
From mary.ietf.barnes@gmail.com  Mon Apr  1 08:48:42 2013
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Date: Mon, 1 Apr 2013 10:48:40 -0500
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From: Mary Barnes <mary.ietf.barnes@gmail.com>
To: Parthasarathi R <partha@parthasarathi.co.in>
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Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or websocket
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Partha,

It's not entirely clear to me that RTCWEB is the "compelling" use case
for MSRP over websockets.  I'd be interested to hear who might think
that is the case.  I can envision other Real Time applications using
web servers to make use of this, as well.

I don't see any requirements in the RTCWEB WG (charter or
requirements) that indicate this is a compelling use case:
http://www.ietf.org/id/draft-ietf-rtcweb-use-cases-and-requirements-10.txt
There are no use cases for IM, Chat or MSRP specifically that I could
find and I couldn't find anything in the RTCWEB archives:
http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requirements-10
If there has been discussion, please provide a pointer.

The discussion of MSRP over websockets should continue on this mailing
list.  Per the dispatching of IETF-86 topics, the current proposal is
to AD sponsor this document unless there are very compelling reasons
not to publish.
http://tools.ietf.org/wg/dispatch/trac/wiki

The proposal had been for the general discussion of "foo" over
websockets to happen on the RAI list.    The last I looked there are
at least 200 people subscribed to DISPATCH that are not on the RAI
area mailing list.  I strongly encourage folks with an interest in the
topic to make sure they are subscribe to the RAI area mailing list:
https://www.ietf.org/mailman/listinfo/rai

Regards,
Mary.

On Sun, Mar 31, 2013 at 11:52 AM, Parthasarathi R
<partha@parthasarathi.co.in> wrote:
> The compelling reason for this draft is RTCWeb. In case of non-RTCWeb,
> TCP Shall be used which avoids the overhead introduced due to
> WebSocket. Please clarify in case WebSocket has some specific
> advantage over TCP in non-RTCWeb environment.
>
> In case dispatch or BFCPbis are not the right WG to make recommendation,
> then let us discuss in RTCWeb WG itself. Also, Please suggest if any
> other mailing list like (rai@ietf.org) is the right place to discuss.
>
> Thanks
> Partha
>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>> Behalf Of Mary Barnes
>> Sent: Thursday, March 28, 2013 8:28 PM
>> To: DRAGE, Keith (Keith)
>> Cc: dispatch@ietf.org
>> Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or
>> websocket
>>
>> I don't think it's the job of dispatch or BFCPbis WG to define or even
>> make recommendations for RTCWEB.  That choice is up to the RTCWEB WG
>> and it shouldn't be influencing this work.  There are applications
>> other than RTCWEB that can and will make use of BFCPbis over
>> websockets.  I really don't understand why we are getting so hung up
>> on this - it's the BFCP messages themselves that are important - how
>> one chooses to transport is up to the application.
>>
>> Mary.
>>
>> On Thu, Mar 28, 2013 at 7:35 AM, DRAGE, Keith (Keith)
>> <keith.drage@alcatel-lucent.com> wrote:
>> > From what I have seen of the discussion so far, there appears to be a
>> body of support that there is a use case for both BFCP and MSRP over
>> websocket, for use in a non-rtcweb environment.
>> >
>> > If when we get to the rtcweb environment we allow this, and also
>> allow a usage over datachannel version, we essentially have two
>> solutions in the market place.
>> >
>> > If that occurs, what guidance will we give to a rtcweb developer as
>> to which to implement?
>> >
>> > Or do we envisage that only one will be allowed (and if so which) in
>> an rtcweb environment?
>> >
>> >
>> >
>> > Regards
>> >
>> > Keith
>> > _______________________________________________
>> > dispatch mailing list
>> > dispatch@ietf.org
>> > https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>

From partha@parthasarathi.co.in  Mon Apr  1 10:12:05 2013
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Mary Barnes'" <mary.ietf.barnes@gmail.com>
References: <949EF20990823C4C85C18D59AA11AD8B01D5F4@FR712WXCHMBA11.zeu.alcatel-lucent.com>	<CAHBDyN5zZ3f6mjfXVXbvvvB7FTTKPWfDHvC9CaHQgau4_8uL1w@mail.gmail.com>	<004301ce2e30$2317e700$6947b500$@co.in> <CAHBDyN6mBMwE2pFMh-iT4i302pqRDvBXkn+7NctrdVrVkJ+2iQ@mail.gmail.com>
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Date: Mon, 1 Apr 2013 22:41:55 +0530
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Cc: 'DISPATCH' <dispatch@ietf.org>
Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or websocket
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Mary,

I think that my point on RTCWEB does not come out well.
Browser has the limitation of including TCP applications (like 
BFCP over TCP) directly. To achieve this Datachannel or WebSocket 
Shall be used which is under discussion here. As you mentioned, 
RTCWeb WG does not cover these usecases. The discussion 
required in RTCWeb WG whether BFCP and MSRP usecase are qualified
as usecase or as the additional usecase (sec 8 of 
draft-ietf-rtcweb-use-cases-and-requirements-10) as other 
conference related usecases exist in additional usecase already. 

Please note that browser limitation does not apply for the generic
Web (HTTP) client. In your envisioned Real time applications using 
web servers, web client shall use TCP for BFCP & MSRP as web client
MUST support TCP protocol. It is not clear to me why websocket is
required for those real time applications. 

As you suggest, Let us discuss in RAI for further discussion on
"foo" over WebSocket or Datachannel.

Thanks
Partha

> -----Original Message-----
> From: Mary Barnes [mailto:mary.ietf.barnes@gmail.com]
> Sent: Monday, April 01, 2013 9:19 PM
> To: Parthasarathi R
> Cc: DRAGE, Keith (Keith); DISPATCH
> Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or
> websocket
> 
> Partha,
> 
> It's not entirely clear to me that RTCWEB is the "compelling" use case
> for MSRP over websockets.  I'd be interested to hear who might think
> that is the case.  I can envision other Real Time applications using
> web servers to make use of this, as well.
> 
> I don't see any requirements in the RTCWEB WG (charter or
> requirements) that indicate this is a compelling use case:
> http://www.ietf.org/id/draft-ietf-rtcweb-use-cases-and-requirements-
> 10.txt
> There are no use cases for IM, Chat or MSRP specifically that I could
> find and I couldn't find anything in the RTCWEB archives:
> http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-
> requirements-10
> If there has been discussion, please provide a pointer.
> 
> The discussion of MSRP over websockets should continue on this mailing
> list.  Per the dispatching of IETF-86 topics, the current proposal is
> to AD sponsor this document unless there are very compelling reasons
> not to publish.
> http://tools.ietf.org/wg/dispatch/trac/wiki
> 
> The proposal had been for the general discussion of "foo" over
> websockets to happen on the RAI list.    The last I looked there are
> at least 200 people subscribed to DISPATCH that are not on the RAI
> area mailing list.  I strongly encourage folks with an interest in the
> topic to make sure they are subscribe to the RAI area mailing list:
> https://www.ietf.org/mailman/listinfo/rai
> 
> Regards,
> Mary.
> 
> On Sun, Mar 31, 2013 at 11:52 AM, Parthasarathi R
> <partha@parthasarathi.co.in> wrote:
> > The compelling reason for this draft is RTCWeb. In case of non-
> RTCWeb,
> > TCP Shall be used which avoids the overhead introduced due to
> > WebSocket. Please clarify in case WebSocket has some specific
> > advantage over TCP in non-RTCWeb environment.
> >
> > In case dispatch or BFCPbis are not the right WG to make
> recommendation,
> > then let us discuss in RTCWeb WG itself. Also, Please suggest if any
> > other mailing list like (rai@ietf.org) is the right place to discuss.
> >
> > Thanks
> > Partha
> >
> >> -----Original Message-----
> >> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org]
> On
> >> Behalf Of Mary Barnes
> >> Sent: Thursday, March 28, 2013 8:28 PM
> >> To: DRAGE, Keith (Keith)
> >> Cc: dispatch@ietf.org
> >> Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or
> >> websocket
> >>
> >> I don't think it's the job of dispatch or BFCPbis WG to define or
> even
> >> make recommendations for RTCWEB.  That choice is up to the RTCWEB WG
> >> and it shouldn't be influencing this work.  There are applications
> >> other than RTCWEB that can and will make use of BFCPbis over
> >> websockets.  I really don't understand why we are getting so hung up
> >> on this - it's the BFCP messages themselves that are important - how
> >> one chooses to transport is up to the application.
> >>
> >> Mary.
> >>
> >> On Thu, Mar 28, 2013 at 7:35 AM, DRAGE, Keith (Keith)
> >> <keith.drage@alcatel-lucent.com> wrote:
> >> > From what I have seen of the discussion so far, there appears to
> be a
> >> body of support that there is a use case for both BFCP and MSRP over
> >> websocket, for use in a non-rtcweb environment.
> >> >
> >> > If when we get to the rtcweb environment we allow this, and also
> >> allow a usage over datachannel version, we essentially have two
> >> solutions in the market place.
> >> >
> >> > If that occurs, what guidance will we give to a rtcweb developer
> as
> >> to which to implement?
> >> >
> >> > Or do we envisage that only one will be allowed (and if so which)
> in
> >> an rtcweb environment?
> >> >
> >> >
> >> >
> >> > Regards
> >> >
> >> > Keith
> >> > _______________________________________________
> >> > dispatch mailing list
> >> > dispatch@ietf.org
> >> > https://www.ietf.org/mailman/listinfo/dispatch
> >> _______________________________________________
> >> dispatch mailing list
> >> dispatch@ietf.org
> >> https://www.ietf.org/mailman/listinfo/dispatch
> >


From ibc@aliax.net  Tue Apr  2 04:50:52 2013
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 2 Apr 2013 13:50:31 +0200
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Cc: dispatch@ietf.org
Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or websocket
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2013/3/31 Parthasarathi R <partha@parthasarathi.co.in>:
> The compelling reason for this draft is RTCWeb.

Why? where did you get such a conclusion from?


> In case of non-RTCWeb,
> TCP Shall be used which avoids the overhead introduced due to
> WebSocket. Please clarify in case WebSocket has some specific
> advantage over TCP in non-RTCWeb environment.
>
> In case dispatch or BFCPbis are not the right WG to make recommendation,
> then let us discuss in RTCWeb WG itself.

The RTCWeb WG has nothing to do with *what* is sent over a DataChannel
communication.



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

From ibc@aliax.net  Tue Apr  2 05:02:22 2013
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 2 Apr 2013 14:02:01 +0200
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To: Parthasarathi R <partha@parthasarathi.co.in>
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Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or websocket
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2013/4/1 Parthasarathi R <partha@parthasarathi.co.in>:
>
> I think that my point on RTCWEB does not come out well.
> Browser has the limitation of including TCP applications (like
> BFCP over TCP) directly. To achieve this Datachannel or WebSocket
> Shall be used which is under discussion here. As you mentioned,
> RTCWeb WG does not cover these usecases. The discussion
> required in RTCWeb WG whether BFCP and MSRP usecase are qualified
> as usecase or as the additional usecase (sec 8 of
> draft-ietf-rtcweb-use-cases-and-requirements-10) as other
> conference related usecases exist in additional usecase already.

You can send BFCP messages over WebSocket and you can also send them
over a WebRTC DataChannel "connection". So this is not a "WebRTC use
case" that should be defined in
draft-ietf-rtcweb-use-cases-and-requirements because anyone can send
*whatever* he wants over a DataChannel connection or over a WebSocket
connection.

Please don't disturb the RTCWEB WG with BFCP stuff.


> In your envisioned Real time applications using
> web servers, web client shall use TCP for BFCP & MSRP as web client
> MUST support TCP protocol.

"web client MUST support TCP protocol" ??

Sorry but, is that a joke? It's obvious that a web browser cannot open
a raw TCP connection, that's why we need WebSocket as the *standard*
realtime and bidirectional communication mechanism for applications
running in web browsers.


> It is not clear to me why websocket is
> required for those real time applications.

Realtime bidirectional communication from browser to server and vice
versa. What other *standard* mechanism you know for realtime
bidirectional communication in web applications? Please let me know
(reply please).

Regards.




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

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FYI.


-------- Original Message --------
Subject: I-D Action: draft-saintandre-sip-xmpp-core-04.txt
Date: Tue, 02 Apr 2013 21:00:20 -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):
Addresses and Error Conditions
	Author(s)       : Peter Saint-Andre
                          Avshalom Houri
                          Joe Hildebrand
	Filename        : draft-saintandre-sip-xmpp-core-04.txt
	Pages           : 14
	Date            : 2013-04-02

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-04

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


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

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FYI.


-------- Original Message --------
Subject: I-D Action: draft-saintandre-sip-xmpp-presence-05.txt
Date: Tue, 02 Apr 2013 21:02:57 -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): Presence
	Author(s)       : Peter Saint-Andre
                          Avshalom Houri
                          Joe Hildebrand
	Filename        : draft-saintandre-sip-xmpp-presence-05.txt
	Pages           : 20
	Date            : 2013-04-02

Abstract:
   This document defines a bi-directional protocol mapping for the
   exchange of presence information between the Session Initiation
   Protocol (SIP) and the Extensible Messaging and Presence Protocol
   (XMPP).


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

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

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


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

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Subject: Re: [dispatch] Comments on draft-saintandre-sip-xmpp-presence-04
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Michael, I think your feedback has been addressed in -05. Please check.

On 3/9/13 5:29 PM, Michael Lundberg wrote:
> Peter,
> 
> Thanks for resubmitting the SIP-XMPP presence I-D.  I think this
> and the other documents are important as there are a lot of
> interworking issues today with XMPP and SIP/SIMPLE implementations,
> and I think these documents can help reduce some of those
> interworking challenges.
> 
> In order to help ensure interworking between different 
> implementations, I think Sections 5.2 and 5.3 need to have some 
> stronger â€˜MUSTâ€™ language instead of SHOULDs, MAYs, etc.  This will 
> enforce proper and complete mappings between implementations that 
> conform to this document.
> 
> While I donâ€™t think the document needs to enforce the use of the 
> <show/> element mapping discussed in 5.2, I think if
> implementations are going to use the mapping, this document should
> enforce the use of a specific namespace and describe the mapping of
> <show> values specified in RFC 6121 (i.e., away, dnd, chat, and
> xa).  This will at least allow a minimum set of standard mappings
> between XMPP and SIP, which can be expanded on in future
> document(s).  Section 5.3 could then also discuss how this mapping
> can be accomplished from SIP to XMPP if the SIP endpoints/devices
> support this same namespace.  One of the big interworking
> challenges with todayâ€™s implementations is the use of different
> namespaces given the extensibility of XML-based presence standards
> (e.g., PIDF, RPID), which allows individuals to create their own
> namespace.  This flexibility is good for ingenuity, but problematic
> for interworking between different implementations.  I think
> standardizing a basic set of common status information using a 
> specific namespace is needed, and this document is a good place to
> do that.
> 
> In my opinion, I do not think Section 6 should be part of this
> base document.  While I like the idea of encapsulating the SIP
> <presence> information into the XMPP message, I think this content
> should be a consideration for implementing â€˜rich presenceâ€™.  As
> written, this method is only applicable for SIP to XMPP
> translation, and if the focus of this document is just â€˜basicâ€™
> presence information interworking, the SIP to XMPP mapping in
> Section 5.3 already describes a method for how this can be
> accomplished.
> 
> Regards, Michael _______________________________________________ 
> dispatch mailing list dispatch@ietf.org 
> https://www.ietf.org/mailman/listinfo/dispatch
> 

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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: =?UTF-8?Q?'I=C3=B1aki_Baz_Castillo'?= <ibc@aliax.net>
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Cc: 'DISPATCH' <dispatch@ietf.org>
Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or websocket
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Inaki,

As Mary suggested, Let us discuss data channel vs WebSocket in
RAI list.

Please look into snip from your earlier mail which answer your=20
own query of Why BFCP over Websocket has to be discussed in RTCWeb=20
WG.

<snip1>
anyone can send
> *whatever* he wants over a DataChannel connection or over a WebSocket
> connection.
>=20
> Please don't disturb the RTCWEB WG with BFCP stuff.
>
</snip1>

<snip2>  It's obvious that a web browser cannot open
> a raw TCP connection, that's why we need WebSocket as the *standard*
> realtime and bidirectional communication mechanism for applications
> running in web browsers. </snip2>


Thanks
Partha

> -----Original Message-----
> From: I=C3=B1aki Baz Castillo [mailto:ibc@aliax.net]
> Sent: Tuesday, April 02, 2013 5:32 PM
> To: Parthasarathi R
> Cc: Mary Barnes; DISPATCH
> Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or
> websocket
>=20
> 2013/4/1 Parthasarathi R <partha@parthasarathi.co.in>:
> >
> > I think that my point on RTCWEB does not come out well.
> > Browser has the limitation of including TCP applications (like
> > BFCP over TCP) directly. To achieve this Datachannel or WebSocket
> > Shall be used which is under discussion here. As you mentioned,
> > RTCWeb WG does not cover these usecases. The discussion
> > required in RTCWeb WG whether BFCP and MSRP usecase are qualified
> > as usecase or as the additional usecase (sec 8 of
> > draft-ietf-rtcweb-use-cases-and-requirements-10) as other
> > conference related usecases exist in additional usecase already.
>=20
> You can send BFCP messages over WebSocket and you can also send them
> over a WebRTC DataChannel "connection". So this is not a "WebRTC use
> case" that should be defined in
> draft-ietf-rtcweb-use-cases-and-requirements because anyone can send
> *whatever* he wants over a DataChannel connection or over a WebSocket
> connection.
>=20
> Please don't disturb the RTCWEB WG with BFCP stuff.
>=20
>=20
> > In your envisioned Real time applications using
> > web servers, web client shall use TCP for BFCP & MSRP as web client
> > MUST support TCP protocol.
>=20
> "web client MUST support TCP protocol" ??
>=20
> Sorry but, is that a joke? It's obvious that a web browser cannot open
> a raw TCP connection, that's why we need WebSocket as the *standard*
> realtime and bidirectional communication mechanism for applications
> running in web browsers.
>=20
>=20
> > It is not clear to me why websocket is
> > required for those real time applications.
>=20
> Realtime bidirectional communication from browser to server and vice
> versa. What other *standard* mechanism you know for realtime
> bidirectional communication in web applications? Please let me know
> (reply please).
>=20
> Regards.
>=20
>=20
>=20
>=20
> --
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 3 Apr 2013 19:27:31 +0200
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Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or websocket
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--001a11c27e3269a3bb04d9782c0c
Content-Type: text/plain; charset=UTF-8
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Thanks Partha. As I expected you don't reply to my question (you never do
it). Please let me try it again:


> It is not clear to me why websocket is
> required for those real time applications.

Realtime bidirectional communication from browser to server and vice
versa. What other *standard* mechanism you know for realtime
bidirectional communication in web applications? Please let me know
(reply please).


Regards.



2013/4/3 Parthasarathi R <partha@parthasarathi.co.in>

> Inaki,
>
> As Mary suggested, Let us discuss data channel vs WebSocket in
> RAI list.
>
> Please look into snip from your earlier mail which answer your
> own query of Why BFCP over Websocket has to be discussed in RTCWeb
> WG.
>
> <snip1>
> anyone can send
> > *whatever* he wants over a DataChannel connection or over a WebSocket
> > connection.
> >
> > Please don't disturb the RTCWEB WG with BFCP stuff.
> >
> </snip1>
>
> <snip2>  It's obvious that a web browser cannot open
> > a raw TCP connection, that's why we need WebSocket as the *standard*
> > realtime and bidirectional communication mechanism for applications
> > running in web browsers. </snip2>
>
>
> Thanks
> Partha
>
> > -----Original Message-----
> > From: I=C3=B1aki Baz Castillo [mailto:ibc@aliax.net]
> > Sent: Tuesday, April 02, 2013 5:32 PM
> > To: Parthasarathi R
> > Cc: Mary Barnes; DISPATCH
> > Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or
> > websocket
> >
> > 2013/4/1 Parthasarathi R <partha@parthasarathi.co.in>:
> > >
> > > I think that my point on RTCWEB does not come out well.
> > > Browser has the limitation of including TCP applications (like
> > > BFCP over TCP) directly. To achieve this Datachannel or WebSocket
> > > Shall be used which is under discussion here. As you mentioned,
> > > RTCWeb WG does not cover these usecases. The discussion
> > > required in RTCWeb WG whether BFCP and MSRP usecase are qualified
> > > as usecase or as the additional usecase (sec 8 of
> > > draft-ietf-rtcweb-use-cases-and-requirements-10) as other
> > > conference related usecases exist in additional usecase already.
> >
> > You can send BFCP messages over WebSocket and you can also send them
> > over a WebRTC DataChannel "connection". So this is not a "WebRTC use
> > case" that should be defined in
> > draft-ietf-rtcweb-use-cases-and-requirements because anyone can send
> > *whatever* he wants over a DataChannel connection or over a WebSocket
> > connection.
> >
> > Please don't disturb the RTCWEB WG with BFCP stuff.
> >
> >
> > > In your envisioned Real time applications using
> > > web servers, web client shall use TCP for BFCP & MSRP as web client
> > > MUST support TCP protocol.
> >
> > "web client MUST support TCP protocol" ??
> >
> > Sorry but, is that a joke? It's obvious that a web browser cannot open
> > a raw TCP connection, that's why we need WebSocket as the *standard*
> > realtime and bidirectional communication mechanism for applications
> > running in web browsers.
> >
> >
> > > It is not clear to me why websocket is
> > > required for those real time applications.
> >
> > Realtime bidirectional communication from browser to server and vice
> > versa. What other *standard* mechanism you know for realtime
> > bidirectional communication in web applications? Please let me know
> > (reply please).
> >
> > Regards.
> >
> >
> >
> >
> > --
> > I=C3=B1aki Baz Castillo
> > <ibc@aliax.net>
>
>


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

--001a11c27e3269a3bb04d9782c0c
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Thanks Partha. As I expected you don&#39;t reply to my que=
stion (you never do it). Please let me try it again:<div><br></div><div><br=
></div><div><div class=3D"im" style=3D"font-family:arial,sans-serif;font-si=
ze:13px">

&gt; It is not clear to me why websocket is<br>&gt; required for those real=
 time applications.<br><br></div><span style=3D"font-family:arial,sans-seri=
f;font-size:13px">Realtime bidirectional communication from browser to serv=
er and vice</span><br style=3D"font-family:arial,sans-serif;font-size:13px"=
>

<span style=3D"font-family:arial,sans-serif;font-size:13px">versa. What oth=
er *standard* mechanism you know for realtime</span><br style=3D"font-famil=
y:arial,sans-serif;font-size:13px"><span style=3D"font-family:arial,sans-se=
rif;font-size:13px">bidirectional communication in web applications? Please=
 let me know</span><br style=3D"font-family:arial,sans-serif;font-size:13px=
">

<span style=3D"font-family:arial,sans-serif;font-size:13px">(reply please).=
</span><br style=3D"font-family:arial,sans-serif;font-size:13px"></div><div=
><span style=3D"font-family:arial,sans-serif;font-size:13px"><br></span></d=
iv>

<div><span style=3D"font-family:arial,sans-serif;font-size:13px"><br></span=
></div><div style><span style=3D"font-family:arial,sans-serif;font-size:13p=
x">Regards.</span></div><div style><span style=3D"font-family:arial,sans-se=
rif;font-size:13px"><br>

</span></div></div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_q=
uote">2013/4/3 Parthasarathi R <span dir=3D"ltr">&lt;<a href=3D"mailto:part=
ha@parthasarathi.co.in" target=3D"_blank">partha@parthasarathi.co.in</a>&gt=
;</span><br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Inaki,<br>
<br>
As Mary suggested, Let us discuss data channel vs WebSocket in<br>
RAI list.<br>
<br>
Please look into snip from your earlier mail which answer your<br>
own query of Why BFCP over Websocket has to be discussed in RTCWeb<br>
WG.<br>
<br>
&lt;snip1&gt;<br>
<div class=3D"im">anyone can send<br>
&gt; *whatever* he wants over a DataChannel connection or over a WebSocket<=
br>
&gt; connection.<br>
&gt;<br>
&gt; Please don&#39;t disturb the RTCWEB WG with BFCP stuff.<br>
&gt;<br>
</div>&lt;/snip1&gt;<br>
<br>
&lt;snip2&gt; =C2=A0It&#39;s obvious that a web browser cannot open<br>
<div class=3D"im">&gt; a raw TCP connection, that&#39;s why we need WebSock=
et as the *standard*<br>
&gt; realtime and bidirectional communication mechanism for applications<br=
>
</div>&gt; running in web browsers. &lt;/snip2&gt;<br>
<br>
<br>
Thanks<br>
Partha<br>
<div class=3D"im HOEnZb"><br>
&gt; -----Original Message-----<br>
&gt; From: I=C3=B1aki Baz Castillo [mailto:<a href=3D"mailto:ibc@aliax.net"=
>ibc@aliax.net</a>]<br>
&gt; Sent: Tuesday, April 02, 2013 5:32 PM<br>
&gt; To: Parthasarathi R<br>
</div><div class=3D"im HOEnZb">&gt; Cc: Mary Barnes; DISPATCH<br>
&gt; Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or<br>
&gt; websocket<br>
&gt;<br>
</div><div class=3D"HOEnZb"><div class=3D"h5">&gt; 2013/4/1 Parthasarathi R=
 &lt;<a href=3D"mailto:partha@parthasarathi.co.in">partha@parthasarathi.co.=
in</a>&gt;:<br>
&gt; &gt;<br>
&gt; &gt; I think that my point on RTCWEB does not come out well.<br>
&gt; &gt; Browser has the limitation of including TCP applications (like<br=
>
&gt; &gt; BFCP over TCP) directly. To achieve this Datachannel or WebSocket=
<br>
&gt; &gt; Shall be used which is under discussion here. As you mentioned,<b=
r>
&gt; &gt; RTCWeb WG does not cover these usecases. The discussion<br>
&gt; &gt; required in RTCWeb WG whether BFCP and MSRP usecase are qualified=
<br>
&gt; &gt; as usecase or as the additional usecase (sec 8 of<br>
&gt; &gt; draft-ietf-rtcweb-use-cases-and-requirements-10) as other<br>
&gt; &gt; conference related usecases exist in additional usecase already.<=
br>
&gt;<br>
&gt; You can send BFCP messages over WebSocket and you can also send them<b=
r>
&gt; over a WebRTC DataChannel &quot;connection&quot;. So this is not a &qu=
ot;WebRTC use<br>
&gt; case&quot; that should be defined in<br>
&gt; draft-ietf-rtcweb-use-cases-and-requirements because anyone can send<b=
r>
&gt; *whatever* he wants over a DataChannel connection or over a WebSocket<=
br>
&gt; connection.<br>
&gt;<br>
&gt; Please don&#39;t disturb the RTCWEB WG with BFCP stuff.<br>
&gt;<br>
&gt;<br>
&gt; &gt; In your envisioned Real time applications using<br>
&gt; &gt; web servers, web client shall use TCP for BFCP &amp; MSRP as web =
client<br>
&gt; &gt; MUST support TCP protocol.<br>
&gt;<br>
&gt; &quot;web client MUST support TCP protocol&quot; ??<br>
&gt;<br>
&gt; Sorry but, is that a joke? It&#39;s obvious that a web browser cannot =
open<br>
&gt; a raw TCP connection, that&#39;s why we need WebSocket as the *standar=
d*<br>
&gt; realtime and bidirectional communication mechanism for applications<br=
>
&gt; running in web browsers.<br>
&gt;<br>
&gt;<br>
&gt; &gt; It is not clear to me why websocket is<br>
&gt; &gt; required for those real time applications.<br>
&gt;<br>
&gt; Realtime bidirectional communication from browser to server and vice<b=
r>
&gt; versa. What other *standard* mechanism you know for realtime<br>
&gt; bidirectional communication in web applications? Please let me know<br=
>
&gt; (reply please).<br>
&gt;<br>
&gt; Regards.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; --<br>
&gt; I=C3=B1aki Baz Castillo<br>
&gt; &lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
<br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
I=C3=B1aki Baz Castillo<br>&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.n=
et</a>&gt;
</div>

--001a11c27e3269a3bb04d9782c0c--

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Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or websocket
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2013/4/3 Parthasarathi R <partha@parthasarathi.co.in>

> Please look into snip from your earlier mail which answer your
> own query of Why BFCP over Websocket has to be discussed in RTCWeb
> WG.
>

Regardless to this, as you know "SIP over WebSocket" (
http://datatracker.ietf.org/doc/draft-ietf-sipcore-sip-websocket/) is in
Last Call status, and we required nothing from RTCWEB WG for that, so the
same for "BFCP over WebSocket" or "Anything related to SIP over WebSocket".

Best regards.

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

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<div dir=3D"ltr"><div class=3D"gmail_extra">2013/4/3 Parthasarathi R <span =
dir=3D"ltr">&lt;<a href=3D"mailto:partha@parthasarathi.co.in" target=3D"_bl=
ank">partha@parthasarathi.co.in</a>&gt;</span><br><div class=3D"gmail_quote=
"><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;borde=
r-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid=
;padding-left:1ex">

<div id=3D":48v">Please look into snip from your earlier mail which answer =
your<br>
own query of Why BFCP over Websocket has to be discussed in RTCWeb<br>
WG.</div></blockquote></div><br>Regardless to this, as you know &quot;SIP o=
ver WebSocket&quot; (<a href=3D"http://datatracker.ietf.org/doc/draft-ietf-=
sipcore-sip-websocket/">http://datatracker.ietf.org/doc/draft-ietf-sipcore-=
sip-websocket/</a>) is in Last Call status, and we required nothing from RT=
CWEB WG for that, so the same for &quot;BFCP over WebSocket&quot; or &quot;=
Anything related to SIP over WebSocket&quot;.<br>

<br clear=3D"all"><div style>Best regards.</div><div style><br></div>-- <br=
>I=C3=B1aki Baz Castillo<br>&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.=
net</a>&gt;
</div></div>

--047d7bdc9612a5bf4104d9783573--

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Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or websocket
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On 4/4/13 1:27 AM, Iñaki Baz Castillo wrote:
> Thanks Partha. As I expected you don't reply to my question (you never
> do it). Please let me try it again:
>
>
>  > It is not clear to me why websocket is
>  > required for those real time applications.
>
> Realtime bidirectional communication from browser to server and vice
> versa. What other *standard* mechanism you know for realtime
> bidirectional communication in web applications? Please let me know
> (reply please).

I see a significant difference between "media" streams such as msrp and 
bfcp, and the sip over websocket spec.

For sip over websocket there is little expectation that the target of 
the websocket is the ultimate destination. It is more likely to be an 
outbound proxy. So interop with peers that aren't web servers is preserved.

That may be true for msrp too, since a caller-specific msrp relay can be 
used to ensure interop.

It is more of a problem for bfcp, since AFAIK there is no spec for a 
bfcp relay. In general you can't know in advance if the peer in your sip 
call is a web server, so you can't depend on being able to set up a 
websocket to that peer.

I am not decided on what is the right answer for bfcp. But I have 
succeeded in getting the issue discussed more widely.

	Thanks,
	Paul

> Regards.
>
>
>
> 2013/4/3 Parthasarathi R <partha@parthasarathi.co.in
> <mailto:partha@parthasarathi.co.in>>
>
>     Inaki,
>
>     As Mary suggested, Let us discuss data channel vs WebSocket in
>     RAI list.
>
>     Please look into snip from your earlier mail which answer your
>     own query of Why BFCP over Websocket has to be discussed in RTCWeb
>     WG.
>
>     <snip1>
>     anyone can send
>      > *whatever* he wants over a DataChannel connection or over a WebSocket
>      > connection.
>      >
>      > Please don't disturb the RTCWEB WG with BFCP stuff.
>      >
>     </snip1>
>
>     <snip2>  It's obvious that a web browser cannot open
>      > a raw TCP connection, that's why we need WebSocket as the *standard*
>      > realtime and bidirectional communication mechanism for applications
>      > running in web browsers. </snip2>
>
>
>     Thanks
>     Partha
>
>      > -----Original Message-----
>      > From: Iñaki Baz Castillo [mailto:ibc@aliax.net
>     <mailto:ibc@aliax.net>]
>      > Sent: Tuesday, April 02, 2013 5:32 PM
>      > To: Parthasarathi R
>      > Cc: Mary Barnes; DISPATCH
>      > Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or
>      > websocket
>      >
>      > 2013/4/1 Parthasarathi R <partha@parthasarathi.co.in
>     <mailto:partha@parthasarathi.co.in>>:
>      > >
>      > > I think that my point on RTCWEB does not come out well.
>      > > Browser has the limitation of including TCP applications (like
>      > > BFCP over TCP) directly. To achieve this Datachannel or WebSocket
>      > > Shall be used which is under discussion here. As you mentioned,
>      > > RTCWeb WG does not cover these usecases. The discussion
>      > > required in RTCWeb WG whether BFCP and MSRP usecase are qualified
>      > > as usecase or as the additional usecase (sec 8 of
>      > > draft-ietf-rtcweb-use-cases-and-requirements-10) as other
>      > > conference related usecases exist in additional usecase already.
>      >
>      > You can send BFCP messages over WebSocket and you can also send them
>      > over a WebRTC DataChannel "connection". So this is not a "WebRTC use
>      > case" that should be defined in
>      > draft-ietf-rtcweb-use-cases-and-requirements because anyone can send
>      > *whatever* he wants over a DataChannel connection or over a WebSocket
>      > connection.
>      >
>      > Please don't disturb the RTCWEB WG with BFCP stuff.
>      >
>      >
>      > > In your envisioned Real time applications using
>      > > web servers, web client shall use TCP for BFCP & MSRP as web client
>      > > MUST support TCP protocol.
>      >
>      > "web client MUST support TCP protocol" ??
>      >
>      > Sorry but, is that a joke? It's obvious that a web browser cannot
>     open
>      > a raw TCP connection, that's why we need WebSocket as the *standard*
>      > realtime and bidirectional communication mechanism for applications
>      > running in web browsers.
>      >
>      >
>      > > It is not clear to me why websocket is
>      > > required for those real time applications.
>      >
>      > Realtime bidirectional communication from browser to server and vice
>      > versa. What other *standard* mechanism you know for realtime
>      > bidirectional communication in web applications? Please let me know
>      > (reply please).
>      >
>      > Regards.
>      >
>      >
>      >
>      >
>      > --
>      > Iñaki Baz Castillo
>      > <ibc@aliax.net <mailto:ibc@aliax.net>>
>
>
>
>
> --
> Iñaki Baz Castillo
> <ibc@aliax.net <mailto:ibc@aliax.net>>
>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or websocket
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2013/4/3 Paul Kyzivat <pkyzivat@alum.mit.edu>
>
> On 4/4/13 1:27 AM, I=C3=B1aki Baz Castillo wrote:
>>
>> Realtime bidirectional communication from browser to server and vice
>> versa. What other *standard* mechanism you know for realtime
>> bidirectional communication in web applications? Please let me know
>> (reply please).
>
>
> I see a significant difference between "media" streams such as msrp and b=
fcp, and the sip over websocket spec.
>
> For sip over websocket there is little expectation that the target of the=
 websocket is the ultimate destination. It is more likely to be an outbound=
 proxy. So interop with peers that aren't web servers is preserved.
>
> That may be true for msrp too, since a caller-specific msrp relay can be =
used to ensure interop.

Agreed.


> It is more of a problem for bfcp, since AFAIK there is no spec for a bfcp=
 relay. In general you can't know in advance if the peer in your sip call i=
s a web server, so you can't depend on being able to set up a websocket to =
that peer.

It could be just a WebSocket server (without Web component) but it
changes nothing. I agree with your point and, may be what it would
make sense is to (also) define the concept of BFCP relay.


> I am not decided on what is the right answer for bfcp. But I have succeed=
ed in getting the issue discussed more widely.

Using BFCP over DataChannel would not help a lot here since,
obviously, a WebRTC browser/client would not be able to establish a
BFCP session with a non WebRTC client (i.e. a "common" SIP device).
So, again, the concept of a BFCP relay seems needed regardless
WebSocket or DataChannel is defined for transporting BFCP data.

Anyhow, and trying to be realistic: are there *real* use cases in
which BFCP is talked directly between SIP endpoints (instead of "SIP
endpoint <--> SIP conference server")? I expect (and may be I'm wrong)
that the answer is "no" so, in that case, defining BFCP over WebSocket
seems a much more appropriate, easy to implement and realistic
approach (implementing a WebSocket interface in a server is much more
easier than implementing a WebRTC DataChannel client, which involves
SCTP over DTLS over UDP).


Regards.


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

From lorenzo@meetecho.com  Wed Apr  3 14:58:26 2013
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Date: Wed, 3 Apr 2013 23:58:21 +0200
From: Lorenzo Miniero <lorenzo@meetecho.com>
To: =?ISO-8859-1?B?SfFha2k=?= Baz Castillo <ibc@aliax.net>
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Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or websocket
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On Wed, 3 Apr 2013 23:36:31 +0200
I=F1aki Baz Castillo <ibc@aliax.net> wrote:

> 2013/4/3 Paul Kyzivat <pkyzivat@alum.mit.edu>
> >
> > On 4/4/13 1:27 AM, I=F1aki Baz Castillo wrote:
> >>
> >> Realtime bidirectional communication from browser to server and vice
> >> versa. What other *standard* mechanism you know for realtime
> >> bidirectional communication in web applications? Please let me know
> >> (reply please).
> >
> >
> > I see a significant difference between "media" streams such as msrp and=
 bfcp, and the sip over websocket spec.
> >
> > For sip over websocket there is little expectation that the target of t=
he websocket is the ultimate destination. It is more likely to be an outbou=
nd proxy. So interop with peers that aren't web servers is preserved.
> >
> > That may be true for msrp too, since a caller-specific msrp relay can b=
e used to ensure interop.
>=20
> Agreed.
>=20
>=20
> > It is more of a problem for bfcp, since AFAIK there is no spec for a bf=
cp relay. In general you can't know in advance if the peer in your sip call=
 is a web server, so you can't depend on being able to set up a websocket t=
o that peer.
>=20
> It could be just a WebSocket server (without Web component) but it
> changes nothing. I agree with your point and, may be what it would
> make sense is to (also) define the concept of BFCP relay.
>=20
>=20
> > I am not decided on what is the right answer for bfcp. But I have succe=
eded in getting the issue discussed more widely.
>=20
> Using BFCP over DataChannel would not help a lot here since,
> obviously, a WebRTC browser/client would not be able to establish a
> BFCP session with a non WebRTC client (i.e. a "common" SIP device).
> So, again, the concept of a BFCP relay seems needed regardless
> WebSocket or DataChannel is defined for transporting BFCP data.
>=20
> Anyhow, and trying to be realistic: are there *real* use cases in
> which BFCP is talked directly between SIP endpoints (instead of "SIP
> endpoint <--> SIP conference server")? I expect (and may be I'm wrong)
> that the answer is "no" so, in that case, defining BFCP over WebSocket
> seems a much more appropriate, easy to implement and realistic
> approach (implementing a WebSocket interface in a server is much more
> easier than implementing a WebRTC DataChannel client, which involves
> SCTP over DTLS over UDP).
>=20
>=20


I do agree on your realistic view (more on that below), but I also do share=
 Paul's confusion in part... to digress a little, I remember there was some=
 debate at the time when BFCP was being specified, about where a floor cont=
rol server was to be deployed in a conferencing environment: some (includin=
g me) thought it better colocated with the application server (the one hand=
ling the application logic), others with the media server (the one enforcin=
g the decisions). This new discussion partly seems to take the same way, es=
pecially if we rephrase it in slightly a different way: is BFCP more of a "=
signalling" protocol, or a media protocol? The very fact that BFCPBIS is co=
mpleting the works on a UDP transport for BFCP makes this question less eas=
y to answer to.

I still think it's closer to the former as, wherever decisions are taken, t=
here's application logic enforcing something at the media level (which is h=
ow we implemented it as well, by the way). This makes me more keen to view =
a websocket approach being used for BFCP, especially if we assume the floor=
 control server somewhere on the server side and as such not colocated with=
 any of the WebRTC-compliant browsers: this is similar to how we currently =
implement the web based interface to our conferencing platform, where simpl=
e HTTP interactions are used to wrap a BFCP participant that is originated =
server side, and it seems to work fine for our needs. Besides, as you say, =
no change needed on legacy devices as they'd keep on "talking" BFCP with th=
e FCS the old way.

That said, though, I may have this view because of how, just as you, I'm cu=
rrently associating BFCP with the conferencing world (well, it was born in =
XCON after all), which may be a wrong assumption: people may be using BFCP =
for entirely different scenarios as well, and the example made by someone a=
 few posts ago (I think it was Victor but I may be wrong) about one of two =
peers acting as the floor control server in a call may be a perfectly valid=
 scenario for it being used differently than what I'm used to. I also remem=
ber Adam Roach proposing a Call Completion service based on BFCP a few year=
s ago, which is just another scenario that may change the roles a bit.

Such a shift in the roles may very well motivate a use of datachannels for =
BFCP as well, especially if we think of scenarios where no centralized cont=
rol will be needed, or at least no centralized control outside of the WebRT=
C peers that are part of the conversation. Of course, if an application DOE=
S need some control on the server side, this means that this server would n=
eed to implement the backend of a datachannel; and yes, it would make thing=
s definitelt harder when interoperating with legacy devices, but as some se=
em to advocate, WebRTC should be a fresh start, and should not look back to=
o much. Whether this would be a dealbreaker for some I don't know, but I gu=
ess that, as there already are non-browser WebRTC implementations for media=
 out there, it won't be long before datachannels will be handled as well: m=
aybe it won't be optimal, but someone will make it work one way or the othe=
r.

Lorenzo


> Regards.
>=20
>=20
> --
> I=F1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


--=20
Lorenzo Miniero, COB

Meetecho s.r.l.
Web Conferencing and Collaboration Tools
http://www.meetecho.com

From stpeter@stpeter.im  Wed Apr  3 18:56:05 2013
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

RFC 4575 (Section 5.6.3) defines the concept of a role in a
conference, but does not specify or define any roles.

5.6.3. <roles>

   This element MAY contain a set of human-readable strings describing
   the roles of the user in the conference.  Note that this information
   is applicable for human consumption only.  This specification does
   not define the set of possible conferencing roles or the semantics
   associated with each.  It is expected that future conferencing
   specifications will define these and the corresponding schema
   extensions, as appropriate.

RFC 6501 (Section 4.6.5.2) specifies a few values for the role
element, but does not actually define what those values mean.

4.6.5.2. <roles>

   A <role> provides the context for the set of conference operations
   that a participant can perform.  This element can contain one or more
   of the following values: "administrator", "moderator", "user",
   "participant", "observer", and "none".  A role of "none" indicates
   that any role is assigned.  The <roles> semantic definition is out of
   the scope of this document and is subject to future policy documents.
   This element can be extended with new roles in future documents.

(As far as I know, those "future policy documents" have not been
published.)

More recently, draft-groves-clue-capture-attr specifies several
additional roles: "manager", "chairman", "secretary", "lecturer", and
"audience" (so-called person roles) as well as "speaker" and
"controller" (so-called conference roles). Personally, I happen to
have an interest in defining a few other roles, such as "host",
"presenter", "scribe", and "panelist".

Unfortunately there's quite a bit of overlap and confusion here. For
instance, one could argue that "host", "manager", and "controller"
(perhaps even "chairman") are the same as or very similar to
"administrator" or "moderator" from RFC 6501. There might also be
overlap between "lecturer", "speaker", and "presenter" (and between
"speaker" and "panelist").

Furthermore, I'm concerned that various implementations will define
their own roles, blithely unaware that very similar (or even identical
but differently named roles) are already in use by other implementations.

It strikes me that one solution here would be to establish an IANA
registry for conference roles, with a registration policy (cf. RFC
5226) of at least Expert Review and perhaps Specification Required (I
think RFC Required is a bit heavy). IMHO this would help to prevent
continued confusion and the minting of new roles that overlap with
existing roles.

Thoughts?

Peter

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

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Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or websocket
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On Wed, Apr 3, 2013 at 11:58 PM, Lorenzo Miniero <lorenzo@meetecho.com> wro=
te:
> That said, though, I may have this view because of how, just as you, I'm =
currently associating BFCP with the conferencing world (well, it was born i=
n XCON after all), which may be a wrong assumption: people may be using BFC=
P for entirely different scenarios as well, and the example made by someone=
 a few posts ago (I think it was Victor but I may be wrong) about one of tw=
o peers acting as the floor control server in a call may be a perfectly val=
id scenario for it being used differently than what I'm used to.

It was not me -- all my use cases include a conf server. On the client
side, we have WebRTC and non-WebRTC web endpoints (all of them
supporting WebSockets already) as well as "legacy" BFCP devices.

Cheers,
-Victor

From lorenzo@meetecho.com  Wed Apr  3 23:18:34 2013
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From: Lorenzo Miniero <lorenzo@meetecho.com>
To: Victor Pascual Avila <victor.pascual.avila@gmail.com>
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Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or websocket
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On Thu, 4 Apr 2013 07:40:03 +0200
Victor Pascual Avila <victor.pascual.avila@gmail.com> wrote:

> On Wed, Apr 3, 2013 at 11:58 PM, Lorenzo Miniero <lorenzo@meetecho.com> wrote:
> > That said, though, I may have this view because of how, just as you, I'm currently associating BFCP with the conferencing world (well, it was born in XCON after all), which may be a wrong assumption: people may be using BFCP for entirely different scenarios as well, and the example made by someone a few posts ago (I think it was Victor but I may be wrong) about one of two peers acting as the floor control server in a call may be a perfectly valid scenario for it being used differently than what I'm used to.
> 
> It was not me -- all my use cases include a conf server. On the client
> side, we have WebRTC and non-WebRTC web endpoints (all of them
> supporting WebSockets already) as well as "legacy" BFCP devices.
> 
> Cheers,
> -Victor


Ops you're right, sorry about that! As they say here in Napoli, "that's how you send people to jail" :-)

I just checked the archives and it was Charles Eckel that referred to the possibility in a couple of posts.

Cheers,
Lorenzo

-- 
Lorenzo Miniero, COB

Meetecho s.r.l.
Web Conferencing and Collaboration Tools
http://www.meetecho.com

From gonzalo.camarillo@ericsson.com  Thu Apr  4 00:07:23 2013
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: [dispatch] Fwd: [RAI] Call for volunteers to chair WGs
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Hi,

this email (see below) is also relevant to DISPATCH.

Cheers,

Gonzalo


-------- Original Message --------
Subject: [RAI] Call for volunteers to chair WGs
Date: Thu, 4 Apr 2013 10:03:24 +0300
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
To: <rai@ietf.org>

Folks,

as you know, every now and then we need to fill chair positions in both
new and existing WGs. If you could be interested in chairing a WG in the
RAI area at some point, please send us and email.

We understand people are typically interested in a specific WG. It is
also typical to get management approval to chair a particular WG.
Nevertheless, knowing that you may be interested in a chair position and
which areas you are mostly working on would be useful to us.

Additionally, every time we agree to charter a new WG (in DISPATCH or in
a BOF) you are interested in chairing, you should let us know as well.

Thanks,

Gonzalo

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





From sergio.garcia.murillo@gmail.com  Thu Apr  4 02:04:34 2013
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Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or websocket
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El 04/04/2013 7:40, Victor Pascual Avila escribió:
> On Wed, Apr 3, 2013 at 11:58 PM, Lorenzo Miniero <lorenzo@meetecho.com> wrote:
>> That said, though, I may have this view because of how, just as you, I'm currently associating BFCP with the conferencing world (well, it was born in XCON after all), which may be a wrong assumption: people may be using BFCP for entirely different scenarios as well, and the example made by someone a few posts ago (I think it was Victor but I may be wrong) about one of two peers acting as the floor control server in a call may be a perfectly valid scenario for it being used differently than what I'm used to.
> It was not me -- all my use cases include a conf server. On the client
> side, we have WebRTC and non-WebRTC web endpoints (all of them
> supporting WebSockets already) as well as "legacy" BFCP devices.
>
>

Also, there will be use cases for BFCP endpoints not involved in the 
media flow at all.

Best regards
Sergio

From oej@edvina.net  Thu Apr  4 13:13:12 2013
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Subject: [dispatch] SIP SNMP MIB and new transports
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Hi!

RFC 4780 defines an OID for transports. If we add new transports =
(websockets) we need to expand this table. The RFC in itself doesn't =
outline a procedure for extending it.=20

Should the draft for websockets transport extend the MIB?

In the Kamailio SNMP implementation, I have used bit 6 for ws and 7 for =
wss.

/O

------

SipTCTransportProtocol ::=3D TEXTUAL-CONVENTION
    STATUS      current
    DESCRIPTION
       "This convention is a bit map.  Each bit represents a transport
        protocol.  If a bit has value 1, then that selected transport
        protocol is in some way dependent on the context of the object
        using this convention.  If a bit has value 0, then that
        transport protocol is not selected.  Combinations of bits can
        be set when multiple transport protocols are selected.

        bit 0: a protocol other than those defined here
        bit 1: User Datagram Protocol
        bit 2: Transmission Control Protocol
        bit 3: Stream Control Transmission Protocol
        bit 4: Transport Layer Security Protocol over TCP
        bit 5: Transport Layer Security Protocol over SCTP
       "
    REFERENCE "
RFC 3261, Section 18 and RFC 4168
"
    SYNTAX      BITS {
                  other(0),  -- none of the following
                  udp(1),
                  tcp(2),
                  sctp(3),   --=20
RFC4168

                  tlsTcp(4),
                  tlsSctp(5) --=20
RFC 4168

                }=

From stpeter@stpeter.im  Thu Apr  4 19:38:25 2013
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Subject: [dispatch] Fwd: I-D Action: draft-saintandre-sip-xmpp-im-03.txt
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FYI.


-------- Original Message --------
Subject: I-D Action: draft-saintandre-sip-xmpp-im-03.txt
Date: Thu, 04 Apr 2013 19:37:18 -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): Instant
Messaging
	Author(s)       : Peter Saint-Andre
                          Avshalom Houri
                          Joe Hildebrand
	Filename        : draft-saintandre-sip-xmpp-im-03.txt
	Pages           : 10
	Date            : 2013-04-04

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


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-saintandre-sip-xmpp-im-03

A diff from the previous version is available at:
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FYI.


-------- Original Message --------
Subject: I-D Action: draft-saintandre-sip-xmpp-chat-05.txt
Date: Thu, 04 Apr 2013 20:09:11 -0700
From: internet-drafts@ietf.org
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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
                          Eddy Gavita
                          Nazin Hossain
                          Salvatore Loreto
	Filename        : draft-saintandre-sip-xmpp-chat-05.txt
	Pages           : 12
	Date            : 2013-04-04

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-05

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


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FYI.


-------- Original Message --------
Subject: I-D Action: draft-ivov-xmpp-cusax-04.txt
Date: Thu, 04 Apr 2013 20:16:30 -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           : CUSAX: Combined Use of the Session Initiation
Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP)
	Author(s)       : Emil Ivov
                          Peter Saint-Andre
                          Enrico Marocco
	Filename        : draft-ivov-xmpp-cusax-04.txt
	Pages           : 11
	Date            : 2013-04-04

Abstract:
   This document describes suggested practices for combined use of the
   Session Initiation Protocol (SIP) and the Extensible Messaging and
   Presence Protocol (XMPP).  Such practices aim to provide a single
   fully featured real-time communication service by using complementary
   subsets of features from each of the protocols.  Typically such
   subsets would include telephony capabilities from SIP and instant
   messaging and presence capabilities from XMPP.  This specification
   does not define any new protocols or syntax for either SIP or XMPP.
   However, implementing it may require modifying or at least
   reconfiguring existing client and server-side software.  Also, it is
   not the purpose of this document to make recommendations as to
   whether or not such combined use should be preferred to the
   mechanisms provided natively by each protocol (for example, SIP's
   SIMPLE or XMPP's Jingle).  It merely aims to provide guidance to
   those who are interested in such a combined use.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ivov-xmpp-cusax

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

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


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From gonzalo.camarillo@ericsson.com  Thu Apr  4 23:13:11 2013
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: [dispatch] Acronym for the SIP-XMPP effort
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Folks,

I am going to handle the chartering of the SIP-XMPP effort, as discussed
in Orlando. As usual, we need to choose an acronym for the group. Among
the acronyms I have heard, I like this one best so far:

  SIXPAC - SIP Integration with XMPP in Presence Aware Contexts

While, per the RAI tradition, it refers to an alcoholic drink, usually
beers, it also refers to the nice six packs all RAI contributors are
working on during the spring in order to show off at the beach in summer :-)

Any other suggestions?

With respect to the chairs for this effort, we will be choosing them
soon as well. So, if you are interested but have not volunteered yet per
my other email (from yesterday), now it would be a great time to drop me
a note.

Thanks,

Gonzalo

From br@brianrosen.net  Fri Apr  5 06:20:37 2013
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Cc: DISPATCH <dispatch@ietf.org>
Subject: Re: [dispatch] conference roles
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> It strikes me that one solution here would be to establish an IANA
> registry for conference roles, with a registration policy (cf. RFC
> 5226) of at least Expert Review and perhaps Specification Required (I
> think RFC Required is a bit heavy). IMHO this would help to prevent
> continued confusion and the minting of new roles that overlap with
> existing roles.
I like this idea.

Brian

From salvatore.loreto@ericsson.com  Fri Apr  5 06:55:18 2013
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Subject: Re: [dispatch] conference roles
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On 4/4/13 4:56 AM, Peter Saint-Andre wrote:
> Furthermore, I'm concerned that various implementations will define
> their own roles, blithely unaware that very similar (or even identical
> but differently named roles) are already in use by other implementations.
>
> It strikes me that one solution here would be to establish an IANA
> registry for conference roles, with a registration policy (cf. RFC
> 5226) of at least Expert Review and perhaps Specification Required (I
> think RFC Required is a bit heavy). IMHO this would help to prevent
> continued confusion and the minting of new roles that overlap with
> existing roles.
>
> Thoughts?
I do think is a good idea to establish a IANA registry for conference roles
and I concur on the fact that RFC Required is a bit too much (and 
unnecessary)

cheers
Salvatore


From pkyzivat@alum.mit.edu  Fri Apr  5 08:25:02 2013
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Date: Fri, 05 Apr 2013 23:25:39 +0800
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [dispatch] SIP SNMP MIB and new transports
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Olle,

On 4/5/13 4:12 AM, Olle E. Johansson wrote:
> Hi!
>
> RFC 4780 defines an OID for transports. If we add new transports (websockets) we need to expand this table. The RFC in itself doesn't outline a procedure for extending it.
>
> Should the draft for websockets transport extend the MIB?

It is too bad this wasn't raised sooner, e.g. during WGLC.
draft-ietf-sipcore-sip-websocket-08 is now with the editor,
so it is really too late to address this in that draft now.

So if something needs to be done about this then I guess it will require 
a separate draft.

For the moment, ISTM the obvious work around is:

   bit 0: a protocol other than those defined here

> In the Kamailio SNMP implementation, I have used bit 6 for ws and 7 for wss.

I know *nothing* about mibs, but when looking at 4780 I don't see any 
evidence that it was designed with a mechanism for extension as new 
transports are defined. Is there some way to fix that, and link it to a 
new IANA registry?

	Thanks,
	Paul

> /O
>
> ------
>
> SipTCTransportProtocol ::= TEXTUAL-CONVENTION
>      STATUS      current
>      DESCRIPTION
>         "This convention is a bit map.  Each bit represents a transport
>          protocol.  If a bit has value 1, then that selected transport
>          protocol is in some way dependent on the context of the object
>          using this convention.  If a bit has value 0, then that
>          transport protocol is not selected.  Combinations of bits can
>          be set when multiple transport protocols are selected.
>
>          bit 0: a protocol other than those defined here
>          bit 1: User Datagram Protocol
>          bit 2: Transmission Control Protocol
>          bit 3: Stream Control Transmission Protocol
>          bit 4: Transport Layer Security Protocol over TCP
>          bit 5: Transport Layer Security Protocol over SCTP
>         "
>      REFERENCE "
> RFC 3261, Section 18 and RFC 4168
> "
>      SYNTAX      BITS {
>                    other(0),  -- none of the following
>                    udp(1),
>                    tcp(2),
>                    sctp(3),   --
> RFC4168
>
>                    tlsTcp(4),
>                    tlsSctp(5) --
> RFC 4168
>
>                  }
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>


From stpeter@stpeter.im  Fri Apr  5 08:30:52 2013
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From: Peter Saint-Andre <stpeter@stpeter.im>
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On 4/5/13 7:57 AM, Salvatore Loreto wrote:
> On 4/4/13 4:56 AM, Peter Saint-Andre wrote:
>> Furthermore, I'm concerned that various implementations will define
>> their own roles, blithely unaware that very similar (or even identical
>> but differently named roles) are already in use by other implementations.
>>
>> It strikes me that one solution here would be to establish an IANA
>> registry for conference roles, with a registration policy (cf. RFC
>> 5226) of at least Expert Review and perhaps Specification Required (I
>> think RFC Required is a bit heavy). IMHO this would help to prevent
>> continued confusion and the minting of new roles that overlap with
>> existing roles.
>>
>> Thoughts?
> I do think is a good idea to establish a IANA registry for conference roles
> and I concur on the fact that RFC Required is a bit too much (and
> unnecessary)

OK. I might write a brief I-D about this sometime soon.

Peter

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



From pkyzivat@alum.mit.edu  Fri Apr  5 08:31:04 2013
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Date: Fri, 05 Apr 2013 23:31:42 +0800
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Subject: Re: [dispatch] conference roles
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This also seems reasonable to me. But I would be conservative about 
policy for new ones. I don't think we know enough to describe criteria 
for an expert, and I expect there to be a lot of bad ideas to come up 
here. So I think I would go for Standards Action for now, to ensure 
strong debate.

Even then, I think we need to think hard about definitions for the first 
batch, that clearly distinguish one role from another.

	Thanks,
	Paul

On 4/5/13 9:20 PM, Brian Rosen wrote:
>> It strikes me that one solution here would be to establish an IANA
>> registry for conference roles, with a registration policy (cf. RFC
>> 5226) of at least Expert Review and perhaps Specification Required (I
>> think RFC Required is a bit heavy). IMHO this would help to prevent
>> continued confusion and the minting of new roles that overlap with
>> existing roles.
> I like this idea.
>
> Brian
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>


From stpeter@stpeter.im  Fri Apr  5 08:37:38 2013
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On 4/5/13 9:31 AM, Paul Kyzivat wrote:
> This also seems reasonable to me. But I would be conservative about
> policy for new ones. I don't think we know enough to describe criteria
> for an expert, and I expect there to be a lot of bad ideas to come up
> here. So I think I would go for Standards Action for now, to ensure
> strong debate.

You make a good point about being able to provide clear guidelines to an
expert. We either need to figure that out during the process of defining
this IANA registry or use the standards process to become clear on each
registration request as it comes in.

> Even then, I think we need to think hard about definitions for the first
> batch, that clearly distinguish one role from another.

Agreed.

Peter

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



From jmpolk@cisco.com  Fri Apr  5 12:35:57 2013
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Subject: Re: [dispatch] Acronym for the SIP-XMPP effort
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At 01:13 AM 4/5/2013, Gonzalo Camarillo wrote:
>I am going to handle the chartering of the SIP-XMPP effort, as discussed
>in Orlando. As usual, we need to choose an acronym for the group. Among
>the acronyms I have heard, I like this one best so far:
>
>   SIXPAC - SIP Integration with XMPP in Presence Aware Contexts

wfm

James



From michaellundberg.ietf@gmail.com  Fri Apr  5 13:13:36 2013
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Subject: Re: [dispatch] Comments on draft-saintandre-sip-xmpp-presence-04
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Peter,

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

Regards,
Michael

On Wed, Apr 3, 2013 at 12:07 AM, Peter Saint-Andre <stpeter@stpeter.im> wro=
te:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Michael, I think your feedback has been addressed in -05. Please check.
>
> On 3/9/13 5:29 PM, Michael Lundberg wrote:
>> Peter,
>>
>> Thanks for resubmitting the SIP-XMPP presence I-D.  I think this
>> and the other documents are important as there are a lot of
>> interworking issues today with XMPP and SIP/SIMPLE implementations,
>> and I think these documents can help reduce some of those
>> interworking challenges.
>>
>> In order to help ensure interworking between different
>> implementations, I think Sections 5.2 and 5.3 need to have some
>> stronger =91MUST=92 language instead of SHOULDs, MAYs, etc.  This will
>> enforce proper and complete mappings between implementations that
>> conform to this document.
>>
>> While I don=92t think the document needs to enforce the use of the
>> <show/> element mapping discussed in 5.2, I think if
>> implementations are going to use the mapping, this document should
>> enforce the use of a specific namespace and describe the mapping of
>> <show> values specified in RFC 6121 (i.e., away, dnd, chat, and
>> xa).  This will at least allow a minimum set of standard mappings
>> between XMPP and SIP, which can be expanded on in future
>> document(s).  Section 5.3 could then also discuss how this mapping
>> can be accomplished from SIP to XMPP if the SIP endpoints/devices
>> support this same namespace.  One of the big interworking
>> challenges with today=92s implementations is the use of different
>> namespaces given the extensibility of XML-based presence standards
>> (e.g., PIDF, RPID), which allows individuals to create their own
>> namespace.  This flexibility is good for ingenuity, but problematic
>> for interworking between different implementations.  I think
>> standardizing a basic set of common status information using a
>> specific namespace is needed, and this document is a good place to
>> do that.
>>
>> In my opinion, I do not think Section 6 should be part of this
>> base document.  While I like the idea of encapsulating the SIP
>> <presence> information into the XMPP message, I think this content
>> should be a consideration for implementing =91rich presence=92.  As
>> written, this method is only applicable for SIP to XMPP
>> translation, and if the focus of this document is just =91basic=92
>> presence information interworking, the SIP to XMPP mapping in
>> Section 5.3 already describes a method for how this can be
>> accomplished.
>>
>> Regards, Michael _______________________________________________
>> dispatch mailing list dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
>
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From ron.even.tlv@gmail.com  Fri Apr  5 13:46:30 2013
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Peter Saint-Andre'" <stpeter@stpeter.im>, "'DISPATCH'" <dispatch@ietf.org>
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Date: Fri, 5 Apr 2013 23:46:00 +0300
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Subject: Re: [dispatch] conference roles
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Hi,
I think that we first need to understand the scope of the role. 
RFC 4575 talks about the role of the user. This is the conference
participant.
There is also RFC4796 that defines the content attribute which talks about
the content of an RTP stream.
In CLUE we are talking about a capture which is a specific stream again and
not about the participant role.
So if we proceed we first need to specify what we mean by role.

Roni Even.

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
Of Peter Saint-Andre
Sent: 04 April, 2013 4:56 AM
To: DISPATCH
Subject: [dispatch] conference roles

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

RFC 4575 (Section 5.6.3) defines the concept of a role in a conference, but
does not specify or define any roles.

5.6.3. <roles>

   This element MAY contain a set of human-readable strings describing
   the roles of the user in the conference.  Note that this information
   is applicable for human consumption only.  This specification does
   not define the set of possible conferencing roles or the semantics
   associated with each.  It is expected that future conferencing
   specifications will define these and the corresponding schema
   extensions, as appropriate.

RFC 6501 (Section 4.6.5.2) specifies a few values for the role element, but
does not actually define what those values mean.

4.6.5.2. <roles>

   A <role> provides the context for the set of conference operations
   that a participant can perform.  This element can contain one or more
   of the following values: "administrator", "moderator", "user",
   "participant", "observer", and "none".  A role of "none" indicates
   that any role is assigned.  The <roles> semantic definition is out of
   the scope of this document and is subject to future policy documents.
   This element can be extended with new roles in future documents.

(As far as I know, those "future policy documents" have not been
published.)

More recently, draft-groves-clue-capture-attr specifies several additional
roles: "manager", "chairman", "secretary", "lecturer", and "audience"
(so-called person roles) as well as "speaker" and "controller" (so-called
conference roles). Personally, I happen to have an interest in defining a
few other roles, such as "host", "presenter", "scribe", and "panelist".

Unfortunately there's quite a bit of overlap and confusion here. For
instance, one could argue that "host", "manager", and "controller"
(perhaps even "chairman") are the same as or very similar to "administrator"
or "moderator" from RFC 6501. There might also be overlap between
"lecturer", "speaker", and "presenter" (and between "speaker" and
"panelist").

Furthermore, I'm concerned that various implementations will define their
own roles, blithely unaware that very similar (or even identical but
differently named roles) are already in use by other implementations.

It strikes me that one solution here would be to establish an IANA registry
for conference roles, with a registration policy (cf. RFC
5226) of at least Expert Review and perhaps Specification Required (I think
RFC Required is a bit heavy). IMHO this would help to prevent continued
confusion and the minting of new roles that overlap with existing roles.

Thoughts?

Peter

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

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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Lorenzo Miniero <lorenzo@meetecho.com>, Victor Pascual Avila <victor.pascual.avila@gmail.com>
Thread-Topic: [dispatch] Use of BFCP or MSRP over datachannel or websocket
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Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or websocket
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> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Lorenzo Miniero
> Sent: Wednesday, April 03, 2013 11:19 PM
> To: Victor Pascual Avila
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or websocket
>=20
> On Thu, 4 Apr 2013 07:40:03 +0200
> Victor Pascual Avila <victor.pascual.avila@gmail.com> wrote:
>=20
> > On Wed, Apr 3, 2013 at 11:58 PM, Lorenzo Miniero
> <lorenzo@meetecho.com> wrote:
> > > That said, though, I may have this view because of how, just as you, =
I'm
> currently associating BFCP with the conferencing world (well, it was born=
 in
> XCON after all), which may be a wrong assumption: people may be using
> BFCP for entirely different scenarios as well, and the example made by
> someone a few posts ago (I think it was Victor but I may be wrong) about
> one of two peers acting as the floor control server in a call may be a
> perfectly valid scenario for it being used differently than what I'm used=
 to.
> >
> > It was not me -- all my use cases include a conf server. On the client
> > side, we have WebRTC and non-WebRTC web endpoints (all of them
> > supporting WebSockets already) as well as "legacy" BFCP devices.
> >
> > Cheers,
> > -Victor
>=20
>=20
> Ops you're right, sorry about that! As they say here in Napoli, "that's h=
ow
> you send people to jail" :-)
>=20
> I just checked the archives and it was Charles Eckel that referred to the
> possibility in a couple of posts.

Yes, and I still stand by BFCP being useful and implemented for point to po=
int and peer to peer use cases, not just for conference calls between a cli=
ent and a conference server. If you want additional details, please have a =
look at the following IMTC Best Practice shared with the IETF through a lia=
ison statement last year:
https://datatracker.ietf.org/documents/LIAISON/liaison-2012-05-31-imtc-the-=
ietf-imtc-work-on-sip-feature-parity-with-h323-attachment-3.pdf

That said, I am not saying BFCP over websockets is a bad idea. Rather, I am=
 saying it is not sufficient for all the use cases for which BFCP is used t=
oday. Perhaps that is okay, and we just need to define something else for t=
he other use cases, and applications need to know when to use one method as=
 opposed to another. This adds complexity other than that of implementing B=
FCP over websockets.

Cheers,
Charles
=20
> Cheers,
> Lorenzo
>=20
> --
> Lorenzo Miniero, COB
>=20
> Meetecho s.r.l.
> Web Conferencing and Collaboration Tools
> http://www.meetecho.com
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

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From: worley@ariadne.com (Dale R. Worley)
Sender: worley@ariadne.com (Dale R. Worley)
To: Peter Saint-Andre <stpeter@stpeter.im>
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Subject: Re: [dispatch] conference roles
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> From: Peter Saint-Andre <stpeter@stpeter.im>
> 
> It strikes me that one solution here would be to establish an IANA
> registry for conference roles, with a registration policy (cf. RFC
> 5226) of at least Expert Review and perhaps Specification Required (I
> think RFC Required is a bit heavy). IMHO this would help to prevent
> continued confusion and the minting of new roles that overlap with
> existing roles.
> 
> Thoughts?

My thought is that more work than this is needed.  We may not need an
RFC to review and define a good set of role names, but I think we need
to start with an I-D that discusses all the issues and desiderata of
how to define role names, and establishes an initial set.  Only that
sort of writing, discussion, and analysis will solidify knowledge of
what how such a registry should operate.

Dale

From spromano@unina.it  Fri Apr  5 15:40:25 2013
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From: Simon Pietro Romano <spromano@unina.it>
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FYI.

Simon

Inizio messaggio inoltrato:

> Da: Simon Pietro Romano <spromano@unina.it>
> Oggetto: Re: [clue] [dispatch] conference roles
> Data: 06 aprile 2013 00:39:19 GMT+02:00
> A: Peter Saint-Andre <stpeter@stpeter.im>
> Cc: clue@ietf.org
>=20
> Hi Peter,
>=20
> thank you for raising such an interesting discussion. This reminds me =
of a specific thread dedicated to policies and roles inside XCON, back =
in 2007. I expressed my personal view on this subject in a (long) e-mail =
(http://www.ietf.org/mail-archive/web/xcon/current/msg01919.html) you =
might want to have a look at. As you mention in your e-mail "future =
policy documents" have not been published.
>=20
> Cheers,
>=20
> Simon
>=20
>=20
> Il giorno 04/apr/2013, alle ore 16:42, Peter Saint-Andre ha scritto:
>=20
>> FYI.
>>=20
>>=20
>> -------- Original Message --------
>> Subject: [dispatch] conference roles
>> Date: Wed, 03 Apr 2013 19:56:01 -0600
>> From: Peter Saint-Andre <stpeter@stpeter.im>
>> To: DISPATCH <dispatch@ietf.org>
>>=20
>> RFC 4575 (Section 5.6.3) defines the concept of a role in a
>> conference, but does not specify or define any roles.
>>=20
>> 5.6.3. <roles>
>>=20
>>   This element MAY contain a set of human-readable strings describing
>>   the roles of the user in the conference.  Note that this =
information
>>   is applicable for human consumption only.  This specification does
>>   not define the set of possible conferencing roles or the semantics
>>   associated with each.  It is expected that future conferencing
>>   specifications will define these and the corresponding schema
>>   extensions, as appropriate.
>>=20
>> RFC 6501 (Section 4.6.5.2) specifies a few values for the role
>> element, but does not actually define what those values mean.
>>=20
>> 4.6.5.2. <roles>
>>=20
>>   A <role> provides the context for the set of conference operations
>>   that a participant can perform.  This element can contain one or =
more
>>   of the following values: "administrator", "moderator", "user",
>>   "participant", "observer", and "none".  A role of "none" indicates
>>   that any role is assigned.  The <roles> semantic definition is out =
of
>>   the scope of this document and is subject to future policy =
documents.
>>   This element can be extended with new roles in future documents.
>>=20
>> (As far as I know, those "future policy documents" have not been
>> published.)
>>=20
>> More recently, draft-groves-clue-capture-attr specifies several
>> additional roles: "manager", "chairman", "secretary", "lecturer", and
>> "audience" (so-called person roles) as well as "speaker" and
>> "controller" (so-called conference roles). Personally, I happen to
>> have an interest in defining a few other roles, such as "host",
>> "presenter", "scribe", and "panelist".
>>=20
>> Unfortunately there's quite a bit of overlap and confusion here. For
>> instance, one could argue that "host", "manager", and "controller"
>> (perhaps even "chairman") are the same as or very similar to
>> "administrator" or "moderator" from RFC 6501. There might also be
>> overlap between "lecturer", "speaker", and "presenter" (and between
>> "speaker" and "panelist").
>>=20
>> Furthermore, I'm concerned that various implementations will define
>> their own roles, blithely unaware that very similar (or even =
identical
>> but differently named roles) are already in use by other =
implementations.
>>=20
>> It strikes me that one solution here would be to establish an IANA
>> registry for conference roles, with a registration policy (cf. RFC
>> 5226) of at least Expert Review and perhaps Specification Required (I
>> think RFC Required is a bit heavy). IMHO this would help to prevent
>> continued confusion and the minting of new roles that overlap with
>> existing roles.
>>=20
>> Thoughts?
>>=20
>> Peter
>>=20
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>=20
>>=20
>> _______________________________________________
>> clue mailing list
>> clue@ietf.org
>> https://www.ietf.org/mailman/listinfo/clue
>>=20
>=20
>                      					       _\\|//_
>                            				      ( O-O )
>    ~~~~~~~~~~~~~~~~~~~~~~o00~~(_)~~00o~~~~~~~~~~~~~~~~~~~~~~~~
>                     				Simon Pietro Romano
>              				 Universita' di Napoli Federico =
II
>                 		     Computer Engineering Department=20
> 	             Phone: +39 081 7683823 -- Fax: +39 081 7683816
>                                            e-mail: spromano@unina.it
>=20
> 		    <<Molti mi dicono che lo scoraggiamento =CB l'alibi =
degli=20
> 		    idioti. Ci rifletto un istante; e mi scoraggio>>. =
Magritte.
>                			                     oooO
>   ~~~~~~~~~~~~~~~~~~~~~~~(   )~~~ Oooo~~~~~~~~~~~~~~~~~~~~~~~~~
> 					                 \ (            =
(   )
> 			                                  \_)          ) =
/
>                                                                        =
(_/
>=20
>=20
>=20
>=20
>=20

                     					       _\\|//_
                           				      ( O-O )
   ~~~~~~~~~~~~~~~~~~~~~~o00~~(_)~~00o~~~~~~~~~~~~~~~~~~~~~~~~
                    				Simon Pietro Romano
             				 Universita' di Napoli Federico =
II
                		     Computer Engineering Department=20
	             Phone: +39 081 7683823 -- Fax: +39 081 7683816
                                           e-mail: spromano@unina.it

		    <<Molti mi dicono che lo scoraggiamento =CB l'alibi =
degli=20
		    idioti. Ci rifletto un istante; e mi scoraggio>>. =
Magritte.
               			                     oooO
  ~~~~~~~~~~~~~~~~~~~~~~~(   )~~~ Oooo~~~~~~~~~~~~~~~~~~~~~~~~~
					                 \ (            =
(   )
			                                  \_)          ) =
/
                                                                       =
(_/






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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
">FYI.<div><br></div><div>Simon<br><div><br><div>Inizio messaggio =
inoltrato:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px;"><span =
style=3D"font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, 0, =
1.0);"><b>Da: </b></span><span style=3D"font-family:'Helvetica'; =
font-size:medium;">Simon Pietro Romano &lt;<a =
href=3D"mailto:spromano@unina.it">spromano@unina.it</a>&gt;<br></span></di=
v><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px;"><span style=3D"font-family:'Helvetica'; =
font-size:medium; color:rgba(0, 0, 0, 1.0);"><b>Oggetto: =
</b></span><span style=3D"font-family:'Helvetica'; =
font-size:medium;"><b>Re: [clue] [dispatch] conference =
roles</b><br></span></div><div style=3D"margin-top: 0px; margin-right: =
0px; margin-bottom: 0px; margin-left: 0px;"><span =
style=3D"font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, 0, =
1.0);"><b>Data: </b></span><span style=3D"font-family:'Helvetica'; =
font-size:medium;">06 aprile 2013 00:39:19 =
GMT+02:00<br></span></div><div style=3D"margin-top: 0px; margin-right: =
0px; margin-bottom: 0px; margin-left: 0px;"><span =
style=3D"font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, 0, =
1.0);"><b>A: </b></span><span style=3D"font-family:'Helvetica'; =
font-size:medium;">Peter Saint-Andre &lt;<a =
href=3D"mailto:stpeter@stpeter.im">stpeter@stpeter.im</a>&gt;<br></span></=
div><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: =
0px; margin-left: 0px;"><span style=3D"font-family:'Helvetica'; =
font-size:medium; color:rgba(0, 0, 0, 1.0);"><b>Cc: </b></span><span =
style=3D"font-family:'Helvetica'; font-size:medium;"><a =
href=3D"mailto:clue@ietf.org">clue@ietf.org</a><br></span></div><br><div =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; ">Hi =
Peter,<div><br></div><div>thank you for raising such an interesting =
discussion. This reminds me of a specific thread dedicated to policies =
and roles inside XCON, back in 2007. I expressed my personal view on =
this subject in a (long) e-mail (<a =
href=3D"http://www.ietf.org/mail-archive/web/xcon/current/msg01919.html">h=
ttp://www.ietf.org/mail-archive/web/xcon/current/msg01919.html</a>)&nbsp;y=
ou might want to have a look at. As you mention in your e-mail "future =
policy documents" have not been =
published.</div><div><br></div><div>Cheers,</div><div><br></div><div>Simon=
</div><div><br></div><div><br><div><div>Il giorno 04/apr/2013, alle ore =
16:42, Peter Saint-Andre ha scritto:</div><br =
class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><div>FYI.<br><br><br>-------- Original Message =
--------<br>Subject: [dispatch] conference roles<br>Date: Wed, 03 Apr =
2013 19:56:01 -0600<br>From: Peter Saint-Andre &lt;<a =
href=3D"mailto:stpeter@stpeter.im">stpeter@stpeter.im</a>&gt;<br>To: =
DISPATCH &lt;<a =
href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a>&gt;<br><br>RFC =
4575 (Section 5.6.3) defines the concept of a role in a<br>conference, =
but does not specify or define any roles.<br><br>5.6.3. =
&lt;roles&gt;<br><br> &nbsp;&nbsp;This element MAY contain a set of =
human-readable strings describing<br> &nbsp;&nbsp;the roles of the user =
in the conference. &nbsp;Note that this information<br> &nbsp;&nbsp;is =
applicable for human consumption only. &nbsp;This specification does<br> =
&nbsp;&nbsp;not define the set of possible conferencing roles or the =
semantics<br> &nbsp;&nbsp;associated with each. &nbsp;It is expected =
that future conferencing<br> &nbsp;&nbsp;specifications will define =
these and the corresponding schema<br> &nbsp;&nbsp;extensions, as =
appropriate.<br><br>RFC 6501 (Section 4.6.5.2) specifies a few values =
for the role<br>element, but does not actually define what those values =
mean.<br><br>4.6.5.2. &lt;roles&gt;<br><br> &nbsp;&nbsp;A &lt;role&gt; =
provides the context for the set of conference operations<br> =
&nbsp;&nbsp;that a participant can perform. &nbsp;This element can =
contain one or more<br> &nbsp;&nbsp;of the following values: =
"administrator", "moderator", "user",<br> &nbsp;&nbsp;"participant", =
"observer", and "none". &nbsp;A role of "none" indicates<br> =
&nbsp;&nbsp;that any role is assigned. &nbsp;The &lt;roles&gt; semantic =
definition is out of<br> &nbsp;&nbsp;the scope of this document and is =
subject to future policy documents.<br> &nbsp;&nbsp;This element can be =
extended with new roles in future documents.<br><br>(As far as I know, =
those "future policy documents" have not been<br>published.)<br><br>More =
recently, draft-groves-clue-capture-attr specifies several<br>additional =
roles: "manager", "chairman", "secretary", "lecturer", and<br>"audience" =
(so-called person roles) as well as "speaker" and<br>"controller" =
(so-called conference roles). Personally, I happen to<br>have an =
interest in defining a few other roles, such as "host",<br>"presenter", =
"scribe", and "panelist".<br><br>Unfortunately there's quite a bit of =
overlap and confusion here. For<br>instance, one could argue that =
"host", "manager", and "controller"<br>(perhaps even "chairman") are the =
same as or very similar to<br>"administrator" or "moderator" from RFC =
6501. There might also be<br>overlap between "lecturer", "speaker", and =
"presenter" (and between<br>"speaker" and =
"panelist").<br><br>Furthermore, I'm concerned that various =
implementations will define<br>their own roles, blithely unaware that =
very similar (or even identical<br>but differently named roles) are =
already in use by other implementations.<br><br>It strikes me that one =
solution here would be to establish an IANA<br>registry for conference =
roles, with a registration policy (cf. RFC<br>5226) of at least Expert =
Review and perhaps Specification Required (I<br>think RFC Required is a =
bit heavy). IMHO this would help to prevent<br>continued confusion and =
the minting of new roles that overlap with<br>existing =
roles.<br><br>Thoughts?<br><br>Peter<br><br>______________________________=
_________________<br>dispatch mailing list<br><a =
href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf.o=
rg/mailman/listinfo/dispatch</a><br><br><br>______________________________=
_________________<br>clue mailing =
list<br>clue@ietf.org<br>https://www.ietf.org/mailman/listinfo/clue<br><br=
></div></blockquote></div><br><div apple-content-edited=3D"true">
<div style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; font-family: Helvetica; font-style: =
normal; font-variant: normal; font-weight: normal; letter-spacing: =
normal; line-height: normal; orphans: 2; text-align: -webkit-auto; =
text-indent: 0px; text-transform: none; white-space: normal; widows: 2; =
word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><div><div>&nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;<span =
class=3D"Apple-tab-span" style=3D"white-space: pre; ">				=
	</span><span class=3D"Apple-converted-space">&nbsp;</span>&nbsp; =
&nbsp; &nbsp; _\\|//_</div><div>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;<span =
class=3D"Apple-tab-span" style=3D"white-space: pre; ">				=
</span>&nbsp; &nbsp; &nbsp;&nbsp;( O-O )</div><div>&nbsp; =
&nbsp;~~~~~~~~~~~~~~~~~~~~~~o00~~(_)~~00o~~~~~~~~~~~~~~~~~~~~~~~~</div><di=
v>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;<span class=3D"Apple-converted-space">&nbsp;</span><span =
class=3D"Apple-tab-span" style=3D"white-space: pre; ">				=
</span>Simon Pietro Romano</div><div>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;<span class=3D"Apple-tab-span" style=3D"white-space: pre; =
">				</span><span =
class=3D"Apple-converted-space">&nbsp;</span>Universita' di Napoli =
Federico II</div><div>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&nbsp;<span class=3D"Apple-tab-span" style=3D"white-space: pre; ">	=
	</span>&nbsp; &nbsp; &nbsp;Computer Engineering =
Department&nbsp;</div><div><span class=3D"Apple-tab-span" =
style=3D"white-space: pre; ">	</span>&nbsp; &nbsp; &nbsp;&nbsp; &nbsp; =
&nbsp; &nbsp; Phone: +39 081 7683823 -- Fax: +39 081 =
7683816</div><div>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;e-mail: <a =
href=3D"mailto:spromano@unina.it">spromano@unina.it</a></div><div><br></di=
v><div><span class=3D"Apple-tab-span" style=3D"white-space: pre; ">		=
</span>&nbsp; &nbsp; &lt;&lt;Molti mi dicono che lo scoraggiamento =CB =
l'alibi degli&nbsp;</div><div><span class=3D"Apple-tab-span" =
style=3D"white-space: pre; ">		</span>&nbsp;&nbsp; =
&nbsp;idioti. Ci rifletto un istante; e mi scoraggio&gt;&gt;. =
Magritte.</div><div>&nbsp; &nbsp; &nbsp; &nbsp;&nbsp; &nbsp; &nbsp; =
&nbsp;<span class=3D"Apple-converted-space">&nbsp;</span><span =
class=3D"Apple-tab-span" style=3D"white-space: pre; ">			=
</span>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;oooO</div><div>&nbsp; ~~~~~~~~~~~~~~~~~~~~~~~( &nbsp; =
)~~~&nbsp;Oooo~~~~~~~~~~~~~~~~~~~~~~~~~</div><div><span =
class=3D"Apple-tab-span" style=3D"white-space: pre; ">				=
	</span>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;\ ( &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;( &nbsp; =
)</div><div><span class=3D"Apple-tab-span" style=3D"white-space: pre; ">	=
		</span>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
\_) &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;) /</div><div>&nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp;(_/</div></div><div><br></div></div></span><br =
class=3D"Apple-interchange-newline"></div><br =
class=3D"Apple-interchange-newline"><br =
class=3D"Apple-interchange-newline">
</div>
<br></div></div></blockquote></div><br><div apple-content-edited=3D"true">=

<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
color: rgb(0, 0, 0); font-family: Helvetica; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: =
0px; text-transform: none; white-space: normal; widows: 2; word-spacing: =
0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; color: =
rgb(0, 0, 0); font-family: Helvetica; font-style: normal; font-variant: =
normal; font-weight: normal; letter-spacing: normal; line-height: =
normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; =
text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
Helvetica; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; orphans: 2; =
text-align: -webkit-auto; text-indent: 0px; text-transform: none; =
white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><div><div>&nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;<span =
class=3D"Apple-tab-span" style=3D"white-space: pre; ">				=
	</span><span class=3D"Apple-converted-space">&nbsp;</span>&nbsp; =
&nbsp; &nbsp; _\\|//_</div><div>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;<span =
class=3D"Apple-tab-span" style=3D"white-space: pre; ">				=
</span>&nbsp; &nbsp; &nbsp;&nbsp;( O-O )</div><div>&nbsp; =
&nbsp;~~~~~~~~~~~~~~~~~~~~~~o00~~(_)~~00o~~~~~~~~~~~~~~~~~~~~~~~~</div><di=
v>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;<span class=3D"Apple-converted-space">&nbsp;</span><span =
class=3D"Apple-tab-span" style=3D"white-space: pre; ">				=
</span>Simon Pietro Romano</div><div>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;<span class=3D"Apple-tab-span" style=3D"white-space: pre; =
">				</span><span =
class=3D"Apple-converted-space">&nbsp;</span>Universita' di Napoli =
Federico II</div><div>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&nbsp;<span class=3D"Apple-tab-span" style=3D"white-space: pre; ">	=
	</span>&nbsp; &nbsp; &nbsp;Computer Engineering =
Department&nbsp;</div><div><span class=3D"Apple-tab-span" =
style=3D"white-space: pre; ">	</span>&nbsp; &nbsp; &nbsp;&nbsp; &nbsp; =
&nbsp; &nbsp; Phone: +39 081 7683823 -- Fax: +39 081 =
7683816</div><div>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;e-mail: <a =
href=3D"mailto:spromano@unina.it">spromano@unina.it</a></div><div><br></di=
v><div><span class=3D"Apple-tab-span" style=3D"white-space: pre; ">		=
</span>&nbsp; &nbsp; &lt;&lt;Molti mi dicono che lo scoraggiamento =CB =
l'alibi degli&nbsp;</div><div><span class=3D"Apple-tab-span" =
style=3D"white-space: pre; ">		</span>&nbsp;&nbsp; =
&nbsp;idioti. Ci rifletto un istante; e mi scoraggio&gt;&gt;. =
Magritte.</div><div>&nbsp; &nbsp; &nbsp; &nbsp;&nbsp; &nbsp; &nbsp; =
&nbsp;<span class=3D"Apple-converted-space">&nbsp;</span><span =
class=3D"Apple-tab-span" style=3D"white-space: pre; ">			=
</span>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;oooO</div><div>&nbsp; ~~~~~~~~~~~~~~~~~~~~~~~( &nbsp; =
)~~~&nbsp;Oooo~~~~~~~~~~~~~~~~~~~~~~~~~</div><div><span =
class=3D"Apple-tab-span" style=3D"white-space: pre; ">				=
	</span>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;\ ( &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;( &nbsp; =
)</div><div><span class=3D"Apple-tab-span" style=3D"white-space: pre; ">	=
		</span>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
\_) &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;) /</div><div>&nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp;(_/</div></div><div><br></div></div></span><br =
class=3D"Apple-interchange-newline"></div></span><br =
class=3D"Apple-interchange-newline"></span><br =
class=3D"Apple-interchange-newline">
</div>
<br></div></body></html>=

--Apple-Mail=_F8DF8A88-3E87-49AF-A658-7C7CB1DFBE1A--

From dromasca@avaya.com  Sun Apr  7 01:20:43 2013
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From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "dispatch@ietf.org" <dispatch@ietf.org>
Thread-Topic: [dispatch] SIP SNMP MIB and new transports
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Date: Sun, 7 Apr 2013 08:20:20 +0000
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Subject: Re: [dispatch] SIP SNMP MIB and new transports
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Hi,

> Is there some way to fix that, and link it to a
> new IANA registry?

Yes, there is a way. It involves updating RFC 4780 by placing the SipTCTran=
sportProtocol under IANA change control. Actually it may be a good idea to =
transfer all the TCs in the SIP MIB under the control of IANA, so that they=
 can be extended if and when there is a need to add new values to the enume=
rations or bitmaps. Thus the TCs do become the registries. The IANA conside=
rations policy section also needs to be updated in order to define the RFC =
5226 policy for updating the registries (for example ' Expert Review (or De=
signated Expert))'

It's not a horrendous task if it is focused on this specific change and the=
 impulse to make other updates to RFC 4780 is resisted.=20

Regards,

Dan
(Technical Advisor for the SIP WG when the SIP MIB was written)

> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Paul Kyzivat
> Sent: Friday, April 05, 2013 6:26 PM
> To: dispatch@ietf.org
> Subject: Re: [dispatch] SIP SNMP MIB and new transports
>=20
> Olle,
>=20
> On 4/5/13 4:12 AM, Olle E. Johansson wrote:
> > Hi!
> >
> > RFC 4780 defines an OID for transports. If we add new transports
> (websockets) we need to expand this table. The RFC in itself doesn't
> outline a procedure for extending it.
> >
> > Should the draft for websockets transport extend the MIB?
>=20
> It is too bad this wasn't raised sooner, e.g. during WGLC.
> draft-ietf-sipcore-sip-websocket-08 is now with the editor, so it is
> really too late to address this in that draft now.
>=20
> So if something needs to be done about this then I guess it will require
> a separate draft.
>=20
> For the moment, ISTM the obvious work around is:
>=20
>    bit 0: a protocol other than those defined here
>=20
> > In the Kamailio SNMP implementation, I have used bit 6 for ws and 7
> for wss.
>=20
> I know *nothing* about mibs, but when looking at 4780 I don't see any
> evidence that it was designed with a mechanism for extension as new
> transports are defined. Is there some way to fix that, and link it to a
> new IANA registry?
>=20
> 	Thanks,
> 	Paul
>=20
> > /O
> >
> > ------
> >
> > SipTCTransportProtocol ::=3D TEXTUAL-CONVENTION
> >      STATUS      current
> >      DESCRIPTION
> >         "This convention is a bit map.  Each bit represents a
> transport
> >          protocol.  If a bit has value 1, then that selected transport
> >          protocol is in some way dependent on the context of the
> object
> >          using this convention.  If a bit has value 0, then that
> >          transport protocol is not selected.  Combinations of bits can
> >          be set when multiple transport protocols are selected.
> >
> >          bit 0: a protocol other than those defined here
> >          bit 1: User Datagram Protocol
> >          bit 2: Transmission Control Protocol
> >          bit 3: Stream Control Transmission Protocol
> >          bit 4: Transport Layer Security Protocol over TCP
> >          bit 5: Transport Layer Security Protocol over SCTP
> >         "
> >      REFERENCE "
> > RFC 3261, Section 18 and RFC 4168
> > "
> >      SYNTAX      BITS {
> >                    other(0),  -- none of the following
> >                    udp(1),
> >                    tcp(2),
> >                    sctp(3),   --
> > RFC4168
> >
> >                    tlsTcp(4),
> >                    tlsSctp(5) --
> > RFC 4168
> >
> >                  }
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> >
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

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To: Peter Saint-Andre <stpeter@stpeter.im>
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Subject: Re: [dispatch] Fwd: I-D Action: draft-saintandre-sip-xmpp-im-03.txt
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Hi Peter,

I did a quick read of this version, in section 4, shouldn't we be =
looking at the RURI, instead of the To header?



On Apr 5, 2013, at 4:38 AM, Peter Saint-Andre wrote:

> FYI.
>=20
>=20
> -------- Original Message --------
> Subject: I-D Action: draft-saintandre-sip-xmpp-im-03.txt
> Date: Thu, 04 Apr 2013 19:37:18 -0700
> From: internet-drafts@ietf.org
> Reply-To: internet-drafts@ietf.org
> To: i-d-announce@ietf.org
>=20
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>=20
>=20
> 	Title           : Interworking between the Session Initiation =
Protocol
> (SIP) and the Extensible Messaging and Presence Protocol (XMPP): =
Instant
> Messaging
> 	Author(s)       : Peter Saint-Andre
>                          Avshalom Houri
>                          Joe Hildebrand
> 	Filename        : draft-saintandre-sip-xmpp-im-03.txt
> 	Pages           : 10
> 	Date            : 2013-04-04
>=20
> Abstract:
>   This document defines a bidirectional protocol mapping for the
>   exchange of single instant messages between the Session Initiation
>   Protocol (SIP) and the Extensible Messaging and Presence Protocol
>   (XMPP).
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-saintandre-sip-xmpp-im
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-saintandre-sip-xmpp-im-03
>=20
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-saintandre-sip-xmpp-im-03
>=20
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft directories: http://www.ietf.org/shadow.html
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>=20
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> _______________________________________________
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> https://www.ietf.org/mailman/listinfo/dispatch

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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To: Peter Saint-Andre <stpeter@stpeter.im>
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Subject: Re: [dispatch] Fwd: I-D Action: draft-saintandre-sip-xmpp-chat-05.txt
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Hi Peter,

Are you planning to include mappings for MSRP REPORT chunks and XEP-0184 =
delivery receipts? (not sure if I asked this earlier, I may have =
forgotten).


On Apr 5, 2013, at 5:10 AM, Peter Saint-Andre wrote:

> FYI.
>=20
>=20
> -------- Original Message --------
> Subject: I-D Action: draft-saintandre-sip-xmpp-chat-05.txt
> Date: Thu, 04 Apr 2013 20:09:11 -0700
> From: internet-drafts@ietf.org
> Reply-To: internet-drafts@ietf.org
> To: i-d-announce@ietf.org
>=20
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>=20
>=20
> 	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
>                          Eddy Gavita
>                          Nazin Hossain
>                          Salvatore Loreto
> 	Filename        : draft-saintandre-sip-xmpp-chat-05.txt
> 	Pages           : 12
> 	Date            : 2013-04-04
>=20
> 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).
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-saintandre-sip-xmpp-chat
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-saintandre-sip-xmpp-chat-05
>=20
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-saintandre-sip-xmpp-chat-05
>=20
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
> https://www.ietf.org/mailman/listinfo/i-d-announce
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> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>=20
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> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

--
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AG Projects




From saul@ag-projects.com  Sun Apr  7 10:44:32 2013
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Hi Peter,

Here are some comments on revision 5:

- Section 3.3.1: The initial SUBSCRIBE should have an empty to tag.

- Section 4.3: Tuple id should be ID-orchard

- Section 4.3, table 7: The reference to <body> looks like a c&p error =
;-)

I think we should make sue of the 'contact' child element of the PIDF =
tuple to provide the full (translated as per the core draft) URI of the =
presence entity. From/To headers cannot have GRUUs in them IIRC, so the =
contact element seems like the right place to hold this piece of =
information. This way the endpoint which processes this presence =
information will learn how to contact *this* particular instance.

In addition, the draft currently only considers the case where a single =
NOTIFY is generated for a presence stanza, but if RLS is used a presence =
agent will aggregate multiple PIDF documents in a single RLMI document =
and send it in a single NOTIFY, so not relying in the =46rom header =
would help.



Regards,

On Apr 3, 2013, at 6:03 AM, Peter Saint-Andre wrote:

> FYI.
>=20
>=20
> -------- Original Message --------
> Subject: I-D Action: draft-saintandre-sip-xmpp-presence-05.txt
> Date: Tue, 02 Apr 2013 21:02:57 -0700
> From: internet-drafts@ietf.org
> Reply-To: internet-drafts@ietf.org
> To: i-d-announce@ietf.org
>=20
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>=20
>=20
> 	Title           : Interworking between the Session Initiation =
Protocol
> (SIP) and the Extensible Messaging and Presence Protocol (XMPP): =
Presence
> 	Author(s)       : Peter Saint-Andre
>                          Avshalom Houri
>                          Joe Hildebrand
> 	Filename        : draft-saintandre-sip-xmpp-presence-05.txt
> 	Pages           : 20
> 	Date            : 2013-04-02
>=20
> Abstract:
>   This document defines a bi-directional protocol mapping for the
>   exchange of presence information between the Session Initiation
>   Protocol (SIP) and the Extensible Messaging and Presence Protocol
>   (XMPP).
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-saintandre-sip-xmpp-presence
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-saintandre-sip-xmpp-presence-05
>=20
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-saintandre-sip-xmpp-presence-05=

>=20
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
> https://www.ietf.org/mailman/listinfo/i-d-announce
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> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>=20
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

--
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References: <949EF20990823C4C85C18D59AA11AD8B01D5F4@FR712WXCHMBA11.zeu.alcatel-lucent.com> <CAHBDyN5zZ3f6mjfXVXbvvvB7FTTKPWfDHvC9CaHQgau4_8uL1w@mail.gmail.com> <004301ce2e30$2317e700$6947b500$@co.in> <CAHBDyN6mBMwE2pFMh-iT4i302pqRDvBXkn+7NctrdVrVkJ+2iQ@mail.gmail.com> <001201ce2efc$0695a5f0$13c0f1d0$@co.in> <CALiegfkZD1tO+=FXukZ+Vu32pHYeqzSYmCB7UX8wEK4iSzUp-A@mail.gmail.com> <003601ce3088$618e3c00$24aab400$@co.in> <CALiegf=QrFKTFfjC24e7ZdPrAE7oMprihgdROGyj-D6pzGZSeA@mail.gmail.com> <515C98AC.7020600@alum.mit.edu> <CALiegf=HvRFccYZfECL-c8pdiz3gSXULt4HpyboF2HF_-T0vQA@mail.gmail.com> <20130403235821.12f4e8a4@rainpc> <CAGTXFp-7-65wWOMcatcT+qFFjC3N1B880+8N5_n242kbrVE3JQ@mail.gmail.com> <20130404081830.6a01c45d@rainpc> <92B7E61ADAC1BB4F941F943788C08828047E386E@xmb-aln-x08.cisco.com>
Date: Sun, 7 Apr 2013 13:32:08 -0500
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To: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
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Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or websocket
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On Fri, Apr 5, 2013 at 5:09 PM, Charles Eckel (eckelcu)
<eckelcu@cisco.com> wrote:
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>> Behalf Of Lorenzo Miniero
>> Sent: Wednesday, April 03, 2013 11:19 PM
>> To: Victor Pascual Avila
>> Cc: dispatch@ietf.org
>> Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or websocke=
t
>>
>> On Thu, 4 Apr 2013 07:40:03 +0200
>> Victor Pascual Avila <victor.pascual.avila@gmail.com> wrote:
>>
>> > On Wed, Apr 3, 2013 at 11:58 PM, Lorenzo Miniero
>> <lorenzo@meetecho.com> wrote:
>> > > That said, though, I may have this view because of how, just as you,=
 I'm
>> currently associating BFCP with the conferencing world (well, it was bor=
n in
>> XCON after all), which may be a wrong assumption: people may be using
>> BFCP for entirely different scenarios as well, and the example made by
>> someone a few posts ago (I think it was Victor but I may be wrong) about
>> one of two peers acting as the floor control server in a call may be a
>> perfectly valid scenario for it being used differently than what I'm use=
d to.
>> >
>> > It was not me -- all my use cases include a conf server. On the client
>> > side, we have WebRTC and non-WebRTC web endpoints (all of them
>> > supporting WebSockets already) as well as "legacy" BFCP devices.
>> >
>> > Cheers,
>> > -Victor
>>
>>
>> Ops you're right, sorry about that! As they say here in Napoli, "that's =
how
>> you send people to jail" :-)
>>
>> I just checked the archives and it was Charles Eckel that referred to th=
e
>> possibility in a couple of posts.
>
> Yes, and I still stand by BFCP being useful and implemented for point to =
point and peer to peer use cases, not just for conference calls between a c=
lient and a conference server. If you want additional details, please have =
a look at the following IMTC Best Practice shared with the IETF through a l=
iaison statement last year:
> https://datatracker.ietf.org/documents/LIAISON/liaison-2012-05-31-imtc-th=
e-ietf-imtc-work-on-sip-feature-parity-with-h323-attachment-3.pdf
>
> That said, I am not saying BFCP over websockets is a bad idea. Rather, I =
am saying it is not sufficient for all the use cases for which BFCP is used=
 today. Perhaps that is okay, and we just need to define something else for=
 the other use cases, and applications need to know when to use one method =
as opposed to another. This adds complexity other than that of implementing=
 BFCP over websockets.
[MB] Right. I don't think anyone's suggesting BFCP over websockets
would subsume other usages.  We have different application
environments that need some of the same building blocks, but the
flexibility in "transports" is a good thing IMHO. [/MB]
>
> Cheers,
> Charles
>
>> Cheers,
>> Lorenzo
>>
>> --
>> Lorenzo Miniero, COB
>>
>> Meetecho s.r.l.
>> Web Conferencing and Collaboration Tools
>> http://www.meetecho.com
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
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From: "Gonzalo Salgueiro (gsalguei)" <gsalguei@cisco.com>
To: Mary Barnes <mary.ietf.barnes@gmail.com>
Thread-Topic: [dispatch] Use of BFCP or MSRP over datachannel or websocket
Thread-Index: AQHOM75EZYNQxCKQpECoMrHnIZ8oyZjLNIbo
Date: Sun, 7 Apr 2013 20:25:12 +0000
Message-ID: <1B5F7E07-2C42-437C-9855-C734D930B0E2@cisco.com>
References: <949EF20990823C4C85C18D59AA11AD8B01D5F4@FR712WXCHMBA11.zeu.alcatel-lucent.com> <CAHBDyN5zZ3f6mjfXVXbvvvB7FTTKPWfDHvC9CaHQgau4_8uL1w@mail.gmail.com> <004301ce2e30$2317e700$6947b500$@co.in> <CAHBDyN6mBMwE2pFMh-iT4i302pqRDvBXkn+7NctrdVrVkJ+2iQ@mail.gmail.com> <001201ce2efc$0695a5f0$13c0f1d0$@co.in> <CALiegfkZD1tO+=FXukZ+Vu32pHYeqzSYmCB7UX8wEK4iSzUp-A@mail.gmail.com> <003601ce3088$618e3c00$24aab400$@co.in> <CALiegf=QrFKTFfjC24e7ZdPrAE7oMprihgdROGyj-D6pzGZSeA@mail.gmail.com> <515C98AC.7020600@alum.mit.edu> <CALiegf=HvRFccYZfECL-c8pdiz3gSXULt4HpyboF2HF_-T0vQA@mail.gmail.com> <20130403235821.12f4e8a4@rainpc> <CAGTXFp-7-65wWOMcatcT+qFFjC3N1B880+8N5_n242kbrVE3JQ@mail.gmail.com> <20130404081830.6a01c45d@rainpc> <92B7E61ADAC1BB4F941F943788C08828047E386E@xmb-aln-x08.cisco.com>, <CAHBDyN6NY0XGGXchi=U+tpOv+zKb3HMoJFodnJe8k2Kes1Di2w@mail.gmail.com>
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Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or websocket
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On Apr 7, 2013, at 2:32 PM, "Mary Barnes" <mary.ietf.barnes@gmail.com> wrot=
e:

 <snip>

>> Yes, and I still stand by BFCP being useful and implemented for point to=
 point and peer to peer use cases, not just for conference calls between a =
client and a conference server. If you want additional details, please have=
 a look at the following IMTC Best Practice shared with the IETF through a =
liaison statement last year:
>> https://datatracker.ietf.org/documents/LIAISON/liaison-2012-05-31-imtc-t=
he-ietf-imtc-work-on-sip-feature-parity-with-h323-attachment-3.pdf
>>=20
>> That said, I am not saying BFCP over websockets is a bad idea. Rather, I=
 am saying it is not sufficient for all the use cases for which BFCP is use=
d today. Perhaps that is okay, and we just need to define something else fo=
r the other use cases, and applications need to know when to use one method=
 as opposed to another. This adds complexity other than that of implementin=
g BFCP over websockets.
> [MB] Right. I don't think anyone's suggesting BFCP over websockets
> would subsume other usages.  We have different application
> environments that need some of the same building blocks, but the
> flexibility in "transports" is a good thing IMHO. [/MB]

This is pretty much my same feeling on this.  There are customer use cases,=
 proven instances where it is the superior solution, and it plays nicely wi=
th legacy implementations.  I see no reason not to pursue this, despite any=
 (IMO grossly overestimated) perceived additional complexity introduced as =
a result of an additional transport choice.

Gonzalo=

From stpeter@stpeter.im  Sun Apr  7 15:15:28 2013
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Subject: [dispatch] Fwd: I-D Action: draft-saintandre-impp-call-info-01.txt
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FYI.


-------- Original Message --------
Subject: I-D Action: draft-saintandre-impp-call-info-01.txt
Date: Sun, 07 Apr 2013 15:13:45 -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           : Instant Messaging and Presence Purpose for the
Call-Info Header in the Session Initiation Protocol (SIP)
	Author(s)       : Peter Saint-Andre
	Filename        : draft-saintandre-impp-call-info-01.txt
	Pages           : 4
	Date            : 2013-04-07

Abstract:
   This document defines and registers a value of "impp" ("instant
   messaging and presence protocol") for the "purpose" header field
   parameter of the Call-Info header field in the Session Initiation
   Protocol (SIP).


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-saintandre-impp-call-info

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-saintandre-impp-call-info-01

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-saintandre-impp-call-info-01


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

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From Christian.Groves@nteczone.com  Sun Apr  7 22:43:35 2013
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Date: Mon, 08 Apr 2013 15:43:29 +1000
From: Christian Groves <Christian.Groves@nteczone.com>
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Subject: Re: [dispatch] [clue] Fwd:  conference roles
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Perhaps I should have kept this on the dispatch list?

Christian

On 8/04/2013 3:40 PM, Christian Groves wrote:
> Hello Peter,
>
> I had envisaged that from the CLUE perspective that an IANA registry 
> would be establish for Role (and some of the attributes). I'm not sure 
> that it is possible to harmonise RFC4575, RFC6501 and CLUE as I think 
> they have different semantics.
>
> RFC4575 seems just to have a free text field that's human readable. 
> The goal of the CLUE "role" attribute is to have a set of tags which 
> are human readable and able to used in an automatic way.
>
> RFC6501 talks about "conference operations that a participant can 
> perform" I think this is semantically different to what is being 
> proposed in CLUE which is more about the person/s in a capture (i.e. 
> those depicted in the RTP stream).
>
> If it helps lessen the confusion we could change the name of the CLUE 
> parameter from "role" to "Character", "Captured Persons" or "Person 
> represented by media", etc.? With the semantic that the media contains 
> a person who is acting in that character/role.
>
> I think part of the challenge will be to define the values. A person 
> can have multiple roles/characters, e.g. a chairman who is acting as a 
> scribe. Some have a similar function but have a different connotation 
> e.g. presenter versus lecturer. I think to move forward we need to 
> address these sorts of things. The previous RFCs seem to have largely 
> ignored the problem in the final text. I'm not implying that it wasn't 
> considered in their development.
>
> Regards, Christian
>
> On 5/04/2013 1:42 AM, Peter Saint-Andre wrote:
>> FYI.
>>
>>
>> -------- Original Message --------
>> Subject: [dispatch] conference roles
>> Date: Wed, 03 Apr 2013 19:56:01 -0600
>> From: Peter Saint-Andre <stpeter@stpeter.im>
>> To: DISPATCH <dispatch@ietf.org>
>>
>> RFC 4575 (Section 5.6.3) defines the concept of a role in a
>> conference, but does not specify or define any roles.
>>
>> 5.6.3. <roles>
>>
>>     This element MAY contain a set of human-readable strings describing
>>     the roles of the user in the conference.  Note that this information
>>     is applicable for human consumption only.  This specification does
>>     not define the set of possible conferencing roles or the semantics
>>     associated with each.  It is expected that future conferencing
>>     specifications will define these and the corresponding schema
>>     extensions, as appropriate.
>>
>> RFC 6501 (Section 4.6.5.2) specifies a few values for the role
>> element, but does not actually define what those values mean.
>>
>> 4.6.5.2. <roles>
>>
>>     A <role> provides the context for the set of conference operations
>>     that a participant can perform.  This element can contain one or 
>> more
>>     of the following values: "administrator", "moderator", "user",
>>     "participant", "observer", and "none".  A role of "none" indicates
>>     that any role is assigned.  The <roles> semantic definition is 
>> out of
>>     the scope of this document and is subject to future policy 
>> documents.
>>     This element can be extended with new roles in future documents.
>>
>> (As far as I know, those "future policy documents" have not been
>> published.)
>>
>> More recently, draft-groves-clue-capture-attr specifies several
>> additional roles: "manager", "chairman", "secretary", "lecturer", and
>> "audience" (so-called person roles) as well as "speaker" and
>> "controller" (so-called conference roles). Personally, I happen to
>> have an interest in defining a few other roles, such as "host",
>> "presenter", "scribe", and "panelist".
>>
>> Unfortunately there's quite a bit of overlap and confusion here. For
>> instance, one could argue that "host", "manager", and "controller"
>> (perhaps even "chairman") are the same as or very similar to
>> "administrator" or "moderator" from RFC 6501. There might also be
>> overlap between "lecturer", "speaker", and "presenter" (and between
>> "speaker" and "panelist").
>>
>> Furthermore, I'm concerned that various implementations will define
>> their own roles, blithely unaware that very similar (or even identical
>> but differently named roles) are already in use by other 
>> implementations.
>>
>> It strikes me that one solution here would be to establish an IANA
>> registry for conference roles, with a registration policy (cf. RFC
>> 5226) of at least Expert Review and perhaps Specification Required (I
>> think RFC Required is a bit heavy). IMHO this would help to prevent
>> continued confusion and the minting of new roles that overlap with
>> existing roles.
>>
>> Thoughts?
>>
>> Peter
>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
>>
>> _______________________________________________
>> clue mailing list
>> clue@ietf.org
>> https://www.ietf.org/mailman/listinfo/clue
>>
>


From Markus.Isomaki@nokia.com  Mon Apr  8 03:47:33 2013
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Subject: Re: [dispatch] Acronym for the SIP-XMPP effort
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Hi Gonzalo,

SIXPAC was the name of an earlier proposal to charter a WG about combined u=
se of SIP and XMPP at the endpoint. It never made it to a real WG, but some=
thing similar still exists in the form of the CUSAX proposal (http://datatr=
acker.ietf.org/doc/draft-ivov-xmpp-cusax/).=20

Since this effort we are now chartering is about translation/mapping/gatewa=
ying between SIP and XMPP, it may be best to use a different name for it to=
 avoid confusion with the combined usage work.=20

Markus


-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of ext Gonzalo Camarillo
Sent: 05 April, 2013 09:13
To: DISPATCH
Subject: [dispatch] Acronym for the SIP-XMPP effort

Folks,

I am going to handle the chartering of the SIP-XMPP effort, as discussed in=
 Orlando. As usual, we need to choose an acronym for the group. Among the a=
cronyms I have heard, I like this one best so far:

  SIXPAC - SIP Integration with XMPP in Presence Aware Contexts

While, per the RAI tradition, it refers to an alcoholic drink, usually beer=
s, it also refers to the nice six packs all RAI contributors are working on=
 during the spring in order to show off at the beach in summer :-)

Any other suggestions?

With respect to the chairs for this effort, we will be choosing them soon a=
s well. So, if you are interested but have not volunteered yet per my other=
 email (from yesterday), now it would be a great time to drop me a note.

Thanks,

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

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From: "Shekh-Yusef, Rifaat (Rifaat)" <rifatyu@avaya.com>
To: "dispatch@ietf.org" <dispatch@ietf.org>
Thread-Topic: [dispatch] draft-saintandre-sip-xmpp-media-02
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Hi Peter,

I did a quick review for the sip-xmpp-media-02 document and have the follow=
ing comments/questions:

Section 2.1, bullet 3:
This bullet states "The SIP signaling channel is transported over UDP, ..."=
. While SIP can be transported over UDP, it is also widely transported over=
 TCP and TLS. You may want to reword this sentence and clarify that stateme=
nt.

Section 2.2.1, Jingle Action to SIP Method mapping table:
content-accept, session-accept: I think that the SIP part should be "INVITE=
 response (18x)" instead of "... (1xx)"
content-add, content-modify, content-remove: how about UPDATE?

Section 2.3.1, first jingle,
you have two "from" attributes; I think that the second one should be the "=
to" attribute instead.

Section 2.3.1, the SDP in the first INVITE sent from the GW to the target, =
the m line should include the codecs offered by this SDP.

Section 2.3.1., 180 Ringing,
The "To" header seems to include a tag with the same value as of the resour=
ce of the "to" attribute in the Jingle session-initiation.
It is likely that these values will be different, and it would be up to the=
 GW to map these values.

General comments:
* This document seems to assume that the user part of the SIP id and XMPP i=
d is the same.
I understand that this document is trying to address the media mapping, but=
 is there any document that deals with the mapping of ids, and does it addr=
ess the use case of a user having different user parts for his SIP and XMPP=
 ids?
* A SIP INVITE could be forked and multiple responses sent to the originato=
r. Are you going to cover this use case? does this change the mapping table=
?


Regards,
Rifaat


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<p class=3D"MsoNormal">Hi Peter,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I did a quick review for the sip-xmpp-media-02 docum=
ent and have the following comments/questions:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Section 2.1, bullet 3: <o:p></o:p></p>
<p class=3D"MsoNormal">This bullet states &quot;The SIP signaling channel i=
s transported over UDP, ...&quot;. While SIP can be transported over UDP, i=
t is also widely transported over TCP and TLS. You may want to reword this =
sentence and clarify that statement.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Section 2.2.1, Jingle Action to SIP Method mapping t=
able:<o:p></o:p></p>
<p class=3D"MsoNormal">content-accept, session-accept: I think that the SIP=
 part should be &quot;INVITE response (18x)&quot; instead of &quot;... (1xx=
)&quot;<o:p></o:p></p>
<p class=3D"MsoNormal">content-add, content-modify, content-remove: how abo=
ut UPDATE?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Section 2.3.1, first jingle,<o:p></o:p></p>
<p class=3D"MsoNormal">you have two &quot;from&quot; attributes; I think th=
at the second one should be the &quot;to&quot; attribute instead.<o:p></o:p=
></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Section 2.3.1, the SDP in the first INVITE sent from=
 the GW to the target, the m line should include the codecs offered by this=
 SDP.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Section 2.3.1., 180 Ringing,<o:p></o:p></p>
<p class=3D"MsoNormal">The &quot;To&quot; header seems to include a tag wit=
h the same value as of the resource of the &quot;to&quot; attribute in the =
Jingle session-initiation.<o:p></o:p></p>
<p class=3D"MsoNormal">It is likely that these values will be different, an=
d it would be up to the GW to map these values.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">General comments:<o:p></o:p></p>
<p class=3D"MsoNormal">* This document seems to assume that the user part o=
f the SIP id and XMPP id is the same.<o:p></o:p></p>
<p class=3D"MsoNormal">I understand that this document is trying to address=
 the media mapping, but is there any document that deals with the mapping o=
f ids, and does it address the use case of a user having different user par=
ts for his SIP and XMPP ids?<o:p></o:p></p>
<p class=3D"MsoNormal">* A SIP INVITE could be forked and multiple response=
s sent to the originator. Are you going to cover this use case? does this c=
hange the mapping table?<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>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal">Rifaat<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</body>
</html>

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From info@vonniman.com  Mon Apr  8 07:37:40 2013
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Date: Mon, 08 Apr 2013 16:40:23 +0200
From: "info@vonniman.com" <info@vonniman.com>
To: <Markus.Isomaki@nokia.com>, <Gonzalo.Camarillo@ericsson.com>, <dispatch@ietf.org>
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Thread-Topic: [dispatch] Acronym for the SIP-XMPP effort
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Subject: Re: [dispatch] Acronym for the SIP-XMPP effort
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SIXPACK+1
DOUBLE-SIXPACK

A hard choice...
/Bruno


On 08/04/13 12:47, "Markus.Isomaki@nokia.com" <Markus.Isomaki@nokia.com>
wrote:

>Hi Gonzalo,
>
>SIXPAC was the name of an earlier proposal to charter a WG about combined
>use of SIP and XMPP at the endpoint. It never made it to a real WG, but
>something similar still exists in the form of the CUSAX proposal
>(http://datatracker.ietf.org/doc/draft-ivov-xmpp-cusax/).
>
>Since this effort we are now chartering is about
>translation/mapping/gatewaying between SIP and XMPP, it may be best to
>use a different name for it to avoid confusion with the combined usage
>work. 
>
>Markus
>
>
>-----Original Message-----
>From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>Behalf Of ext Gonzalo Camarillo
>Sent: 05 April, 2013 09:13
>To: DISPATCH
>Subject: [dispatch] Acronym for the SIP-XMPP effort
>
>Folks,
>
>I am going to handle the chartering of the SIP-XMPP effort, as discussed
>in Orlando. As usual, we need to choose an acronym for the group. Among
>the acronyms I have heard, I like this one best so far:
>
>  SIXPAC - SIP Integration with XMPP in Presence Aware Contexts
>
>While, per the RAI tradition, it refers to an alcoholic drink, usually
>beers, it also refers to the nice six packs all RAI contributors are
>working on during the spring in order to show off at the beach in summer
>:-)
>
>Any other suggestions?
>
>With respect to the chairs for this effort, we will be choosing them soon
>as well. So, if you are interested but have not volunteered yet per my
>other email (from yesterday), now it would be a great time to drop me a
>note.
>
>Thanks,
>
>Gonzalo
>_______________________________________________
>dispatch mailing list
>dispatch@ietf.org
>https://www.ietf.org/mailman/listinfo/dispatch
>_______________________________________________
>dispatch mailing list
>dispatch@ietf.org
>https://www.ietf.org/mailman/listinfo/dispatch



From stpeter@stpeter.im  Mon Apr  8 10:27:44 2013
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Subject: Re: [dispatch] draft-saintandre-sip-xmpp-media-02
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On 4/8/13 7:41 AM, Shekh-Yusef, Rifaat (Rifaat) wrote:

> I did a quick review for the sip-xmpp-media-02 document and have the following comments/questions:

Thanks, Rifaat! This document still needs quite a bit of work. SaÃºl
Ibarra and I are working to update draft-saintandre-sip-xmpp-groupchat
first, but we'll move on to draft-saintandre-sip-xmpp-media after that.
I'll reply to your message in more detail later this week.

Peter

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



From stpeter@stpeter.im  Mon Apr  8 10:50:10 2013
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Subject: Re: [dispatch] [clue] Fwd:  conference roles
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On 4/7/13 11:43 PM, Christian Groves wrote:
> Perhaps I should have kept this on the dispatch list?

Perhaps. At the least, it would be good to have one place to talk
about this topic.

> On 8/04/2013 3:40 PM, Christian Groves wrote:
>> Hello Peter,
>> 
>> I had envisaged that from the CLUE perspective that an IANA 
>> registry would be establish for Role (and some of the
>> attributes). I'm not sure that it is possible to harmonise
>> RFC4575, RFC6501 and CLUE as I think they have different
>> semantics.
>> 
>> RFC4575 seems just to have a free text field that's human
>> readable. The goal of the CLUE "role" attribute is to have a set
>> of tags which are human readable and able to used in an automatic
>> way.
>> 
>> RFC6501 talks about "conference operations that a participant
>> can perform" I think this is semantically different to what is
>> being proposed in CLUE which is more about the person/s in a
>> capture (i.e. those depicted in the RTP stream).
>> 
>> If it helps lessen the confusion we could change the name of the
>>  CLUE parameter from "role" to "Character", "Captured Persons" or
>>  "Person represented by media", etc.? With the semantic that the
>>  media contains a person who is acting in that character/role.

Apparently I got confused by the use of the term "role" in
draft-groves-clue-capture-attr because to me it seems like an
extension to the roles defined in RFC 6501, which provided some new
values for the same construct defined in RFC 4575. What you're saying
is that the CLUE roles proposed in draft-groves-clue-capture-attr are
actually a completely different construct. That is, RFC 4575 and RFC
6501 define roles as essentially bundles of permissions to perform
certain operations, whereas the roles / characters in
draft-groves-clue-capture-attr have nothing to do with conference
control. Correct?

>> I think part of the challenge will be to define the values. A 
>> person can have multiple roles/characters, e.g. a chairman who is
>>  acting as a scribe. Some have a similar function but have a 
>> different connotation e.g. presenter versus lecturer. I think to
>>  move forward we need to address these sorts of things. The
>> previous RFCs seem to have largely ignored the problem in the
>> final text.

Agreed.

Just FYI, in the XMPP community we had thought about the concept of
"hats", which are roles you can play in a conference (and you can play
multiple roles, just as you can wear different hats). We have a very
basic spec for hats here:

http://xmpp.org/extensions/xep-0317.html

Rather than continuing to pursue the hats idea, I've been thinking
that it might be better to settle on using conference-info syntax from
RFC 4575 (since that would improve interoperability across SIP systems
and XMPP systems -- there's nothing SIP-specific about conference-info
itself). Unfortunately, it appears that the conference-info syntax is
clear but the semantics are not. I'm happy to put some cycles into
clarifying the semantics if there's interest.

(Now it seems that is a separate topic from the CLUE work, since CLUE
isn't actually using the RFC 4575 construct. So discussing this on the
DISPATCH list seems correct.)

Peter

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

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Date: Mon, 8 Apr 2013 15:38:56 -0400
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Subject: Re: [dispatch] Acronym for the SIP-XMPP effort
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STOX - SIP-to-XMPP?



On Mon, Apr 8, 2013 at 6:47 AM, <Markus.Isomaki@nokia.com> wrote:

> Hi Gonzalo,
>
> SIXPAC was the name of an earlier proposal to charter a WG about combined
> use of SIP and XMPP at the endpoint. It never made it to a real WG, but
> something similar still exists in the form of the CUSAX proposal (
> http://datatracker.ietf.org/doc/draft-ivov-xmpp-cusax/).
>
> Since this effort we are now chartering is about
> translation/mapping/gatewaying between SIP and XMPP, it may be best to use
> a different name for it to avoid confusion with the combined usage work.
>
> Markus
>
>
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of ext Gonzalo Camarillo
> Sent: 05 April, 2013 09:13
> To: DISPATCH
> Subject: [dispatch] Acronym for the SIP-XMPP effort
>
> Folks,
>
> I am going to handle the chartering of the SIP-XMPP effort, as discussed
> in Orlando. As usual, we need to choose an acronym for the group. Among the
> acronyms I have heard, I like this one best so far:
>
>   SIXPAC - SIP Integration with XMPP in Presence Aware Contexts
>
> While, per the RAI tradition, it refers to an alcoholic drink, usually
> beers, it also refers to the nice six packs all RAI contributors are
> working on during the spring in order to show off at the beach in summer :-)
>
> Any other suggestions?
>
> With respect to the chairs for this effort, we will be choosing them soon
> as well. So, if you are interested but have not volunteered yet per my
> other email (from yesterday), now it would be a great time to drop me a
> note.
>
> Thanks,
>
> Gonzalo
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

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STOX - SIP-to-XMPP?<div><br></div><div><br><br><div class=3D"gmail_quote">O=
n Mon, Apr 8, 2013 at 6:47 AM,  <span dir=3D"ltr">&lt;<a href=3D"mailto:Mar=
kus.Isomaki@nokia.com" target=3D"_blank">Markus.Isomaki@nokia.com</a>&gt;</=
span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi Gonzalo,<br>
<br>
SIXPAC was the name of an earlier proposal to charter a WG about combined u=
se of SIP and XMPP at the endpoint. It never made it to a real WG, but some=
thing similar still exists in the form of the CUSAX proposal (<a href=3D"ht=
tp://datatracker.ietf.org/doc/draft-ivov-xmpp-cusax/" target=3D"_blank">htt=
p://datatracker.ietf.org/doc/draft-ivov-xmpp-cusax/</a>).<br>

<br>
Since this effort we are now chartering is about translation/mapping/gatewa=
ying between SIP and XMPP, it may be best to use a different name for it to=
 avoid confusion with the combined usage work.<br>
<br>
Markus<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
-----Original Message-----<br>
From: <a href=3D"mailto:dispatch-bounces@ietf.org">dispatch-bounces@ietf.or=
g</a> [mailto:<a href=3D"mailto:dispatch-bounces@ietf.org">dispatch-bounces=
@ietf.org</a>] On Behalf Of ext Gonzalo Camarillo<br>
Sent: 05 April, 2013 09:13<br>
To: DISPATCH<br>
Subject: [dispatch] Acronym for the SIP-XMPP effort<br>
<br>
Folks,<br>
<br>
I am going to handle the chartering of the SIP-XMPP effort, as discussed in=
 Orlando. As usual, we need to choose an acronym for the group. Among the a=
cronyms I have heard, I like this one best so far:<br>
<br>
=A0 SIXPAC - SIP Integration with XMPP in Presence Aware Contexts<br>
<br>
While, per the RAI tradition, it refers to an alcoholic drink, usually beer=
s, it also refers to the nice six packs all RAI contributors are working on=
 during the spring in order to show off at the beach in summer :-)<br>

<br>
Any other suggestions?<br>
<br>
With respect to the chairs for this effort, we will be choosing them soon a=
s well. So, if you are interested but have not volunteered yet per my other=
 email (from yesterday), now it would be a great time to drop me a note.<br=
>

<br>
Thanks,<br>
<br>
Gonzalo<br>
_______________________________________________<br>
dispatch mailing list<br>
<a href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" target=3D"_blank=
">https://www.ietf.org/mailman/listinfo/dispatch</a><br>
_______________________________________________<br>
dispatch mailing list<br>
<a href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" target=3D"_blank=
">https://www.ietf.org/mailman/listinfo/dispatch</a><br>
</div></div></blockquote></div><br></div>

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From: "Christou, Christos [USA]" <christou_chris@bah.com>
To: Richard Barnes <rlb@ipv.sx>, "Markus.Isomaki@nokia.com" <Markus.Isomaki@nokia.com>
Thread-Topic: [External]  Re: [dispatch] Acronym for the SIP-XMPP effort
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I like it.

From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Richard Barnes
Sent: Monday, April 08, 2013 3:39 PM
To: Markus.Isomaki@nokia.com
Cc: dispatch@ietf.org
Subject: [External] Re: [dispatch] Acronym for the SIP-XMPP effort

STOX - SIP-to-XMPP?


On Mon, Apr 8, 2013 at 6:47 AM, <Markus.Isomaki@nokia.com<mailto:Markus.Iso=
maki@nokia.com>> wrote:
Hi Gonzalo,

SIXPAC was the name of an earlier proposal to charter a WG about combined u=
se of SIP and XMPP at the endpoint. It never made it to a real WG, but some=
thing similar still exists in the form of the CUSAX proposal (http://datatr=
acker.ietf.org/doc/draft-ivov-xmpp-cusax/).

Since this effort we are now chartering is about translation/mapping/gatewa=
ying between SIP and XMPP, it may be best to use a different name for it to=
 avoid confusion with the combined usage work.

Markus


-----Original Message-----
From: dispatch-bounces@ietf.org<mailto:dispatch-bounces@ietf.org> [mailto:d=
ispatch-bounces@ietf.org<mailto:dispatch-bounces@ietf.org>] On Behalf Of ex=
t Gonzalo Camarillo
Sent: 05 April, 2013 09:13
To: DISPATCH
Subject: [dispatch] Acronym for the SIP-XMPP effort

Folks,

I am going to handle the chartering of the SIP-XMPP effort, as discussed in=
 Orlando. As usual, we need to choose an acronym for the group. Among the a=
cronyms I have heard, I like this one best so far:

  SIXPAC - SIP Integration with XMPP in Presence Aware Contexts

While, per the RAI tradition, it refers to an alcoholic drink, usually beer=
s, it also refers to the nice six packs all RAI contributors are working on=
 during the spring in order to show off at the beach in summer :-)

Any other suggestions?

With respect to the chairs for this effort, we will be choosing them soon a=
s well. So, if you are interested but have not volunteered yet per my other=
 email (from yesterday), now it would be a great time to drop me a note.

Thanks,

Gonzalo
_______________________________________________
dispatch mailing list
dispatch@ietf.org<mailto:dispatch@ietf.org>
https://www.ietf.org/mailman/listinfo/dispatch
_______________________________________________
dispatch mailing list
dispatch@ietf.org<mailto:dispatch@ietf.org>
https://www.ietf.org/mailman/listinfo/dispatch


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	{mso-style-priority:99;
	color:blue;
	text-decoration:underline;}
a:visited, span.MsoHyperlinkFollowed
	{mso-style-priority:99;
	color:purple;
	text-decoration:underline;}
p.MsoAcetate, li.MsoAcetate, div.MsoAcetate
	{mso-style-priority:99;
	mso-style-link:"Balloon Text Char";
	margin:0in;
	margin-bottom:.0001pt;
	font-size:8.0pt;
	font-family:"Tahoma","sans-serif";}
span.BalloonTextChar
	{mso-style-name:"Balloon Text Char";
	mso-style-priority:99;
	mso-style-link:"Balloon Text";
	font-family:"Tahoma","sans-serif";}
span.EmailStyle19
	{mso-style-type:personal-reply;
	font-family:"Calibri","sans-serif";
	color:#1F497D;}
.MsoChpDefault
	{mso-style-type:export-only;
	font-family:"Calibri","sans-serif";}
@page WordSection1
	{size:8.5in 11.0in;
	margin:1.0in 1.0in 1.0in 1.0in;}
div.WordSection1
	{page:WordSection1;}
--></style><!--[if gte mso 9]><xml>
<o:shapedefaults v:ext=3D"edit" spidmax=3D"1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext=3D"edit">
<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]-->
</head>
<body lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">I like it.<o:p></o:p></sp=
an></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> dispatch=
-bounces@ietf.org [mailto:dispatch-bounces@ietf.org]
<b>On Behalf Of </b>Richard Barnes<br>
<b>Sent:</b> Monday, April 08, 2013 3:39 PM<br>
<b>To:</b> Markus.Isomaki@nokia.com<br>
<b>Cc:</b> dispatch@ietf.org<br>
<b>Subject:</b> [External] Re: [dispatch] Acronym for the SIP-XMPP effort<o=
:p></o:p></span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">STOX - SIP-to-XMPP?<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On Mon, Apr 8, 2013 at 6:47 AM, &lt;<a href=3D"mailt=
o:Markus.Isomaki@nokia.com" target=3D"_blank">Markus.Isomaki@nokia.com</a>&=
gt; wrote:<o:p></o:p></p>
<p class=3D"MsoNormal">Hi Gonzalo,<br>
<br>
SIXPAC was the name of an earlier proposal to charter a WG about combined u=
se of SIP and XMPP at the endpoint. It never made it to a real WG, but some=
thing similar still exists in the form of the CUSAX proposal (<a href=3D"ht=
tp://datatracker.ietf.org/doc/draft-ivov-xmpp-cusax/" target=3D"_blank">htt=
p://datatracker.ietf.org/doc/draft-ivov-xmpp-cusax/</a>).<br>
<br>
Since this effort we are now chartering is about translation/mapping/gatewa=
ying between SIP and XMPP, it may be best to use a different name for it to=
 avoid confusion with the combined usage work.<br>
<br>
Markus<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><br>
<br>
-----Original Message-----<br>
From: <a href=3D"mailto:dispatch-bounces@ietf.org">dispatch-bounces@ietf.or=
g</a> [mailto:<a href=3D"mailto:dispatch-bounces@ietf.org">dispatch-bounces=
@ietf.org</a>] On Behalf Of ext Gonzalo Camarillo<br>
Sent: 05 April, 2013 09:13<br>
To: DISPATCH<br>
Subject: [dispatch] Acronym for the SIP-XMPP effort<br>
<br>
Folks,<br>
<br>
I am going to handle the chartering of the SIP-XMPP effort, as discussed in=
 Orlando. As usual, we need to choose an acronym for the group. Among the a=
cronyms I have heard, I like this one best so far:<br>
<br>
&nbsp; SIXPAC - SIP Integration with XMPP in Presence Aware Contexts<br>
<br>
While, per the RAI tradition, it refers to an alcoholic drink, usually beer=
s, it also refers to the nice six packs all RAI contributors are working on=
 during the spring in order to show off at the beach in summer :-)<br>
<br>
Any other suggestions?<br>
<br>
With respect to the chairs for this effort, we will be choosing them soon a=
s well. So, if you are interested but have not volunteered yet per my other=
 email (from yesterday), now it would be a great time to drop me a note.<br=
>
<br>
Thanks,<br>
<br>
Gonzalo<br>
_______________________________________________<br>
dispatch mailing list<br>
<a href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" target=3D"_blank=
">https://www.ietf.org/mailman/listinfo/dispatch</a><br>
_______________________________________________<br>
dispatch mailing list<br>
<a href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" target=3D"_blank=
">https://www.ietf.org/mailman/listinfo/dispatch</a><o:p></o:p></p>
</div>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</body>
</html>

--_000_462D69106F82344E95BC83996D8039D25A02059DASHBDAG1M1resou_--

From stpeter@stpeter.im  Mon Apr  8 19:03:51 2013
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Date: Mon, 08 Apr 2013 20:03:45 -0600
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Hash: SHA1

On 4/8/13 11:50 AM, Peter Saint-Andre wrote:

> Just FYI, in the XMPP community we had thought about the concept
> of "hats", which are roles you can play in a conference (and you
> can play multiple roles, just as you can wear different hats). We
> have a very basic spec for hats here:
> 
> http://xmpp.org/extensions/xep-0317.html
> 
> Rather than continuing to pursue the hats idea, I've been thinking 
> that it might be better to settle on using conference-info syntax
> from RFC 4575 (since that would improve interoperability across SIP
> systems and XMPP systems -- there's nothing SIP-specific about
> conference-info itself). Unfortunately, it appears that the
> conference-info syntax is clear but the semantics are not. I'm
> happy to put some cycles into clarifying the semantics if there's
> interest.

BTW, looking at RFC 4575, I notice that it defines a <roles> element
but doesn't describe the syntax very precisely:

4.6.5.2. <roles>

   A <role> provides the context for the set of conference operations
   that a participant can perform.  This element can contain one or more
   of the following values: "administrator", "moderator", "user",
   "participant", "observer", and "none".  A role of "none" indicates
   that any role is assigned.  The <roles> semantic definition is out of
   the scope of this document and is subject to future policy documents.
   This element can be extended with new roles in future documents.

That seems to imply that the syntax is:

  <roles>
    <role>foo</role>
  </roles>

But the examples have:

  <roles>
    <entry>foo</entry>
  </roles>

And is the syntax a comma-separated or space-separated list of values,
or is it multiple child elements (whether <role/> or <entry/>)?

Anyone up for 4575bis??? ;-)

Peter

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


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From Christian.Groves@nteczone.com  Mon Apr  8 19:27:29 2013
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Subject: Re: [dispatch] [clue] Fwd:  conference roles
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Hello Peter,

Do you mean RFC6501bis? :-)

Christian

On 9/04/2013 12:03 PM, Peter Saint-Andre wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> On 4/8/13 11:50 AM, Peter Saint-Andre wrote:
>
>> Just FYI, in the XMPP community we had thought about the concept
>> of "hats", which are roles you can play in a conference (and you
>> can play multiple roles, just as you can wear different hats). We
>> have a very basic spec for hats here:
>>
>> http://xmpp.org/extensions/xep-0317.html
>>
>> Rather than continuing to pursue the hats idea, I've been thinking
>> that it might be better to settle on using conference-info syntax
>> from RFC 4575 (since that would improve interoperability across SIP
>> systems and XMPP systems -- there's nothing SIP-specific about
>> conference-info itself). Unfortunately, it appears that the
>> conference-info syntax is clear but the semantics are not. I'm
>> happy to put some cycles into clarifying the semantics if there's
>> interest.
> BTW, looking at RFC 4575, I notice that it defines a <roles> element
> but doesn't describe the syntax very precisely:
>
> 4.6.5.2. <roles>
>
>     A <role> provides the context for the set of conference operations
>     that a participant can perform.  This element can contain one or more
>     of the following values: "administrator", "moderator", "user",
>     "participant", "observer", and "none".  A role of "none" indicates
>     that any role is assigned.  The <roles> semantic definition is out of
>     the scope of this document and is subject to future policy documents.
>     This element can be extended with new roles in future documents.
>
> That seems to imply that the syntax is:
>
>    <roles>
>      <role>foo</role>
>    </roles>
>
> But the examples have:
>
>    <roles>
>      <entry>foo</entry>
>    </roles>
>
> And is the syntax a comma-separated or space-separated list of values,
> or is it multiple child elements (whether <role/> or <entry/>)?
>
> Anyone up for 4575bis??? ;-)
>
> Peter
>
> - -- 
> Peter Saint-Andre
> https://stpeter.im/
>
>
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> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>


From stpeter@stpeter.im  Mon Apr  8 19:30:38 2013
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Hash: SHA1

On 4/8/13 8:27 PM, Christian Groves wrote:
> Hello Peter,
> 
> Do you mean RFC6501bis? :-)

Maybe both, given that RFC 6501 elements refine, extend, or are
interleaved with the root <conference-info/> element from RFC 4575.

(But yes, I was quoting from RFC 6501 there...)

Peter

- -- 
Peter Saint-Andre
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From gonzalo.camarillo@ericsson.com  Tue Apr  9 05:47:05 2013
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Date: Tue, 9 Apr 2013 15:46:58 +0300
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Cc: dispatch@ietf.org
Subject: Re: [dispatch] Acronym for the SIP-XMPP effort
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Hi,

I agree we want to avoid creating confusion by re-using an acronym. I am
fine with using STOX instead. It is clearly more financial oriented.

Thanks,

Gonzalo

On 08/04/2013 10:38 PM, Richard Barnes wrote:
> STOX - SIP-to-XMPP?
> 
> 
> 
> On Mon, Apr 8, 2013 at 6:47 AM, <Markus.Isomaki@nokia.com
> <mailto:Markus.Isomaki@nokia.com>> wrote:
> 
>     Hi Gonzalo,
> 
>     SIXPAC was the name of an earlier proposal to charter a WG about
>     combined use of SIP and XMPP at the endpoint. It never made it to a
>     real WG, but something similar still exists in the form of the CUSAX
>     proposal (http://datatracker.ietf.org/doc/draft-ivov-xmpp-cusax/).
> 
>     Since this effort we are now chartering is about
>     translation/mapping/gatewaying between SIP and XMPP, it may be best
>     to use a different name for it to avoid confusion with the combined
>     usage work.
> 
>     Markus
> 
> 
>     -----Original Message-----
>     From: dispatch-bounces@ietf.org <mailto:dispatch-bounces@ietf.org>
>     [mailto:dispatch-bounces@ietf.org
>     <mailto:dispatch-bounces@ietf.org>] On Behalf Of ext Gonzalo Camarillo
>     Sent: 05 April, 2013 09:13
>     To: DISPATCH
>     Subject: [dispatch] Acronym for the SIP-XMPP effort
> 
>     Folks,
> 
>     I am going to handle the chartering of the SIP-XMPP effort, as
>     discussed in Orlando. As usual, we need to choose an acronym for the
>     group. Among the acronyms I have heard, I like this one best so far:
> 
>       SIXPAC - SIP Integration with XMPP in Presence Aware Contexts
> 
>     While, per the RAI tradition, it refers to an alcoholic drink,
>     usually beers, it also refers to the nice six packs all RAI
>     contributors are working on during the spring in order to show off
>     at the beach in summer :-)
> 
>     Any other suggestions?
> 
>     With respect to the chairs for this effort, we will be choosing them
>     soon as well. So, if you are interested but have not volunteered yet
>     per my other email (from yesterday), now it would be a great time to
>     drop me a note.
> 
>     Thanks,
> 
>     Gonzalo
>     _______________________________________________
>     dispatch mailing list
>     dispatch@ietf.org <mailto:dispatch@ietf.org>
>     https://www.ietf.org/mailman/listinfo/dispatch
>     _______________________________________________
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> 
> 


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To: Peter Saint-Andre <stpeter@stpeter.im>
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Cc: DISPATCH <dispatch@ietf.org>
Subject: Re: [dispatch] Fwd: I-D Action: draft-saintandre-impp-call-info-01.txt
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Folks,

based on the discussions we have had on this draft (messages with "URI
schemes in P-Asserted-Identity" in their Subjects), I am going to AD
sponsor this draft.

Cheers,

Gonzalo

On 08/04/2013 1:15 AM, Peter Saint-Andre wrote:
> FYI.
> 
> 
> -------- Original Message --------
> Subject: I-D Action: draft-saintandre-impp-call-info-01.txt
> Date: Sun, 07 Apr 2013 15:13:45 -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           : Instant Messaging and Presence Purpose for the
> Call-Info Header in the Session Initiation Protocol (SIP)
> 	Author(s)       : Peter Saint-Andre
> 	Filename        : draft-saintandre-impp-call-info-01.txt
> 	Pages           : 4
> 	Date            : 2013-04-07
> 
> Abstract:
>    This document defines and registers a value of "impp" ("instant
>    messaging and presence protocol") for the "purpose" header field
>    parameter of the Call-Info header field in the Session Initiation
>    Protocol (SIP).
> 
> 
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-saintandre-impp-call-info
> 
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-saintandre-impp-call-info-01
> 
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-saintandre-impp-call-info-01
> 
> 
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
> 
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
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> 
> 
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> 


From pkyzivat@alum.mit.edu  Tue Apr  9 06:57:54 2013
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Subject: Re: [dispatch] [clue] Fwd:  conference roles
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Coming late to the discussion...

I agree that what CLUE is considering is a different usage than what RFC 
4575 is talking about. But in both cases we are talking about attributes 
that are applied to *something*. For CLUE it is Captures that we are 
attaching attributes to while in RFC 4575 it is participants.

If it makes sense, and I'm not yet sure it does in this case, we could 
have a definition of "role" that can be used by CLUE to describe 
captures and is also used in the conference event package to describe 
participants.

My point is that the "name" of this attribute need not fully define what 
we mean. Rather it is the name together with the thing it is applied to. 
So I don't have a problem calling this "role". (But I also have no 
problem calling it something else.)

In CLUE we tried this once - with "content" based on reusing the values 
from a=content. But then we decided that definition was too narrow and 
extending it enough would have undesirable consequences. It may turn out 
that sharing/reusing a definition for "role" is also problematic. But we 
can at least investigate it.

     Thanks,
     Paul


On 4/9/13 1:50 AM, Peter Saint-Andre wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> On 4/7/13 11:43 PM, Christian Groves wrote:
>> Perhaps I should have kept this on the dispatch list?
>
> Perhaps. At the least, it would be good to have one place to talk
> about this topic.
>
>> On 8/04/2013 3:40 PM, Christian Groves wrote:
>>> Hello Peter,
>>>
>>> I had envisaged that from the CLUE perspective that an IANA
>>> registry would be establish for Role (and some of the
>>> attributes). I'm not sure that it is possible to harmonise
>>> RFC4575, RFC6501 and CLUE as I think they have different
>>> semantics.
>>>
>>> RFC4575 seems just to have a free text field that's human
>>> readable. The goal of the CLUE "role" attribute is to have a set
>>> of tags which are human readable and able to used in an automatic
>>> way.
>>>
>>> RFC6501 talks about "conference operations that a participant
>>> can perform" I think this is semantically different to what is
>>> being proposed in CLUE which is more about the person/s in a
>>> capture (i.e. those depicted in the RTP stream).
>>>
>>> If it helps lessen the confusion we could change the name of the
>>>   CLUE parameter from "role" to "Character", "Captured Persons" or
>>>   "Person represented by media", etc.? With the semantic that the
>>>   media contains a person who is acting in that character/role.
>
> Apparently I got confused by the use of the term "role" in
> draft-groves-clue-capture-attr because to me it seems like an
> extension to the roles defined in RFC 6501, which provided some new
> values for the same construct defined in RFC 4575. What you're saying
> is that the CLUE roles proposed in draft-groves-clue-capture-attr are
> actually a completely different construct. That is, RFC 4575 and RFC
> 6501 define roles as essentially bundles of permissions to perform
> certain operations, whereas the roles / characters in
> draft-groves-clue-capture-attr have nothing to do with conference
> control. Correct?
>
>>> I think part of the challenge will be to define the values. A
>>> person can have multiple roles/characters, e.g. a chairman who is
>>>   acting as a scribe. Some have a similar function but have a
>>> different connotation e.g. presenter versus lecturer. I think to
>>>   move forward we need to address these sorts of things. The
>>> previous RFCs seem to have largely ignored the problem in the
>>> final text.
>
> Agreed.
>
> Just FYI, in the XMPP community we had thought about the concept of
> "hats", which are roles you can play in a conference (and you can play
> multiple roles, just as you can wear different hats). We have a very
> basic spec for hats here:
>
> http://xmpp.org/extensions/xep-0317.html
>
> Rather than continuing to pursue the hats idea, I've been thinking
> that it might be better to settle on using conference-info syntax from
> RFC 4575 (since that would improve interoperability across SIP systems
> and XMPP systems -- there's nothing SIP-specific about conference-info
> itself). Unfortunately, it appears that the conference-info syntax is
> clear but the semantics are not. I'm happy to put some cycles into
> clarifying the semantics if there's interest.
>
> (Now it seems that is a separate topic from the CLUE work, since CLUE
> isn't actually using the RFC 4575 construct. So discussing this on the
> DISPATCH list seems correct.)
>
> Peter
>
> - --
> Peter Saint-Andre
> https://stpeter.im/
>
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Date: Tue, 9 Apr 2013 09:15:11 -0500
Message-ID: <CAHBDyN5vXFM4P-np53EciEbs+ObsD3a0-73OxPs+tTKZNDLygw@mail.gmail.com>
From: Mary Barnes <mary.ietf.barnes@gmail.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Cc: DISPATCH <dispatch@ietf.org>
Subject: Re: [dispatch] [clue] Fwd: conference roles
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Well, I had to join in the thread since it seems appropriate for Mary
to respond to a thread with Peter & Paul ;)   I do not think that the
CLUE usage is any different really than XCON conference package
defined role -  in terms of the "roles" being both human readable and
the semantics being unique to an application  and can be based on
policies.  The latter was the XCON/conference intent as clearly stated
in  section 4.6.5.2 of 6501. It's the right thing for SIP based
conferencing to be using this information and since CLUE is based on
SIP, we need to have these things consistent.

Regards,
Mary.

On Tue, Apr 9, 2013 at 8:57 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
> Coming late to the discussion...
>
> I agree that what CLUE is considering is a different usage than what RFC
> 4575 is talking about. But in both cases we are talking about attributes
> that are applied to *something*. For CLUE it is Captures that we are
> attaching attributes to while in RFC 4575 it is participants.
>
> If it makes sense, and I'm not yet sure it does in this case, we could have
> a definition of "role" that can be used by CLUE to describe captures and is
> also used in the conference event package to describe participants.
>
> My point is that the "name" of this attribute need not fully define what we
> mean. Rather it is the name together with the thing it is applied to. So I
> don't have a problem calling this "role". (But I also have no problem
> calling it something else.)
>
> In CLUE we tried this once - with "content" based on reusing the values from
> a=content. But then we decided that definition was too narrow and extending
> it enough would have undesirable consequences. It may turn out that
> sharing/reusing a definition for "role" is also problematic. But we can at
> least investigate it.
>
>     Thanks,
>     Paul
>
>
>
> On 4/9/13 1:50 AM, Peter Saint-Andre wrote:
>>
>> -----BEGIN PGP SIGNED MESSAGE-----
>> Hash: SHA1
>>
>> On 4/7/13 11:43 PM, Christian Groves wrote:
>>>
>>> Perhaps I should have kept this on the dispatch list?
>>
>>
>> Perhaps. At the least, it would be good to have one place to talk
>> about this topic.
>>
>>> On 8/04/2013 3:40 PM, Christian Groves wrote:
>>>>
>>>> Hello Peter,
>>>>
>>>> I had envisaged that from the CLUE perspective that an IANA
>>>> registry would be establish for Role (and some of the
>>>> attributes). I'm not sure that it is possible to harmonise
>>>> RFC4575, RFC6501 and CLUE as I think they have different
>>>> semantics.
>>>>
>>>> RFC4575 seems just to have a free text field that's human
>>>> readable. The goal of the CLUE "role" attribute is to have a set
>>>> of tags which are human readable and able to used in an automatic
>>>> way.
>>>>
>>>> RFC6501 talks about "conference operations that a participant
>>>> can perform" I think this is semantically different to what is
>>>> being proposed in CLUE which is more about the person/s in a
>>>> capture (i.e. those depicted in the RTP stream).
>>>>
>>>> If it helps lessen the confusion we could change the name of the
>>>>   CLUE parameter from "role" to "Character", "Captured Persons" or
>>>>   "Person represented by media", etc.? With the semantic that the
>>>>   media contains a person who is acting in that character/role.
>>
>>
>> Apparently I got confused by the use of the term "role" in
>> draft-groves-clue-capture-attr because to me it seems like an
>> extension to the roles defined in RFC 6501, which provided some new
>> values for the same construct defined in RFC 4575. What you're saying
>> is that the CLUE roles proposed in draft-groves-clue-capture-attr are
>> actually a completely different construct. That is, RFC 4575 and RFC
>> 6501 define roles as essentially bundles of permissions to perform
>> certain operations, whereas the roles / characters in
>> draft-groves-clue-capture-attr have nothing to do with conference
>> control. Correct?
>>
>>>> I think part of the challenge will be to define the values. A
>>>> person can have multiple roles/characters, e.g. a chairman who is
>>>>   acting as a scribe. Some have a similar function but have a
>>>> different connotation e.g. presenter versus lecturer. I think to
>>>>   move forward we need to address these sorts of things. The
>>>> previous RFCs seem to have largely ignored the problem in the
>>>> final text.
>>
>>
>> Agreed.
>>
>> Just FYI, in the XMPP community we had thought about the concept of
>> "hats", which are roles you can play in a conference (and you can play
>> multiple roles, just as you can wear different hats). We have a very
>> basic spec for hats here:
>>
>> http://xmpp.org/extensions/xep-0317.html
>>
>> Rather than continuing to pursue the hats idea, I've been thinking
>> that it might be better to settle on using conference-info syntax from
>> RFC 4575 (since that would improve interoperability across SIP systems
>> and XMPP systems -- there's nothing SIP-specific about conference-info
>> itself). Unfortunately, it appears that the conference-info syntax is
>> clear but the semantics are not. I'm happy to put some cycles into
>> clarifying the semantics if there's interest.
>>
>> (Now it seems that is a separate topic from the CLUE work, since CLUE
>> isn't actually using the RFC 4575 construct. So discussing this on the
>> DISPATCH list seems correct.)
>>
>> Peter
>>
>> - --
>> Peter Saint-Andre
>> https://stpeter.im/
>>
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>
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From pkyzivat@alum.mit.edu  Tue Apr  9 08:13:42 2013
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Subject: Re: [dispatch] Acronym for the SIP-XMPP effort
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On 4/9/13 8:46 PM, Gonzalo Camarillo wrote:
> Hi,
>
> I agree we want to avoid creating confusion by re-using an acronym. I am
> fine with using STOX instead. It is clearly more financial oriented.

Hmm - I was thinking it was referring to antique torture devices.

	Paul

> Thanks,
>
> Gonzalo
>
> On 08/04/2013 10:38 PM, Richard Barnes wrote:
>> STOX - SIP-to-XMPP?
>>
>>
>>
>> On Mon, Apr 8, 2013 at 6:47 AM, <Markus.Isomaki@nokia.com
>> <mailto:Markus.Isomaki@nokia.com>> wrote:
>>
>>      Hi Gonzalo,
>>
>>      SIXPAC was the name of an earlier proposal to charter a WG about
>>      combined use of SIP and XMPP at the endpoint. It never made it to a
>>      real WG, but something similar still exists in the form of the CUSAX
>>      proposal (http://datatracker.ietf.org/doc/draft-ivov-xmpp-cusax/).
>>
>>      Since this effort we are now chartering is about
>>      translation/mapping/gatewaying between SIP and XMPP, it may be best
>>      to use a different name for it to avoid confusion with the combined
>>      usage work.
>>
>>      Markus
>>
>>
>>      -----Original Message-----
>>      From: dispatch-bounces@ietf.org <mailto:dispatch-bounces@ietf.org>
>>      [mailto:dispatch-bounces@ietf.org
>>      <mailto:dispatch-bounces@ietf.org>] On Behalf Of ext Gonzalo Camarillo
>>      Sent: 05 April, 2013 09:13
>>      To: DISPATCH
>>      Subject: [dispatch] Acronym for the SIP-XMPP effort
>>
>>      Folks,
>>
>>      I am going to handle the chartering of the SIP-XMPP effort, as
>>      discussed in Orlando. As usual, we need to choose an acronym for the
>>      group. Among the acronyms I have heard, I like this one best so far:
>>
>>        SIXPAC - SIP Integration with XMPP in Presence Aware Contexts
>>
>>      While, per the RAI tradition, it refers to an alcoholic drink,
>>      usually beers, it also refers to the nice six packs all RAI
>>      contributors are working on during the spring in order to show off
>>      at the beach in summer :-)
>>
>>      Any other suggestions?
>>
>>      With respect to the chairs for this effort, we will be choosing them
>>      soon as well. So, if you are interested but have not volunteered yet
>>      per my other email (from yesterday), now it would be a great time to
>>      drop me a note.
>>
>>      Thanks,
>>
>>      Gonzalo
>>      _______________________________________________
>>      dispatch mailing list
>>      dispatch@ietf.org <mailto:dispatch@ietf.org>
>>      https://www.ietf.org/mailman/listinfo/dispatch
>>      _______________________________________________
>>      dispatch mailing list
>>      dispatch@ietf.org <mailto:dispatch@ietf.org>
>>      https://www.ietf.org/mailman/listinfo/dispatch
>>
>>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>


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To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>, Richard Barnes <rlb@ipv.sx>
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Subject: Re: [dispatch] Acronym for the SIP-XMPP effort
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At 07:46 AM 4/9/2013, Gonzalo Camarillo wrote:
>Hi,
>
>I agree we want to avoid creating confusion by re-using an acronym. I am
>fine with using STOX instead. It is clearly more financial oriented.

think of it this way, following (buying/selling/owning/investing 
in/being stuck with/buying high - selling low) STOX (i.e., stocks) 
will *make* you drink... so it fits our theme indirectly.

James


>Thanks,
>
>Gonzalo
>
>On 08/04/2013 10:38 PM, Richard Barnes wrote:
> > STOX - SIP-to-XMPP?
> >
> >
> >
> > On Mon, Apr 8, 2013 at 6:47 AM, <Markus.Isomaki@nokia.com
> > <mailto:Markus.Isomaki@nokia.com>> wrote:
> >
> >     Hi Gonzalo,
> >
> >     SIXPAC was the name of an earlier proposal to charter a WG about
> >     combined use of SIP and XMPP at the endpoint. It never made it to a
> >     real WG, but something similar still exists in the form of the CUSAX
> >     proposal (http://datatracker.ietf.org/doc/draft-ivov-xmpp-cusax/).
> >
> >     Since this effort we are now chartering is about
> >     translation/mapping/gatewaying between SIP and XMPP, it may be best
> >     to use a different name for it to avoid confusion with the combined
> >     usage work.
> >
> >     Markus
> >
> >
> >     -----Original Message-----
> >     From: dispatch-bounces@ietf.org <mailto:dispatch-bounces@ietf.org>
> >     [mailto:dispatch-bounces@ietf.org
> >     <mailto:dispatch-bounces@ietf.org>] On Behalf Of ext Gonzalo Camarillo
> >     Sent: 05 April, 2013 09:13
> >     To: DISPATCH
> >     Subject: [dispatch] Acronym for the SIP-XMPP effort
> >
> >     Folks,
> >
> >     I am going to handle the chartering of the SIP-XMPP effort, as
> >     discussed in Orlando. As usual, we need to choose an acronym for the
> >     group. Among the acronyms I have heard, I like this one best so far:
> >
> >       SIXPAC - SIP Integration with XMPP in Presence Aware Contexts
> >
> >     While, per the RAI tradition, it refers to an alcoholic drink,
> >     usually beers, it also refers to the nice six packs all RAI
> >     contributors are working on during the spring in order to show off
> >     at the beach in summer :-)
> >
> >     Any other suggestions?
> >
> >     With respect to the chairs for this effort, we will be choosing them
> >     soon as well. So, if you are interested but have not volunteered yet
> >     per my other email (from yesterday), now it would be a great time to
> >     drop me a note.
> >
> >     Thanks,
> >
> >     Gonzalo
> >     _______________________________________________
> >     dispatch mailing list
> >     dispatch@ietf.org <mailto:dispatch@ietf.org>
> >     https://www.ietf.org/mailman/listinfo/dispatch
> >     _______________________________________________
> >     dispatch mailing list
> >     dispatch@ietf.org <mailto:dispatch@ietf.org>
> >     https://www.ietf.org/mailman/listinfo/dispatch
> >
> >
>
>_______________________________________________
>dispatch mailing list
>dispatch@ietf.org
>https://www.ietf.org/mailman/listinfo/dispatch


From Christian.Groves@nteczone.com  Tue Apr  9 17:03:09 2013
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Subject: Re: [dispatch] [clue] Fwd: conference roles
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4.6.5.2 RFC6501 also says "A <role> provides the context for the set of 
conference operations that a participant can perform." To me this is a 
semantic. I think the issue is that it presents a set of values that 
have no definition and says that policy is defined in the future. The 
description so sparse its hard to see what its actually used for. Policy 
could be set by the endpoint manufacturer, user, service provider, a 
standards body etc.

Regards, Christian


On 10/04/2013 12:15 AM, Mary Barnes wrote:
> Well, I had to join in the thread since it seems appropriate for Mary
> to respond to a thread with Peter & Paul ;)   I do not think that the
> CLUE usage is any different really than XCON conference package
> defined role -  in terms of the "roles" being both human readable and
> the semantics being unique to an application  and can be based on
> policies.  The latter was the XCON/conference intent as clearly stated
> in  section 4.6.5.2 of 6501. It's the right thing for SIP based
> conferencing to be using this information and since CLUE is based on
> SIP, we need to have these things consistent.
>
> Regards,
> Mary.
>
> On Tue, Apr 9, 2013 at 8:57 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>> Coming late to the discussion...
>>
>> I agree that what CLUE is considering is a different usage than what RFC
>> 4575 is talking about. But in both cases we are talking about attributes
>> that are applied to *something*. For CLUE it is Captures that we are
>> attaching attributes to while in RFC 4575 it is participants.
>>
>> If it makes sense, and I'm not yet sure it does in this case, we could have
>> a definition of "role" that can be used by CLUE to describe captures and is
>> also used in the conference event package to describe participants.
>>
>> My point is that the "name" of this attribute need not fully define what we
>> mean. Rather it is the name together with the thing it is applied to. So I
>> don't have a problem calling this "role". (But I also have no problem
>> calling it something else.)
>>
>> In CLUE we tried this once - with "content" based on reusing the values from
>> a=content. But then we decided that definition was too narrow and extending
>> it enough would have undesirable consequences. It may turn out that
>> sharing/reusing a definition for "role" is also problematic. But we can at
>> least investigate it.
>>
>>      Thanks,
>>      Paul
>>
>>
>>
>> On 4/9/13 1:50 AM, Peter Saint-Andre wrote:
>>> -----BEGIN PGP SIGNED MESSAGE-----
>>> Hash: SHA1
>>>
>>> On 4/7/13 11:43 PM, Christian Groves wrote:
>>>> Perhaps I should have kept this on the dispatch list?
>>>
>>> Perhaps. At the least, it would be good to have one place to talk
>>> about this topic.
>>>
>>>> On 8/04/2013 3:40 PM, Christian Groves wrote:
>>>>> Hello Peter,
>>>>>
>>>>> I had envisaged that from the CLUE perspective that an IANA
>>>>> registry would be establish for Role (and some of the
>>>>> attributes). I'm not sure that it is possible to harmonise
>>>>> RFC4575, RFC6501 and CLUE as I think they have different
>>>>> semantics.
>>>>>
>>>>> RFC4575 seems just to have a free text field that's human
>>>>> readable. The goal of the CLUE "role" attribute is to have a set
>>>>> of tags which are human readable and able to used in an automatic
>>>>> way.
>>>>>
>>>>> RFC6501 talks about "conference operations that a participant
>>>>> can perform" I think this is semantically different to what is
>>>>> being proposed in CLUE which is more about the person/s in a
>>>>> capture (i.e. those depicted in the RTP stream).
>>>>>
>>>>> If it helps lessen the confusion we could change the name of the
>>>>>    CLUE parameter from "role" to "Character", "Captured Persons" or
>>>>>    "Person represented by media", etc.? With the semantic that the
>>>>>    media contains a person who is acting in that character/role.
>>>
>>> Apparently I got confused by the use of the term "role" in
>>> draft-groves-clue-capture-attr because to me it seems like an
>>> extension to the roles defined in RFC 6501, which provided some new
>>> values for the same construct defined in RFC 4575. What you're saying
>>> is that the CLUE roles proposed in draft-groves-clue-capture-attr are
>>> actually a completely different construct. That is, RFC 4575 and RFC
>>> 6501 define roles as essentially bundles of permissions to perform
>>> certain operations, whereas the roles / characters in
>>> draft-groves-clue-capture-attr have nothing to do with conference
>>> control. Correct?
>>>
>>>>> I think part of the challenge will be to define the values. A
>>>>> person can have multiple roles/characters, e.g. a chairman who is
>>>>>    acting as a scribe. Some have a similar function but have a
>>>>> different connotation e.g. presenter versus lecturer. I think to
>>>>>    move forward we need to address these sorts of things. The
>>>>> previous RFCs seem to have largely ignored the problem in the
>>>>> final text.
>>>
>>> Agreed.
>>>
>>> Just FYI, in the XMPP community we had thought about the concept of
>>> "hats", which are roles you can play in a conference (and you can play
>>> multiple roles, just as you can wear different hats). We have a very
>>> basic spec for hats here:
>>>
>>> http://xmpp.org/extensions/xep-0317.html
>>>
>>> Rather than continuing to pursue the hats idea, I've been thinking
>>> that it might be better to settle on using conference-info syntax from
>>> RFC 4575 (since that would improve interoperability across SIP systems
>>> and XMPP systems -- there's nothing SIP-specific about conference-info
>>> itself). Unfortunately, it appears that the conference-info syntax is
>>> clear but the semantics are not. I'm happy to put some cycles into
>>> clarifying the semantics if there's interest.
>>>
>>> (Now it seems that is a separate topic from the CLUE work, since CLUE
>>> isn't actually using the RFC 4575 construct. So discussing this on the
>>> DISPATCH list seems correct.)
>>>
>>> Peter
>>>
>>> - --
>>> Peter Saint-Andre
>>> https://stpeter.im/
>>>
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>>> _______________________________________________
>>> dispatch mailing list
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>>>
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From dan@itsyscom.com  Wed Apr 10 01:28:23 2013
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Date: Wed, 10 Apr 2013 10:28:17 +0200
From: Dan Christian Bogos <dan@itsyscom.com>
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Subject: [dispatch] I-D Action: draft-ivov-xmpp-cusax-04.txt
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Dear All,

After reading your draft I would like to express here my interest in the 
integration you have proposed.
I have already used it in a couple of scenarios and, despite the latest 
trends in complicating things and analyzing all kind of network high 
availability scenarios ... it simply works.

If you find them useful, here are some quick thoughts from me on related:

  * In order to avoid disturbing the security of one protocol when 
combining it with the other, one should not attempt to unify the 
authentication mechanisms or the data driving them.
  * In real-life networks I have met the need of combining CUSAX with 
keeping support for features available complementary in both protocols, 
since the ideal case of dual-stack protocol implementation in the agents 
was quite rare. I am sure in the future these implementations will be 
more easy to find but, in order to transition existing networks towards 
CUSAX support, failover mechanisms still need to defined. One simple 
approach would be flag-ing "exceptions" on both server as well as CUSAX 
aware UAs.

Another useful feature I find by combining SIP and XMPP is improving 
ways to standardize interaction with SIP Proxies/B2BUAs through XMPP, as 
example of applicability being third-party-call-control by using non SIP 
means. I am not sure this is the domain targeted by CUSAX but still 
votes for SIP+XMPP interaction.

Once again, thanks for this!

DanB


With regards,

ITsysCOM, inh.
Dan Christian Bogos

Tiroler str. 9
D-83435 Bad Reichenhall
Germany
Tel.: +49 (0) 8651/7174963
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email: dan@itsyscom.com
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From stephane.cazeaux@orange.com  Fri Apr 12 14:07:35 2013
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References: <20130215171708.27730.58906.idtracker@ietfa.amsl.com> <CD442C6C.B156%vpascual@acmepacket.com> <CAGTXFp-Ee_XsD0NdmrFxjqyUN9cP-+Q2XNAKV35MMTtZJxUEcA@mail.gmail.com> <5126925E.2010102@alum.mit.edu> <FEE7A6136F518B4B87037A58924AB6B0067596FB@FTRDMB03.rd.francetelecom.fr> <51296D15.1030105@alum.mit.edu> <FEE7A6136F518B4B87037A58924AB6B00675A13A@FTRDMB03.rd.francetelecom.fr> <CAGTXFp_ScEvzdJXCBS5NtqOtvbXZ1Bo7XPLjB-BZns_JLPj8eQ@mail.gmail.com> <51348999.1000506@gmail.com> <CALiegf=J6sGG0qeyPx97U8frBycqKrdDm3mYJF_rpGvAi2z1TQ@mail.gmail.com> <D8CA0A9D-6BF2-4488-84A5-02252E93DC39@iii.ca> <CALiegfkhHFp47QtLwGvgS7BRW7K4oou3DHRmeEvoz+CTSrnOqw@mail.gmail.com> <513DDE61.7010605@nostrum.com> <CALiegfm3LzhFRiZxLaa0rzTJ+SVVh2dAJwQbTiieFYGdq6Lhuw@mail.gmail.com> <FEE7A6136F518B4B87037A58924AB6B00679C2F8@FTRDMB03.rd.francetelecom.fr>
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Cc: dispatch@ietf.org, Cullen Jennings <fluffy@iii.ca>
Subject: Re: [dispatch] New Version Notification for draft-pascual-dispatch-bfcp-websocket-00.txt
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Sure but, is it really friendly? Would somebody design such a binary
protocol from scratch to be handled by JS?

--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>
El 12/04/2013 23:07, <stephane.cazeaux@orange.com> escribi=C3=B3:

> Hi all
>
> I come back on the PS note below. It is possible to handle binary data in
> JS code using TypedArrays (
> http://www.khronos.org/registry/typedarray/specs/latest/). The Uint8Array
> typed array is appropriate for BFCP.
> It is already widely supported in browsers (http://caniuse.com/typedarray=
s
> ).
>
> St=C3=A9phane
>
>
> -----Message d'origine-----
> De : I=C3=B1aki Baz Castillo [mailto:ibc@aliax.net]
> Envoy=C3=A9 : lundi 11 mars 2013 15:28
> =C3=80 : Adam Roach
> Cc : Cullen Jennings; dispatch@ietf.org; CAZEAUX Stephane OLNC/OLPS
> Objet : Re: [dispatch] New Version Notification for
> draft-pascual-dispatch-bfcp-websocket-00.txt
>
> 2013/3/11 Adam Roach <adam@nostrum.com>:
> > All I think you'll see is that the browser -- just like any other
> > endpoint that doesn't support BFCP -- would decline the BFCP media
> section.
>
> Hopefully yes :)
>
> Then the problem is how the JS code *parses* and *mangles* the SDP to
> interact with the BFPC media section, which is hard to accomplish given t=
he
> SDP format... but that's another subject :)
>
> PS: Anyhow I would like to insist that, if BFPC means a "binary protocol"=
,
> then it will be really hard to implement it at JavaScript level.
>
>
> --
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>
>

--e89a8f5034aa36baee04da30e0f9
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<p dir=3D"ltr">Sure but, is it really friendly? Would somebody design such =
a binary protocol from scratch to be handled by JS?<br></p>
<p dir=3D"ltr">--<br>
I=C3=B1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;</p>
<div class=3D"gmail_quote">El 12/04/2013 23:07,  &lt;<a href=3D"mailto:step=
hane.cazeaux@orange.com">stephane.cazeaux@orange.com</a>&gt; escribi=C3=B3:=
<br type=3D"attribution"><blockquote class=3D"gmail_quote" style=3D"margin:=
0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi all<br>
<br>
I come back on the PS note below. It is possible to handle binary data in J=
S code using TypedArrays (<a href=3D"http://www.khronos.org/registry/typeda=
rray/specs/latest/" target=3D"_blank">http://www.khronos.org/registry/typed=
array/specs/latest/</a>). The Uint8Array typed array is appropriate for BFC=
P.<br>

It is already widely supported in browsers (<a href=3D"http://caniuse.com/t=
ypedarrays" target=3D"_blank">http://caniuse.com/typedarrays</a>).<br>
<br>
St=C3=A9phane<br>
<br>
<br>
-----Message d&#39;origine-----<br>
De=C2=A0: I=C3=B1aki Baz Castillo [mailto:<a href=3D"mailto:ibc@aliax.net">=
ibc@aliax.net</a>]<br>
Envoy=C3=A9=C2=A0: lundi 11 mars 2013 15:28<br>
=C3=80=C2=A0: Adam Roach<br>
Cc=C2=A0: Cullen Jennings; <a href=3D"mailto:dispatch@ietf.org">dispatch@ie=
tf.org</a>; CAZEAUX Stephane OLNC/OLPS<br>
Objet=C2=A0: Re: [dispatch] New Version Notification for draft-pascual-disp=
atch-bfcp-websocket-00.txt<br>
<br>
2013/3/11 Adam Roach &lt;<a href=3D"mailto:adam@nostrum.com">adam@nostrum.c=
om</a>&gt;:<br>
&gt; All I think you&#39;ll see is that the browser -- just like any other<=
br>
&gt; endpoint that doesn&#39;t support BFCP -- would decline the BFCP media=
 section.<br>
<br>
Hopefully yes :)<br>
<br>
Then the problem is how the JS code *parses* and *mangles* the SDP to inter=
act with the BFPC media section, which is hard to accomplish given the SDP =
format... but that&#39;s another subject :)<br>
<br>
PS: Anyhow I would like to insist that, if BFPC means a &quot;binary protoc=
ol&quot;, then it will be really hard to implement it at JavaScript level.<=
br>
<br>
<br>
--<br>
I=C3=B1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
</blockquote></div>

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Subject: Re: [dispatch] New Version Notification for draft-pascual-dispatch-bfcp-websocket-00.txt
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On 4/12/13 5:49 PM, Iñaki Baz Castillo wrote:
> Sure but, is it really friendly? Would somebody design such a binary
> protocol from scratch to be handled by JS?

No, probably not.
But since the protocol is already defined and implemented, it isn't 
unreasonable to expect JS to deal with it, rather than having everybody 
else adapt to JS limitations.

	Thanks,
	Paul

> --
> Iñaki Baz Castillo
> <ibc@aliax.net <mailto:ibc@aliax.net>>
>
> El 12/04/2013 23:07, <stephane.cazeaux@orange.com
> <mailto:stephane.cazeaux@orange.com>> escribió:
>
>     Hi all
>
>     I come back on the PS note below. It is possible to handle binary
>     data in JS code using TypedArrays
>     (http://www.khronos.org/registry/typedarray/specs/latest/). The
>     Uint8Array typed array is appropriate for BFCP.
>     It is already widely supported in browsers
>     (http://caniuse.com/typedarrays).
>
>     Stéphane
>
>
>     -----Message d'origine-----
>     De : Iñaki Baz Castillo [mailto:ibc@aliax.net <mailto:ibc@aliax.net>]
>     Envoyé : lundi 11 mars 2013 15:28
>     À : Adam Roach
>     Cc : Cullen Jennings; dispatch@ietf.org <mailto:dispatch@ietf.org>;
>     CAZEAUX Stephane OLNC/OLPS
>     Objet : Re: [dispatch] New Version Notification for
>     draft-pascual-dispatch-bfcp-websocket-00.txt
>
>     2013/3/11 Adam Roach <adam@nostrum.com <mailto:adam@nostrum.com>>:
>      > All I think you'll see is that the browser -- just like any other
>      > endpoint that doesn't support BFCP -- would decline the BFCP
>     media section.
>
>     Hopefully yes :)
>
>     Then the problem is how the JS code *parses* and *mangles* the SDP
>     to interact with the BFPC media section, which is hard to accomplish
>     given the SDP format... but that's another subject :)
>
>     PS: Anyhow I would like to insist that, if BFPC means a "binary
>     protocol", then it will be really hard to implement it at JavaScript
>     level.
>
>
>     --
>     Iñaki Baz Castillo
>     <ibc@aliax.net <mailto:ibc@aliax.net>>
>
>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 15 Apr 2013 13:01:18 +0200
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Subject: Re: [dispatch] FW: New Version Notification for draft-pascual-dispatch-bfcp-websocket-00.txt
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Section 6 has some typos:

- Missing dot at the end of paragraphs 1, 3 and 4.

- s/WS/WSS/g in the 4th paragraph.


2013/2/21 Victor Pascual Avila <victor.pascual.avila@gmail.com>:
> Hi all,
>
> We have submitted a new Internet draft to describe the WebSocket
> Protocol as a Transport for the Binary Floor Control Protocol (BFCP).
> We are looking forward to your review comments.
>
> http://tools.ietf.org/html/draft-pascual-dispatch-bfcp-websocket-00
>
> Best regards,
> -Victor
>
>
> On 2/15/13 6:17 PM, "internet-drafts@ietf.org" <internet-drafts@ietf.org>
> wrote:
>
>>
>>A new version of I-D, draft-pascual-dispatch-bfcp-websocket-00.txt
>>has been successfully submitted by Victor Pascual and posted to the
>>IETF repository.
>>
>>Filename:       draft-pascual-dispatch-bfcp-websocket
>>Revision:       00
>>Title:          The WebSocket Protocol as a Transport for the Binary Floo=
r
>>Control Protocol (BFCP)
>>Creation date:  2013-02-15
>>Group:          Individual Submission
>>Number of pages: 9
>>URL:
>>http://www.ietf.org/internet-drafts/draft-pascual-dispatch-bfcp-websocket=
-
>>00.txt
>>Status:
>>http://datatracker.ietf.org/doc/draft-pascual-dispatch-bfcp-websocket
>>Htmlized:
>>http://tools.ietf.org/html/draft-pascual-dispatch-bfcp-websocket-00
>>
>>
>>Abstract:
>>   The WebSocket protocol enables two-way realtime communication between
>>   clients and servers.  This document specifies a new WebSocket sub-
>>   protocol as a reliable transport mechanism between Binary Floor
>>   Control Protocol (BFCP) entities to enable usage of BFCP in new
>>   scenarios.  This document normatively updates
>>   [I-D.draft-ietf-bfcpbis-rfc4582bis] and
>>   [I-D.draft-ietf-bfcpbis-rfc4583bis]
>>
>>
>>
>>
>>
>>The IETF Secretariat
>>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch



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

From keith.drage@alcatel-lucent.com  Mon Apr 15 05:15:07 2013
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Mary Barnes <mary.ietf.barnes@gmail.com>
Thread-Topic: [dispatch] Use of BFCP or MSRP over datachannel or websocket
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Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or websocket
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It appears to me that we are fine in the area of non-WebRTC usage. We know =
what we are doing.

When it comes to webRTC usage, then we have two options. That implies to me=
 either this document needs to include considerations of why it might be us=
ed rather than the data channel versus webrtc, or it needs to be decided th=
at some other document should cover it - but webrtc owns no such document.

Dispatch to me needs to decide whether this document should have those cons=
iderations, or whether some other document should be provided.

Regards

Keith


> -----Original Message-----
> From: Mary Barnes [mailto:mary.ietf.barnes@gmail.com]
> Sent: 28 March 2013 14:58
> To: DRAGE, Keith (Keith)
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or websocket
>=20
> I don't think it's the job of dispatch or BFCPbis WG to define or even
> make recommendations for RTCWEB.  That choice is up to the RTCWEB WG
> and it shouldn't be influencing this work.  There are applications
> other than RTCWEB that can and will make use of BFCPbis over
> websockets.  I really don't understand why we are getting so hung up
> on this - it's the BFCP messages themselves that are important - how
> one chooses to transport is up to the application.
>=20
> Mary.
>=20
> On Thu, Mar 28, 2013 at 7:35 AM, DRAGE, Keith (Keith)
> <keith.drage@alcatel-lucent.com> wrote:
> > From what I have seen of the discussion so far, there appears to be a
> body of support that there is a use case for both BFCP and MSRP over
> websocket, for use in a non-rtcweb environment.
> >
> > If when we get to the rtcweb environment we allow this, and also allow =
a
> usage over datachannel version, we essentially have two solutions in the
> market place.
> >
> > If that occurs, what guidance will we give to a rtcweb developer as to
> which to implement?
> >
> > Or do we envisage that only one will be allowed (and if so which) in an
> rtcweb environment?
> >
> >
> >
> > Regards
> >
> > Keith
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 15 Apr 2013 14:27:08 +0200
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Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or websocket
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2013/4/15 DRAGE, Keith (Keith) <keith.drage@alcatel-lucent.com>:
> It appears to me that we are fine in the area of non-WebRTC usage. We kno=
w what we are doing.
>
> When it comes to webRTC usage, then we have two options. That implies to =
me either this document needs to include considerations of why it might be =
used rather than the data channel versus webrtc, or it needs to be decided =
that some other document should cover it - but webrtc owns no such document=
.
>
> Dispatch to me needs to decide whether this document should have those co=
nsiderations, or whether some other document should be provided.

I agree.

Just one consideration more: This draft is just intended for a BFCP
WebSocket Client to establish a BFCP session with a BFCP WebSocket
Server (terms defined within the drat [*]), thus the draft should
state that it's not possible for a BFCP WebSocket Client to establish
a session with another BFCP WebSocket Client or with any other BFCP
peer that is not a BFCP WebSocket Server (and given WebSocket nature I
don't expect at all that there will be BCFP nodes behaving as both a
WS client and a WS server).

Another option would be defining the concept of "BCFP relay server"
but IMHO that is out of the scope of this specification. May be
another document for this purpose?


Regards.


[*] http://tools.ietf.org/html/draft-pascual-dispatch-bfcp-websocket-00#sec=
tion-2.1


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

From gsalguei@cisco.com  Mon Apr 15 08:58:39 2013
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From: Gonzalo Salgueiro <gsalguei@cisco.com>
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Cc: dispatch@ietf.org
Subject: Re: [dispatch] Use of BFCP or MSRP over datachannel or websocket
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On Apr 15, 2013, at 8:27 AM, I=F1aki Baz Castillo <ibc@aliax.net> wrote:

> 2013/4/15 DRAGE, Keith (Keith) <keith.drage@alcatel-lucent.com>:
>> It appears to me that we are fine in the area of non-WebRTC usage. We =
know what we are doing.
>>=20
>> When it comes to webRTC usage, then we have two options. That implies =
to me either this document needs to include considerations of why it =
might be used rather than the data channel versus webrtc, or it needs to =
be decided that some other document should cover it - but webrtc owns no =
such document.
>>=20
>> Dispatch to me needs to decide whether this document should have =
those considerations, or whether some other document should be provided.
>=20
> I agree.

+1

Having a section in this document discussing typical use cases, scope, =
limitations and advantages/disadvantages of other similar  transport =
mechanisms seems entirely reasonable for this document.

> Just one consideration more: This draft is just intended for a BFCP
> WebSocket Client to establish a BFCP session with a BFCP WebSocket
> Server (terms defined within the drat [*]), thus the draft should
> state that it's not possible for a BFCP WebSocket Client to establish
> a session with another BFCP WebSocket Client or with any other BFCP
> peer that is not a BFCP WebSocket Server (and given WebSocket nature I
> don't expect at all that there will be BCFP nodes behaving as both a
> WS client and a WS server).
>=20
> Another option would be defining the concept of "BCFP relay server"
> but IMHO that is out of the scope of this specification. May be
> another document for this purpose?

I think this is a future consideration for another document.

Gonzalo

>=20
>=20
> Regards.
>=20
>=20
> [*] =
http://tools.ietf.org/html/draft-pascual-dispatch-bfcp-websocket-00#sectio=
n-2.1
>=20
>=20
> --
> I=F1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20


From aaron@thinkingphones.com  Mon Apr 15 14:17:00 2013
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From: Aaron Evans <aaron@thinkingphones.com>
To: "dispatch@ietf.org" <dispatch@ietf.org>
Thread-Topic: [dispatch] Fwd: I-D Action: draft-ivov-xmpp-cusax-04.txt
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Subject: Re: [dispatch] Fwd: I-D Action: draft-ivov-xmpp-cusax-04.txt
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Emil,

I read through the latest draft and I think it looks great.  =20

This document formalizes a lot of the concepts that we've been trying to im=
plement with our UCaaS offering.   I also see some other ideas in here that=
 I think we will consider integrating into our service.=20

-aaron

------------------------------------------------------------
Aaron Evans
VP, Engineering
P: 613.482.0179=A0 C: 613.355.7263=20
aaron@thinkingphones.com


-----Original Message-----
From: Emil Ivov [mailto:emcho@jitsi.org]=20
Sent: Friday, April 05, 2013 2:41 PM
To: Aaron Evans
Subject: Fwd: [dispatch] Fwd: I-D Action: draft-ivov-xmpp-cusax-04.txt

Hey Aaron,

I already mentioned our CUSAX document.

Since you guys have been actively using this model, could I please ask you =
to have a look at it and send us any feedback on the DISPATCH <dispatch@iet=
f.org> mailing list? It's relatively short and at this point even just sayi=
ng that you've read it and it makes sense to you is helpful.

Cheers,
Emil


-------- Original Message --------
Subject: [dispatch] Fwd: I-D Action: draft-ivov-xmpp-cusax-04.txt
Date: Thu, 04 Apr 2013 21:19:17 -0600
From: Peter Saint-Andre <stpeter@stpeter.im>
To: DISPATCH <dispatch@ietf.org>

FYI.


-------- Original Message --------
Subject: I-D Action: draft-ivov-xmpp-cusax-04.txt
Date: Thu, 04 Apr 2013 20:16:30 -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 director=
ies.


	Title           : CUSAX: Combined Use of the Session Initiation
Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP)
	Author(s)       : Emil Ivov
                          Peter Saint-Andre
                          Enrico Marocco
	Filename        : draft-ivov-xmpp-cusax-04.txt
	Pages           : 11
	Date            : 2013-04-04

Abstract:
   This document describes suggested practices for combined use of the
   Session Initiation Protocol (SIP) and the Extensible Messaging and
   Presence Protocol (XMPP).  Such practices aim to provide a single
   fully featured real-time communication service by using complementary
   subsets of features from each of the protocols.  Typically such
   subsets would include telephony capabilities from SIP and instant
   messaging and presence capabilities from XMPP.  This specification
   does not define any new protocols or syntax for either SIP or XMPP.
   However, implementing it may require modifying or at least
   reconfiguring existing client and server-side software.  Also, it is
   not the purpose of this document to make recommendations as to
   whether or not such combined use should be preferred to the
   mechanisms provided natively by each protocol (for example, SIP's
   SIMPLE or XMPP's Jingle).  It merely aims to provide guidance to
   those who are interested in such a combined use.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ivov-xmpp-cusax

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

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


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

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From oej@edvina.net  Sun Apr 21 04:25:39 2013
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Subject: [dispatch] Fwd: DANE SRV draft and SIP
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Hi!

Looking at DANE for SIP, I mailed out some thoughts on the DANE mailing =
list. If we come to
the conclusion that RFC 5922 will need to be updated - will this work be =
done in Dispatch
or Dane or somewhere else?

DANE also has impacts on SIP Identity as an alternative discovery =
mechanism for
domain certificates (replacing HTTPS).

/O

Vidarebefordrat brev:

> Fr=E5n: "Olle E. Johansson" <oej@edvina.net>
> =C4mne: DANE SRV draft and SIP
> Datum: 21 april 2013 12:27:47 CEST
> Till: dane WG list <dane@ietf.org>
> Kopia: "Olle E. Johansson" <oej@edvina.net>
>=20
> Hi!
>=20
> Looking at the DANE SRV draft with my SIP eyes I realize that there's =
a lot of work to be done to get this
> to work with SIP.=20
>=20
> SIP has no StartTLS, we use NAPTR records to select transport. The =
NAPTR points to a SRV
> name that resolves into a set of host names.
>=20
> The SIP Domain certificates RFC - http://tools.ietf.org/html/rfc5922 - =
specificies matching a=20
> given SIP URI with a certificate. The matching is done on service =
domain either with=20
> a ALT name URI, like sip:ietf.org or a DNS name, like ietf.org - but =
not the host name.
>=20
> I personally agree with the policy in the DANE SRV draft that we =
should match on the
> SRV hostname used to get A/AAAA records when using DNSsec. For this to =
work,
> the path from the SIP URI over NAPTR to SRV and hostnames needs to be =
fully
> signed in and verified in the DNS. This is going to require an update =
to 5922.
>=20
> The final question is how to handle this without SNI. The certificates =
for both
> DANE verification and RFC 5922 plus "old-style" verification with the =
sip
> domain in the CN seems like a complicated mess to manage.
>=20
> Food for thought on a sunny Sunday.
>=20
> The SRV draft is not clear on how to use Subject AltNames of various =
types,
> and doesn't mention NAPTR. I am not personally aware of other =
protocols using
> this setup, so maybe this requires a very SIP specfic draft, following =
the wake of the
> smtp work.
>=20
> /O
>=20


From emil@sip-communicator.org  Sun Apr 21 13:43:12 2013
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Date: Sun, 21 Apr 2013 22:43:07 +0200
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Subject: [dispatch] Announcing a MUC with RFC4575
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Hey all,

I have been looking at RFC4575 and wondering what the best way would be
for a mixer to announce that a conference has an associated chatroom
(e.g. an XMPP MUC) available at a certain address. This would allow UAs
that support MUCs can join there too.

>From what I see the most reasonable way to do this would be a
<conf-uri>, like this for example:

<conf-uris>
  <entry>
   <uri>sip:conf545@h323.example.com</uri>
   <display-text>TTI Bridge</display-text>
   <purpose>participation</purpose>
  </entry>
  <entry>
   <uri>xmpp:conf545@conference.example.com</uri>
   <display-text>Multi-User Chat 545</display-text>
   <purpose>impp-participation</purpose>
  </entry>
</conf-uris>

Another option would be to use the same <entry> but place it ina a
<service-uris> element instead:

<service-uris>
  <entry>
   <uri>http://sharepoint/salesgroup/</uri>
   <purpose>web-page</purpose>
  </entry>
  <entry>
   <uri>xmpp:conf545@conference.example.com</uri>
   <display-text>Multi-User Chat 545</display-text>
   <purpose>impp-participation</purpose>
  </entry>
</service-uris>

Does any of the above make sense? In case no other mechanism allows for
the same thing in a simpler way, then could we maybe put one of the two
on paper?

Emil
-- 
https://jitsi.org

From saul@ag-projects.com  Mon Apr 22 00:53:31 2013
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Subject: Re: [dispatch] Announcing a MUC with RFC4575
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Hi Emil!

On Apr 21, 2013, at 10:43 PM, Emil Ivov wrote:

> Hey all,
>=20
> I have been looking at RFC4575 and wondering what the best way would =
be
> for a mixer to announce that a conference has an associated chatroom
> (e.g. an XMPP MUC) available at a certain address. This would allow =
UAs
> that support MUCs can join there too.
>=20
> =46rom what I see the most reasonable way to do this would be a
> <conf-uri>, like this for example:
>=20
> <conf-uris>
>  <entry>
>   <uri>sip:conf545@h323.example.com</uri>
>   <display-text>TTI Bridge</display-text>
>   <purpose>participation</purpose>
>  </entry>
>  <entry>
>   <uri>xmpp:conf545@conference.example.com</uri>
>   <display-text>Multi-User Chat 545</display-text>
>   <purpose>impp-participation</purpose>
>  </entry>
> </conf-uris>
>=20

I think the above does make sense, now, would the same URI be usable =
with Jingle or is intended for MUC only?

> Another option would be to use the same <entry> but place it ina a
> <service-uris> element instead:
>=20
> <service-uris>
>  <entry>
>   <uri>http://sharepoint/salesgroup/</uri>
>   <purpose>web-page</purpose>
>  </entry>
>  <entry>
>   <uri>xmpp:conf545@conference.example.com</uri>
>   <display-text>Multi-User Chat 545</display-text>
>   <purpose>impp-participation</purpose>
>  </entry>
> </service-uris>
>=20
> Does any of the above make sense? In case no other mechanism allows =
for
> the same thing in a simpler way, then could we maybe put one of the =
two
> on paper?
>=20

This second approach would require a new entry in the IANA registry for =
that 'purpose' element right? In any case, this one could also make =
sense, I guess it depends on the scenario.

Assuming your scenario is Jitsi's use case, that is, the audio / video =
conference is held by some entity and the MUC by some other, IMHO this =
second approach makes more sense.


Regards,

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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From: "Shekh-Yusef, Rifaat (Rifaat)" <rifatyu@avaya.com>
To: =?iso-8859-1?Q?Sa=FAl_Ibarra_Corretg=E9?= <saul@ag-projects.com>, "Emil Ivov" <emcho@jitsi.org>
Thread-Topic: [dispatch] Announcing a MUC with RFC4575
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Subject: Re: [dispatch] Announcing a MUC with RFC4575
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Emil,

You may want to look at the following CCMP draft:
https://datatracker.ietf.org/doc/draft-yusef-dispatch-ccmp-indication/?incl=
ude_text=3D1

We considered the two options you mentioned below and decided that the serv=
ices-uris option is more appropriate.
As Sa=FAl Ibarra Corretg=E9 mentioned this requires IANA registration for t=
he new purpose value, but it can be done in the same document (see section =
4 of the document above).

Regards,
 Rifaat


> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Sa=FAl Ibarra Corretg=E9
> Sent: Monday, April 22, 2013 3:53 AM
> To: Emil Ivov
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] Announcing a MUC with RFC4575
>=20
> Hi Emil!
>=20
> On Apr 21, 2013, at 10:43 PM, Emil Ivov wrote:
>=20
> > Hey all,
> >
> > I have been looking at RFC4575 and wondering what the best way would
> be
> > for a mixer to announce that a conference has an associated chatroom
> > (e.g. an XMPP MUC) available at a certain address. This would allow
> UAs
> > that support MUCs can join there too.
> >
> > From what I see the most reasonable way to do this would be a
> > <conf-uri>, like this for example:
> >
> > <conf-uris>
> >  <entry>
> >   <uri>sip:conf545@h323.example.com</uri>
> >   <display-text>TTI Bridge</display-text>
> >   <purpose>participation</purpose>
> >  </entry>
> >  <entry>
> >   <uri>xmpp:conf545@conference.example.com</uri>
> >   <display-text>Multi-User Chat 545</display-text>
> >   <purpose>impp-participation</purpose>
> >  </entry>
> > </conf-uris>
> >
>=20
> I think the above does make sense, now, would the same URI be usable
> with Jingle or is intended for MUC only?
>=20
> > Another option would be to use the same <entry> but place it ina a
> > <service-uris> element instead:
> >
> > <service-uris>
> >  <entry>
> >   <uri>http://sharepoint/salesgroup/</uri>
> >   <purpose>web-page</purpose>
> >  </entry>
> >  <entry>
> >   <uri>xmpp:conf545@conference.example.com</uri>
> >   <display-text>Multi-User Chat 545</display-text>
> >   <purpose>impp-participation</purpose>
> >  </entry>
> > </service-uris>
> >
> > Does any of the above make sense? In case no other mechanism allows
> for
> > the same thing in a simpler way, then could we maybe put one of the
> two
> > on paper?
> >
>=20
> This second approach would require a new entry in the IANA registry for
> that 'purpose' element right? In any case, this one could also make
> sense, I guess it depends on the scenario.
>=20
> Assuming your scenario is Jitsi's use case, that is, the audio / video
> conference is held by some entity and the MUC by some other, IMHO this
> second approach makes more sense.
>=20
>=20
> Regards,
>=20
> --
> Sa=FAl Ibarra Corretg=E9
> AG Projects
>=20
>=20
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

From mary.ietf.barnes@gmail.com  Mon Apr 22 08:18:47 2013
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To: "Olle E. Johansson" <oej@edvina.net>
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Subject: Re: [dispatch] Fwd: DANE SRV draft and SIP
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Updates to 5922 should happen in the RAI area.  We can discuss on
DISPATCH the need for the updates and then decide exactly where the
work should be done.  I would think the work would be done in SIPCORE,
since 5922 updates RFC 3261.

Thanks,
Mary.

On Sun, Apr 21, 2013 at 6:25 AM, Olle E. Johansson <oej@edvina.net> wrote:
> Hi!
>
> Looking at DANE for SIP, I mailed out some thoughts on the DANE mailing l=
ist. If we come to
> the conclusion that RFC 5922 will need to be updated - will this work be =
done in Dispatch
> or Dane or somewhere else?
>
> DANE also has impacts on SIP Identity as an alternative discovery mechani=
sm for
> domain certificates (replacing HTTPS).
>
> /O
>
> Vidarebefordrat brev:
>
>> Fr=E5n: "Olle E. Johansson" <oej@edvina.net>
>> =C4mne: DANE SRV draft and SIP
>> Datum: 21 april 2013 12:27:47 CEST
>> Till: dane WG list <dane@ietf.org>
>> Kopia: "Olle E. Johansson" <oej@edvina.net>
>>
>> Hi!
>>
>> Looking at the DANE SRV draft with my SIP eyes I realize that there's a =
lot of work to be done to get this
>> to work with SIP.
>>
>> SIP has no StartTLS, we use NAPTR records to select transport. The NAPTR=
 points to a SRV
>> name that resolves into a set of host names.
>>
>> The SIP Domain certificates RFC - http://tools.ietf.org/html/rfc5922 - s=
pecificies matching a
>> given SIP URI with a certificate. The matching is done on service domain=
 either with
>> a ALT name URI, like sip:ietf.org or a DNS name, like ietf.org - but not=
 the host name.
>>
>> I personally agree with the policy in the DANE SRV draft that we should =
match on the
>> SRV hostname used to get A/AAAA records when using DNSsec. For this to w=
ork,
>> the path from the SIP URI over NAPTR to SRV and hostnames needs to be fu=
lly
>> signed in and verified in the DNS. This is going to require an update to=
 5922.
>>
>> The final question is how to handle this without SNI. The certificates f=
or both
>> DANE verification and RFC 5922 plus "old-style" verification with the si=
p
>> domain in the CN seems like a complicated mess to manage.
>>
>> Food for thought on a sunny Sunday.
>>
>> The SRV draft is not clear on how to use Subject AltNames of various typ=
es,
>> and doesn't mention NAPTR. I am not personally aware of other protocols =
using
>> this setup, so maybe this requires a very SIP specfic draft, following t=
he wake of the
>> smtp work.
>>
>> /O
>>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

From oej@edvina.net  Mon Apr 22 08:34:11 2013
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Subject: Re: [dispatch] Fwd: DANE SRV draft and SIP
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22 apr 2013 kl. 17:18 skrev Mary Barnes <mary.ietf.barnes@gmail.com>:

> Updates to 5922 should happen in the RAI area.  We can discuss on
> DISPATCH the need for the updates and then decide exactly where the
> work should be done.  I would think the work would be done in SIPCORE,
> since 5922 updates RFC 3261.

A very important change is the SIP URI to certificate matching.
In SIP, we have specified in RFC 5922 that the certificate should match
the requested domain, not the host name choosen by the RFC 3263 process.

In the current DANE drafts for SRV records the matching is done by
the host name choosen as a result in the SRV lookups. This requires
a validated trusted chain from the NAPTR down to the SRV and
A/AAAA lookup.

I can see the point of not having to get new or updated certificates for
every domain added to a hosted service.=20


5922 states the following:

SIP domain identity:  An identity (e.g., "sip:example.com") contained
      in an X.509 certificate bound to a subject that identifies the
      subject as an authoritative SIP server for a domain.

With DNSsec, the signed DNS zone binds a domain to a server,
and the X.509 certificates identifies the server, but doesn't
say anything about being authoritative for a domain.

Validating certs by host name and binding authority in DNS=20
is a normative change to 5922.

I haven't spent any bandwidth on considering SIP identity yet,
but it is applicable there too.

/O

>=20
> Thanks,
> Mary.
>=20
> On Sun, Apr 21, 2013 at 6:25 AM, Olle E. Johansson <oej@edvina.net> =
wrote:
>> Hi!
>>=20
>> Looking at DANE for SIP, I mailed out some thoughts on the DANE =
mailing list. If we come to
>> the conclusion that RFC 5922 will need to be updated - will this work =
be done in Dispatch
>> or Dane or somewhere else?
>>=20
>> DANE also has impacts on SIP Identity as an alternative discovery =
mechanism for
>> domain certificates (replacing HTTPS).
>>=20
>> /O
>>=20
>> Vidarebefordrat brev:
>>=20
>>> Fr=E5n: "Olle E. Johansson" <oej@edvina.net>
>>> =C4mne: DANE SRV draft and SIP
>>> Datum: 21 april 2013 12:27:47 CEST
>>> Till: dane WG list <dane@ietf.org>
>>> Kopia: "Olle E. Johansson" <oej@edvina.net>
>>>=20
>>> Hi!
>>>=20
>>> Looking at the DANE SRV draft with my SIP eyes I realize that =
there's a lot of work to be done to get this
>>> to work with SIP.
>>>=20
>>> SIP has no StartTLS, we use NAPTR records to select transport. The =
NAPTR points to a SRV
>>> name that resolves into a set of host names.
>>>=20
>>> The SIP Domain certificates RFC - http://tools.ietf.org/html/rfc5922 =
- specificies matching a
>>> given SIP URI with a certificate. The matching is done on service =
domain either with
>>> a ALT name URI, like sip:ietf.org or a DNS name, like ietf.org - but =
not the host name.
>>>=20
>>> I personally agree with the policy in the DANE SRV draft that we =
should match on the
>>> SRV hostname used to get A/AAAA records when using DNSsec. For this =
to work,
>>> the path from the SIP URI over NAPTR to SRV and hostnames needs to =
be fully
>>> signed in and verified in the DNS. This is going to require an =
update to 5922.
>>>=20
>>> The final question is how to handle this without SNI. The =
certificates for both
>>> DANE verification and RFC 5922 plus "old-style" verification with =
the sip
>>> domain in the CN seems like a complicated mess to manage.
>>>=20
>>> Food for thought on a sunny Sunday.
>>>=20
>>> The SRV draft is not clear on how to use Subject AltNames of various =
types,
>>> and doesn't mention NAPTR. I am not personally aware of other =
protocols using
>>> this setup, so maybe this requires a very SIP specfic draft, =
following the wake of the
>>> smtp work.
>>>=20
>>> /O
>>>=20
>>=20
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch


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Cc: "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] Fwd: DANE SRV draft and SIP
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On 04/22/2013 10:34 AM, Olle E. Johansson wrote:
> A very important change is the SIP URI to certificate matching.
> In SIP, we have specified in RFC 5922 that the certificate should match
> the requested domain, not the host name choosen by the RFC 3263 process.

Olle: Indeed.  I think we have to be careful here before changing
rfc5922.  The mechanism in rfc5922 was designed specifically to
validate domains, not individual hosts.  More inline (at the end).

> In the current DANE drafts for SRV records the matching is done by
> the host name choosen as a result in the SRV lookups. This requires
> a validated trusted chain from the NAPTR down to the SRV and
> A/AAAA lookup.
>
> I can see the point of not having to get new or updated certificates for
> every domain added to a hosted service.
>
> 5922 states the following:
>
> SIP domain identity:  An identity (e.g., "sip:example.com") contained
>        in an X.509 certificate bound to a subject that identifies the
>        subject as an authoritative SIP server for a domain.
>
> With DNSsec, the signed DNS zone binds a domain to a server,
> and the X.509 certificates identifies the server, but doesn't
> say anything about being authoritative for a domain.
>
> Validating certs by host name and binding authority in DNS
> is a normative change to 5922.

I would like to understand DANE's approach better before we
proceed to modify rfc5922.  Unfortunately, I am swamped for the next
couple of weeks, but I will like to spare some time after that to
understand better your motivation.

Thanks,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60563 (USA)
Email: vkg@{bell-labs.com,acm.org} / vijay.gurbani@alcatel-lucent.com
Web:   http://ect.bell-labs.com/who/vkg/

From keith.drage@alcatel-lucent.com  Mon Apr 22 09:42:58 2013
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Mary Barnes <mary.ietf.barnes@gmail.com>, "Olle E. Johansson" <oej@edvina.net>
Thread-Topic: [dispatch] Fwd: DANE SRV draft and SIP
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Subject: Re: [dispatch] Fwd: DANE SRV draft and SIP
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While there was never a formal mailing list poll (because we never got to t=
he point of needing a 3261bis), I think there was a considerable body of op=
inion during the development of domain-certs that the material would form p=
art of any 3261bis work. That I think lends support to any further work bei=
ng done in sipcore.

I do suggest you look back in the archives for the mailing list discussion =
on domain-certs. You'll find it on the sip (not sipcore) mailing list archi=
ve. You'll find the WG discussion between Feb 2008 and April 2009 with the =
IESG approval discussion continuing until May 2010.

Regards

Keith

> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Mary Barnes
> Sent: 22 April 2013 16:19
> To: Olle E. Johansson
> Cc: dispatch@ietf.org list
> Subject: Re: [dispatch] Fwd: DANE SRV draft and SIP
>=20
> Updates to 5922 should happen in the RAI area.  We can discuss on
> DISPATCH the need for the updates and then decide exactly where the
> work should be done.  I would think the work would be done in SIPCORE,
> since 5922 updates RFC 3261.
>=20
> Thanks,
> Mary.
>=20
> On Sun, Apr 21, 2013 at 6:25 AM, Olle E. Johansson <oej@edvina.net> wrote=
:
> > Hi!
> >
> > Looking at DANE for SIP, I mailed out some thoughts on the DANE mailing
> list. If we come to
> > the conclusion that RFC 5922 will need to be updated - will this work b=
e
> done in Dispatch
> > or Dane or somewhere else?
> >
> > DANE also has impacts on SIP Identity as an alternative discovery
> mechanism for
> > domain certificates (replacing HTTPS).
> >
> > /O
> >
> > Vidarebefordrat brev:
> >
> >> Fr=E5n: "Olle E. Johansson" <oej@edvina.net>
> >> =C4mne: DANE SRV draft and SIP
> >> Datum: 21 april 2013 12:27:47 CEST
> >> Till: dane WG list <dane@ietf.org>
> >> Kopia: "Olle E. Johansson" <oej@edvina.net>
> >>
> >> Hi!
> >>
> >> Looking at the DANE SRV draft with my SIP eyes I realize that there's =
a
> lot of work to be done to get this
> >> to work with SIP.
> >>
> >> SIP has no StartTLS, we use NAPTR records to select transport. The
> NAPTR points to a SRV
> >> name that resolves into a set of host names.
> >>
> >> The SIP Domain certificates RFC - http://tools.ietf.org/html/rfc5922 -
> specificies matching a
> >> given SIP URI with a certificate. The matching is done on service
> domain either with
> >> a ALT name URI, like sip:ietf.org or a DNS name, like ietf.org - but
> not the host name.
> >>
> >> I personally agree with the policy in the DANE SRV draft that we shoul=
d
> match on the
> >> SRV hostname used to get A/AAAA records when using DNSsec. For this to
> work,
> >> the path from the SIP URI over NAPTR to SRV and hostnames needs to be
> fully
> >> signed in and verified in the DNS. This is going to require an update
> to 5922.
> >>
> >> The final question is how to handle this without SNI. The certificates
> for both
> >> DANE verification and RFC 5922 plus "old-style" verification with the
> sip
> >> domain in the CN seems like a complicated mess to manage.
> >>
> >> Food for thought on a sunny Sunday.
> >>
> >> The SRV draft is not clear on how to use Subject AltNames of various
> types,
> >> and doesn't mention NAPTR. I am not personally aware of other protocol=
s
> using
> >> this setup, so maybe this requires a very SIP specfic draft, following
> the wake of the
> >> smtp work.
> >>
> >> /O
> >>
> >
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

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

On 4/22/13 9:34 AM, Olle E. Johansson wrote:
> 
> 22 apr 2013 kl. 17:18 skrev Mary Barnes
> <mary.ietf.barnes@gmail.com>:
> 
>> Updates to 5922 should happen in the RAI area.  We can discuss
>> on DISPATCH the need for the updates and then decide exactly
>> where the work should be done.  I would think the work would be
>> done in SIPCORE, since 5922 updates RFC 3261.
> 
> A very important change is the SIP URI to certificate matching. In
> SIP, we have specified in RFC 5922 that the certificate should
> match the requested domain, not the host name choosen by the RFC
> 3263 process.
> 
> In the current DANE drafts for SRV records the matching is done by 
> the host name choosen as a result in the SRV lookups. This
> requires a validated trusted chain from the NAPTR down to the SRV
> and A/AAAA lookup.
> 
> I can see the point of not having to get new or updated
> certificates for every domain added to a hosted service.

Yes, this is a huge problem for multi-tenanted services. We've been
thinking about this for a while in the XMPP world. I suggest you have
a look at http://datatracker.ietf.org/doc/draft-ietf-xmpp-dna/ (also
in the RAI area, of course!) for some related considerations. XMPP too
uses SRV (not NAPTR) but draft-dane-srv doesn't quite meet the needs
of XMPP, either (Matt Miller and I are providing feedback to the DANE
WG as needed).

Peter


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Subject: Re: [dispatch] DANE SRV draft and SIP
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22 apr 2013 kl. 18:42 skrev "DRAGE, Keith (Keith)" =
<keith.drage@alcatel-lucent.com>:

> While there was never a formal mailing list poll (because we never got =
to the point of needing a 3261bis), I think there was a considerable =
body of opinion during the development of domain-certs that the material =
would form part of any 3261bis work. That I think lends support to any =
further work being done in sipcore.
>=20
> I do suggest you look back in the archives for the mailing list =
discussion on domain-certs. You'll find it on the sip (not sipcore) =
mailing list archive. You'll find the WG discussion between Feb 2008 and =
April 2009 with the IESG approval discussion continuing until May 2010.

Keith,
Thank you for the reference.

Note that I'm not saying that  RFC 5922 is wrong. The issue at hand is =
that the DANE groups current RFC
suggests a solution not compatible with 5922. We need to decide which =
way to go.

We could recommend that the DANE way is used when DNSsec and DANE =
validation is possible,
and keep RFC 5922 for other cases. Or update the recommendation in 5922 =
to make sip with TLS
better in regards to hosting larger amounts of domains.

In any case, I think we need to do something to support DANE in SIP.

Regards,

/Olle
>=20
> Regards
>=20
> Keith
>=20
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>> Behalf Of Mary Barnes
>> Sent: 22 April 2013 16:19
>> To: Olle E. Johansson
>> Cc: dispatch@ietf.org list
>> Subject: Re: [dispatch] Fwd: DANE SRV draft and SIP
>>=20
>> Updates to 5922 should happen in the RAI area.  We can discuss on
>> DISPATCH the need for the updates and then decide exactly where the
>> work should be done.  I would think the work would be done in =
SIPCORE,
>> since 5922 updates RFC 3261.
>>=20
>> Thanks,
>> Mary.
>>=20
>> On Sun, Apr 21, 2013 at 6:25 AM, Olle E. Johansson <oej@edvina.net> =
wrote:
>>> Hi!
>>>=20
>>> Looking at DANE for SIP, I mailed out some thoughts on the DANE =
mailing
>> list. If we come to
>>> the conclusion that RFC 5922 will need to be updated - will this =
work be
>> done in Dispatch
>>> or Dane or somewhere else?
>>>=20
>>> DANE also has impacts on SIP Identity as an alternative discovery
>> mechanism for
>>> domain certificates (replacing HTTPS).
>>>=20
>>> /O
>>>=20
>>> Vidarebefordrat brev:
>>>=20
>>>> Fr=E5n: "Olle E. Johansson" <oej@edvina.net>
>>>> =C4mne: DANE SRV draft and SIP
>>>> Datum: 21 april 2013 12:27:47 CEST
>>>> Till: dane WG list <dane@ietf.org>
>>>> Kopia: "Olle E. Johansson" <oej@edvina.net>
>>>>=20
>>>> Hi!
>>>>=20
>>>> Looking at the DANE SRV draft with my SIP eyes I realize that =
there's a
>> lot of work to be done to get this
>>>> to work with SIP.
>>>>=20
>>>> SIP has no StartTLS, we use NAPTR records to select transport. The
>> NAPTR points to a SRV
>>>> name that resolves into a set of host names.
>>>>=20
>>>> The SIP Domain certificates RFC - =
http://tools.ietf.org/html/rfc5922 -
>> specificies matching a
>>>> given SIP URI with a certificate. The matching is done on service
>> domain either with
>>>> a ALT name URI, like sip:ietf.org or a DNS name, like ietf.org - =
but
>> not the host name.
>>>>=20
>>>> I personally agree with the policy in the DANE SRV draft that we =
should
>> match on the
>>>> SRV hostname used to get A/AAAA records when using DNSsec. For this =
to
>> work,
>>>> the path from the SIP URI over NAPTR to SRV and hostnames needs to =
be
>> fully
>>>> signed in and verified in the DNS. This is going to require an =
update
>> to 5922.
>>>>=20
>>>> The final question is how to handle this without SNI. The =
certificates
>> for both
>>>> DANE verification and RFC 5922 plus "old-style" verification with =
the
>> sip
>>>> domain in the CN seems like a complicated mess to manage.
>>>>=20
>>>> Food for thought on a sunny Sunday.
>>>>=20
>>>> The SRV draft is not clear on how to use Subject AltNames of =
various
>> types,
>>>> and doesn't mention NAPTR. I am not personally aware of other =
protocols
>> using
>>>> this setup, so maybe this requires a very SIP specfic draft, =
following
>> the wake of the
>>>> smtp work.
>>>>=20
>>>> /O
>>>>=20
>>>=20
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch


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

On 4/22/13 12:18 PM, Olle E. Johansson wrote:
> 
> 22 apr 2013 kl. 18:42 skrev "DRAGE, Keith (Keith)"
> <keith.drage@alcatel-lucent.com>:
> 
>> While there was never a formal mailing list poll (because we
>> never got to the point of needing a 3261bis), I think there was a
>> considerable body of opinion during the development of
>> domain-certs that the material would form part of any 3261bis
>> work. That I think lends support to any further work being done
>> in sipcore.
>> 
>> I do suggest you look back in the archives for the mailing list
>> discussion on domain-certs. You'll find it on the sip (not
>> sipcore) mailing list archive. You'll find the WG discussion
>> between Feb 2008 and April 2009 with the IESG approval discussion
>> continuing until May 2010.
> 
> Keith, Thank you for the reference.
> 
> Note that I'm not saying that  RFC 5922 is wrong. The issue at hand
> is that the DANE groups current RFC suggests a solution not
> compatible with 5922. We need to decide which way to go.
> 
> We could recommend that the DANE way is used when DNSsec and DANE
> validation is possible, and keep RFC 5922 for other cases.

Well, it's clear to me that we wouldn't allow checking of the derived
domain (in RFC 6125 terms) unless DNSSEC validation succeeds. Since
that is currently a rare event, we'd just continue to do what RFC 5922
says, which in draft-ietf-xmpp-dna we call the PKI prooftype.

> Or update the recommendation in 5922 to make sip with TLS better in
> regards to hosting larger amounts of domains.

I think that is somewhat a separate issue from defining the DANE
prooftype, because other prooftypes might be possible or more
deployable in the short term, such as the POSH prooftype that Matt
Miller and I have defined in draft-miller-xmpp-posh-prooftype (but
which is not specific to XMPP).

Peter

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From oej@edvina.net  Mon Apr 22 12:18:38 2013
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Subject: Re: [dispatch] DANE SRV draft and SIP
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22 apr 2013 kl. 20:28 skrev Peter Saint-Andre <stpeter@stpeter.im>:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
> 
> On 4/22/13 12:18 PM, Olle E. Johansson wrote:
>> 
>> 22 apr 2013 kl. 18:42 skrev "DRAGE, Keith (Keith)"
>> <keith.drage@alcatel-lucent.com>:
>> 
>>> While there was never a formal mailing list poll (because we
>>> never got to the point of needing a 3261bis), I think there was a
>>> considerable body of opinion during the development of
>>> domain-certs that the material would form part of any 3261bis
>>> work. That I think lends support to any further work being done
>>> in sipcore.
>>> 
>>> I do suggest you look back in the archives for the mailing list
>>> discussion on domain-certs. You'll find it on the sip (not
>>> sipcore) mailing list archive. You'll find the WG discussion
>>> between Feb 2008 and April 2009 with the IESG approval discussion
>>> continuing until May 2010.
>> 
>> Keith, Thank you for the reference.
>> 
>> Note that I'm not saying that  RFC 5922 is wrong. The issue at hand
>> is that the DANE groups current RFC suggests a solution not
>> compatible with 5922. We need to decide which way to go.
>> 
>> We could recommend that the DANE way is used when DNSsec and DANE
>> validation is possible, and keep RFC 5922 for other cases.
> 
> Well, it's clear to me that we wouldn't allow checking of the derived
> domain (in RFC 6125 terms) unless DNSSEC validation succeeds. Since
> that is currently a rare event, we'd just continue to do what RFC 5922
> says, which in draft-ietf-xmpp-dna we call the PKI prooftype.
I'm not sure that the certificates would be compatible, which is a problem.
A non-DNSsec client following RFC 5922 requires one certificate with domain
names and the DNSsec/DANE client would require host names in the
same certificate - unless we have certificate selection (TNI).

Being tricky one could possibly have the host name in the CN
and add SIP domain URIs in subj alt names.

> 
>> Or update the recommendation in 5922 to make sip with TLS better in
>> regards to hosting larger amounts of domains.
> 
> I think that is somewhat a separate issue from defining the DANE
> prooftype, because other prooftypes might be possible or more
> deployable in the short term, such as the POSH prooftype that Matt
> Miller and I have defined in draft-miller-xmpp-posh-prooftype (but
> which is not specific to XMPP).
I need to read up on your drafts, but how do you handle certificates
for one server supporting multiple prooftypes? Are they compatible
in XMPP?

/O
> 
> Peter
> 
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Date: Mon, 22 Apr 2013 21:22:50 +0200
From: Emil Ivov <emcho@jitsi.org>
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Cc: "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] Announcing a MUC with RFC4575
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Hey Rifaat, Sa=FAl

On 22.04.13, 15:29, Shekh-Yusef, Rifaat (Rifaat) wrote:
> Emil,
>=20
> You may want to look at the following CCMP draft:
> https://datatracker.ietf.org/doc/draft-yusef-dispatch-ccmp-indication/?=
include_text=3D1
>=20
> We considered the two options you mentioned below and decided that the =
services-uris option is more appropriate.
> As Sa=FAl Ibarra Corretg=E9 mentioned this requires IANA registration f=
or the new purpose value, but it can be done in the same document (see se=
ction 4 of the document above).

Just to make sure I understand: are you suggesting that you could add an
impp purpose for service-uris in the above document?

If so, then this certainly works for me!

Thanks!
Emil

>=20
> Regards,
>  Rifaat
>=20
>=20
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>> Behalf Of Sa=FAl Ibarra Corretg=E9
>> Sent: Monday, April 22, 2013 3:53 AM
>> To: Emil Ivov
>> Cc: dispatch@ietf.org
>> Subject: Re: [dispatch] Announcing a MUC with RFC4575
>>
>> Hi Emil!
>>
>> On Apr 21, 2013, at 10:43 PM, Emil Ivov wrote:
>>
>>> Hey all,
>>>
>>> I have been looking at RFC4575 and wondering what the best way would
>> be
>>> for a mixer to announce that a conference has an associated chatroom
>>> (e.g. an XMPP MUC) available at a certain address. This would allow
>> UAs
>>> that support MUCs can join there too.
>>>
>>> From what I see the most reasonable way to do this would be a
>>> <conf-uri>, like this for example:
>>>
>>> <conf-uris>
>>>  <entry>
>>>   <uri>sip:conf545@h323.example.com</uri>
>>>   <display-text>TTI Bridge</display-text>
>>>   <purpose>participation</purpose>
>>>  </entry>
>>>  <entry>
>>>   <uri>xmpp:conf545@conference.example.com</uri>
>>>   <display-text>Multi-User Chat 545</display-text>
>>>   <purpose>impp-participation</purpose>
>>>  </entry>
>>> </conf-uris>
>>>
>>
>> I think the above does make sense, now, would the same URI be usable
>> with Jingle or is intended for MUC only?
>>
>>> Another option would be to use the same <entry> but place it ina a
>>> <service-uris> element instead:
>>>
>>> <service-uris>
>>>  <entry>
>>>   <uri>http://sharepoint/salesgroup/</uri>
>>>   <purpose>web-page</purpose>
>>>  </entry>
>>>  <entry>
>>>   <uri>xmpp:conf545@conference.example.com</uri>
>>>   <display-text>Multi-User Chat 545</display-text>
>>>   <purpose>impp-participation</purpose>
>>>  </entry>
>>> </service-uris>
>>>
>>> Does any of the above make sense? In case no other mechanism allows
>> for
>>> the same thing in a simpler way, then could we maybe put one of the
>> two
>>> on paper?
>>>
>>
>> This second approach would require a new entry in the IANA registry fo=
r
>> that 'purpose' element right? In any case, this one could also make
>> sense, I guess it depends on the scenario.
>>
>> Assuming your scenario is Jitsi's use case, that is, the audio / video=

>> conference is held by some entity and the MUC by some other, IMHO this=

>> second approach makes more sense.
>>
>>
>> Regards,
>>
>> --
>> Sa=FAl Ibarra Corretg=E9
>> AG Projects
>>
>>
>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

--=20
https://jitsi.org


From stpeter@stpeter.im  Mon Apr 22 12:32:41 2013
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Date: Mon, 22 Apr 2013 13:32:39 -0600
From: Peter Saint-Andre <stpeter@stpeter.im>
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Cc: "dispatch@ietf.org list" <dispatch@ietf.org>, mamille2@cisco.com
Subject: Re: [dispatch] DANE SRV draft and SIP
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Hash: SHA1

On 4/22/13 1:18 PM, Olle E. Johansson wrote:
> 
> 22 apr 2013 kl. 20:28 skrev Peter Saint-Andre
> <stpeter@stpeter.im>:
> 
> On 4/22/13 12:18 PM, Olle E. Johansson wrote:
>>>> 
>>>> 22 apr 2013 kl. 18:42 skrev "DRAGE, Keith (Keith)" 
>>>> <keith.drage@alcatel-lucent.com>:
>>>> 
>>>>> While there was never a formal mailing list poll (because
>>>>> we never got to the point of needing a 3261bis), I think
>>>>> there was a considerable body of opinion during the
>>>>> development of domain-certs that the material would form
>>>>> part of any 3261bis work. That I think lends support to any
>>>>> further work being done in sipcore.
>>>>> 
>>>>> I do suggest you look back in the archives for the mailing
>>>>> list discussion on domain-certs. You'll find it on the sip
>>>>> (not sipcore) mailing list archive. You'll find the WG
>>>>> discussion between Feb 2008 and April 2009 with the IESG
>>>>> approval discussion continuing until May 2010.
>>>> 
>>>> Keith, Thank you for the reference.
>>>> 
>>>> Note that I'm not saying that  RFC 5922 is wrong. The issue
>>>> at hand is that the DANE groups current RFC suggests a
>>>> solution not compatible with 5922. We need to decide which
>>>> way to go.
>>>> 
>>>> We could recommend that the DANE way is used when DNSsec and
>>>> DANE validation is possible, and keep RFC 5922 for other
>>>> cases.
> 
> Well, it's clear to me that we wouldn't allow checking of the
> derived domain (in RFC 6125 terms) unless DNSSEC validation
> succeeds. Since that is currently a rare event, we'd just continue
> to do what RFC 5922 says, which in draft-ietf-xmpp-dna we call the
> PKI prooftype.
>> I'm not sure that the certificates would be compatible, which is
>> a problem. A non-DNSsec client following RFC 5922 requires one
>> certificate with domain names and the DNSsec/DANE client would
>> require host names in the same certificate - unless we have
>> certificate selection (TNI).
> 
>> Being tricky one could possibly have the host name in the CN and
>> add SIP domain URIs in subj alt names.

In practice, large hosting providers either (1) don't offer
certificates at all or (2) offer a certificate with the hostname /
derived domain instead of the source domain. In the case of (2), this
forces the client developer to offer a special registration method or,
even worse, forces the user to override security settings (so that,
for example, everyone just "knows" that gmail.com users are actually
going to be presented with a certificate for talk.google.com). Both of
those are bad outcomes.

>>>> Or update the recommendation in 5922 to make sip with TLS
>>>> better in regards to hosting larger amounts of domains.
> 
> I think that is somewhat a separate issue from defining the DANE 
> prooftype, because other prooftypes might be possible or more 
> deployable in the short term, such as the POSH prooftype that Matt 
> Miller and I have defined in draft-miller-xmpp-posh-prooftype (but 
> which is not specific to XMPP).
>> I need to read up on your drafts, but how do you handle
>> certificates for one server supporting multiple prooftypes? Are
>> they compatible in XMPP?

I see no reason why they can't be.

Peter
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From: "Shekh-Yusef, Rifaat (Rifaat)" <rifatyu@avaya.com>
To: Emil Ivov <emcho@jitsi.org>
Thread-Topic: [dispatch] Announcing a MUC with RFC4575
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Subject: Re: [dispatch] Announcing a MUC with RFC4575
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> Just to make sure I understand: are you suggesting that you could add an
> impp purpose for service-uris in the above document?
>=20
> If so, then this certainly works for me!

No, that is not what I am suggesting.
What I am saying is that when you create your document to define your impp,=
 you can register the new purpose value in the same document in the IANA se=
ction, similar to what we did in section 4 of our CCMP document.

Regards,
 Rifaat




> -----Original Message-----
> From: Emil Ivov [mailto:emcho@jitsi.org]
> Sent: Monday, April 22, 2013 3:23 PM
> To: Shekh-Yusef, Rifaat (Rifaat)
> Cc: Sa=FAl Ibarra Corretg=E9; dispatch@ietf.org
> Subject: Re: [dispatch] Announcing a MUC with RFC4575
>=20
> Hey Rifaat, Sa=FAl
>=20
> On 22.04.13, 15:29, Shekh-Yusef, Rifaat (Rifaat) wrote:
> > Emil,
> >
> > You may want to look at the following CCMP draft:
> > https://datatracker.ietf.org/doc/draft-yusef-dispatch-ccmp-
> indication/?include_text=3D1
> >
> > We considered the two options you mentioned below and decided that the
> services-uris option is more appropriate.
> > As Sa=FAl Ibarra Corretg=E9 mentioned this requires IANA registration f=
or
> the new purpose value, but it can be done in the same document (see
> section 4 of the document above).
>=20
> Just to make sure I understand: are you suggesting that you could add an
> impp purpose for service-uris in the above document?
>=20
> If so, then this certainly works for me!
>=20
> Thanks!
> Emil
>=20
> >
> > Regards,
> >  Rifaat
> >
> >
> >> -----Original Message-----
> >> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> >> Behalf Of Sa=FAl Ibarra Corretg=E9
> >> Sent: Monday, April 22, 2013 3:53 AM
> >> To: Emil Ivov
> >> Cc: dispatch@ietf.org
> >> Subject: Re: [dispatch] Announcing a MUC with RFC4575
> >>
> >> Hi Emil!
> >>
> >> On Apr 21, 2013, at 10:43 PM, Emil Ivov wrote:
> >>
> >>> Hey all,
> >>>
> >>> I have been looking at RFC4575 and wondering what the best way would
> >> be
> >>> for a mixer to announce that a conference has an associated chatroom
> >>> (e.g. an XMPP MUC) available at a certain address. This would allow
> >> UAs
> >>> that support MUCs can join there too.
> >>>
> >>> From what I see the most reasonable way to do this would be a
> >>> <conf-uri>, like this for example:
> >>>
> >>> <conf-uris>
> >>>  <entry>
> >>>   <uri>sip:conf545@h323.example.com</uri>
> >>>   <display-text>TTI Bridge</display-text>
> >>>   <purpose>participation</purpose>
> >>>  </entry>
> >>>  <entry>
> >>>   <uri>xmpp:conf545@conference.example.com</uri>
> >>>   <display-text>Multi-User Chat 545</display-text>
> >>>   <purpose>impp-participation</purpose>
> >>>  </entry>
> >>> </conf-uris>
> >>>
> >>
> >> I think the above does make sense, now, would the same URI be usable
> >> with Jingle or is intended for MUC only?
> >>
> >>> Another option would be to use the same <entry> but place it ina a
> >>> <service-uris> element instead:
> >>>
> >>> <service-uris>
> >>>  <entry>
> >>>   <uri>http://sharepoint/salesgroup/</uri>
> >>>   <purpose>web-page</purpose>
> >>>  </entry>
> >>>  <entry>
> >>>   <uri>xmpp:conf545@conference.example.com</uri>
> >>>   <display-text>Multi-User Chat 545</display-text>
> >>>   <purpose>impp-participation</purpose>
> >>>  </entry>
> >>> </service-uris>
> >>>
> >>> Does any of the above make sense? In case no other mechanism allows
> >> for
> >>> the same thing in a simpler way, then could we maybe put one of the
> >> two
> >>> on paper?
> >>>
> >>
> >> This second approach would require a new entry in the IANA registry
> for
> >> that 'purpose' element right? In any case, this one could also make
> >> sense, I guess it depends on the scenario.
> >>
> >> Assuming your scenario is Jitsi's use case, that is, the audio /
> video
> >> conference is held by some entity and the MUC by some other, IMHO
> this
> >> second approach makes more sense.
> >>
> >>
> >> Regards,
> >>
> >> --
> >> Sa=FAl Ibarra Corretg=E9
> >> AG Projects
> >>
> >>
> >>
> >> _______________________________________________
> >> dispatch mailing list
> >> dispatch@ietf.org
> >> https://www.ietf.org/mailman/listinfo/dispatch
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> >
>=20
> --
> https://jitsi.org


From stpeter@stpeter.im  Mon Apr 22 12:48:30 2013
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Subject: Re: [dispatch] Announcing a MUC with RFC4575
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On 4/22/13 1:36 PM, Shekh-Yusef, Rifaat (Rifaat) wrote:
>> Just to make sure I understand: are you suggesting that you could add an
>> impp purpose for service-uris in the above document?
>>
>> If so, then this certainly works for me!
> 
> No, that is not what I am suggesting.
> What I am saying is that when you create your document to define your impp, you can register the new purpose value in the same document in the IANA section, similar to what we did in section 4 of our CCMP document.

https://datatracker.ietf.org/doc/draft-saintandre-impp-call-info/
registers the "impp" value for Call-Info, but not for other message
header fields.

Peter


From stpeter@stpeter.im  Mon Apr 22 13:44:38 2013
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On 4/21/13 2:43 PM, Emil Ivov wrote:
> Hey all,
> 
> I have been looking at RFC4575 and wondering what the best way would be
> for a mixer to announce that a conference has an associated chatroom
> (e.g. an XMPP MUC) available at a certain address. This would allow UAs
> that support MUCs can join there too.
> 
> From what I see the most reasonable way to do this would be a
> <conf-uri>, like this for example:
> 
> <conf-uris>
>   <entry>
>    <uri>sip:conf545@h323.example.com</uri>
>    <display-text>TTI Bridge</display-text>
>    <purpose>participation</purpose>
>   </entry>
>   <entry>
>    <uri>xmpp:conf545@conference.example.com</uri>
>    <display-text>Multi-User Chat 545</display-text>
>    <purpose>impp-participation</purpose>
>   </entry>
> </conf-uris>
> 
> Another option would be to use the same <entry> but place it ina a
> <service-uris> element instead:
> 
> <service-uris>
>   <entry>
>    <uri>http://sharepoint/salesgroup/</uri>
>    <purpose>web-page</purpose>
>   </entry>
>   <entry>
>    <uri>xmpp:conf545@conference.example.com</uri>
>    <display-text>Multi-User Chat 545</display-text>
>    <purpose>impp-participation</purpose>
>   </entry>
> </service-uris>
> 
> Does any of the above make sense? In case no other mechanism allows for
> the same thing in a simpler way, then could we maybe put one of the two
> on paper?

Somehow "impp-participation" doesn't sound right to me. What you're
talking about is a multiparty text conference that is associated with or
auxiliary to the main conference. It seems to me that the text
conference could be an XMPP MUC room, an IRC channel, an MSRP multiparty
chat session, etc. So I would suggest something like "groupchat" (since
the primary purpose here is multiparty text chat, not one-to-one IM and
presence as they are traditionally understood).

Peter


Peter


From emil@sip-communicator.org  Mon Apr 22 13:59:35 2013
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Date: Mon, 22 Apr 2013 22:59:29 +0200
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Subject: Re: [dispatch] Announcing a MUC with RFC4575
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On 22.04.13, 22:44, Peter Saint-Andre wrote:
> On 4/21/13 2:43 PM, Emil Ivov wrote:
>> Hey all,
>>
>> I have been looking at RFC4575 and wondering what the best way would be
>> for a mixer to announce that a conference has an associated chatroom
>> (e.g. an XMPP MUC) available at a certain address. This would allow UAs
>> that support MUCs can join there too.
>>
>> From what I see the most reasonable way to do this would be a
>> <conf-uri>, like this for example:
>>
>> <conf-uris>
>>   <entry>
>>    <uri>sip:conf545@h323.example.com</uri>
>>    <display-text>TTI Bridge</display-text>
>>    <purpose>participation</purpose>
>>   </entry>
>>   <entry>
>>    <uri>xmpp:conf545@conference.example.com</uri>
>>    <display-text>Multi-User Chat 545</display-text>
>>    <purpose>impp-participation</purpose>
>>   </entry>
>> </conf-uris>
>>
>> Another option would be to use the same <entry> but place it ina a
>> <service-uris> element instead:
>>
>> <service-uris>
>>   <entry>
>>    <uri>http://sharepoint/salesgroup/</uri>
>>    <purpose>web-page</purpose>
>>   </entry>
>>   <entry>
>>    <uri>xmpp:conf545@conference.example.com</uri>
>>    <display-text>Multi-User Chat 545</display-text>
>>    <purpose>impp-participation</purpose>
>>   </entry>
>> </service-uris>
>>
>> Does any of the above make sense? In case no other mechanism allows for
>> the same thing in a simpler way, then could we maybe put one of the two
>> on paper?
> 
> Somehow "impp-participation" doesn't sound right to me.

That was just a stab really. I don't really have a preference for the
name ... unless it would mean less overhead: if for example we could
have just reused the "impp" purpose from draft-saintandre-impp-call-info
then that would have been great. However, as you point out, it currently
only applies to Call-Info and we would need another draft that looks
pretty much the same way (like Rifaat suggested).

> What you're
> talking about is a multiparty text conference that is associated with or
> auxiliary to the main conference. It seems to me that the text
> conference could be an XMPP MUC room, an IRC channel, an MSRP multiparty
> chat session, etc.

Correct.

> So I would suggest something like "groupchat" (since
> the primary purpose here is multiparty text chat, not one-to-one IM and
> presence as they are traditionally understood).

Well, it seems to me that a URI with an "impp" purpose that has a 1-to-1
relation with a conference can only be a group chat and it would also
give us some uniformity in the naming conventions. But again, I don't
really mind what the exact name would be as long as a name is defined.

Cheers,
Emil
> 
> Peter
> 
> 
> Peter
> 
> 

-- 
https://jitsi.org

From stpeter@stpeter.im  Mon Apr 22 14:06:58 2013
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On 4/22/13 2:59 PM, Emil Ivov wrote:
> 
> 
> On 22.04.13, 22:44, Peter Saint-Andre wrote:
>> On 4/21/13 2:43 PM, Emil Ivov wrote:
>>> Hey all,
>>>
>>> I have been looking at RFC4575 and wondering what the best way would be
>>> for a mixer to announce that a conference has an associated chatroom
>>> (e.g. an XMPP MUC) available at a certain address. This would allow UAs
>>> that support MUCs can join there too.
>>>
>>> From what I see the most reasonable way to do this would be a
>>> <conf-uri>, like this for example:
>>>
>>> <conf-uris>
>>>   <entry>
>>>    <uri>sip:conf545@h323.example.com</uri>
>>>    <display-text>TTI Bridge</display-text>
>>>    <purpose>participation</purpose>
>>>   </entry>
>>>   <entry>
>>>    <uri>xmpp:conf545@conference.example.com</uri>
>>>    <display-text>Multi-User Chat 545</display-text>
>>>    <purpose>impp-participation</purpose>
>>>   </entry>
>>> </conf-uris>
>>>
>>> Another option would be to use the same <entry> but place it ina a
>>> <service-uris> element instead:
>>>
>>> <service-uris>
>>>   <entry>
>>>    <uri>http://sharepoint/salesgroup/</uri>
>>>    <purpose>web-page</purpose>
>>>   </entry>
>>>   <entry>
>>>    <uri>xmpp:conf545@conference.example.com</uri>
>>>    <display-text>Multi-User Chat 545</display-text>
>>>    <purpose>impp-participation</purpose>
>>>   </entry>
>>> </service-uris>
>>>
>>> Does any of the above make sense? In case no other mechanism allows for
>>> the same thing in a simpler way, then could we maybe put one of the two
>>> on paper?
>>
>> Somehow "impp-participation" doesn't sound right to me.
> 
> That was just a stab really. I don't really have a preference for the
> name ... unless it would mean less overhead: if for example we could
> have just reused the "impp" purpose from draft-saintandre-impp-call-info
> then that would have been great. However, as you point out, it currently
> only applies to Call-Info and we would need another draft that looks
> pretty much the same way (like Rifaat suggested).
> 
>> What you're
>> talking about is a multiparty text conference that is associated with or
>> auxiliary to the main conference. It seems to me that the text
>> conference could be an XMPP MUC room, an IRC channel, an MSRP multiparty
>> chat session, etc.
> 
> Correct.
> 
>> So I would suggest something like "groupchat" (since
>> the primary purpose here is multiparty text chat, not one-to-one IM and
>> presence as they are traditionally understood).
> 
> Well, it seems to me that a URI with an "impp" purpose that has a 1-to-1
> relation with a conference can only be a group chat and it would also
> give us some uniformity in the naming conventions. But again, I don't
> really mind what the exact name would be as long as a name is defined.

The "impp" acronym goes back to the IMPP WG and its main output, RFC
2778 and RFC 2779, which talked about one-to-one IM and one-to-one
presence. IMHO groupchat / multiparty text conferencing / whatever you
want to call it is quite a different beast.

Peter



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Subject: Re: [dispatch] Announcing a MUC with RFC4575
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On 22.04.13, 23:06, Peter Saint-Andre wrote:
>> Well, it seems to me that a URI with an "impp" purpose that has a 1-to-1
>> relation with a conference can only be a group chat and it would also
>> give us some uniformity in the naming conventions. But again, I don't
>> really mind what the exact name would be as long as a name is defined.
> 
> The "impp" acronym goes back to the IMPP WG and its main output, RFC
> 2778 and RFC 2779, which talked about one-to-one IM and one-to-one
> presence. IMHO groupchat / multiparty text conferencing / whatever you
> want to call it is quite a different beast.

OK then, so can we agree that a "groupchat" purpose would be a
reasonable resolution to this and that a short draft specifying it would
make sense?

Some guidance from the chairs would be very much appreciated ;)

Cheers,
Emil


-- 
https://jitsi.org

From pkyzivat@alum.mit.edu  Mon Apr 22 14:21:20 2013
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Subject: Re: [dispatch] Announcing a MUC with RFC4575
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On 4/22/13 4:44 PM, Peter Saint-Andre wrote:
> On 4/21/13 2:43 PM, Emil Ivov wrote:
>> Hey all,
>>
>> I have been looking at RFC4575 and wondering what the best way would be
>> for a mixer to announce that a conference has an associated chatroom
>> (e.g. an XMPP MUC) available at a certain address. This would allow UAs
>> that support MUCs can join there too.
>>
>>  From what I see the most reasonable way to do this would be a
>> <conf-uri>, like this for example:
>>
>> <conf-uris>
>>    <entry>
>>     <uri>sip:conf545@h323.example.com</uri>
>>     <display-text>TTI Bridge</display-text>
>>     <purpose>participation</purpose>
>>    </entry>
>>    <entry>
>>     <uri>xmpp:conf545@conference.example.com</uri>
>>     <display-text>Multi-User Chat 545</display-text>
>>     <purpose>impp-participation</purpose>
>>    </entry>
>> </conf-uris>
>>
>> Another option would be to use the same <entry> but place it ina a
>> <service-uris> element instead:
>>
>> <service-uris>
>>    <entry>
>>     <uri>http://sharepoint/salesgroup/</uri>
>>     <purpose>web-page</purpose>
>>    </entry>
>>    <entry>
>>     <uri>xmpp:conf545@conference.example.com</uri>
>>     <display-text>Multi-User Chat 545</display-text>
>>     <purpose>impp-participation</purpose>
>>    </entry>
>> </service-uris>
>>
>> Does any of the above make sense? In case no other mechanism allows for
>> the same thing in a simpler way, then could we maybe put one of the two
>> on paper?
>
> Somehow "impp-participation" doesn't sound right to me. What you're
> talking about is a multiparty text conference that is associated with or
> auxiliary to the main conference. It seems to me that the text
> conference could be an XMPP MUC room, an IRC channel, an MSRP multiparty
> chat session, etc. So I would suggest something like "groupchat" (since
> the primary purpose here is multiparty text chat, not one-to-one IM and
> presence as they are traditionally understood).

Yes, any of these could be possible, though probably if it was MSRP it 
wouldn't be a separate URI from the main conference URL.

So the mechanism should be work for any of those. 
<purpose>groupchat</purpose> together with the URI scheme to distinguish 
among them should work. (As long as we believe everyone knows that 
"chat" means "type" rather than "talk informally".)

	Thanks,
	Paul


From jonathan@vidyo.com  Tue Apr 23 07:41:56 2013
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From: Jonathan Lennox <jonathan@vidyo.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
Date: Tue, 23 Apr 2013 10:41:54 -0400
Thread-Topic: [dispatch] Announcing a MUC with RFC4575
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On Apr 22, 2013, at 5:21 PM, Paul Kyzivat wrote:

> On 4/22/13 4:44 PM, Peter Saint-Andre wrote:
>>=20
>> Somehow "impp-participation" doesn't sound right to me. What you're
>> talking about is a multiparty text conference that is associated with or
>> auxiliary to the main conference. It seems to me that the text
>> conference could be an XMPP MUC room, an IRC channel, an MSRP multiparty
>> chat session, etc. So I would suggest something like "groupchat" (since
>> the primary purpose here is multiparty text chat, not one-to-one IM and
>> presence as they are traditionally understood).
>=20
> Yes, any of these could be possible, though probably if it was MSRP it=20
> wouldn't be a separate URI from the main conference URL.
>=20
> So the mechanism should be work for any of those.=20
> <purpose>groupchat</purpose> together with the URI scheme to distinguish=
=20
> among them should work. (As long as we believe everyone knows that=20
> "chat" means "type" rather than "talk informally".)

Maybe something with "text" in the name would be clearer -- "grouptextchat"=
?

(Also, I think the bike shed should be orange.)

--
Jonathan Lennox
jonathan@vidyo.com



From mary.ietf.barnes@gmail.com  Tue Apr 23 07:50:54 2013
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Subject: Re: [dispatch] Announcing a MUC with RFC4575
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This would also be something useful for the XCON chat, which is
defined in draft-boulton-xcon-session-chat (we will shortly be
updating this and posting for discussion on this mailing list).  I
think "grouptextchat" would work, although I don't find it
particularly readable, but I don't think abbreviating is useful, so I
don't have another suggestion.

Mary.

On Tue, Apr 23, 2013 at 9:41 AM, Jonathan Lennox <jonathan@vidyo.com> wrote:
>
> On Apr 22, 2013, at 5:21 PM, Paul Kyzivat wrote:
>
>> On 4/22/13 4:44 PM, Peter Saint-Andre wrote:
>>>
>>> Somehow "impp-participation" doesn't sound right to me. What you're
>>> talking about is a multiparty text conference that is associated with or
>>> auxiliary to the main conference. It seems to me that the text
>>> conference could be an XMPP MUC room, an IRC channel, an MSRP multiparty
>>> chat session, etc. So I would suggest something like "groupchat" (since
>>> the primary purpose here is multiparty text chat, not one-to-one IM and
>>> presence as they are traditionally understood).
>>
>> Yes, any of these could be possible, though probably if it was MSRP it
>> wouldn't be a separate URI from the main conference URL.
>>
>> So the mechanism should be work for any of those.
>> <purpose>groupchat</purpose> together with the URI scheme to distinguish
>> among them should work. (As long as we believe everyone knows that
>> "chat" means "type" rather than "talk informally".)
>
> Maybe something with "text" in the name would be clearer -- "grouptextchat"?
>
> (Also, I think the bike shed should be orange.)
>
> --
> Jonathan Lennox
> jonathan@vidyo.com
>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

From emil@sip-communicator.org  Tue Apr 23 08:33:18 2013
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Date: Tue, 23 Apr 2013 17:33:13 +0200
From: Emil Ivov <emcho@jitsi.org>
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Cc: Jonathan Lennox <jonathan@vidyo.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] Announcing a MUC with RFC4575
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Hey Mary,

On 23.04.13, 16:50, Mary Barnes wrote:
> This would also be something useful for the XCON chat, which is
> defined in draft-boulton-xcon-session-chat (we will shortly be
> updating this and posting for discussion on this mailing list).  I
> think "grouptextchat" would work, although I don't find it
> particularly readable, but I don't think abbreviating is useful, so I
> don't have another suggestion.

I suppose I am once again missing something so I'd better ask :)

Are you saying that you would find a draft that defines the value
helpful for draft-boulton-xcon-session-chat or are you planning on
adding the definition in the draft itself?

The former could be quicker I suppose and I'd be happy to write the
document, but the latter also works for me.

Cheers,
Emil

> 
> Mary.
> 
> On Tue, Apr 23, 2013 at 9:41 AM, Jonathan Lennox <jonathan@vidyo.com> wrote:
>>
>> On Apr 22, 2013, at 5:21 PM, Paul Kyzivat wrote:
>>
>>> On 4/22/13 4:44 PM, Peter Saint-Andre wrote:
>>>>
>>>> Somehow "impp-participation" doesn't sound right to me. What you're
>>>> talking about is a multiparty text conference that is associated with or
>>>> auxiliary to the main conference. It seems to me that the text
>>>> conference could be an XMPP MUC room, an IRC channel, an MSRP multiparty
>>>> chat session, etc. So I would suggest something like "groupchat" (since
>>>> the primary purpose here is multiparty text chat, not one-to-one IM and
>>>> presence as they are traditionally understood).
>>>
>>> Yes, any of these could be possible, though probably if it was MSRP it
>>> wouldn't be a separate URI from the main conference URL.
>>>
>>> So the mechanism should be work for any of those.
>>> <purpose>groupchat</purpose> together with the URI scheme to distinguish
>>> among them should work. (As long as we believe everyone knows that
>>> "chat" means "type" rather than "talk informally".)
>>
>> Maybe something with "text" in the name would be clearer -- "grouptextchat"?
>>
>> (Also, I think the bike shed should be orange.)
>>
>> --
>> Jonathan Lennox
>> jonathan@vidyo.com
>>
>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

-- 
https://jitsi.org

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Subject: Re: [dispatch] Announcing a MUC with RFC4575
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On Tue, Apr 23, 2013 at 10:33 AM, Emil Ivov <emcho@jitsi.org> wrote:
> Hey Mary,
>
> On 23.04.13, 16:50, Mary Barnes wrote:
>> This would also be something useful for the XCON chat, which is
>> defined in draft-boulton-xcon-session-chat (we will shortly be
>> updating this and posting for discussion on this mailing list).  I
>> think "grouptextchat" would work, although I don't find it
>> particularly readable, but I don't think abbreviating is useful, so I
>> don't have another suggestion.
>
> I suppose I am once again missing something so I'd better ask :)
>
> Are you saying that you would find a draft that defines the value
> helpful for draft-boulton-xcon-session-chat or are you planning on
> adding the definition in the draft itself?
>
> The former could be quicker I suppose and I'd be happy to write the
> document, but the latter also works for me.
[MB] I agree with you.  I think we can write a simple doc to define
this and it can be referenced by these other docs as necessary (since
it isn't unique to any of these chat applications).
[/MB]
>
> Cheers,
> Emil
>
>>
>> Mary.
>>
>> On Tue, Apr 23, 2013 at 9:41 AM, Jonathan Lennox <jonathan@vidyo.com> wrote:
>>>
>>> On Apr 22, 2013, at 5:21 PM, Paul Kyzivat wrote:
>>>
>>>> On 4/22/13 4:44 PM, Peter Saint-Andre wrote:
>>>>>
>>>>> Somehow "impp-participation" doesn't sound right to me. What you're
>>>>> talking about is a multiparty text conference that is associated with or
>>>>> auxiliary to the main conference. It seems to me that the text
>>>>> conference could be an XMPP MUC room, an IRC channel, an MSRP multiparty
>>>>> chat session, etc. So I would suggest something like "groupchat" (since
>>>>> the primary purpose here is multiparty text chat, not one-to-one IM and
>>>>> presence as they are traditionally understood).
>>>>
>>>> Yes, any of these could be possible, though probably if it was MSRP it
>>>> wouldn't be a separate URI from the main conference URL.
>>>>
>>>> So the mechanism should be work for any of those.
>>>> <purpose>groupchat</purpose> together with the URI scheme to distinguish
>>>> among them should work. (As long as we believe everyone knows that
>>>> "chat" means "type" rather than "talk informally".)
>>>
>>> Maybe something with "text" in the name would be clearer -- "grouptextchat"?
>>>
>>> (Also, I think the bike shed should be orange.)
>>>
>>> --
>>> Jonathan Lennox
>>> jonathan@vidyo.com
>>>
>>>
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
>
> --
> https://jitsi.org

From stpeter@stpeter.im  Tue Apr 23 10:57:55 2013
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Subject: Re: [dispatch] Announcing a MUC with RFC4575
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On 4/23/13 9:36 AM, Mary Barnes wrote:
> On Tue, Apr 23, 2013 at 10:33 AM, Emil Ivov <emcho@jitsi.org>
> wrote:
>> Hey Mary,
>> 
>> On 23.04.13, 16:50, Mary Barnes wrote:
>>> This would also be something useful for the XCON chat, which
>>> is defined in draft-boulton-xcon-session-chat (we will shortly
>>> be updating this and posting for discussion on this mailing
>>> list).  I think "grouptextchat" would work, although I don't
>>> find it particularly readable, but I don't think abbreviating
>>> is useful, so I don't have another suggestion.
>> 
>> I suppose I am once again missing something so I'd better ask :)
>> 
>> Are you saying that you would find a draft that defines the
>> value helpful for draft-boulton-xcon-session-chat or are you
>> planning on adding the definition in the draft itself?
>> 
>> The former could be quicker I suppose and I'd be happy to write
>> the document, but the latter also works for me.
> [MB] I agree with you.  I think we can write a simple doc to
> define this and it can be referenced by these other docs as
> necessary (since it isn't unique to any of these chat
> applications). [/MB]

+1

Peter



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Subject: Re: [dispatch] Announcing a MUC with RFC4575
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There it is then:

http://tools.ietf.org/html/draft-ivov-grouptextchat-purpose

Cheers,
Emil

-------- Original Message --------
Subject: New Version Notification for
draft-ivov-grouptextchat-purpose-00.txt
Date: Tue, 23 Apr 2013 19:09:01 -0700
From: internet-drafts@ietf.org
To: Emil Ivov <emcho@jitsi.org>


A new version of I-D, draft-ivov-grouptextchat-purpose-00.txt
has been successfully submitted by Emil Ivov and posted to the
IETF repository.

Filename:	 draft-ivov-grouptextchat-purpose
Revision:	 00
Title:		 A Group Text Chat Purpose for Conference and Service URIs in
the Session Initiation Protocol (SIP) Event Package for Conference State
Creation date:	 2013-04-24
Group:		 Individual Submission
Number of pages: 5
URL:
http://www.ietf.org/internet-drafts/draft-ivov-grouptextchat-purpose-00.txt
Status:
http://datatracker.ietf.org/doc/draft-ivov-grouptextchat-purpose
Htmlized:
http://tools.ietf.org/html/draft-ivov-grouptextchat-purpose-00


Abstract:
   This document defines and registers a value of "grouptextchat"
   ("Group Text Chat") value for the URI <purpose> element of SIP's
   Conference Event Package [RFC4575].





The IETF Secretariat

On 23.04.13, 19:57, Peter Saint-Andre wrote:
> On 4/23/13 9:36 AM, Mary Barnes wrote:
>> On Tue, Apr 23, 2013 at 10:33 AM, Emil Ivov <emcho@jitsi.org>
>> wrote:
>>> Hey Mary,
>>>
>>> On 23.04.13, 16:50, Mary Barnes wrote:
>>>> This would also be something useful for the XCON chat, which
>>>> is defined in draft-boulton-xcon-session-chat (we will shortly
>>>> be updating this and posting for discussion on this mailing
>>>> list).  I think "grouptextchat" would work, although I don't
>>>> find it particularly readable, but I don't think abbreviating
>>>> is useful, so I don't have another suggestion.
>>>
>>> I suppose I am once again missing something so I'd better ask :)
>>>
>>> Are you saying that you would find a draft that defines the
>>> value helpful for draft-boulton-xcon-session-chat or are you
>>> planning on adding the definition in the draft itself?
>>>
>>> The former could be quicker I suppose and I'd be happy to write
>>> the document, but the latter also works for me.
>> [MB] I agree with you.  I think we can write a simple doc to
>> define this and it can be referenced by these other docs as
>> necessary (since it isn't unique to any of these chat
>> applications). [/MB]
> 
> +1
> 
> Peter
> 
> 
> 

-- 
https://jitsi.org

From saul@ag-projects.com  Tue Apr 23 23:48:31 2013
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Subject: Re: [dispatch] Announcing a MUC with RFC4575
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On Apr 24, 2013, at 4:10 AM, Emil Ivov wrote:

> There it is then:
>=20
> http://tools.ietf.org/html/draft-ivov-grouptextchat-purpose
>=20

+1

> Cheers,
> Emil
>=20
> -------- Original Message --------
> Subject: New Version Notification for
> draft-ivov-grouptextchat-purpose-00.txt
> Date: Tue, 23 Apr 2013 19:09:01 -0700
> From: internet-drafts@ietf.org
> To: Emil Ivov <emcho@jitsi.org>
>=20
>=20
> A new version of I-D, draft-ivov-grouptextchat-purpose-00.txt
> has been successfully submitted by Emil Ivov and posted to the
> IETF repository.
>=20
> Filename:	 draft-ivov-grouptextchat-purpose
> Revision:	 00
> Title:		 A Group Text Chat Purpose for Conference and =
Service URIs in
> the Session Initiation Protocol (SIP) Event Package for Conference =
State
> Creation date:	 2013-04-24
> Group:		 Individual Submission
> Number of pages: 5
> URL:
> =
http://www.ietf.org/internet-drafts/draft-ivov-grouptextchat-purpose-00.tx=
t
> Status:
> http://datatracker.ietf.org/doc/draft-ivov-grouptextchat-purpose
> Htmlized:
> http://tools.ietf.org/html/draft-ivov-grouptextchat-purpose-00
>=20
>=20
> Abstract:
>   This document defines and registers a value of "grouptextchat"
>   ("Group Text Chat") value for the URI <purpose> element of SIP's
>   Conference Event Package [RFC4575].
>=20
>=20
>=20
>=20
>=20
> The IETF Secretariat
>=20
> On 23.04.13, 19:57, Peter Saint-Andre wrote:
>> On 4/23/13 9:36 AM, Mary Barnes wrote:
>>> On Tue, Apr 23, 2013 at 10:33 AM, Emil Ivov <emcho@jitsi.org>
>>> wrote:
>>>> Hey Mary,
>>>>=20
>>>> On 23.04.13, 16:50, Mary Barnes wrote:
>>>>> This would also be something useful for the XCON chat, which
>>>>> is defined in draft-boulton-xcon-session-chat (we will shortly
>>>>> be updating this and posting for discussion on this mailing
>>>>> list).  I think "grouptextchat" would work, although I don't
>>>>> find it particularly readable, but I don't think abbreviating
>>>>> is useful, so I don't have another suggestion.
>>>>=20
>>>> I suppose I am once again missing something so I'd better ask :)
>>>>=20
>>>> Are you saying that you would find a draft that defines the
>>>> value helpful for draft-boulton-xcon-session-chat or are you
>>>> planning on adding the definition in the draft itself?
>>>>=20
>>>> The former could be quicker I suppose and I'd be happy to write
>>>> the document, but the latter also works for me.
>>> [MB] I agree with you.  I think we can write a simple doc to
>>> define this and it can be referenced by these other docs as
>>> necessary (since it isn't unique to any of these chat
>>> applications). [/MB]
>>=20
>> +1
>>=20
>> Peter
>>=20
>>=20
>>=20
>=20
> --=20
> https://jitsi.org
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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Subject: Re: [dispatch] Announcing a MUC with RFC4575
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On 4/24/13 9:48 AM, Saúl Ibarra Corretgé wrote:
> On Apr 24, 2013, at 4:10 AM, Emil Ivov wrote:
>
>> There it is then:
>>
>> http://tools.ietf.org/html/draft-ivov-grouptextchat-purpose

great!

/Sal


From fluffy@cisco.com  Wed Apr 24 07:29:54 2013
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Emil Ivov <emcho@jitsi.org>
Thread-Topic: [dispatch] Announcing a MUC with RFC4575
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On Apr 23, 2013, at 8:10 PM, Emil Ivov <emcho@jitsi.org> wrote:

> There it is then:
>=20
> http://tools.ietf.org/html/draft-ivov-grouptextchat-purpose
>=20
> Cheers,
> Emil

I like it.=20

Now how would people like to dispatch this? My preference would probably be=
 if one of the authors could talk to ADs about AD sponsoring this and if th=
ey seem like they might be OK with that, just do a quick call on the dispat=
ch list for any objects then send it that way.=20

Cullen=20




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From: Mary Barnes <mary.ietf.barnes@gmail.com>
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Subject: Re: [dispatch] Announcing a MUC with RFC4575
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On Wed, Apr 24, 2013 at 9:29 AM, Cullen Jennings (fluffy)
<fluffy@cisco.com> wrote:
>
> On Apr 23, 2013, at 8:10 PM, Emil Ivov <emcho@jitsi.org> wrote:
>
>> There it is then:
>>
>> http://tools.ietf.org/html/draft-ivov-grouptextchat-purpose
>>
>> Cheers,
>> Emil
>
> I like it.
>
> Now how would people like to dispatch this? My preference would probably =
be if one of the authors could talk to ADs about AD sponsoring this and if =
they seem like they might be OK with that, just do a quick call on the disp=
atch list for any objects then send it that way.
>
> Cullen
[MB] My thoughts exactly.  I've cc'ed the RAI ADs to see if they have
any issues with this approach. [/MB]

Mary

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

From fluffy@iii.ca  Wed Apr 24 07:37:45 2013
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To: Olle E. Johansson <oej@edvina.net>
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Subject: [dispatch] Brigining SIP+DANE to Dispatch
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I'm keen on someone writing a draft that specifies how to do DANE with =
SIP and bringing that draft to dispatch.=20

However, I don't think you want to do that as un update to 5922. I think =
you want it to be just a new draft that is an extension to SIP and does =
not update or replace any of the existing draft.=20

I think Dispatch would be a good place to start the discussion around it =
and decide how if this was a sip core thing, or a new small WG, or =
something else. There are a few different problems that can be tacked. =
One is validating that the TLS connection has the right cert. The other =
is more 4474 style identity issues. I think it will be good to see =
discussion on the dispatch list of the scope of problems we want to =
solve with DANE.=20

Cullen <with my Dispatch co-chair hat on>



On Apr 21, 2013, at 5:25 AM, Olle E. Johansson <oej@edvina.net> wrote:

> Hi!
>=20
> Looking at DANE for SIP, I mailed out some thoughts on the DANE =
mailing list. If we come to
> the conclusion that RFC 5922 will need to be updated - will this work =
be done in Dispatch
> or Dane or somewhere else?
>=20
> DANE also has impacts on SIP Identity as an alternative discovery =
mechanism for
> domain certificates (replacing HTTPS).
>=20
> /O
>=20
> Vidarebefordrat brev:
>=20
>> Fr=E5n: "Olle E. Johansson" <oej@edvina.net>
>> =C4mne: DANE SRV draft and SIP
>> Datum: 21 april 2013 12:27:47 CEST
>> Till: dane WG list <dane@ietf.org>
>> Kopia: "Olle E. Johansson" <oej@edvina.net>
>>=20
>> Hi!
>>=20
>> Looking at the DANE SRV draft with my SIP eyes I realize that there's =
a lot of work to be done to get this
>> to work with SIP.=20
>>=20
>> SIP has no StartTLS, we use NAPTR records to select transport. The =
NAPTR points to a SRV
>> name that resolves into a set of host names.
>>=20
>> The SIP Domain certificates RFC - http://tools.ietf.org/html/rfc5922 =
- specificies matching a=20
>> given SIP URI with a certificate. The matching is done on service =
domain either with=20
>> a ALT name URI, like sip:ietf.org or a DNS name, like ietf.org - but =
not the host name.
>>=20
>> I personally agree with the policy in the DANE SRV draft that we =
should match on the
>> SRV hostname used to get A/AAAA records when using DNSsec. For this =
to work,
>> the path from the SIP URI over NAPTR to SRV and hostnames needs to be =
fully
>> signed in and verified in the DNS. This is going to require an update =
to 5922.
>>=20
>> The final question is how to handle this without SNI. The =
certificates for both
>> DANE verification and RFC 5922 plus "old-style" verification with the =
sip
>> domain in the CN seems like a complicated mess to manage.
>>=20
>> Food for thought on a sunny Sunday.
>>=20
>> The SRV draft is not clear on how to use Subject AltNames of various =
types,
>> and doesn't mention NAPTR. I am not personally aware of other =
protocols using
>> this setup, so maybe this requires a very SIP specfic draft, =
following the wake of the
>> smtp work.
>>=20
>> /O
>>=20
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


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Subject: Re: [dispatch] Brigining SIP+DANE to Dispatch
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24 apr 2013 kl. 16:37 skrev Cullen Jennings <fluffy@iii.ca>:

>=20
> I'm keen on someone writing a draft that specifies how to do DANE with =
SIP and bringing that draft to dispatch.=20
>=20
> However, I don't think you want to do that as un update to 5922. I =
think you want it to be just a new draft that is an extension to SIP and =
does not update or replace any of the existing draft.=20
>=20
> I think Dispatch would be a good place to start the discussion around =
it and decide how if this was a sip core thing, or a new small WG, or =
something else. There are a few different problems that can be tacked. =
One is validating that the TLS connection has the right cert. The other =
is more 4474 style identity issues. I think it will be good to see =
discussion on the dispatch list of the scope of problems we want to =
solve with DANE.=20
>=20
Ok, thanks for the feedback. There's some discussion already on the DANE =
list to coordinate with the work on SMTP and XMPP.

I'm trying to collect my thoughts here and will come back with some =
ideas for discussion.

/O

> Cullen <with my Dispatch co-chair hat on>
>=20
>=20
>=20
> On Apr 21, 2013, at 5:25 AM, Olle E. Johansson <oej@edvina.net> wrote:
>=20
>> Hi!
>>=20
>> Looking at DANE for SIP, I mailed out some thoughts on the DANE =
mailing list. If we come to
>> the conclusion that RFC 5922 will need to be updated - will this work =
be done in Dispatch
>> or Dane or somewhere else?
>>=20
>> DANE also has impacts on SIP Identity as an alternative discovery =
mechanism for
>> domain certificates (replacing HTTPS).
>>=20
>> /O
>>=20
>> Vidarebefordrat brev:
>>=20
>>> Fr=E5n: "Olle E. Johansson" <oej@edvina.net>
>>> =C4mne: DANE SRV draft and SIP
>>> Datum: 21 april 2013 12:27:47 CEST
>>> Till: dane WG list <dane@ietf.org>
>>> Kopia: "Olle E. Johansson" <oej@edvina.net>
>>>=20
>>> Hi!
>>>=20
>>> Looking at the DANE SRV draft with my SIP eyes I realize that =
there's a lot of work to be done to get this
>>> to work with SIP.=20
>>>=20
>>> SIP has no StartTLS, we use NAPTR records to select transport. The =
NAPTR points to a SRV
>>> name that resolves into a set of host names.
>>>=20
>>> The SIP Domain certificates RFC - http://tools.ietf.org/html/rfc5922 =
- specificies matching a=20
>>> given SIP URI with a certificate. The matching is done on service =
domain either with=20
>>> a ALT name URI, like sip:ietf.org or a DNS name, like ietf.org - but =
not the host name.
>>>=20
>>> I personally agree with the policy in the DANE SRV draft that we =
should match on the
>>> SRV hostname used to get A/AAAA records when using DNSsec. For this =
to work,
>>> the path from the SIP URI over NAPTR to SRV and hostnames needs to =
be fully
>>> signed in and verified in the DNS. This is going to require an =
update to 5922.
>>>=20
>>> The final question is how to handle this without SNI. The =
certificates for both
>>> DANE verification and RFC 5922 plus "old-style" verification with =
the sip
>>> domain in the CN seems like a complicated mess to manage.
>>>=20
>>> Food for thought on a sunny Sunday.
>>>=20
>>> The SRV draft is not clear on how to use Subject AltNames of various =
types,
>>> and doesn't mention NAPTR. I am not personally aware of other =
protocols using
>>> this setup, so maybe this requires a very SIP specfic draft, =
following the wake of the
>>> smtp work.
>>>=20
>>> /O
>>>=20
>>=20
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch


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Date: Wed, 24 Apr 2013 17:32:08 -0400
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Subject: Re: [dispatch] Announcing a MUC with RFC4575
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That sounds fine to me.
--Richard


On Wed, Apr 24, 2013 at 10:32 AM, Mary Barnes <mary.ietf.barnes@gmail.com>wrote:

> On Wed, Apr 24, 2013 at 9:29 AM, Cullen Jennings (fluffy)
> <fluffy@cisco.com> wrote:
> >
> > On Apr 23, 2013, at 8:10 PM, Emil Ivov <emcho@jitsi.org> wrote:
> >
> >> There it is then:
> >>
> >> http://tools.ietf.org/html/draft-ivov-grouptextchat-purpose
> >>
> >> Cheers,
> >> Emil
> >
> > I like it.
> >
> > Now how would people like to dispatch this? My preference would probably
> be if one of the authors could talk to ADs about AD sponsoring this and if
> they seem like they might be OK with that, just do a quick call on the
> dispatch list for any objects then send it that way.
> >
> > Cullen
> [MB] My thoughts exactly.  I've cc'ed the RAI ADs to see if they have
> any issues with this approach. [/MB]
>
> Mary
>
> >
> >
> >
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
>

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<div dir=3D"ltr">That sounds fine to me. =A0<div style>--Richard</div></div=
><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Wed, Apr =
24, 2013 at 10:32 AM, Mary Barnes <span dir=3D"ltr">&lt;<a href=3D"mailto:m=
ary.ietf.barnes@gmail.com" target=3D"_blank">mary.ietf.barnes@gmail.com</a>=
&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im">On Wed, Apr 24, 2013 at 9:=
29 AM, Cullen Jennings (fluffy)<br>
&lt;<a href=3D"mailto:fluffy@cisco.com">fluffy@cisco.com</a>&gt; wrote:<br>
&gt;<br>
&gt; On Apr 23, 2013, at 8:10 PM, Emil Ivov &lt;<a href=3D"mailto:emcho@jit=
si.org">emcho@jitsi.org</a>&gt; wrote:<br>
&gt;<br>
&gt;&gt; There it is then:<br>
&gt;&gt;<br>
&gt;&gt; <a href=3D"http://tools.ietf.org/html/draft-ivov-grouptextchat-pur=
pose" target=3D"_blank">http://tools.ietf.org/html/draft-ivov-grouptextchat=
-purpose</a><br>
&gt;&gt;<br>
&gt;&gt; Cheers,<br>
&gt;&gt; Emil<br>
&gt;<br>
&gt; I like it.<br>
&gt;<br>
&gt; Now how would people like to dispatch this? My preference would probab=
ly be if one of the authors could talk to ADs about AD sponsoring this and =
if they seem like they might be OK with that, just do a quick call on the d=
ispatch list for any objects then send it that way.<br>

&gt;<br>
&gt; Cullen<br>
</div>[MB] My thoughts exactly. =A0I&#39;ve cc&#39;ed the RAI ADs to see if=
 they have<br>
any issues with this approach. [/MB]<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Mary<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; dispatch mailing list<br>
&gt; <a href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" target=3D"_=
blank">https://www.ietf.org/mailman/listinfo/dispatch</a><br>
</div></div></blockquote></div><br></div>

--089e0149d1b6b31ee304db2208d3--

From johnsonhammond2@hushmail.com  Sat Apr 27 10:08:10 2013
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Subject: [dispatch] Biggest Fake Conference in Computer Science
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Biggest Fake Conference in Computer Science


We are researchers from different parts of the world and conducted a study on  
the worldâ€™s biggest bogus computer science conference WORLDCOMP 
( http://sites.google.com/site/worlddump1 ) organized by Prof. Hamid Arabnia 
from University of Georgia, USA.


We submitted a fake paper to WORLDCOMP 2011 and again (the same paper 
with a modified title) to WORLDCOMP 2012. This paper had numerous 
fundamental mistakes. Sample statements from that paper include: 

(1). Binary logic is fuzzy logic and vice versa
(2). Pascal developed fuzzy logic
(3). Object oriented languages do not exhibit any polymorphism or inheritance
(4). TCP and IP are synonyms and are part of OSI model 
(5). Distributed systems deal with only one computer
(6). Laptop is an example for a super computer
(7). Operating system is an example for computer hardware


Also, our paper did not express any conceptual meaning.  However, it 
was accepted both the times without any modifications (and without 
any reviews) and we were invited to submit the final paper and a 
payment of $500+ fee to present the paper. We decided to use the 
fee for better purposes than making Prof. Hamid Arabnia (Chairman 
of WORLDCOMP) rich. After that, we received few reminders from 
WORLDCOMP to pay the fee but we never responded. 


We MUST say that you should look at the above website if you have any thoughts 
to submit a paper to WORLDCOMP.  DBLP and other indexing agencies have stopped 
indexing WORLDCOMPâ€™s proceedings since 2011 due to its fakeness. See 
http://www.informatik.uni-trier.de/~ley/db/conf/icai/index.html for of one of the 
conferences of WORLDCOMP and notice that there is no listing after 2010. See Section 2 of
http://sites.google.com/site/dumpconf for comments from well-known researchers 
about WORLDCOMP. 


The status of your WORLDCOMP papers can be changed from scientific
to other (i.e., junk or non-technical) at any time. Better not to have a paper than 
having it in WORLDCOMP and spoil the resume and peace of mind forever!


Our study revealed that WORLDCOMP is a money making business, 
using University of Georgia mask, for Prof. Hamid Arabnia. He is throwing 
out a small chunk of that money (around 20 dollars per paper published 
in WORLDCOMPâ€™s proceedings) to his puppet (Mr. Ashu Solo or A.M.G. Solo) 
who publicizes WORLDCOMP and also defends it at various forums, using 
fake/anonymous names. The puppet uses fake names and defames other conferences
to divert traffic to WORLDCOMP. He also makes anonymous phone calls and tries to 
threaten the critiques of WORLDCOMP (See Item 7 of Section 5 of above website). 
That is, the puppet does all his best to get a maximum number of papers published 
at WORLDCOMP to get more money into his (and Prof. Hamid Arabniaâ€™s) pockets. 


Monte Carlo Resort (the venue of WORLDCOMP for more than 10 years, until 2012) has 
refused to provide the venue for WORLDCOMPâ€™13 because of the fears of their image 
being tarnished due to WORLDCOMPâ€™s fraudulent activities. That is why WORLDCOMPâ€™13 
is taking place at a different resort. WORLDCOMP will not be held after 2013. 


The draft paper submission deadline is over but still there are no committee 
members, no reviewers, and there is no conference Chairman. The only contact 
details available on WORLDCOMPâ€™s website is just an email address! 

Let us make a direct request to Prof. Hamid arabnia: publish all reviews for 
all the papers (after blocking identifiable details) since 2000 conference. Reveal 
the names and affiliations of all the reviewers (for each year) and how many 
papers each reviewer had reviewed on average. We also request him to look at 
the Open Challenge (Section 6) at https://sites.google.com/site/moneycomp1 


Sorry for posting to multiple lists. Spreading the word is the only way to stop 
this bogus conference. Please forward this message to other mailing lists and people. 


We are shocked with Prof. Hamid Arabnia and his puppetâ€™s activities 
http://worldcomp-fake-bogus.blogspot.com   Search Google using the 
keyword worldcomp fake for additional links.


From prvs=183141684c=aallen@blackberry.com  Sun Apr 28 23:29:24 2013
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From: Andrew Allen <aallen@blackberry.com>
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Thread-Topic: Notes and Summary of April 17 call on draft-montemurro-gsma-imei-urn and draft-allen-dispatch-imei-urn-as-instanceid 
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Subject: [dispatch] Notes and Summary of April 17 call on draft-montemurro-gsma-imei-urn and draft-allen-dispatch-imei-urn-as-instanceid
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--_000_BBF5DDFE515C3946BC18D733B20DAD2338D602E7XMB104ADSrimnet_
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In order to make progress on draft-montemurro-gsma-imei-urn and draft-allen-=
dispatch-imei-urn-as-instanceid a conference call was held on Wednesday

April 17 attended by Culllen Jennings, Atle Monrad, John Luc Bakker and myse=
lf.



Below are the combined notes and summary from Cullen and I of this call on t=
he IMEI drafts:





1) change draft-allen-dispatch-imei-urn-as-instanceid to say IMEI is only us=
ed in either a register request or also in an emergency invite request



Cullen will then be OK with it from a privacy point of view as long as it ex=
plains the trade offs.



It should be highlighted in draft-montemurro-gsma-imei-urn that using the IM=
EI as a security credential is a bad thing even though some popular commerci=
al applications are actually doing this and some service providers are using=
 knowledge of the IMEI to verify the legitimacy of caller being the authoriz=
ed user/owner of the device. The IMEI is usually extremely easy to obtain by=
 anyone who has very brief physical access to the device as it is often prin=
ted on the device beneath the battery and can be displayed by using a well-k=
nown key press sequence.



Change draft-allen-dispatch-imei-urn-as-instanceid to not be open to any ext=
ension to URN but instead require a draft updating this draft if any paramet=
er other than ver=3D0 was to be included.  Also add text in draft-montemurro=
-gsma-imei-urn and, draft-aindicating that extensions to the IMEI parameters=
 will require an update of draft-allen-dispatch-imei-urn-as-instanceid.





2) Write a new separate draft that tries to deal with the OMA push to talk c=
ase.



This would have the UA include the IMEI in an INVITE request in very particu=
lar instances where the UA knew that request was going to the POC server and=
 that the POC server would remove it. I'm not sure what I would think of thi=
s draft - Cullen might hate it - but regardless of if folks thought it was g=
ood or bad, splitting it into a separate problem can allow the use in the fi=
rst draft to proceed. Cullen would want this draft to be very clear on how a=
 UA knew it was "safe" to put the IMEI in a INVITE and Cullen does not think=
 the IME should ever be in an anonymous INVITE.  The current text around you=
 only include the IMEI if you know it is safe does not seem implementable so=
 Cullen would prefer something where it was really clear when the UA did it=
 and when it did not. Cullen thinks this is much easier to do in the specifi=
c case of push to talk than trying to solve it for all uses of an INVITE so=
 this should be easier to get people on board with than the current draft.



Response to Cullen on the IME should never be in an anonymous INVITE point.=
 - in 3GPP networks there is really no such thing as an anonymous INVITE sin=
ce the P-CSCF always knows the identity of the UE and will always include a=
 P-Asserted-Identity header field. There are instead requests where privacy=
 is requested and the  P-Asserted-Identity header field is removed at the ed=
ge of the trust domain. Thus in my view there is no need for a UE in the PoC=
 case where it knows the PoC Server will remove the instance ID to prohibit=
 inclusion of the IMEI. Cullen any comments to that?



3) Update draft-montemurro-gsma-imei-urn security section to clarify that th=
e IMEI-SV is not updated based on the OS version or when new applications ar=
e added but represents the software version of lower layer software that has=
 completed certification testing.



I am in the process of updating the drafts as described above.


Comments on the above are of course welcome

Andrew

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<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">In order to make progress on draft-montemurro-gsma=
-imei-urn and draft-allen-dispatch-imei-urn-as-instanceid a conference call=
 was held on Wednesday
<o:p></o:p></p>
<p class=3D"MsoPlainText">April 17 attended by Culllen Jennings, Atle Monrad=
, John Luc Bakker and myself.<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">Below are the combined notes and summary from Cull=
en and I of this call on the IMEI drafts:<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">1) change draft-allen-dispatch-imei-urn-as-instanc=
eid to say IMEI is only used in either a register request or also in an emer=
gency invite request
<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">Cullen will then be OK with it from a privacy poin=
t of view as long as it explains the trade offs.
<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">It should be highlighted in draft-montemurro-gsma-=
imei-urn that using the IMEI as a security credential is a bad thing even th=
ough some popular commercial applications are actually doing this and some s=
ervice providers are using knowledge
 of the IMEI to verify the legitimacy of caller being the authorized user/ow=
ner of the device. The IMEI is usually extremely easy to obtain by anyone wh=
o has very brief physical access to the device as it is often printed on the=
 device beneath the battery and
 can be displayed by using a well-known key press sequence.<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">Change draft-allen-dispatch-imei-urn-as-instanceid=
 to not be open to any extension to URN but instead require a draft updating=
 this draft if any parameter other than ver=3D0 was to be included.&nbsp; Al=
so add text in draft-montemurro-gsma-imei-urn
 and, draft-aindicating that extensions to the IMEI parameters will require=
 an update of draft-allen-dispatch-imei-urn-as-instanceid.<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">2) Write a new separate draft that tries to deal w=
ith the OMA push to talk case.
<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">This would have the UA include the IMEI in an INVI=
TE request in very particular instances where the UA knew that request was g=
oing to the POC server and that the POC server would remove it. I'm not sure=
 what I would think of this draft
 - Cullen might hate it - but regardless of if folks thought it was good or=
 bad, splitting it into a separate problem can allow the use in the first dr=
aft to proceed. Cullen would want this draft to be very clear on how a UA kn=
ew it was &quot;safe&quot; to put the IMEI
 in a INVITE and Cullen does not think the IME should ever be in an anonymou=
s INVITE.&nbsp; The current text around you only include the IMEI if you kno=
w it is safe does not seem implementable so Cullen would prefer something wh=
ere it was really clear when the UA
 did it and when it did not. Cullen thinks this is much easier to do in the=
 specific case of push to talk than trying to solve it for all uses of an IN=
VITE so this should be easier to get people on board with than the current d=
raft.
<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">Response to Cullen on the IME should never be in a=
n anonymous INVITE point. - in 3GPP networks there is really no such thing a=
s an anonymous INVITE since the P-CSCF always knows the identity of the UE a=
nd will always include a P-Asserted-Identity
 header field. There are instead requests where privacy is requested and the=
&nbsp; P-Asserted-Identity header field is removed at the edge of the trust=
 domain. Thus in my view there is no need for a UE in the PoC case where it=
 knows the PoC Server will remove the
 instance ID to prohibit inclusion of the IMEI. Cullen any comments to that?=
<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">3) Update draft-montemurro-gsma-imei-urn security=
 section to clarify that the IMEI-SV is not updated based on the OS version=
 or when new applications are added but represents the software version of l=
ower layer software that has completed
 certification testing.<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">I am in the process of updating the drafts as desc=
ribed above.<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Comments on the above are of course welcome<o:p></o:p=
></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Andrew<o:p></o:p></p>
</div>
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From oej@edvina.net  Mon Apr 29 02:31:19 2013
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I don't know if this is the proper mailing list, but I will start here =
anyway :-)

http://tools.ietf.org/html/rfc5626

This URL gives me=20
""" % args) IOError: [Errno 28] No space left on device

/O=

From oej@edvina.net  Mon Apr 29 02:49:37 2013
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Subject: [dispatch] SIP outbound and in-dialog requests
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In the Kamailio project, we've worked on a registrar and edge-proxy =
implementation of SIP outbound (RFC 5626) for a while.

We now stumble into the handling of in-dialog requests.

The RFC is clear how to handle dialog-forming requests and failover in =
regards to these. But it doesn't really say anything
clear on how to handle in-dialog requests in case of a failed flow.

The route set, based on the Path header at registration, now contains a =
flow token for a failed flow. This route set can
not change, even though we know that we have a secondary flow to the =
same instance ID that works (another Reg-id).
The dialog has to die since we can not route according to the route set.

I do remember a discussion about this on a SIPit event but don't =
remember if we found a solution.

Any ideas on how to handle this situation? Anyone working on this issue?

/O=

From keith.drage@alcatel-lucent.com  Mon Apr 29 05:34:38 2013
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Cullen Jennings <fluffy@iii.ca>, "Olle E. Johansson" <oej@edvina.net>
Thread-Topic: [dispatch] Brigining SIP+DANE to Dispatch
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I regarded the destination group as a given, i.e. SIPCORE.

I agree that a draft would be useful to confirm that, but then I see the ke=
y DISPATCH decision as to whether it is dispatchable, rather than where it =
is going.

Keith

> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Cullen Jennings
> Sent: 24 April 2013 15:38
> To: Olle E. Johansson
> Cc: dispatch@ietf.org list
> Subject: [dispatch] Brigining SIP+DANE to Dispatch
>=20
>=20
> I'm keen on someone writing a draft that specifies how to do DANE with SI=
P
> and bringing that draft to dispatch.
>=20
> However, I don't think you want to do that as un update to 5922. I think
> you want it to be just a new draft that is an extension to SIP and does
> not update or replace any of the existing draft.
>=20
> I think Dispatch would be a good place to start the discussion around it
> and decide how if this was a sip core thing, or a new small WG, or
> something else. There are a few different problems that can be tacked. On=
e
> is validating that the TLS connection has the right cert. The other is
> more 4474 style identity issues. I think it will be good to see discussio=
n
> on the dispatch list of the scope of problems we want to solve with DANE.
>=20
> Cullen <with my Dispatch co-chair hat on>
>=20
>=20
>=20
> On Apr 21, 2013, at 5:25 AM, Olle E. Johansson <oej@edvina.net> wrote:
>=20
> > Hi!
> >
> > Looking at DANE for SIP, I mailed out some thoughts on the DANE mailing
> list. If we come to
> > the conclusion that RFC 5922 will need to be updated - will this work b=
e
> done in Dispatch
> > or Dane or somewhere else?
> >
> > DANE also has impacts on SIP Identity as an alternative discovery
> mechanism for
> > domain certificates (replacing HTTPS).
> >
> > /O
> >
> > Vidarebefordrat brev:
> >
> >> Fr=E5n: "Olle E. Johansson" <oej@edvina.net>
> >> =C4mne: DANE SRV draft and SIP
> >> Datum: 21 april 2013 12:27:47 CEST
> >> Till: dane WG list <dane@ietf.org>
> >> Kopia: "Olle E. Johansson" <oej@edvina.net>
> >>
> >> Hi!
> >>
> >> Looking at the DANE SRV draft with my SIP eyes I realize that there's =
a
> lot of work to be done to get this
> >> to work with SIP.
> >>
> >> SIP has no StartTLS, we use NAPTR records to select transport. The
> NAPTR points to a SRV
> >> name that resolves into a set of host names.
> >>
> >> The SIP Domain certificates RFC - http://tools.ietf.org/html/rfc5922 -
> specificies matching a
> >> given SIP URI with a certificate. The matching is done on service
> domain either with
> >> a ALT name URI, like sip:ietf.org or a DNS name, like ietf.org - but
> not the host name.
> >>
> >> I personally agree with the policy in the DANE SRV draft that we shoul=
d
> match on the
> >> SRV hostname used to get A/AAAA records when using DNSsec. For this to
> work,
> >> the path from the SIP URI over NAPTR to SRV and hostnames needs to be
> fully
> >> signed in and verified in the DNS. This is going to require an update
> to 5922.
> >>
> >> The final question is how to handle this without SNI. The certificates
> for both
> >> DANE verification and RFC 5922 plus "old-style" verification with the
> sip
> >> domain in the CN seems like a complicated mess to manage.
> >>
> >> Food for thought on a sunny Sunday.
> >>
> >> The SRV draft is not clear on how to use Subject AltNames of various
> types,
> >> and doesn't mention NAPTR. I am not personally aware of other protocol=
s
> using
> >> this setup, so maybe this requires a very SIP specfic draft, following
> the wake of the
> >> smtp work.
> >>
> >> /O
> >>
> >
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

From mary.ietf.barnes@gmail.com  Mon Apr 29 05:45:58 2013
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I sent an email to ietf-action (@ietf.org) for the problem.

On Mon, Apr 29, 2013 at 4:31 AM, Olle E. Johansson <oej@edvina.net> wrote:
> I don't know if this is the proper mailing list, but I will start here anyway :-)
>
> http://tools.ietf.org/html/rfc5626
>
> This URL gives me
> """ % args) IOError: [Errno 28] No space left on device
>
> /O
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

From pkyzivat@alum.mit.edu  Mon Apr 29 07:24:53 2013
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Subject: Re: [dispatch] Brigining SIP+DANE to Dispatch
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On 4/29/13 8:33 AM, DRAGE, Keith (Keith) wrote:
> I regarded the destination group as a given, i.e. SIPCORE.
>
> I agree that a draft would be useful to confirm that, but then I see the key DISPATCH decision as to whether it is dispatchable, rather than where it is going.

I agree that the point where this should be addressed, if dispatched, is 
sipcore.

	Thanks,
	Paul (as co-chair of sipcore)

> Keith
>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>> Behalf Of Cullen Jennings
>> Sent: 24 April 2013 15:38
>> To: Olle E. Johansson
>> Cc: dispatch@ietf.org list
>> Subject: [dispatch] Brigining SIP+DANE to Dispatch
>>
>>
>> I'm keen on someone writing a draft that specifies how to do DANE with SIP
>> and bringing that draft to dispatch.
>>
>> However, I don't think you want to do that as un update to 5922. I think
>> you want it to be just a new draft that is an extension to SIP and does
>> not update or replace any of the existing draft.
>>
>> I think Dispatch would be a good place to start the discussion around it
>> and decide how if this was a sip core thing, or a new small WG, or
>> something else. There are a few different problems that can be tacked. One
>> is validating that the TLS connection has the right cert. The other is
>> more 4474 style identity issues. I think it will be good to see discussion
>> on the dispatch list of the scope of problems we want to solve with DANE.
>>
>> Cullen <with my Dispatch co-chair hat on>
>>
>>
>>
>> On Apr 21, 2013, at 5:25 AM, Olle E. Johansson <oej@edvina.net> wrote:
>>
>>> Hi!
>>>
>>> Looking at DANE for SIP, I mailed out some thoughts on the DANE mailing
>> list. If we come to
>>> the conclusion that RFC 5922 will need to be updated - will this work be
>> done in Dispatch
>>> or Dane or somewhere else?
>>>
>>> DANE also has impacts on SIP Identity as an alternative discovery
>> mechanism for
>>> domain certificates (replacing HTTPS).
>>>
>>> /O
>>>
>>> Vidarebefordrat brev:
>>>
>>>> Från: "Olle E. Johansson" <oej@edvina.net>
>>>> Ämne: DANE SRV draft and SIP
>>>> Datum: 21 april 2013 12:27:47 CEST
>>>> Till: dane WG list <dane@ietf.org>
>>>> Kopia: "Olle E. Johansson" <oej@edvina.net>
>>>>
>>>> Hi!
>>>>
>>>> Looking at the DANE SRV draft with my SIP eyes I realize that there's a
>> lot of work to be done to get this
>>>> to work with SIP.
>>>>
>>>> SIP has no StartTLS, we use NAPTR records to select transport. The
>> NAPTR points to a SRV
>>>> name that resolves into a set of host names.
>>>>
>>>> The SIP Domain certificates RFC - http://tools.ietf.org/html/rfc5922 -
>> specificies matching a
>>>> given SIP URI with a certificate. The matching is done on service
>> domain either with
>>>> a ALT name URI, like sip:ietf.org or a DNS name, like ietf.org - but
>> not the host name.
>>>>
>>>> I personally agree with the policy in the DANE SRV draft that we should
>> match on the
>>>> SRV hostname used to get A/AAAA records when using DNSsec. For this to
>> work,
>>>> the path from the SIP URI over NAPTR to SRV and hostnames needs to be
>> fully
>>>> signed in and verified in the DNS. This is going to require an update
>> to 5922.
>>>>
>>>> The final question is how to handle this without SNI. The certificates
>> for both
>>>> DANE verification and RFC 5922 plus "old-style" verification with the
>> sip
>>>> domain in the CN seems like a complicated mess to manage.
>>>>
>>>> Food for thought on a sunny Sunday.
>>>>
>>>> The SRV draft is not clear on how to use Subject AltNames of various
>> types,
>>>> and doesn't mention NAPTR. I am not personally aware of other protocols
>> using
>>>> this setup, so maybe this requires a very SIP specfic draft, following
>> the wake of the
>>>> smtp work.
>>>>
>>>> /O
>>>>
>>>
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>


From oej@edvina.net  Mon Apr 29 07:34:30 2013
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To: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [dispatch] Brigining SIP+DANE to Dispatch
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29 apr 2013 kl. 16:24 skrev Paul Kyzivat <pkyzivat@alum.mit.edu>:

> On 4/29/13 8:33 AM, DRAGE, Keith (Keith) wrote:
>> I regarded the destination group as a given, i.e. SIPCORE.
>>=20
>> I agree that a draft would be useful to confirm that, but then I see =
the key DISPATCH decision as to whether it is dispatchable, rather than =
where it is going.
>=20
> I agree that the point where this should be addressed, if dispatched, =
is sipcore.

Sounds like there is a plan. Now let's focus on getting this process =
done :-)

Thanks for all the feedback!

/O
>=20
> 	Thanks,
> 	Paul (as co-chair of sipcore)
>=20
>> Keith
>>=20
>>> -----Original Message-----
>>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] =
On
>>> Behalf Of Cullen Jennings
>>> Sent: 24 April 2013 15:38
>>> To: Olle E. Johansson
>>> Cc: dispatch@ietf.org list
>>> Subject: [dispatch] Brigining SIP+DANE to Dispatch
>>>=20
>>>=20
>>> I'm keen on someone writing a draft that specifies how to do DANE =
with SIP
>>> and bringing that draft to dispatch.
>>>=20
>>> However, I don't think you want to do that as un update to 5922. I =
think
>>> you want it to be just a new draft that is an extension to SIP and =
does
>>> not update or replace any of the existing draft.
>>>=20
>>> I think Dispatch would be a good place to start the discussion =
around it
>>> and decide how if this was a sip core thing, or a new small WG, or
>>> something else. There are a few different problems that can be =
tacked. One
>>> is validating that the TLS connection has the right cert. The other =
is
>>> more 4474 style identity issues. I think it will be good to see =
discussion
>>> on the dispatch list of the scope of problems we want to solve with =
DANE.
>>>=20
>>> Cullen <with my Dispatch co-chair hat on>
>>>=20
>>>=20
>>>=20
>>> On Apr 21, 2013, at 5:25 AM, Olle E. Johansson <oej@edvina.net> =
wrote:
>>>=20
>>>> Hi!
>>>>=20
>>>> Looking at DANE for SIP, I mailed out some thoughts on the DANE =
mailing
>>> list. If we come to
>>>> the conclusion that RFC 5922 will need to be updated - will this =
work be
>>> done in Dispatch
>>>> or Dane or somewhere else?
>>>>=20
>>>> DANE also has impacts on SIP Identity as an alternative discovery
>>> mechanism for
>>>> domain certificates (replacing HTTPS).
>>>>=20
>>>> /O
>>>>=20
>>>> Vidarebefordrat brev:
>>>>=20
>>>>> Fr=E5n: "Olle E. Johansson" <oej@edvina.net>
>>>>> =C4mne: DANE SRV draft and SIP
>>>>> Datum: 21 april 2013 12:27:47 CEST
>>>>> Till: dane WG list <dane@ietf.org>
>>>>> Kopia: "Olle E. Johansson" <oej@edvina.net>
>>>>>=20
>>>>> Hi!
>>>>>=20
>>>>> Looking at the DANE SRV draft with my SIP eyes I realize that =
there's a
>>> lot of work to be done to get this
>>>>> to work with SIP.
>>>>>=20
>>>>> SIP has no StartTLS, we use NAPTR records to select transport. The
>>> NAPTR points to a SRV
>>>>> name that resolves into a set of host names.
>>>>>=20
>>>>> The SIP Domain certificates RFC - =
http://tools.ietf.org/html/rfc5922 -
>>> specificies matching a
>>>>> given SIP URI with a certificate. The matching is done on service
>>> domain either with
>>>>> a ALT name URI, like sip:ietf.org or a DNS name, like ietf.org - =
but
>>> not the host name.
>>>>>=20
>>>>> I personally agree with the policy in the DANE SRV draft that we =
should
>>> match on the
>>>>> SRV hostname used to get A/AAAA records when using DNSsec. For =
this to
>>> work,
>>>>> the path from the SIP URI over NAPTR to SRV and hostnames needs to =
be
>>> fully
>>>>> signed in and verified in the DNS. This is going to require an =
update
>>> to 5922.
>>>>>=20
>>>>> The final question is how to handle this without SNI. The =
certificates
>>> for both
>>>>> DANE verification and RFC 5922 plus "old-style" verification with =
the
>>> sip
>>>>> domain in the CN seems like a complicated mess to manage.
>>>>>=20
>>>>> Food for thought on a sunny Sunday.
>>>>>=20
>>>>> The SRV draft is not clear on how to use Subject AltNames of =
various
>>> types,
>>>>> and doesn't mention NAPTR. I am not personally aware of other =
protocols
>>> using
>>>>> this setup, so maybe this requires a very SIP specfic draft, =
following
>>> the wake of the
>>>>> smtp work.
>>>>>=20
>>>>> /O
>>>>>=20
>>>>=20
>>>> _______________________________________________
>>>> dispatch mailing list
>>>> dispatch@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>=20
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>=20
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 29 Apr 2013 21:47:31 +0200
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Subject: Re: [dispatch] SIP outbound and in-dialog requests
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2013/4/29 Olle E. Johansson <oej@edvina.net>:
> The route set, based on the Path header at registration, now contains a f=
low token for a failed flow. This route set can
> not change, even though we know that we have a secondary flow to the same=
 instance ID that works (another Reg-id).
> The dialog has to die since we can not route according to the route set.
>
> I do remember a discussion about this on a SIPit event but don't remember=
 if we found a solution.
>
> Any ideas on how to handle this situation? Anyone working on this issue?

The only solution is using GRUU so if the client looses the connection
during a dialog, it registers again and the location-server (i.e. the
registrar) would route in-dialog requests via the new Route set.

Tested ;)


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

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29 apr 2013 kl. 21:47 skrev I=F1aki Baz Castillo <ibc@aliax.net>:

> 2013/4/29 Olle E. Johansson <oej@edvina.net>:
>> The route set, based on the Path header at registration, now contains =
a flow token for a failed flow. This route set can
>> not change, even though we know that we have a secondary flow to the =
same instance ID that works (another Reg-id).
>> The dialog has to die since we can not route according to the route =
set.
>>=20
>> I do remember a discussion about this on a SIPit event but don't =
remember if we found a solution.
>>=20
>> Any ideas on how to handle this situation? Anyone working on this =
issue?
>=20
> The only solution is using GRUU so if the client looses the connection
> during a dialog, it registers again and the location-server (i.e. the
> registrar) would route in-dialog requests via the new Route set.

Which means that you actually change the in-dialog route set?

/O=

From ibc@aliax.net  Mon Apr 29 13:49:49 2013
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Subject: Re: [dispatch] SIP outbound and in-dialog requests
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Yes! And it breaks RFC3261 :)

--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>
El 29/04/2013 21:56, "Olle E. Johansson" <oej@edvina.net> escribi=C3=B3:

>
> 29 apr 2013 kl. 21:47 skrev I=C3=B1aki Baz Castillo <ibc@aliax.net>:
>
> > 2013/4/29 Olle E. Johansson <oej@edvina.net>:
> >> The route set, based on the Path header at registration, now contains =
a
> flow token for a failed flow. This route set can
> >> not change, even though we know that we have a secondary flow to the
> same instance ID that works (another Reg-id).
> >> The dialog has to die since we can not route according to the route se=
t.
> >>
> >> I do remember a discussion about this on a SIPit event but don't
> remember if we found a solution.
> >>
> >> Any ideas on how to handle this situation? Anyone working on this issu=
e?
> >
> > The only solution is using GRUU so if the client looses the connection
> > during a dialog, it registers again and the location-server (i.e. the
> > registrar) would route in-dialog requests via the new Route set.
>
> Which means that you actually change the in-dialog route set?
>
> /O

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<p dir=3D"ltr">Yes! And it breaks RFC3261 :)<br></p>
<p dir=3D"ltr">--<br>
I=C3=B1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;</p>
<div class=3D"gmail_quote">El 29/04/2013 21:56, &quot;Olle E. Johansson&quo=
t; &lt;<a href=3D"mailto:oej@edvina.net">oej@edvina.net</a>&gt; escribi=C3=
=B3:<br type=3D"attribution"><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
29 apr 2013 kl. 21:47 skrev I=C3=B1aki Baz Castillo &lt;<a href=3D"mailto:i=
bc@aliax.net">ibc@aliax.net</a>&gt;:<br>
<br>
&gt; 2013/4/29 Olle E. Johansson &lt;<a href=3D"mailto:oej@edvina.net">oej@=
edvina.net</a>&gt;:<br>
&gt;&gt; The route set, based on the Path header at registration, now conta=
ins a flow token for a failed flow. This route set can<br>
&gt;&gt; not change, even though we know that we have a secondary flow to t=
he same instance ID that works (another Reg-id).<br>
&gt;&gt; The dialog has to die since we can not route according to the rout=
e set.<br>
&gt;&gt;<br>
&gt;&gt; I do remember a discussion about this on a SIPit event but don&#39=
;t remember if we found a solution.<br>
&gt;&gt;<br>
&gt;&gt; Any ideas on how to handle this situation? Anyone working on this =
issue?<br>
&gt;<br>
&gt; The only solution is using GRUU so if the client looses the connection=
<br>
&gt; during a dialog, it registers again and the location-server (i.e. the<=
br>
&gt; registrar) would route in-dialog requests via the new Route set.<br>
<br>
Which means that you actually change the in-dialog route set?<br>
<br>
/O</blockquote></div>

--047d7b5d57dae3376104db860575--

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Subject: Re: [dispatch] SIP outbound and in-dialog requests
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29 apr 2013 kl. 22:49 skrev I=F1aki Baz Castillo <ibc@aliax.net>:

> Yes! And it breaks RFC3261 :)
>=20
That's what we agreed on the Kamailio developer mailing lists too.

Is this the only response that is out there? To break RFC 3261 and apply =
a route change to an existing dialog?
Or follow RFC 3261 and let the dialog fail?

/O
> --
> I=F1aki Baz Castillo
> <ibc@aliax.net>
>=20
> El 29/04/2013 21:56, "Olle E. Johansson" <oej@edvina.net> escribi=F3:
>=20
> 29 apr 2013 kl. 21:47 skrev I=F1aki Baz Castillo <ibc@aliax.net>:
>=20
> > 2013/4/29 Olle E. Johansson <oej@edvina.net>:
> >> The route set, based on the Path header at registration, now =
contains a flow token for a failed flow. This route set can
> >> not change, even though we know that we have a secondary flow to =
the same instance ID that works (another Reg-id).
> >> The dialog has to die since we can not route according to the route =
set.
> >>
> >> I do remember a discussion about this on a SIPit event but don't =
remember if we found a solution.
> >>
> >> Any ideas on how to handle this situation? Anyone working on this =
issue?
> >
> > The only solution is using GRUU so if the client looses the =
connection
> > during a dialog, it registers again and the location-server (i.e. =
the
> > registrar) would route in-dialog requests via the new Route set.
>=20
> Which means that you actually change the in-dialog route set?
>=20
> /O


--Apple-Mail=_A11FBC95-D7E5-4927-96E3-FB2846834428
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	charset=iso-8859-1

<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Diso-8859-1"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><br><div><div>29 apr 2013 kl. 22:49 skrev I=F1aki Baz Castillo &lt;<a =
href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;:</div><br><blockquote =
type=3D"cite"><p dir=3D"ltr">Yes! And it breaks RFC3261 =
:)<br></p></blockquote><div>That's what we agreed on the Kamailio =
developer mailing lists too.</div><div><br></div>Is this the only =
response that is out there? To break RFC 3261 and apply a route change =
to an existing dialog?</div><div>Or follow RFC 3261 and let the dialog =
fail?</div><div><br></div><div>/O<br><blockquote type=3D"cite"><p =
dir=3D"ltr">--<br>
I=F1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;</p>
<div class=3D"gmail_quote">El 29/04/2013 21:56, "Olle E. Johansson" =
&lt;<a href=3D"mailto:oej@edvina.net">oej@edvina.net</a>&gt; =
escribi=F3:<br type=3D"attribution"><blockquote class=3D"gmail_quote" =
style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
29 apr 2013 kl. 21:47 skrev I=F1aki Baz Castillo &lt;<a =
href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;:<br>
<br>
&gt; 2013/4/29 Olle E. Johansson &lt;<a =
href=3D"mailto:oej@edvina.net">oej@edvina.net</a>&gt;:<br>
&gt;&gt; The route set, based on the Path header at registration, now =
contains a flow token for a failed flow. This route set can<br>
&gt;&gt; not change, even though we know that we have a secondary flow =
to the same instance ID that works (another Reg-id).<br>
&gt;&gt; The dialog has to die since we can not route according to the =
route set.<br>
&gt;&gt;<br>
&gt;&gt; I do remember a discussion about this on a SIPit event but =
don't remember if we found a solution.<br>
&gt;&gt;<br>
&gt;&gt; Any ideas on how to handle this situation? Anyone working on =
this issue?<br>
&gt;<br>
&gt; The only solution is using GRUU so if the client looses the =
connection<br>
&gt; during a dialog, it registers again and the location-server (i.e. =
the<br>
&gt; registrar) would route in-dialog requests via the new Route =
set.<br>
<br>
Which means that you actually change the in-dialog route set?<br>
<br>
/O</blockquote></div>
</blockquote></div><br></body></html>=

--Apple-Mail=_A11FBC95-D7E5-4927-96E3-FB2846834428--

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 30 Apr 2013 11:41:43 +0200
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Subject: Re: [dispatch] SIP outbound and in-dialog requests
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2013/4/30 Olle E. Johansson <oej@edvina.net>:
> Is this the only response that is out there? To break RFC 3261 and apply =
a
> route change to an existing dialog?

Is not cool enough? :)

Right, RFC 5627 should say "something" about it.


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