specific services =
like SIP or XMPP. You do not know if they are gateways or something =
else. The local domain policy can detect a remote domain capability and =
route based its own policy.
>=20
>> - If we wanted to take destination-domain gateways into =
consideration, what would the implications be to the mapping specs?=20
>=20
> One can only decide what to do for its own domain and advertise it =
into the DNS. One cannot guess or know about topology of a foreign =
domain, can't he?
>=20
> Adrian
>=20
>> Regards,
>> 	Markus=20
>>=20
>> [1] http://datatracker.ietf.org/doc/draft-saintandre-sip-xmpp-core/
>>=20
>>=20
>>=20
>>=20
>> _______________________________________________
>> stox mailing list
>> stox@ietf.org
>> https://www.ietf.org/mailman/listinfo/stox
>>=20
>=20


From Markus.Isomaki@nokia.com  Tue Jun  4 09:53:08 2013
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From: <Markus.Isomaki@nokia.com>
To: <ag@ag-projects.com>
Thread-Topic: [Stox] sip-xmpp-core: gateway in source or destination domain?
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Date: Tue, 4 Jun 2013 14:41:05 +0000
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Subject: Re: [Stox] sip-xmpp-core: gateway in source or destination domain?
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Hi Adrian,

I agree with everything you say below, and that has also been my assumption=
 of how the deployments will work.

I'm probably just reading the sip-xmpp-core too literally. If so, it would =
be useful in the next update of that document to make it a bit clearer. =20

Markus=20

-----Original Message-----
From: ext Adrian Georgescu [mailto:ag@ag-projects.com]=20
Sent: 04 June, 2013 16:36
To: Isomaki Markus (Nokia-CIC/Espoo)
Cc: stox@ietf.org
Subject: Re: [Stox] sip-xmpp-core: gateway in source or destination domain?


On Jun 4, 2013, at 3:25 PM, <Markus.Isomaki@nokia.com> wrote:

> Hi Adrian,
>=20
> Thanks. One clarification about your deployment:
>=20
> Adrian Georgescu [mailto:ag@ag-projects.com] wrote:
>=20
>>> - Is this how the current deployments in general work? Is that enough?
>=20
>> I deployed this service and and this is how it works.  It is enough in m=
y experience and doe snot require any new addressing scheme or location ser=
vice.
>=20
> So, when one domain runs SIP and the other domain runs XMPP, in order to =
talk to each other, each domain is responsible for deploying a gateway that=
 translates outgoing messages from their "native" protocol to the other pro=
tocol? It is not enough to just deploy a gateway in one of the domains (and=
 have that domain advertise both SIP and XMPP in DNS)?
>=20
> Current "sip-xmpp-core" draft explicitly assumes that SIP messages are tr=
anslated to XMPP in the SIP domain, while XMPP messages are translated to S=
IP in the XMPP domain.=20

This is most likely not possible in real life because it would require that=
 both domains have such gateway in place and in practice the chance of this=
 happening is close to zero (unless adoption of such solution goes through =
the roof, unlikely). Most likely the SIP/XMPP gateway is deployed only by t=
hose who care about such interoperability, the remote domains may not care =
and they should not know or be required to do anything about this, it shoul=
d just work.

My use case is the following.=20

I serve SIP users and I want to provide my users with the possibility to in=
teract with remote XMPP domains without having to know or configure anythin=
g in their SIP clients. For this purpose, I have configured my SIP server t=
o route outgoing requests for that XMPP remote domain through the gateway I=
 control. For receiving incoming requests from any remote XMPP domains, I c=
onfigured the proper XMPP related  DNS records to point to the same gateway=
 (its XMPP server listening interface). This was it works both ways and the=
 remote XMPP domain does not need to have a gateway not does he  need to kn=
ow anything about my topology or the fact that I have a gateway, it just ta=
lks native XMPP as it would federate with any other XMPP foreign domain.


>=20
> What I was wondering is how that assumption in practice affects the draft=
-saintandre-sip-xmpp-* specs in general. To my understanding the mappings a=
re the same no matter in which trust domain the gateway is located. The sip=
-xmpp-core does talk about the DNS operations which are certainly affected =
depending on where the translation happens.=20

Again, requiring both domains to have such gateway is both unnecessary and =
unrealistic. Nobody can be forced to deploy a gateway just because a foreig=
n domain would lkike or require so. If the remote domain cares about talkin=
g my protocol, they should deploy a gateway on their own and it should just=
 talk to my service natively.

Adrian


> Markus
>=20
>=20
> -----Original Message-----
> From: ext Adrian Georgescu [mailto:ag@ag-projects.com]
> Sent: 04 June, 2013 15:30
> To: Isomaki Markus (Nokia-CIC/Espoo)
> Cc: stox@ietf.org
> Subject: Re: [Stox] sip-xmpp-core: gateway in source or destination domai=
n?
>=20
>=20
> On Jun 4, 2013, at 12:46 PM, <Markus.Isomaki@nokia.com> <Markus.Isomaki@n=
okia.com> wrote:
>=20
>> Hi,
>>=20
>> The sip-xmpp-core draft [1] says:
>>=20
>> "We further assume that protocol translation  will occur within a=20
>> gateway in the source domain, so that information  generated by the=20
>> user of an XMPP service will be translated by a  gateway within the=20
>> trust domain of that XMPP service, and information  generated by the=20
>> user of a SIP service will be translated by a  gateway within the=20
>> trust domain of that SIP service."
>>=20
>> In the context of STOX, I have a few questions about this:
>> - Is this how the current deployments in general work? Is that enough?
>=20
> I deployed this service and and this is how it works.  It is enough in my=
 experience and doe snot require any new addressing scheme or location serv=
ice.=20
>=20
>> - Should we keep this assumption or also explicitly consider the case wh=
ere the gateway would be in the destination domain?
>=20
> Not sure how one could know what gateways the foreign domain has and for =
what purpose. One can only lookup DNS records for specific services like SI=
P or XMPP. You do not know if they are gateways or something else. The loca=
l domain policy can detect a remote domain capability and route based its o=
wn policy.
>=20
>> - If we wanted to take destination-domain gateways into consideration, w=
hat would the implications be to the mapping specs?=20
>=20
> One can only decide what to do for its own domain and advertise it into t=
he DNS. One cannot guess or know about topology of a foreign domain, can't =
he?
>=20
> Adrian
>=20
>> Regards,
>> 	Markus
>>=20
>> [1] http://datatracker.ietf.org/doc/draft-saintandre-sip-xmpp-core/
>>=20
>>=20
>>=20
>>=20
>> _______________________________________________
>> stox mailing list
>> stox@ietf.org
>> https://www.ietf.org/mailman/listinfo/stox
>>=20
>=20


From stpeter@stpeter.im  Tue Jun  4 10:11:23 2013
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Subject: Re: [Stox] sip-xmpp-core: gateway in source or destination domain?
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On 6/4/13 6:29 AM, Adrian Georgescu wrote:
> 
> On Jun 4, 2013, at 12:46 PM, <Markus.Isomaki@nokia.com>
> <Markus.Isomaki@nokia.com> wrote:
> 
>> Hi,
>> 
>> The sip-xmpp-core draft [1] says:
>> 
>> "We further assume that protocol translation will occur within a
>> gateway in the source domain, so that information generated by the
>> user of an XMPP service will be translated by a gateway within the
>> trust domain of that XMPP service, and information generated by the
>> user of a SIP service will be translated by a gateway within the
>> trust domain of that SIP service."
>> 
>> In the context of STOX, I have a few questions about this: - Is
>> this how the current deployments in general work? Is that enough?
> 
> I deployed this service and and this is how it works.  It is enough
> in my experience and doe snot require any new addressing scheme or
> location service.
> 
>> - Should we keep this assumption or also explicitly consider the
>> case where the gateway would be in the destination domain?
> 
> Not sure how one could know what gateways the foreign domain has and
> for what purpose. One can only lookup DNS records for specific
> services like SIP or XMPP. You do not know if they are gateways or
> something else. The local domain policy can detect a remote domain
> capability and route based its own policy.

Agreed. Some services / products have implemented destination-side
gateways, also called connection managers. A client or peer server would
connect via its preferred protocol (say, SIP) because the service
advertised support for it, but the service would translate the SIP
traffic into XMPP at the connection manager and natively operate using
XMPP within the other parts / core of the service. A similar approach
could be taken for a SIP-based system (offer an XMPP connection manager)
or for a system that uses some other / proprietary protocol internally
(in which case the service might have connection managers for both SIP
and XMPP). But those details are service-specific and I agree with
Adrian that we really can't say much about those internals, if anything.

>> - If we wanted to take destination-domain gateways into
>> consideration, what would the implications be to the mapping specs?
>> 
> 
> One can only decide what to do for its own domain and advertise it
> into the DNS. One cannot guess or know about topology of a foreign
> domain, can't he?

No. But I don't think we need to limit the mappings to source-side
translation -- that was just a simplifying assumption we made for the
sake of the examples, not an architectural recommendation. So that's
something we need to make clearer in the core spec.

Peter

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



From stpeter@stpeter.im  Tue Jun  4 10:13:24 2013
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Cc: stox@ietf.org, "<Markus.Isomaki@nokia.com>" <Markus.Isomaki@nokia.com>
Subject: Re: [Stox] sip-xmpp-core: gateway in source or destination domain?
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On 6/4/13 9:02 AM, Peter Saint-Andre wrote:
> On 6/4/13 6:29 AM, Adrian Georgescu wrote:
>>
>> On Jun 4, 2013, at 12:46 PM, <Markus.Isomaki@nokia.com>
>> <Markus.Isomaki@nokia.com> wrote:
>>
>>> Hi,
>>>
>>> The sip-xmpp-core draft [1] says:
>>>
>>> "We further assume that protocol translation will occur within a
>>> gateway in the source domain, so that information generated by the
>>> user of an XMPP service will be translated by a gateway within the
>>> trust domain of that XMPP service, and information generated by the
>>> user of a SIP service will be translated by a gateway within the
>>> trust domain of that SIP service."
>>>
>>> In the context of STOX, I have a few questions about this: - Is
>>> this how the current deployments in general work? Is that enough?
>>
>> I deployed this service and and this is how it works.  It is enough
>> in my experience and doe snot require any new addressing scheme or
>> location service.
>>
>>> - Should we keep this assumption or also explicitly consider the
>>> case where the gateway would be in the destination domain?
>>
>> Not sure how one could know what gateways the foreign domain has and
>> for what purpose. One can only lookup DNS records for specific
>> services like SIP or XMPP. You do not know if they are gateways or
>> something else. The local domain policy can detect a remote domain
>> capability and route based its own policy.
> 
> Agreed. Some services / products have implemented destination-side
> gateways, also called connection managers. A client or peer server would
> connect via its preferred protocol (say, SIP) because the service
> advertised support for it, but the service would translate the SIP
> traffic into XMPP at the connection manager and natively operate using
> XMPP within the other parts / core of the service. A similar approach
> could be taken for a SIP-based system (offer an XMPP connection manager)
> or for a system that uses some other / proprietary protocol internally
> (in which case the service might have connection managers for both SIP
> and XMPP). But those details are service-specific and I agree with
> Adrian that we really can't say much about those internals, if anything.
> 
>>> - If we wanted to take destination-domain gateways into
>>> consideration, what would the implications be to the mapping specs?
>>>
>>
>> One can only decide what to do for its own domain and advertise it
>> into the DNS. One cannot guess or know about topology of a foreign
>> domain, can't he?
> 
> No. But I don't think we need to limit the mappings to source-side
> translation -- that was just a simplifying assumption we made for the
> sake of the examples, not an architectural recommendation. So that's
> something we need to make clearer in the core spec.

Here is proposed text.

OLD
   This document assumes that a gateway will translate directly from one
   protocol to the other.  We further assume that protocol translation
   will occur within a gateway in the source domain, so that information
   generated by the user of an XMPP service will be translated by a
   gateway within the trust domain of that XMPP service, and information
   generated by the user of a SIP service will be translated by a
   gateway within the trust domain of that SIP service.

NEW
   This document assumes that a gateway will translate directly from one
   protocol to the other.  For the sake of the examples, we further
   assume that protocol translation will occur within a gateway in the
   source domain, so that information generated by the user of an XMPP
   service will be translated by a gateway within the trust domain of
   that XMPP service, and information generated by the user of a SIP
   service will be translated by a gateway within the trust domain of
   that SIP service.  However, nothing in this document ought to be
   taken as recommending against protocol translation at the destination
   domain.

Peter

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



From stpeter@stpeter.im  Tue Jun  4 10:44:38 2013
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On 6/4/13 7:25 AM, Markus.Isomaki@nokia.com wrote:

> Current "sip-xmpp-core" draft explicitly assumes that SIP messages
> are translated to XMPP in the SIP domain, while XMPP messages are
> translated to SIP in the XMPP domain.

Poorly phrased text. Please see my proposed fix.

Peter

-- 
Peter Saint-Andre
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From stpeter@stpeter.im  Tue Jun  4 11:16:41 2013
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Subject: Re: [Stox] WG Review: SIP-TO-XMPP (stox)
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On 6/4/13 2:55 AM, Romascanu, Dan (Dan) wrote:
> Hi Peter, 
> 
> Thank you for your answers. 
> 
> Please see in-line. 
> 
> Regards,
> 
> Dan
> 
> 
> 
> 
>> -----Original Message-----
>> From: Peter Saint-Andre [mailto:stpeter@stpeter.im]
>> Sent: Tuesday, June 04, 2013 1:33 AM
>> To: Romascanu, Dan (Dan)
>> Cc: The IESG; stox@ietf.org
>> Subject: Re: [Stox] WG Review: SIP-TO-XMPP (stox)
>>
>> On 6/3/13 4:58 AM, Romascanu, Dan (Dan) wrote:
>>> Hi,
>>>
>>> I have two comments concerning this charter.
>>>
>>> 1. The way the Objectives section is written does not make clear
>>> whether the current scope of stox is dully defined as making out of
>>> the draft-saintandre-sip-xmpp-* Proposed Standard RFCs, or whether
>>> other proposals for the same mapping functionality (if submitted) will
>>> be considered.
>>
>> Your phrase "dully defined" is spot on: this topic is so dull that I
>> very much doubt that other proposals will be forthcoming! However, you
>> are right that the charter should probably say that those I-D are likely
>> (but not certain) starting points.
>>
>> How about this?
>>
>> OLD
>> These documents are fairly stable and the authors have received feedback
>> from a number of implementers over the years.  However, implementers do
>> not always know about these documents because they are Internet-Drafts
>> and sometimes they have become expired due to inactivity.  Thus it would
>> be helpful to polish them off and publish them as RFCs.
>>
>> NEW
>> Those documents are fairly stable and the authors have received feedback
>> from a number of implementers over the years.  However, implementers do
>> not always know that such mapping specifications exist, because they are
>> Internet-Drafts and sometimes they have become expired due to
>> inactivity.  Thus it would be helpful to publish a set of mapping
>> specifications as RFCs; the foregoing Internet-Drafts provide likely
>> starting points, but other proposals are welcome as per normal IETF
>> working group processes.
>>
> [[DR]] looks good

Great.

>>> 2. It is very unusual that there are no proposed milestones for a
>>> document submitted for public review. I am pretty sure that the IESG
>>> will not approved the charter without seeing the milestones scheduled,
>>> so I suggest that they are made public before the IESG discussion.
>>
>> Personally I think that they can all be submitted for IESG review before
>> IETF 88, and perhaps even well before that.
>>
> [[DR]] so October 2013 as submission date for all documents? I suggest to add these to the proposed charter (unless somebody disagrees)

It might not be good to send all of the documents to the IESG at the
same time. What do IESG members and the chairs think about the order?
Would it make sense to hold WGLC, IETF Last Call, and IESG review about
the whole bundle at the same time? I worry slightly about pushing them
all through simultaneously, but I suppose the total page count is not
huge (smaller than RFC 3261 or RFC 6120).

Peter

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



From stpeter@stpeter.im  Wed Jun  5 14:30:26 2013
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Subject: Re: [Stox] sip-xmpp-media: scope, transparency, extensibility?
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On 6/4/13 6:00 AM, Markus.Isomaki@nokia.com wrote:
> Hi,
> 
> A high-level comment about sip-xmpp-media [1]:
> 
> SDP/Jingle Offer/Answer and ICE are moving targets by themselves. For
> instance right now MMUSIC WG is working on new features such as
> “trickle-ICE” (adding new candidates on the fly – I know this is already
> supported in Jingle) and “bundle” (multiplexing several RTP
> sessions/streams into a single transport-level packet flow).
> 
> We need to ensure that the sip-xmpp-media RFC-to-become is generic
> enough to stay valid even when such extensions are developed, and we
> have a story on how and where the SIP-XMPP mappings to various
> extensions can be developed if needed.
> 
> For instance “bundle”, AFAIK, is going to be entirely SDP/Jingle level
> issue, so presumably it will be enough that after the MMUSIC SDP spec is
> done, a mapping spec to Jingle is written in XSF? On the other hand,
> trickle-ICE, AFAIK, will require the use of SIP INFO (gasp! ;-) method
> to carry the new candidates. That seems like something that would
> require an extension (“ a new RFC”) to sip-xmpp-media in the IETF,
> explaining how Jingle’s trickle is mapped to SIP’s trickle.  

Markus, thanks for your comments.

I've been thinking about this recently, too. I am now thinking that we
might want to remove the media signaling translation from the charter.
There's quite a bit of flux in this area these days, and tracking that
won't be easy (e.g., do we perhaps just need a way to send SDP over
XMPP?). I think we can finish all of the other charter items quickly.
I'm not as convinced about "sip-xmpp-media".

Peter

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



From emil@sip-communicator.org  Wed Jun  5 14:45:57 2013
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Subject: Re: [Stox] sip-xmpp-media: scope, transparency, extensibility?
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Hey Markus,

On 04.06.13, 14:00, Markus.Isomaki@nokia.com wrote:
> On the other hand, trickle-ICE, AFAIK, will require the use of SIP INFO
> (gasp! ;-) method to carry the new candidates.

Yes, but no one seems to mind and we actually seem to have rough 
consensus on the INFO. So I think this could be mapped relatively easily 
to transport-info-s

Emil

-- 
https://jitsi.org

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Subject: Re: [Stox] sip-xmpp-media: scope, transparency, extensibility?
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On 05.06.13, 23:29, Peter Saint-Andre wrote:
> On 6/4/13 6:00 AM, Markus.Isomaki@nokia.com wrote:
>> Hi,
>>
>> A high-level comment about sip-xmpp-media [1]:
>>
>> SDP/Jingle Offer/Answer and ICE are moving targets by themselves. For
>> instance right now MMUSIC WG is working on new features such as
>> =93trickle-ICE=94 (adding new candidates on the fly =96 I know this is=
 already
>> supported in Jingle) and =93bundle=94 (multiplexing several RTP
>> sessions/streams into a single transport-level packet flow).
>>
>> We need to ensure that the sip-xmpp-media RFC-to-become is generic
>> enough to stay valid even when such extensions are developed, and we
>> have a story on how and where the SIP-XMPP mappings to various
>> extensions can be developed if needed.
>>
>> For instance =93bundle=94, AFAIK, is going to be entirely SDP/Jingle l=
evel
>> issue, so presumably it will be enough that after the MMUSIC SDP spec =
is
>> done, a mapping spec to Jingle is written in XSF? On the other hand,
>> trickle-ICE, AFAIK, will require the use of SIP INFO (gasp! ;-) method=

>> to carry the new candidates. That seems like something that would
>> require an extension (=93 a new RFC=94) to sip-xmpp-media in the IETF,=

>> explaining how Jingle=92s trickle is mapped to SIP=92s trickle.
>
> Markus, thanks for your comments.
>
> I've been thinking about this recently, too. I am now thinking that we
> might want to remove the media signaling translation from the charter.
> There's quite a bit of flux in this area these days, and tracking that
> won't be easy (e.g., do we perhaps just need a way to send SDP over
> XMPP?). I think we can finish all of the other charter items quickly.
> I'm not as convinced about "sip-xmpp-media".

Couldn't we maybe just leave this for a later milestone rather than=20
completely dropping it? People are already working on different gateways =

to SIP/SDP and WebRTC and I believe that once the RTCWEB and CLUE cases=20
have been addressed, there wouldn't be that much work.

Also, depending on the plan that gets chosen by RTCWEB/MMUSIC things=20
could be really simple. This would be the case with Plan B or No Plan.

Cheers,
Emil

--=20
https://jitsi.org


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Subject: Re: [Stox] sip-xmpp-media: scope, transparency, extensibility?
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On 6/5/13 3:49 PM, Emil Ivov wrote:
> 
> 
> On 05.06.13, 23:29, Peter Saint-Andre wrote:
>> On 6/4/13 6:00 AM, Markus.Isomaki@nokia.com wrote:
>>> Hi,
>>>
>>> A high-level comment about sip-xmpp-media [1]:
>>>
>>> SDP/Jingle Offer/Answer and ICE are moving targets by themselves. For
>>> instance right now MMUSIC WG is working on new features such as
>>> “trickle-ICE” (adding new candidates on the fly – I know this is already
>>> supported in Jingle) and “bundle” (multiplexing several RTP
>>> sessions/streams into a single transport-level packet flow).
>>>
>>> We need to ensure that the sip-xmpp-media RFC-to-become is generic
>>> enough to stay valid even when such extensions are developed, and we
>>> have a story on how and where the SIP-XMPP mappings to various
>>> extensions can be developed if needed.
>>>
>>> For instance “bundle”, AFAIK, is going to be entirely SDP/Jingle level
>>> issue, so presumably it will be enough that after the MMUSIC SDP spec is
>>> done, a mapping spec to Jingle is written in XSF? On the other hand,
>>> trickle-ICE, AFAIK, will require the use of SIP INFO (gasp! ;-) method
>>> to carry the new candidates. That seems like something that would
>>> require an extension (“ a new RFC”) to sip-xmpp-media in the IETF,
>>> explaining how Jingle’s trickle is mapped to SIP’s trickle.
>>
>> Markus, thanks for your comments.
>>
>> I've been thinking about this recently, too. I am now thinking that we
>> might want to remove the media signaling translation from the charter.
>> There's quite a bit of flux in this area these days, and tracking that
>> won't be easy (e.g., do we perhaps just need a way to send SDP over
>> XMPP?). I think we can finish all of the other charter items quickly.
>> I'm not as convinced about "sip-xmpp-media".
> 
> Couldn't we maybe just leave this for a later milestone rather than
> completely dropping it? People are already working on different gateways
> to SIP/SDP and WebRTC and I believe that once the RTCWEB and CLUE cases
> have been addressed, there wouldn't be that much work.
> 
> Also, depending on the plan that gets chosen by RTCWEB/MMUSIC things
> could be really simple. This would be the case with Plan B or No Plan.

That's one approach. The other approach is to drop it for now, blast
through the easy stuff on the charter, then recharter.

Peter

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



From emil@sip-communicator.org  Wed Jun  5 15:06:36 2013
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Date: Thu, 06 Jun 2013 00:06:27 +0200
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Cc: stox@ietf.org
Subject: Re: [Stox] sip-xmpp-media: scope, transparency, extensibility?
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On 05.06.13, 23:59, Peter Saint-Andre wrote:
> On 6/5/13 3:49 PM, Emil Ivov wrote:
>>
>>
>> On 05.06.13, 23:29, Peter Saint-Andre wrote:
>>> On 6/4/13 6:00 AM, Markus.Isomaki@nokia.com wrote:
>>>> Hi,
>>>>
>>>> A high-level comment about sip-xmpp-media [1]:
>>>>
>>>> SDP/Jingle Offer/Answer and ICE are moving targets by themselves. Fo=
r
>>>> instance right now MMUSIC WG is working on new features such as
>>>> =93trickle-ICE=94 (adding new candidates on the fly =96 I know this =
is already
>>>> supported in Jingle) and =93bundle=94 (multiplexing several RTP
>>>> sessions/streams into a single transport-level packet flow).
>>>>
>>>> We need to ensure that the sip-xmpp-media RFC-to-become is generic
>>>> enough to stay valid even when such extensions are developed, and we=

>>>> have a story on how and where the SIP-XMPP mappings to various
>>>> extensions can be developed if needed.
>>>>
>>>> For instance =93bundle=94, AFAIK, is going to be entirely SDP/Jingle=
 level
>>>> issue, so presumably it will be enough that after the MMUSIC SDP spe=
c is
>>>> done, a mapping spec to Jingle is written in XSF? On the other hand,=

>>>> trickle-ICE, AFAIK, will require the use of SIP INFO (gasp! ;-) meth=
od
>>>> to carry the new candidates. That seems like something that would
>>>> require an extension (=93 a new RFC=94) to sip-xmpp-media in the IET=
F,
>>>> explaining how Jingle=92s trickle is mapped to SIP=92s trickle.
>>>
>>> Markus, thanks for your comments.
>>>
>>> I've been thinking about this recently, too. I am now thinking that w=
e
>>> might want to remove the media signaling translation from the charter=
=2E
>>> There's quite a bit of flux in this area these days, and tracking tha=
t
>>> won't be easy (e.g., do we perhaps just need a way to send SDP over
>>> XMPP?). I think we can finish all of the other charter items quickly.=

>>> I'm not as convinced about "sip-xmpp-media".
>>
>> Couldn't we maybe just leave this for a later milestone rather than
>> completely dropping it? People are already working on different gatewa=
ys
>> to SIP/SDP and WebRTC and I believe that once the RTCWEB and CLUE case=
s
>> have been addressed, there wouldn't be that much work.
>>
>> Also, depending on the plan that gets chosen by RTCWEB/MMUSIC things
>> could be really simple. This would be the case with Plan B or No Plan.=

>
> That's one approach. The other approach is to drop it for now, blast
> through the easy stuff on the charter, then recharter.

I'd appreciate a couple of pointers about IETF best practices here.

Is there something wrong with a dormant WG? Is this considered bad=20
practice and generally avoided?

Because if not, wouldn't it be easier to blast through the easy=20
milestones then wait until ready to start work on the SDP translation.=20
This way, once we are ready we won't have to go through the hassle of=20
mustering consensus, finding victims for chairs, rechartering and most=20
importantly: thinking of a new name!

Emil

--=20
https://jitsi.org


From stpeter@stpeter.im  Wed Jun  5 15:15:58 2013
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Subject: Re: [Stox] sip-xmpp-media: scope, transparency, extensibility?
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On 6/5/13 4:06 PM, Emil Ivov wrote:
> 
> 
> On 05.06.13, 23:59, Peter Saint-Andre wrote:
>> On 6/5/13 3:49 PM, Emil Ivov wrote:
>>>
>>>
>>> On 05.06.13, 23:29, Peter Saint-Andre wrote:
>>>> On 6/4/13 6:00 AM, Markus.Isomaki@nokia.com wrote:
>>>>> Hi,
>>>>>
>>>>> A high-level comment about sip-xmpp-media [1]:
>>>>>
>>>>> SDP/Jingle Offer/Answer and ICE are moving targets by themselves. For
>>>>> instance right now MMUSIC WG is working on new features such as
>>>>> “trickle-ICE” (adding new candidates on the fly – I know this is
>>>>> already
>>>>> supported in Jingle) and “bundle” (multiplexing several RTP
>>>>> sessions/streams into a single transport-level packet flow).
>>>>>
>>>>> We need to ensure that the sip-xmpp-media RFC-to-become is generic
>>>>> enough to stay valid even when such extensions are developed, and we
>>>>> have a story on how and where the SIP-XMPP mappings to various
>>>>> extensions can be developed if needed.
>>>>>
>>>>> For instance “bundle”, AFAIK, is going to be entirely SDP/Jingle level
>>>>> issue, so presumably it will be enough that after the MMUSIC SDP
>>>>> spec is
>>>>> done, a mapping spec to Jingle is written in XSF? On the other hand,
>>>>> trickle-ICE, AFAIK, will require the use of SIP INFO (gasp! ;-) method
>>>>> to carry the new candidates. That seems like something that would
>>>>> require an extension (“ a new RFC”) to sip-xmpp-media in the IETF,
>>>>> explaining how Jingle’s trickle is mapped to SIP’s trickle.
>>>>
>>>> Markus, thanks for your comments.
>>>>
>>>> I've been thinking about this recently, too. I am now thinking that we
>>>> might want to remove the media signaling translation from the charter.
>>>> There's quite a bit of flux in this area these days, and tracking that
>>>> won't be easy (e.g., do we perhaps just need a way to send SDP over
>>>> XMPP?). I think we can finish all of the other charter items quickly.
>>>> I'm not as convinced about "sip-xmpp-media".
>>>
>>> Couldn't we maybe just leave this for a later milestone rather than
>>> completely dropping it? People are already working on different gateways
>>> to SIP/SDP and WebRTC and I believe that once the RTCWEB and CLUE cases
>>> have been addressed, there wouldn't be that much work.
>>>
>>> Also, depending on the plan that gets chosen by RTCWEB/MMUSIC things
>>> could be really simple. This would be the case with Plan B or No Plan.
>>
>> That's one approach. The other approach is to drop it for now, blast
>> through the easy stuff on the charter, then recharter.
> 
> I'd appreciate a couple of pointers about IETF best practices here.
> 
> Is there something wrong with a dormant WG? Is this considered bad
> practice and generally avoided?
> 
> Because if not, wouldn't it be easier to blast through the easy
> milestones then wait until ready to start work on the SDP translation.
> This way, once we are ready we won't have to go through the hassle of
> mustering consensus, finding victims for chairs, rechartering and most
> importantly: thinking of a new name!

The IESG likes to see progress. If the WG quickly finishes its other
milestones, then rechartering to do the media mapping would be fairly
straightforward.

Peter

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



From emil@sip-communicator.org  Wed Jun  5 15:20:43 2013
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Subject: Re: [Stox] sip-xmpp-media: scope, transparency, extensibility?
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On 06.06.13, 00:15, Peter Saint-Andre wrote:
> On 6/5/13 4:06 PM, Emil Ivov wrote:
>>
>>
>> On 05.06.13, 23:59, Peter Saint-Andre wrote:
>>> On 6/5/13 3:49 PM, Emil Ivov wrote:
>>>>
>>>>
>>>> On 05.06.13, 23:29, Peter Saint-Andre wrote:
>>>>> On 6/4/13 6:00 AM, Markus.Isomaki@nokia.com wrote:
>>>>>> Hi,
>>>>>>
>>>>>> A high-level comment about sip-xmpp-media [1]:
>>>>>>
>>>>>> SDP/Jingle Offer/Answer and ICE are moving targets by themselves. =
For
>>>>>> instance right now MMUSIC WG is working on new features such as
>>>>>> =93trickle-ICE=94 (adding new candidates on the fly =96 I know thi=
s is
>>>>>> already
>>>>>> supported in Jingle) and =93bundle=94 (multiplexing several RTP
>>>>>> sessions/streams into a single transport-level packet flow).
>>>>>>
>>>>>> We need to ensure that the sip-xmpp-media RFC-to-become is generic=

>>>>>> enough to stay valid even when such extensions are developed, and =
we
>>>>>> have a story on how and where the SIP-XMPP mappings to various
>>>>>> extensions can be developed if needed.
>>>>>>
>>>>>> For instance =93bundle=94, AFAIK, is going to be entirely SDP/Jing=
le level
>>>>>> issue, so presumably it will be enough that after the MMUSIC SDP
>>>>>> spec is
>>>>>> done, a mapping spec to Jingle is written in XSF? On the other han=
d,
>>>>>> trickle-ICE, AFAIK, will require the use of SIP INFO (gasp! ;-) me=
thod
>>>>>> to carry the new candidates. That seems like something that would
>>>>>> require an extension (=93 a new RFC=94) to sip-xmpp-media in the I=
ETF,
>>>>>> explaining how Jingle=92s trickle is mapped to SIP=92s trickle.
>>>>>
>>>>> Markus, thanks for your comments.
>>>>>
>>>>> I've been thinking about this recently, too. I am now thinking that=
 we
>>>>> might want to remove the media signaling translation from the chart=
er.
>>>>> There's quite a bit of flux in this area these days, and tracking t=
hat
>>>>> won't be easy (e.g., do we perhaps just need a way to send SDP over=

>>>>> XMPP?). I think we can finish all of the other charter items quickl=
y.
>>>>> I'm not as convinced about "sip-xmpp-media".
>>>>
>>>> Couldn't we maybe just leave this for a later milestone rather than
>>>> completely dropping it? People are already working on different gate=
ways
>>>> to SIP/SDP and WebRTC and I believe that once the RTCWEB and CLUE ca=
ses
>>>> have been addressed, there wouldn't be that much work.
>>>>
>>>> Also, depending on the plan that gets chosen by RTCWEB/MMUSIC things=

>>>> could be really simple. This would be the case with Plan B or No Pla=
n.
>>>
>>> That's one approach. The other approach is to drop it for now, blast
>>> through the easy stuff on the charter, then recharter.
>>
>> I'd appreciate a couple of pointers about IETF best practices here.
>>
>> Is there something wrong with a dormant WG? Is this considered bad
>> practice and generally avoided?
>>
>> Because if not, wouldn't it be easier to blast through the easy
>> milestones then wait until ready to start work on the SDP translation.=

>> This way, once we are ready we won't have to go through the hassle of
>> mustering consensus, finding victims for chairs, rechartering and most=

>> importantly: thinking of a new name!
>
> The IESG likes to see progress. If the WG quickly finishes its other
> milestones, then rechartering to do the media mapping would be fairly
> straightforward.

OK, I see.

Emil


--=20
https://jitsi.org


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To: Peter Saint-Andre <stpeter@stpeter.im>
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Subject: Re: [Stox] sip-xmpp-media: scope, transparency, extensibility?
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Hi,

On Jun 5, 2013, at 11:29 PM, Peter Saint-Andre <stpeter@stpeter.im> =
wrote:

> On 6/4/13 6:00 AM, Markus.Isomaki@nokia.com wrote:
>> Hi,
>>=20
>> A high-level comment about sip-xmpp-media [1]:
>>=20
>> SDP/Jingle Offer/Answer and ICE are moving targets by themselves. For
>> instance right now MMUSIC WG is working on new features such as
>> =93trickle-ICE=94 (adding new candidates on the fly =96 I know this =
is already
>> supported in Jingle) and =93bundle=94 (multiplexing several RTP
>> sessions/streams into a single transport-level packet flow).
>>=20
>> We need to ensure that the sip-xmpp-media RFC-to-become is generic
>> enough to stay valid even when such extensions are developed, and we
>> have a story on how and where the SIP-XMPP mappings to various
>> extensions can be developed if needed.
>>=20
>> For instance =93bundle=94, AFAIK, is going to be entirely SDP/Jingle =
level
>> issue, so presumably it will be enough that after the MMUSIC SDP spec =
is
>> done, a mapping spec to Jingle is written in XSF? On the other hand,
>> trickle-ICE, AFAIK, will require the use of SIP INFO (gasp! ;-) =
method
>> to carry the new candidates. That seems like something that would
>> require an extension (=93 a new RFC=94) to sip-xmpp-media in the =
IETF,
>> explaining how Jingle=92s trickle is mapped to SIP=92s trickle. =20
>=20
> Markus, thanks for your comments.
>=20
> I've been thinking about this recently, too. I am now thinking that we
> might want to remove the media signaling translation from the charter.
> There's quite a bit of flux in this area these days, and tracking that
> won't be easy (e.g., do we perhaps just need a way to send SDP over
> XMPP?). I think we can finish all of the other charter items quickly.
> I'm not as convinced about "sip-xmpp-media".
>=20

Those are all new things mainly pushed by WebRTC which is also a moving =
target at this point IMHO.

XEP-0176 already specifies a way to tell endpoints that they should use =
the "legacy" offer/answer model and not send candidates in a =
session-info [0]. The XEP actually mentions it in the context of a SIP - =
XMPP gateway. Moreover, Emil and I already have working and =
interoperable implementations using this mechanism, FWIW.

Could we perhaps have sip-xmpp-media be the base spec which uses the =
RFC3264 model and later extend it with ICE trickle and friends, or is =
that a no go?


[0]: http://xmpp.org/extensions/xep-0176.html#support-sdp

--
Sa=FAl Ibarra Corretg=E9
AG Projects




From emcho@sip-communicator.org  Thu Jun  6 00:36:03 2013
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From: Emil Ivov <emcho@jitsi.org>
Date: Thu, 6 Jun 2013 09:35:28 +0200
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Cc: stox@ietf.org, Peter Saint-Andre <stpeter@stpeter.im>
Subject: Re: [Stox] sip-xmpp-media: scope, transparency, extensibility?
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+1

On Thu, Jun 6, 2013 at 9:28 AM, Sa=FAl Ibarra Corretg=E9
<saul@ag-projects.com> wrote:
> Hi,
>
> On Jun 5, 2013, at 11:29 PM, Peter Saint-Andre <stpeter@stpeter.im> wrote=
:
>
>> On 6/4/13 6:00 AM, Markus.Isomaki@nokia.com wrote:
>>> Hi,
>>>
>>> A high-level comment about sip-xmpp-media [1]:
>>>
>>> SDP/Jingle Offer/Answer and ICE are moving targets by themselves. For
>>> instance right now MMUSIC WG is working on new features such as
>>> =93trickle-ICE=94 (adding new candidates on the fly =96 I know this is =
already
>>> supported in Jingle) and =93bundle=94 (multiplexing several RTP
>>> sessions/streams into a single transport-level packet flow).
>>>
>>> We need to ensure that the sip-xmpp-media RFC-to-become is generic
>>> enough to stay valid even when such extensions are developed, and we
>>> have a story on how and where the SIP-XMPP mappings to various
>>> extensions can be developed if needed.
>>>
>>> For instance =93bundle=94, AFAIK, is going to be entirely SDP/Jingle le=
vel
>>> issue, so presumably it will be enough that after the MMUSIC SDP spec i=
s
>>> done, a mapping spec to Jingle is written in XSF? On the other hand,
>>> trickle-ICE, AFAIK, will require the use of SIP INFO (gasp! ;-) method
>>> to carry the new candidates. That seems like something that would
>>> require an extension (=93 a new RFC=94) to sip-xmpp-media in the IETF,
>>> explaining how Jingle=92s trickle is mapped to SIP=92s trickle.
>>
>> Markus, thanks for your comments.
>>
>> I've been thinking about this recently, too. I am now thinking that we
>> might want to remove the media signaling translation from the charter.
>> There's quite a bit of flux in this area these days, and tracking that
>> won't be easy (e.g., do we perhaps just need a way to send SDP over
>> XMPP?). I think we can finish all of the other charter items quickly.
>> I'm not as convinced about "sip-xmpp-media".
>>
>
> Those are all new things mainly pushed by WebRTC which is also a moving t=
arget at this point IMHO.
>
> XEP-0176 already specifies a way to tell endpoints that they should use t=
he "legacy" offer/answer model and not send candidates in a session-info [0=
]. The XEP actually mentions it in the context of a SIP - XMPP gateway. Mor=
eover, Emil and I already have working and interoperable implementations us=
ing this mechanism, FWIW.
>
> Could we perhaps have sip-xmpp-media be the base spec which uses the RFC3=
264 model and later extend it with ICE trickle and friends, or is that a no=
 go?
>
>
> [0]: http://xmpp.org/extensions/xep-0176.html#support-sdp
>
> --
> Sa=FAl Ibarra Corretg=E9
> AG Projects
>
>
>
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox



--=20
Emil Ivov, Ph.D.                       67000 Strasbourg,
Project Lead                           France
Jitsi
emcho@jitsi.org                        PHONE: +33.1.77.62.43.30
http://jitsi.org                       FAX:   +33.1.77.62.47.31

From Markus.Isomaki@nokia.com  Thu Jun  6 07:25:59 2013
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From: <Markus.Isomaki@nokia.com>
To: <saul@ag-projects.com>, <stpeter@stpeter.im>
Thread-Topic: [Stox] sip-xmpp-media: scope, transparency, extensibility?
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Subject: Re: [Stox] sip-xmpp-media: scope, transparency, extensibility?
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Hi Saul,

> Could we perhaps have sip-xmpp-media be the base spec which uses the RFC3=
264 model and later extend it with ICE trickle and friends, or is that a no=
 go?

Yes, that could be a good approach, and that was what I was also initially =
thinking. We just need to make sure that the scope of the baseline sip-xmpp=
-media is clear, and have a clean enough extensibility model for it, and go=
od enough guidance on how the gateways can deal with a) feature disparity b=
etween SIP and XMPP clients they serve and b) features unknown to the gatew=
ay itself. I suppose it would be anyway a good idea to write somewhere some=
thing about the transparency or non-transparency of SIP-XMPP gateways wrt. =
new features. For instance a SIP proxy can - in theory - be fairly transpar=
ent to many types of new features (new headers, new header parameters, new =
payloads), but it is much harder to do that in a gateway.=20

Markus =20


-----Original Message-----
From: stox-bounces@ietf.org [mailto:stox-bounces@ietf.org] On Behalf Of ext=
 Sa=FAl Ibarra Corretg=E9
Sent: 06 June, 2013 10:29
To: Peter Saint-Andre
Cc: stox@ietf.org
Subject: Re: [Stox] sip-xmpp-media: scope, transparency, extensibility?

Hi,

On Jun 5, 2013, at 11:29 PM, Peter Saint-Andre <stpeter@stpeter.im> wrote:

> On 6/4/13 6:00 AM, Markus.Isomaki@nokia.com wrote:
>> Hi,
>>=20
>> A high-level comment about sip-xmpp-media [1]:
>>=20
>> SDP/Jingle Offer/Answer and ICE are moving targets by themselves. For=20
>> instance right now MMUSIC WG is working on new features such as=20
>> "trickle-ICE" (adding new candidates on the fly - I know this is=20
>> already supported in Jingle) and "bundle" (multiplexing several RTP=20
>> sessions/streams into a single transport-level packet flow).
>>=20
>> We need to ensure that the sip-xmpp-media RFC-to-become is generic=20
>> enough to stay valid even when such extensions are developed, and we=20
>> have a story on how and where the SIP-XMPP mappings to various=20
>> extensions can be developed if needed.
>>=20
>> For instance "bundle", AFAIK, is going to be entirely SDP/Jingle=20
>> level issue, so presumably it will be enough that after the MMUSIC=20
>> SDP spec is done, a mapping spec to Jingle is written in XSF? On the=20
>> other hand, trickle-ICE, AFAIK, will require the use of SIP INFO=20
>> (gasp! ;-) method to carry the new candidates. That seems like=20
>> something that would require an extension (" a new RFC") to=20
>> sip-xmpp-media in the IETF, explaining how Jingle's trickle is mapped to=
 SIP's trickle.
>=20
> Markus, thanks for your comments.
>=20
> I've been thinking about this recently, too. I am now thinking that we=20
> might want to remove the media signaling translation from the charter.
> There's quite a bit of flux in this area these days, and tracking that=20
> won't be easy (e.g., do we perhaps just need a way to send SDP over=20
> XMPP?). I think we can finish all of the other charter items quickly.
> I'm not as convinced about "sip-xmpp-media".
>=20

Those are all new things mainly pushed by WebRTC which is also a moving tar=
get at this point IMHO.

XEP-0176 already specifies a way to tell endpoints that they should use the=
 "legacy" offer/answer model and not send candidates in a session-info [0].=
 The XEP actually mentions it in the context of a SIP - XMPP gateway. Moreo=
ver, Emil and I already have working and interoperable implementations usin=
g this mechanism, FWIW.

Could we perhaps have sip-xmpp-media be the base spec which uses the RFC326=
4 model and later extend it with ICE trickle and friends, or is that a no g=
o?


[0]: http://xmpp.org/extensions/xep-0176.html#support-sdp

--
Sa=FAl Ibarra Corretg=E9
AG Projects



_______________________________________________
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stox@ietf.org
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From stpeter@stpeter.im  Fri Jun  7 12:32:35 2013
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Subject: Re: [Stox] sip-xmpp-media: scope, transparency, extensibility?
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On 6/6/13 8:25 AM, Markus.Isomaki@nokia.com wrote:
> Hi Saul,
> 
>> Could we perhaps have sip-xmpp-media be the base spec which uses the RFC3264 model and later extend it with ICE trickle and friends, or is that a no go?
> 
> Yes, that could be a good approach, and that was what I was also initially thinking. We just need to make sure that the scope of the baseline sip-xmpp-media is clear, and have a clean enough extensibility model for it, and good enough guidance on how the gateways can deal with a) feature disparity between SIP and XMPP clients they serve and b) features unknown to the gateway itself. I suppose it would be anyway a good idea to write somewhere something about the transparency or non-transparency of SIP-XMPP gateways wrt. new features. For instance a SIP proxy can - in theory - be fairly transparent to many types of new features (new headers, new header parameters, new payloads), but it is much harder to do that in a gateway. 

Maybe I'll leave it up to Saúl and Emil to write that I-D. :-)

Peter

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



From emil@sip-communicator.org  Fri Jun  7 12:37:33 2013
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Cc: stox@ietf.org, saul@ag-projects.com, Markus.Isomaki@nokia.com
Subject: Re: [Stox] sip-xmpp-media: scope, transparency, extensibility?
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On 07.06.13, 21:31, Peter Saint-Andre wrote:
> On 6/6/13 8:25 AM, Markus.Isomaki@nokia.com wrote:
>> Hi Saul,
>>
>>> Could we perhaps have sip-xmpp-media be the base spec which uses the =
RFC3264 model and later extend it with ICE trickle and friends, or is tha=
t a no go?
>>
>> Yes, that could be a good approach, and that was what I was also initi=
ally thinking. We just need to make sure that the scope of the baseline s=
ip-xmpp-media is clear, and have a clean enough extensibility model for i=
t, and good enough guidance on how the gateways can deal with a) feature =
disparity between SIP and XMPP clients they serve and b) features unknown=
 to the gateway itself. I suppose it would be anyway a good idea to write=
 somewhere something about the transparency or non-transparency of SIP-XM=
PP gateways wrt. new features. For instance a SIP proxy can - in theory -=
 be fairly transparent to many types of new features (new headers, new he=
ader parameters, new payloads), but it is much harder to do that in a gat=
eway.
>
> Maybe I'll leave it up to Sa=FAl and Emil to write that I-D. :-)

Yeah, that should teach them! ;)

Personally I don't mind helping up, so sure, count me in! Sa=FAl already =

has implemented a Jingle gateway, so we should probably be fine if we=20
just documented that.

Emil


--=20
https://jitsi.org


From stpeter@stpeter.im  Fri Jun  7 12:40:56 2013
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Subject: Re: [Stox] sip-xmpp-media: scope, transparency, extensibility?
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On 6/7/13 1:37 PM, Emil Ivov wrote:
> 
> 
> On 07.06.13, 21:31, Peter Saint-Andre wrote:
>> On 6/6/13 8:25 AM, Markus.Isomaki@nokia.com wrote:
>>> Hi Saul,
>>>
>>>> Could we perhaps have sip-xmpp-media be the base spec which uses the
>>>> RFC3264 model and later extend it with ICE trickle and friends, or
>>>> is that a no go?
>>>
>>> Yes, that could be a good approach, and that was what I was also
>>> initially thinking. We just need to make sure that the scope of the
>>> baseline sip-xmpp-media is clear, and have a clean enough
>>> extensibility model for it, and good enough guidance on how the
>>> gateways can deal with a) feature disparity between SIP and XMPP
>>> clients they serve and b) features unknown to the gateway itself. I
>>> suppose it would be anyway a good idea to write somewhere something
>>> about the transparency or non-transparency of SIP-XMPP gateways wrt.
>>> new features. For instance a SIP proxy can - in theory - be fairly
>>> transparent to many types of new features (new headers, new header
>>> parameters, new payloads), but it is much harder to do that in a
>>> gateway.
>>
>> Maybe I'll leave it up to Saúl and Emil to write that I-D. :-)
> 
> Yeah, that should teach them! ;)
> 
> Personally I don't mind helping up, so sure, count me in! Saúl already
> has implemented a Jingle gateway, so we should probably be fine if we
> just documented that.

Sure, these mapping specs are mostly just documenting what folks are
already doing in their gateway implementations.

Peter

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



From stpeter@stpeter.im  Tue Jun 11 10:04:48 2013
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Subject: [Stox] revised STOX charter proposal
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Based on feedback received on the stox@ietf.org list and from IESG
members, here is a revised charter proposal for the STOX WG. Your
feedback is welcome.

Peter

###

SIP-TO-XMPP (stox)
------------------------------------------------
Current Status: Proposed WG

Chairs:
  Markus Isomaki <markus.isomaki at nokia.com>
  Yana Stamcheva <yana at jitsi.org>

Assigned Area Director:
  Gonzalo Camarillo <gonzalo.camarillo at ericsson.com>

Mailing list
  Address: stox at ietf.org
  To Subscribe: https://www.ietf.org/mailman/listinfo/stox
  Archive: http://www.ietf.org/mail-archive/web/stox/

Charter:

Problem Statement

The IETF has defined two signalling technologies that can be used
for multimedia session negotiation, instant messaging, presence,
file transfer, capabilities discovery, notifications, and other types
of real-time functionality:

o  The Session Initiation Protocol (SIP), along with various SIP
   extensions developed within the SIP for Instant Messaging and
   Presence Leveraging Extensions (SIMPLE) Working Group.

o  The Extensible Messaging and Presence Protocol (XMPP), along
   with various XMPP extensions developed by the IETF as well as by
   the XMPP Standards Foundation.

SIP has been focused primarily on media session negotiation (e.g.,
audio and video), whereas XMPP has been focused primarily on messaging
and presence.  As a result, the technologies are mostly complementary.
However, there is also some overlap between SIP and XMPP, since there
are SIP extensions for messaging, presence, groupchat, file transfer,
etc., and there are XMPP extensions for multimedia session negotiation.
This overlap has practical implications, since some deployed services
use SIP for both media and messaging/presence, whereas other deployed
services use XMPP for both messaging/presence and media.  When such
services wish to exchange information, they often need to translate
their native protocol (either SIP or XMPP) to the other protocol
(either XMPP or SIP).

Implementers needing to perform such protocol mappings have often worked
out their own heuristics for doing so.  Unfortunately, these heuristics
are not always consistent, which can lead to interoperability problems.

Objectives

To make it easier for implementers to enable interworking between
SIP-based systems and XMPP-based systems, several Internet-Drafts have
defined guidelines for protocol mapping between SIP and XMPP, starting
with draft-saintandre-xmpp-simple-00 in early 2004.  The current
documents are:

draft-saintandre-sip-xmpp-core
draft-saintandre-sip-xmpp-presence
draft-saintandre-sip-xmpp-im
draft-saintandre-sip-xmpp-chat
draft-saintandre-sip-xmpp-groupchat
draft-saintandre-sip-xmpp-media

Those documents are fairly stable and the authors have received feedback
from a number of implementers over the years.  However, implementers do
not always know that such mapping specifications exist, because they are
Internet-Drafts and sometimes they have become expired due to
inactivity.  Thus it would be helpful to publish a set of mapping
specifications as RFCs; the foregoing Internet-Drafts provide likely
starting points, but other proposals are welcome as per normal IETF
working group processes.

It might also be helpful to at some point publish additional documents
in the same series, covering topics like capabilities discovery and
file transfer.  However, any such work would require a recharter.

The group shall not be tasked with defining any new protocols, only with
specifying mappings between existing protocols that have been defined
for SIP and XMPP.

Deliverables

1. Address mapping and error handling
2. Presence mapping
3. Mapping for single instant messages
4. Mapping for one-to-one text chat sessions
5. Mapping for multi-user text chat sessions
6. Mapping for media signaling

All of the foregoing deliverables are standards track, since they are
subject to interoperability testing.

Milestones:

Jun 2013   Accept starting-point mapping specifications as WG items

Aug 2013   Start Working Group Last Call on mapping specifications

Oct 2013   Submit mapping specifications to the IESG

###



From Markus.Isomaki@nokia.com  Wed Jun 12 23:57:04 2013
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Thread-Topic: [Stox] sip-xmpp-media: scope, transparency, extensibility?
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Cc: stox@ietf.org, saul@ag-projects.com
Subject: Re: [Stox] sip-xmpp-media: scope, transparency, extensibility?
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Hi,

Peter Saint-Andre [mailto:stpeter@stpeter.im] wrote:=20
>
>On 6/7/13 1:37 PM, Emil Ivov wrote:
>>=20
>>=20
>> Yeah, that should teach them! ;)
>>=20
>> Personally I don't mind helping up, so sure, count me in! Sa=FAl already=
=20
>> has implemented a Jingle gateway, so we should probably be fine if we=20
>> just documented that.
>
>Sure, these mapping specs are mostly just documenting what folks are alrea=
dy doing in their gateway implementations.
>

And I don't think either it needs to be more complicated than that. Just tr=
y to document the scope of the gateway's mapping capabilities as well as po=
ssible, and think about how we can later on extend them to support new feat=
ures. The real-time media is the most interesting part in this sense.

And indeed, it's good to have volunteer co-authors for STOX documents.=20

Markus=20

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Subject: [Stox] Milestones changed for stox WG
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Added milestone "Accept starting-point mapping specifications as WG
items", due June 2013.

Added milestone "Start Working Group Last Call on mapping
specifications", due August 2013.

Added milestone "Submit mapping specifications to the IESG", due
October 2013.

URL: http://datatracker.ietf.org/wg/stox/charter/

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Subject: [Stox] Fwd: I-D Action: draft-saintandre-sip-xmpp-core-05.txt
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FYI. This version incorporates feedback from Markus.

Peter


-------- Original Message --------
Subject: I-D Action: draft-saintandre-sip-xmpp-core-05.txt
Date: Thu, 13 Jun 2013 20:49:50 -0700
From: internet-drafts@ietf.org
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To: i-d-announce@ietf.org


A New Internet-Draft is available from the on-line Internet-Drafts
directories.


	Title           : Interworking between the Session Initiation Protocol
(SIP) and the Extensible Messaging and Presence Protocol (XMPP):
Addresses and Error Conditions
	Author(s)       : Peter Saint-Andre
                          Avshalom Houri
                          Joe Hildebrand
	Filename        : draft-saintandre-sip-xmpp-core-05.txt
	Pages           : 14
	Date            : 2013-06-13

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-saintandre-sip-xmpp-core

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-saintandre-sip-xmpp-core-05

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-saintandre-sip-xmpp-core-05


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From stpeter@stpeter.im  Thu Jun 13 21:28:36 2013
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Subject: [Stox] Fwd: I-D Action: draft-saintandre-sip-xmpp-chat-06.txt
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FYI. Fixes to many of the examples and a few of the mapping tables.

Peter


-------- Original Message --------
Subject: I-D Action: draft-saintandre-sip-xmpp-chat-06.txt
Date: Thu, 13 Jun 2013 21:25:40 -0700
From: internet-drafts@ietf.org
Reply-To: internet-drafts@ietf.org
To: i-d-announce@ietf.org


A New Internet-Draft is available from the on-line Internet-Drafts
directories.


	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-saintandre-sip-xmpp-chat-06.txt
	Pages           : 13
	Date            : 2013-06-13

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-saintandre-sip-xmpp-chat

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

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


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

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From saul@ag-projects.com  Fri Jun 14 00:32:59 2013
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>=20
> Here is proposed text.
>=20
> OLD
>   This document assumes that a gateway will translate directly from =
one
>   protocol to the other.  We further assume that protocol translation
>   will occur within a gateway in the source domain, so that =
information
>   generated by the user of an XMPP service will be translated by a
>   gateway within the trust domain of that XMPP service, and =
information
>   generated by the user of a SIP service will be translated by a
>   gateway within the trust domain of that SIP service.
>=20
> NEW
>   This document assumes that a gateway will translate directly from =
one
>   protocol to the other.  For the sake of the examples, we further
>   assume that protocol translation will occur within a gateway in the
>   source domain, so that information generated by the user of an XMPP
>   service will be translated by a gateway within the trust domain of
>   that XMPP service, and information generated by the user of a SIP
>   service will be translated by a gateway within the trust domain of
>   that SIP service.  However, nothing in this document ought to be
>   taken as recommending against protocol translation at the =
destination
>   domain.
>=20

+1.

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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Subject: Re: [Stox] sip-xmpp-media: scope, transparency, extensibility?
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On Jun 7, 2013, at 9:31 PM, Peter Saint-Andre wrote:

> On 6/6/13 8:25 AM, Markus.Isomaki@nokia.com wrote:
>> Hi Saul,
>>=20
>>> Could we perhaps have sip-xmpp-media be the base spec which uses the =
RFC3264 model and later extend it with ICE trickle and friends, or is =
that a no go?
>>=20
>> Yes, that could be a good approach, and that was what I was also =
initially thinking. We just need to make sure that the scope of the =
baseline sip-xmpp-media is clear, and have a clean enough extensibility =
model for it, and good enough guidance on how the gateways can deal with =
a) feature disparity between SIP and XMPP clients they serve and b) =
features unknown to the gateway itself. I suppose it would be anyway a =
good idea to write somewhere something about the transparency or =
non-transparency of SIP-XMPP gateways wrt. new features. For instance a =
SIP proxy can - in theory - be fairly transparent to many types of new =
features (new headers, new header parameters, new payloads), but it is =
much harder to do that in a gateway.=20
>=20
> Maybe I'll leave it up to Sa=FAl and Emil to write that I-D. :-)
>=20

Sorry, catching up... Sure, I already volunteered to do this and I still =
plan to :-)

Basically, as Emil said down below it would start as a description of =
what we ended up implementing and then try to iron out (some of) the =
problems we found along the way.

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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Cc: stox@ietf.org, Markus.Isomaki@nokia.com
Subject: Re: [Stox] sip-xmpp-media: scope, transparency, extensibility?
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On 6/14/13 1:36 AM, Saúl Ibarra Corretgé wrote:
> 
> On Jun 7, 2013, at 9:31 PM, Peter Saint-Andre wrote:
> 
>> On 6/6/13 8:25 AM, Markus.Isomaki@nokia.com wrote:
>>> Hi Saul,
>>>
>>>> Could we perhaps have sip-xmpp-media be the base spec which uses the RFC3264 model and later extend it with ICE trickle and friends, or is that a no go?
>>>
>>> Yes, that could be a good approach, and that was what I was also initially thinking. We just need to make sure that the scope of the baseline sip-xmpp-media is clear, and have a clean enough extensibility model for it, and good enough guidance on how the gateways can deal with a) feature disparity between SIP and XMPP clients they serve and b) features unknown to the gateway itself. I suppose it would be anyway a good idea to write somewhere something about the transparency or non-transparency of SIP-XMPP gateways wrt. new features. For instance a SIP proxy can - in theory - be fairly transparent to many types of new features (new headers, new header parameters, new payloads), but it is much harder to do that in a gateway. 
>>
>> Maybe I'll leave it up to Saúl and Emil to write that I-D. :-)
>>
> 
> Sorry, catching up... Sure, I already volunteered to do this and I still plan to :-)
> 
> Basically, as Emil said down below it would start as a description of what we ended up implementing and then try to iron out (some of) the problems we found along the way.

Great. If you're able to use parts of draft-saintandre-sip-xmpp-media as
a starting point, feel free!

Peter

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



From iesg-secretary@ietf.org  Fri Jun 14 08:58:22 2013
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Subject: [Stox] WG Action: Formed SIP-TO-XMPP (stox)
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A new IETF working group has been formed in the Real-time Applications
and Infrastructure Area. For additional information please contact the
Area Directors or the WG Chairs.

SIP-TO-XMPP (stox)
------------------------------------------------
Current Status: Proposed WG

Chairs:
  Markus Isomaki <markus.isomaki@nokia.com>
  Yana Stamcheva <yana@jitsi.org>

Assigned Area Director:
  Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>

Mailing list
  Address: stox@ietf.org
  To Subscribe: https://www.ietf.org/mailman/listinfo/stox
  Archive: http://www.ietf.org/mail-archive/web/stox/

Charter:

Problem Statement

The IETF has defined two signalling technologies that can be used
for multimedia session negotiation, instant messaging, presence,
file transfer, capabilities discovery, notifications, and other types
of real-time functionality:

o  The Session Initiation Protocol (SIP), along with various SIP
   extensions developed within the SIP for Instant Messaging and
   Presence Leveraging Extensions (SIMPLE) Working Group.

o  The Extensible Messaging and Presence Protocol (XMPP), along
   with various XMPP extensions developed by the IETF as well as by
   the XMPP Standards Foundation.

SIP has been focused primarily on media session negotiation (e.g.,
audio and video), whereas XMPP has been focused primarily on messaging
and presence.  As a result, the technologies are mostly complementary.
However, there is also some overlap between SIP and XMPP, since there
are SIP extensions for messaging, presence, groupchat, file transfer,
etc., and there are XMPP extensions for multimedia session negotiation.
This overlap has practical implications, since some deployed services
use SIP for both media and messaging/presence, whereas other deployed
services use XMPP for both messaging/presence and media.  When such
services wish to exchange information, they often need to translate
their native protocol (either SIP or XMPP) to the other protocol
(either XMPP or SIP).

Implementers needing to perform such protocol mappings have often worked
out their own heuristics for doing so.  Unfortunately, these heuristics
are not always consistent, which can lead to interoperability problems.

Objectives

To make it easier for implementers to enable interworking between
SIP-based systems and XMPP-based systems, several Internet-Drafts have
defined guidelines for protocol mapping between SIP and XMPP, starting
with draft-saintandre-xmpp-simple-00 in early 2004.  The current
documents are:

draft-saintandre-sip-xmpp-core
draft-saintandre-sip-xmpp-presence
draft-saintandre-sip-xmpp-im
draft-saintandre-sip-xmpp-chat
draft-saintandre-sip-xmpp-groupchat
draft-saintandre-sip-xmpp-media

Those documents are fairly stable and the authors have received feedback
from a number of implementers over the years.  However, implementers do
not always know that such mapping specifications exist, because they are
Internet-Drafts and sometimes they have become expired due to
inactivity.  Thus it would be helpful to publish a set of mapping
specifications as RFCs; the foregoing Internet-Drafts provide likely
starting points, but other proposals are welcome as per normal IETF
working group processes.

It might also be helpful to at some point publish additional documents
in the same series, covering topics like capabilities discovery and
file transfer.  However, any such work would require a recharter.

The group shall not be tasked with defining any new protocols, only with
specifying mappings between existing protocols that have been defined
for SIP and XMPP.

Deliverables

1. Address mapping and error handling
2. Presence mapping
3. Mapping for single instant messages
4. Mapping for one-to-one text chat sessions
5. Mapping for multi-user text chat sessions
6. Mapping for media signaling

All of the foregoing deliverables are standards track, since they are
subject to interoperability testing.

Milestones:

Jun 2013  Accept starting-point mapping specifications as WG items
Aug 2013  Start Working Group Last Call on mapping specifications
Oct 2013  Submit mapping specifications to the IESG

From ietf-secretariat-reply@ietf.org  Fri Jun 14 08:59:29 2013
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Subject: [Stox] Milestones changed for stox WG
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Added milestone "Accept starting-point mapping specifications as WG
items", due June 2013.

Added milestone "Start Working Group Last Call on mapping
specifications", due August 2013.

Added milestone "Submit mapping specifications to the IESG", due
October 2013.

URL: http://datatracker.ietf.org/wg/stox/charter/

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Subject: [Stox] STOX WG officially approved - charter and work plan
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Hi all,

The SIP-to-XMPP (STOX) WG has now been officially approved by the IESG. Our=
 charter can be found at <http://datatracker.ietf.org/wg/stox/charter/>. My=
self (markus.isomaki@nokia.com) and Yana Stamcheva (yana@jitsi.org) will ac=
t as co-chairs.=20

The plan outlined in the charter is that this will be a focused and relativ=
ely straight-forward effort that we aim to finish in just a few months. The=
 charter has these milestones:

Jun 2013  Accept starting-point mapping specifications as WG items
Aug 2013  Start Working Group Last Call on mapping specifications
Oct 2013  Submit mapping specifications to the IESG

Yana and I have additionally outlined a bit more detailed plan:

- June: One week consensus call on adopting the current drafts as WG items.
- Latest by July 8: Submit current drafts as -00 WG drafts.
- July: Ask people to comment and raise open issues about the drafts, ask c=
o-authors to prepare presentations about open issues for IETF 87. Start rec=
ruiting reviewers and if needed, co-authors.=20
- IETF 87: Meet  for 1 hour. Discuss open issues. Agree on the deadline for=
 reviews and WG last call on the documents. Decide if all documents can pro=
ceed in the same timeline, or whether some phasing will be needed.=20
- By end of August: Start the official WG last call.=20
- September - October: Process last call comments and finalize the document=
s for the IESG review.=20
- IETF 88: Meet if there is interest and energy to discuss about recharteri=
ng, i.e. adding new WG items.

Let us know if you have any feedback to this.

I will next send a few mails to the list about specific topics such as our =
first milestone, i.e. the acceptance of the current drafts as WG items.

Markus=20



From Markus.Isomaki@nokia.com  Wed Jun 19 03:30:40 2013
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From: <Markus.Isomaki@nokia.com>
To: <stox@ietf.org>
Thread-Topic: Consensus call: Adopting the "current" drafts as WG documents to match the planned deliverables
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Date: Wed, 19 Jun 2013 10:30:26 +0000
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Subject: [Stox] Consensus call: Adopting the "current" drafts as WG documents to match the planned deliverables
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Hi,

The chartered deliverables of STOX are:

1. Address mapping and error handling
2. Presence mapping
3. Mapping for single instant messages
4. Mapping for one-to-one text chat sessions
5. Mapping for multi-user text chat sessions
6. Mapping for media signaling

The plan is that a separate RFC will be published for each of these.=20

As the SIP-to-XMPP mapping effort has been going on for some years already =
even before the WG for it has been setup, we already have a set of relative=
ly mature (individual) drafts to match these deliverables:

draft-saintandre-sip-xmpp-core (Peter Saint-Andre, Avshalom Houri, Joe Hild=
ebrand)
draft-saintandre-sip-xmpp-presence (Peter Saint-Andre, Avshalom Houri, Joe =
Hildebrand)
draft-saintandre-sip-xmpp-im (Peter Saint-Andre, Avshalom Houri, Joe Hildeb=
rand)
draft-saintandre-sip-xmpp-chat (Peter Saint-Andre, Salvatore Loreto, Eddy G=
avita, Nazin Hossain)
draft-saintandre-sip-xmpp-groupchat (Peter Saint-Andre, Salvatore Loreto, S=
aul Ibarra Corretge, Fabio Forno)
draft-saintandre-sip-xmpp-media (Peter Saint-Andre)

The first milestone of our charter is:

Jun 2013  Accept starting-point mapping specifications as WG items

To meet this on time, we are hereby starting a consensus call on adopting t=
hese drafts as WG items, so that they can be submitted as such by the IETF =
87 -00 cut-off date (July 8). So, please send your comments on this to the =
STOX list by the end of next week (June 30), after which the chairs determi=
ne if there is a consensus for the WG adoption of these documents. Please s=
end also simple "yes" or "+1" statements to the list, since we want to also=
 see how many active participants we currently have. If you have issues abo=
ut why the current drafts should not be adopted, please raise them ASAP.=20

The other question is about the editors/co-authors of the WG documents. By =
default we will continue with the same list of co-authors as in the individ=
ual documents. So, we would like all the current co-authors to send a note =
to the chairs whether they are willing to continue (and let us know about y=
our realistic time commitment too). One change has already been agreed: sip=
-xmpp-media draft will be taken over by Saul Ibarra Corretge and Emil Ivov.=
 In case someone else is willing to volunteers as a co-chair for one or mor=
e of the drafts, let the chairs know.

The 'sip-xmpp-media' draft (or the to-be RFC) will still also need a decisi=
on about its scope. We'll send a separate mail on that.

Regards,
	Markus & Yana






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From: <Markus.Isomaki@nokia.com>
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Thread-Topic: Maturity of the documents
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Hi,

The chairs have started a consensus call on adopting the 'draft-saintandre-=
sip-xmpp-' documents as WG items. Please send your feedback on it by replyi=
ng to the separate mail about the topic.

Somewhat independently of the WG adoption, here is my current understanding=
 of the maturity of these documents (as a co-chair). This is based on my ow=
n reading as well as discussion with some of the co-authors.


These documents seem to be in a good shape and will mainly just need review=
s:



saintandre-sip-xmpp-core

saintandre-sip-xmpp-presence

saintandre-sip-xmpp-im

saintandre-sip-xmpp-chat



This document still has open issues. The co-authors are already working on =
an update, but further work will still be needed:



saintandre-sip-xmpp-groupchat



This document requires most work still. It is in my understanding the most =
complex piece of these mappings:


saintandre-sip-xmpp-media

This is just a very informal categorization and people are of course welcom=
e to raise any issues or send improvement suggestions with any of these doc=
uments still even if we adopt them as WG documents.

Markus


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<p class=3D"MsoNormal">Hi,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The chairs have started a consensus call on adopting=
 the &#8216;draft-saintandre-sip-xmpp-&#8217; documents as WG items. Please=
 send your feedback on it by replying to the separate mail about the topic.=
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Somewhat independently of the WG adoption, here is m=
y current understanding of the maturity of these documents (as a co-chair).=
 This is based on my own reading as well as discussion with some of the co-=
authors.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">These documents seem to be in a good shape and wi=
ll mainly just need reviews:<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">saintandre-sip-xmpp-core<o:p></o:p></p>
<p class=3D"MsoPlainText">saintandre-sip-xmpp-presence<o:p></o:p></p>
<p class=3D"MsoPlainText">saintandre-sip-xmpp-im<o:p></o:p></p>
<p class=3D"MsoPlainText">saintandre-sip-xmpp-chat<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">This document still has open issues. The co-autho=
rs are already working on an update, but further work will still be needed:=
<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">saintandre-sip-xmpp-groupchat<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">This document requires most work still. It is in =
my understanding the most complex piece of these mappings:<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">saintandre-sip-xmpp-media<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">This is just a very informal categorization and peop=
le are of course welcome to raise any issues or send improvement suggestion=
s with any of these documents still even if we adopt them as WG documents.<=
o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Markus<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</body>
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From yana@sip-communicator.org  Wed Jun 19 04:04:42 2013
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Subject: Re: [Stox] STOX WG officially approved - charter and work plan
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The wiki is now also updated and you can find all related information =
regrouped here: http://trac.tools.ietf.org/wg/stox/trac/wiki.

Yana

On Jun 19, 2013, at 11:57 AM, Markus.Isomaki@nokia.com wrote:

> Hi all,
>=20
> The SIP-to-XMPP (STOX) WG has now been officially approved by the =
IESG. Our charter can be found at =
<http://datatracker.ietf.org/wg/stox/charter/>. Myself =
(markus.isomaki@nokia.com) and Yana Stamcheva (yana@jitsi.org) will act =
as co-chairs.=20
>=20
> The plan outlined in the charter is that this will be a focused and =
relatively straight-forward effort that we aim to finish in just a few =
months. The charter has these milestones:
>=20
> Jun 2013  Accept starting-point mapping specifications as WG items
> Aug 2013  Start Working Group Last Call on mapping specifications
> Oct 2013  Submit mapping specifications to the IESG
>=20
> Yana and I have additionally outlined a bit more detailed plan:
>=20
> - June: One week consensus call on adopting the current drafts as WG =
items.
> - Latest by July 8: Submit current drafts as -00 WG drafts.
> - July: Ask people to comment and raise open issues about the drafts, =
ask co-authors to prepare presentations about open issues for IETF 87. =
Start recruiting reviewers and if needed, co-authors.=20
> - IETF 87: Meet  for 1 hour. Discuss open issues. Agree on the =
deadline for reviews and WG last call on the documents. Decide if all =
documents can proceed in the same timeline, or whether some phasing will =
be needed.=20
> - By end of August: Start the official WG last call.=20
> - September - October: Process last call comments and finalize the =
documents for the IESG review.=20
> - IETF 88: Meet if there is interest and energy to discuss about =
rechartering, i.e. adding new WG items.
>=20
> Let us know if you have any feedback to this.
>=20
> I will next send a few mails to the list about specific topics such as =
our first milestone, i.e. the acceptance of the current drafts as WG =
items.
>=20
> Markus=20
>=20
>=20


From Markus.Isomaki@nokia.com  Wed Jun 19 04:56:44 2013
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From: <Markus.Isomaki@nokia.com>
To: <stox@ietf.org>, <emcho@jitsi.org>, <saul@ag-projects.com>, <stpeter@stpeter.im>
Thread-Topic: Mapping for media signaling: still about the scope
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Subject: [Stox] Mapping for media signaling: still about the scope
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Hi Emil, Saul, Peter, and others,

Recently I raised some questions about the scope of the 'sip-xmpp-media' an=
d the related deliverable on the 'mapping for media signaling': http://www.=
ietf.org/mail-archive/web/stox/current/msg00025.html

Now that STOX is officially up and running I'd like us to make an actual pl=
an on what the scope of this document should be, i.e. what features are exp=
licitly going to be part of this mapping spec. Although people have success=
fully implemented SIP-to-XMPP media signaling gateways, I at least see this=
 topic to have some complexity due to:

-          SDP offer/answer is a moving target as new features are being ad=
ded as we speak, including "bundle" and "trickle-ICE". We can't know about =
what features will be added later on.

-          There are features currently supported in XMPP that are not curr=
ently  supported in SIP (and perhaps vice versa?), such as the trickle-ICE =
support in Jingle.

-          Regardless of what is supported in each protocol in general, the=
 actual SIP and XMPP clients will support different sets of features.

Ideally, some things might be mapped transparently in the gateway in an "al=
gorithmic" manner. And ideally, the clients supporting different feature se=
ts can negotiate end-to-end without needing for the gateway to mediate. How=
ever, I'm not sure how easy this will be in practice.

Actually one reason why I see these complications may be that I don't have =
a very good overall understanding about how SDP O/A (as in RFC 3264 syntax =
and semantics) is mapped to XMPP/Jingle (as in its XML based syntax) are in=
 general kept "in synch". For instance, if IETF extends SDP O/A in some way=
, will that always require an explicit corresponding extension XEP by the X=
SF, or is there some more straightforward "algorithmic" way to convey somet=
hing new on the SIP/SDP side to the XMPP/Jingle side.

In the earlier discussion the following proposal by Saul got some support:

> Could we perhaps have sip-xmpp-media be the base spec which uses the RFC3=
264 model and later extend it with ICE trickle and friends

This also sounds fine to me on the high level. As people have already imple=
mented these gateways, writing a document about how they do their work shou=
ld be doable. However, we should elaborate this plan a bit beyond the one l=
ine of text.

Emil, Saul: You have agreed to work as co-authors of this document. Thanks =
;-) Would you be able to write a short proposal about:

-          What end-to-end media setup features should the mapping spec exp=
licitly cover.

-          What will happen to features supported on one or either side of =
the gateway that the gateway is not aware of - can some things be "transpar=
ently" mapped or will the gateway become the "blocking factor" in the middl=
e. (I.e., will the session setup be limited to the features the gateway exp=
licitly needs to "understand".)

-          How will we extend the mapping spec in the future and how should=
 the initial spec take this into account.

This would be very useful so we can set the expectations about what we can/=
should deliver in October, and what is possibly left for the future. Let me=
 know if this sounds reasonable.

Thanks,
                Markus

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<p class=3D"MsoNormal">Hi Emil, Saul, Peter, and others,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Recently I raised some questions about the scope of =
the &#8216;sip-xmpp-media&#8217; and the related deliverable on the &#8216;=
mapping for media signaling&#8217;:
<a href=3D"http://www.ietf.org/mail-archive/web/stox/current/msg00025.html"=
>http://www.ietf.org/mail-archive/web/stox/current/msg00025.html</a><o:p></=
o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Now that STOX is officially up and running I&#8217;d=
 like us to make an actual plan on what the scope of this document should b=
e, i.e. what features are explicitly going to be part of this mapping spec.=
 Although people have successfully implemented
 SIP-to-XMPP media signaling gateways, I at least see this topic to have so=
me complexity due to:<o:p></o:p></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-18.0pt;mso-list:l0 leve=
l1 lfo1"><![if !supportLists]><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><![endif]>SDP offer/answer is a moving target as new features=
 are being added as we speak, including &#8220;bundle&#8221; and &#8220;tri=
ckle-ICE&#8221;. We can&#8217;t know about what features will be added late=
r on.
<o:p></o:p></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-18.0pt;mso-list:l0 leve=
l1 lfo1"><![if !supportLists]><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><![endif]>There are features currently supported in XMPP that=
 are not currently &nbsp;supported in SIP (and perhaps vice versa?), such a=
s the trickle-ICE support in Jingle.<o:p></o:p></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-18.0pt;mso-list:l0 leve=
l1 lfo1"><![if !supportLists]><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><![endif]>Regardless of what is supported in each protocol in=
 general, the actual SIP and XMPP clients will support different sets of fe=
atures.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Ideally, some things might be mapped transparently i=
n the gateway in an &#8220;algorithmic&#8221; manner. And ideally, the clie=
nts supporting different feature sets can negotiate end-to-end without need=
ing for the gateway to mediate. However, I&#8217;m not
 sure how easy this will be in practice.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Actually one reason why I see these complications ma=
y be that I don&#8217;t have a very good overall understanding about how SD=
P O/A (as in RFC 3264 syntax and semantics) is mapped to XMPP/Jingle (as in=
 its XML based syntax) are in general kept
 &#8220;in synch&#8221;. For instance, if IETF extends SDP O/A in some way,=
 will that always require an explicit corresponding extension XEP by the XS=
F, or is there some more straightforward &#8220;algorithmic&#8221; way to c=
onvey something new on the SIP/SDP side to the XMPP/Jingle
 side.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">In the earlier discussion the following proposal by =
Saul got some support:
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">&gt; Could we perhaps have sip-xmpp-media be the bas=
e spec which uses the RFC3264 model and later extend it with ICE trickle an=
d friends<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">This also sounds fine to me on the high level. As pe=
ople have already implemented these gateways, writing a document about how =
they do their work should be doable. However, we should elaborate this plan=
 a bit beyond the one line of text.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Emil, Saul: You have agreed to work as co-authors of=
 this document. Thanks ;-) Would you be able to write a short proposal abou=
t:<o:p></o:p></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-18.0pt;mso-list:l0 leve=
l1 lfo1"><![if !supportLists]><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><![endif]>What end-to-end media setup features should the map=
ping spec explicitly cover.<o:p></o:p></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-18.0pt;mso-list:l0 leve=
l1 lfo1"><![if !supportLists]><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><![endif]>What will happen to features supported on one or ei=
ther side of the gateway that the gateway is not aware of &#8211; can some =
things be &#8220;transparently&#8221; mapped or will the gateway become the=
 &#8220;blocking factor&#8221; in the middle. (I.e., will the
 session setup be limited to the features the gateway explicitly needs to &=
#8220;understand&#8221;.)<o:p></o:p></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-18.0pt;mso-list:l0 leve=
l1 lfo1"><![if !supportLists]><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><![endif]>How will we extend the mapping spec in the future a=
nd how should the initial spec take this into account.
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">This would be very useful so we can set the expectat=
ions about what we can/should deliver in October, and what is possibly left=
 for the future. Let me know if this sounds reasonable.
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Thanks,<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Markus <o:p></o:p></p>
</div>
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Date: Wed, 19 Jun 2013 09:28:02 -0500
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From: Mary Barnes <mary.ietf.barnes@gmail.com>
To: "markus.isomaki@nokia.com" <Markus.Isomaki@nokia.com>
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Subject: Re: [Stox] Maturity of the documents
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I fully support this proposal.  Although, there were concerns raised
during chartering that it seemed that the WG would just fell swoop
adopt this documents - I personally feel that enough time has elapsed
for someone to do that if they had the intention.  But, it might be
good to set a deadline on this proposal for adopting these as WG
documents and make that the same deadline for anyone with alternative
proposals towards the milestones.

Regards,
Mary.

On Wed, Jun 19, 2013 at 6:01 AM,  <Markus.Isomaki@nokia.com> wrote:
> Hi,
>
>
>
> The chairs have started a consensus call on adopting the
> =91draft-saintandre-sip-xmpp-=92 documents as WG items. Please send your
> feedback on it by replying to the separate mail about the topic.
>
>
>
> Somewhat independently of the WG adoption, here is my current understandi=
ng
> of the maturity of these documents (as a co-chair). This is based on my o=
wn
> reading as well as discussion with some of the co-authors.
>
>
>
> These documents seem to be in a good shape and will mainly just need
> reviews:
>
>
>
> saintandre-sip-xmpp-core
>
> saintandre-sip-xmpp-presence
>
> saintandre-sip-xmpp-im
>
> saintandre-sip-xmpp-chat
>
>
>
> This document still has open issues. The co-authors are already working o=
n
> an update, but further work will still be needed:
>
>
>
> saintandre-sip-xmpp-groupchat
>
>
>
> This document requires most work still. It is in my understanding the mos=
t
> complex piece of these mappings:
>
>
>
> saintandre-sip-xmpp-media
>
>
>
> This is just a very informal categorization and people are of course welc=
ome
> to raise any issues or send improvement suggestions with any of these
> documents still even if we adopt them as WG documents.
>
>
>
> Markus
>
>
>
>
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox
>

From mary.ietf.barnes@gmail.com  Wed Jun 19 07:31:23 2013
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Date: Wed, 19 Jun 2013 09:31:07 -0500
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From: Mary Barnes <mary.ietf.barnes@gmail.com>
To: "markus.isomaki@nokia.com" <Markus.Isomaki@nokia.com>
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Subject: Re: [Stox] Maturity of the documents
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Sorry - I should have read all the emails as I see the deadline in this ema=
il:
http://www.ietf.org/mail-archive/web/stox/current/msg00058.html
So, maybe just be explicit that this is the deadline for any
alternative proposals.

Thanks,
Mary.

On Wed, Jun 19, 2013 at 9:28 AM, Mary Barnes <mary.ietf.barnes@gmail.com> w=
rote:
> I fully support this proposal.  Although, there were concerns raised
> during chartering that it seemed that the WG would just fell swoop
> adopt this documents - I personally feel that enough time has elapsed
> for someone to do that if they had the intention.  But, it might be
> good to set a deadline on this proposal for adopting these as WG
> documents and make that the same deadline for anyone with alternative
> proposals towards the milestones.
>
> Regards,
> Mary.
>
> On Wed, Jun 19, 2013 at 6:01 AM,  <Markus.Isomaki@nokia.com> wrote:
>> Hi,
>>
>>
>>
>> The chairs have started a consensus call on adopting the
>> =91draft-saintandre-sip-xmpp-=92 documents as WG items. Please send your
>> feedback on it by replying to the separate mail about the topic.
>>
>>
>>
>> Somewhat independently of the WG adoption, here is my current understand=
ing
>> of the maturity of these documents (as a co-chair). This is based on my =
own
>> reading as well as discussion with some of the co-authors.
>>
>>
>>
>> These documents seem to be in a good shape and will mainly just need
>> reviews:
>>
>>
>>
>> saintandre-sip-xmpp-core
>>
>> saintandre-sip-xmpp-presence
>>
>> saintandre-sip-xmpp-im
>>
>> saintandre-sip-xmpp-chat
>>
>>
>>
>> This document still has open issues. The co-authors are already working =
on
>> an update, but further work will still be needed:
>>
>>
>>
>> saintandre-sip-xmpp-groupchat
>>
>>
>>
>> This document requires most work still. It is in my understanding the mo=
st
>> complex piece of these mappings:
>>
>>
>>
>> saintandre-sip-xmpp-media
>>
>>
>>
>> This is just a very informal categorization and people are of course wel=
come
>> to raise any issues or send improvement suggestions with any of these
>> documents still even if we adopt them as WG documents.
>>
>>
>>
>> Markus
>>
>>
>>
>>
>> _______________________________________________
>> stox mailing list
>> stox@ietf.org
>> https://www.ietf.org/mailman/listinfo/stox
>>

From stpeter@stpeter.im  Wed Jun 19 10:09:15 2013
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Subject: Re: [Stox] STOX WG officially approved - charter and work plan
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Thanks for the detailed plan and information. I'm looking forward to
working with you all on getting this done. :-)

On 6/19/13 5:04 AM, Yana Stamcheva wrote:
> The wiki is now also updated and you can find all related information regrouped here: http://trac.tools.ietf.org/wg/stox/trac/wiki.
> 
> Yana
> 
> On Jun 19, 2013, at 11:57 AM, Markus.Isomaki@nokia.com wrote:
> 
>> Hi all,
>>
>> The SIP-to-XMPP (STOX) WG has now been officially approved by the IESG. Our charter can be found at <http://datatracker.ietf.org/wg/stox/charter/>. Myself (markus.isomaki@nokia.com) and Yana Stamcheva (yana@jitsi.org) will act as co-chairs. 
>>
>> The plan outlined in the charter is that this will be a focused and relatively straight-forward effort that we aim to finish in just a few months. The charter has these milestones:
>>
>> Jun 2013  Accept starting-point mapping specifications as WG items
>> Aug 2013  Start Working Group Last Call on mapping specifications
>> Oct 2013  Submit mapping specifications to the IESG
>>
>> Yana and I have additionally outlined a bit more detailed plan:
>>
>> - June: One week consensus call on adopting the current drafts as WG items.
>> - Latest by July 8: Submit current drafts as -00 WG drafts.
>> - July: Ask people to comment and raise open issues about the drafts, ask co-authors to prepare presentations about open issues for IETF 87. Start recruiting reviewers and if needed, co-authors. 
>> - IETF 87: Meet  for 1 hour. Discuss open issues. Agree on the deadline for reviews and WG last call on the documents. Decide if all documents can proceed in the same timeline, or whether some phasing will be needed. 
>> - By end of August: Start the official WG last call. 
>> - September - October: Process last call comments and finalize the documents for the IESG review. 
>> - IETF 88: Meet if there is interest and energy to discuss about rechartering, i.e. adding new WG items.
>>
>> Let us know if you have any feedback to this.
>>
>> I will next send a few mails to the list about specific topics such as our first milestone, i.e. the acceptance of the current drafts as WG items.
>>
>> Markus 
>>
>>
> 
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox
> 


From saul@ag-projects.com  Wed Jun 19 10:10:30 2013
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Subject: Re: [Stox] STOX WG officially approved - charter and work plan
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Likewise!

On Jun 19, 2013, at 7:09 PM, Peter Saint-Andre <stpeter@stpeter.im> =
wrote:

> Thanks for the detailed plan and information. I'm looking forward to
> working with you all on getting this done. :-)
>=20
> On 6/19/13 5:04 AM, Yana Stamcheva wrote:
>> The wiki is now also updated and you can find all related information =
regrouped here: http://trac.tools.ietf.org/wg/stox/trac/wiki.
>>=20
>> Yana
>>=20
>> On Jun 19, 2013, at 11:57 AM, Markus.Isomaki@nokia.com wrote:
>>=20
>>> Hi all,
>>>=20
>>> The SIP-to-XMPP (STOX) WG has now been officially approved by the =
IESG. Our charter can be found at =
<http://datatracker.ietf.org/wg/stox/charter/>. Myself =
(markus.isomaki@nokia.com) and Yana Stamcheva (yana@jitsi.org) will act =
as co-chairs.=20
>>>=20
>>> The plan outlined in the charter is that this will be a focused and =
relatively straight-forward effort that we aim to finish in just a few =
months. The charter has these milestones:
>>>=20
>>> Jun 2013  Accept starting-point mapping specifications as WG items
>>> Aug 2013  Start Working Group Last Call on mapping specifications
>>> Oct 2013  Submit mapping specifications to the IESG
>>>=20
>>> Yana and I have additionally outlined a bit more detailed plan:
>>>=20
>>> - June: One week consensus call on adopting the current drafts as WG =
items.
>>> - Latest by July 8: Submit current drafts as -00 WG drafts.
>>> - July: Ask people to comment and raise open issues about the =
drafts, ask co-authors to prepare presentations about open issues for =
IETF 87. Start recruiting reviewers and if needed, co-authors.=20
>>> - IETF 87: Meet  for 1 hour. Discuss open issues. Agree on the =
deadline for reviews and WG last call on the documents. Decide if all =
documents can proceed in the same timeline, or whether some phasing will =
be needed.=20
>>> - By end of August: Start the official WG last call.=20
>>> - September - October: Process last call comments and finalize the =
documents for the IESG review.=20
>>> - IETF 88: Meet if there is interest and energy to discuss about =
rechartering, i.e. adding new WG items.
>>>=20
>>> Let us know if you have any feedback to this.
>>>=20
>>> I will next send a few mails to the list about specific topics such =
as our first milestone, i.e. the acceptance of the current drafts as WG =
items.
>>>=20
>>> Markus=20
>>>=20
>>>=20
>>=20
>> _______________________________________________
>> stox mailing list
>> stox@ietf.org
>> https://www.ietf.org/mailman/listinfo/stox
>>=20
>=20
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox

--
Sa=FAl Ibarra Corretg=E9
AG Projects




From stpeter@stpeter.im  Wed Jun 19 10:24:38 2013
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On 6/19/13 5:01 AM, Markus.Isomaki@nokia.com wrote:

> This document still has open issues. The co-authors are already working
> on an update, but further work will still be needed:
> 
> saintandre-sip-xmpp-groupchat

Saúl, Salvatore, and I plan to submit a revised I-D for the groupchat
document by the end of this week.

> This document requires most work still. It is in my understanding the
> most complex piece of these mappings:
> 
> saintandre-sip-xmpp-media

Yes, we need to have more discussion about that one. There is a great
deal of flux happening in this space right now (e.g., WebRTC using raw
SDP -- or not, as being discussed right now in the RTCWEB WG).

Peter

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



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Subject: Re: [Stox] Mapping for media signaling: still about the scope
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Hey Markus

On 19.06.13, 13:56, Markus.Isomaki@nokia.com wrote:
> Hi Emil, Saul, Peter, and others,
>
> Recently I raised some questions about the scope of the =91sip-xmpp-med=
ia=92
> and the related deliverable on the =91mapping for media signaling=92:
> http://www.ietf.org/mail-archive/web/stox/current/msg00025.html
>
> Now that STOX is officially up and running I=92d like us to make an act=
ual
> plan on what the scope of this document should be, i.e. what features
> are explicitly going to be part of this mapping spec. Although people
> have successfully implemented SIP-to-XMPP media signaling gateways, I a=
t
> least see this topic to have some complexity due to:
>
> -SDP offer/answer is a moving target as new features are being added as=

> we speak,

Well ... it is and it isn't. 3264 isn't really changing and all=20
extensions are supposed to keep backward compatibility with it.

> including =93bundle=94 and =93trickle-ICE=94. We can=92t know about wha=
t
> features will be added later on.

No, but we could include some that we know of ... like "bundle" and=20
"trickle-ICE" :).

Trickle ICE would be easier because we already have it in Jingle and the =

SIP version shouldn't take long.

> -There are features currently supported in XMPP that are not currently
>   supported in SIP (and perhaps vice versa?), such as the trickle-ICE
> support in Jingle.

As mentioned. I don't think trickle-ICE would be a problem. I believe=20
separating transport and encoding information would be a bigger hurdle=20
but if we accept that urn:ietf:rfc:3264 means one shouldn't do this.

> -Regardless of what is supported in each protocol in general, the actua=
l
> SIP and XMPP clients will support different sets of features.

Right.

> Ideally, some things might be mapped transparently in the gateway in an=

> =93algorithmic=94 manner.

I don't believe this would be possible.

> And ideally, the clients supporting different
> feature sets can negotiate end-to-end without needing for the gateway t=
o
> mediate. However, I=92m not sure how easy this will be in practice.
>
> Actually one reason why I see these complications may be that I don=92t=

> have a very good overall understanding about how SDP O/A (as in RFC 326=
4
> syntax and semantics) is mapped to XMPP/Jingle (as in its XML based
> syntax) are in general kept =93in synch=94. For instance, if IETF exten=
ds
> SDP O/A in some way, will that always require an explicit corresponding=

> extension XEP by the XSF, or is there some more straightforward
> =93algorithmic=94 way to convey something new on the SIP/SDP side to th=
e
> XMPP/Jingle side.

Well, the good thing about O/A extensions is that they generally=20
preserve backward compatibility. This is valid for things like ICE,=20
trickle ICE, rtcp-mux, bundle. So as long as we support 3264 we should=20
be OK. I think :)

> In the earlier discussion the following proposal by Saul got some suppo=
rt:
>
>  > Could we perhaps have sip-xmpp-media be the base spec which uses the=

> RFC3264 model and later extend it with ICE trickle and friends
>
> This also sounds fine to me on the high level. As people have already
> implemented these gateways, writing a document about how they do their
> work should be doable. However, we should elaborate this plan a bit
> beyond the one line of text.
>
> Emil, Saul: You have agreed to work as co-authors of this document.
> Thanks ;-) Would you be able to write a short proposal about:

Sure!

> -What end-to-end media setup features should the mapping spec explicitl=
y
> cover.

OK, so how about starting with 3261, 3264, 4585, 5285, 5761.

I think we can also add trickle ICE since I don't believe we have much=20
work left for SIP.

Personally I wouldn't even mind having BUNDLE, we'd just need to have=20
the feature in Jingle.

> -What will happen to features supported on one or either side of the
> gateway that the gateway is not aware of=96 can some things be
> =93transparently=94 mapped or will the gateway become the =93blocking f=
actor=94
> in the middle. (I.e., will the session setup be limited to the features=

> the gateway explicitly needs to =93understand=94.)

I don't think the gateway could just forward anything because this would =

imply a straightforward mapping from any SDP attribute to a Jingle elemen=
t.

Maybe gateways could behave as regular SIP B2BUAs: they forward the=20
things that they understand and they remove those that they don't=20
understand or support.

Thoughts?

> -How will we extend the mapping spec in the future and how should the
> initial spec take this into account.

As in administratively?

Emil

> This would be very useful so we can set the expectations about what we
> can/should deliver in October, and what is possibly left for the future=
=2E
> Let me know if this sounds reasonable.
>
> Thanks,
>
>                  Markus
>

--=20
https://jitsi.org


From fippo@goodadvice.pages.de  Wed Jun 19 11:16:14 2013
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Subject: Re: [Stox] Mapping for media signaling: still about the scope
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Am 19.06.2013 20:00, schrieb Emil Ivov: [...]
> OK, so how about starting with 3261, 3264, 4585, 5285, 5761.

That is the core i'm using in my sdp<->jingle mapping for webrtc 
purposes. Add 4568 (sdes, a=crypto), 5763 (a=fingerprint for dtls-srtp) 
and possibly 5576 (a=ssrc) to that list.

> I think we can also add trickle ICE since I don't believe we have much
> work left for SIP.
>
> Personally I wouldn't even mind having BUNDLE, we'd just need to have
> the feature in Jingle.

You can exploit the same-icepwd/ufrag property from 
http://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-04#section-8.1 
-- I haven't checked for backward compability issues though. That way it 
seems we'd only need to have a disco feature for it along with a 
paragraph in XEP-0176.

cheers

philipp

From stpeter@stpeter.im  Wed Jun 19 15:35:44 2013
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

While working on the sip-xmpp-groupchat spec today, Saúl and I found
what I think is an open issue wiht the sip-xmpp-core spec: it doesn't
say how you determine whether a remote domain supports SIP or XMPP (or
both). This is needed for interdomain routing of messages. Thus I
think the sip-xmpp-core spec needs to contain something like the text
in Section 11.2 of RFC 3921:

###

11.2. Outbound Stanzas


   If the hostname of the domain identifier portion of the address
   contained in the 'to' attribute of an outbound stanza matches a
   hostname of the server itself, the server MUST deliver the stanza to
   a local entity according the rules for Inbound Stanzas (Section
   11.1).

   If the hostname of the domain identifier portion of the address
   contained in the 'to' attribute of an outbound stanza does not match
   a hostname of the server itself, the server MUST attempt to route the
   stanza to the foreign domain.  The recommended order of actions is as
   follows:

   1.  First attempt to resolve the foreign hostname using an [SRV]
       Service of "xmpp-server" and Proto of "tcp", resulting in
       resource records such as "_xmpp-server._tcp.example.com.", as
       specified in [XMPP-CORE].

   2.  If the "xmpp-server" address record resolution fails, attempt to
       resolve the "_im" or "_pres" [SRV] Service as specified in
       [IMP-SRV], using the "_im" Service for <message/> stanzas and the
       "_pres" Service for <presence/> stanzas (it is up to the
       implementation how to handle <iq/> stanzas).  This will result in
       one or more resolutions of the form "_im.<proto>.example.com." or
       "_pres.<proto>.example.com.", where "<proto>" would be a label
       registered in the Instant Messaging SRV Protocol Label registry
       or the Presence SRV Protocol Label registry: either "_xmpp" for
       an XMPP-aware domain or some other IANA-registered label (e.g.,
       "_simple") for a non-XMPP-aware domain.

   3.  If both SRV address record resolutions fail, attempt to perform a
       normal IPv4/IPv6 address record resolution to determine the IP
       address using the "xmpp-server" port of 5269 registered with the
       IANA, as specified in [XMPP-CORE].

   Administrators of server deployments are strongly encouraged to keep
   the _im._xmpp, _pres._xmpp, and _xmpp._tcp SRV records properly
   synchronized, since different implementations might perform the "_im"
   and "_pres" lookups before the "xmpp-server" lookup.

###

However, we'd need text describing the discovery and decision process
in both directions (and IMHO we need to drop the _im and _pres SRV
stuff, as we did when working on RFC 6121).

Thoughts?

Peter

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


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From saul@ag-projects.com  Wed Jun 19 23:23:05 2013
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To: Peter Saint-Andre <stpeter@stpeter.im>
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Subject: Re: [Stox] core: connecting to remote domain
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Hi!

On Jun 20, 2013, at 12:35 AM, Peter Saint-Andre <stpeter@stpeter.im> =
wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>=20
> While working on the sip-xmpp-groupchat spec today, Sa=FAl and I found
> what I think is an open issue wiht the sip-xmpp-core spec: it doesn't
> say how you determine whether a remote domain supports SIP or XMPP (or
> both). This is needed for interdomain routing of messages. Thus I
> think the sip-xmpp-core spec needs to contain something like the text
> in Section 11.2 of RFC 3921:
>=20
> ###
>=20
> 11.2. Outbound Stanzas
>=20
>=20
>   If the hostname of the domain identifier portion of the address
>   contained in the 'to' attribute of an outbound stanza matches a
>   hostname of the server itself, the server MUST deliver the stanza to
>   a local entity according the rules for Inbound Stanzas (Section
>   11.1).
>=20
>   If the hostname of the domain identifier portion of the address
>   contained in the 'to' attribute of an outbound stanza does not match
>   a hostname of the server itself, the server MUST attempt to route =
the
>   stanza to the foreign domain.  The recommended order of actions is =
as
>   follows:
>=20
>   1.  First attempt to resolve the foreign hostname using an [SRV]
>       Service of "xmpp-server" and Proto of "tcp", resulting in
>       resource records such as "_xmpp-server._tcp.example.com.", as
>       specified in [XMPP-CORE].
>=20
>   2.  If the "xmpp-server" address record resolution fails, attempt to
>       resolve the "_im" or "_pres" [SRV] Service as specified in
>       [IMP-SRV], using the "_im" Service for <message/> stanzas and =
the
>       "_pres" Service for <presence/> stanzas (it is up to the
>       implementation how to handle <iq/> stanzas).  This will result =
in
>       one or more resolutions of the form "_im.<proto>.example.com." =
or
>       "_pres.<proto>.example.com.", where "<proto>" would be a label
>       registered in the Instant Messaging SRV Protocol Label registry
>       or the Presence SRV Protocol Label registry: either "_xmpp" for
>       an XMPP-aware domain or some other IANA-registered label (e.g.,
>       "_simple") for a non-XMPP-aware domain.
>=20
>   3.  If both SRV address record resolutions fail, attempt to perform =
a
>       normal IPv4/IPv6 address record resolution to determine the IP
>       address using the "xmpp-server" port of 5269 registered with the
>       IANA, as specified in [XMPP-CORE].
>=20
>   Administrators of server deployments are strongly encouraged to keep
>   the _im._xmpp, _pres._xmpp, and _xmpp._tcp SRV records properly
>   synchronized, since different implementations might perform the =
"_im"
>   and "_pres" lookups before the "xmpp-server" lookup.
>=20
> ###
>=20

Well, the problem is that in practice a given domain would have records =
for both SIP and XMPP, so which one would we check first? I'm not sure =
if we can define an unequivocal mechanism to choose the right target, so =
I'm tempted to suggest not to define it for the moment and leave it as a =
local policy item. If we come up with a way to do it later on, maybe we =
can create a new document updating sip-xmpp core?

> However, we'd need text describing the discovery and decision process
> in both directions (and IMHO we need to drop the _im and _pres SRV
> stuff, as we did when working on RFC 6121).
>=20

I have never seen _im and _pres out in the wild, so I'm +1 on removing =
the mention to them.

--
Sa=FAl Ibarra Corretg=E9
AG Projects




From ag@ag-projects.com  Wed Jun 19 23:28:56 2013
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From: Adrian Georgescu <ag@ag-projects.com>
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Cc: "stox@ietf.org" <stox@ietf.org>, Peter Saint-Andre <stpeter@stpeter.im>
Subject: Re: [Stox] core: connecting to remote domain
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XMPP has defined DNS records for both C2S and S2S connections. One =
domain may have a server to server XMPP capability but would not accept =
client connections (missing correspondent  DNS record), this would give =
me a clue that that foreign domain has such XMPP 2 SIP gateway in place =
and if I am a client of that domain I know I cannot use an XMPP client =
for it. For SIP however, is not clear as the same DNS records are used =
both by clients and servers to locate the Proxy/Registrar for a given =
domain. This is the missing piece that makes a heuristic approach not =
100% accurate.=20

Adrian


On Jun 20, 2013, at 8:22 AM, Sa=FAl Ibarra Corretg=E9 =
<saul@ag-projects.com> wrote:

> Hi!
>=20
> On Jun 20, 2013, at 12:35 AM, Peter Saint-Andre <stpeter@stpeter.im> =
wrote:
>=20
>> -----BEGIN PGP SIGNED MESSAGE-----
>> Hash: SHA1
>>=20
>> While working on the sip-xmpp-groupchat spec today, Sa=FAl and I =
found
>> what I think is an open issue wiht the sip-xmpp-core spec: it doesn't
>> say how you determine whether a remote domain supports SIP or XMPP =
(or
>> both). This is needed for interdomain routing of messages. Thus I
>> think the sip-xmpp-core spec needs to contain something like the text
>> in Section 11.2 of RFC 3921:
>>=20
>> ###
>>=20
>> 11.2. Outbound Stanzas
>>=20
>>=20
>>  If the hostname of the domain identifier portion of the address
>>  contained in the 'to' attribute of an outbound stanza matches a
>>  hostname of the server itself, the server MUST deliver the stanza to
>>  a local entity according the rules for Inbound Stanzas (Section
>>  11.1).
>>=20
>>  If the hostname of the domain identifier portion of the address
>>  contained in the 'to' attribute of an outbound stanza does not match
>>  a hostname of the server itself, the server MUST attempt to route =
the
>>  stanza to the foreign domain.  The recommended order of actions is =
as
>>  follows:
>>=20
>>  1.  First attempt to resolve the foreign hostname using an [SRV]
>>      Service of "xmpp-server" and Proto of "tcp", resulting in
>>      resource records such as "_xmpp-server._tcp.example.com.", as
>>      specified in [XMPP-CORE].
>>=20
>>  2.  If the "xmpp-server" address record resolution fails, attempt to
>>      resolve the "_im" or "_pres" [SRV] Service as specified in
>>      [IMP-SRV], using the "_im" Service for <message/> stanzas and =
the
>>      "_pres" Service for <presence/> stanzas (it is up to the
>>      implementation how to handle <iq/> stanzas).  This will result =
in
>>      one or more resolutions of the form "_im.<proto>.example.com." =
or
>>      "_pres.<proto>.example.com.", where "<proto>" would be a label
>>      registered in the Instant Messaging SRV Protocol Label registry
>>      or the Presence SRV Protocol Label registry: either "_xmpp" for
>>      an XMPP-aware domain or some other IANA-registered label (e.g.,
>>      "_simple") for a non-XMPP-aware domain.
>>=20
>>  3.  If both SRV address record resolutions fail, attempt to perform =
a
>>      normal IPv4/IPv6 address record resolution to determine the IP
>>      address using the "xmpp-server" port of 5269 registered with the
>>      IANA, as specified in [XMPP-CORE].
>>=20
>>  Administrators of server deployments are strongly encouraged to keep
>>  the _im._xmpp, _pres._xmpp, and _xmpp._tcp SRV records properly
>>  synchronized, since different implementations might perform the =
"_im"
>>  and "_pres" lookups before the "xmpp-server" lookup.
>>=20
>> ###
>>=20
>=20
> Well, the problem is that in practice a given domain would have =
records for both SIP and XMPP, so which one would we check first? I'm =
not sure if we can define an unequivocal mechanism to choose the right =
target, so I'm tempted to suggest not to define it for the moment and =
leave it as a local policy item. If we come up with a way to do it later =
on, maybe we can create a new document updating sip-xmpp core?
>=20
>> However, we'd need text describing the discovery and decision process
>> in both directions (and IMHO we need to drop the _im and _pres SRV
>> stuff, as we did when working on RFC 6121).
>>=20
>=20
> I have never seen _im and _pres out in the wild, so I'm +1 on removing =
the mention to them.
>=20
> --
> Sa=FAl Ibarra Corretg=E9
> AG Projects
>=20
>=20
>=20
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox
>=20


From saul@ag-projects.com  Wed Jun 19 23:41:02 2013
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Subject: Re: [Stox] core: connecting to remote domain
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On Jun 20, 2013, at 8:28 AM, Adrian Georgescu <ag@ag-projects.com> =
wrote:

> XMPP has defined DNS records for both C2S and S2S connections. One =
domain may have a server to server XMPP capability but would not accept =
client connections (missing correspondent  DNS record), this would give =
me a clue that that foreign domain has such XMPP 2 SIP gateway in place =
and if I am a client of that domain I know I cannot use an XMPP client =
for it. For SIP however, is not clear as the same DNS records are used =
both by clients and servers to locate the Proxy/Registrar for a given =
domain. This is the missing piece that makes a heuristic approach not =
100% accurate.=20
>=20

The absence of C2S records could give you a hint, but some gateways may =
actually implement it thus allowing you to be on both SIP and XMPP at =
the same time from the same domain, so you'll still be confronted with =
the same problem. Not to mention deployments where SIP and XMPP are =
deployed in parallel for the same domain, which I think it's quite =
common these days.

Earlier this year, at XMPP summit, someone mentioned that S-NAPTR could =
be used for this. Looks like it actually would. See =
http://tools.ietf.org/html/rfc3958#section-4.3

I have never seen that deployed though.

--
Sa=FAl Ibarra Corretg=E9
AG Projects




From salvatore.loreto@ericsson.com  Thu Jun 20 02:11:21 2013
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Subject: Re: [Stox] core: connecting to remote domain
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Hi,

On 6/20/13 9:22 AM, Saúl Ibarra Corretgé wrote:
> Hi!
>
> On Jun 20, 2013, at 12:35 AM, Peter Saint-Andre <stpeter@stpeter.im> wrote:
>
>> -----BEGIN PGP SIGNED MESSAGE-----
>> Hash: SHA1
>>
>> While working on the sip-xmpp-groupchat spec today, Saúl and I found
>> what I think is an open issue wiht the sip-xmpp-core spec: it doesn't
>> say how you determine whether a remote domain supports SIP or XMPP (or
>> both). This is needed for interdomain routing of messages. Thus I
>> think the sip-xmpp-core spec needs to contain something like the text
>> in Section 11.2 of RFC 3921:
>>
>> ###
>>
>> 11.2. Outbound Stanzas
>>
>>
>>    If the hostname of the domain identifier portion of the address
>>    contained in the 'to' attribute of an outbound stanza matches a
>>    hostname of the server itself, the server MUST deliver the stanza to
>>    a local entity according the rules for Inbound Stanzas (Section
>>    11.1).
>>
>>    If the hostname of the domain identifier portion of the address
>>    contained in the 'to' attribute of an outbound stanza does not match
>>    a hostname of the server itself, the server MUST attempt to route the
>>    stanza to the foreign domain.  The recommended order of actions is as
>>    follows:
>>
>>    1.  First attempt to resolve the foreign hostname using an [SRV]
>>        Service of "xmpp-server" and Proto of "tcp", resulting in
>>        resource records such as "_xmpp-server._tcp.example.com.", as
>>        specified in [XMPP-CORE].
>>
>>    2.  If the "xmpp-server" address record resolution fails, attempt to
>>        resolve the "_im" or "_pres" [SRV] Service as specified in
>>        [IMP-SRV], using the "_im" Service for <message/> stanzas and the
>>        "_pres" Service for <presence/> stanzas (it is up to the
>>        implementation how to handle <iq/> stanzas).  This will result in
>>        one or more resolutions of the form "_im.<proto>.example.com." or
>>        "_pres.<proto>.example.com.", where "<proto>" would be a label
>>        registered in the Instant Messaging SRV Protocol Label registry
>>        or the Presence SRV Protocol Label registry: either "_xmpp" for
>>        an XMPP-aware domain or some other IANA-registered label (e.g.,
>>        "_simple") for a non-XMPP-aware domain.
>>
>>    3.  If both SRV address record resolutions fail, attempt to perform a
>>        normal IPv4/IPv6 address record resolution to determine the IP
>>        address using the "xmpp-server" port of 5269 registered with the
>>        IANA, as specified in [XMPP-CORE].
>>
>>    Administrators of server deployments are strongly encouraged to keep
>>    the _im._xmpp, _pres._xmpp, and _xmpp._tcp SRV records properly
>>    synchronized, since different implementations might perform the "_im"
>>    and "_pres" lookups before the "xmpp-server" lookup.
>>
>> ###
>>
> Well, the problem is that in practice a given domain would have records for both SIP and XMPP, so which one would we check first? I'm not sure if we can define an unequivocal mechanism to choose the right target, so I'm tempted to suggest not to define it for the moment and leave it as a local policy item.

if a domain has a record for both SIP and XMMP
then if you are an XMPP client, it seems normal to me try first the XMPP one
(and the other way around if I am a SIP UA)

or I am missing something here?

>   If we come up with a way to do it later on, maybe we can create a new document updating sip-xmpp core?
>
>> However, we'd need text describing the discovery and decision process
>> in both directions (and IMHO we need to drop the _im and _pres SRV
>> stuff, as we did when working on RFC 6121).
>>
> I have never seen _im and _pres out in the wild, so I'm +1 on removing the mention to them.
+1 on removing them

/Salvatore

From salvatore.loreto@ericsson.com  Thu Jun 20 02:17:03 2013
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Date: Thu, 20 Jun 2013 12:16:50 +0300
From: Salvatore Loreto <salvatore.loreto@ericsson.com>
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Subject: Re: [Stox] core: connecting to remote domain
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On 6/20/13 9:28 AM, Adrian Georgescu wrote:
> XMPP has defined DNS records for both C2S and S2S connections. One domain may have a server to server XMPP capability but would not accept client connections (missing correspondent  DNS record), this would give me a clue that that foreign domain has such XMPP 2 SIP gateway in place and if I am a client of that domain I know I cannot use an XMPP client for it.
sorry but what would be the reason to have (and expose thru DNS) an 
XMPP2SIP gateway and then not let it to be accessible from a client?
are you assuming that a XMPP2SIP gateway is always only an outbound one?

/Salvatore

> For SIP however, is not clear as the same DNS records are used both by clients and servers to locate the Proxy/Registrar for a given domain. This is the missing piece that makes a heuristic approach not 100% accurate.
>
> Adrian
>
>
> On Jun 20, 2013, at 8:22 AM, Saúl Ibarra Corretgé <saul@ag-projects.com> wrote:
>
>> Hi!
>>
>> On Jun 20, 2013, at 12:35 AM, Peter Saint-Andre <stpeter@stpeter.im> wrote:
>>
>>> -----BEGIN PGP SIGNED MESSAGE-----
>>> Hash: SHA1
>>>
>>> While working on the sip-xmpp-groupchat spec today, Saúl and I found
>>> what I think is an open issue wiht the sip-xmpp-core spec: it doesn't
>>> say how you determine whether a remote domain supports SIP or XMPP (or
>>> both). This is needed for interdomain routing of messages. Thus I
>>> think the sip-xmpp-core spec needs to contain something like the text
>>> in Section 11.2 of RFC 3921:
>>>
>>> ###
>>>
>>> 11.2. Outbound Stanzas
>>>
>>>
>>>   If the hostname of the domain identifier portion of the address
>>>   contained in the 'to' attribute of an outbound stanza matches a
>>>   hostname of the server itself, the server MUST deliver the stanza to
>>>   a local entity according the rules for Inbound Stanzas (Section
>>>   11.1).
>>>
>>>   If the hostname of the domain identifier portion of the address
>>>   contained in the 'to' attribute of an outbound stanza does not match
>>>   a hostname of the server itself, the server MUST attempt to route the
>>>   stanza to the foreign domain.  The recommended order of actions is as
>>>   follows:
>>>
>>>   1.  First attempt to resolve the foreign hostname using an [SRV]
>>>       Service of "xmpp-server" and Proto of "tcp", resulting in
>>>       resource records such as "_xmpp-server._tcp.example.com.", as
>>>       specified in [XMPP-CORE].
>>>
>>>   2.  If the "xmpp-server" address record resolution fails, attempt to
>>>       resolve the "_im" or "_pres" [SRV] Service as specified in
>>>       [IMP-SRV], using the "_im" Service for <message/> stanzas and the
>>>       "_pres" Service for <presence/> stanzas (it is up to the
>>>       implementation how to handle <iq/> stanzas).  This will result in
>>>       one or more resolutions of the form "_im.<proto>.example.com." or
>>>       "_pres.<proto>.example.com.", where "<proto>" would be a label
>>>       registered in the Instant Messaging SRV Protocol Label registry
>>>       or the Presence SRV Protocol Label registry: either "_xmpp" for
>>>       an XMPP-aware domain or some other IANA-registered label (e.g.,
>>>       "_simple") for a non-XMPP-aware domain.
>>>
>>>   3.  If both SRV address record resolutions fail, attempt to perform a
>>>       normal IPv4/IPv6 address record resolution to determine the IP
>>>       address using the "xmpp-server" port of 5269 registered with the
>>>       IANA, as specified in [XMPP-CORE].
>>>
>>>   Administrators of server deployments are strongly encouraged to keep
>>>   the _im._xmpp, _pres._xmpp, and _xmpp._tcp SRV records properly
>>>   synchronized, since different implementations might perform the "_im"
>>>   and "_pres" lookups before the "xmpp-server" lookup.
>>>
>>> ###
>>>
>> Well, the problem is that in practice a given domain would have records for both SIP and XMPP, so which one would we check first? I'm not sure if we can define an unequivocal mechanism to choose the right target, so I'm tempted to suggest not to define it for the moment and leave it as a local policy item. If we come up with a way to do it later on, maybe we can create a new document updating sip-xmpp core?
>>
>>> However, we'd need text describing the discovery and decision process
>>> in both directions (and IMHO we need to drop the _im and _pres SRV
>>> stuff, as we did when working on RFC 6121).
>>>
>> I have never seen _im and _pres out in the wild, so I'm +1 on removing the mention to them.
>>
>> --
>> Saúl Ibarra Corretgé
>> AG Projects
>>
>>
>>
>> _______________________________________________
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>> https://www.ietf.org/mailman/listinfo/stox
>>
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Hi Sal,

Salvatore Loreto wrote:
>=20
> On 6/20/13 9:28 AM, Adrian Georgescu wrote:
> > XMPP has defined DNS records for both C2S and S2S connections. One
> > domain may have a server to server XMPP capability but would not accept
> > client connections (missing correspondent  DNS record), this would give=
 me a
> > clue that that foreign domain has such XMPP 2 SIP gateway in place and =
if I
> > am a client of that domain I know I cannot use an XMPP client for it.
>
> sorry but what would be the reason to have (and expose thru DNS) an
> XMPP2SIP gateway and then not let it to be accessible from a client?
> are you assuming that a XMPP2SIP gateway is always only an outbound one?
>=20

I suppose it would be the case where the domain only runs a SIP service int=
ernally, and just offers an XMPP gateway to external domains. So, if you ha=
ve an XMPP-only client, you would have to get an account for some other dom=
ain to be able to communicate with this SIP-only domain.

Markus


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Hi,=20

Sure. So, if you are not happy with the set of current 'draft-saintandre-si=
p-xmpp-' documents but would like to make your own proposal, please=20
1.) Raise the issues on the STOX list by June 30 and explain briefly what i=
s the different approach you would like to propose.
2.) Submit a draft by the -00 deadline for IETF 87, that is July 8, so we c=
an discuss that at the meeting.

Please make sure your proposal fits the charter, which is quite focused.

Markus

> -----Original Message-----
> From: ext Mary Barnes [mailto:mary.ietf.barnes@gmail.com]
> Sent: 19 June, 2013 17:31
> To: Isomaki Markus (Nokia-CIC/Espoo)
> Cc: stox@ietf.org
> Subject: Re: [Stox] Maturity of the documents
>=20
> Sorry - I should have read all the emails as I see the deadline in this e=
mail:
> http://www.ietf.org/mail-archive/web/stox/current/msg00058.html
> So, maybe just be explicit that this is the deadline for any alternative
> proposals.
>=20
> Thanks,
> Mary.
>=20
> On Wed, Jun 19, 2013 at 9:28 AM, Mary Barnes
> <mary.ietf.barnes@gmail.com> wrote:
> > I fully support this proposal.  Although, there were concerns raised
> > during chartering that it seemed that the WG would just fell swoop
> > adopt this documents - I personally feel that enough time has elapsed
> > for someone to do that if they had the intention.  But, it might be
> > good to set a deadline on this proposal for adopting these as WG
> > documents and make that the same deadline for anyone with alternative
> > proposals towards the milestones.
> >
> > Regards,
> > Mary.
> >
> > On Wed, Jun 19, 2013 at 6:01 AM,  <Markus.Isomaki@nokia.com> wrote:
> >> Hi,
> >>
> >>
> >>
> >> The chairs have started a consensus call on adopting the
> >> 'draft-saintandre-sip-xmpp-' documents as WG items. Please send your
> >> feedback on it by replying to the separate mail about the topic.
> >>
> >>
> >>
> >> Somewhat independently of the WG adoption, here is my current
> >> understanding of the maturity of these documents (as a co-chair).
> >> This is based on my own reading as well as discussion with some of the=
 co-
> authors.
> >>
> >>
> >>
> >> These documents seem to be in a good shape and will mainly just need
> >> reviews:
> >>
> >>
> >>
> >> saintandre-sip-xmpp-core
> >>
> >> saintandre-sip-xmpp-presence
> >>
> >> saintandre-sip-xmpp-im
> >>
> >> saintandre-sip-xmpp-chat
> >>
> >>
> >>
> >> This document still has open issues. The co-authors are already
> >> working on an update, but further work will still be needed:
> >>
> >>
> >>
> >> saintandre-sip-xmpp-groupchat
> >>
> >>
> >>
> >> This document requires most work still. It is in my understanding the
> >> most complex piece of these mappings:
> >>
> >>
> >>
> >> saintandre-sip-xmpp-media
> >>
> >>
> >>
> >> This is just a very informal categorization and people are of course
> >> welcome to raise any issues or send improvement suggestions with any
> >> of these documents still even if we adopt them as WG documents.
> >>
> >>
> >>
> >> Markus
> >>
> >>
> >>
> >>
> >> _______________________________________________
> >> stox mailing list
> >> stox@ietf.org
> >> https://www.ietf.org/mailman/listinfo/stox
> >>

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From: Adrian Georgescu <ag@ag-projects.com>
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To: <Markus.Isomaki@nokia.com>
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Cc: salvatore.loreto@ericsson.com, stox@ietf.org
Subject: Re: [Stox] core: connecting to remote domain
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On Jun 20, 2013, at 12:37 PM, <Markus.Isomaki@nokia.com> wrote:

> Hi Sal,
>=20
> Salvatore Loreto wrote:
>>=20
>> On 6/20/13 9:28 AM, Adrian Georgescu wrote:
>>> XMPP has defined DNS records for both C2S and S2S connections. One
>>> domain may have a server to server XMPP capability but would not =
accept
>>> client connections (missing correspondent  DNS record), this would =
give me a
>>> clue that that foreign domain has such XMPP 2 SIP gateway in place =
and if I
>>> am a client of that domain I know I cannot use an XMPP client for =
it.
>>=20
>> sorry but what would be the reason to have (and expose thru DNS) an
>> XMPP2SIP gateway and then not let it to be accessible from a client?

If the gateway can accept XMPP client connections, I see no problem. Is =
just that in my experience the XMPP client connections are handled by =
the native XMPP server of the XMPP domain and the gateway has limited =
functionality and will only be used for the translation to SIP and =
nothing else. Of course one can have a monolith server that handles =
everything but this does not change the topology of the call flow, there =
are still two different functions, handling the C2S connection and =
bridging outside to another server using a S2S connection.

>> are you assuming that a XMPP2SIP gateway is always only an outbound =
one?

It s used for both directions of course, I cannot see why would be used =
just in one direction.

> I suppose it would be the case where the domain only runs a SIP =
service internally, and just offers an XMPP gateway to external domains.

This is my scenario.

> So, if you have an XMPP-only client, you would have to get an account =
for some other domain to be able to communicate with this SIP-only =
domain.


Yes, this is a scenario where an operator provides a service using only =
one of the protocols (typically SIP) and only uses gateway for bridging =
traffic transparently to other XMPP domains without users having to know =
that the foreign user is using either SIP or XMPP. It is the SIP server =
that has the logic to make this routing decision and the gateway has a =
trusted relations ship with this server to avoid fraudulent usage of =
such service. The clients only talk to their respective native servers, =
not directly to the gateway. Same as clients do not talk directly to =
PSTN gateways, they rely on their servers to make such decisions.
=20


>=20
> Markus
>=20
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox
>=20


From Markus.Isomaki@nokia.com  Thu Jun 20 04:13:07 2013
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From: <Markus.Isomaki@nokia.com>
To: <emcho@jitsi.org>
Thread-Topic: Mapping for media signaling: still about the scope
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Subject: Re: [Stox] Mapping for media signaling: still about the scope
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Hi Emil,

See inline.

Emil Ivov [mailto:emcho@jitsi.org] wrote:

> On 19.06.13, 13:56, Markus.Isomaki@nokia.com wrote:
> > Hi Emil, Saul, Peter, and others,
> >
> > Recently I raised some questions about the scope of the 'sip-xmpp-media=
'
> > and the related deliverable on the 'mapping for media signaling':
> > http://www.ietf.org/mail-archive/web/stox/current/msg00025.html
> >
> > Now that STOX is officially up and running I'd like us to make an
> > actual plan on what the scope of this document should be, i.e. what
> > features are explicitly going to be part of this mapping spec.
> > Although people have successfully implemented SIP-to-XMPP media
> > signaling gateways, I at least see this topic to have some complexity d=
ue to:
> >
> > -SDP offer/answer is a moving target as new features are being added
> > as we speak,
>=20
> Well ... it is and it isn't. 3264 isn't really changing and all extension=
s are
> supposed to keep backward compatibility with it.
>=20

Right. That's good.

> > including "bundle" and "trickle-ICE". We can't know about what
> > features will be added later on.
>=20
> No, but we could include some that we know of ... like "bundle" and "tric=
kle-
> ICE" :).
>=20
> Trickle ICE would be easier because we already have it in Jingle and the =
SIP
> version shouldn't take long.
>=20

That's one of the "administrative" decisions we should make about the "firs=
t release" of the media signaling mapping *RFC*. Do we only take in feature=
s/extensions that are completed RFCs or XSF XEPs, so that we for quite big =
certainty can get a mature document to IESG in October. And then extend it =
with new features once they complete, perhaps one or two years later, perha=
ps in multiple small steps. Or, do we already take features like "bundle" a=
nd "trickle-ICE" in, and aim to get the document to IESG at some later poin=
t, for instance March 2014. Trickle-ICE for SIP may not take long, but I be=
t it will at least take longer than this WG is supposed to live ;-)

I don't have an opinion about this as a co-chair. This is a question to peo=
ple implementing and deploying the gateways or relying on them. What is the=
 most useful initial scope for you? How important it is to have an RFC out =
this year vs. covering as many features as possible? I mean technically a d=
raft might be as useful to an implementer as an RFC on those parts that are=
 stable while others are developed, but would there be some interoperabilit=
y/technical/political/marketing/getting-actually-something-completed reason=
 to get an RFC out ASAP?

The only guidance we have at this point is that our charter expects us to c=
omplete in October, and in this case, unlike the normal IETF WG process, I =
at least as a co-chair intend to take that seriously :-)

> > -There are features currently supported in XMPP that are not currently
> >   supported in SIP (and perhaps vice versa?), such as the trickle-ICE
> > support in Jingle.
>=20
> As mentioned. I don't think trickle-ICE would be a problem. I believe
> separating transport and encoding information would be a bigger hurdle bu=
t
> if we accept that urn:ietf:rfc:3264 means one shouldn't do this.
>=20
> > -Regardless of what is supported in each protocol in general, the
> > actual SIP and XMPP clients will support different sets of features.
>=20
> Right.
>=20
> > Ideally, some things might be mapped transparently in the gateway in
> > an "algorithmic" manner.
>=20
> I don't believe this would be possible.
>=20

Good, then my understanding seems to be correct.

> > And ideally, the clients supporting different feature sets can
> > negotiate end-to-end without needing for the gateway to mediate.
> > However, I'm not sure how easy this will be in practice.
> >
> > Actually one reason why I see these complications may be that I don't
> > have a very good overall understanding about how SDP O/A (as in RFC
> > 3264 syntax and semantics) is mapped to XMPP/Jingle (as in its XML
> > based
> > syntax) are in general kept "in synch". For instance, if IETF extends
> > SDP O/A in some way, will that always require an explicit
> > corresponding extension XEP by the XSF, or is there some more
> > straightforward "algorithmic" way to convey something new on the
> > SIP/SDP side to the XMPP/Jingle side.
>=20
> Well, the good thing about O/A extensions is that they generally preserve
> backward compatibility. This is valid for things like ICE, trickle ICE, r=
tcp-mux,
> bundle. So as long as we support 3264 we should be OK. I think :)
>=20

Yes, let's stick to that theory ;-)

...

>=20
> > -What will happen to features supported on one or either side of the
> > gateway that the gateway is not aware of- can some things be
> > "transparently" mapped or will the gateway become the "blocking factor"
> > in the middle. (I.e., will the session setup be limited to the
> > features the gateway explicitly needs to "understand".)
>=20
> I don't think the gateway could just forward anything because this would
> imply a straightforward mapping from any SDP attribute to a Jingle elemen=
t.
>=20
> Maybe gateways could behave as regular SIP B2BUAs: they forward the
> things that they understand and they remove those that they don't
> understand or support.
>=20

OK, that simplifies things as well. (Again was just checking I'm not missin=
g any end-to-end magic that might exist. But apparently not ;-)

> Thoughts?
>=20
> > -How will we extend the mapping spec in the future and how should the
> > initial spec take this into account.
>=20
> As in administratively?
>=20

In some sense yes. I would expect perhaps some brief section that gives gui=
delines to the authors who in the future want to extend this mapping spec w=
ith some new feature.

And would be rather do a full-fledged '-bis' spec at some point or a bunch =
or incremental extension drafts on top of the baseline. As an example "An e=
xtension to SIP-XMPP media signaling mapping to support trickle-ICE".=20

> Emil
>=20

Markus

From Markus.Isomaki@nokia.com  Thu Jun 20 04:17:27 2013
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Hi,

To my understanding this is now the most relevant discussion to have regard=
ing the STOX media signaling mapping draft. So, please send your input and =
let's try to converge on this so that our authors-to-be Emil, Saul (and Pet=
er) will know what to include in the next draft version.

Markus

> -----Original Message-----
> From: stox-bounces@ietf.org [mailto:stox-bounces@ietf.org] On Behalf Of
> ext Philipp Hancke
> Sent: 19 June, 2013 21:16
> To: stox@ietf.org
> Subject: Re: [Stox] Mapping for media signaling: still about the scope
>=20
> Am 19.06.2013 20:00, schrieb Emil Ivov: [...]
> > OK, so how about starting with 3261, 3264, 4585, 5285, 5761.
>=20
> That is the core i'm using in my sdp<->jingle mapping for webrtc purposes=
.
> Add 4568 (sdes, a=3Dcrypto), 5763 (a=3Dfingerprint for dtls-srtp) and pos=
sibly 5576
> (a=3Dssrc) to that list.
>=20
> > I think we can also add trickle ICE since I don't believe we have much
> > work left for SIP.
> >
> > Personally I wouldn't even mind having BUNDLE, we'd just need to have
> > the feature in Jingle.
>=20
> You can exploit the same-icepwd/ufrag property from
> http://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-
> 04#section-8.1
> -- I haven't checked for backward compability issues though. That way it
> seems we'd only need to have a disco feature for it along with a paragrap=
h in
> XEP-0176.
>=20
> cheers
>=20
> philipp
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox

From stpeter@stpeter.im  Thu Jun 20 10:55:58 2013
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From: Peter Saint-Andre <stpeter@stpeter.im>
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Cc: salvatore.loreto@ericsson.com, stox@ietf.org, Markus.Isomaki@nokia.com
Subject: Re: [Stox] core: connecting to remote domain
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On 6/20/13 4:53 AM, Adrian Georgescu wrote:
> 
> On Jun 20, 2013, at 12:37 PM, <Markus.Isomaki@nokia.com> wrote:
> 
>> Hi Sal,
>> 
>> Salvatore Loreto wrote:
>>> 
>>> On 6/20/13 9:28 AM, Adrian Georgescu wrote:
>>>> XMPP has defined DNS records for both C2S and S2S connections.
>>>> One domain may have a server to server XMPP capability but
>>>> would not accept client connections (missing correspondent  DNS
>>>> record), this would give me a clue that that foreign domain has
>>>> such XMPP 2 SIP gateway in place and if I am a client of that
>>>> domain I know I cannot use an XMPP client for it.
>>> 
>>> sorry but what would be the reason to have (and expose thru DNS)
>>> an XMPP2SIP gateway and then not let it to be accessible from a
>>> client?
> 
> If the gateway can accept XMPP client connections, I see no problem.
> Is just that in my experience the XMPP client connections are handled
> by the native XMPP server of the XMPP domain and the gateway has
> limited functionality and will only be used for the translation to
> SIP and nothing else. Of course one can have a monolith server that
> handles everything but this does not change the topology of the call
> flow, there are still two different functions, handling the C2S
> connection and bridging outside to another server using a S2S
> connection.
> 
>>> are you assuming that a XMPP2SIP gateway is always only an
>>> outbound one?
> 
> It s used for both directions of course, I cannot see why would be
> used just in one direction.
> 
>> I suppose it would be the case where the domain only runs a SIP
>> service internally, and just offers an XMPP gateway to external
>> domains.
> 
> This is my scenario.
> 
>> So, if you have an XMPP-only client, you would have to get an
>> account for some other domain to be able to communicate with this
>> SIP-only domain.
> 
> 
> Yes, this is a scenario where an operator provides a service using
> only one of the protocols (typically SIP) and only uses gateway for
> bridging traffic transparently to other XMPP domains without users
> having to know that the foreign user is using either SIP or XMPP. It
> is the SIP server that has the logic to make this routing decision
> and the gateway has a trusted relations ship with this server to
> avoid fraudulent usage of such service. The clients only talk to
> their respective native servers, not directly to the gateway. Same as
> clients do not talk directly to PSTN gateways, they rely on their
> servers to make such decisions.

Right, that makes sense.

I was thinking about the interdomain communication: if the SIP user
romeo@example.com sends a message to juliet@example.org, he doesn't need
to know or care if Juliet uses SIP or XMPP, but the service at
example.org needs to determine how it will route the message to Juliet
(i.e., using SIP or XMPP). So it needs to complete some kind of
discovery or it needs to be configured to "just know" that example.org
is (say) a SIP service.

However, while completing a copy edit on the groupchat spec this morning
I noticed that we already have the following text:

   Upon receiving the INVITE, the SIP-to-XMPP gateway needs to determine
   the identity of the remote domain, which it does by performing one or
   more DNS SRV lookups [RFC2782].  The SIP-to-XMPP gateway SHOULD
   resolve the address present in the To header of the INVITE to an im
   URI, then follow the rules in [RFC3861] regarding the "_im" SRV
   service for the target domain contained in the To header.  If SRV
   address resolution fails for the "_im" service, the SIP-to-XMPP
   gateway MAY attempt a lookup for the "_xmpp-server" service as
   specified in [RFC6120] or MAY return an error to the sender (i.e. 502
   Bad Gateway).

Now, that text is probably old (we've been working on these mapping
specs for a long time) and thus probably predates the updated XMPP RFCs.
If we're not going to recommend use of the "_im" and "_pres" SRV
records, then connections to remote domains a likely a matter for
implementation and deployment. However, I'd hate to do all this work on
the mapping specs and then leave the hard federation bits up to folks in
the field who might not be able to easily configure these services.

Peter

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



From stpeter@stpeter.im  Thu Jun 20 15:02:48 2013
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As promised, we've submitted a revised version of the groupchat spec.
This closes all the open issues we know about. Reviews and feedback are
welcome.

Peter


-------- Original Message --------
Subject: I-D Action: draft-saintandre-sip-xmpp-groupchat-04.txt
Date: Thu, 20 Jun 2013 15:00:45 -0700
From: internet-drafts@ietf.org
Reply-To: internet-drafts@ietf.org
To: i-d-announce@ietf.org


A New Internet-Draft is available from the on-line Internet-Drafts
directories.


	Title           : Interworking between the Session Initiation Protocol
(SIP) and the Extensible Messaging and Presence Protocol (XMPP): Groupchat
	Author(s)       : Peter Saint-Andre
                          Saul Ibarra
                          Salvatore Loreto
	Filename        : draft-saintandre-sip-xmpp-groupchat-04.txt
	Pages           : 32
	Date            : 2013-06-20

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


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

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

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


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

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Subject: Re: [Stox] core: connecting to remote domain
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>> Yes, this is a scenario where an operator provides a service using
>> only one of the protocols (typically SIP) and only uses gateway for
>> bridging traffic transparently to other XMPP domains without users
>> having to know that the foreign user is using either SIP or XMPP. It
>> is the SIP server that has the logic to make this routing decision
>> and the gateway has a trusted relations ship with this server to
>> avoid fraudulent usage of such service. The clients only talk to
>> their respective native servers, not directly to the gateway. Same as
>> clients do not talk directly to PSTN gateways, they rely on their
>> servers to make such decisions.
>=20
> Right, that makes sense.
>=20
> I was thinking about the interdomain communication: if the SIP user
> romeo@example.com sends a message to juliet@example.org, he doesn't =
need
> to know or care if Juliet uses SIP or XMPP, but the service at
> example.org needs to determine how it will route the message to Juliet
> (i.e., using SIP or XMPP). So it needs to complete some kind of
> discovery or it needs to be configured to "just know" that example.org
> is (say) a SIP service.
>=20
> However, while completing a copy edit on the groupchat spec this =
morning
> I noticed that we already have the following text:
>=20
>   Upon receiving the INVITE, the SIP-to-XMPP gateway needs to =
determine
>   the identity of the remote domain, which it does by performing one =
or
>   more DNS SRV lookups [RFC2782].  The SIP-to-XMPP gateway SHOULD
>   resolve the address present in the To header of the INVITE to an im
>   URI, then follow the rules in [RFC3861] regarding the "_im" SRV
>   service for the target domain contained in the To header.  If SRV
>   address resolution fails for the "_im" service, the SIP-to-XMPP
>   gateway MAY attempt a lookup for the "_xmpp-server" service as
>   specified in [RFC6120] or MAY return an error to the sender (i.e. =
502
>   Bad Gateway).

In practice we do lookup _xmmp_server, I  personally never saw a =
deployment of those _im records in the real world as of yet but others =
may confirm if this is the rule or the exception. Does anyone know a =
public service using those?

> Now, that text is probably old (we've been working on these mapping
> specs for a long time) and thus probably predates the updated XMPP =
RFCs.
> If we're not going to recommend use of the "_im" and "_pres" SRV
> records, then connections to remote domains a likely a matter for
> implementation and deployment.

It is a matter of implementation

> However, I'd hate to do all this work on
> the mapping specs and then leave the hard federation bits up to folks =
in
> the field who might not be able to easily configure these services.

Nothing breaks if a gateway tries to lookup those records too, it is =
harmless yet useless as far as I can see because of lack of adoption.=20

> Peter
>=20
> --=20
> Peter Saint-Andre
> https://stpeter.im/
>=20
>=20
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox
>=20


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On 6/21/13 12:49 AM, Adrian Georgescu wrote:
>>> Yes, this is a scenario where an operator provides a service
>>> using only one of the protocols (typically SIP) and only uses
>>> gateway for bridging traffic transparently to other XMPP domains
>>> without users having to know that the foreign user is using
>>> either SIP or XMPP. It is the SIP server that has the logic to
>>> make this routing decision and the gateway has a trusted
>>> relations ship with this server to avoid fraudulent usage of such
>>> service. The clients only talk to their respective native
>>> servers, not directly to the gateway. Same as clients do not talk
>>> directly to PSTN gateways, they rely on their servers to make
>>> such decisions.
>> 
>> Right, that makes sense.
>> 
>> I was thinking about the interdomain communication: if the SIP
>> user romeo@example.com sends a message to juliet@example.org, he
>> doesn't need to know or care if Juliet uses SIP or XMPP, but the
>> service at example.org needs to determine how it will route the
>> message to Juliet (i.e., using SIP or XMPP). So it needs to
>> complete some kind of discovery or it needs to be configured to
>> "just know" that example.org is (say) a SIP service.
>> 
>> However, while completing a copy edit on the groupchat spec this
>> morning I noticed that we already have the following text:
>> 
>> Upon receiving the INVITE, the SIP-to-XMPP gateway needs to
>> determine the identity of the remote domain, which it does by
>> performing one or more DNS SRV lookups [RFC2782].  The SIP-to-XMPP
>> gateway SHOULD resolve the address present in the To header of the
>> INVITE to an im URI, then follow the rules in [RFC3861] regarding
>> the "_im" SRV service for the target domain contained in the To
>> header.  If SRV address resolution fails for the "_im" service, the
>> SIP-to-XMPP gateway MAY attempt a lookup for the "_xmpp-server"
>> service as specified in [RFC6120] or MAY return an error to the
>> sender (i.e. 502 Bad Gateway).
> 
> In practice we do lookup _xmmp_server, I  personally never saw a
> deployment of those _im records in the real world as of yet but
> others may confirm if this is the rule or the exception. Does anyone
> know a public service using those?

I deployed them at jabber.org, but we might be the only ones. :-)

>> Now, that text is probably old (we've been working on these
>> mapping specs for a long time) and thus probably predates the
>> updated XMPP RFCs. If we're not going to recommend use of the "_im"
>> and "_pres" SRV records, then connections to remote domains a
>> likely a matter for implementation and deployment.
> 
> It is a matter of implementation

Although I'd like to have something more automated, I think you're right.

>> However, I'd hate to do all this work on the mapping specs and then
>> leave the hard federation bits up to folks in the field who might
>> not be able to easily configure these services.
> 
> Nothing breaks if a gateway tries to lookup those records too, it is
> harmless yet useless as far as I can see because of lack of adoption.

Sadly, yes.

Peter

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



From Markus.Isomaki@nokia.com  Fri Jun 28 01:22:40 2013
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Hi,

Please notice:

In the preliminary agenda for IETF 87, STOX WG has been reserved a slot at =
the 1700-1830 session on Thursday:
http://tools.ietf.org/agenda/87/#THURSDAY

Markus=20


