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Subject: [dispatch] PLEASE NOTE (Fwd:  IETF-76 DISPATCH WG deadlines)
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(apologies for cross-post: reply-to set to dispatch)

The first deadline below is next Monday. Please get in touch with Mary  
and Gonzalo if you are going to have trouble hitting that date.

RjS

Begin forwarded message:

> From: "Mary Barnes" <mary.barnes@nortel.com>
> Date: August 21, 2009 11:25:16 AM CDT
> To: <dispatch@ietf.org>
> Cc: rai-ads@tools.ietf.org
> Subject: [dispatch] IETF-76 DISPATCH WG deadlines
>
> Hi all,
>
> As was done for IETF-75, we are setting earlier deadlines for  
> discussing topics in DISPATCH prior to the WG session at IETF-76.
>
> We will again follow the deadlines for BoFs:
> http://www.ietf.org/meeting/cutoff-dates.html#76
>
> This provides enough time to dispatch topics prior to the meeting as  
> appropriate - e.g., mini-bofs, official bofs, other WGs, new WGs,   
> mailing lists, etc. Thus, allowing us to have more focused and  
> constructive discussions on a smaller set of topics prior to the WG  
> session.
>
> Please note that the dates are much sooner than you might expect as  
> the time between IETF-75 and IETF-76 is 3 weeks less than the time  
> between IETF-74 and IETF-75. The document deadlines are 9 and 10  
> weeks away from next Monday (August 24th).
>
>
> Thus, the following are the proposed deadlines:
>
> Sept 7th  - Cutoff date to notify the chairs and DISPATCH WG mailing  
> list of plans to submit a proposal.
> --------------------------------------------------------------------------------------------------------
>
> This will help folks with similar topics to work together before the  
> meeting. If a preliminary charter proposal is available at this  
> point, please include it.
>
>
> September 21st - Cutoff for charter proposals for topics.
> --------------------------------------------------------
>
> A charter proposal must consist of a minimum of a problem statement  
> and at least one milestone/deliverable. This deadline will allow  
> time for consideration of proposals that could potentially be  
> "dispatched" prior to the WG session.
>
>
> September 28th - Topics that are to be the focus of IETF-76 are  
> announced.
> ---------------------------------------------------------------------------
>
> The idea here is to focus the discussion over the weeks preceding  
> IETF-76 on these main topics, noting that any new or updates to any  
> documents are due 3-4 weeks later.  This will ensure we have  
> constructive discussions before the meeting and are actually able to  
> determine consensus as to where work should be progressed (e.g.,  
> separate BoF, a new WG, an existing WG, individual document, etc.)  
> at IETF-76. Note, that the exact disposition for a topic may (per  
> the usual process) require follow-up and confirmation by the ADS and/ 
> or IESG (e.g., for a new WG, Bof, individually sponsored document,  
> etc.) or with another WG to ensure agreement with the DISPATCH WG  
> consensus for the topic.
>
>
> As discussed previously, the intention is not necessarily to  
> preclude folks submitting documents on other topics, however, it is  
> very unlikely there would be agenda time allocated to documents/ 
> topics submitted after the deadlines. We can include one or two  
> slides on these topics in the DISPATCH WG chair charts so that the  
> authors can gather other interested parties to contribute to the work.
>
> Regards,
> Mary and Gonzalo
> DISPATCH WG co-chairs
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><div>(apologies for cross-post: =
reply-to set to dispatch)</div><div><br></div>The first deadline below =
is next Monday. Please get in touch with Mary =
and&nbsp;Gonzalo&nbsp;if&nbsp;you&nbsp;are&nbsp;going&nbsp;to&nbsp;have&nb=
sp;trouble&nbsp;hitting&nbsp;that&nbsp;date.<div><br></div><div>RjS<br><di=
v><br><div>Begin forwarded message:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" color=3D"#000000" =
style=3D"font: 12.0px Helvetica; color: #000000"><b>From: =
</b></font><font face=3D"Helvetica" size=3D"3" style=3D"font: 12.0px =
Helvetica">"Mary Barnes" &lt;<a =
href=3D"mailto:mary.barnes@nortel.com">mary.barnes@nortel.com</a>&gt;</fon=
t></div><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: =
0px; margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" =
color=3D"#000000" style=3D"font: 12.0px Helvetica; color: =
#000000"><b>Date: </b></font><font face=3D"Helvetica" size=3D"3" =
style=3D"font: 12.0px Helvetica">August 21, 2009 11:25:16 AM =
CDT</font></div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><font face=3D"Helvetica" =
size=3D"3" color=3D"#000000" style=3D"font: 12.0px Helvetica; color: =
#000000"><b>To: </b></font><font face=3D"Helvetica" size=3D"3" =
style=3D"font: 12.0px Helvetica">&lt;<a =
href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a>&gt;</font></div><d=
iv style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" color=3D"#000000" =
style=3D"font: 12.0px Helvetica; color: #000000"><b>Cc: </b></font><font =
face=3D"Helvetica" size=3D"3" style=3D"font: 12.0px Helvetica"><a =
href=3D"mailto:rai-ads@tools.ietf.org">rai-ads@tools.ietf.org</a></font></=
div><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: =
0px; margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" =
color=3D"#000000" style=3D"font: 12.0px Helvetica; color: =
#000000"><b>Subject: </b></font><font face=3D"Helvetica" size=3D"3" =
style=3D"font: 12.0px Helvetica"><b>[dispatch] IETF-76 DISPATCH WG =
deadlines</b></font></div><div style=3D"margin-top: 0px; margin-right: =
0px; margin-bottom: 0px; margin-left: 0px; min-height: 14px; =
"><br></div> </div> <div> <!-- Converted from text/rtf format =
--><p><font size=3D"2" face=3D"Courier New">Hi all,</font> </p><p><font =
size=3D"2" face=3D"Courier New">As was done for IETF-75, we are setting =
earlier deadlines for discussing topics in DISPATCH prior to the WG =
session at IETF-76.&nbsp; </font></p><p><font size=3D"2" face=3D"Courier =
New">We will again follow the deadlines for BoFs:</font> <br><a =
href=3D"http://www.ietf.org/meeting/cutoff-dates.html#76"><u><font =
color=3D"#0000FF" size=3D"2" face=3D"Courier =
New">http://www.ietf.org/meeting/cutoff-dates.html#76</font></u></a> =
</p><p><font size=3D"2" face=3D"Courier New">This provides enough time =
to dispatch topics prior to the meeting as appropriate - e.g., =
mini-bofs, official bofs, other WGs, new WGs,&nbsp; mailing lists, etc. =
Thus, allowing us to have more focused and constructive discussions on a =
smaller set of topics prior to the WG session. </font></p><p><font =
size=3D"2" face=3D"Courier New">Please note that the dates are much =
sooner than you might expect as the time between IETF-75 and IETF-76 is =
3 weeks less than the time between IETF-74 and IETF-75. The document =
deadlines are 9 and 10 weeks away from next Monday (August 24th).&nbsp; =
</font></p> <br><p><font size=3D"2" face=3D"Courier New">Thus, the =
following are the proposed deadlines:</font> </p><p><font size=3D"2" =
face=3D"Courier New">Sept 7th&nbsp; - Cutoff date to notify the chairs =
and DISPATCH WG mailing list of plans to submit a proposal.</font> =
<br><font size=3D"2" face=3D"Courier =
New">---------------------------------------------------------------------=
----------------------------------- </font> </p><p><font size=3D"2" =
face=3D"Courier New">This will help folks with similar topics to work =
together before the meeting. If a preliminary charter proposal is =
available at this point, please include it.</font></p> <br><p><font =
size=3D"2" face=3D"Courier New">September 21st - Cutoff for charter =
proposals for topics. </font> <br><font size=3D"2" face=3D"Courier =
New">--------------------------------------------------------</font> =
</p><p><font size=3D"2" face=3D"Courier New">A charter proposal must =
consist of a minimum of a problem statement and at least one =
milestone/deliverable. This deadline will allow time for consideration =
of proposals that could potentially be "dispatched" prior to the WG =
session.</font></p> <br><p><font size=3D"2" face=3D"Courier =
New">September 28th - Topics that are to be the focus of IETF-76 are =
announced. </font> <br><font size=3D"2" face=3D"Courier =
New">---------------------------------------------------------------------=
------ </font> </p><p><font size=3D"2" face=3D"Courier New">The idea =
here is to focus the discussion over the weeks preceding IETF-76 on =
these main topics, noting that any new or updates to any documents are =
due 3-4 weeks later.&nbsp; This will ensure we have constructive =
discussions before the meeting and are actually able to determine =
consensus as to where work should be progressed (e.g., separate BoF, a =
new WG, an existing WG, individual document, etc.) at IETF-76. Note, =
that the exact disposition for a topic may (per the usual process) =
require follow-up and confirmation by the ADS and/or IESG (e.g., for a =
new WG, Bof, individually sponsored document, etc.) or with another WG =
to ensure agreement with the DISPATCH WG consensus for the =
topic.</font></p> <br><p><font size=3D"2" face=3D"Courier New">As =
discussed previously, the intention is not necessarily to preclude folks =
submitting documents on other topics, however, it is very unlikely there =
would be agenda time allocated to documents/topics submitted after the =
deadlines. We can include one or two slides on these topics in the =
DISPATCH WG chair charts so that the authors can gather other interested =
parties to contribute to the work. </font></p><p><font size=3D"2" =
face=3D"Courier New">Regards,</font> <br><font size=3D"2" face=3D"Courier =
New">Mary and Gonzalo</font> <br><font size=3D"2" face=3D"Courier =
New">DISPATCH WG co-chairs </font> </p> </div> =
_______________________________________________<br>dispatch mailing =
list<br><a =
href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a><br>https://www.iet=
f.org/mailman/listinfo/dispatch<br></blockquote></div><br></div></body></h=
tml>=

--Apple-Mail-292-860984748--

From alan@sipstation.com  Wed Sep  2 15:12:52 2009
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Date: Wed, 02 Sep 2009 16:51:50 -0500
From: Alan Johnston <alan@sipstation.com>
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Subject: [dispatch] Call Control UUI for SIP Issue
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Based on feedback during my presentation in Stockholm, I think I  have a 
good idea how to modify the proposed charter text.

One issue perhaps needs a little more discussion so I'd like to kick 
that off here.

The question is, for some kind of information, when should a new SIP 
header field be used, when should this UUI transport mechanism be used, 
and when should a proprietary header field be used.  It seemed like we  
had some differing opinions on this with respect to UUI.  I think which 
approach depends on the information being transported.  Here's my take 
on this:

New SIP header field to transport user information.  I think a new SIP 
header field is needed when:
1. SIP behavior is changed
2. The information is generated and consumed at the SIP layer by SIP 
elements
3. SIP elements besides the UAC and UAS might be interested and consume 
the information
4. There are specific privacy or cross-RAI issues involved with the 
information being transported
5. Interoperability is important

UUI Transport mechanism.  I think this mechanism is suitable when:
1. The information is generated and consumed by an application using SIP 
during session setup but not necessarily even SIP aware.
2. SIP behavior is unchanged
3. Generally only the UAC and UAS are interested in the information
4. The information is expected to survive retargeting, redirection, and 
retargeting
5. SIP elements may need to apply policy about passing and screening the 
information
6. Multivendor interop is important

Proprietary header field (the old P-header)
1. The information is only of interest to UAC and UAS
2. No SIP elements need apply policy to the information
3. Single vendor or closed system application

If we disagree strongly on these points, lets have that discussion now, 
and this may help us formulate charter text about the mechanism itself.

Thanks,
Alan


From pkyzivat@cisco.com  Thu Sep  3 08:43:02 2009
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From: Paul Kyzivat <pkyzivat@cisco.com>
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Cc: "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] Call Control UUI for SIP Issue
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Alan,

I agree it is a good idea to partition the use cases in this way.

I'm a little confused about the criteria you give for UUI below, because
- I thought one of the reasons originally given for the proposed
   mechanism was that it was needed before call establishment for
   purpose of making routing decisions. If so, that means that it
   impacts sip behavior.

- Also, isn't the applying of policy by sip elements also a change
   to sip behavior?

	Thanks,
	Paul

Alan Johnston wrote:
> Based on feedback during my presentation in Stockholm, I think I  have a 
> good idea how to modify the proposed charter text.
> 
> One issue perhaps needs a little more discussion so I'd like to kick 
> that off here.
> 
> The question is, for some kind of information, when should a new SIP 
> header field be used, when should this UUI transport mechanism be used, 
> and when should a proprietary header field be used.  It seemed like we  
> had some differing opinions on this with respect to UUI.  I think which 
> approach depends on the information being transported.  Here's my take 
> on this:
> 
> New SIP header field to transport user information.  I think a new SIP 
> header field is needed when:
> 1. SIP behavior is changed
> 2. The information is generated and consumed at the SIP layer by SIP 
> elements
> 3. SIP elements besides the UAC and UAS might be interested and consume 
> the information
> 4. There are specific privacy or cross-RAI issues involved with the 
> information being transported
> 5. Interoperability is important
> 
> UUI Transport mechanism.  I think this mechanism is suitable when:
> 1. The information is generated and consumed by an application using SIP 
> during session setup but not necessarily even SIP aware.
> 2. SIP behavior is unchanged
> 3. Generally only the UAC and UAS are interested in the information
> 4. The information is expected to survive retargeting, redirection, and 
> retargeting
> 5. SIP elements may need to apply policy about passing and screening the 
> information
> 6. Multivendor interop is important
> 
> Proprietary header field (the old P-header)
> 1. The information is only of interest to UAC and UAS
> 2. No SIP elements need apply policy to the information
> 3. Single vendor or closed system application
> 
> If we disagree strongly on these points, lets have that discussion now, 
> and this may help us formulate charter text about the mechanism itself.
> 
> Thanks,
> Alan
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

From alan@sipstation.com  Thu Sep  3 13:26:50 2009
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Paul,

Thanks for your comments - see mine below.

Thanks,
Alan

Paul Kyzivat wrote:
> Alan,
>
> I agree it is a good idea to partition the use cases in this way.
>
> I'm a little confused about the criteria you give for UUI below, because
> - I thought one of the reasons originally given for the proposed
>   mechanism was that it was needed before call establishment for
>   purpose of making routing decisions. If so, that means that it
>   impacts sip behavior.

The information does need to be rendered and parsed at the time of call 
establishment - that is why it is carried in an INVITE.  But it is not 
really used for SIP routing decisions.  I suppose it could be used in 
deciding whether to answer a call or reject it, but again I'd say that 
is not SIP behavior.

>
> - Also, isn't the applying of policy by sip elements also a change
>   to sip behavior?

No, the policy is just whether to pass the information element up the 
protocol stack or whether to block its  transport within a domain.  The 
policy isn't, because this type of UUI is present, do some special SIP 
operation or invoke a special SIP feature.

So, here's an example of what I mean by "changing SIP behavior".  Lets 
say we had this UUI mechanism but we didn't have the Refer-Sub: no 
header field.  One could argue that suppressing the REFER implicit 
subscription is just a piece of user information, which is generated by 
the UAC and consumed by the UAS, so why not transport it using the UUI 
mechanism.  Well, this information changes SIP behavior in that no 
NOTIFY is sent.  Hence, this would not be a suitable usage for this UUI 
mechanism.  

>
>     Thanks,
>     Paul
>
> Alan Johnston wrote:
>> Based on feedback during my presentation in Stockholm, I think I  
>> have a good idea how to modify the proposed charter text.
>>
>> One issue perhaps needs a little more discussion so I'd like to kick 
>> that off here.
>>
>> The question is, for some kind of information, when should a new SIP 
>> header field be used, when should this UUI transport mechanism be 
>> used, and when should a proprietary header field be used.  It seemed 
>> like we  had some differing opinions on this with respect to UUI.  I 
>> think which approach depends on the information being transported.  
>> Here's my take on this:
>>
>> New SIP header field to transport user information.  I think a new 
>> SIP header field is needed when:
>> 1. SIP behavior is changed
>> 2. The information is generated and consumed at the SIP layer by SIP 
>> elements
>> 3. SIP elements besides the UAC and UAS might be interested and 
>> consume the information
>> 4. There are specific privacy or cross-RAI issues involved with the 
>> information being transported
>> 5. Interoperability is important
>>
>> UUI Transport mechanism.  I think this mechanism is suitable when:
>> 1. The information is generated and consumed by an application using 
>> SIP during session setup but not necessarily even SIP aware.
>> 2. SIP behavior is unchanged
>> 3. Generally only the UAC and UAS are interested in the information
>> 4. The information is expected to survive retargeting, redirection, 
>> and retargeting
>> 5. SIP elements may need to apply policy about passing and screening 
>> the information
>> 6. Multivendor interop is important
>>
>> Proprietary header field (the old P-header)
>> 1. The information is only of interest to UAC and UAS
>> 2. No SIP elements need apply policy to the information
>> 3. Single vendor or closed system application
>>
>> If we disagree strongly on these points, lets have that discussion 
>> now, and this may help us formulate charter text about the mechanism 
>> itself.
>>
>> Thanks,
>> Alan
>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
>


From scott.lawrence@nortel.com  Thu Sep  3 13:38:36 2009
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On Fri, 2009-08-21 at 11:25 -0500, Mary Barnes wrote:

> Sept 7th  - Cutoff date to notify the chairs and DISPATCH WG mailing
> list of plans to submit a proposal. 

I plan to respin and expand my draft problem statement on third party
authentication, and would like to plan time to discuss it.



From dyork@voxeo.com  Thu Sep  3 14:21:01 2009
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Victor and John,

In getting caught up on all that's been going on in DISPATCH over the  
past bit, I will only say that *yes*, I think this is both helpful and  
going in the right direction.  Please do continue your work on it.  I  
do not currently have further feedback beyond saying that.

Thanks for doing this,
Dan
On Jul 21, 2009, at 12:17 PM, Victor Pascual Avila wrote:

> As mentioned in the below message, the authors would appreciate
> feedback as to whether this is helpful and going in the right
> direction.
>
> Thanks,
> -Victor
>
> 2009/6/29 Elwell, John <john.elwell@siemens-enterprise.com>:
>> There was an extensive discussion at IETF#74 on SIP Identity, where  
>> a good many views were put forward. The only consensus seemed to be  
>> to continue to discuss the topic.
>>
>> One of the themes during that discussion was to focus more on  
>> requirements, which the authors have attempted to do in this new  
>> draft:
>> http://www.ietf.org/internet-drafts/draft-elwell-dispatch-identity-reqs-00.txt
>>
>> We would appreciate feedback as to whether this is helpful and  
>> going in the right direction, as well as detailed comments.
>>
>> I had hoped to do this a lot earlier, to trigger a discussion in  
>> time to get something set up for DISPATCH at IETF#75, but I failed,  
>> and also I failed to talk to the many of you who have interest in  
>> the topic and expressed opinions in the past. But since the  
>> deadline is approaching, I thought it best to post the draft and  
>> let discussion continue on that basis.
>>
>> John
>>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

-- 
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From gonzalo.camarillo@ericsson.com  Fri Sep  4 00:54:36 2009
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FYI.

Gonzalo

-------- Original Message --------
Subject: 	Nomcom 2009-10: Reminder Call for Nominations, Local Office 
hours, Nominee Questionnaires available
Date: 	Mon, 31 Aug 2009 15:23:03 -0700 (PDT)
From: 	Mary Barnes <mary.barnes@nortel.com>
To: 	IETF Announcement list <ietf-announce@ietf.org>
CC: 	ietf@ietf.org


Hi all,

This email covers 3 topics:
1) Reminder: Call for Nominations
2) New: Local Office Hours
3) New: Questionnaires available for nominees

Best Regards,
Mary Barnes
nomcom-chair@ietf.org
mary.h.barnes@gmail.com
mary.barnes@nortel.com


===============================================================================
1) Call for Nominations
------------------------
The nomination period ends in less than 3 weeks on Sept. 18th, 2009. We
appreciate the folks that have taken the time to make nominations thus
far. But, we do need more nominations. Please use the online tool to
nominate - it's easy and secure:
https://wiki.tools.ietf.org/group/nomcom/09/nominate

Details on the open positions, as well as other details and options for
making nominations are available on the Nomcom homepage:
https://wiki.tools.ietf.org/group/nomcom/09/

Please consider that the sooner you make the nominations, the more time
your nominee(s) will have to complete the necessary questionnaire (item 3
below).  As well, please consider nominating more than one person for a
particular position. You will have the opportunity to provide additional
feedback later and it's important to consider that not all nominees will
be able to accept the nomination.


2) Local office hours
-----------------------

In order to facilitate additional feedback, the Nomcom is planning to
make themselves available for office areas at various geographic locations
for 3 weeks in September, starting on the 8th.

Below please find the list of locations where Nomcom members will be
available for these f2f meetings . Unless dates are identified below, the
Nomcom member is generally available for part of the day during the weeks
of Sept 8-11, Sept 14-18 and Sept 21-25. Also, languages other than
English in which the Nomcom member is fluent are identfied.

Please contact a Nomcom member in a specific geographic location to
arrange a convenient meeting time and place. Most Nomcom members are
flexible as to meeting locations - i.e., we can travel to your office,
meet at our offices or somewhere in between.

As a reminder folks can always contact any Nomcom member to provide
feedback at anytime - i.e., you don't need to participate in these f2f
sessions to provide feedback.


Belgium:
      Dimitri Papadimitriou - dimitri.papadimitriou@alcatel-lucent.be (Sept
21-25) (Languages: French)

Boston, Mass, USA:
      Stephen Kent - kent@bbn.com  (Sept 16-18)

Boulder, CO, USA:
      Wassim Haddad - wmhaddad@gmail.com (Sept 14-18)

Dallas/Ft. Worth, TX, USA:
      Mary Barnes  - mary.h.barnes@gmail.com
      Lucy Yong - lucyyong@huawei.com  (Languages: Chinese)

Helsinki, Finland:
       Simo Veikkolainen - simo.veikkolainen@nokia.com (Languages: Finnish)


Ithaca, NY, USA:
       Scott Brim - scott.brim@gmail.com

Montevideo, Uruguay:
       Roque Gagliano - roque@lacnic.net (Sept 14-18, 21-25) (Languages:
Spanish, Portuguese)

Montreal, Quebec, Canada
      Wassim Haddad - wmhaddad@gmail.com (Sept 8-11)
      -- Can also be available in Ottawa if folks are interested

Paris, France:
       Dimitri Papadimitriou - dimitri.papadimitriou@alcatel-lucent.be
(Sept 15-18)  (Languages: French)

San Diego, CA, USA:
       Randall Gellens - rg+ietf@qualcomm.com
       Dave Crocker - dcrocker@bbiw.net  (Sept 16-18)

Silicon Valley/SF Bay,  CA, USA:
       Dave Crocker - dcrocker@bbiw.net  (Sept 8-11, Sept 14-15, Sept
21-25)
       Dorothy Gellert - Dorothy.gellert@gmail.com

3) Questionnaires available for nominees:

For folks that have been notified that they have been nominated for any
of the positions, the questionnaires are now available on the Nomcom09
tools website:
https://wiki.tools.ietf.org/group/nomcom/09/iab-questionnaire
https://wiki.tools.ietf.org/group/nomcom/09/iaoc-questionnaire
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From Simo.Veikkolainen@nokia.com  Fri Sep  4 05:10:13 2009
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From: <Simo.Veikkolainen@nokia.com>
To: <dispatch@ietf.org>, <mary.barnes@nortel.com>, <gonzalo.camarillo@ericsson.com>
Date: Fri, 4 Sep 2009 14:09:04 +0200
Thread-Topic: New topic proposal for DISPATCH
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Cc: Markus.Isomaki@nokia.com
Subject: [dispatch] New topic proposal for DISPATCH
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--_000_B23B311878A0B4438F5F09F47E01AAE03B10AD6263NOKEUMSG01mgd_
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Hi,

We would like to propose a new DISPATCH topic on how SIP and XMPP can be us=
ed together in a complementary fashion.

We have been working on a solution which combines SIP based VoIP and XMPP b=
ased IM and Presence. The requirements and a proposed solution is outlined =
in http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01.

The motivation for this work comes from experience which shows that most st=
andards based VoIP deployments use SIP. At the same time, the Extensible Me=
ssaging and Presence Protocol (XMPP) is widely used in instant messaging an=
d presence services. An interesting scenario arises when a SIP based voice =
(and video) service is combined together with an XMPP based instant messagi=
ng and presence service.

Note that the goal in this work is not SIP-XMPP protocol translation, but t=
o define protocol extensions and best practises such that SIP based VoIP an=
d XMPP based IM and presence could be seamlessly combined and offered to th=
e end user. For rapid deployment, we assume no changes in the server infras=
tructure, and instead impose new requirements and protocol extensions only =
to the endpoints.

We would like to get some discussion going on the use case itself as well a=
s on the solution. Also, we would like to hear you thoughts on DISPATCH or =
a new "dispatched" mini-WG as the home for such work.

Exact charter proposal and problem statemen is to follow.

Regards,
Simo and Markus



--_000_B23B311878A0B4438F5F09F47E01AAE03B10AD6263NOKEUMSG01mgd_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
<meta name=3D"Generator" content=3D"Microsoft Exchange Server">
<!-- converted from rtf -->
<style><!-- .EmailQuote { margin-left: 1pt; padding-left: 4pt; border-left:=
 #800000 2px solid; } --></style>
</head>
<body>
<font face=3D"Arial, sans-serif" size=3D"2">
<div>Hi,</div>
<div>&nbsp;</div>
<div>We would like to propose a new DISPATCH topic on how SIP and XMPP can =
be used together in a complementary fashion.</div>
<div>&nbsp;</div>
<div>We have been working on a solution which combines SIP based VoIP and X=
MPP based IM and Presence. The requirements and a proposed solution is outl=
ined in <a href=3D"http://tools.ietf.org/html/draft-veikkolainen-sip-voip-x=
mpp-im-01"><font color=3D"#0000FF"><u>http://tools.ietf.org/html/draft-veik=
kolainen-sip-voip-xmpp-im-01</u></font></a>.</div>
<div>&nbsp;</div>
<div>The motivation for this work comes from experience which shows that mo=
st standards based VoIP deployments use SIP. At the same time, the Extensib=
le Messaging and Presence Protocol (XMPP) is widely used in instant messagi=
ng and presence services. An interesting
scenario arises when a SIP based voice (and video) service is combined toge=
ther with an XMPP based instant messaging and presence service. </div>
<div>&nbsp;</div>
<div>Note that the goal in this work is not SIP-XMPP protocol translation, =
but to define protocol extensions and best practises such that SIP based Vo=
IP and XMPP based IM and presence could be seamlessly combined and offered =
to the end user. For rapid deployment,
we assume no changes in the server infrastructure, and instead impose new r=
equirements and protocol extensions only to the endpoints.</div>
<div>&nbsp;</div>
<div>We would like to get some discussion going on the use case itself as w=
ell as on the solution. Also, we would like to hear you thoughts on DISPATC=
H or a new &quot;dispatched&quot; mini-WG as the home for such work.</div>
<div>&nbsp;</div>
<div>Exact charter proposal and problem statemen is to follow. </div>
<div>&nbsp;</div>
<div>Regards,</div>
<div>Simo and Markus</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
</font>
</body>
</html>

--_000_B23B311878A0B4438F5F09F47E01AAE03B10AD6263NOKEUMSG01mgd_--

From oej@edvina.net  Fri Sep  4 05:14:58 2009
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Cc: dispatch@ietf.org, Markus.Isomaki@nokia.com, Camarillo Gonzalo <gonzalo.camarillo@ericsson.com>
Subject: Re: [dispatch] New topic proposal for DISPATCH
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4 sep 2009 kl. 14.09 skrev <Simo.Veikkolainen@nokia.com>:

> Hi,
>
> We would like to propose a new DISPATCH topic on how SIP and XMPP  
> can be used together in a complementary fashion.
>
> We have been working on a solution which combines SIP based VoIP and  
> XMPP based IM and Presence. The requirements and a proposed solution  
> is outlined in http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01 
> .
>
> The motivation for this work comes from experience which shows that  
> most standards based VoIP deployments use SIP. At the same time, the  
> Extensible Messaging and Presence Protocol (XMPP) is widely used in  
> instant messaging and presence services. An interesting scenario  
> arises when a SIP based voice (and video) service is combined  
> together with an XMPP based instant messaging and presence service.
>
> Note that the goal in this work is not SIP-XMPP protocol  
> translation, but to define protocol extensions and best practises  
> such that SIP based VoIP and XMPP based IM and presence could be  
> seamlessly combined and offered to the end user. For rapid  
> deployment, we assume no changes in the server infrastructure, and  
> instead impose new requirements and protocol extensions only to the  
> endpoints.
>
> Exact charter proposal and problem statemen is to follow.

The XMPP forum has a few drafts on this, so we should coordinate with  
them. There are a lot of issues to sort out. We had a workshop on this  
topic with Asterisk, Kamailio and a few other products at Inria in  
France and saw a lot of issues to sort out both in regards to IM and  
presence.

For Asterisk we have both SIP and XMPP telephony, for Kamailio we have  
modules that try to bridge presence updates and IM.

I think it's hard to separate protocol translation. It's all about  
making it work together seamlessly.

/O

From vkg@alcatel-lucent.com  Fri Sep  4 07:53:27 2009
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Subject: [dispatch] SIP recording proposal
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On Aug 21, 2009, Mary Barnes wrote:

 > Thus, the following are the proposed deadlines:
 >
 > Sept 7th  - Cutoff date to notify the chairs and DISPATCH WG
 > mailing list of plans to submit a proposal.
 > -------------------------------------------------------------
 >
 > This will help folks with similar topics to work together before
 > the meeting. If a preliminary charter proposal is available at
 > this point, please include it.

Mary, Gonzalo: This is a quick notification of our intent to
continue the work on SIP recording that was started in the
Stockholm IETF.

We are currently working on a charter proposal and will submit
that by the next deadline -- Sept. 21, 2009.

The media (slides, audio recording, minutes, drafts) relevant
to the recording work discussed in Stockholm can be retrieved
from http://www.softarmor.com/dispatch/IETF75/.

Thanks,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/

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Cc: "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] Call Control UUI for SIP Issue
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Alan,

Based on your answers, I'm ok with your categorization of cases.

	Thanks,
	Paul

Alan Johnston wrote:
> Paul,
> 
> Thanks for your comments - see mine below.
> 
> Thanks,
> Alan
> 
> Paul Kyzivat wrote:
>> Alan,
>>
>> I agree it is a good idea to partition the use cases in this way.
>>
>> I'm a little confused about the criteria you give for UUI below, because
>> - I thought one of the reasons originally given for the proposed
>>   mechanism was that it was needed before call establishment for
>>   purpose of making routing decisions. If so, that means that it
>>   impacts sip behavior.
> 
> The information does need to be rendered and parsed at the time of call 
> establishment - that is why it is carried in an INVITE.  But it is not 
> really used for SIP routing decisions.  I suppose it could be used in 
> deciding whether to answer a call or reject it, but again I'd say that 
> is not SIP behavior.
> 
>>
>> - Also, isn't the applying of policy by sip elements also a change
>>   to sip behavior?
> 
> No, the policy is just whether to pass the information element up the 
> protocol stack or whether to block its  transport within a domain.  The 
> policy isn't, because this type of UUI is present, do some special SIP 
> operation or invoke a special SIP feature.
> 
> So, here's an example of what I mean by "changing SIP behavior".  Lets 
> say we had this UUI mechanism but we didn't have the Refer-Sub: no 
> header field.  One could argue that suppressing the REFER implicit 
> subscription is just a piece of user information, which is generated by 
> the UAC and consumed by the UAS, so why not transport it using the UUI 
> mechanism.  Well, this information changes SIP behavior in that no 
> NOTIFY is sent.  Hence, this would not be a suitable usage for this UUI 
> mechanism. 
>>
>>     Thanks,
>>     Paul
>>
>> Alan Johnston wrote:
>>> Based on feedback during my presentation in Stockholm, I think I  
>>> have a good idea how to modify the proposed charter text.
>>>
>>> One issue perhaps needs a little more discussion so I'd like to kick 
>>> that off here.
>>>
>>> The question is, for some kind of information, when should a new SIP 
>>> header field be used, when should this UUI transport mechanism be 
>>> used, and when should a proprietary header field be used.  It seemed 
>>> like we  had some differing opinions on this with respect to UUI.  I 
>>> think which approach depends on the information being transported.  
>>> Here's my take on this:
>>>
>>> New SIP header field to transport user information.  I think a new 
>>> SIP header field is needed when:
>>> 1. SIP behavior is changed
>>> 2. The information is generated and consumed at the SIP layer by SIP 
>>> elements
>>> 3. SIP elements besides the UAC and UAS might be interested and 
>>> consume the information
>>> 4. There are specific privacy or cross-RAI issues involved with the 
>>> information being transported
>>> 5. Interoperability is important
>>>
>>> UUI Transport mechanism.  I think this mechanism is suitable when:
>>> 1. The information is generated and consumed by an application using 
>>> SIP during session setup but not necessarily even SIP aware.
>>> 2. SIP behavior is unchanged
>>> 3. Generally only the UAC and UAS are interested in the information
>>> 4. The information is expected to survive retargeting, redirection, 
>>> and retargeting
>>> 5. SIP elements may need to apply policy about passing and screening 
>>> the information
>>> 6. Multivendor interop is important
>>>
>>> Proprietary header field (the old P-header)
>>> 1. The information is only of interest to UAC and UAS
>>> 2. No SIP elements need apply policy to the information
>>> 3. Single vendor or closed system application
>>>
>>> If we disagree strongly on these points, lets have that discussion 
>>> now, and this may help us formulate charter text about the mechanism 
>>> itself.
>>>
>>> Thanks,
>>> Alan
>>>
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>
>>
> 
> 

From hsinnrei@adobe.com  Fri Sep  4 08:42:21 2009
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From: Henry Sinnreich <hsinnrei@adobe.com>
To: "Simo.Veikkolainen@nokia.com" <Simo.Veikkolainen@nokia.com>, "dispatch@ietf.org" <dispatch@ietf.org>, "mary.barnes@nortel.com" <mary.barnes@nortel.com>, "gonzalo.camarillo@ericsson.com" <gonzalo.camarillo@ericsson.com>
Date: Fri, 4 Sep 2009 08:41:58 -0700
Thread-Topic: [dispatch] New topic proposal for DISPATCH
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Subject: Re: [dispatch] New topic proposal for DISPATCH
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I would like to second such work, but a far deeper look under the hood is n=
ecessary, so we don't do another Elbonia project.
http://dilbert.com/strips/comic/2009-09-01/

Here are some not very mature thoughts on the scope of the work.
The most interesting, ROI estimate and REST architecture support are at the=
 end.

Servers

Fundamentally, we need a registrar, a proxy and a presence server: Three in=
 total

 *   What will be the duplication of servers and where specifically?
 *   How does the SIP registrar communicate with the XMPP server(s)?
 *   What may be the routing protocol(s) inside a SIP+XMMP network? Think o=
f IMS as a stressing use case.

User Agents


 *   Same as with servers: How many and what protocol RFCs are there in the=
 combination?
 *   Which of the "services" can be placed in the applications in the endpo=
ints or in the network application protocols, such as SIP and XMPP. This is=
 the topic on infrastructure vs. endpoint applications.

Application Protocol Stacks


 *   How many protocol stacks does this involve and more to the point, how =
many RFCs?
 *   For SIP alone, see http://rfc3261.net/. What will be the total combine=
d RFC count?

NAT Traversal


 *   Will STUN/TURN/ICE work the same for SIP and for XMPP?

Return on Investment (ROI): Cost and effort by developers. More network inf=
rastructure, not even counting B2B NEs.


 *   What is the additional time/money required for developers that know SI=
P to also learn XMPP and vice versa? This is all about return on investment=
.

Support for a REST-full architecture

The WWW has a REST architecture was formulated in 2000, after the emergence=
 of SIP and XMMP.  As a result of its architecture, the WWW has had a trans=
formational effect on the following (this is not a complete list). Some SIP=
 work recommends already recommends REST, such as in RFC 4240 and in http:/=
/tools.ietf.org/html/draft-griffin-bliss-rest-00


 *   Web applications from mail to office apps
 *   Internet applications previously served by dedicated network applicati=
ons protocols
 *   Many other industries, from banking to newsprint to travel. And Oh, so=
cial networks, they also have registrars...

Now please keep in mind the WWW architecture uses basically only 3 componen=
ts*: The URI for addressing, HTTP for transport and HTML for data rendering=
, augmented by some other languages (XML) and namespaces. Can we benefit so=
mething from the WWW architecture? If yes what specifically?

Heresy: If the web developers could use HTTP only, what other application l=
evel protocols do we really need except RTP/UDP?
This includes in my mind TLS and DTLS for secure transport or even better I=
MO: HIP for many reasons.

* T.V. Raman: "Toward 2W, Beyond Web 2.0", Communications of the ACM, Febru=
ary 2009

Many or most of these points may be moot or make no sense, but maybe should=
 be cleared up, in case there are more naive people like me that may be int=
erested.

My strict personal 2 cents.

FRANK comments are most welcome.

Henry



On 9/4/09 7:09 AM, "Simo.Veikkolainen@nokia.com" <Simo.Veikkolainen@nokia.c=
om> wrote:

Hi,

We would like to propose a new DISPATCH topic on how SIP and XMPP can be us=
ed together in a complementary fashion.

We have been working on a solution which combines SIP based VoIP and XMPP b=
ased IM and Presence. The requirements and a proposed solution is outlined =
in http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01 <http:=
//tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01> .

The motivation for this work comes from experience which shows that most st=
andards based VoIP deployments use SIP. At the same time, the Extensible Me=
ssaging and Presence Protocol (XMPP) is widely used in instant messaging an=
d presence services. An interesting scenario arises when a SIP based voice =
(and video) service is combined together with an XMPP based instant messagi=
ng and presence service.

Note that the goal in this work is not SIP-XMPP protocol translation, but t=
o define protocol extensions and best practises such that SIP based VoIP an=
d XMPP based IM and presence could be seamlessly combined and offered to th=
e end user. For rapid deployment, we assume no changes in the server infras=
tructure, and instead impose new requirements and protocol extensions only =
to the endpoints.

We would like to get some discussion going on the use case itself as well a=
s on the solution. Also, we would like to hear you thoughts on DISPATCH or =
a new "dispatched" mini-WG as the home for such work.

Exact charter proposal and problem statemen is to follow.

Regards,
Simo and Markus




--_000_C6C69F765AB1hsinnreiadobecom_
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<HTML>
<HEAD>
<TITLE>Re: [dispatch] New topic proposal for DISPATCH</TITLE>
</HEAD>
<BODY>
<FONT SIZE=3D"2"><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN ST=
YLE=3D'font-size:14pt'>I would like to second such work, but a far deeper l=
ook under the hood is necessary, so we don&#8217;t do another Elbonia proje=
ct.<BR>
<a href=3D"http://dilbert.com/strips/comic/2009-09-01/">http://dilbert.com/=
strips/comic/2009-09-01/</a><BR>
<BR>
Here are some not very mature thoughts on the scope of the work. <BR>
The most interesting, ROI estimate and REST architecture support are at the=
 end.<BR>
<BR>
Servers<BR>
<BR>
Fundamentally, we need a registrar, a proxy and a presence server: Three in=
 total<BR>
</SPAN></FONT></FONT><UL><LI><FONT SIZE=3D"2"><FONT FACE=3D"Calibri, Verdan=
a, Helvetica, Arial"><SPAN STYLE=3D'font-size:14pt'>What will be the duplic=
ation of servers and where specifically?
</SPAN></FONT></FONT><LI><FONT SIZE=3D"2"><FONT FACE=3D"Calibri, Verdana, H=
elvetica, Arial"><SPAN STYLE=3D'font-size:14pt'>How does the SIP registrar =
communicate with the XMPP server(s)?
</SPAN></FONT></FONT><LI><FONT SIZE=3D"2"><FONT FACE=3D"Calibri, Verdana, H=
elvetica, Arial"><SPAN STYLE=3D'font-size:14pt'>What may be the routing pro=
tocol(s) inside a SIP+XMMP network? Think of IMS as a stressing use case.<B=
R>
</SPAN></FONT></FONT></UL><FONT SIZE=3D"2"><FONT FACE=3D"Calibri, Verdana, =
Helvetica, Arial"><SPAN STYLE=3D'font-size:14pt'><BR>
User Agents<BR>
<BR>
</SPAN></FONT></FONT><UL><LI><FONT SIZE=3D"2"><FONT FACE=3D"Calibri, Verdan=
a, Helvetica, Arial"><SPAN STYLE=3D'font-size:14pt'>Same as with servers: H=
ow many and what protocol RFCs are there in the combination?
</SPAN></FONT></FONT><LI><FONT SIZE=3D"2"><FONT FACE=3D"Calibri, Verdana, H=
elvetica, Arial"><SPAN STYLE=3D'font-size:14pt'>Which of the &#8220;service=
s&#8221; can be placed in the applications in the endpoints or in the netwo=
rk application protocols, such as SIP and XMPP. This is the topic on infras=
tructure vs. endpoint applications.<BR>
</SPAN></FONT></FONT></UL><FONT SIZE=3D"2"><FONT FACE=3D"Calibri, Verdana, =
Helvetica, Arial"><SPAN STYLE=3D'font-size:14pt'><BR>
Application Protocol Stacks<BR>
<BR>
</SPAN></FONT></FONT><UL><LI><FONT SIZE=3D"2"><FONT FACE=3D"Calibri, Verdan=
a, Helvetica, Arial"><SPAN STYLE=3D'font-size:14pt'>How many protocol stack=
s does this involve and more to the point, how many RFCs?
</SPAN></FONT></FONT><LI><FONT SIZE=3D"2"><FONT FACE=3D"Calibri, Verdana, H=
elvetica, Arial"><SPAN STYLE=3D'font-size:14pt'>For SIP alone, see <a href=
=3D"http://rfc3261.net/">http://rfc3261.net/</a>. What will be the total co=
mbined RFC count?<BR>
</SPAN></FONT></FONT></UL><FONT SIZE=3D"2"><FONT FACE=3D"Calibri, Verdana, =
Helvetica, Arial"><SPAN STYLE=3D'font-size:14pt'><BR>
NAT Traversal<BR>
<BR>
</SPAN></FONT></FONT><UL><LI><FONT SIZE=3D"2"><FONT FACE=3D"Calibri, Verdan=
a, Helvetica, Arial"><SPAN STYLE=3D'font-size:14pt'>Will STUN/TURN/ICE work=
 the same for SIP and for XMPP?<BR>
</SPAN></FONT></FONT></UL><FONT SIZE=3D"2"><FONT FACE=3D"Calibri, Verdana, =
Helvetica, Arial"><SPAN STYLE=3D'font-size:14pt'><BR>
Return on Investment (ROI): Cost and effort by developers. More network inf=
rastructure, not even counting B2B NEs.<BR>
<BR>
</SPAN></FONT></FONT><UL><LI><FONT SIZE=3D"2"><FONT FACE=3D"Calibri, Verdan=
a, Helvetica, Arial"><SPAN STYLE=3D'font-size:14pt'>What is the additional =
time/money required for developers that know SIP to also learn XMPP and vic=
e versa? This is all about return on investment.<BR>
</SPAN></FONT></FONT></UL><FONT SIZE=3D"2"><FONT FACE=3D"Calibri, Verdana, =
Helvetica, Arial"><SPAN STYLE=3D'font-size:14pt'><BR>
Support for a REST-full architecture<BR>
<BR>
The WWW has a REST architecture was formulated in 2000, after the emergence=
 of SIP and XMMP. &nbsp;As a result of its architecture, the WWW has had a =
transformational effect on the following (this is not a complete list). Som=
e SIP work recommends already recommends REST, such as in RFC 4240 and in <=
a href=3D"http://tools.ietf.org/html/draft-griffin-bliss-rest-00">http://to=
ols.ietf.org/html/draft-griffin-bliss-rest-00</a><BR>
&nbsp;<BR>
</SPAN></FONT></FONT><UL><LI><FONT SIZE=3D"2"><FONT FACE=3D"Calibri, Verdan=
a, Helvetica, Arial"><SPAN STYLE=3D'font-size:14pt'>Web applications from m=
ail to office apps=20
</SPAN></FONT></FONT><LI><FONT SIZE=3D"2"><FONT FACE=3D"Calibri, Verdana, H=
elvetica, Arial"><SPAN STYLE=3D'font-size:14pt'>Internet applications previ=
ously served by dedicated network applications protocols
</SPAN></FONT></FONT><LI><FONT SIZE=3D"2"><FONT FACE=3D"Calibri, Verdana, H=
elvetica, Arial"><SPAN STYLE=3D'font-size:14pt'>Many other industries, from=
 banking to newsprint to travel. And Oh, social networks, they also have re=
gistrars...<BR>
</SPAN></FONT></FONT></UL><FONT SIZE=3D"2"><FONT FACE=3D"Calibri, Verdana, =
Helvetica, Arial"><SPAN STYLE=3D'font-size:14pt'><BR>
Now please keep in mind the WWW architecture uses basically only 3 componen=
ts*: The URI for addressing, HTTP for transport and HTML for data rendering=
, augmented by some other languages (XML) and namespaces. Can we benefit so=
mething from the WWW architecture? If yes what specifically?<BR>
<BR>
Heresy: If the web developers could use HTTP only, what other application l=
evel protocols do we really need except RTP/UDP?<BR>
This includes in my mind TLS and DTLS for secure transport or even better I=
MO: HIP for many reasons.<BR>
<BR>
* T.V. Raman: &#8220;Toward 2W, Beyond Web 2.0&#8221;, Communications of th=
e ACM, February 2009<BR>
<BR>
Many or most of these points may be moot or make no sense, but maybe should=
 be cleared up, in case there are more naive people like me that may be int=
erested.<BR>
<BR>
My strict personal 2 cents.<BR>
<BR>
FRANK comments are most welcome.<BR>
<BR>
Henry<BR>
</SPAN></FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPA=
N STYLE=3D'font-size:18pt'><BR>
<BR>
<BR>
On 9/4/09 7:09 AM, &quot;<a href=3D"Simo.Veikkolainen@nokia.com">Simo.Veikk=
olainen@nokia.com</a>&quot; &lt;<a href=3D"Simo.Veikkolainen@nokia.com">Sim=
o.Veikkolainen@nokia.com</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Aria=
l">Hi,<BR>
&nbsp;<BR>
We would like to propose a new DISPATCH topic on how SIP and XMPP can be us=
ed together in a complementary fashion.<BR>
&nbsp;<BR>
We have been working on a solution which combines SIP based VoIP and XMPP b=
ased IM and Presence. The requirements and a proposed solution is outlined =
in <FONT COLOR=3D"#0000FF"><U><a href=3D"http://tools.ietf.org/html/draft-v=
eikkolainen-sip-voip-xmpp-im-01">http://tools.ietf.org/html/draft-veikkolai=
nen-sip-voip-xmpp-im-01</a></U></FONT> &lt;<a href=3D"http://tools.ietf.org=
/html/draft-veikkolainen-sip-voip-xmpp-im-01">http://tools.ietf.org/html/dr=
aft-veikkolainen-sip-voip-xmpp-im-01</a>&gt; .<BR>
&nbsp;<BR>
The motivation for this work comes from experience which shows that most st=
andards based VoIP deployments use SIP. At the same time, the Extensible Me=
ssaging and Presence Protocol (XMPP) is widely used in instant messaging an=
d presence services. An interesting scenario arises when a SIP based voice =
(and video) service is combined together with an XMPP based instant messagi=
ng and presence service. <BR>
&nbsp;<BR>
Note that the goal in this work is not SIP-XMPP protocol translation, but t=
o define protocol extensions and best practises such that SIP based VoIP an=
d XMPP based IM and presence could be seamlessly combined and offered to th=
e end user. For rapid deployment, we assume no changes in the server infras=
tructure, and instead impose new requirements and protocol extensions only =
to the endpoints.<BR>
&nbsp;<BR>
We would like to get some discussion going on the use case itself as well a=
s on the solution. Also, we would like to hear you thoughts on DISPATCH or =
a new &quot;dispatched&quot; mini-WG as the home for such work.<BR>
&nbsp;<BR>
Exact charter proposal and problem statemen is to follow. <BR>
&nbsp;<BR>
Regards,<BR>
Simo and Markus<BR>
&nbsp;<BR>
&nbsp;<BR>
</FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><BR>
</FONT></SPAN></BLOCKQUOTE>
</BODY>
</HTML>


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Subject: [dispatch] IETF-76 DISPATCH WG deadlines : New Topic Domain Registration
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Dispatch Chairs:  We would like to reserve some time during DISPATCH to
discuss the topic of domain registration in SIP.

A draft from Hadriel Kaplan that fully describes the issue will be
forthcoming but the problem statement can be described in the following
excerpt:    


   In some environments, notably the Small-to-Medium Business (SMB) market,
SIP devices in the Enterprise are effectively IP-PBX's,receiving PRI-type
SIP trunk services from a SIP Service Provider (SSP).  The SIP IP-PBX device
is authoritative for its local domain of AoR's.   
    
   Even if the IP-PBX has its own Domain Name, not every small business
owner has a DNS, or has the technical capability to provision DNS entries
(e.g., SRV's) appropriately in their DNS, or has the ability to do so
through any third party which hosts their domain name(s),or may not wish to
make their SIP device address known publicly. Furthermore, the SMB may not
have its own Domain Name at all, and   instead merely has an IP Address. 
    
   Because a single SSP may support multiple thousands of such SMB IP-
PBX's, it is impractical and cost-prohibitive to manually provision  their
IP Addresses in every SIP node along the path which needs to   reach the SMB
IP-PBX customer.  Instead, a dynamic reachability mechanism is needed. 
    
   Such a mechanism already exists for SIP: the REGISTER transaction. A
REGISTER request transaction provides dynamic reachability information of an
AoR for its authoritative domain.   
    
   Since a REGISTER request mechanism is generally supported by most SIP
Service Providers' equipment, it has become common practice for  small
IP-PBX's to use the mechanism as a means of providing their   reachability
information.  In some cases interoperability problems have arisen, due to
differing expectations and implementations of how such a mechanism should
work in practice.  


Richard Shockey
PSTN Mobile: +1 703.593.2683
<mailto:richard(at)shockey.us>
skype/AIM: rshockey101 
LinkedIn : http://www.linkedin.com/in/rshockey101


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From: "Richard Shockey" <richard@shockey.us>
To: <sipcore@ietf.org>, <dispatch@ietf.org>
Date: Fri, 4 Sep 2009 16:05:53 -0400
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Subject: [dispatch] Results of a SIPforum effort to understand problems with Fax and SIP
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The SIPforum has recently issued a report on ongoing issues with SIP and Fax
that may be of interest to the larger SIP community. Work on understanding
the problems with FAX and SIP are continuing and members of the IETF SIP
community are invited to participate in future discussions.


SIPforum FoIP Task Group Charter

The proposed charter of the SIP Forum FoIP task group is to investigate
ongoing issues with the deployment of fax services, specifically ITU T.38,
in SIP networks. SIP networks cannot adequately replace analog PSTN in
enterprises unless essential services such as FAX are accommodated.

The use of ITU T.37 email based store and forward has not been successful
due to the specific legal status fax enjoys in law and that fax
communications has 3rd party confirmation of transmission and/or delivery in
carrier billing records (non repudiation). Classic FAX (T.30) over G.711 has
not proved to be reliable and SIP communications , in the future, may use
other codecs that have been proven to break T.30 such as G.729 and other
high compression codecs like SPEEDEX etc.

The SIP Forum FoIP task group is chartered to accomplish the following
tasks:

1. Fully document what the current issues are surrounding ITU T.38 in SIP
networks.
a. What interoperability testing procedures currently exist.
b. What are the common factors in T.38 failure such as page length or lack
of ECM support in carrier gateways and ATA's.
c. Network packet loss considerations.

2. Determine what solutions are currently available to address the problem.

3. Determine if the problem can be solved within the scope of existing IETF
SIP and ITU T.38 Recommendations.

4. If the problems can be solved using existing standards by tightening
requirements, document the procedures vendors and carriers need to implement
in an appropriate SIP Forum Technical Recommendation.

5. If, in the judgment of the SIP Forum FoIP WG, existing IETF and or ITU
standards need to be modified, develop a strongly worded recommendation to
the appropriate Standards Development Organization (SDO) on what the SIP
Forum FoIP WG has discovered and recommend appropriate action by the SDO to
remedy the issue.

Information about the FoIP Task Group

http://www.sipforum.org/content/view/310/252/

The Fax Problem Statement Report.

http://www.sipforum.org/component/option,com_docman/task,doc_download/gid,30
3/Itemid,75/




Richard Shockey
Member Board of Directors SIPforum
Co-Chair SIPforum Technical Task Groups

PSTN Mobile: +1 703.593.2683
<mailto:richard(at)shockey.us>
skype/AIM: rshockey101 
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From aallen@rim.com  Mon Sep  7 03:37:06 2009
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From: "Andrew Allen" <aallen@rim.com>
To: <mary.barnes@nortel.com>, <Gonzalo.Camarillo@ericsson.com>, <dispatch@ietf.org>
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Can we request agenda time in Dispatch to discuss how to proceed with draft-=
boucadair-dispatch-ipv6-atypes?

Thanks
Andrew

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From salvatore.loreto@ericsson.com  Mon Sep  7 04:26:45 2009
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From: Salvatore Loreto <salvatore.loreto@ericsson.com>
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Subject: Re: [dispatch] IETF-76 DISPATCH WG deadlines
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Hi,
 
We would like to propose a new DISPATCH topic on "Disaggregated media":

    the ability for a user to create a  multimedia session combining
    different media streams,
    coming from different devices under his or her control, so that they
    are treated
    by the far end of the session as a single media session.

The analysis of what types of disaggregated media can be implemented 
using existing protocol mechanisms,
the pros and cons of using each of those mechanisms and also the 
scenarios that are not covered by current mechanisms
are all described in :
http://tools.ietf.org/id/draft-loreto-dispatch-disaggregated-media-00.txt

This topic has been discussed on the mailing receiving quite high interest
So we would like to hear you thoughts on DISPATCH or a new "dispatched" 
mini-WG as the home for such work.

Charter proposal and problem statement is to follow.

best regards
Salvatore Loreto



Mary Barnes wrote:
>
> Hi all,
>
> As was done for IETF-75, we are setting earlier deadlines for 
> discussing topics in DISPATCH prior to the WG session at IETF-76. 
>
> We will again follow the deadlines for BoFs:
> _http://www.ietf.org/meeting/cutoff-dates.html#76_
>
> This provides enough time to dispatch topics prior to the meeting as 
> appropriate - e.g., mini-bofs, official bofs, other WGs, new WGs,  
> mailing lists, etc. Thus, allowing us to have more focused and 
> constructive discussions on a smaller set of topics prior to the WG 
> session.
>
> Please note that the dates are much sooner than you might expect as 
> the time between IETF-75 and IETF-76 is 3 weeks less than the time 
> between IETF-74 and IETF-75. The document deadlines are 9 and 10 weeks 
> away from next Monday (August 24th). 
>
>
> Thus, the following are the proposed deadlines:
>
> Sept 7th  - Cutoff date to notify the chairs and DISPATCH WG mailing 
> list of plans to submit a proposal.
> -------------------------------------------------------------------------------------------------------- 
>
>
> This will help folks with similar topics to work together before the 
> meeting. If a preliminary charter proposal is available at this point, 
> please include it.
>
>
> September 21st - Cutoff for charter proposals for topics.
> --------------------------------------------------------
>
> A charter proposal must consist of a minimum of a problem statement 
> and at least one milestone/deliverable. This deadline will allow time 
> for consideration of proposals that could potentially be "dispatched" 
> prior to the WG session.
>
>
> September 28th - Topics that are to be the focus of IETF-76 are 
> announced.
> --------------------------------------------------------------------------- 
>
>
> The idea here is to focus the discussion over the weeks preceding 
> IETF-76 on these main topics, noting that any new or updates to any 
> documents are due 3-4 weeks later.  This will ensure we have 
> constructive discussions before the meeting and are actually able to 
> determine consensus as to where work should be progressed (e.g., 
> separate BoF, a new WG, an existing WG, individual document, etc.) at 
> IETF-76. Note, that the exact disposition for a topic may (per the 
> usual process) require follow-up and confirmation by the ADS and/or 
> IESG (e.g., for a new WG, Bof, individually sponsored document, etc.) 
> or with another WG to ensure agreement with the DISPATCH WG consensus 
> for the topic.
>
>
> As discussed previously, the intention is not necessarily to preclude 
> folks submitting documents on other topics, however, it is very 
> unlikely there would be agenda time allocated to documents/topics 
> submitted after the deadlines. We can include one or two slides on 
> these topics in the DISPATCH WG chair charts so that the authors can 
> gather other interested parties to contribute to the work.
>
> Regards,
> Mary and Gonzalo
> DISPATCH WG co-chairs
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>   


From kumiko@cs.columbia.edu  Mon Sep  7 07:17:43 2009
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From: Kumiko Ono <kumiko@cs.columbia.edu>
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Subject: [dispatch] a new proposal for reference to an earlier communication
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Hello,

We are going to submit a proposal for reference to an earlier  
communication.

This is to reduce false positives during SPIT filtering. Even if  
caller IDs are not part of the recipient's white lists,
the recipients could label incoming calls using the reference, e.g.,  
the Message-ID of an email sent from the caller to the recipient.

Although the objectives are different, similar proposals in terms of  
references to another communication (mostly limited to a dialog) have  
been discussed
like in "draft-worley-references-04."

We would like to hear your thoughts of our proposal.

Thanks,
Kumiko







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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; ">Hello,<div><br></div><div>We =
are going to submit a proposal for reference to an earlier =
communication.</div><div><br></div><div>This is to reduce false =
positives during SPIT filtering. Even if caller IDs are not part of the =
recipient's white lists,</div><div>the recipients could label incoming =
calls using the reference, e.g., the Message-ID of an email sent from =
the caller to the recipient.</div><div><br></div><div>Although the =
objectives are different,&nbsp;similar proposals in terms of references =
to another communication (mostly limited to a dialog) have been =
discussed&nbsp;</div><div>like&nbsp;in&nbsp;"draft-worley-references-04."<=
/div><div><br></div><div>We would like to hear your thoughts of our =
proposal.</div><div><br></div><div>Thanks,</div><div>Kumiko</div><div><br>=
</div><div><div style=3D"text-align: auto;margin-top: 0px; margin-right: =
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From mary.barnes@nortel.com  Mon Sep  7 08:45:53 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Andrew Allen" <aallen@rim.com>, <Gonzalo.Camarillo@ericsson.com>, <dispatch@ietf.org>
Subject: Re: [dispatch] draft-boucadair-dispatch-ipv6-atypes
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Hi Andrew,

At this point in time, we're not handling agenda requests per se. Right
now, we want to know the topics for which folks want feedback PRIOR to
IETF-76 so that we can use the agenda time to discuss issues that can't
be resolved on the mailing list before then. We can take this as a
placeholder for this topic and thus, we would expect to see a detailed
problem statement, etc. on or before Sept. 21st., per the IETF-76
DISPATCH deadlines:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00500.html

Just a note that when I summarized the status of the post-IETF-75
topics, this one had received no ML discussion nor indication of
interest in working on this problem:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00501.html
So, you really need to get some WG feedback on this proposal if you want
it to move forward. =20

Thanks,
Mary.=20

-----Original Message-----
From: Andrew Allen [mailto:aallen@rim.com]=20
Sent: Monday, September 07, 2009 5:37 AM
To: Barnes, Mary (RICH2:AR00); Gonzalo.Camarillo@ericsson.com;
dispatch@ietf.org
Cc: mohamed.boucadair@orange-ftgroup.com
Subject: draft-boucadair-dispatch-ipv6-atypes


Can we request agenda time in Dispatch to discuss how to proceed with
draft-boucadair-dispatch-ipv6-atypes?

Thanks
Andrew

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From volkerh@bell-labs.com  Tue Sep  8 19:00:04 2009
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Cc: "NOEL, ERIC C, ATTLABS" <ecnoel@att.com>, DISPATCH <dispatch@ietf.org>, SIP-OVERLOAD <sip-overload@ietf.org>
Subject: Re: [dispatch] [sip-overload] Revised Charter Proposal for SIP Overload
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Carolyn,

thanks for your feedback! I'll update the charter proposal to address 
the comment.

Thanks,

Volker

JOHNSON, CAROLYN R, ATTLABS wrote:
> Volker,
> Sorry for the delay, this got stuck in my drafts folder while I was out
> on vacation.
> 
> One comment on the last dashed item:
> - The second mechanism aims at preventing overload in SIP servers by
> distributing load control filters to SIP servers that throttle calls
> based on their destination domain, telephone number prefix or for a
> specific user. Such filters can be used, for example, to throttle calls
> to a hotline during a ticket-giveaway event.
> 
> I suggest replacing "telephone number" with "destination routing" so as
> not to limit to applications that use telephone prefixes. I believe it
> will be more general without losing the applicability to telephone IDs.
> 
> The charter is clearly laid out. 
> Regards,
> 
> Carolyn
>  
> 
> -----Original Message-----
> From: sip-overload-bounces@ietf.org
> [mailto:sip-overload-bounces@ietf.org] On Behalf Of Volker Hilt
> Sent: Thursday, August 20, 2009 9:52 AM
> To: DISPATCH; SIP-OVERLOAD
> Subject: [sip-overload] Revised Charter Proposal for SIP Overload
> 
> All,
> 
> below is an revised SIP Overload Control charter proposal. I have
> received a few comments that are reflected in the new proposal.
> 
> All comments, thoughts, feedback is very welcome!
> 
> Thanks,
> 
> Volker
> 
> 
> 
> SIP Overload Control WG Charter
> -------------------------------
> 
> As with any network element, a Session Initiation Protocol (SIP) server
> can suffer from overload when the number of SIP messages it receives
> exceeds the number of messages it can process. Overload is said to occur
> when a SIP server does not have sufficient resources to process all
> incoming SIP messages.  These resources can include CPU, memory, network
> bandwidth, input/output, or disk resources.
> 
> Overload can pose a serious problem for a network of SIP servers. During
> periods of overload, the throughput of a network of SIP servers can be
> significantly degraded.  In fact, overload may lead to a situation in
> which the throughput drops down to zero or a small fraction of the
> original processing capacity.  This is often called congestion collapse.
> 
> The objective of a SIP overload control mechanism is to enable SIP
> servers to operate close to their capacity limit during times of
> overload.
> 
> The SIP protocol provides a limited mechanism for overload control
> through its 503 (Service Unavailable) response code and the Retry-After
> header.  However, this mechanism cannot fully prevent overload of a SIP
> server and it cannot prevent congestion collapse.  In fact, its on/off
> behavior may cause traffic to oscillate and shift between SIP servers
> and thereby worsen an overload condition.  A detailed discussion of the
> SIP overload problem, the problems with the 503 (Service Unavailable)
> response code and the Retry-After header and the requirements for a SIP
> overload control mechanism can be found in RFC5390.
> 
> The objective of the proposed working group is to develop mechanisms for
> SIP overload control. Two complementary approaches exist for handling
> overload situations:
> - The first mechanism aims at preventing overload in SIP servers by
> adjusting the incoming load to a level that is acceptable for this
> server. This mechanism uses implicit and/or explicit feedback to
> determine when an overload condition occurs in a SIP server and a
> reduction in load is required.
> - The second mechanism aims at preventing overload in SIP servers by
> distributing load control filters to SIP servers that throttle calls
> based on their destination domain, telephone number prefix or for a
> specific user. Such filters can be used, for example, to throttle calls
> to a hotline during a ticket-giveaway event.
> 
> Specifically, the proposed working group will develop the following
> deliverables:
> 1. A document describing key design considerations for a SIP overload
> control mechanism.
> 2. A specification for an SIP overload control mechanism based on
> implicit/explicit feedback.
> 3. A specification for a SIP load filtering mechanism.
> 
> 
> 
> _______________________________________________
> sip-overload mailing list
> sip-overload@ietf.org
> https://www.ietf.org/mailman/listinfo/sip-overload


From dean.willis@softarmor.com  Tue Sep  8 19:48:44 2009
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From: Dean Willis <dean.willis@softarmor.com>
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Subject: Re: [dispatch] IETF-76 DISPATCH WG deadlines : New Topic Domain Registration
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On Sep 4, 2009, at 2:53 PM, Richard Shockey wrote:

> Dispatch Chairs:  We would like to reserve some time during DISPATCH  
> to
> discuss the topic of domain registration in SIP.

It seems like you're proposing a deep and fundamental rewrite of the  
Internet's name architecture, replacing RFC 3263 and its use of DNS  
with some sort of SIP REGISTER technique.

Or maybe I'm missing it entirely, and what you're proposing is a way  
to use SIP REGISTER messages to populate a DNS SRV record within a  
registry?

This raises the question of delegation to that registry -- the  
registry has to be authoritative for all of  example.com in order to  
be able to host an srv record for example.com. So your mythical SMB is  
still going to require a DNS service provider that has to be  
meaningfully provisioned. It's just a question of the outsourcing  
contract, and I fail to see a need for protocol work here. After all,  
we have dynamic DNS, and it works pretty darned well for me today.

So what you talking 'bout, Shockey?

--
Dean

From Simo.Veikkolainen@nokia.com  Wed Sep  9 01:09:58 2009
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From: <Simo.Veikkolainen@nokia.com>
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Date: Wed, 9 Sep 2009 10:09:51 +0200
Thread-Topic: [dispatch] New topic proposal for DISPATCH
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=20

>-----Original Message-----
>From: ext Olle E. Johansson [mailto:oej@edvina.net]=20
>Sent: 04 September, 2009 15:14
>To: Veikkolainen Simo (Nokia-D/Helsinki)
>Cc: dispatch@ietf.org; Mary Barnes; Camarillo Gonzalo; Isomaki=20
>Markus (Nokia-CIC/Espoo); Saint-Andre Peter
>Subject: Re: [dispatch] New topic proposal for DISPATCH
>
>
>4 sep 2009 kl. 14.09 skrev <Simo.Veikkolainen@nokia.com>:
>
>> Hi,
>>
>> We would like to propose a new DISPATCH topic on how SIP and=20
>XMPP can=20
>> be used together in a complementary fashion.
>>
>> We have been working on a solution which combines SIP based VoIP and=20
>> XMPP based IM and Presence. The requirements and a proposed solution=20
>> is outlined in=20
>> http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01
>> .
>>
>> The motivation for this work comes from experience which shows that=20
>> most standards based VoIP deployments use SIP. At the same time, the=20
>> Extensible Messaging and Presence Protocol (XMPP) is widely used in=20
>> instant messaging and presence services. An interesting scenario=20
>> arises when a SIP based voice (and video) service is=20
>combined together=20
>> with an XMPP based instant messaging and presence service.
>>
>> Note that the goal in this work is not SIP-XMPP protocol=20
>translation,=20
>> but to define protocol extensions and best practises such that SIP=20
>> based VoIP and XMPP based IM and presence could be=20
>seamlessly combined=20
>> and offered to the end user. For rapid deployment, we assume no=20
>> changes in the server infrastructure, and instead impose new=20
>> requirements and protocol extensions only to the endpoints.
>>
>> Exact charter proposal and problem statemen is to follow.
>
>The XMPP forum has a few drafts on this, so we should=20
>coordinate with them. There are a lot of issues to sort out.=20
>We had a workshop on this topic with Asterisk, Kamailio and a=20
>few other products at Inria in France and saw a lot of issues=20
>to sort out both in regards to IM and presence.

We have presented the above draft in the XMPP WG BoF in San Francisco earli=
er this year. I agree both the IETF and the XSF would need to be aware, but=
 preferably there should be one place where the concepting happens. We thou=
ght DISPATCH might be the right place for this.=20

>For Asterisk we have both SIP and XMPP telephony, for Kamailio=20
>we have modules that try to bridge presence updates and IM.
>
>I think it's hard to separate protocol translation. It's all=20
>about making it work together seamlessly.

Our idea was to specify a solution which admittedly leaves some aspects unt=
ouched, but - on the other hand - would offer a solution to an issue we see=
 is relevant and which would be rapidly deployable. There are already quite=
 a few drafts available on protocol interworking (draft-saintandre-sip-xmpp=
-*).

Simo

>/O
>=

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From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
To: "Gilad Shaham" <gshaham@juniper.net>
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Hi Gilad
=20
Thanks for reviewing the document. Here my comments with <Ranjit>
Regards=20
Ranjit=20
=20

________________________________

From: Gilad Shaham [mailto:gshaham@juniper.net]=20
Sent: Tuesday, August 18, 2009 5:50 AM
To: Avasarala Ranjit-A20990; dispatch@ietf.org; sip@ietf.org
Subject: RE: [Sip] New version
of"draft-avasasarala-dispatch-comm-diversion-info" draft submitted



Hi,

=20

See some comments

=20

Page 5:

   "... For e.g. the subscriber wanted to diverted all incoming calls to
voice-mail,
   between 3.00 p.m. to 4.00 p.m.  Yet, by mistake she configures the
   time-duration as 3.00 to 4.00 p.m"

Some of the sentence needs restructuring and I also don't fully
understand the example. Is it AM-PM or wrong field was configured?=20

<Ranjit> Agreed.  Will correct the sentence.=20

=20

Page 12:

What if time-range is missing? What should be the default? Sounds to me
the default should be the current time with no end date.=20

<Ranjit> if time-range is missing, then notifications for all
communication diversions are sent.=20

=20

Page 14:

In Comm-div-info-selection-criteria there are several disable-*
subsections, yet their text describes these element gives the subscriber
option of adding information. Shouldn't this be for omitting information
or alternatively, call these elements "enable-*" or did I misunderstand
the purpose.=20

<Ranjit>  E.g disable-originating-user-info -> this element gives the
subscriber the option of adding originating-user-info element to the
notification information. The default value is false which means that
the subscriber wants the originating-user-info element to be present as
part of the notification information. If the value is set to TRUE, then
originating-user-info element is removed from the notification
information document.=20

=20

Page 16:

             <user-name>Boss</originating-user-name>

Should be

             <user-name>Boss</user-name>=20
<Ranjit> Corrected.
=20

It might be also useful to see an example of periodic request.=20

<Ranjit> Will see if I can add one.=20

=20

Page 21:

503 is there, but I don't see 500. Some implementations will avoid 503
and use 500 due to discussion related to this
http://tools.ietf.org/html/draft-hilt-sip-correction-503-01 (now
expired, but still affected some vendor decisions). I might be able to
think of some scenario that involves 502, but I assume this is a result
of the diversion implementation itself so maybe that's the context of
this discussion.=20

<Ranjit> Will add 500 also.=20

=20

Thanks

=20

Thanks

Gilad

=20

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Avasarala Ranjit-A20990
Sent: Monday, August 17, 2009 10:07 AM
To: dispatch@ietf.org; sip@ietf.org
Subject: [Sip] New version
of"draft-avasasarala-dispatch-comm-diversion-info" draft submitted

=20

Hi All=20

We have submitted an updated version of
draft-avasasarala-dispatch-comm-diversion-info=20

It can be accessed at:
http://www.ietf.org/internet-drafts/draft-avasarala-dispatch-comm-div-no
tification-01.txt
<http://www.ietf.org/internet-drafts/draft-avasarala-dispatch-comm-div-n
otification-01.txt>=20

This draft proposes a SIP Event package for Communication Diversions
Notification Information and conforms to procedures and schema described
in 3GPP TS 24.604.=20

Please review and comment

Regards=20
Ranjit=20


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<DIV dir=3Dltr align=3Dleft><SPAN class=3D054262809-09092009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Hi Gilad</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D054262809-09092009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D054262809-09092009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Thanks for reviewing the document. Here my =
comments with=20
<STRONG>&lt;Ranjit&gt;</STRONG></FONT></SPAN></DIV>
<DIV><SPAN lang=3Den-us><FONT face=3DArial color=3D#0000ff=20
size=3D2>Regards</FONT></SPAN> <BR><SPAN lang=3Den-us><FONT face=3DArial =
color=3D#0000ff=20
size=3D2>Ranjit</FONT></SPAN> </DIV>
<DIV>&nbsp;</DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Gilad Shaham =
[mailto:gshaham@juniper.net]=20
<BR><B>Sent:</B> Tuesday, August 18, 2009 5:50 AM<BR><B>To:</B> =
Avasarala=20
Ranjit-A20990; dispatch@ietf.org; sip@ietf.org<BR><B>Subject:</B> RE: =
[Sip] New=20
version of"draft-avasasarala-dispatch-comm-diversion-info" draft=20
submitted<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV class=3DSection1>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial">Hi,<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">See some=20
comments<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Page=20
5:<o:p></o:p></SPAN></FONT></P><PRE><FONT face=3D"Courier New" =
size=3D2><SPAN style=3D"FONT-SIZE: 10pt">&nbsp;&nbsp; &#8220;... For =
e.g. the subscriber wanted to diverted all incoming calls to =
voice-mail,<o:p></o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier =
New" size=3D2><SPAN style=3D"FONT-SIZE: 10pt">&nbsp;&nbsp; between 3.00 =
p.m. to 4.00 p.m.&nbsp; Yet, by mistake she configures =
the<o:p></o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier New" =
size=3D2><SPAN style=3D"FONT-SIZE: 10pt">&nbsp;&nbsp; time-duration as =
3.00 to 4.00 p.m&#8221;<o:p></o:p></SPAN></FONT></PRE>
<P class=3DMsoNormal><FONT color=3Dnavy><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Some of the =
sentence=20
needs restructuring and I also don&#8217;t fully understand the example. =
Is it AM-PM=20
or wrong field was configured?<FONT color=3D#0000ff><SPAN=20
class=3D054262809-09092009>&nbsp;</SPAN></FONT></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dnavy><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><FONT=20
color=3D#0000ff><SPAN class=3D054262809-09092009><STRONG>&lt;Ranjit&gt;=20
Agreed.&nbsp; Will correct the=20
sentence.</STRONG>&nbsp;</SPAN><o:p></o:p></FONT></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Page=20
12:<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dnavy><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">What if =
time-range is=20
missing? What should be the default? Sounds to me the default should be =
the=20
current time with no end date.<FONT color=3D#0000ff><SPAN=20
class=3D054262809-09092009>&nbsp;</SPAN></FONT></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dnavy><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><FONT=20
color=3D#0000ff><SPAN class=3D054262809-09092009><STRONG>&lt;Ranjit&gt; =
if=20
time-range is missing, then&nbsp;notifications for all communication =
diversions=20
are sent.</STRONG>&nbsp;</SPAN><o:p></o:p></FONT></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Page=20
14:<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dnavy><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">In=20
Comm-div-info-selection-criteria there are several disable-* =
subsections, yet=20
their text describes these element gives the subscriber option of adding =

information. Shouldn&#8217;t this be for omitting information or =
alternatively, call=20
these elements &#8220;enable-*&#8221; or did I misunderstand the =
purpose.<FONT=20
color=3D#0000ff><SPAN=20
class=3D054262809-09092009>&nbsp;</SPAN></FONT></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dnavy><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><FONT=20
color=3D#0000ff><SPAN class=3D054262809-09092009><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><FONT=20
color=3D#0000ff><SPAN =
class=3D054262809-09092009><STRONG>&lt;Ranjit&gt;&nbsp; E.g=20
disable-originating-user-info -&gt; this&nbsp;element gives the =
subscriber the=20
option of adding&nbsp;originating-user-info element to the notification=20
information. The default value is&nbsp;false which means that the =
subscriber=20
wants the originating-user-info element to be present as part of the=20
notification information. If the value is set to TRUE, then=20
originating-user-info element is removed from the notification =
information=20
document.</STRONG></SPAN></FONT></SPAN>&nbsp;</SPAN><o:p></o:p></FONT></S=
PAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Page=20
16:<o:p></o:p></SPAN></FONT></P><PRE><FONT face=3D"Courier New" =
size=3D2><SPAN style=3D"FONT-SIZE: =
10pt">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp; =
&lt;user-name&gt;Boss&lt;/originating-user-name&gt;<o:p></o:p></SPAN></FO=
NT></PRE>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Should=20
be<o:p></o:p></SPAN></FONT></P><PRE><FONT face=3D"Courier New"><SPAN =
style=3D"FONT-SIZE: =
10pt">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp; &lt;user-name&gt;Boss&lt;/user-name&gt;<FONT face=3DArial><FONT =
color=3D#0000ff><SPAN =
class=3D054262809-09092009>&nbsp;</SPAN></FONT></FONT></SPAN></FONT></PRE=
><PRE><FONT face=3D"Courier New"><SPAN style=3D"FONT-SIZE: 10pt"><FONT =
face=3DArial><FONT color=3D#0000ff><SPAN =
class=3D054262809-09092009><SPAN style=3D"FONT-SIZE: 10pt; COLOR: navy; =
FONT-FAMILY: Arial"><FONT color=3D#0000ff><SPAN =
class=3D054262809-09092009><STRONG>&lt;Ranjit&gt; =
Corrected.</STRONG></SPAN></FONT></SPAN></SPAN></FONT></FONT></SPAN></FON=
T></PRE><PRE><FONT face=3D"Courier New"><SPAN style=3D"FONT-SIZE: =
10pt"><FONT face=3DArial><FONT color=3D#0000ff><SPAN =
class=3D054262809-09092009><SPAN style=3D"FONT-SIZE: 10pt; COLOR: navy; =
FONT-FAMILY: Arial"><FONT color=3D#0000ff><SPAN =
class=3D054262809-09092009></SPAN></FONT></SPAN>&nbsp;</SPAN><o:p></o:p><=
/FONT></FONT></SPAN></FONT></PRE>
<P class=3DMsoNormal><FONT color=3Dnavy><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">It might be =
also useful=20
to see an example of periodic request.<FONT color=3D#0000ff><SPAN=20
class=3D054262809-09092009>&nbsp;</SPAN></FONT></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dnavy><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><FONT=20
color=3D#0000ff><SPAN class=3D054262809-09092009><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><FONT=20
color=3D#0000ff><SPAN=20
class=3D054262809-09092009><STRONG>&lt;Ranjit&gt;&nbsp;Will&nbsp;see if =
I can add=20
one.</STRONG></SPAN></FONT></SPAN>&nbsp;</SPAN><o:p></o:p></FONT></SPAN><=
/FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Page=20
21:<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dnavy><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">503 is there, =
but I=20
don&#8217;t see 500. Some implementations will avoid 503 and use 500 due =
to discussion=20
related to this <A=20
href=3D"http://tools.ietf.org/html/draft-hilt-sip-correction-503-01">http=
://tools.ietf.org/html/draft-hilt-sip-correction-503-01</A>=20
(now expired, but still affected some vendor decisions). I might be able =
to=20
think of some scenario that involves 502, but I assume this is a result =
of the=20
diversion implementation itself so maybe that&#8217;s the context of =
this=20
discussion.<FONT color=3D#0000ff><SPAN=20
class=3D054262809-09092009>&nbsp;</SPAN></FONT></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dnavy><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><FONT=20
color=3D#0000ff><SPAN class=3D054262809-09092009><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><FONT=20
color=3D#0000ff><SPAN =
class=3D054262809-09092009><STRONG>&lt;Ranjit&gt;&nbsp;Will=20
add 500=20
also.</STRONG></SPAN></FONT></SPAN>&nbsp;</SPAN></FONT></SPAN></FONT></P>=

<P class=3DMsoNormal><FONT color=3Dnavy><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><FONT=20
color=3D#0000ff><SPAN=20
class=3D054262809-09092009><STRONG></STRONG></SPAN></FONT></SPAN></FONT>&=
nbsp;</P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><SPAN=20
class=3D054262809-09092009></SPAN><o:p><SPAN =
class=3D054262809-09092009><FONT=20
color=3D#0000ff>Thanks</FONT></SPAN></o:p></SPAN></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial">Thanks<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial">Gilad<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<DIV>
<DIV class=3DMsoNormal style=3D"TEXT-ALIGN: center" align=3Dcenter><FONT =

face=3D"Times New Roman" size=3D3><SPAN style=3D"FONT-SIZE: 12pt">
<HR tabIndex=3D-1 align=3Dcenter width=3D"100%" SIZE=3D2>
</SPAN></FONT></DIV>
<P class=3DMsoNormal><B><FONT face=3DTahoma size=3D2><SPAN=20
style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; FONT-FAMILY: =
Tahoma">From:</SPAN></FONT></B><FONT=20
face=3DTahoma size=3D2><SPAN style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Tahoma">=20
sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <B><SPAN=20
style=3D"FONT-WEIGHT: bold">On Behalf Of </SPAN></B>Avasarala=20
Ranjit-A20990<BR><B><SPAN style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> =
Monday,=20
August 17, 2009 10:07 AM<BR><B><SPAN style=3D"FONT-WEIGHT: =
bold">To:</SPAN></B>=20
dispatch@ietf.org; sip@ietf.org<BR><B><SPAN=20
style=3D"FONT-WEIGHT: bold">Subject:</SPAN></B> [Sip] New version=20
of"draft-avasasarala-dispatch-comm-diversion-info" draft=20
submitted</SPAN></FONT><o:p></o:p></P></DIV>
<P class=3DMsoNormal><FONT face=3D"Times New Roman" size=3D3><SPAN=20
style=3D"FONT-SIZE: 12pt"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dblue size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">Hi =
All</SPAN></FONT>=20
<o:p></o:p></P>
<P><FONT face=3DArial color=3Dblue size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">We have =
submitted an=20
updated version of draft-avasasarala-dispatch-comm-diversion-info=20
</SPAN></FONT><o:p></o:p></P>
<P><FONT face=3DArial color=3Dblue size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">It can be =
accessed at:=20
</SPAN></FONT><FONT face=3DArial size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"><A=20
href=3D"http://www.ietf.org/internet-drafts/draft-avasarala-dispatch-comm=
-div-notification-01.txt"=20
target=3D_top jQuery1250492662043=3D"3"><FONT face=3D"Times New Roman" =
size=3D3><SPAN=20
style=3D"FONT-SIZE: 12pt; FONT-FAMILY: 'Times New =
Roman'">http://www.ietf.org/internet-drafts/draft-avasarala-dispatch-comm=
-div-notification-01.txt</SPAN></FONT></A></SPAN></FONT><o:p></o:p></P>
<P><FONT face=3DArial color=3Dblue size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">This draft =
proposes a=20
SIP Event package for Communication Diversions Notification Information =
and=20
conforms to procedures and schema described in 3GPP TS=20
24.604.&nbsp;</SPAN></FONT><o:p></o:p></P>
<P><FONT face=3DArial color=3Dblue size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">Please review =
and=20
comment</SPAN></FONT><o:p></o:p></P>
<P><FONT face=3DArial color=3Dblue size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
Arial">Regards</SPAN></FONT>=20
<BR><FONT face=3DArial color=3Dblue size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
Arial">Ranjit</SPAN></FONT>=20
<o:p></o:p></P></DIV></BODY></HTML>

------_=_NextPart_001_01CA3130.C421AE9C--

From vkg@alcatel-lucent.com  Wed Sep  9 09:09:20 2009
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Date: Wed, 09 Sep 2009 11:09:48 -0500
From: "Vijay K. Gurbani" <vkg@alcatel-lucent.com>
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Cc: "Jain, Rajnish" <Rajnish.Jain@ipc.com>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: [dispatch] TCP and SCTP connection reuse draft
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Folks: Before the summer, we had released a new
TCP/SCTP connection reuse draft in the (now defunct) SIP
WG.  With the formation of dispatch, we were advised to
submit it to dispatch for a decision on where to handle
this work.

Essentially, the draft documents the practice of reusing TCP
(and SCTP) connections that is employed by some SIP
implementations today.  TCP and SCTP transport reuse were
originally a part of the draft-ietf-sip-connect-reuse draft,
but were taken out before we progressed that draft to the IESG.

Can we kindly ask the dispatch WG to review and comment on
the the TCP and SCTP connection reuse draft?  All comments
received from the first iteration of the draft in the SIP WG
have been duly addressed in this version.

It will also be helpful if the WG and chairs could provide
some guidance on how to move this work forward (i.e.,
AD-sponsored individual draft, take it to sipcore, etc.)

The draft can be retrieved from:

http://tools.ietf.org/html/draft-jain-dispatch-sip-transport-connection-reuse-00

Thank you,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/

From L.Liess@telekom.de  Thu Sep 10 07:52:17 2009
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thread-topic: Topic for DISPATCH: Alert-Info URNs 
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From: <L.Liess@telekom.de>
To: <mary.barnes@nortel.com>, <Gonzalo.Camarillo@ericsson.com>, <dispatch@ietf.org>
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Subject: [dispatch] Topic for DISPATCH: Alert-Info URNs
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Mary, Gonzalo,

At the IETF-75 we had a presentation and a discussion in BLISS on the =
"Alert-Info URNs" proposal, described in =
http://tools.ietf.org/id/draft-alexeitsev-bliss-alert-info-urns-02.txt.  =
During the discussion, it was recommended to submit the work to DISPATCH =
and to document the interop problems.=20

So, we would like to propose this as a new topic for DISPATCH.=20

The motivation for this work comes from interoperability problems when =
SIP services require the semantic for a ring- / ringback-tone to be =
transmitted in in a SIP-message. =20

The RFC 3261 allows an UAS or proxy to provide a specific ring- =
/ringback-tone as a reference (e.g. HTTP URI) which can be played to the =
user in the Alert-Info header. This mechanism does not ensure =
interoperability when there is no common understanding of the referenced =
content (different countries or vendors, hearing impaired) or when the =
user has his own tones configured in the end device.=20

For interoperability of services as call waiting, call forwarding, blind =
transfer, automatic callback, call hold or emergency annoncements sent =
over PBX systems, a mechanism is needed which allows the UAs and proxies =
to transmit an indication which contains the semantic of the rendering =
instead of the rendering itself and allows the receiving UA to decide =
about the concrete rendering.=20

The charter proposal and the updated draft are to follow.=20

Kind regards
Laura

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Subject: Re: [dispatch] TCP and SCTP connection reuse draft
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Vijay

wrt the transport aspects of this ID, shouldn't the folks in the 
transport area be contacted?

TCPM WG currently controls any TCP related efforts, and TSVWG 
controls all SCTP related efforts.

James

At 11:09 AM 9/9/2009, Vijay K. Gurbani wrote:
>Folks: Before the summer, we had released a new
>TCP/SCTP connection reuse draft in the (now defunct) SIP
>WG.  With the formation of dispatch, we were advised to
>submit it to dispatch for a decision on where to handle
>this work.
>
>Essentially, the draft documents the practice of reusing TCP
>(and SCTP) connections that is employed by some SIP
>implementations today.  TCP and SCTP transport reuse were
>originally a part of the draft-ietf-sip-connect-reuse draft,
>but were taken out before we progressed that draft to the IESG.
>
>Can we kindly ask the dispatch WG to review and comment on
>the the TCP and SCTP connection reuse draft?  All comments
>received from the first iteration of the draft in the SIP WG
>have been duly addressed in this version.
>
>It will also be helpful if the WG and chairs could provide
>some guidance on how to move this work forward (i.e.,
>AD-sponsored individual draft, take it to sipcore, etc.)
>
>The draft can be retrieved from:
>
>http://tools.ietf.org/html/draft-jain-dispatch-sip-transport-connection-reuse-00
>
>Thank you,
>
>- vijay
>--
>Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
>1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
>Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
>Web:   http://ect.bell-labs.com/who/vkg/
>_______________________________________________
>dispatch mailing list
>dispatch@ietf.org
>https://www.ietf.org/mailman/listinfo/dispatch


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Cc: "dispatch@ietf.org" <dispatch@ietf.org>, "Jain, Rajnish" <Rajnish.Jain@ipc.com>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [dispatch] TCP and SCTP connection reuse draft
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James M. Polk wrote:
> Vijay
> 
> wrt the transport aspects of this ID, shouldn't the folks in the 
> transport area be contacted?
> TCPM WG currently controls any TCP related efforts, and TSVWG controls 
> all SCTP related efforts.

James: That's an interesting comment.  However, we are not
proposing any changes to TCP or SCTP, so I am not sure that
TCPM and TSVWG will be interested in this particular work.

That said, I think TSVWG may be interested in eye-balling
this document since SCTP is (still) relatively new and
not many SIP implementations exist that do SCTP connection reuse
today.  Maybe this draft could be expert reviewed by someone
from TSVWG to make sure that our use of SCTP is indeed
within the norm?  I think this may be a good idea.

Thank you for your time and comment.

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/

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Subject: Re: [dispatch] TCP and SCTP connection reuse draft
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At 04:53 PM 9/10/2009, Vijay K. Gurbani wrote:
>James M. Polk wrote:
>>Vijay
>>wrt the transport aspects of this ID, shouldn't the folks in the 
>>transport area be contacted?
>>TCPM WG currently controls any TCP related efforts, and TSVWG 
>>controls all SCTP related efforts.
>
>James: That's an interesting comment.  However, we are not
>proposing any changes to TCP or SCTP, so I am not sure that
>TCPM and TSVWG will be interested in this particular work.
>
>That said, I think TSVWG may be interested in eye-balling
>this document since SCTP is (still) relatively new and
>not many SIP implementations exist that do SCTP connection reuse
>today.  Maybe this draft could be expert reviewed by someone
>from TSVWG to make sure that our use of SCTP is indeed
>within the norm?  I think this may be a good idea.

I think this may be a good idea too...

I wonder who you can get to ask for volunteers in TSVWG?

;-)

James


>Thank you for your time and comment.
>
>- vijay
>--
>Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
>1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
>Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
>Web:   http://ect.bell-labs.com/who/vkg/


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Hi Henry,

Thanks for your feedback :) Let me answer to the excellent points you raise=
.

First of all, I would encourage you and everyone to carefully read the Intr=
oduction chapter of the draft (http://tools.ietf.org/html/draft-veikkolaine=
n-sip-voip-xmpp-im-01) to be clear on what the proposal is about. That is, =
we are not trying to boil the ocean and define all possible ways how SIP an=
d XMPP infrastructures could be combined, but instead try to define a minim=
alistic approach whereby a client can use both SIP and XMPP in an integrate=
d manner WITHOUT requiring ANY changes on the servers that are already depl=
oyed.

See further comments below with [Markus]:

________________________________
From: ext Henry Sinnreich [mailto:hsinnrei@adobe.com]
Sent: 04 September, 2009 18:42
To: Veikkolainen Simo (Nokia-D/Helsinki); dispatch@ietf.org; mary.barnes@no=
rtel.com; gonzalo.camarillo@ericsson.com
Cc: Isomaki Markus (Nokia-CIC/Espoo); Olle E. Johansson
Subject: Re: [dispatch] New topic proposal for DISPATCH

I would like to second such work, but a far deeper look under the hood is n=
ecessary, so we don't do another Elbonia project.
http://dilbert.com/strips/comic/2009-09-01/

Here are some not very mature thoughts on the scope of the work.
The most interesting, ROI estimate and REST architecture support are at the=
 end.

Servers

Fundamentally, we need a registrar, a proxy and a presence server: Three in=
 total

 *   What will be the duplication of servers and where specifically?
 *   How does the SIP registrar communicate with the XMPP server(s)?
 *   What may be the routing protocol(s) inside a SIP+XMMP network? Think o=
f IMS as a stressing use case.

[Markus] Our starting point is that we already have quite many SIP VoIP dep=
loyments out there, as long as quite many XMPP presence & IM deployments. S=
o, the servers you mention already exist. If one now wants to develop a cli=
ent that does VoIP, IM and presence together, it should be possible to util=
ize these existing infrastrcutures and not have to wait for new ones to be =
deployed (as for instance we have waited quite many years for SIMPLE-based =
Presence servers to get deployed but compared to XMPP there's very little i=
n use). The idea is to define a couple of E2E (as in nothing that SIP proxi=
es or XMPP servers would have to support) protocol extensions which would a=
llow the client to nicely combine SIP VoIP and XMPP presence & IM.

There is no need for SIP and XMPP servers to know about each other or route=
 protocol messages between each other. All the logic is put into the endpoi=
nts. No need to worry about IMS either, except that in theory the client ca=
n use IMS for its VoIP service (similar to any other SIP infra).

User Agents

 *   Same as with servers: How many and what protocol RFCs are there in the=
 combination?
 *   Which of the "services" can be placed in the applications in the endpo=
ints or in the network application protocols, such as SIP and XMPP. This is=
 the topic on infrastructure vs. endpoint applications.

[Markus] Just take any current SIP VoIP UA implementation, XMPP client impl=
ementation and glue them together and you're already pretty far. Add a coup=
le of extensions as proposed and it's done. From there onwards you have a p=
retty powerful machinery to add applications by using SIP session setup and=
 XMPP message delivery capabilities.

Application Protocol Stacks

 *   How many protocol stacks does this involve and more to the point, how =
many RFCs?
 *   For SIP alone, see http://rfc3261.net/. What will be the total combine=
d RFC count?

[Markus] All what we expect is already pretty widely supported and even ava=
ilable as open source: basic SIP VoIP and XMPP IM and presence is enough to=
 start with.

NAT Traversal

 *   Will STUN/TURN/ICE work the same for SIP and for XMPP?

 [Markus] In our proposal SIP would be used for setting up voice & video se=
ssions. So, you would need NAT traversal for that. How that's done is outsi=
de the scope but either B2BUA or ICE based mechanisms would work as far as =
they work in general. For XMPP we would not need to worry about NAT travers=
al beyond how its done in that protocol by default. For a client there is a=
 caveat that it has to keepalive the TCP connections with BOTH SIP and XMPP=
 servers so it would be recommended to synchronize those (send them at the =
same time) for instance in a battery powered device.

Return on Investment (ROI): Cost and effort by developers. More network inf=
rastructure, not even counting B2B NEs.

 *   What is the additional time/money required for developers that know SI=
P to also learn XMPP and vice versa? This is all about return on investment=
.

[Markus] This is a problem for sure, even more so for the people writing st=
andard specs than code :)  Hopefully the developer would just have add a co=
uple of simple extensions to existing SIP and XMPP implementations.

Support for a REST-full architecture

The WWW has a REST architecture was formulated in 2000, after the emergence=
 of SIP and XMMP.  As a result of its architecture, the WWW has had a trans=
formational effect on the following (this is not a complete list). Some SIP=
 work recommends already recommends REST, such as in RFC 4240 and in http:/=
/tools.ietf.org/html/draft-griffin-bliss-rest-00

 *   Web applications from mail to office apps
 *   Internet applications previously served by dedicated network applicati=
ons protocols
 *   Many other industries, from banking to newsprint to travel. And Oh, so=
cial networks, they also have registrars...

Now please keep in mind the WWW architecture uses basically only 3 componen=
ts*: The URI for addressing, HTTP for transport and HTML for data rendering=
, augmented by some other languages (XML) and namespaces. Can we benefit so=
mething from the WWW architecture? If yes what specifically?

Heresy: If the web developers could use HTTP only, what other application l=
evel protocols do we really need except RTP/UDP?
This includes in my mind TLS and DTLS for secure transport or even better I=
MO: HIP for many reasons.

[Markus] Personally I agree that REST+HTTP+Javascript is a superior toolset=
 compared to SIP, XMPP, IMAP and so on to develop many applications. Howeve=
r, there are still plenty of SIP based VoIP and XMPP based IM and presence =
deployments out there and I don't expect the web apps to replace all of the=
m in the near future :)

* T.V. Raman: "Toward 2W, Beyond Web 2.0", Communications of the ACM, Febru=
ary 2009

Many or most of these points may be moot or make no sense, but maybe should=
 be cleared up, in case there are more naive people like me that may be int=
erested.

My strict personal 2 cents.

FRANK comments are most welcome.

Henry



On 9/4/09 7:09 AM, "Simo.Veikkolainen@nokia.com" <Simo.Veikkolainen@nokia.c=
om> wrote:

Hi,

We would like to propose a new DISPATCH topic on how SIP and XMPP can be us=
ed together in a complementary fashion.

We have been working on a solution which combines SIP based VoIP and XMPP b=
ased IM and Presence. The requirements and a proposed solution is outlined =
in http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01 <http:=
//tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01> .

The motivation for this work comes from experience which shows that most st=
andards based VoIP deployments use SIP. At the same time, the Extensible Me=
ssaging and Presence Protocol (XMPP) is widely used in instant messaging an=
d presence services. An interesting scenario arises when a SIP based voice =
(and video) service is combined together with an XMPP based instant messagi=
ng and presence service.

Note that the goal in this work is not SIP-XMPP protocol translation, but t=
o define protocol extensions and best practises such that SIP based VoIP an=
d XMPP based IM and presence could be seamlessly combined and offered to th=
e end user. For rapid deployment, we assume no changes in the server infras=
tructure, and instead impose new requirements and protocol extensions only =
to the endpoints.

We would like to get some discussion going on the use case itself as well a=
s on the solution. Also, we would like to hear you thoughts on DISPATCH or =
a new "dispatched" mini-WG as the home for such work.

Exact charter proposal and problem statemen is to follow.

Regards,
Simo and Markus




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<DIV dir=3Dltr align=3Dleft><SPAN class=3D185332211-11092009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Hi Henry,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D185332211-11092009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D185332211-11092009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Thanks for your feedback :) Let me answer to the e=
xcellent=20
points you raise.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D185332211-11092009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D185332211-11092009><FONT face=3DA=
rial=20
color=3D#0000ff><FONT size=3D2>First of all, I would encourage you and ever=
yone to=20
carefully read the Introduction chapter of the draft (</FONT><A=20
href=3D"http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01">=
<FONT=20
size=3D2>http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01<=
/FONT></A><FONT=20
size=3D2>) to be clear on what the proposal is about. That is, we are not t=
rying=20
to boil the ocean and define all possible ways how SIP and XMPP infrastruct=
ures=20
could be combined, but instead try to define a minimalistic approach whereb=
y a=20
client can use both SIP and XMPP in an integrated manner WITHOUT requiring =
ANY=20
changes on the servers that are already deployed.</FONT></FONT></SPAN></DIV=
>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D185332211-11092009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D185332211-11092009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>See further comments below with=20
[Markus]:</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px soli=
d; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> ext Henry Sinnreich=20
  [mailto:hsinnrei@adobe.com] <BR><B>Sent:</B> 04 September, 2009=20
  18:42<BR><B>To:</B> Veikkolainen Simo (Nokia-D/Helsinki); dispatch@ietf.o=
rg;=20
  mary.barnes@nortel.com; gonzalo.camarillo@ericsson.com<BR><B>Cc:</B> Isom=
aki=20
  Markus (Nokia-CIC/Espoo); Olle E. Johansson<BR><B>Subject:</B> Re: [dispa=
tch]=20
  New topic proposal for DISPATCH<BR></FONT><BR></DIV>
  <DIV></DIV><FONT size=3D2><FONT face=3D"Calibri, Verdana, Helvetica, Aria=
l"><SPAN=20
  style=3D"FONT-SIZE: 14pt">I would like to second such work, but a far dee=
per=20
  look under the hood is necessary, so we don&#8217;t do another Elbonia=20
  project.<BR><A=20
  href=3D"http://dilbert.com/strips/comic/2009-09-01/">http://dilbert.com/s=
trips/comic/2009-09-01/</A><BR><BR>Here=20
  are some not very mature thoughts on the scope of the work. <BR>The most=
=20
  interesting, ROI estimate and REST architecture support are at the=20
  end.<BR><BR>Servers<BR><BR>Fundamentally, we need a registrar, a proxy an=
d a=20
  presence server: Three in total<BR></SPAN></FONT></FONT>
  <UL>
    <LI><FONT size=3D2><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><S=
PAN=20
    style=3D"FONT-SIZE: 14pt">What will be the duplication of servers and w=
here=20
    specifically? </SPAN></FONT></FONT>
    <LI><FONT size=3D2><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><S=
PAN=20
    style=3D"FONT-SIZE: 14pt">How does the SIP registrar communicate with t=
he XMPP=20
    server(s)? </SPAN></FONT></FONT>
    <LI><FONT size=3D2><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><S=
PAN=20
    style=3D"FONT-SIZE: 14pt">What may be the routing protocol(s) inside a=
=20
    SIP+XMMP network? Think of IMS as a stressing use=20
    case.<BR></SPAN></FONT></FONT></LI></UL>
  <DIV><FONT size=3D2><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SP=
AN=20
  style=3D"FONT-SIZE: 14pt"><SPAN class=3D185332211-11092009><FONT face=3DA=
rial=20
  color=3D#0000ff size=3D2>[Markus] Our starting point is that we already h=
ave quite=20
  many SIP VoIP deployments out there, as long as quite many XMPP presence =
&amp;=20
  IM deployments. So, the servers you mention already&nbsp;exist. If one no=
w=20
  wants to develop a client that does VoIP, IM and presence together, it sh=
ould=20
  be possible to utilize these existing infrastrcutures and not have to wai=
t for=20
  new ones to be deployed (as for instance we have waited quite many years =
for=20
  SIMPLE-based Presence servers to get deployed but compared to XMPP there'=
s=20
  very little&nbsp;in use). The idea is to define a couple of&nbsp;E2E (as =
in=20
  nothing that SIP proxies or XMPP servers would have to support) protocol=
=20
  extensions which would allow the client to nicely combine SIP VoIP and XM=
PP=20
  presence &amp; IM.&nbsp;</FONT></SPAN></SPAN></FONT></FONT></DIV>
  <DIV><FONT size=3D2><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SP=
AN=20
  style=3D"FONT-SIZE: 14pt"><SPAN=20
  class=3D185332211-11092009></SPAN></SPAN></FONT></FONT>&nbsp;</DIV>
  <DIV><FONT size=3D2><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SP=
AN=20
  style=3D"FONT-SIZE: 14pt"><SPAN class=3D185332211-11092009><FONT face=3DA=
rial=20
  color=3D#0000ff size=3D2>There is no need for SIP and XMPP servers to kno=
w about=20
  each other or route protocol messages between each other. All the logic i=
s put=20
  into the endpoints. No need to worry about IMS either, except=20
  that&nbsp;in&nbsp;theory the client can use IMS for its VoIP&nbsp;service=
=20
  (similar to any other SIP=20
  infra).</FONT>&nbsp;</SPAN></SPAN></FONT></FONT></DIV>
  <DIV><FONT size=3D2><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SP=
AN=20
  style=3D"FONT-SIZE: 14pt"><SPAN class=3D185332211-11092009>&nbsp;</SPAN><=
BR>User=20
  Agents</DIV></SPAN></FONT></FONT>
  <UL>
    <LI><FONT size=3D2><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><S=
PAN=20
    style=3D"FONT-SIZE: 14pt">Same as with servers: How many and what proto=
col=20
    RFCs are there in the combination? </SPAN></FONT></FONT>
    <LI><FONT size=3D2><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><S=
PAN=20
    style=3D"FONT-SIZE: 14pt">Which of the &#8220;services&#8221; can be pl=
aced in the=20
    applications in the endpoints or in the network application protocols, =
such=20
    as SIP and XMPP. This is the topic on infrastructure vs. endpoint=20
    applications.<BR></SPAN></FONT></FONT></LI></UL>
  <DIV><FONT size=3D2><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SP=
AN=20
  style=3D"FONT-SIZE: 14pt"><SPAN class=3D185332211-11092009><FONT face=3DA=
rial=20
  color=3D#0000ff size=3D2>[Markus] Just take any current SIP VoIP=20
  UA&nbsp;implementation, XMPP&nbsp;client implementation and glue them tog=
ether=20
  and you're already pretty far. Add a couple of extensions as proposed and=
 it's=20
  done. From there onwards you have a pretty powerful machinery to add=20
  applications by using SIP session setup and XMPP message delivery=20
  capabilities. </FONT></SPAN></SPAN></FONT></FONT></DIV>
  <DIV><FONT size=3D2><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SP=
AN=20
  style=3D"FONT-SIZE: 14pt"><SPAN=20
  class=3D185332211-11092009>&nbsp;</SPAN><BR>Application Protocol=20
  Stacks</DIV></SPAN></FONT></FONT>
  <UL>
    <LI><FONT size=3D2><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><S=
PAN=20
    style=3D"FONT-SIZE: 14pt">How many protocol stacks does this involve an=
d more=20
    to the point, how many RFCs? </SPAN></FONT></FONT>
    <LI><FONT size=3D2><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><S=
PAN=20
    style=3D"FONT-SIZE: 14pt">For SIP alone, see <A=20
    href=3D"http://rfc3261.net/">http://rfc3261.net/</A>. What will be the =
total=20
    combined RFC count?<BR></SPAN></FONT></FONT></LI></UL>
  <DIV><FONT size=3D2><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SP=
AN=20
  style=3D"FONT-SIZE: 14pt"><SPAN class=3D185332211-11092009><FONT face=3DA=
rial=20
  color=3D#0000ff size=3D2>[Markus] All what we&nbsp;expect is already pret=
ty widely=20
  supported and even available as open source: basic SIP VoIP and XMPP IM a=
nd=20
  presence is enough&nbsp;to start with.=20
  &nbsp;</FONT></SPAN></SPAN></FONT></FONT></DIV>
  <DIV><FONT size=3D2><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SP=
AN=20
  style=3D"FONT-SIZE: 14pt"><SPAN class=3D185332211-11092009>&nbsp;</SPAN><=
BR>NAT=20
  Traversal</DIV></SPAN></FONT></FONT>
  <UL>
    <LI><FONT size=3D2><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><S=
PAN=20
    style=3D"FONT-SIZE: 14pt">Will STUN/TURN/ICE work the same for SIP and =
for=20
    XMPP?<BR></SPAN></FONT></FONT><FONT size=3D2><FONT=20
    face=3D"Calibri, Verdana, Helvetica, Arial"><SPAN=20
    style=3D"FONT-SIZE: 14pt"><SPAN class=3D185332211-11092009><FONT face=
=3DArial=20
    color=3D#0000ff size=3D2></FONT></SPAN></SPAN></FONT></FONT></LI></UL>
  <DIV><FONT size=3D2><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SP=
AN=20
  style=3D"FONT-SIZE: 14pt"><SPAN class=3D185332211-11092009>&nbsp;<FONT fa=
ce=3DArial=20
  color=3D#0000ff size=3D2>[Markus] In our proposal SIP would be used for s=
etting up=20
  voice &amp; video sessions. So, you would need NAT traversal for that. Ho=
w=20
  that's done is outside the scope but either B2BUA or ICE based mechanisms=
=20
  would work as far as they work in general. For XMPP we would not need to =
worry=20
  about NAT traversal beyond how its done in that protocol by default. For =
a=20
  client there is a caveat that it has to keepalive the TCP connections wit=
h=20
  BOTH SIP and XMPP servers so it would be recommended to synchronize those=
=20
  (send them at the same time)&nbsp;for instance in a battery powered=20
  device.</FONT></SPAN></SPAN></FONT></FONT></DIV>
  <DIV><FONT size=3D2><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SP=
AN=20
  style=3D"FONT-SIZE: 14pt"><SPAN class=3D185332211-11092009>&nbsp;</SPAN><=
BR>Return=20
  on Investment (ROI): Cost and effort by developers. More network=20
  infrastructure, not even counting B2B NEs.<BR></DIV></SPAN></FONT></FONT>
  <UL>
    <LI><FONT size=3D2><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><S=
PAN=20
    style=3D"FONT-SIZE: 14pt">What is the additional time/money required fo=
r=20
    developers that know SIP to also learn XMPP and vice versa? This is all=
=20
    about return on investment.<BR></SPAN></FONT></FONT></LI></UL>
  <DIV><FONT size=3D2><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SP=
AN=20
  style=3D"FONT-SIZE: 14pt"><SPAN class=3D185332211-11092009><FONT face=3DA=
rial=20
  color=3D#0000ff size=3D2>[Markus] This is a problem for sure,&nbsp;even m=
ore so=20
  for the people writing standard specs than code :)&nbsp; Hopefully the=20
  developer would just have add a couple of simple extensions to&nbsp;exist=
ing=20
  SIP and XMPP implementations.&nbsp;</FONT></SPAN></SPAN></FONT></FONT></D=
IV>
  <DIV><FONT size=3D2><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SP=
AN=20
  style=3D"FONT-SIZE: 14pt"><SPAN=20
  class=3D185332211-11092009>&nbsp;</SPAN><BR>Support for a REST-full=20
  architecture<BR><BR>The WWW has a REST architecture was formulated in 200=
0,=20
  after the emergence of SIP and XMMP. &nbsp;As a result of its architectur=
e,=20
  the WWW has had a transformational effect on the following (this is not a=
=20
  complete list). Some SIP work recommends already recommends REST, such as=
 in=20
  RFC 4240 and in <A=20
  href=3D"http://tools.ietf.org/html/draft-griffin-bliss-rest-00">http://to=
ols.ietf.org/html/draft-griffin-bliss-rest-00</A><BR></DIV></SPAN></FONT></=
FONT>
  <UL>
    <LI><FONT size=3D2><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><S=
PAN=20
    style=3D"FONT-SIZE: 14pt">Web applications from mail to office apps=20
    </SPAN></FONT></FONT>
    <LI><FONT size=3D2><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><S=
PAN=20
    style=3D"FONT-SIZE: 14pt">Internet applications previously served by de=
dicated=20
    network applications protocols </SPAN></FONT></FONT>
    <LI><FONT size=3D2><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><S=
PAN=20
    style=3D"FONT-SIZE: 14pt">Many other industries, from banking to newspr=
int to=20
    travel. And Oh, social networks, they also have=20
    registrars...<BR></SPAN></FONT></FONT></LI></UL><FONT size=3D2><FONT=20
  face=3D"Calibri, Verdana, Helvetica, Arial"><SPAN style=3D"FONT-SIZE: 14p=
t">
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT><FONT face=3DAria=
l=20
  color=3D#0000ff size=3D2></FONT><FONT face=3DArial color=3D#0000ff=20
  size=3D2></FONT><BR>Now please keep in mind the WWW architecture uses bas=
ically=20
  only 3 components*: The URI for addressing, HTTP for transport and HTML f=
or=20
  data rendering, augmented by some other languages (XML) and namespaces. C=
an we=20
  benefit something from the WWW architecture? If yes what=20
  specifically?<BR><BR>Heresy: If the web developers could use HTTP only, w=
hat=20
  other application level protocols do we really need except RTP/UDP?<BR>Th=
is=20
  includes in my mind TLS and DTLS for secure transport or even better IMO:=
 HIP=20
  for many reasons.<BR><SPAN class=3D185332211-11092009><FONT face=3DArial=
=20
  color=3D#0000ff size=3D2>&nbsp;</FONT></SPAN></DIV>
  <DIV><SPAN class=3D185332211-11092009><FONT face=3DArial color=3D#0000ff=
=20
  size=3D2>[Markus] Personally I agree that REST+HTTP+Javascript is a super=
ior=20
  toolset compared to SIP, XMPP, IMAP and so on to develop many application=
s.=20
  However, there are still plenty of SIP based VoIP and XMPP based IM and=20
  presence deployments out there and I don't expect the web apps to replace=
 all=20
  of them in the near future :)</FONT></SPAN></DIV>
  <DIV><SPAN class=3D185332211-11092009>&nbsp;</SPAN><BR>* T.V. Raman: &#82=
20;Toward 2W,=20
  Beyond Web 2.0&#8221;, Communications of the ACM, February 2009<BR><BR>Ma=
ny or most=20
  of these points may be moot or make no sense, but maybe should be cleared=
 up,=20
  in case there are more naive people like me that may be interested.<BR><B=
R>My=20
  strict personal 2 cents.<BR><BR>FRANK comments are most=20
  welcome.<BR><BR>Henry<BR></SPAN></FONT></FONT><FONT=20
  face=3D"Calibri, Verdana, Helvetica, Arial"><SPAN=20
  style=3D"FONT-SIZE: 18pt"><BR><BR><BR>On 9/4/09 7:09 AM, "<A=20
  href=3D"Simo.Veikkolainen@nokia.com">Simo.Veikkolainen@nokia.com</A>" &lt=
;<A=20
  href=3D"Simo.Veikkolainen@nokia.com">Simo.Veikkolainen@nokia.com</A>&gt;=
=20
  wrote:<BR><BR></DIV></SPAN></FONT>
  <BLOCKQUOTE><SPAN style=3D"FONT-SIZE: 18pt"><FONT=20
    face=3DArial>Hi,<BR>&nbsp;<BR>We would like to propose a new DISPATCH t=
opic on=20
    how SIP and XMPP can be used together in a complementary=20
    fashion.<BR>&nbsp;<BR>We have been working on a solution which combines=
 SIP=20
    based VoIP and XMPP based IM and Presence. The requirements and a propo=
sed=20
    solution is outlined in <FONT color=3D#0000ff><U><A=20
    href=3D"http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-=
01">http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01</A></=
U></FONT>=20
    &lt;<A=20
    href=3D"http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-=
01">http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01</A>&g=
t;=20
    .<BR>&nbsp;<BR>The motivation for this work comes from experience which=
=20
    shows that most standards based VoIP deployments use SIP. At the same t=
ime,=20
    the Extensible Messaging and Presence Protocol (XMPP) is widely used in=
=20
    instant messaging and presence services. An interesting scenario arises=
 when=20
    a SIP based voice (and video) service is combined together with an XMPP=
=20
    based instant messaging and presence service. <BR>&nbsp;<BR>Note that t=
he=20
    goal in this work is not SIP-XMPP protocol translation, but to define=20
    protocol extensions and best practises such that SIP based VoIP and XMP=
P=20
    based IM and presence could be seamlessly combined and offered to the e=
nd=20
    user. For rapid deployment, we assume no changes in the server=20
    infrastructure, and instead impose new requirements and protocol extens=
ions=20
    only to the endpoints.<BR>&nbsp;<BR>We would like to get some discussio=
n=20
    going on the use case itself as well as on the solution. Also, we would=
 like=20
    to hear you thoughts on DISPATCH or a new "dispatched" mini-WG as the h=
ome=20
    for such work.<BR>&nbsp;<BR>Exact charter proposal and problem statemen=
 is=20
    to follow. <BR>&nbsp;<BR>Regards,<BR>Simo and=20
    Markus<BR>&nbsp;<BR>&nbsp;<BR></FONT><FONT=20
    face=3D"Calibri, Verdana, Helvetica, Arial"><BR></FONT></SPAN></BLOCKQU=
OTE></BLOCKQUOTE></BODY></HTML>

--_000_B3F72E5548B10A4A8E6F4795430F8418125E6740DFNOKEUMSG02mgd_--

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Cc: dispatch@ietf.org, 'Gonzalo Camarillo' <gonzalo.camarillo@ericsson.com>, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [dispatch] IETF-76 DISPATCH WG deadlines : New Topic Domain	Registration
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Dean,

The plan is not to rewrite the Internet's name architecture nor replace 
RFC 3263.

The plan is to standardize a function that has been shown to be needed 
in the marketplace for the deployment of SIP in certain applications.

And yes, dynamic DNS is an alternative, it does work, but again, there 
are reasons why it is not being used in this certain application.

Anyway, lets start this conversation properly when we have a draft.

Thanks,
Alan


Dean Willis wrote:
>
> On Sep 4, 2009, at 2:53 PM, Richard Shockey wrote:
>
>> Dispatch Chairs:  We would like to reserve some time during DISPATCH to
>> discuss the topic of domain registration in SIP.
>
> It seems like you're proposing a deep and fundamental rewrite of the 
> Internet's name architecture, replacing RFC 3263 and its use of DNS 
> with some sort of SIP REGISTER technique.
>
> Or maybe I'm missing it entirely, and what you're proposing is a way 
> to use SIP REGISTER messages to populate a DNS SRV record within a 
> registry?
>
> This raises the question of delegation to that registry -- the 
> registry has to be authoritative for all of  example.com in order to 
> be able to host an srv record for example.com. So your mythical SMB is 
> still going to require a DNS service provider that has to be 
> meaningfully provisioned. It's just a question of the outsourcing 
> contract, and I fail to see a need for protocol work here. After all, 
> we have dynamic DNS, and it works pretty darned well for me today.
>
> So what you talking 'bout, Shockey?
>
> -- 
> Dean
> _______________________________________________
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From: "John-Luc Bakker" <jbakker@rim.com>
To: "Gilad Shaham" <gshaham@juniper.net>, "Avasarala Ranjit-A20990" <ranjit@motorola.com>, <dispatch@ietf.org>
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Hi Gilad,

 

Regarding:

 

> Page 21:

> 503 is there, but I don't see 500. Some implementations will avoid 503
and use 500 due to 

> discussion related to this
http://tools.ietf.org/html/draft-hilt-sip-correction-503-01 (now
expired,

>  but still affected some vendor decisions). I might be able to think
of some scenario that 

> involves 502, but I assume this is a result of the diversion
implementation itself so maybe 

> that's the context of this discussion.

 

The referenced expired draft mentions overload situations. 503

(Service Unavailable) response code is provided as a remedy for servers
under

Overload, the draft says.  

 

The 503 code in this document is to be interpreted as "subscriber not
reachable" (e.g. see RFC 4458).

Moreover, sending a NOTIFY with an XML document indicating a diversion
(re-directions or forwarding) 

of an incoming communication session request due to a server in a
overload situation, may even amplify 

the overload condition.

 

Regards,

 

            John-Luc

 

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Gilad Shaham
Sent: Monday, August 17, 2009 7:20 PM
To: Avasarala Ranjit-A20990; dispatch@ietf.org; sip@ietf.org
Subject: Re: [Sip] New
versionof"draft-avasasarala-dispatch-comm-diversion-info" draft
submitted

 

Hi,

 

See some comments

 

Page 5:

   "... For e.g. the subscriber wanted to diverted all incoming calls to
voice-mail,
   between 3.00 p.m. to 4.00 p.m.  Yet, by mistake she configures the
   time-duration as 3.00 to 4.00 p.m"

Some of the sentence needs restructuring and I also don't fully
understand the example. Is it AM-PM or wrong field was configured?

 

Page 12:

What if time-range is missing? What should be the default? Sounds to me
the default should be the current time with no end date.

 

Page 14:

In Comm-div-info-selection-criteria there are several disable-*
subsections, yet their text describes these element gives the subscriber
option of adding information. Shouldn't this be for omitting information
or alternatively, call these elements "enable-*" or did I misunderstand
the purpose.

 

Page 16:

             <user-name>Boss</originating-user-name>

Should be

             <user-name>Boss</user-name>

It might be also useful to see an example of periodic request.

 

Page 21:

503 is there, but I don't see 500. Some implementations will avoid 503
and use 500 due to discussion related to this
http://tools.ietf.org/html/draft-hilt-sip-correction-503-01 (now
expired, but still affected some vendor decisions). I might be able to
think of some scenario that involves 502, but I assume this is a result
of the diversion implementation itself so maybe that's the context of
this discussion.

 

Thanks

Gilad

 

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Avasarala Ranjit-A20990
Sent: Monday, August 17, 2009 10:07 AM
To: dispatch@ietf.org; sip@ietf.org
Subject: [Sip] New version
of"draft-avasasarala-dispatch-comm-diversion-info" draft submitted

 

Hi All 

We have submitted an updated version of
draft-avasasarala-dispatch-comm-diversion-info 

It can be accessed at:
http://www.ietf.org/internet-drafts/draft-avasarala-dispatch-comm-div-no
tification-01.txt
<http://www.ietf.org/internet-drafts/draft-avasarala-dispatch-comm-div-n
otification-01.txt> 

This draft proposes a SIP Event package for Communication Diversions
Notification Information and conforms to procedures and schema described
in 3GPP TS 24.604. 

Please review and comment

Regards 
Ranjit 


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<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>Hi Gilad,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>Regarding:<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>&gt; Page 21:<o:p></o:p></span></font><=
/p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>&gt; 503 is there, but I don&#8217;t se=
e
500. Some implementations will avoid 503 and use 500 due to <o:p></o:p></spa=
n></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>&gt; discussion related to this <a
href=3D"http://tools.ietf.org/html/draft-hilt-sip-correction-503-01"
title=3D"blocked::http://tools.ietf.org/html/draft-hilt-sip-correction-503-0=
1">http://tools.ietf.org/html/draft-hilt-sip-correction-503-01</a>
(now expired,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>&gt; &nbsp;but still affected some vend=
or
decisions). I might be able to think of some scenario that <o:p></o:p></span=
></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>&gt; involves 502, but I assume this is=
 a
result of the diversion implementation itself so maybe <o:p></o:p></span></f=
ont></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>&gt; that&#8217;s the context of this
discussion.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>The referenced expired draft mentions
overload situations. 503<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>(Service Unavailable) response code is
provided as a remedy for servers under<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>Overload, the draft says.&nbsp; <o:p></=
o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>The 503 code in this document is to be=
 interpreted
as &#8220;subscriber not reachable&#8221; (e.g. see RFC 4458).<o:p></o:p></s=
pan></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>Moreover, sending a NOTIFY with an XML
document indicating a diversion (re-directions or forwarding) <o:p></o:p></s=
pan></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>of an incoming communication session re=
quest
due to a server in a overload situation, may even amplify <o:p></o:p></span>=
</font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>the overload condition.<o:p></o:p></spa=
n></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>Regards,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp; John-Luc<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font size=
=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span style=3D'font-siz=
e:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'>
sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <b><span style=3D'font-we=
ight:
bold'>On Behalf Of </span></b>Gilad Shaham<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Monday, August 17, 2009=
 7:20
PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Avasarala Ranjit-A20990;
dispatch@ietf.org; sip@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [Sip] New
versionof&quot;draft-avasasarala-dispatch-comm-diversion-info&quot; draft
submitted</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D'=
font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>Hi,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>See some comments<o:p></o:p></span></fo=
nt></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>Page 5:<o:p></o:p></span></font></p>

<pre><font size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>&n=
bsp;&nbsp; &#8220;... For e.g. the subscriber wanted to diverted all incomin=
g calls to voice-mail,<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>&nbsp;&nbsp;=
 between 3.00 p.m. to 4.00 p.m.&nbsp; Yet, by mistake she configures the<o:p=
></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>&nbsp;&nbsp;=
 time-duration as 3.00 to 4.00 p.m&#8221;<o:p></o:p></span></font></pre>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>Some of the sentence needs restructurin=
g
and I also don&#8217;t fully understand the example. Is it AM-PM or wrong fi=
eld
was configured?<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>Page 12:<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>What if time-range is missing? What sho=
uld
be the default? Sounds to me the default should be the current time with no=
 end
date.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>Page 14:<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>In Comm-div-info-selection-criteria the=
re
are several disable-* subsections, yet their text describes these element gi=
ves
the subscriber option of adding information. Shouldn&#8217;t this be for
omitting information or alternatively, call these elements
&#8220;enable-*&#8221; or did I misunderstand the purpose.<o:p></o:p></span>=
</font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>Page 16:<o:p></o:p></span></font></p>

<pre><font size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;u=
ser-name&gt;Boss&lt;/originating-user-name&gt;<o:p></o:p></span></font></pre=
>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>Should be<o:p></o:p></span></font></p>

<pre><font size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;u=
ser-name&gt;Boss&lt;/user-name&gt;<o:p></o:p></span></font></pre>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>It might be also useful to see an examp=
le
of periodic request.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>Page 21:<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>503 is there, but I don&#8217;t see 500=
.
Some implementations will avoid 503 and use 500 due to discussion related to
this <a href=3D"http://tools.ietf.org/html/draft-hilt-sip-correction-503-01"=
>http://tools.ietf.org/html/draft-hilt-sip-correction-503-01</a>
(now expired, but still affected some vendor decisions). I might be able to
think of some scenario that involves 502, but I assume this is a result of t=
he
diversion implementation itself so maybe that&#8217;s the context of this
discussion.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>Thanks<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'>Gilad<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font size=
=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span style=3D'font-siz=
e:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> sip-bounc=
es@ietf.org
[mailto:sip-bounces@ietf.org] <b><span style=3D'font-weight:bold'>On Behalf=
 Of </span></b>Avasarala
Ranjit-A20990<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Monday, August 17, 2009
10:07 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> dispatch@ietf.org;
sip@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> [Sip] New version
of&quot;draft-avasasarala-dispatch-comm-diversion-info&quot; draft submitted=
</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D'=
font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span style=3D=
'font-size:
10.0pt;font-family:Arial;color:blue'>Hi All</span></font> <o:p></o:p></p>

<p><font size=3D2 color=3Dblue face=3DArial><span style=3D'font-size:10.0pt;=
font-family:
Arial;color:blue'>We have submitted an updated version of
draft-avasasarala-dispatch-comm-diversion-info </span></font><o:p></o:p></p>

<p><font size=3D2 color=3Dblue face=3DArial><span style=3D'font-size:10.0pt;=
font-family:
Arial;color:blue'>It can be accessed at: </span></font><font size=3D2 face=
=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'><a
href=3D"http://www.ietf.org/internet-drafts/draft-avasarala-dispatch-comm-di=
v-notification-01.txt"
target=3D"_top" jQuery1250492662043=3D3><font size=3D3 face=3D"Times New Rom=
an"><span
style=3D'font-size:12.0pt;font-family:"Times New Roman"'>http://www.ietf.org=
/internet-drafts/draft-avasarala-dispatch-comm-div-notification-01.txt</span=
></font></a></span></font><o:p></o:p></p>

<p><font size=3D2 color=3Dblue face=3DArial><span style=3D'font-size:10.0pt;=
font-family:
Arial;color:blue'>This draft proposes a SIP Event package for Communication
Diversions Notification Information and conforms to procedures and schema
described in 3GPP TS 24.604.&nbsp;</span></font><o:p></o:p></p>

<p><font size=3D2 color=3Dblue face=3DArial><span style=3D'font-size:10.0pt;=
font-family:
Arial;color:blue'>Please review and comment</span></font><o:p></o:p></p>

<p><font size=3D2 color=3Dblue face=3DArial><span style=3D'font-size:10.0pt;=
font-family:
Arial;color:blue'>Regards</span></font> <br>
<font size=3D2 color=3Dblue face=3DArial><span style=3D'font-size:10.0pt;fon=
t-family:
Arial;color:blue'>Ranjit</span></font> <o:p></o:p></p>

</div>

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From: Henry Sinnreich <hsinnrei@adobe.com>
To: "Markus.Isomaki@nokia.com" <Markus.Isomaki@nokia.com>, "Simo.Veikkolainen@nokia.com" <Simo.Veikkolainen@nokia.com>, "dispatch@ietf.org" <dispatch@ietf.org>, "mary.barnes@nortel.com" <mary.barnes@nortel.com>, "gonzalo.camarillo@ericsson.com" <gonzalo.camarillo@ericsson.com>
Date: Fri, 11 Sep 2009 10:41:02 -0700
Thread-Topic: [dispatch] New topic proposal for DISPATCH
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Hi Markus,

Thanks for your thoughtful reply and I agree there are some scenarios where=
 it makes sense.
More important, doing this in the endpoints is the right approach IMO and a=
s you know, I will always defend the Internet end-to-end principle :-)

This distributed approach is also IMO a good answer to the Interdomain Scal=
ing problem for IM
( I-D.draft-ietf-simple-interdomain-scaling-analysis) and I have copied one=
 of its authors here.

This raises the questions:


 1.  Why stop at SIP and XMPP, and not add other popular IM protocols, as i=
s already done by quite a few products in the market?
 2.  Can your approach be generalized in a more generic way?

If endpoints are accommodating several IM protocols, an IETF, Dispatch reco=
mmended GENERIC approach would make many people feel more comfortable about=
 it. Your I-D could be used as a tested example for a generic approach.

Ranting on preferring endpoint applications to more application protocols

I still believe creative application developers face the choice of either i=
nventing new applications or dedicating their time to learning application =
protocols and they cannot do both. Learning and following both SIP and XMPP=
 to write and update code is quite a challenge.
Is there a better way?
For this reason, one single application protocol for IM and voice is enough=
.
Proof: Almost all the applications that make people buy smart phones, use o=
nline office apps and enterprise apps run on HTTP(S) and their application =
developers don't have to worry about knowing how HTTP works. As a result, H=
TTP has been also used to even carry streaming media, web conferencing and =
IM. Just to avoid spending a lifetime to learn and follow application proto=
cols. But HTTP based apps dominate the application space. This is however f=
or another discussion elsewhere.

My strictly personal 2 cents.

Thanks again,

Henry


On 9/11/09 8:11 AM, "Markus.Isomaki@nokia.com" <Markus.Isomaki@nokia.com> w=
rote:

Hi Henry,

Thanks for your feedback :) Let me answer to the excellent points you raise=
.

First of all, I would encourage you and everyone to carefully read the Intr=
oduction chapter of the draft (http://tools.ietf.org/html/draft-veikkolaine=
n-sip-voip-xmpp-im-01 <http://tools.ietf.org/html/draft-veikkolainen-sip-vo=
ip-xmpp-im-01> ) to be clear on what the proposal is about. That is, we are=
 not trying to boil the ocean and define all possible ways how SIP and XMPP=
 infrastructures could be combined, but instead try to define a minimalisti=
c approach whereby a client can use both SIP and XMPP in an integrated mann=
er WITHOUT requiring ANY changes on the servers that are already deployed.

See further comments below with [Markus]:



________________________________
From: ext Henry Sinnreich  [mailto:hsinnrei@adobe.com]
Sent: 04 September, 2009  18:42
To: Veikkolainen Simo (Nokia-D/Helsinki); dispatch@ietf.org;  mary.barnes@n=
ortel.com; gonzalo.camarillo@ericsson.com
Cc: Isomaki  Markus (Nokia-CIC/Espoo); Olle E. Johansson
Subject: Re: [dispatch]  New topic proposal for DISPATCH


I would like to second such work, but a far deeper  look under the hood is =
necessary, so we don't do another Elbonia  project.
http://dilbert.com/strips/comic/2009-09-01/

Here  are some not very mature thoughts on the scope of the work.
The most  interesting, ROI estimate and REST architecture support are at th=
e  end.

Servers

Fundamentally, we need a registrar, a proxy and a  presence server: Three i=
n total


 *   What will be the duplication of servers and where  specifically?
 *   How does the SIP registrar communicate with the XMPP  server(s)?
 *   What may be the routing protocol(s) inside a  SIP+XMMP network? Think =
of IMS as a stressing use  case.

[Markus] Our starting point is that we already have quite  many SIP VoIP de=
ployments out there, as long as quite many XMPP presence &  IM deployments.=
 So, the servers you mention already exist. If one now  wants to develop a =
client that does VoIP, IM and presence together, it should  be possible to =
utilize these existing infrastrcutures and not have to wait for  new ones t=
o be deployed (as for instance we have waited quite many years for  SIMPLE-=
based Presence servers to get deployed but compared to XMPP there's  very l=
ittle in use). The idea is to define a couple of E2E (as in  nothing that S=
IP proxies or XMPP servers would have to support) protocol  extensions whic=
h would allow the client to nicely combine SIP VoIP and XMPP  presence & IM=
.



There is no need for SIP and XMPP servers to know about  each other or rout=
e protocol messages between each other. All the logic is put  into the endp=
oints. No need to worry about IMS either, except  that in theory the client=
 can use IMS for its VoIP service  (similar to any other SIP  infra).


User  Agents


 *   Same as with servers: How many and what protocol  RFCs are there in th=
e combination?
 *   Which of the "services" can be placed in the  applications in the endp=
oints or in the network application protocols, such  as SIP and XMPP. This =
is the topic on infrastructure vs. endpoint  applications.

[Markus] Just take any current SIP VoIP  UA implementation, XMPP client imp=
lementation and glue them together  and you're already pretty far. Add a co=
uple of extensions as proposed and it's  done. From there onwards you have =
a pretty powerful machinery to add  applications by using SIP session setup=
 and XMPP message delivery  capabilities.


Application Protocol  Stacks


 *   How many protocol stacks does this involve and more  to the point, how=
 many RFCs?
 *   For SIP alone, see http://rfc3261.net/. What will be the total  combin=
ed RFC count?

[Markus] All what we expect is already pretty widely  supported and even av=
ailable as open source: basic SIP VoIP and XMPP IM and  presence is enough =
to start with.


NAT  Traversal


 *   Will STUN/TURN/ICE work the same for SIP and for  XMPP?

[Markus] In our proposal SIP would be used for setting up  voice & video se=
ssions. So, you would need NAT traversal for that. How  that's done is outs=
ide the scope but either B2BUA or ICE based mechanisms  would work as far a=
s they work in general. For XMPP we would not need to worry  about NAT trav=
ersal beyond how its done in that protocol by default. For a  client there =
is a caveat that it has to keepalive the TCP connections with  BOTH SIP and=
 XMPP servers so it would be recommended to synchronize those  (send them a=
t the same time) for instance in a battery powered  device.


Return  on Investment (ROI): Cost and effort by developers. More network  i=
nfrastructure, not even counting B2B NEs.


 *   What is the additional time/money required for  developers that know S=
IP to also learn XMPP and vice versa? This is all  about return on investme=
nt.

[Markus] This is a problem for sure, even more so  for the people writing s=
tandard specs than code :)  Hopefully the  developer would just have add a =
couple of simple extensions to existing  SIP and XMPP implementations.


Support for a REST-full  architecture

The WWW has a REST architecture was formulated in 2000,  after the emergenc=
e of SIP and XMMP.  As a result of its architecture,  the WWW has had a tra=
nsformational effect on the following (this is not a  complete list). Some =
SIP work recommends already recommends REST, such as in  RFC 4240 and in ht=
tp://tools.ietf.org/html/draft-griffin-bliss-rest-00


 *   Web applications from mail to office apps
 *   Internet applications previously served by dedicated  network applicat=
ions protocols
 *   Many other industries, from banking to newsprint to  travel. And Oh, s=
ocial networks, they also have  registrars...


Now please keep in mind the WWW architecture uses basically  only 3 compone=
nts*: The URI for addressing, HTTP for transport and HTML for  data renderi=
ng, augmented by some other languages (XML) and namespaces. Can we  benefit=
 something from the WWW architecture? If yes what  specifically?

Heresy: If the web developers could use HTTP only, what  other application =
level protocols do we really need except RTP/UDP?
This  includes in my mind TLS and DTLS for secure transport or even better =
IMO: HIP  for many reasons.


[Markus] Personally I agree that REST+HTTP+Javascript is a superior  toolse=
t compared to SIP, XMPP, IMAP and so on to develop many applications.  Howe=
ver, there are still plenty of SIP based VoIP and XMPP based IM and  presen=
ce deployments out there and I don't expect the web apps to replace all  of=
 them in the near future :)


* T.V. Raman: "Toward 2W,  Beyond Web 2.0", Communications of the ACM, Febr=
uary 2009

Many or most  of these points may be moot or make no sense, but maybe shoul=
d be cleared up,  in case there are more naive people like me that may be i=
nterested.

My  strict personal 2 cents.

FRANK comments are most  welcome.

Henry



On 9/4/09 7:09 AM, "Simo.Veikkolainen@nokia.com" <Simo.Veikkolainen@nokia.c=
om>  wrote:


Hi,

We would like to propose a new DISPATCH topic on  how SIP and XMPP can be u=
sed together in a complementary  fashion.

We have been working on a solution which combines SIP  based VoIP and XMPP =
based IM and Presence. The requirements and a proposed  solution is outline=
d in http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01  <ht=
tp://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01>  .

The motivation for this work comes from experience which  shows that most s=
tandards based VoIP deployments use SIP. At the same time,  the Extensible =
Messaging and Presence Protocol (XMPP) is widely used in  instant messaging=
 and presence services. An interesting scenario arises when  a SIP based vo=
ice (and video) service is combined together with an XMPP  based instant me=
ssaging and presence service.

Note that the  goal in this work is not SIP-XMPP protocol translation, but =
to define  protocol extensions and best practises such that SIP based VoIP =
and XMPP  based IM and presence could be seamlessly combined and offered to=
 the end  user. For rapid deployment, we assume no changes in the server  i=
nfrastructure, and instead impose new requirements and protocol extensions =
 only to the endpoints.

We would like to get some discussion  going on the use case itself as well =
as on the solution. Also, we would like  to hear you thoughts on DISPATCH o=
r a new "dispatched" mini-WG as the home  for such work.

Exact charter proposal and problem statemen is  to follow.

Regards,
Simo and  Markus





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<HEAD>
<TITLE>Re: [dispatch] New topic proposal for DISPATCH</TITLE>
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<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
18pt'>Hi Markus,<BR>
<BR>
Thanks for your thoughtful reply and I agree there are some scenarios where=
 it makes sense.<BR>
More important, doing this in the endpoints is the right approach IMO and a=
s you know, I will always defend the Internet end-to-end principle :-)<BR>
<BR>
This distributed approach is also IMO a good answer to the Interdomain Scal=
ing problem for IM <BR>
( I-D.draft-ietf-simple-interdomain-scaling-analysis) and I have copied one=
 of its authors here.<BR>
<BR>
This raises the questions: <BR>
<BR>
</SPAN></FONT><OL><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SP=
AN STYLE=3D'font-size:18pt'>Why stop at SIP and XMPP, and not add other pop=
ular IM protocols, as is already done by quite a few products in the market=
?
</SPAN></FONT><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN S=
TYLE=3D'font-size:18pt'>Can your approach be generalized in a more generic =
way?<BR>
</SPAN></FONT></OL><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN =
STYLE=3D'font-size:18pt'><BR>
If endpoints are accommodating several IM protocols, an IETF, Dispatch reco=
mmended GENERIC approach would make many people feel more comfortable about=
 it. Your I-D could be used as a tested example for a generic approach.<BR>
<BR>
Ranting on preferring endpoint applications to more application protocols<B=
R>
<BR>
I still believe creative application developers face the choice of either i=
nventing new applications or dedicating their time to learning application =
protocols and they cannot do both. Learning and following both SIP and XMPP=
 to write and update code is quite a challenge. <BR>
Is there a better way? <BR>
For this reason, one single application protocol for IM and voice is enough=
.<BR>
Proof: Almost all the applications that make people buy smart phones, use o=
nline office apps and enterprise apps run on HTTP(S) and their application =
developers don&#8217;t have to worry about knowing how HTTP works. As a res=
ult, HTTP has been also used to even carry streaming media, web conferencin=
g and IM. Just to avoid spending a lifetime to learn and follow application=
 protocols. But HTTP based apps dominate the application space. This is how=
ever for another discussion elsewhere. <BR>
<BR>
My strictly personal 2 cents.<BR>
<BR>
Thanks again,<BR>
<BR>
Henry <BR>
<BR>
<BR>
On 9/11/09 8:11 AM, &quot;<a href=3D"Markus.Isomaki@nokia.com">Markus.Isoma=
ki@nokia.com</a>&quot; &lt;<a href=3D"Markus.Isomaki@nokia.com">Markus.Isom=
aki@nokia.com</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT COLOR=3D"#00=
00FF"><FONT FACE=3D"Arial">Hi Henry,<BR>
</FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"> <BR>
</FONT><FONT COLOR=3D"#0000FF"><FONT FACE=3D"Arial">Thanks for your feedbac=
k :) Let me answer to the excellent points you raise.<BR>
</FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"> <BR>
</FONT><FONT COLOR=3D"#0000FF"><FONT FACE=3D"Arial">First of all, I would e=
ncourage you and everyone to carefully read the Introduction chapter of the=
 draft (<a href=3D"http://tools.ietf.org/html/draft-veikkolainen-sip-voip-x=
mpp-im-01">http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-0=
1</a> &lt;<a href=3D"http://tools.ietf.org/html/draft-veikkolainen-sip-voip=
-xmpp-im-01">http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im=
-01</a>&gt; ) to be clear on what the proposal is about. That is, we are no=
t trying to boil the ocean and define all possible ways how SIP and XMPP in=
frastructures could be combined, but instead try to define a minimalistic a=
pproach whereby a client can use both SIP and XMPP in an integrated manner =
WITHOUT requiring ANY changes on the servers that are already deployed.<BR>
</FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"> <BR>
</FONT><FONT COLOR=3D"#0000FF"><FONT FACE=3D"Arial">See further comments be=
low with [Markus]:<BR>
</FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><BR>
</FONT></SPAN><BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Cali=
bri, Verdana, Helvetica, Arial"> <BR>
&nbsp;<BR>
<HR ALIGN=3DCENTER SIZE=3D"3" WIDTH=3D"100%"> </FONT><FONT FACE=3D"Tahoma, =
Verdana, Helvetica, Arial"><B>From:</B> ext Henry Sinnreich &nbsp;[<a href=
=3D"mailto:hsinnrei@adobe.com">mailto:hsinnrei@adobe.com</a>] <BR>
<B>Sent:</B> 04 September, 2009 &nbsp;18:42<BR>
<B>To:</B> Veikkolainen Simo (Nokia-D/Helsinki); <a href=3D"dispatch@ietf.o=
rg">dispatch@ietf.org</a>; &nbsp;<a href=3D"mary.barnes@nortel.com">mary.ba=
rnes@nortel.com</a>; <a href=3D"gonzalo.camarillo@ericsson.com">gonzalo.cam=
arillo@ericsson.com</a><BR>
<B>Cc:</B> Isomaki &nbsp;Markus (Nokia-CIC/Espoo); Olle E. Johansson<BR>
<B>Subject:</B> Re: [dispatch] &nbsp;New topic proposal for DISPATCH<BR>
</FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><BR>
&nbsp;<BR>
</FONT></SPAN><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><FONT SIZE=
=3D"2"><SPAN STYLE=3D'font-size:14pt'>I would like to second such work, but=
 a far deeper &nbsp;look under the hood is necessary, so we don&#8217;t do =
another Elbonia &nbsp;project.<BR>
<a href=3D"http://dilbert.com/strips/comic/2009-09-01/">http://dilbert.com/=
strips/comic/2009-09-01/</a><BR>
<BR>
Here &nbsp;are some not very mature thoughts on the scope of the work. <BR>
The most &nbsp;interesting, ROI estimate and REST architecture support are =
at the &nbsp;end.<BR>
<BR>
Servers<BR>
<BR>
Fundamentally, we need a registrar, a proxy and a &nbsp;presence server: Th=
ree in total<BR>
</SPAN></FONT><SPAN STYLE=3D'font-size:18pt'> <BR>
</SPAN></FONT><UL><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><FO=
NT SIZE=3D"2"><SPAN STYLE=3D'font-size:14pt'>What will be the duplication o=
f servers and where &nbsp;specifically? </SPAN></FONT><SPAN STYLE=3D'font-s=
ize:18pt'>=20
</SPAN></FONT><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><FONT S=
IZE=3D"2"><SPAN STYLE=3D'font-size:14pt'>How does the SIP registrar communi=
cate with the XMPP &nbsp;server(s)? </SPAN></FONT><SPAN STYLE=3D'font-size:=
18pt'>=20
</SPAN></FONT><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><FONT S=
IZE=3D"2"><SPAN STYLE=3D'font-size:14pt'>What may be the routing protocol(s=
) inside a &nbsp;SIP+XMMP network? Think of IMS as a stressing use &nbsp;ca=
se.<BR>
</SPAN></FONT></FONT></UL><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:18pt'> <BR>
</SPAN></FONT><SPAN STYLE=3D'font-size:18pt'><FONT COLOR=3D"#0000FF"><FONT =
FACE=3D"Arial">[Markus] Our starting point is that we already have quite &n=
bsp;many SIP VoIP deployments out there, as long as quite many XMPP presenc=
e &amp; &nbsp;IM deployments. So, the servers you mention already exist. If=
 one now &nbsp;wants to develop a client that does VoIP, IM and presence to=
gether, it should &nbsp;be possible to utilize these existing infrastrcutur=
es and not have to wait for &nbsp;new ones to be deployed (as for instance =
we have waited quite many years for &nbsp;SIMPLE-based Presence servers to =
get deployed but compared to XMPP there's &nbsp;very little in use). The id=
ea is to define a couple of E2E (as in &nbsp;nothing that SIP proxies or XM=
PP servers would have to support) protocol &nbsp;extensions which would all=
ow the client to nicely combine SIP VoIP and XMPP &nbsp;presence &amp; IM. =
<BR>
</FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"> <BR>
&nbsp;<BR>
&nbsp;<BR>
</FONT><FONT COLOR=3D"#0000FF"><FONT FACE=3D"Arial">There is no need for SI=
P and XMPP servers to know about &nbsp;each other or route protocol message=
s between each other. All the logic is put &nbsp;into the endpoints. No nee=
d to worry about IMS either, except &nbsp;that in theory the client can use=
 IMS for its VoIP service &nbsp;(similar to any other SIP &nbsp;infra).</FO=
NT></FONT></SPAN><FONT SIZE=3D"2"><FONT FACE=3D"Calibri, Verdana, Helvetica=
, Arial"><SPAN STYLE=3D'font-size:14pt'> <BR>
</SPAN></FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPA=
N STYLE=3D'font-size:18pt'> <BR>
</SPAN><FONT SIZE=3D"2"><SPAN STYLE=3D'font-size:14pt'> <BR>
User &nbsp;Agents<BR>
</SPAN></FONT><SPAN STYLE=3D'font-size:18pt'> <BR>
</SPAN></FONT><UL><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><FO=
NT SIZE=3D"2"><SPAN STYLE=3D'font-size:14pt'>Same as with servers: How many=
 and what protocol &nbsp;RFCs are there in the combination? </SPAN></FONT><=
SPAN STYLE=3D'font-size:18pt'>=20
</SPAN></FONT><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><FONT S=
IZE=3D"2"><SPAN STYLE=3D'font-size:14pt'>Which of the &#8220;services&#8221=
; can be placed in the &nbsp;applications in the endpoints or in the networ=
k application protocols, such &nbsp;as SIP and XMPP. This is the topic on i=
nfrastructure vs. endpoint &nbsp;applications.<BR>
</SPAN></FONT></FONT></UL><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:18pt'> <BR>
</SPAN></FONT><SPAN STYLE=3D'font-size:18pt'><FONT COLOR=3D"#0000FF"><FONT =
FACE=3D"Arial">[Markus] Just take any current SIP VoIP &nbsp;UA implementat=
ion, XMPP client implementation and glue them together &nbsp;and you're alr=
eady pretty far. Add a couple of extensions as proposed and it's &nbsp;done=
. From there onwards you have a pretty powerful machinery to add &nbsp;appl=
ications by using SIP session setup and XMPP message delivery &nbsp;capabil=
ities. <BR>
</FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"> <BR>
</FONT></SPAN><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><FONT SIZE=
=3D"2"><SPAN STYLE=3D'font-size:14pt'> <BR>
Application Protocol &nbsp;Stacks<BR>
</SPAN></FONT><SPAN STYLE=3D'font-size:18pt'> <BR>
</SPAN></FONT><UL><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><FO=
NT SIZE=3D"2"><SPAN STYLE=3D'font-size:14pt'>How many protocol stacks does =
this involve and more &nbsp;to the point, how many RFCs? </SPAN></FONT><SPA=
N STYLE=3D'font-size:18pt'>=20
</SPAN></FONT><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><FONT S=
IZE=3D"2"><SPAN STYLE=3D'font-size:14pt'>For SIP alone, see <a href=3D"http=
://rfc3261.net/">http://rfc3261.net/</a>. What will be the total &nbsp;comb=
ined RFC count?<BR>
</SPAN></FONT></FONT></UL><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:18pt'> <BR>
</SPAN></FONT><SPAN STYLE=3D'font-size:18pt'><FONT COLOR=3D"#0000FF"><FONT =
FACE=3D"Arial">[Markus] All what we expect is already pretty widely &nbsp;s=
upported and even available as open source: basic SIP VoIP and XMPP IM and =
&nbsp;presence is enough to start with. &nbsp;&nbsp;<BR>
</FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"> <BR>
</FONT></SPAN><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><FONT SIZE=
=3D"2"><SPAN STYLE=3D'font-size:14pt'> <BR>
NAT &nbsp;Traversal<BR>
</SPAN></FONT><SPAN STYLE=3D'font-size:18pt'> <BR>
</SPAN></FONT><UL><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><FO=
NT SIZE=3D"2"><SPAN STYLE=3D'font-size:14pt'>Will STUN/TURN/ICE work the sa=
me for SIP and for &nbsp;XMPP?<BR>
</SPAN></FONT></FONT></UL><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:18pt'> <BR>
</SPAN><FONT SIZE=3D"2"><SPAN STYLE=3D'font-size:14pt'> </SPAN></FONT></FON=
T><FONT COLOR=3D"#0000FF"><FONT FACE=3D"Arial"><SPAN STYLE=3D'font-size:18p=
t'>[Markus] In our proposal SIP would be used for setting up &nbsp;voice &a=
mp; video sessions. So, you would need NAT traversal for that. How &nbsp;th=
at's done is outside the scope but either B2BUA or ICE based mechanisms &nb=
sp;would work as far as they work in general. For XMPP we would not need to=
 worry &nbsp;about NAT traversal beyond how its done in that protocol by de=
fault. For a &nbsp;client there is a caveat that it has to keepalive the TC=
P connections with &nbsp;BOTH SIP and XMPP servers so it would be recommend=
ed to synchronize those &nbsp;(send them at the same time) for instance in =
a battery powered &nbsp;device.<BR>
</SPAN></FONT></FONT><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Calibri, =
Verdana, Helvetica, Arial"> <BR>
</FONT></SPAN><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><FONT SIZE=
=3D"2"><SPAN STYLE=3D'font-size:14pt'> <BR>
Return &nbsp;on Investment (ROI): Cost and effort by developers. More netwo=
rk &nbsp;infrastructure, not even counting B2B NEs.<BR>
</SPAN></FONT><SPAN STYLE=3D'font-size:18pt'> <BR>
</SPAN></FONT><UL><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><FO=
NT SIZE=3D"2"><SPAN STYLE=3D'font-size:14pt'>What is the additional time/mo=
ney required for &nbsp;developers that know SIP to also learn XMPP and vice=
 versa? This is all &nbsp;about return on investment.<BR>
</SPAN></FONT></FONT></UL><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:18pt'> <BR>
</SPAN></FONT><SPAN STYLE=3D'font-size:18pt'><FONT COLOR=3D"#0000FF"><FONT =
FACE=3D"Arial">[Markus] This is a problem for sure, even more so &nbsp;for =
the people writing standard specs than code :) &nbsp;Hopefully the &nbsp;de=
veloper would just have add a couple of simple extensions to existing &nbsp=
;SIP and XMPP implementations. <BR>
</FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"> <BR>
</FONT></SPAN><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><FONT SIZE=
=3D"2"><SPAN STYLE=3D'font-size:14pt'> <BR>
Support for a REST-full &nbsp;architecture<BR>
<BR>
The WWW has a REST architecture was formulated in 2000, &nbsp;after the eme=
rgence of SIP and XMMP. &nbsp;As a result of its architecture, &nbsp;the WW=
W has had a transformational effect on the following (this is not a &nbsp;c=
omplete list). Some SIP work recommends already recommends REST, such as in=
 &nbsp;RFC 4240 and in <a href=3D"http://tools.ietf.org/html/draft-griffin-=
bliss-rest-00">http://tools.ietf.org/html/draft-griffin-bliss-rest-00</a><B=
R>
</SPAN></FONT><SPAN STYLE=3D'font-size:18pt'> <BR>
</SPAN></FONT><UL><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><FO=
NT SIZE=3D"2"><SPAN STYLE=3D'font-size:14pt'>Web applications from mail to =
office apps &nbsp;</SPAN></FONT><SPAN STYLE=3D'font-size:18pt'>=20
</SPAN></FONT><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><FONT S=
IZE=3D"2"><SPAN STYLE=3D'font-size:14pt'>Internet applications previously s=
erved by dedicated &nbsp;network applications protocols </SPAN></FONT><SPAN=
 STYLE=3D'font-size:18pt'>=20
</SPAN></FONT><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><FONT S=
IZE=3D"2"><SPAN STYLE=3D'font-size:14pt'>Many other industries, from bankin=
g to newsprint to &nbsp;travel. And Oh, social networks, they also have &nb=
sp;registrars...<BR>
</SPAN></FONT></FONT></UL><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><FONT SIZE=3D"2"><SPAN STYLE=3D'font-size:14pt'> <BR>
<BR>
Now please keep in mind the WWW architecture uses basically &nbsp;only 3 co=
mponents*: The URI for addressing, HTTP for transport and HTML for &nbsp;da=
ta rendering, augmented by some other languages (XML) and namespaces. Can w=
e &nbsp;benefit something from the WWW architecture? If yes what &nbsp;spec=
ifically?<BR>
<BR>
Heresy: If the web developers could use HTTP only, what &nbsp;other applica=
tion level protocols do we really need except RTP/UDP?<BR>
This &nbsp;includes in my mind TLS and DTLS for secure transport or even be=
tter IMO: HIP &nbsp;for many reasons.<BR>
</SPAN></FONT></FONT><FONT COLOR=3D"#0000FF"><FONT FACE=3D"Arial"><SPAN STY=
LE=3D'font-size:18pt'> <BR>
</SPAN></FONT></FONT><FONT SIZE=3D"2"><FONT FACE=3D"Calibri, Verdana, Helve=
tica, Arial"><SPAN STYLE=3D'font-size:14pt'> <BR>
</SPAN></FONT></FONT><FONT COLOR=3D"#0000FF"><FONT FACE=3D"Arial"><SPAN STY=
LE=3D'font-size:18pt'>[Markus] Personally I agree that REST+HTTP+Javascript=
 is a superior &nbsp;toolset compared to SIP, XMPP, IMAP and so on to devel=
op many applications. &nbsp;However, there are still plenty of SIP based Vo=
IP and XMPP based IM and &nbsp;presence deployments out there and I don't e=
xpect the web apps to replace all &nbsp;of them in the near future :)<BR>
</SPAN></FONT></FONT><FONT SIZE=3D"2"><FONT FACE=3D"Calibri, Verdana, Helve=
tica, Arial"><SPAN STYLE=3D'font-size:14pt'> <BR>
&nbsp;<BR>
* T.V. Raman: &#8220;Toward 2W, &nbsp;Beyond Web 2.0&#8221;, Communications=
 of the ACM, February 2009<BR>
<BR>
Many or most &nbsp;of these points may be moot or make no sense, but maybe =
should be cleared up, &nbsp;in case there are more naive people like me tha=
t may be interested.<BR>
<BR>
My &nbsp;strict personal 2 cents.<BR>
<BR>
FRANK comments are most &nbsp;welcome.<BR>
<BR>
Henry<BR>
</SPAN></FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPA=
N STYLE=3D'font-size:18pt'><BR>
<BR>
<BR>
On 9/4/09 7:09 AM, &quot;<a href=3D"Simo.Veikkolainen@nokia.com">Simo.Veikk=
olainen@nokia.com</a>&quot; &lt;<a href=3D"Simo.Veikkolainen@nokia.com">Sim=
o.Veikkolainen@nokia.com</a>&gt; &nbsp;wrote:<BR>
<BR>
&nbsp;<BR>
</SPAN></FONT><BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FONT FACE=3D"Aria=
l">Hi,<BR>
&nbsp;<BR>
We would like to propose a new DISPATCH topic on &nbsp;how SIP and XMPP can=
 be used together in a complementary &nbsp;fashion.<BR>
&nbsp;<BR>
We have been working on a solution which combines SIP &nbsp;based VoIP and =
XMPP based IM and Presence. The requirements and a proposed &nbsp;solution =
is outlined in <FONT COLOR=3D"#0000FF"><U><a href=3D"http://tools.ietf.org/=
html/draft-veikkolainen-sip-voip-xmpp-im-01">http://tools.ietf.org/html/dra=
ft-veikkolainen-sip-voip-xmpp-im-01</a></U></FONT> &nbsp;&lt;<a href=3D"htt=
p://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01">http://tool=
s.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01</a>&gt; &nbsp;.<BR>
&nbsp;<BR>
The motivation for this work comes from experience which &nbsp;shows that m=
ost standards based VoIP deployments use SIP. At the same time, &nbsp;the E=
xtensible Messaging and Presence Protocol (XMPP) is widely used in &nbsp;in=
stant messaging and presence services. An interesting scenario arises when =
&nbsp;a SIP based voice (and video) service is combined together with an XM=
PP &nbsp;based instant messaging and presence service. <BR>
&nbsp;<BR>
Note that the &nbsp;goal in this work is not SIP-XMPP protocol translation,=
 but to define &nbsp;protocol extensions and best practises such that SIP b=
ased VoIP and XMPP &nbsp;based IM and presence could be seamlessly combined=
 and offered to the end &nbsp;user. For rapid deployment, we assume no chan=
ges in the server &nbsp;infrastructure, and instead impose new requirements=
 and protocol extensions &nbsp;only to the endpoints.<BR>
&nbsp;<BR>
We would like to get some discussion &nbsp;going on the use case itself as =
well as on the solution. Also, we would like &nbsp;to hear you thoughts on =
DISPATCH or a new &quot;dispatched&quot; mini-WG as the home &nbsp;for such=
 work.<BR>
&nbsp;<BR>
Exact charter proposal and problem statemen is &nbsp;to follow. <BR>
&nbsp;<BR>
Regards,<BR>
Simo and &nbsp;Markus<BR>
&nbsp;<BR>
&nbsp;<BR>
</FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><BR>
</FONT></SPAN></BLOCKQUOTE></BLOCKQUOTE><SPAN STYLE=3D'font-size:18pt'><FON=
T FACE=3D"Calibri, Verdana, Helvetica, Arial"><BR>
</FONT></SPAN></BLOCKQUOTE>
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Subject: Re: [dispatch] New topic proposal for DISPATCH
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I agree that there are pragmatic reasons to pursue the 
coordination/coexistence of sip and xmpp. And pushing responsibility to 
the endpoints remains desirable.

One thing that remains of concern to me in such work is that we end up 
with disjoint address spaces without any straightforward way to 
correlate them. So if I want a single "session" involving both voice via 
SIP and IM via xmpp, then I have great difficulty in discovering URIs 
for the "subordinate" session that correlate with those for the 
"primary" session. (No pejorative intended here - one or the other must 
go first.)

Somehow this problem needs to be resolved.

	Thanks,
	Paul

Henry Sinnreich wrote:
> Hi Markus,
> 
> Thanks for your thoughtful reply and I agree there are some scenarios 
> where it makes sense.
> More important, doing this in the endpoints is the right approach IMO 
> and as you know, I will always defend the Internet end-to-end principle :-)
> 
> This distributed approach is also IMO a good answer to the Interdomain 
> Scaling problem for IM
> ( I-D.draft-ietf-simple-interdomain-scaling-analysis) and I have copied 
> one of its authors here.
> 
> This raises the questions:
> 
>    1. Why stop at SIP and XMPP, and not add other popular IM protocols,
>       as is already done by quite a few products in the market?
>    2. Can your approach be generalized in a more generic way?
> 
> 
> If endpoints are accommodating several IM protocols, an IETF, Dispatch 
> recommended GENERIC approach would make many people feel more 
> comfortable about it. Your I-D could be used as a tested example for a 
> generic approach.
> 
> Ranting on preferring endpoint applications to more application protocols
> 
> I still believe creative application developers face the choice of 
> either inventing new applications or dedicating their time to learning 
> application protocols and they cannot do both. Learning and following 
> both SIP and XMPP to write and update code is quite a challenge.
> Is there a better way?
> For this reason, one single application protocol for IM and voice is enough.
> Proof: Almost all the applications that make people buy smart phones, 
> use online office apps and enterprise apps run on HTTP(S) and their 
> application developers don’t have to worry about knowing how HTTP works. 
> As a result, HTTP has been also used to even carry streaming media, web 
> conferencing and IM. Just to avoid spending a lifetime to learn and 
> follow application protocols. But HTTP based apps dominate the 
> application space. This is however for another discussion elsewhere.
> 
> My strictly personal 2 cents.
> 
> Thanks again,
> 
> Henry
> 
> 
> On 9/11/09 8:11 AM, "Markus.Isomaki@nokia.com" 
> <Markus.Isomaki@nokia.com> wrote:
> 
>     Hi Henry,
> 
>     Thanks for your feedback :) Let me answer to the excellent points
>     you raise.
> 
>     First of all, I would encourage you and everyone to carefully read
>     the Introduction chapter of the draft
>     (http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01
>     <http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01>
>     ) to be clear on what the proposal is about. That is, we are not
>     trying to boil the ocean and define all possible ways how SIP and
>     XMPP infrastructures could be combined, but instead try to define a
>     minimalistic approach whereby a client can use both SIP and XMPP in
>     an integrated manner WITHOUT requiring ANY changes on the servers
>     that are already deployed.
> 
>     See further comments below with [Markus]:
> 
> 
>          
>         ------------------------------------------------------------------------
>         *From:* ext Henry Sinnreich  [mailto:hsinnrei@adobe.com]
>         *Sent:* 04 September, 2009  18:42
>         *To:* Veikkolainen Simo (Nokia-D/Helsinki); dispatch@ietf.org;
>          mary.barnes@nortel.com; gonzalo.camarillo@ericsson.com
>         *Cc:* Isomaki  Markus (Nokia-CIC/Espoo); Olle E. Johansson
>         *Subject:* Re: [dispatch]  New topic proposal for DISPATCH
> 
>          
>         I would like to second such work, but a far deeper  look under
>         the hood is necessary, so we don’t do another Elbonia  project.
>         http://dilbert.com/strips/comic/2009-09-01/
> 
>         Here  are some not very mature thoughts on the scope of the work.
>         The most  interesting, ROI estimate and REST architecture
>         support are at the  end.
> 
>         Servers
> 
>         Fundamentally, we need a registrar, a proxy and a  presence
>         server: Three in total
> 
>             * What will be the duplication of servers and where
>                specifically?
>             * How does the SIP registrar communicate with the XMPP
>                server(s)?
>             * What may be the routing protocol(s) inside a  SIP+XMMP
>               network? Think of IMS as a stressing use  case.
> 
> 
>         [Markus] Our starting point is that we already have quite  many
>         SIP VoIP deployments out there, as long as quite many XMPP
>         presence &  IM deployments. So, the servers you mention already
>         exist. If one now  wants to develop a client that does VoIP, IM
>         and presence together, it should  be possible to utilize these
>         existing infrastrcutures and not have to wait for  new ones to
>         be deployed (as for instance we have waited quite many years for
>          SIMPLE-based Presence servers to get deployed but compared to
>         XMPP there's  very little in use). The idea is to define a
>         couple of E2E (as in  nothing that SIP proxies or XMPP servers
>         would have to support) protocol  extensions which would allow
>         the client to nicely combine SIP VoIP and XMPP  presence & IM.
> 
>          
>          
>         There is no need for SIP and XMPP servers to know about  each
>         other or route protocol messages between each other. All the
>         logic is put  into the endpoints. No need to worry about IMS
>         either, except  that in theory the client can use IMS for its
>         VoIP service  (similar to any other SIP  infra).
> 
> 
>         User  Agents
> 
>             * Same as with servers: How many and what protocol  RFCs are
>               there in the combination?
>             * Which of the “services” can be placed in the  applications
>               in the endpoints or in the network application protocols,
>               such  as SIP and XMPP. This is the topic on infrastructure
>               vs. endpoint  applications.
> 
> 
>         [Markus] Just take any current SIP VoIP  UA implementation, XMPP
>         client implementation and glue them together  and you're already
>         pretty far. Add a couple of extensions as proposed and it's
>          done. From there onwards you have a pretty powerful machinery
>         to add  applications by using SIP session setup and XMPP message
>         delivery  capabilities.
> 
> 
>         Application Protocol  Stacks
> 
>             * How many protocol stacks does this involve and more  to
>               the point, how many RFCs?
>             * For SIP alone, see http://rfc3261.net/. What will be the
>               total  combined RFC count?
> 
> 
>         [Markus] All what we expect is already pretty widely  supported
>         and even available as open source: basic SIP VoIP and XMPP IM
>         and  presence is enough to start with.   
> 
> 
>         NAT  Traversal
> 
>             * Will STUN/TURN/ICE work the same for SIP and for  XMPP?
> 
> 
>         [Markus] In our proposal SIP would be used for setting up  voice
>         & video sessions. So, you would need NAT traversal for that. How
>          that's done is outside the scope but either B2BUA or ICE based
>         mechanisms  would work as far as they work in general. For XMPP
>         we would not need to worry  about NAT traversal beyond how its
>         done in that protocol by default. For a  client there is a
>         caveat that it has to keepalive the TCP connections with  BOTH
>         SIP and XMPP servers so it would be recommended to synchronize
>         those  (send them at the same time) for instance in a battery
>         powered  device.
> 
> 
>         Return  on Investment (ROI): Cost and effort by developers. More
>         network  infrastructure, not even counting B2B NEs.
> 
>             * What is the additional time/money required for  developers
>               that know SIP to also learn XMPP and vice versa? This is
>               all  about return on investment.
> 
> 
>         [Markus] This is a problem for sure, even more so  for the
>         people writing standard specs than code :)  Hopefully the
>          developer would just have add a couple of simple extensions to
>         existing  SIP and XMPP implementations.
> 
> 
>         Support for a REST-full  architecture
> 
>         The WWW has a REST architecture was formulated in 2000,  after
>         the emergence of SIP and XMMP.  As a result of its architecture,
>          the WWW has had a transformational effect on the following
>         (this is not a  complete list). Some SIP work recommends already
>         recommends REST, such as in  RFC 4240 and in
>         http://tools.ietf.org/html/draft-griffin-bliss-rest-00
> 
>             * Web applications from mail to office apps  
>             * Internet applications previously served by dedicated
>                network applications protocols
>             * Many other industries, from banking to newsprint to
>                travel. And Oh, social networks, they also have
>                registrars...
> 
> 
> 
>         Now please keep in mind the WWW architecture uses basically
>          only 3 components*: The URI for addressing, HTTP for transport
>         and HTML for  data rendering, augmented by some other languages
>         (XML) and namespaces. Can we  benefit something from the WWW
>         architecture? If yes what  specifically?
> 
>         Heresy: If the web developers could use HTTP only, what  other
>         application level protocols do we really need except RTP/UDP?
>         This  includes in my mind TLS and DTLS for secure transport or
>         even better IMO: HIP  for many reasons.
> 
> 
>         [Markus] Personally I agree that REST+HTTP+Javascript is a
>         superior  toolset compared to SIP, XMPP, IMAP and so on to
>         develop many applications.  However, there are still plenty of
>         SIP based VoIP and XMPP based IM and  presence deployments out
>         there and I don't expect the web apps to replace all  of them in
>         the near future :)
> 
>          
>         * T.V. Raman: “Toward 2W,  Beyond Web 2.0”, Communications of
>         the ACM, February 2009
> 
>         Many or most  of these points may be moot or make no sense, but
>         maybe should be cleared up,  in case there are more naive people
>         like me that may be interested.
> 
>         My  strict personal 2 cents.
> 
>         FRANK comments are most  welcome.
> 
>         Henry
> 
> 
> 
>         On 9/4/09 7:09 AM, "Simo.Veikkolainen@nokia.com"
>         <Simo.Veikkolainen@nokia.com>  wrote:
> 
>          
> 
>             Hi,
>              
>             We would like to propose a new DISPATCH topic on  how SIP
>             and XMPP can be used together in a complementary  fashion.
>              
>             We have been working on a solution which combines SIP  based
>             VoIP and XMPP based IM and Presence. The requirements and a
>             proposed  solution is outlined in
>             _http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01_
>              <http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01>
>              .
>              
>             The motivation for this work comes from experience which
>              shows that most standards based VoIP deployments use SIP.
>             At the same time,  the Extensible Messaging and Presence
>             Protocol (XMPP) is widely used in  instant messaging and
>             presence services. An interesting scenario arises when  a
>             SIP based voice (and video) service is combined together
>             with an XMPP  based instant messaging and presence service.
>              
>             Note that the  goal in this work is not SIP-XMPP protocol
>             translation, but to define  protocol extensions and best
>             practises such that SIP based VoIP and XMPP  based IM and
>             presence could be seamlessly combined and offered to the end
>              user. For rapid deployment, we assume no changes in the
>             server  infrastructure, and instead impose new requirements
>             and protocol extensions  only to the endpoints.
>              
>             We would like to get some discussion  going on the use case
>             itself as well as on the solution. Also, we would like  to
>             hear you thoughts on DISPATCH or a new "dispatched" mini-WG
>             as the home  for such work.
>              
>             Exact charter proposal and problem statemen is  to follow.
>              
>             Regards,
>             Simo and  Markus
>              
>              
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

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Subject: [dispatch] SIP, IPv6 and IPv4-IPv6 interworking (was RE: draft-boucadair-dispatch-ipv6-atypes)
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=20
Dear Mary, all,

We are definitely aware that unlike sipping ML (http://www.ietf.org/mail-ar=
chive/web/sipping/current/msg16718.html) this draft hasn't been discussed i=
n the DIPATCH mailing list.=20

The topic we are proposing is not restricted to the atypes attribute but co=
vers also issues related to the migration of SIP services to IPv6 and IPv4-=
IPv6 interworking. This is a timely valid topic since IPv4 address exhausti=
on is a real concern. Therefore, the impact on SIP-based services need to b=
e identified and solution proposed.=20

Cheers,
Med


-----Message d'origine-----
De : Mary Barnes [mailto:mary.barnes@nortel.com]=20
Envoy=E9 : lundi 7 septembre 2009 17:48
=C0 : Andrew Allen; Gonzalo.Camarillo@ericsson.com; dispatch@ietf.org
Cc : BOUCADAIR Mohamed NCPI/NAD/TIP
Objet : RE: draft-boucadair-dispatch-ipv6-atypes

Hi Andrew,

At this point in time, we're not handling agenda requests per se. Right now=
, we want to know the topics for which folks want feedback PRIOR to
IETF-76 so that we can use the agenda time to discuss issues that can't be =
resolved on the mailing list before then. We can take this as a placeholder=
 for this topic and thus, we would expect to see a detailed problem stateme=
nt, etc. on or before Sept. 21st., per the IETF-76 DISPATCH deadlines:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00500.html

Just a note that when I summarized the status of the post-IETF-75 topics, t=
his one had received no ML discussion nor indication of interest in working=
 on this problem:
http://www.ietf.org/mail-archive/web/dispatch/current/msg00501.html
So, you really need to get some WG feedback on this proposal if you want it=
 to move forward.=20=20

Thanks,
Mary.=20

-----Original Message-----
From: Andrew Allen [mailto:aallen@rim.com]
Sent: Monday, September 07, 2009 5:37 AM
To: Barnes, Mary (RICH2:AR00); Gonzalo.Camarillo@ericsson.com; dispatch@iet=
f.org
Cc: mohamed.boucadair@orange-ftgroup.com
Subject: draft-boucadair-dispatch-ipv6-atypes


Can we request agenda time in Dispatch to discuss how to proceed with
draft-boucadair-dispatch-ipv6-atypes?

Thanks
Andrew

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Thread-Topic: [dispatch] IETF-76 DISPATCH WG deadlines : CBUS
Thread-Index: Acoy6v86HwZmglQBSaK2ur/X8fpuPwCLowhZ
References: <1ECE0EB50388174790F9694F77522CCF1F9182AD@zrc2hxm0.corp.nortel.com>	<0D5F89FAC29E2C41B98A6A762007F5D0024E504F@GBNTHT12009MSX.gb002.siemens.net>	<019d01ca2d99$67f0a1f0$37d1e5d0$@us> <0F05CF0C-0956-4D32-BD4D-1DEB59DF13FB@softarmor.com> <4AAA5C98.1080502@sipstation.com>
From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: <dispatch@ietf.org>
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Cc: Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>
Subject: Re: [dispatch] IETF-76 DISPATCH WG deadlines : CBUS
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Hi,
=20
Some time ago I submitted a requirement draft for the OMA work on CBUS.
=20
In short, the idea of CBUS is that a client provides a set of conditions =
(which can be related to presence, location etc etc) to a CBUS server. =
Based on the conditions, the CBUS server communicates with presence =
servers, locations servers etc in order to figure out which users =
fulfill the conditions. The CBUS server then provides the client with a =
list of the URIs which fulfilled the conditions. The CBUS can do that on =
a one-time bases, a time-based basis, or whenever the information =
changes (called evaluation parameters). In addition, the client can =
provide a set or URIs, or a reference to URIs, which the CBUS chooses =
from (called candidate URIs).=20
=20
The idea is to provide the URIs in a subscription event package, and the =
associated SUBSCRIBE would contain the conditions (most likely using an =
XML document).
=20
Dean raised a question whether it would be enough to define the new =
event package, and re-use RFC4660/1 to provide the conditions.
=20
Due to summer vacations etc it took a while to look into this, but me =
and some OMA people have now looked into this.
=20
Our initial conclusion is that it is NOT very feasible to re-use RFC =
4660/1.=20
=20
RFC4660/1 provides a filtering mechanism, where the client specifies =
what type of information he wants to receive. However, in CBUS the type =
of information is always the same: the URI list. Instead, the CLIENT =
provides the type of information (conditions) on which the URIs shall be =
selected (and, in addition to that, the evaluation parameter(s) and =
candidate URI(s)).
=20
We did look at whether it would be possible to somehow re-use 4660/1, =
but we thought it would be rather hacky. Some of the 4660/1 XML elements =
would probably be unused, and there would be a need to define new =
extensions - so we thought it would be nicer to define a new XML schema =
instead.
=20
So, I would like to request DISPATCH time to discuss this. We plan to =
provide an updated version of the draft, with clarifications and more =
details etc.
=20
Regards,
=20
Christer

From L.Liess@telekom.de  Mon Sep 14 02:02:43 2009
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From: <L.Liess@telekom.de>
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Subject: [dispatch] Topic for DISPATCH: Alert-Info URNs
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Mary, Gonzalo,

At the IETF-75 we had a presentation and a discussion in BLISS on the =
"Alert-Info URNs" proposal, described in =
http://tools.ietf.org/id/draft-alexeitsev-bliss-alert-info-urns-02.txt.  =
During the discussion, it was recommended to submit the work to DISPATCH =
and to document the interop problems.=20

So, we would like to propose this as a new topic for DISPATCH.=20

The motivation for this work comes from interoperability problems when =
SIP services require the semantic for a ring- / ringback-tone to be =
transmitted in in a SIP-message. =20

The RFC 3261 allows an UAS or proxy to provide a specific ring- =
/ringback-tone as a reference (e.g. HTTP URI) which can be played to the =
user in the Alert-Info header. This mechanism does not ensure =
interoperability when there is no common understanding of the referenced =
content (different countries or vendors, hearing impaired) or when the =
user has his own tones configured in the end device.=20

For interoperability of services as call waiting, call forwarding, blind =
transfer, automatic callback, call hold or emergency annoncements sent =
over PBX systems, a mechanism is needed which allows the UAs and proxies =
to transmit an indication which contains the semantic of the rendering =
instead of the rendering itself and allows the receiving UA to decide =
about the concrete rendering.=20

The charter proposal and the updated draft are to follow.=20

Kind regards
Laura

From L.Liess@telekom.de  Mon Sep 14 02:17:23 2009
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Mary, Gonzalo,

At the IETF-75 we had a presentation and a discussion in BLISS on the
"Alert-Info URNs" proposal, described in
http://tools.ietf.org/id/draft-alexeitsev-bliss-alert-info-urns-02.txt.
During the discussion, it was recommended to submit the work to DISPATCH
and to document the interop problems.=20

So, we would like to propose this as a new topic for DISPATCH.=20

The motivation for this work comes from interoperability problems when
SIP services require the semantic for a ring- / ringback-tone to be
transmitted in in a SIP-message. =20

The RFC 3261 allows an UAS or proxy to provide a specific ring-
/ringback-tone as a reference (e.g. HTTP URI) which can be played to the
user in the Alert-Info header. This mechanism does not ensure
interoperability when there is no common understanding of the referenced
content (different countries or vendors, hearing impaired) or when the
user has his own tones configured in the end device.=20

For interoperability of services as call waiting, call forwarding, blind
transfer, automatic callback, call hold or emergency annoncements sent
over PBX systems, a mechanism is needed which allows the UAs and proxies
to transmit an indication which contains the semantic of the rendering
instead of the rendering itself and allows the receiving UA to decide
about the concrete rendering.=20

The charter proposal and the updated draft are to follow.=20

Kind regards
Laura



From R.Jesske@telekom.de  Mon Sep 14 04:09:42 2009
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Dear all,
A since a couple of weeks a new version of the Reason in Responses draft =
is now available.

http://tools.ietf.org/html/draft-jesske-dispatch-reason-in-responses-00

I have incorporated all comments and the discussion conclusions received =
for this draft.

Comments are welcome.

Best Regards

Roland

Deutsche Telekom Netzproduktion GmbH=20
Zentrum Technik Einf=FChrung=20
Roland Jesske=20
Gateways, Protokolle, Konvergenz, TE32-1
Heinrich-Hertz-Str. 3-7, 64295 Darmstadt,=20
Postfach, 64307 Darmstadt (Postanschrift)=20
+496151 628 2766 (Tel)
+491718618445 (Mobile)=20
E-Mail: r.jesske@telekom.de=20
http://www.telekom.de <http://www.telekom.de/> =20



Registerrechtliche Unternehmensangaben:
Deutsche Telekom Netzproduktion (DT NP) GmbH
Aufsichtsrat: Timotheus H=F6ttges (Vorsitzender)
Gesch=E4ftsf=FChrung: Dr. Bruno Jacobfeuerborn (Vorsitzender), Albert =
Matheis, Klaus Peren
Handelsregister: Amtsgericht Bonn HRB 14190
Sitz der Gesellschaft: Bonn
USt-IdNr.: DE 814645262



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<P><FONT SIZE=3D2 FACE=3D"Arial">Dear all,</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Arial">A since a couple of weeks a new =
version of the Reason in Responses draft is now available.</FONT>
</P>

<P><A =
HREF=3D"http://tools.ietf.org/html/draft-jesske-dispatch-reason-in-respon=
ses-00"><U><FONT COLOR=3D"#0000FF" SIZE=3D2 =
FACE=3D"Arial">http://tools.ietf.org/html/draft-jesske-dispatch-reason-in=
-responses-00</FONT></U></A>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">I have incorporated all comments and =
the discussion conclusions received for this draft.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Comments are welcome.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Best Regards</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Roland</FONT>
</P>

<P><FONT COLOR=3D"#000000" SIZE=3D2 FACE=3D"Arial">Deutsche Telekom =
Netzproduktion GmbH<BR>
Zentrum Technik Einf=FChrung</FONT><BR>
<FONT SIZE=3D2 FACE=3D"Arial">Roland Jesske</FONT><FONT FACE=3D"Times =
New Roman"><BR>
</FONT><FONT SIZE=3D2 FACE=3D"Arial">Gateways, Protokolle, Konvergenz, =
TE32-1</FONT><BR>
<FONT SIZE=3D2 FACE=3D"Arial">Heinrich-Hertz-Str. 3-7, 64295 =
Darmstadt,</FONT><BR>
<FONT SIZE=3D2 FACE=3D"Arial">Postfach, 64307 Darmstadt =
(Postanschrift)</FONT><FONT FACE=3D"Times New Roman"><BR>
</FONT><FONT SIZE=3D2 FACE=3D"Arial">+496151 628 2766 (Tel)</FONT><BR>
<FONT SIZE=3D2 FACE=3D"Arial">+491718618445 (Mobile)</FONT><FONT =
FACE=3D"Times New Roman"><BR>
</FONT><FONT SIZE=3D2 FACE=3D"Arial">E-Mail: </FONT><A =
HREF=3D"mailto:r.jesske@telekom.de"><U><FONT COLOR=3D"#000000" SIZE=3D2 =
FACE=3D"Arial">r.jesske@telekom.de</FONT></U></A><FONT SIZE=3D2 =
FACE=3D"Arial"></FONT>=20

<BR><A HREF=3D"http://www.telekom.de/"><U></U><U></U><U><FONT =
COLOR=3D"#0000FF" SIZE=3D2 =
FACE=3D"Arial">http://www.telekom.de</FONT></U></A><FONT =
COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial"></FONT>=20
</P>
<BR>
<BR>

<P><U><FONT SIZE=3D1 FACE=3D"Arial">Registerrechtliche =
Unternehmensangaben:</FONT></U><BR>
<FONT SIZE=3D1 FACE=3D"Arial">Deutsche Telekom Netzproduktion (DT NP) =
GmbH<BR>
Aufsichtsrat: Timotheus H=F6ttges (Vorsitzender)<BR>
Gesch=E4ftsf=FChrun</FONT><FONT COLOR=3D"#000000" SIZE=3D1 =
FACE=3D"Arial">g: Dr. Bruno Jacobfeuerborn (Vorsitzender), =
Al</FONT><FONT SIZE=3D1 FACE=3D"Arial">bert Matheis, Klaus Peren<BR>
Handelsregister: Amtsgericht Bonn HRB 14190<BR>
Sitz der Gesellschaft: Bonn<BR>
USt-IdNr.: DE 814645262</FONT>
</P>
<BR>

</BODY>
</HTML>
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In-Reply-To: <C6CFF5DE.5C9E%hsinnrei@adobe.com>
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To: Henry Sinnreich <hsinnrei@adobe.com>, Markus.Isomaki@nokia.com, Simo.Veikkolainen@nokia.com
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Cc: dispatch@ietf.org, gonzalo.camarillo@ericsson.com
Subject: Re: [dispatch] New topic proposal for DISPATCH
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I think that this draft is an important start for an important work item=20
that the IETF should do.

--Avshalom




From:
Henry Sinnreich <hsinnrei@adobe.com>
To:
"Markus.Isomaki@nokia.com" <Markus.Isomaki@nokia.com>,=20
"Simo.Veikkolainen@nokia.com" <Simo.Veikkolainen@nokia.com>,=20
"dispatch@ietf.org" <dispatch@ietf.org>, "mary.barnes@nortel.com"=20
<mary.barnes@nortel.com>, "gonzalo.camarillo@ericsson.com"=20
<gonzalo.camarillo@ericsson.com>
Cc:
Avshalom Houri/Haifa/IBM@IBMIL
Date:
11/09/2009 08:41 PM
Subject:
Re: [dispatch] New topic proposal for DISPATCH



Hi Markus,

Thanks for your thoughtful reply and I agree there are some scenarios=20
where it makes sense.
More important, doing this in the endpoints is the right approach IMO and=20
as you know, I will always defend the Internet end-to-end principle :-)

This distributed approach is also IMO a good answer to the Interdomain=20
Scaling problem for IM=20
( I-D.draft-ietf-simple-interdomain-scaling-analysis) and I have copied=20
one of its authors here.

This raises the questions:=20

1.      Why stop at SIP and XMPP, and not add other popular IM protocols,=20
as is already done by quite a few products in the market?=20
2.      Can your approach be generalized in a more generic way?

If endpoints are accommodating several IM protocols, an IETF, Dispatch=20
recommended GENERIC approach would make many people feel more comfortable=20
about it. Your I-D could be used as a tested example for a generic=20
approach.

Ranting on preferring endpoint applications to more application protocols

I still believe creative application developers face the choice of either=20
inventing new applications or dedicating their time to learning=20
application protocols and they cannot do both. Learning and following both =

SIP and XMPP to write and update code is quite a challenge.=20
Is there a better way?=20
For this reason, one single application protocol for IM and voice is=20
enough.
Proof: Almost all the applications that make people buy smart phones, use=20
online office apps and enterprise apps run on HTTP(S) and their=20
application developers don?t have to worry about knowing how HTTP works.=20
As a result, HTTP has been also used to even carry streaming media, web=20
conferencing and IM. Just to avoid spending a lifetime to learn and follow =

application protocols. But HTTP based apps dominate the application space. =

This is however for another discussion elsewhere.=20

My strictly personal 2 cents.

Thanks again,

Henry=20


On 9/11/09 8:11 AM, "Markus.Isomaki@nokia.com" <Markus.Isomaki@nokia.com>=20
wrote:

Hi Henry,

Thanks for your feedback :) Let me answer to the excellent points you=20
raise.

First of all, I would encourage you and everyone to carefully read the=20
Introduction chapter of the draft (
http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01 <
http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01> ) to be =

clear on what the proposal is about. That is, we are not trying to boil=20
the ocean and define all possible ways how SIP and XMPP infrastructures=20
could be combined, but instead try to define a minimalistic approach=20
whereby a client can use both SIP and XMPP in an integrated manner WITHOUT =

requiring ANY changes on the servers that are already deployed.

See further comments below with [Markus]:


=20
From: ext Henry Sinnreich  [mailto:hsinnrei@adobe.com]=20
Sent: 04 September, 2009  18:42
To: Veikkolainen Simo (Nokia-D/Helsinki); dispatch@ietf.org; =20
mary.barnes@nortel.com; gonzalo.camarillo@ericsson.com
Cc: Isomaki  Markus (Nokia-CIC/Espoo); Olle E. Johansson
Subject: Re: [dispatch]  New topic proposal for DISPATCH

=20
I would like to second such work, but a far deeper  look under the hood is =

necessary, so we don?t do another Elbonia  project.
http://dilbert.com/strips/comic/2009-09-01/

Here  are some not very mature thoughts on the scope of the work.=20
The most  interesting, ROI estimate and REST architecture support are at=20
the  end.

Servers

Fundamentally, we need a registrar, a proxy and a  presence server: Three=20
in total

What will be the duplication of servers and where  specifically?=20
How does the SIP registrar communicate with the XMPP  server(s)?=20
What may be the routing protocol(s) inside a  SIP+XMMP network? Think of=20
IMS as a stressing use  case.

[Markus] Our starting point is that we already have quite  many SIP VoIP=20
deployments out there, as long as quite many XMPP presence &  IM=20
deployments. So, the servers you mention already exist. If one now  wants=20
to develop a client that does VoIP, IM and presence together, it should be =

possible to utilize these existing infrastrcutures and not have to wait=20
for  new ones to be deployed (as for instance we have waited quite many=20
years for  SIMPLE-based Presence servers to get deployed but compared to=20
XMPP there's  very little in use). The idea is to define a couple of E2E=20
(as in  nothing that SIP proxies or XMPP servers would have to support)=20
protocol  extensions which would allow the client to nicely combine SIP=20
VoIP and XMPP  presence & IM.=20

=20
=20
There is no need for SIP and XMPP servers to know about  each other or=20
route protocol messages between each other. All the logic is put  into the =

endpoints. No need to worry about IMS either, except  that in theory the=20
client can use IMS for its VoIP service  (similar to any other SIP infra). =




User  Agents

Same as with servers: How many and what protocol  RFCs are there in the=20
combination?=20
Which of the ?services? can be placed in the  applications in the=20
endpoints or in the network application protocols, such  as SIP and XMPP.=20
This is the topic on infrastructure vs. endpoint  applications.

[Markus] Just take any current SIP VoIP  UA implementation, XMPP client=20
implementation and glue them together  and you're already pretty far. Add=20
a couple of extensions as proposed and it's  done. From there onwards you=20
have a pretty powerful machinery to add  applications by using SIP session =

setup and XMPP message delivery  capabilities.=20


Application Protocol  Stacks

How many protocol stacks does this involve and more  to the point, how=20
many RFCs?=20
For SIP alone, see http://rfc3261.net/. What will be the total  combined=20
RFC count?

[Markus] All what we expect is already pretty widely  supported and even=20
available as open source: basic SIP VoIP and XMPP IM and  presence is=20
enough to start with.  =20


NAT  Traversal

Will STUN/TURN/ICE work the same for SIP and for  XMPP?

[Markus] In our proposal SIP would be used for setting up  voice & video=20
sessions. So, you would need NAT traversal for that. How  that's done is=20
outside the scope but either B2BUA or ICE based mechanisms  would work as=20
far as they work in general. For XMPP we would not need to worry  about=20
NAT traversal beyond how its done in that protocol by default. For a=20
client there is a caveat that it has to keepalive the TCP connections with =

 BOTH SIP and XMPP servers so it would be recommended to synchronize those =

 (send them at the same time) for instance in a battery powered  device.


Return  on Investment (ROI): Cost and effort by developers. More network=20
infrastructure, not even counting B2B NEs.

What is the additional time/money required for  developers that know SIP=20
to also learn XMPP and vice versa? This is all  about return on=20
investment.

[Markus] This is a problem for sure, even more so  for the people writing=20
standard specs than code :)  Hopefully the  developer would just have add=20
a couple of simple extensions to existing  SIP and XMPP implementations.=20


Support for a REST-full  architecture

The WWW has a REST architecture was formulated in 2000,  after the=20
emergence of SIP and XMMP.  As a result of its architecture,  the WWW has=20
had a transformational effect on the following (this is not a  complete=20
list). Some SIP work recommends already recommends REST, such as in  RFC=20
4240 and in http://tools.ietf.org/html/draft-griffin-bliss-rest-00

Web applications from mail to office apps  =20
Internet applications previously served by dedicated  network applications =

protocols=20
Many other industries, from banking to newsprint to  travel. And Oh,=20
social networks, they also have  registrars...


Now please keep in mind the WWW architecture uses basically  only 3=20
components*: The URI for addressing, HTTP for transport and HTML for  data =

rendering, augmented by some other languages (XML) and namespaces. Can we=20
benefit something from the WWW architecture? If yes what  specifically?

Heresy: If the web developers could use HTTP only, what  other application =

level protocols do we really need except RTP/UDP?
This  includes in my mind TLS and DTLS for secure transport or even better =

IMO: HIP  for many reasons.


[Markus] Personally I agree that REST+HTTP+Javascript is a superior=20
toolset compared to SIP, XMPP, IMAP and so on to develop many=20
applications.  However, there are still plenty of SIP based VoIP and XMPP=20
based IM and  presence deployments out there and I don't expect the web=20
apps to replace all  of them in the near future :)

=20
* T.V. Raman: ?Toward 2W,  Beyond Web 2.0?, Communications of the ACM,=20
February 2009

Many or most  of these points may be moot or make no sense, but maybe=20
should be cleared up,  in case there are more naive people like me that=20
may be interested.

My  strict personal 2 cents.

FRANK comments are most  welcome.

Henry



On 9/4/09 7:09 AM, "Simo.Veikkolainen@nokia.com" <
Simo.Veikkolainen@nokia.com>  wrote:

=20
Hi,
=20
We would like to propose a new DISPATCH topic on  how SIP and XMPP can be=20
used together in a complementary  fashion.
=20
We have been working on a solution which combines SIP  based VoIP and XMPP =

based IM and Presence. The requirements and a proposed  solution is=20
outlined in=20
http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01  <
http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01>  .
=20
The motivation for this work comes from experience which  shows that most=20
standards based VoIP deployments use SIP. At the same time,  the=20
Extensible Messaging and Presence Protocol (XMPP) is widely used in=20
instant messaging and presence services. An interesting scenario arises=20
when  a SIP based voice (and video) service is combined together with an=20
XMPP  based instant messaging and presence service.=20
=20
Note that the  goal in this work is not SIP-XMPP protocol translation, but =

to define  protocol extensions and best practises such that SIP based VoIP =

and XMPP  based IM and presence could be seamlessly combined and offered=20
to the end  user. For rapid deployment, we assume no changes in the server =

 infrastructure, and instead impose new requirements and protocol=20
extensions  only to the endpoints.
=20
We would like to get some discussion  going on the use case itself as well =

as on the solution. Also, we would like  to hear you thoughts on DISPATCH=20
or a new "dispatched" mini-WG as the home  for such work.
=20
Exact charter proposal and problem statemen is  to follow.=20
=20
Regards,
Simo and  Markus
=20
=20



--=_alternative 006042C7C2257631_=
Content-Type: text/html; charset="ISO-8859-1"
Content-Transfer-Encoding: quoted-printable

<font size=3D2 face=3D"sans-serif">I think that this draft is an important
start for an important work item that the IETF should do.</font>
<br><font size=3D2 face=3D"sans-serif"><br>
--Avshalom<br>
</font>
<br>
<br>
<br>
<table width=3D100%>
<tr valign=3Dtop>
<td><font size=3D1 color=3D#5f5f5f face=3D"sans-serif">From:</font>
<td><font size=3D1 face=3D"sans-serif">Henry Sinnreich &lt;hsinnrei@adobe.c=
om&gt;</font>
<tr valign=3Dtop>
<td><font size=3D1 color=3D#5f5f5f face=3D"sans-serif">To:</font>
<td><font size=3D1 face=3D"sans-serif">&quot;Markus.Isomaki@nokia.com&quot;
&lt;Markus.Isomaki@nokia.com&gt;, &quot;Simo.Veikkolainen@nokia.com&quot;
&lt;Simo.Veikkolainen@nokia.com&gt;, &quot;dispatch@ietf.org&quot; &lt;disp=
atch@ietf.org&gt;,
&quot;mary.barnes@nortel.com&quot; &lt;mary.barnes@nortel.com&gt;, &quot;go=
nzalo.camarillo@ericsson.com&quot;
&lt;gonzalo.camarillo@ericsson.com&gt;</font>
<tr>
<td valign=3Dtop><font size=3D1 color=3D#5f5f5f face=3D"sans-serif">Cc:</fo=
nt>
<td><font size=3D1 face=3D"sans-serif">Avshalom Houri/Haifa/IBM@IBMIL</font>
<tr valign=3Dtop>
<td><font size=3D1 color=3D#5f5f5f face=3D"sans-serif">Date:</font>
<td><font size=3D1 face=3D"sans-serif">11/09/2009 08:41 PM</font>
<tr valign=3Dtop>
<td><font size=3D1 color=3D#5f5f5f face=3D"sans-serif">Subject:</font>
<td><font size=3D1 face=3D"sans-serif">Re: [dispatch] New topic proposal for
DISPATCH</font></table>
<br>
<hr noshade>
<br>
<br>
<br><font size=3D5 face=3D"Calibri">Hi Markus,<br>
<br>
Thanks for your thoughtful reply and I agree there are some scenarios where
it makes sense.<br>
More important, doing this in the endpoints is the right approach IMO and
as you know, I will always defend the Internet end-to-end principle :-)<br>
<br>
This distributed approach is also IMO a good answer to the Interdomain
Scaling problem for IM <br>
( I-D.draft-ietf-simple-interdomain-scaling-analysis) and I have copied
one of its authors here.<br>
<br>
This raises the questions: <br>
</font>
<br><font size=3D2 face=3D"sans-serif">1. &nbsp; &nbsp; &nbsp; &nbsp;</font=
><font size=3D5 face=3D"Calibri">Why
stop at SIP and XMPP, and not add other popular IM protocols, as is already
done by quite a few products in the market? </font>
<br><font size=3D2 face=3D"sans-serif">2. &nbsp; &nbsp; &nbsp; &nbsp;</font=
><font size=3D5 face=3D"Calibri">Can
your approach be generalized in a more generic way?</font>
<br><font size=3D5 face=3D"Calibri"><br>
If endpoints are accommodating several IM protocols, an IETF, Dispatch
recommended GENERIC approach would make many people feel more comfortable
about it. Your I-D could be used as a tested example for a generic approach=
.<br>
<br>
Ranting on preferring endpoint applications to more application protocols<b=
r>
<br>
I still believe creative application developers face the choice of either
inventing new applications or dedicating their time to learning application
protocols and they cannot do both. Learning and following both SIP and
XMPP to write and update code is quite a challenge. <br>
Is there a better way? <br>
For this reason, one single application protocol for IM and voice is enough=
.<br>
Proof: Almost all the applications that make people buy smart phones, use
online office apps and enterprise apps run on HTTP(S) and their application
developers don&#8217;t have to worry about knowing how HTTP works. As a res=
ult,
HTTP has been also used to even carry streaming media, web conferencing
and IM. Just to avoid spending a lifetime to learn and follow application
protocols. But HTTP based apps dominate the application space. This is
however for another discussion elsewhere. <br>
<br>
My strictly personal 2 cents.<br>
<br>
Thanks again,<br>
<br>
Henry <br>
<br>
<br>
On 9/11/09 8:11 AM, &quot;</font><a href=3DMarkus.Isomaki@nokia.com><font s=
ize=3D5 color=3Dblue face=3D"Calibri"><u>Markus.Isomaki@nokia.com</u></font=
></a><font size=3D5 face=3D"Calibri">&quot;
&lt;</font><a href=3DMarkus.Isomaki@nokia.com><font size=3D5 color=3Dblue f=
ace=3D"Calibri"><u>Markus.Isomaki@nokia.com</u></font></a><font size=3D5 fa=
ce=3D"Calibri">&gt;
wrote:<br>
</font>
<br><font size=3D5 color=3Dblue face=3D"Arial">Hi Henry,</font><font size=
=3D5 face=3D"Calibri"><br>
</font><font size=3D5 color=3Dblue face=3D"Arial"><br>
Thanks for your feedback :) Let me answer to the excellent points you raise=
.</font><font size=3D5 face=3D"Calibri"><br>
</font><font size=3D5 color=3Dblue face=3D"Arial"><br>
First of all, I would encourage you and everyone to carefully read the
Introduction chapter of the draft (</font><a href=3D"http://tools.ietf.org/=
html/draft-veikkolainen-sip-voip-xmpp-im-01"><font size=3D5 color=3Dblue fa=
ce=3D"Arial"><u>http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp=
-im-01</u></font></a><font size=3D5 color=3Dblue face=3D"Arial">
&lt;</font><a href=3D"http://tools.ietf.org/html/draft-veikkolainen-sip-voi=
p-xmpp-im-01"><font size=3D5 color=3Dblue face=3D"Arial"><u>http://tools.ie=
tf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01</u></font></a><font size=
=3D5 color=3Dblue face=3D"Arial">&gt;
) to be clear on what the proposal is about. That is, we are not trying
to boil the ocean and define all possible ways how SIP and XMPP infrastruct=
ures
could be combined, but instead try to define a minimalistic approach whereby
a client can use both SIP and XMPP in an integrated manner WITHOUT requiring
ANY changes on the servers that are already deployed.</font><font size=3D5 =
face=3D"Calibri"><br>
</font><font size=3D5 color=3Dblue face=3D"Arial"><br>
See further comments below with [Markus]:</font><font size=3D5 face=3D"Cali=
bri"><br>
</font>
<br><font size=3D5 face=3D"Calibri"><br>
 <br>
</font>
<hr><font size=3D5 face=3D"Tahoma"><b>From:</b> ext Henry Sinnreich &nbsp;[=
</font><a href=3Dmailto:hsinnrei@adobe.com><font size=3D5 color=3Dblue face=
=3D"Tahoma"><u>mailto:hsinnrei@adobe.com</u></font></a><font size=3D5 face=
=3D"Tahoma">]
<b><br>
Sent:</b> 04 September, 2009 &nbsp;18:42<b><br>
To:</b> Veikkolainen Simo (Nokia-D/Helsinki); </font><a href=3Ddispatch@iet=
f.org><font size=3D5 color=3Dblue face=3D"Tahoma"><u>dispatch@ietf.org</u><=
/font></a><font size=3D5 face=3D"Tahoma">;
&nbsp;</font><a href=3Dmary.barnes@nortel.com><font size=3D5 color=3Dblue f=
ace=3D"Tahoma"><u>mary.barnes@nortel.com</u></font></a><font size=3D5 face=
=3D"Tahoma">;
</font><a href=3Dgonzalo.camarillo@ericsson.com><font size=3D5 color=3Dblue=
 face=3D"Tahoma"><u>gonzalo.camarillo@ericsson.com</u></font></a><font size=
=3D5 face=3D"Tahoma"><b><br>
Cc:</b> Isomaki &nbsp;Markus (Nokia-CIC/Espoo); Olle E. Johansson<b><br>
Subject:</b> Re: [dispatch] &nbsp;New topic proposal for DISPATCH</font><fo=
nt size=3D5 face=3D"Calibri"><br>
<br>
 </font><font size=3D4 face=3D"Calibri"><br>
I would like to second such work, but a far deeper &nbsp;look under the
hood is necessary, so we don&#8217;t do another Elbonia &nbsp;project.</fon=
t><font size=3D4 color=3Dblue face=3D"Calibri"><u><br>
</u></font><a href=3D"http://dilbert.com/strips/comic/2009-09-01/"><font si=
ze=3D4 color=3Dblue face=3D"Calibri"><u>http://dilbert.com/strips/comic/200=
9-09-01/</u></font></a><font size=3D4 face=3D"Calibri"><br>
<br>
Here &nbsp;are some not very mature thoughts on the scope of the work.
<br>
The most &nbsp;interesting, ROI estimate and REST architecture support
are at the &nbsp;end.<br>
<br>
Servers<br>
<br>
Fundamentally, we need a registrar, a proxy and a &nbsp;presence server:
Three in total</font><font size=3D5 face=3D"Calibri"><br>
</font>
<ul>
<li><font size=3D4 face=3D"Calibri">What will be the duplication of servers
and where &nbsp;specifically? </font>
<li><font size=3D4 face=3D"Calibri">How does the SIP registrar communicate
with the XMPP &nbsp;server(s)? </font>
<li><font size=3D4 face=3D"Calibri">What may be the routing protocol(s) ins=
ide
a &nbsp;SIP+XMMP network? Think of IMS as a stressing use &nbsp;case.</font=
></ul><font size=3D5 color=3Dblue face=3D"Arial"><br>
[Markus] Our starting point is that we already have quite &nbsp;many SIP
VoIP deployments out there, as long as quite many XMPP presence &amp; &nbsp=
;IM
deployments. So, the servers you mention already exist. If one now &nbsp;wa=
nts
to develop a client that does VoIP, IM and presence together, it should
&nbsp;be possible to utilize these existing infrastrcutures and not have
to wait for &nbsp;new ones to be deployed (as for instance we have waited
quite many years for &nbsp;SIMPLE-based Presence servers to get deployed
but compared to XMPP there's &nbsp;very little in use). The idea is to
define a couple of E2E (as in &nbsp;nothing that SIP proxies or XMPP servers
would have to support) protocol &nbsp;extensions which would allow the
client to nicely combine SIP VoIP and XMPP &nbsp;presence &amp; IM. </font>=
<font size=3D5 face=3D"Calibri"><br>
<br>
 <br>
 </font><font size=3D5 color=3Dblue face=3D"Arial"><br>
There is no need for SIP and XMPP servers to know about &nbsp;each other
or route protocol messages between each other. All the logic is put &nbsp;i=
nto
the endpoints. No need to worry about IMS either, except &nbsp;that in
theory the client can use IMS for its VoIP service &nbsp;(similar to any
other SIP &nbsp;infra).</font><font size=3D4 face=3D"Calibri"> </font><font=
 size=3D5 face=3D"Calibri"><br>
</font><font size=3D4 face=3D"Calibri"><br>
<br>
User &nbsp;Agents</font><font size=3D5 face=3D"Calibri"><br>
</font>
<ul>
<li><font size=3D4 face=3D"Calibri">Same as with servers: How many and what
protocol &nbsp;RFCs are there in the combination? </font>
<li><font size=3D4 face=3D"Calibri">Which of the &#8220;services&#8221; can=
 be placed
in the &nbsp;applications in the endpoints or in the network application
protocols, such &nbsp;as SIP and XMPP. This is the topic on infrastructure
vs. endpoint &nbsp;applications.</font></ul><font size=3D5 color=3Dblue fac=
e=3D"Arial"><br>
[Markus] Just take any current SIP VoIP &nbsp;UA implementation, XMPP client
implementation and glue them together &nbsp;and you're already pretty far.
Add a couple of extensions as proposed and it's &nbsp;done. From there
onwards you have a pretty powerful machinery to add &nbsp;applications
by using SIP session setup and XMPP message delivery &nbsp;capabilities.
</font><font size=3D5 face=3D"Calibri"><br>
</font><font size=3D4 face=3D"Calibri"><br>
<br>
Application Protocol &nbsp;Stacks</font><font size=3D5 face=3D"Calibri"><br>
</font>
<ul>
<li><font size=3D4 face=3D"Calibri">How many protocol stacks does this invo=
lve
and more &nbsp;to the point, how many RFCs? </font>
<li><font size=3D4 face=3D"Calibri">For SIP alone, see </font><a href=3Dhtt=
p://rfc3261.net/><font size=3D4 color=3Dblue face=3D"Calibri"><u>http://rfc=
3261.net/</u></font></a><font size=3D4 face=3D"Calibri">.
What will be the total &nbsp;combined RFC count?</font></ul><font size=3D5 =
color=3Dblue face=3D"Arial"><br>
[Markus] All what we expect is already pretty widely &nbsp;supported and
even available as open source: basic SIP VoIP and XMPP IM and &nbsp;presence
is enough to start with. &nbsp; </font><font size=3D5 face=3D"Calibri"><br>
</font><font size=3D4 face=3D"Calibri"><br>
<br>
NAT &nbsp;Traversal</font><font size=3D5 face=3D"Calibri"><br>
</font>
<ul>
<li><font size=3D4 face=3D"Calibri">Will STUN/TURN/ICE work the same for SIP
and for &nbsp;XMPP?</font></ul><font size=3D5 color=3Dblue face=3D"Arial"><=
br>
[Markus] In our proposal SIP would be used for setting up &nbsp;voice &amp;
video sessions. So, you would need NAT traversal for that. How &nbsp;that's
done is outside the scope but either B2BUA or ICE based mechanisms &nbsp;wo=
uld
work as far as they work in general. For XMPP we would not need to worry
&nbsp;about NAT traversal beyond how its done in that protocol by default.
For a &nbsp;client there is a caveat that it has to keepalive the TCP conne=
ctions
with &nbsp;BOTH SIP and XMPP servers so it would be recommended to synchron=
ize
those &nbsp;(send them at the same time) for instance in a battery powered
&nbsp;device.</font><font size=3D5 face=3D"Calibri"><br>
</font><font size=3D4 face=3D"Calibri"><br>
<br>
Return &nbsp;on Investment (ROI): Cost and effort by developers. More netwo=
rk
&nbsp;infrastructure, not even counting B2B NEs.</font><font size=3D5 face=
=3D"Calibri"><br>
</font>
<ul>
<li><font size=3D4 face=3D"Calibri">What is the additional time/money requi=
red
for &nbsp;developers that know SIP to also learn XMPP and vice versa? This
is all &nbsp;about return on investment.</font></ul><font size=3D5 color=3D=
blue face=3D"Arial"><br>
[Markus] This is a problem for sure, even more so &nbsp;for the people
writing standard specs than code :) &nbsp;Hopefully the &nbsp;developer
would just have add a couple of simple extensions to existing &nbsp;SIP
and XMPP implementations. </font><font size=3D5 face=3D"Calibri"><br>
</font><font size=3D4 face=3D"Calibri"><br>
<br>
Support for a REST-full &nbsp;architecture<br>
<br>
The WWW has a REST architecture was formulated in 2000, &nbsp;after the
emergence of SIP and XMMP. &nbsp;As a result of its architecture, &nbsp;the
WWW has had a transformational effect on the following (this is not a &nbsp=
;complete
list). Some SIP work recommends already recommends REST, such as in &nbsp;R=
FC
4240 and in </font><a href=3D"http://tools.ietf.org/html/draft-griffin-blis=
s-rest-00"><font size=3D4 color=3Dblue face=3D"Calibri"><u>http://tools.iet=
f.org/html/draft-griffin-bliss-rest-00</u></font></a><font size=3D5 face=3D=
"Calibri"><br>
</font>
<ul>
<li><font size=3D4 face=3D"Calibri">Web applications from mail to office ap=
ps
&nbsp;</font><font size=3D5 face=3D"Calibri"> </font>
<li><font size=3D4 face=3D"Calibri">Internet applications previously served
by dedicated &nbsp;network applications protocols </font>
<li><font size=3D4 face=3D"Calibri">Many other industries, from banking to
newsprint to &nbsp;travel. And Oh, social networks, they also have &nbsp;re=
gistrars...</font></ul><font size=3D4 face=3D"Calibri"><br>
<br>
Now please keep in mind the WWW architecture uses basically &nbsp;only
3 components*: The URI for addressing, HTTP for transport and HTML for
&nbsp;data rendering, augmented by some other languages (XML) and namespace=
s.
Can we &nbsp;benefit something from the WWW architecture? If yes what &nbsp=
;specifically?<br>
<br>
Heresy: If the web developers could use HTTP only, what &nbsp;other applica=
tion
level protocols do we really need except RTP/UDP?<br>
This &nbsp;includes in my mind TLS and DTLS for secure transport or even
better IMO: HIP &nbsp;for many reasons.</font><font size=3D5 color=3Dblue f=
ace=3D"Arial"><br>
</font><font size=3D4 face=3D"Calibri"><br>
</font><font size=3D5 color=3Dblue face=3D"Arial"><br>
[Markus] Personally I agree that REST+HTTP+Javascript is a superior &nbsp;t=
oolset
compared to SIP, XMPP, IMAP and so on to develop many applications. &nbsp;H=
owever,
there are still plenty of SIP based VoIP and XMPP based IM and &nbsp;presen=
ce
deployments out there and I don't expect the web apps to replace all &nbsp;=
of
them in the near future :)</font><font size=3D4 face=3D"Calibri"><br>
<br>
 <br>
* T.V. Raman: &#8220;Toward 2W, &nbsp;Beyond Web 2.0&#8221;, Communications=
 of the
ACM, February 2009<br>
<br>
Many or most &nbsp;of these points may be moot or make no sense, but maybe
should be cleared up, &nbsp;in case there are more naive people like me
that may be interested.<br>
<br>
My &nbsp;strict personal 2 cents.<br>
<br>
FRANK comments are most &nbsp;welcome.<br>
<br>
Henry</font><font size=3D5 face=3D"Calibri"><br>
<br>
<br>
<br>
On 9/4/09 7:09 AM, &quot;</font><a href=3DSimo.Veikkolainen@nokia.com><font=
 size=3D5 color=3Dblue face=3D"Calibri"><u>Simo.Veikkolainen@nokia.com</u><=
/font></a><font size=3D5 face=3D"Calibri">&quot;
&lt;</font><a href=3DSimo.Veikkolainen@nokia.com><font size=3D5 color=3Dblu=
e face=3D"Calibri"><u>Simo.Veikkolainen@nokia.com</u></font></a><font size=
=3D5 face=3D"Calibri">&gt;
&nbsp;wrote:<br>
<br>
 </font>
<br><font size=3D5 face=3D"Arial">Hi,<br>
 <br>
We would like to propose a new DISPATCH topic on &nbsp;how SIP and XMPP
can be used together in a complementary &nbsp;fashion.<br>
 <br>
We have been working on a solution which combines SIP &nbsp;based VoIP
and XMPP based IM and Presence. The requirements and a proposed &nbsp;solut=
ion
is outlined in </font><a href=3D"http://tools.ietf.org/html/draft-veikkolai=
nen-sip-voip-xmpp-im-01"><font size=3D5 color=3Dblue face=3D"Arial"><u>http=
://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01</u></font></a=
><font size=3D5 face=3D"Arial">
&nbsp;&lt;</font><a href=3D"http://tools.ietf.org/html/draft-veikkolainen-s=
ip-voip-xmpp-im-01"><font size=3D5 color=3Dblue face=3D"Arial"><u>http://to=
ols.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01</u></font></a><fon=
t size=3D5 face=3D"Arial">&gt;
&nbsp;.<br>
 <br>
The motivation for this work comes from experience which &nbsp;shows that
most standards based VoIP deployments use SIP. At the same time, &nbsp;the
Extensible Messaging and Presence Protocol (XMPP) is widely used in &nbsp;i=
nstant
messaging and presence services. An interesting scenario arises when &nbsp;a
SIP based voice (and video) service is combined together with an XMPP &nbsp=
;based
instant messaging and presence service. <br>
 <br>
Note that the &nbsp;goal in this work is not SIP-XMPP protocol translation,
but to define &nbsp;protocol extensions and best practises such that SIP
based VoIP and XMPP &nbsp;based IM and presence could be seamlessly combined
and offered to the end &nbsp;user. For rapid deployment, we assume no chang=
es
in the server &nbsp;infrastructure, and instead impose new requirements
and protocol extensions &nbsp;only to the endpoints.<br>
 <br>
We would like to get some discussion &nbsp;going on the use case itself
as well as on the solution. Also, we would like &nbsp;to hear you thoughts
on DISPATCH or a new &quot;dispatched&quot; mini-WG as the home &nbsp;for
such work.<br>
 <br>
Exact charter proposal and problem statemen is &nbsp;to follow. <br>
 <br>
Regards,<br>
Simo and &nbsp;Markus<br>
 <br>
 </font><font size=3D5 face=3D"Calibri"><br>
</font>
<br>
<br>
--=_alternative 006042C7C2257631_=--

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On 9/11/09 11:41 AM, Henry Sinnreich wrote:

>    1. Why stop at SIP and XMPP, and not add other popular IM protocols,
>       as is already done by quite a few products in the market?

Because the IETF is a standards organization and it't not our job to
define hackish gateways to things like MySpaceIM? If those people want
to use standardized technologies, it's quite easy for them to do so.

Peter

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https://stpeter.im/


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On 9/14/09 12:42 PM, "Peter Saint-Andre" <stpeter@stpeter.im> wrote:

>Because the IETF is a standards organization and it't not our job to
>define hackish gateways to things like MySpaceIM?

It's in the eye of the beholder what is a hack, a successful innovation or =
a future standard, de facto or from a standards organization. Presence and =
IM started out as proprietary systems long time before standards emerged an=
d unless I am mistaken, proprietary Presence and IM are still dominant or a=
t least significant in the market. Does anyone have numbers on this?

Today's "hack" may be the subject of tomorrow's standards.

Henry

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<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
18pt'>On 9/14/09 12:42 PM, &quot;Peter Saint-Andre&quot; &lt;<a href=3D"stp=
eter@stpeter.im">stpeter@stpeter.im</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:18pt'>&gt;Because the IETF is a standards organiz=
ation and it't not our job to<BR>
&gt;define hackish gateways to things like MySpaceIM? <BR>
<BR>
It&#8217;s in the eye of the beholder what is a hack, a successful innovati=
on or a future standard, de facto or from a standards organization. Presenc=
e and IM started out as proprietary systems long time before standards emer=
ged and unless I am mistaken, proprietary Presence and IM are still dominan=
t or at least significant in the market. Does anyone have numbers on this?<=
BR>
<BR>
Today&#8217;s &#8220;hack&#8221; may be the subject of tomorrow&#8217;s sta=
ndards.<BR>
<BR>
Henry<BR>
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Re: [dispatch] New topic proposal for DISPATCHHi, Henry,

I don't disagree with your note below. I understood Peter's concern to =
be that interworking with a proprietary protocol in an SDO is =
problematic because the protocol we're interworking with can change at =
any time.=20

There were certainly plenty of proprietary e-mail protocols when SMTP =
was produced. My understanding (and I wasn't there) was that we defined =
interworking with X.400, and encouraged people with proprietary =
protocols to develop interworking with either SMTP directly, or with =
X.400 (which we knew how to interwork with SMTP.

That's what I understood Peter to be proposing. Sorry if I =
misunderstand, of course.

Spencer
  ----- Original Message -----=20
  From: Henry Sinnreich=20
  To: Peter Saint-Andre=20
  Cc: dispatch@ietf.org ; gonzalo.camarillo@ericsson.com ; =
Markus.Isomaki@nokia.com ; Simo.Veikkolainen@nokia.com=20
  Sent: Monday, September 14, 2009 2:50 PM
  Subject: Re: [dispatch] New topic proposal for DISPATCH


  On 9/14/09 12:42 PM, "Peter Saint-Andre" <stpeter@stpeter.im> wrote:


    >Because the IETF is a standards organization and it't not our job =
to
    >define hackish gateways to things like MySpaceIM?=20

    It's in the eye of the beholder what is a hack, a successful =
innovation or a future standard, de facto or from a standards =
organization. Presence and IM started out as proprietary systems long =
time before standards emerged and unless I am mistaken, proprietary =
Presence and IM are still dominant or at least significant in the =
market. Does anyone have numbers on this?

    Today's "hack" may be the subject of tomorrow's standards.

    Henry



-------------------------------------------------------------------------=
-----


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

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<HTML><HEAD><TITLE>Re: [dispatch] New topic proposal for =
DISPATCH</TITLE>
<META content=3D"text/html; charset=3Diso-8859-1" =
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<DIV><FONT size=3D2>Hi, Henry,</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>I don't disagree with your note below. I understood =
Peter's=20
concern to be that interworking with a proprietary protocol in an SDO is =

problematic because the protocol we're interworking with can change at =
any time.=20
</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>There were certainly plenty of proprietary e-mail =
protocols=20
when SMTP was produced. My understanding (and I wasn't there) was that =
we=20
defined interworking with X.400, and encouraged people with proprietary=20
protocols to develop interworking with either SMTP directly, or with =
X.400=20
(which we knew how to interwork with SMTP.</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>That's what I understood Peter to be proposing. =
Sorry if I=20
misunderstand, of course.</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>Spencer</FONT></DIV>
<BLOCKQUOTE=20
style=3D"BORDER-LEFT: #000000 2px solid; PADDING-LEFT: 5px; =
PADDING-RIGHT: 0px; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px">
  <DIV style=3D"FONT: 10pt arial">----- Original Message ----- </DIV>
  <DIV=20
  style=3D"FONT: 10pt arial; BACKGROUND: #e4e4e4; font-color: =
black"><B>From:</B>=20
  <A title=3Dhsinnrei@adobe.com href=3D"mailto:hsinnrei@adobe.com">Henry =

  Sinnreich</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A =
title=3Dstpeter@stpeter.im=20
  href=3D"mailto:stpeter@stpeter.im">Peter Saint-Andre</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Cc:</B> <A =
title=3Ddispatch@ietf.org=20
  href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</A> ; <A=20
  title=3Dgonzalo.camarillo@ericsson.com=20
  =
href=3D"mailto:gonzalo.camarillo@ericsson.com">gonzalo.camarillo@ericsson=
.com</A>=20
  ; <A title=3DMarkus.Isomaki@nokia.com=20
  href=3D"mailto:Markus.Isomaki@nokia.com">Markus.Isomaki@nokia.com</A> =
; <A=20
  title=3DSimo.Veikkolainen@nokia.com=20
  =
href=3D"mailto:Simo.Veikkolainen@nokia.com">Simo.Veikkolainen@nokia.com</=
A>=20
  </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Monday, September 14, =
2009 2:50=20
  PM</DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Subject:</B> Re: [dispatch] New =
topic=20
  proposal for DISPATCH</DIV>
  <DIV><BR></DIV><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SPAN =

  style=3D"FONT-SIZE: 18pt">On 9/14/09 12:42 PM, "Peter Saint-Andre" =
&lt;<A=20
  href=3D"mailto:stpeter@stpeter.im">stpeter@stpeter.im</A>&gt;=20
  wrote:<BR><BR></SPAN></FONT>
  <BLOCKQUOTE><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SPAN=20
    style=3D"FONT-SIZE: 18pt">&gt;Because the IETF is a standards =
organization and=20
    it't not our job to<BR>&gt;define hackish gateways to things like =
MySpaceIM?=20
    <BR><BR>It=92s in the eye of the beholder what is a hack, a =
successful=20
    innovation or a future standard, de facto or from a standards =
organization.=20
    Presence and IM started out as proprietary systems long time before=20
    standards emerged and unless I am mistaken, proprietary Presence and =
IM are=20
    still dominant or at least significant in the market. Does anyone =
have=20
    numbers on this?<BR><BR>Today=92s =93hack=94 may be the subject of =
tomorrow=92s=20
    standards.<BR><BR>Henry<BR></SPAN></FONT></BLOCKQUOTE>
  <P>
  <HR>

  <P></P>_______________________________________________<BR>dispatch =
mailing=20
  =
list<BR>dispatch@ietf.org<BR>https://www.ietf.org/mailman/listinfo/dispat=
ch<BR></BLOCKQUOTE></BODY></HTML>

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From: Henry Sinnreich <hsinnrei@adobe.com>
To: Spencer Dawkins <spencer@wonderhamster.org>, Peter Saint-Andre <stpeter@stpeter.im>
Date: Tue, 15 Sep 2009 05:49:50 -0700
Thread-Topic: [dispatch] New topic proposal for DISPATCH
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Subject: Re: [dispatch] New topic proposal for DISPATCH
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Spencer,

Our understanding seems to agree.
Sure, this is worth a longer discussion, but at some other time and place.

Henry


On 9/15/09 7:42 AM, "Spencer Dawkins" <spencer@wonderhamster.org> wrote:

Hi, Henry,

I don't disagree with your note below. I understood Peter's concern to be t=
hat interworking with a proprietary protocol in an SDO is problematic becau=
se the protocol we're interworking with can change at any time.

There were certainly plenty of proprietary e-mail protocols when SMTP was p=
roduced. My understanding (and I wasn't there) was that we defined interwor=
king with X.400, and encouraged people with proprietary protocols to develo=
p interworking with either SMTP directly, or with X.400 (which we knew how =
to interwork with SMTP.

That's what I understood Peter to be proposing. Sorry if I misunderstand, o=
f course.

Spencer

----- Original Message -----

From:  Henry  Sinnreich <mailto:hsinnrei@adobe.com>

To: Peter Saint-Andre <mailto:stpeter@stpeter.im>

Cc: dispatch@ietf.org ; gonzalo.camarillo@ericsson.com  ; Markus.Isomaki@no=
kia.com ; Simo.Veikkolainen@nokia.com

Sent: Monday, September 14, 2009 2:50  PM

Subject: Re: [dispatch] New topic  proposal for DISPATCH


On 9/14/09 12:42 PM, "Peter Saint-Andre" <stpeter@stpeter.im>  wrote:


>Because the IETF is a standards organization and  it't not our job to
>define hackish gateways to things like MySpaceIM?

It's in the eye of the beholder what is a hack, a successful  innovation or=
 a future standard, de facto or from a standards organization.  Presence an=
d IM started out as proprietary systems long time before  standards emerged=
 and unless I am mistaken, proprietary Presence and IM are  still dominant =
or at least significant in the market. Does anyone have  numbers on this?

Today's "hack" may be the subject of tomorrow's  standards.

Henry



________________________________


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


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<HEAD>
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</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
18pt'>Spencer,<BR>
<BR>
Our understanding seems to agree.<BR>
Sure, this is worth a longer discussion, but at some other time and place.<=
BR>
<BR>
Henry<BR>
<BR>
<BR>
On 9/15/09 7:42 AM, &quot;Spencer Dawkins&quot; &lt;<a href=3D"spencer@wond=
erhamster.org">spencer@wonderhamster.org</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:18pt'>Hi, Henry,<BR>
&nbsp;<BR>
I don't disagree with your note below. I understood Peter's concern to be t=
hat interworking with a proprietary protocol in an SDO is problematic becau=
se the protocol we're interworking with can change at any time. <BR>
&nbsp;<BR>
There were certainly plenty of proprietary e-mail protocols when SMTP was p=
roduced. My understanding (and I wasn't there) was that we defined interwor=
king with X.400, and encouraged people with proprietary protocols to develo=
p interworking with either SMTP directly, or with X.400 (which we knew how =
to interwork with SMTP.<BR>
&nbsp;<BR>
That's what I understood Peter to be proposing. Sorry if I misunderstand, o=
f course.<BR>
&nbsp;<BR>
Spencer<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:18pt'> <BR>
----- Original Message ----- <BR>
&nbsp;<BR>
<B>From:</B> &nbsp;Henry &nbsp;Sinnreich &lt;<a href=3D"mailto:hsinnrei@ado=
be.com">mailto:hsinnrei@adobe.com</a>&gt; &nbsp;<BR>
&nbsp;<BR>
<B>To:</B> Peter Saint-Andre &lt;<a href=3D"mailto:stpeter@stpeter.im">mail=
to:stpeter@stpeter.im</a>&gt; &nbsp;<BR>
&nbsp;<BR>
<B>Cc:</B> <a href=3D"dispatch@ietf.org">dispatch@ietf.org</a> ; <a href=3D=
"gonzalo.camarillo@ericsson.com">gonzalo.camarillo@ericsson.com</a> &nbsp;;=
 <a href=3D"Markus.Isomaki@nokia.com">Markus.Isomaki@nokia.com</a> ; <a hre=
f=3D"Simo.Veikkolainen@nokia.com">Simo.Veikkolainen@nokia.com</a> &nbsp;<BR=
>
&nbsp;<BR>
<B>Sent:</B> Monday, September 14, 2009 2:50 &nbsp;PM<BR>
&nbsp;<BR>
<B>Subject:</B> Re: [dispatch] New topic &nbsp;proposal for DISPATCH<BR>
&nbsp;<BR>
<BR>
On 9/14/09 12:42 PM, &quot;Peter Saint-Andre&quot; &lt;<a href=3D"stpeter@s=
tpeter.im">stpeter@stpeter.im</a>&gt; &nbsp;wrote:<BR>
<BR>
&nbsp;<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:18pt'>&gt;Because the IETF is a standards organiz=
ation and &nbsp;it't not our job to<BR>
&gt;define hackish gateways to things like MySpaceIM? &nbsp;<BR>
<BR>
It&#8217;s in the eye of the beholder what is a hack, a successful &nbsp;in=
novation or a future standard, de facto or from a standards organization. &=
nbsp;Presence and IM started out as proprietary systems long time before &n=
bsp;standards emerged and unless I am mistaken, proprietary Presence and IM=
 are &nbsp;still dominant or at least significant in the market. Does anyon=
e have &nbsp;numbers on this?<BR>
<BR>
Today&#8217;s &#8220;hack&#8221; may be the subject of tomorrow&#8217;s &nb=
sp;standards.<BR>
<BR>
Henry<BR>
</SPAN></FONT></BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial=
"><SPAN STYLE=3D'font-size:18pt'> <BR>
<BR>
&nbsp;<BR>
<HR ALIGN=3DCENTER SIZE=3D"3" WIDTH=3D"100%"> <BR>
<BR>
_______________________________________________<BR>
dispatch mailing &nbsp;list<BR>
<a href=3D"dispatch@ietf.org">dispatch@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf=
.org/mailman/listinfo/dispatch</a><BR>
<BR>
</SPAN></FONT></BLOCKQUOTE></BLOCKQUOTE>
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Subject: [dispatch] Cutoff for charter proposals on Monday
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Folks,

this is a reminder of our cutoff for charter proposals, which is on 
Monday (September 21st). Per our previous email: "A charter proposal 
must consist of a minimum of a problem statement and at least one 
milestone/deliverable. This deadline will allow time for consideration 
of proposals that could potentially be "dispatched" prior to the WG 
session."

A few of the topics we received arrived after the previous deadline, 
which was September 7th. We will be considering even those topics that 
arrived a bit late because we realize the deadline was quite tough to 
meet (specially for people that returned from their vacation right 
before the deadline).

We are expecting charter proposals for:

o Third-party authorization
o CBUS
o Session recording
o Identity
o Disaggregated media
o Domain registrations in SIP
o SIP - XMPP
o Alert-info URNs
o Reference to an earlier communication

Let the chairs know if there are topics missing from this list.

Thanks,

Gonzalo
DISPATCH co-chair

From victor.pascual.avila@gmail.com  Fri Sep 18 10:13:02 2009
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Subject: Re: [dispatch] Cutoff for charter proposals on Monday
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Gonzalo,

On Wed, Sep 16, 2009 at 4:12 PM, Gonzalo Camarillo
<Gonzalo.Camarillo@ericsson.com> wrote:
> Folks,
>
> this is a reminder of our cutoff for charter proposals, which is on Monda=
y
> (September 21st). Per our previous email: "A charter proposal must consis=
t
> of a minimum of a problem statement and at least one milestone/deliverabl=
e.
> This deadline will allow time for consideration of proposals that could
> potentially be "dispatched" prior to the WG session."
>
> A few of the topics we received arrived after the previous deadline, whic=
h
> was September 7th. We will be considering even those topics that arrived =
a
> bit late because we realize the deadline was quite tough to meet (special=
ly
> for people that returned from their vacation right before the deadline).
>
> We are expecting charter proposals for:
>
> o Third-party authorization
> o CBUS
> o Session recording
> o Identity

As John mentioned, we don't believe we are in a position yet to propose
a charter. However, we hope to update
http://tools.ietf.org/id/draft-elwell-dispatch-identity-reqs-00.txt
before the meeting.

Cheers,
-Victor

> o Disaggregated media
> o Domain registrations in SIP
> o SIP - XMPP
> o Alert-info URNs
> o Reference to an earlier communication
>
> Let the chairs know if there are topics missing from this list.
>
> Thanks,
>
> Gonzalo
> DISPATCH co-chair
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>



--=20
Victor Pascual =C3=81vila

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To: "'Paul Kyzivat'" <pkyzivat@cisco.com>, "'Henry Sinnreich'" <hsinnrei@adobe.com>
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Cc: Simo.Veikkolainen@nokia.com, dispatch@ietf.org, gonzalo.camarillo@ericsson.com, Markus.Isomaki@nokia.com
Subject: Re: [dispatch] New topic proposal for DISPATCH
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> -----Original Message-----
> From: dispatch-bounces@ietf.org 
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat
> Sent: Friday, September 11, 2009 12:56 PM
> To: Henry Sinnreich
> Cc: dispatch@ietf.org; avshalom@il.ibm.com; 
> gonzalo.camarillo@ericsson.com; Markus.Isomaki@nokia.com; 
> Simo.Veikkolainen@nokia.com
> Subject: Re: [dispatch] New topic proposal for DISPATCH
> 
> I agree that there are pragmatic reasons to pursue the 
> coordination/coexistence of sip and xmpp. And pushing 
> responsibility to 
> the endpoints remains desirable.
> 
> One thing that remains of concern to me in such work is that 
> we end up 
> with disjoint address spaces without any straightforward way to 
> correlate them. So if I want a single "session" involving 
> both voice via 
> SIP and IM via xmpp, then I have great difficulty in discovering URIs 
> for the "subordinate" session that correlate with those for the 
> "primary" session. (No pejorative intended here - one or the 
> other must go first.)

So XMPP has to use E.164's (like SIP deployments do), or SIP
will have to switch to user@host.  I don't forsee either happening.  
If this is deemed necessary then we will have to map between them 
-- somehow.  Send an XMPP message asking "what is your phone
number" (in a computer-parsable manner), perhaps?

-d


> Somehow this problem needs to be resolved.
> 
> 	Thanks,
> 	Paul
> 
> Henry Sinnreich wrote:
> > Hi Markus,
> > 
> > Thanks for your thoughtful reply and I agree there are some 
> scenarios 
> > where it makes sense.
> > More important, doing this in the endpoints is the right 
> approach IMO 
> > and as you know, I will always defend the Internet 
> end-to-end principle :-)
> > 
> > This distributed approach is also IMO a good answer to the 
> Interdomain 
> > Scaling problem for IM
> > ( I-D.draft-ietf-simple-interdomain-scaling-analysis) and I 
> have copied 
> > one of its authors here.
> > 
> > This raises the questions:
> > 
> >    1. Why stop at SIP and XMPP, and not add other popular 
> IM protocols,
> >       as is already done by quite a few products in the market?
> >    2. Can your approach be generalized in a more generic way?
> > 
> > 
> > If endpoints are accommodating several IM protocols, an 
> IETF, Dispatch 
> > recommended GENERIC approach would make many people feel more 
> > comfortable about it. Your I-D could be used as a tested 
> example for a 
> > generic approach.
> > 
> > Ranting on preferring endpoint applications to more 
> application protocols
> > 
> > I still believe creative application developers face the choice of 
> > either inventing new applications or dedicating their time 
> to learning 
> > application protocols and they cannot do both. Learning and 
> following 
> > both SIP and XMPP to write and update code is quite a challenge.
> > Is there a better way?
> > For this reason, one single application protocol for IM and 
> voice is enough.
> > Proof: Almost all the applications that make people buy 
> smart phones, 
> > use online office apps and enterprise apps run on HTTP(S) and their 
> > application developers don't have to worry about knowing 
> how HTTP works. 
> > As a result, HTTP has been also used to even carry 
> streaming media, web 
> > conferencing and IM. Just to avoid spending a lifetime to learn and 
> > follow application protocols. But HTTP based apps dominate the 
> > application space. This is however for another discussion elsewhere.
> > 
> > My strictly personal 2 cents.
> > 
> > Thanks again,
> > 
> > Henry
> > 
> > 
> > On 9/11/09 8:11 AM, "Markus.Isomaki@nokia.com" 
> > <Markus.Isomaki@nokia.com> wrote:
> > 
> >     Hi Henry,
> > 
> >     Thanks for your feedback :) Let me answer to the 
> excellent points
> >     you raise.
> > 
> >     First of all, I would encourage you and everyone to 
> carefully read
> >     the Introduction chapter of the draft
> >     
> (http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01
> >     
> <http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01>
> >     ) to be clear on what the proposal is about. That is, we are not
> >     trying to boil the ocean and define all possible ways 
> how SIP and
> >     XMPP infrastructures could be combined, but instead try 
> to define a
> >     minimalistic approach whereby a client can use both SIP 
> and XMPP in
> >     an integrated manner WITHOUT requiring ANY changes on 
> the servers
> >     that are already deployed.
> > 
> >     See further comments below with [Markus]:
> > 
> > 
> >          
> >         
> --------------------------------------------------------------
> ----------
> >         *From:* ext Henry Sinnreich  [mailto:hsinnrei@adobe.com]
> >         *Sent:* 04 September, 2009  18:42
> >         *To:* Veikkolainen Simo (Nokia-D/Helsinki); 
> dispatch@ietf.org;
> >          mary.barnes@nortel.com; gonzalo.camarillo@ericsson.com
> >         *Cc:* Isomaki  Markus (Nokia-CIC/Espoo); Olle E. Johansson
> >         *Subject:* Re: [dispatch]  New topic proposal for DISPATCH
> > 
> >          
> >         I would like to second such work, but a far deeper  
> look under
> >         the hood is necessary, so we don't do another 
> Elbonia  project.
> >         http://dilbert.com/strips/comic/2009-09-01/
> > 
> >         Here  are some not very mature thoughts on the 
> scope of the work.
> >         The most  interesting, ROI estimate and REST architecture
> >         support are at the  end.
> > 
> >         Servers
> > 
> >         Fundamentally, we need a registrar, a proxy and a  presence
> >         server: Three in total
> > 
> >             * What will be the duplication of servers and where
> >                specifically?
> >             * How does the SIP registrar communicate with the XMPP
> >                server(s)?
> >             * What may be the routing protocol(s) inside a  SIP+XMMP
> >               network? Think of IMS as a stressing use  case.
> > 
> > 
> >         [Markus] Our starting point is that we already have 
> quite  many
> >         SIP VoIP deployments out there, as long as quite many XMPP
> >         presence &  IM deployments. So, the servers you 
> mention already
> >         exist. If one now  wants to develop a client that 
> does VoIP, IM
> >         and presence together, it should  be possible to 
> utilize these
> >         existing infrastrcutures and not have to wait for  
> new ones to
> >         be deployed (as for instance we have waited quite 
> many years for
> >          SIMPLE-based Presence servers to get deployed but 
> compared to
> >         XMPP there's  very little in use). The idea is to define a
> >         couple of E2E (as in  nothing that SIP proxies or 
> XMPP servers
> >         would have to support) protocol  extensions which 
> would allow
> >         the client to nicely combine SIP VoIP and XMPP  
> presence & IM.
> > 
> >          
> >          
> >         There is no need for SIP and XMPP servers to know 
> about  each
> >         other or route protocol messages between each other. All the
> >         logic is put  into the endpoints. No need to worry about IMS
> >         either, except  that in theory the client can use 
> IMS for its
> >         VoIP service  (similar to any other SIP  infra).
> > 
> > 
> >         User  Agents
> > 
> >             * Same as with servers: How many and what 
> protocol  RFCs are
> >               there in the combination?
> >             * Which of the "services" can be placed in the  
> applications
> >               in the endpoints or in the network 
> application protocols,
> >               such  as SIP and XMPP. This is the topic on 
> infrastructure
> >               vs. endpoint  applications.
> > 
> > 
> >         [Markus] Just take any current SIP VoIP  UA 
> implementation, XMPP
> >         client implementation and glue them together  and 
> you're already
> >         pretty far. Add a couple of extensions as proposed and it's
> >          done. From there onwards you have a pretty 
> powerful machinery
> >         to add  applications by using SIP session setup and 
> XMPP message
> >         delivery  capabilities.
> > 
> > 
> >         Application Protocol  Stacks
> > 
> >             * How many protocol stacks does this involve 
> and more  to
> >               the point, how many RFCs?
> >             * For SIP alone, see http://rfc3261.net/. What 
> will be the
> >               total  combined RFC count?
> > 
> > 
> >         [Markus] All what we expect is already pretty 
> widely  supported
> >         and even available as open source: basic SIP VoIP 
> and XMPP IM
> >         and  presence is enough to start with.   
> > 
> > 
> >         NAT  Traversal
> > 
> >             * Will STUN/TURN/ICE work the same for SIP and 
> for  XMPP?
> > 
> > 
> >         [Markus] In our proposal SIP would be used for 
> setting up  voice
> >         & video sessions. So, you would need NAT traversal 
> for that. How
> >          that's done is outside the scope but either B2BUA 
> or ICE based
> >         mechanisms  would work as far as they work in 
> general. For XMPP
> >         we would not need to worry  about NAT traversal 
> beyond how its
> >         done in that protocol by default. For a  client there is a
> >         caveat that it has to keepalive the TCP connections 
> with  BOTH
> >         SIP and XMPP servers so it would be recommended to 
> synchronize
> >         those  (send them at the same time) for instance in 
> a battery
> >         powered  device.
> > 
> > 
> >         Return  on Investment (ROI): Cost and effort by 
> developers. More
> >         network  infrastructure, not even counting B2B NEs.
> > 
> >             * What is the additional time/money required 
> for  developers
> >               that know SIP to also learn XMPP and vice 
> versa? This is
> >               all  about return on investment.
> > 
> > 
> >         [Markus] This is a problem for sure, even more so  for the
> >         people writing standard specs than code :)  Hopefully the
> >          developer would just have add a couple of simple 
> extensions to
> >         existing  SIP and XMPP implementations.
> > 
> > 
> >         Support for a REST-full  architecture
> > 
> >         The WWW has a REST architecture was formulated in 
> 2000,  after
> >         the emergence of SIP and XMMP.  As a result of its 
> architecture,
> >          the WWW has had a transformational effect on the following
> >         (this is not a  complete list). Some SIP work 
> recommends already
> >         recommends REST, such as in  RFC 4240 and in
> >         http://tools.ietf.org/html/draft-griffin-bliss-rest-00
> > 
> >             * Web applications from mail to office apps  
> >             * Internet applications previously served by dedicated
> >                network applications protocols
> >             * Many other industries, from banking to newsprint to
> >                travel. And Oh, social networks, they also have
> >                registrars...
> > 
> > 
> > 
> >         Now please keep in mind the WWW architecture uses basically
> >          only 3 components*: The URI for addressing, HTTP 
> for transport
> >         and HTML for  data rendering, augmented by some 
> other languages
> >         (XML) and namespaces. Can we  benefit something from the WWW
> >         architecture? If yes what  specifically?
> > 
> >         Heresy: If the web developers could use HTTP only, 
> what  other
> >         application level protocols do we really need 
> except RTP/UDP?
> >         This  includes in my mind TLS and DTLS for secure 
> transport or
> >         even better IMO: HIP  for many reasons.
> > 
> > 
> >         [Markus] Personally I agree that REST+HTTP+Javascript is a
> >         superior  toolset compared to SIP, XMPP, IMAP and so on to
> >         develop many applications.  However, there are 
> still plenty of
> >         SIP based VoIP and XMPP based IM and  presence 
> deployments out
> >         there and I don't expect the web apps to replace 
> all  of them in
> >         the near future :)
> > 
> >          
> >         * T.V. Raman: "Toward 2W,  Beyond Web 2.0", 
> Communications of
> >         the ACM, February 2009
> > 
> >         Many or most  of these points may be moot or make 
> no sense, but
> >         maybe should be cleared up,  in case there are more 
> naive people
> >         like me that may be interested.
> > 
> >         My  strict personal 2 cents.
> > 
> >         FRANK comments are most  welcome.
> > 
> >         Henry
> > 
> > 
> > 
> >         On 9/4/09 7:09 AM, "Simo.Veikkolainen@nokia.com"
> >         <Simo.Veikkolainen@nokia.com>  wrote:
> > 
> >          
> > 
> >             Hi,
> >              
> >             We would like to propose a new DISPATCH topic 
> on  how SIP
> >             and XMPP can be used together in a 
> complementary  fashion.
> >              
> >             We have been working on a solution which 
> combines SIP  based
> >             VoIP and XMPP based IM and Presence. The 
> requirements and a
> >             proposed  solution is outlined in
> >             
> _http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01_
> >              
> <http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01>
> >              .
> >              
> >             The motivation for this work comes from experience which
> >              shows that most standards based VoIP 
> deployments use SIP.
> >             At the same time,  the Extensible Messaging and Presence
> >             Protocol (XMPP) is widely used in  instant messaging and
> >             presence services. An interesting scenario 
> arises when  a
> >             SIP based voice (and video) service is combined together
> >             with an XMPP  based instant messaging and 
> presence service.
> >              
> >             Note that the  goal in this work is not 
> SIP-XMPP protocol
> >             translation, but to define  protocol extensions and best
> >             practises such that SIP based VoIP and XMPP  
> based IM and
> >             presence could be seamlessly combined and 
> offered to the end
> >              user. For rapid deployment, we assume no changes in the
> >             server  infrastructure, and instead impose new 
> requirements
> >             and protocol extensions  only to the endpoints.
> >              
> >             We would like to get some discussion  going on 
> the use case
> >             itself as well as on the solution. Also, we 
> would like  to
> >             hear you thoughts on DISPATCH or a new 
> "dispatched" mini-WG
> >             as the home  for such work.
> >              
> >             Exact charter proposal and problem statemen is  
> to follow.
> >              
> >             Regards,
> >             Simo and  Markus
> >              
> >              
> > 
> > 
> > 
> > 
> --------------------------------------------------------------
> ----------
> > 
> > _______________________________________________
> > 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 MILANPA@nortel.com  Mon Sep 21 02:34:14 2009
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From: "Milan Patel" <milanpa@nortel.com>
To: "Gonzalo Camarillo" <Gonzalo.Camarillo@ericsson.com>, "DISPATCH" <dispatch@ietf.org>
Subject: Re: [dispatch] Cutoff for charter proposals on Monday
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Hi,

I would like to add an additional topic to the DISPATCH charter - URI
parameters to describe the calling party category and toll class.

Previously, Calling Party Category (CPC) URI parameter was defined in a
draft by Rohan Mahy in the IPTEL group.=20
There is still a need by carriers for this parameter and it is currently
also used within TISPAN and 3GPP.=20
The motivation for defining the CPC URI parameter is to provide a
standardized method of providing information about the calling party and
the station used to originate a call. Currently, a number of proprietary
solutions are being used as a standardized solution is unavailable.=20

My intention is to revise Rohan's draft:
http://tools.ietf.org/html/draft-mahy-iptel-cpc-06
And to take on board some of the comments received on the old IPTEL
list.=20
My proposal is to provide a draft that defines 2 URI parameters: "cpc"
and "oli" indicating the category and tolls class of the calling party,
allowing mapping to parameters used in ISUP.

Best regards,
Milan

Milan Patel
Carrier Networks Core Standards
Nortel
milanpa@nortel.com
Telephone +44 162 843 2381 / ESN 560 2381
Mobile +44 774 053 9261 / ESN 748 9261

For the Companies listed below, The Institute of Chartered Accountants
in England and Wales authorises A R Bloom, S Harris and C Hill to act as
Insolvency Practitioners under section 390(2)(a) of the Insolvency Act
1986 and the Association of Chartered Certified Accountants authorises A
M Hudson to act as an Insolvency Practitioner under section 390(2)(a) of
the Insolvency Act 1986.

The affairs, business and property of the Companies are being managed by
the Joint Administrators, A R Bloom, S Harris, AM Hudson and C Hill who
act as agents of the Companies only and without personal liability.

The Companies are Nortel Networks UK Limited; Nortel Networks SA; Nortel
GmbH; Nortel Networks France SAS; Nortel Networks NV; Nortel Networks
SpA; Nortel Networks BV; Nortel Networks Polska SP Zoo; Nortel Networks
Hispania SA; Nortel Networks (Austria) GmbH; Nortel Networks sro; Nortel
Networks Engineering Service Kft; Nortel Networks Portugal SA; Nortel
Networks Slovensko sro; Nortel Networks Oy; Nortel Networks Romania SRL;
Nortel Networks AB; Nortel Networks International Finance & Holding BV
-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Gonzalo Camarillo
Sent: 16 September 2009 15:13
To: DISPATCH
Subject: [dispatch] Cutoff for charter proposals on Monday

Folks,

this is a reminder of our cutoff for charter proposals, which is on=20
Monday (September 21st). Per our previous email: "A charter proposal=20
must consist of a minimum of a problem statement and at least one=20
milestone/deliverable. This deadline will allow time for consideration=20
of proposals that could potentially be "dispatched" prior to the WG=20
session."

A few of the topics we received arrived after the previous deadline,=20
which was September 7th. We will be considering even those topics that=20
arrived a bit late because we realize the deadline was quite tough to=20
meet (specially for people that returned from their vacation right=20
before the deadline).

We are expecting charter proposals for:

o Third-party authorization
o CBUS
o Session recording
o Identity
o Disaggregated media
o Domain registrations in SIP
o SIP - XMPP
o Alert-info URNs
o Reference to an earlier communication

Let the chairs know if there are topics missing from this list.

Thanks,

Gonzalo
DISPATCH co-chair
_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch

From salvatore.loreto@ericsson.com  Mon Sep 21 03:50:59 2009
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Subject: [dispatch] Charter Proposal for Disaggregated Media
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Hi there,

below is a charter proposal for a WG on Disaggregated Media.

All comments, thoughts, feedbacks are very welcome!

cheers
Sal



Disaggregated Media WG Charter
------------------------------- 
Disaggregated media refers to the ability for a user to create a
multimedia session combining different media streams, coming from
different devices under his or her control, so that they are treated
by the far end of the session as a single media session.

Generally, a given participant uses a single device to establish (or
participate in) a given multimedia session.  Consequently, the SIP
signaling to manage the multimedia session and the actual media
streams are typically co-located in the same device.  In scenarios
involving disaggregated media, a user wants to establish a single
multimedia session combining different media streams coming from
different devices under his or her control.  This creates a need to
coordinate the exchange of the those media streams within the media
session.

The Message Bus (Mbus) [RFC3259] can be used by an user to coordinate
the different devices under his control to involve them in the call.
The different devices can communicate with one another using Mbus 
messages, and then let only one device, a call control engine, to 
initiate, manage and terminate call control relations to other SIP endpoints. 
All the different user's devices need to support the Mbus protocol.

The Megaco [RFC3525] protocol can be used in a disaggregated media 
scenario to let one of the user's devices act as a Media Gateway Controller,
coordinating all the other devices under the user's control, which in this case
will act as Media Gateways. In this case all the different user's devices 
need to support the Megaco protocol.

The SIP protocol provides 3pcc [RFC3725] as a possible mechanism
to coordinate the exchange of media streams coming from different 
devices under the control of the same user, in the case at least 
one among the different devices under his control supports this
mechanism and is able to become a Controller for the other in the call. 


All the existing mechanisms only cover a subset of the interesting scenarios 
that involve disaggregated media. Scenarios not covered by existing mechanisms 
include the loosely-coupled one where none of the nodes can act as a controller
because it does not support the necessary functionality or because it will not
participate in the whole session.


The objective of the proposed working group is to develop a framework
for Disaggregated Media that include both improvements on existing 
mechanisms (e.g. 3pcc) and the develop of one or more new mechanisms
like a loosely coupled mechanism that does not require the implementation 
of complex logic on any of the terminals.


Specifically, the proposed working group will develop the following
deliverables: 
1. A framework document describing key design considerations for Disaggregated 
   mediamechanism.
2. A specification for a loosely couple mechanism.
3. One or more specifications to improve existing mechanism to coordinate
   disaggregated media.



From christer.holmberg@ericsson.com  Mon Sep 21 04:24:48 2009
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Hi,

Below is the proposed charter for CBUS.

Regards,

Christer

-----------------------------------------

CBUS Charter.
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

Introduction:
-------------

Various OMA service enablers need to be able to retrieve a list of
addresses (URIs), where the users associated with the URIs fulfil
certain conditions.  User information is evaluated against the
conditions, and the matches form a URI list.

The need for the functionality originates from the OMA PoC service.
However, there is ongoing work in OMA to define a common mechanism,
called CBUS enabler (OMA-RD-CBUS-V1_0-20090203-C), to provide the
functionality, so that it can be used for various types of services
(e.g. messaging, gaming, conferencing and advertisement).

The CBUS enabler provides the following functions:

- Support for requestor initiated condition-based URIs selection
requests.

- Administration of conditions for URI selection.

- Interaction with other enablers for retrieval of individual user's
  information (e.g. presence information, location information).

- Evaluation of conditions and selection of matching individual users
  based on one time evaluation.

- Re-evaluation of conditions and re-selection of matching individual
  users based on monitoring.

- Aggregation of one-time evaluation results and monitoring results
  from different information sources.

- Notification of evaluation results to requestor.

The conditions can be based on user information which changes over time
(e.g. presence information or geographical location), but they can also
be based on more static user information (e.g. hobbies or personal
interests).

The entity which acts as requester, and provides the conditions to be
used for the user information evaluation, is called CBUS client.  The
CBUS client usually resides on the user equipment, or an application
server, which supports the CBUS enabler.

The selection of URIs, based on the provided conditions, may be
performed by taking a snapshot, i.e. by making a one-time evaluation of
the user information to find out whether the conditions are fulfilled or
not.  The URI selection may also be performed by continous monitoring of
the user information.


Work and deliverables:
----------------------

The purpose of the work is to, based on the OMA CBUS requirements (OMA-
RD-CBUS-V1_0-20090203-C), procude the SIP protocol information elements
which are needed to implement the interface between the CBUS client and
CBUS server, using the SIP SUBSCRIBE/NOTIFY framework (RFC 3265).


The primary task for the work is to:

1. Procude a mechanism for a subscriber (CBUS client) to, when
subscribing to an event package which contains a list of URI, provide
conditions, candidate URIs and evaluation parameters to the notifier
(CBUS server).=20
The mechanism may be used on defining an XML based message body type,
and SIP header field parameters.

2. Produce an event package, or possible re-use an existing event
package, used to the notifier (CBUS server) to send URI lists (based on
the conditions and candidate URIs) to the subscriber (CBUS client).

The CBUS client actions triggered by the received URI list, and the
interactions between the CBUS server and other enablers, are outside the
scope of the work.


The work is planned to take place in, or in co-operaton with, the
SIPCORE WG.


Goals and milestones:
---------------------

Jan 2010	Working Group Last Call for the document which contains
the mechanism for a
            subscriber to provide the needed CBUS information to the
notifier, and
            contains the event package (if a new package is needed) for
the=20
            notifier to provide the URI list to the subscriber.

Mar 2010    Submit document to IESG as Proposed Standard









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<TITLE>Charter proposed: CBUS</TITLE>
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<BODY>
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<BR>

<P><FONT SIZE=3D2 FACE=3D"Courier New">Hi,</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">Below is the proposed charter for =
CBUS.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">Regards,</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">Christer</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier =
New">-----------------------------------------</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">CBUS Charter.</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">Introduction:</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">-------------</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">Various OMA service enablers need =
to be able to retrieve a list of addresses (URIs), where the users =
associated with the URIs fulfil certain conditions.&nbsp; User =
information is evaluated against the conditions, and the matches form a =
URI list.</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">The need for the functionality =
originates from the OMA PoC service.</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">However, there is ongoing work =
in OMA to define a common mechanism, called CBUS enabler =
(OMA-RD-CBUS-V1_0-20090203-C), to provide the functionality, so that it =
can be used for various types of services (e.g. messaging, gaming, =
conferencing and advertisement).</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">The CBUS enabler provides the =
following functions:</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">- Support for requestor initiated =
condition-based URIs selection requests.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">- Administration of conditions =
for URI selection.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">- Interaction with other enablers =
for retrieval of individual user's</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp; information (e.g. =
presence information, location information).</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">- Evaluation of conditions and =
selection of matching individual users</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp; based on one time =
evaluation.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">- Re-evaluation of conditions and =
re-selection of matching individual</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp; users based on =
monitoring.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">- Aggregation of one-time =
evaluation results and monitoring results</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">&nbsp; from different =
information sources.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">- Notification of evaluation =
results to requestor.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">The conditions can be based on =
user information which changes over time (e.g. presence information or =
geographical location), but they can also be based on more static user =
information (e.g. hobbies or personal interests).</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">The entity which acts as =
requester, and provides the conditions to be used for the user =
information evaluation, is called CBUS client.&nbsp; The CBUS client =
usually resides on the user equipment, or an application server, which =
supports the CBUS enabler.</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">The selection of URIs, based on =
the provided conditions, may be performed by taking a snapshot, i.e. by =
making a one-time evaluation of the user information to find out whether =
the conditions are fulfilled or not.&nbsp; The URI selection may also be =
performed by continous monitoring of the user information.</FONT></P>
<BR>

<P><FONT SIZE=3D2 FACE=3D"Courier New">Work and deliverables:</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">----------------------</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">The purpose of the work is to, =
based on the OMA CBUS requirements (OMA- RD-CBUS-V1_0-20090203-C), =
procude the SIP protocol information elements which are needed to =
implement the interface between the CBUS client and CBUS server, using =
the SIP SUBSCRIBE/NOTIFY framework (RFC 3265).</FONT></P>
<BR>

<P><FONT SIZE=3D2 FACE=3D"Courier New">The primary task for the work is =
to:</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">1. Procude a mechanism for a =
subscriber (CBUS client) to, when subscribing to an event package which =
contains a list of URI, provide conditions, candidate URIs and =
evaluation parameters to the notifier (CBUS server). </FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">The mechanism may be used on =
defining an XML based message body type, and SIP header field =
parameters.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">2. Produce an event package, or =
possible re-use an existing event package, used to the notifier (CBUS =
server) to send URI lists (based on the conditions and candidate URIs) =
to the subscriber (CBUS client).</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">The CBUS client actions triggered =
by the received URI list, and the interactions between the CBUS server =
and other enablers, are outside the scope of the work.</FONT></P>
<BR>

<P><FONT SIZE=3D2 FACE=3D"Courier New">The work is planned to take place =
in, or in co-operaton with, the SIPCORE WG.</FONT>
</P>
<BR>

<P><FONT SIZE=3D2 FACE=3D"Courier New">Goals and milestones:</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">---------------------</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">Jan =
2010&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Working Group Last Call =
for the document which contains the mechanism for a</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
subscriber to provide the needed CBUS information to the notifier, =
and</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
contains the event package (if a new package is needed) for the </FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
notifier to provide the URI list to the subscriber.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">Mar 2010&nbsp;&nbsp;&nbsp; Submit =
document to IESG as Proposed Standard</FONT>
</P>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>

</BODY>
</HTML>
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From Markus.Isomaki@nokia.com  Mon Sep 21 04:42:27 2009
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From: <Markus.Isomaki@nokia.com>
To: <dwing@cisco.com>, <pkyzivat@cisco.com>, <hsinnrei@adobe.com>
Date: Mon, 21 Sep 2009 13:43:07 +0200
Thread-Topic: [dispatch] New topic proposal for DISPATCH
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Cc: Simo.Veikkolainen@nokia.com, dispatch@ietf.org, gonzalo.camarillo@ericsson.com
Subject: Re: [dispatch] New topic proposal for DISPATCH
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Hi Dan,=20

Dan Wing wrote:
>
>Paul Kyzivat wrote:
>>=20
>> I agree that there are pragmatic reasons to pursue the=20
>> coordination/coexistence of sip and xmpp. And pushing responsibility=20
>> to the endpoints remains desirable.
>>=20
>> One thing that remains of concern to me in such work is that=20
>we end up=20
>> with disjoint address spaces without any straightforward way to=20
>> correlate them. So if I want a single "session" involving both voice=20
>> via SIP and IM via xmpp, then I have great difficulty in discovering=20
>> URIs for the "subordinate" session that correlate with those for the=20
>> "primary" session. (No pejorative intended here - one or the other=20
>> must go first.)
>
>So XMPP has to use E.164's (like SIP deployments do), or SIP=20
>will have to switch to user@host.  I don't forsee either happening. =20
>If this is deemed necessary then we will have to map between them
>-- somehow.  Send an XMPP message asking "what is your phone=20
>number" (in a computer-parsable manner), perhaps?
>
>-d
>

I agree. It will be imposible to require a direct or algorithmic mapping be=
tween a SIP URI (in practice most cases a E.164 number as you point out) an=
d a corresponding XMPP URI. However, if a user knows one of them for anothe=
r user, it should be possible to discover the other. In our draft (http://t=
ools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01) we propose a few=
 requirements and solution possibilities. It should be possible to discover=
 "permanent" URIs (SIP AoR, XMPP JID) as well as dynamic ones that only hav=
e meaning in a given context: for instance, if I'm having an IM chat with y=
ou with XMPP, I should be able to tell you a SIP URI you can send INVITE to=
 if you want to add a voice call specifically to that conversation context.=
 It should be enough to understand the semantics of these URIs in the endpo=
ints only, i.e. the infrastructure should not need to know anything about t=
hem beyond "routing". In SIP we could utilize GRUUs if they were available =
(in practice we can't rely on them however), in XMPP we already have the fu=
ll JIDs (xmpp:user@domain/resource) at our disposal.

Markus


From Simo.Veikkolainen@nokia.com  Mon Sep 21 05:27:53 2009
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From: <Simo.Veikkolainen@nokia.com>
To: <dispatch@ietf.org>
Date: Mon, 21 Sep 2009 14:28:43 +0200
Thread-Topic: Charter proposal on combining SIP voice with XMPP IM and presence
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Subject: [dispatch] Charter proposal on combining SIP voice with XMPP IM and presence
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Hello,

Below is the draft charter text for our proposal on combining SIP voice wit=
h XMPP IM and presence.

Comments are very welcome!

Regards,
Simo


-----------------


Combining SIP VoIP and XMPP IM/Presence
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

Currently most standards-based Voice over IP (VoIP) deployments use the Ses=
sion Initiation Protocol (SIP).  In addition to providing basic voice servi=
ce SIP has an extensive support for more advanced telephony features includ=
ing interworking with the traditional Public Switched Telephone Network (PS=
TN).  SIP is also gaining popularity in the field of video communication. A=
t the same time, the Extensible Messaging and Presence Protocol (XMPP) is e=
njoying widespread usage in instant messaging and presence services. =20

Both SIP and XMPP offer extensions for voice as well as IM and presence (XM=
PP voice via the Jingle extension, and SIP IM/presence via SIMPLE protocols=
). However, widespread deployment of these extensions has not so far taken =
place. In order to speed up deployment of integrated VoIP and IM systems, S=
IP based voice and XMPP based IM/Presence could be combined in endpoints to=
 offer a voice, IM and presence service without any changes to existing SIP=
 and XMPP service infrastrucure.=20

The objective of this Working Group is to develop solutions for combining S=
IP based voice with XMPP based IM and Presence such that a unified user exp=
erience can be offered to end user. Specifically, solutions are needed on=20
- addressing; end users should be able to initiate sessions to a user ident=
ity in either SIP or XMPP domain. The corresponding user identity in the ot=
her protocol domain needs to be learned automatically.
- session correlation; endpoints need to be able to correlate voice session=
s with IM/Presence such that they can be presented to the end user in a sea=
mless fashion
- presence; it should be possible to change the XMPP presence status based =
on the user's activity in the SIP domain.

The goal is to rely on existing service infrastructre, and new requirements=
 should be imposed only to the endpoint.=20

Protocol interworking, that is, translation from one protocol to another, i=
s out of scope of this WG.

Milestones
Feb 2010 Problem statement and use cases submitted to IESG as Informational
Dec 2010 Specification on combining SIP based voice and XMPP based IM/Prese=
nce in a seamless manner submitted to IESG as Proposed Standard.

-----------------

From hsinnrei@adobe.com  Mon Sep 21 06:20:34 2009
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On 9/20/09 10:20 PM, "Dan Wing" <dwing@Cisco.COM> wrote:

>So XMPP has to use E.164's (like SIP deployments do), or SIP
>will have to switch to user@host.  I don't forsee either happening.
>If this is deemed necessary then we will have to map between them
>-- somehow.  Send an XMPP message asking "what is your phone
>number" (in a computer-parsable manner), perhaps?

Now why could not the application query the address book to get both the E.=
164 addresses and user@host addresses?

This is not only of interest to na=EFve folks like me, but also illustrates=
 how some key functions are better done in the applications in the endpoint=
s rather than in the network application protocols.
The "old" e2e principle illustrated...

Henry


--_000_C6DCE8015E91hsinnreiadobecom_
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<HTML>
<HEAD>
<TITLE>Re: [dispatch] New topic proposal for DISPATCH</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
18pt'>On 9/20/09 10:20 PM, &quot;Dan Wing&quot; &lt;<a href=3D"dwing@Cisco.=
COM">dwing@Cisco.COM</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:18pt'>&gt;So XMPP has to use E.164's (like SIP de=
ployments do), or SIP<BR>
&gt;will have to switch to user@host. &nbsp;I don't forsee either happening=
. <BR>
&gt;If this is deemed necessary then we will have to map between them<BR>
&gt;-- somehow. &nbsp;Send an XMPP message asking &quot;what is your phone<=
BR>
&gt;number&quot; (in a computer-parsable manner), perhaps?<BR>
<BR>
Now why could not the application query the address book to get both the E.=
164 addresses and user@host addresses? <BR>
<BR>
This is not only of interest to na&iuml;ve folks like me, but also illustra=
tes how some key functions are better done in the applications in the endpo=
ints rather than in the network application protocols. <BR>
The &#8220;old&#8221; e2e principle illustrated...<BR>
<BR>
Henry<BR>
<BR>
</SPAN></FONT></BLOCKQUOTE>
</BODY>
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Subject: Re: [dispatch] New topic proposal for DISPATCH
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inline

Dan Wing wrote:
>  
> 
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org 
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat
>> Sent: Friday, September 11, 2009 12:56 PM
>> To: Henry Sinnreich
>> Cc: dispatch@ietf.org; avshalom@il.ibm.com; 
>> gonzalo.camarillo@ericsson.com; Markus.Isomaki@nokia.com; 
>> Simo.Veikkolainen@nokia.com
>> Subject: Re: [dispatch] New topic proposal for DISPATCH
>>
>> I agree that there are pragmatic reasons to pursue the 
>> coordination/coexistence of sip and xmpp. And pushing 
>> responsibility to 
>> the endpoints remains desirable.
>>
>> One thing that remains of concern to me in such work is that 
>> we end up 
>> with disjoint address spaces without any straightforward way to 
>> correlate them. So if I want a single "session" involving 
>> both voice via 
>> SIP and IM via xmpp, then I have great difficulty in discovering URIs 
>> for the "subordinate" session that correlate with those for the 
>> "primary" session. (No pejorative intended here - one or the 
>> other must go first.)
> 
> So XMPP has to use E.164's (like SIP deployments do), or SIP
> will have to switch to user@host.  I don't forsee either happening.  
> If this is deemed necessary then we will have to map between them 
> -- somehow.  Send an XMPP message asking "what is your phone
> number" (in a computer-parsable manner), perhaps?

Well, there is nothing to prevent XMPP from using E.164+domain name (the 
way most of sip deployments do since nobody seems to like tel:.)

BUT, as things currently stand there is no guarantee that 
sip:alice@atlanta.com and xmpp:alice@atlanta.com refer to the same 
persona. While it would often be a reasonable leap of faith to assume 
they do, the namespaces are typically administered independently.

Having *some* automated way of determining an association seems 
necessary to a good integration. Of course that is exactly what a 
session initiation protocol is all about. Its also something that a 
presence protocol should also be able to do.

	Thanks,
	Paul

> -d
> 
> 
>> Somehow this problem needs to be resolved.
>>
>> 	Thanks,
>> 	Paul
>>
>> Henry Sinnreich wrote:
>>> Hi Markus,
>>>
>>> Thanks for your thoughtful reply and I agree there are some 
>> scenarios 
>>> where it makes sense.
>>> More important, doing this in the endpoints is the right 
>> approach IMO 
>>> and as you know, I will always defend the Internet 
>> end-to-end principle :-)
>>> This distributed approach is also IMO a good answer to the 
>> Interdomain 
>>> Scaling problem for IM
>>> ( I-D.draft-ietf-simple-interdomain-scaling-analysis) and I 
>> have copied 
>>> one of its authors here.
>>>
>>> This raises the questions:
>>>
>>>    1. Why stop at SIP and XMPP, and not add other popular 
>> IM protocols,
>>>       as is already done by quite a few products in the market?
>>>    2. Can your approach be generalized in a more generic way?
>>>
>>>
>>> If endpoints are accommodating several IM protocols, an 
>> IETF, Dispatch 
>>> recommended GENERIC approach would make many people feel more 
>>> comfortable about it. Your I-D could be used as a tested 
>> example for a 
>>> generic approach.
>>>
>>> Ranting on preferring endpoint applications to more 
>> application protocols
>>> I still believe creative application developers face the choice of 
>>> either inventing new applications or dedicating their time 
>> to learning 
>>> application protocols and they cannot do both. Learning and 
>> following 
>>> both SIP and XMPP to write and update code is quite a challenge.
>>> Is there a better way?
>>> For this reason, one single application protocol for IM and 
>> voice is enough.
>>> Proof: Almost all the applications that make people buy 
>> smart phones, 
>>> use online office apps and enterprise apps run on HTTP(S) and their 
>>> application developers don't have to worry about knowing 
>> how HTTP works. 
>>> As a result, HTTP has been also used to even carry 
>> streaming media, web 
>>> conferencing and IM. Just to avoid spending a lifetime to learn and 
>>> follow application protocols. But HTTP based apps dominate the 
>>> application space. This is however for another discussion elsewhere.
>>>
>>> My strictly personal 2 cents.
>>>
>>> Thanks again,
>>>
>>> Henry
>>>
>>>
>>> On 9/11/09 8:11 AM, "Markus.Isomaki@nokia.com" 
>>> <Markus.Isomaki@nokia.com> wrote:
>>>
>>>     Hi Henry,
>>>
>>>     Thanks for your feedback :) Let me answer to the 
>> excellent points
>>>     you raise.
>>>
>>>     First of all, I would encourage you and everyone to 
>> carefully read
>>>     the Introduction chapter of the draft
>>>     
>> (http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01
>>>     
>> <http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01>
>>>     ) to be clear on what the proposal is about. That is, we are not
>>>     trying to boil the ocean and define all possible ways 
>> how SIP and
>>>     XMPP infrastructures could be combined, but instead try 
>> to define a
>>>     minimalistic approach whereby a client can use both SIP 
>> and XMPP in
>>>     an integrated manner WITHOUT requiring ANY changes on 
>> the servers
>>>     that are already deployed.
>>>
>>>     See further comments below with [Markus]:
>>>
>>>
>>>          
>>>         
>> --------------------------------------------------------------
>> ----------
>>>         *From:* ext Henry Sinnreich  [mailto:hsinnrei@adobe.com]
>>>         *Sent:* 04 September, 2009  18:42
>>>         *To:* Veikkolainen Simo (Nokia-D/Helsinki); 
>> dispatch@ietf.org;
>>>          mary.barnes@nortel.com; gonzalo.camarillo@ericsson.com
>>>         *Cc:* Isomaki  Markus (Nokia-CIC/Espoo); Olle E. Johansson
>>>         *Subject:* Re: [dispatch]  New topic proposal for DISPATCH
>>>
>>>          
>>>         I would like to second such work, but a far deeper  
>> look under
>>>         the hood is necessary, so we don't do another 
>> Elbonia  project.
>>>         http://dilbert.com/strips/comic/2009-09-01/
>>>
>>>         Here  are some not very mature thoughts on the 
>> scope of the work.
>>>         The most  interesting, ROI estimate and REST architecture
>>>         support are at the  end.
>>>
>>>         Servers
>>>
>>>         Fundamentally, we need a registrar, a proxy and a  presence
>>>         server: Three in total
>>>
>>>             * What will be the duplication of servers and where
>>>                specifically?
>>>             * How does the SIP registrar communicate with the XMPP
>>>                server(s)?
>>>             * What may be the routing protocol(s) inside a  SIP+XMMP
>>>               network? Think of IMS as a stressing use  case.
>>>
>>>
>>>         [Markus] Our starting point is that we already have 
>> quite  many
>>>         SIP VoIP deployments out there, as long as quite many XMPP
>>>         presence &  IM deployments. So, the servers you 
>> mention already
>>>         exist. If one now  wants to develop a client that 
>> does VoIP, IM
>>>         and presence together, it should  be possible to 
>> utilize these
>>>         existing infrastrcutures and not have to wait for  
>> new ones to
>>>         be deployed (as for instance we have waited quite 
>> many years for
>>>          SIMPLE-based Presence servers to get deployed but 
>> compared to
>>>         XMPP there's  very little in use). The idea is to define a
>>>         couple of E2E (as in  nothing that SIP proxies or 
>> XMPP servers
>>>         would have to support) protocol  extensions which 
>> would allow
>>>         the client to nicely combine SIP VoIP and XMPP  
>> presence & IM.
>>>          
>>>          
>>>         There is no need for SIP and XMPP servers to know 
>> about  each
>>>         other or route protocol messages between each other. All the
>>>         logic is put  into the endpoints. No need to worry about IMS
>>>         either, except  that in theory the client can use 
>> IMS for its
>>>         VoIP service  (similar to any other SIP  infra).
>>>
>>>
>>>         User  Agents
>>>
>>>             * Same as with servers: How many and what 
>> protocol  RFCs are
>>>               there in the combination?
>>>             * Which of the "services" can be placed in the  
>> applications
>>>               in the endpoints or in the network 
>> application protocols,
>>>               such  as SIP and XMPP. This is the topic on 
>> infrastructure
>>>               vs. endpoint  applications.
>>>
>>>
>>>         [Markus] Just take any current SIP VoIP  UA 
>> implementation, XMPP
>>>         client implementation and glue them together  and 
>> you're already
>>>         pretty far. Add a couple of extensions as proposed and it's
>>>          done. From there onwards you have a pretty 
>> powerful machinery
>>>         to add  applications by using SIP session setup and 
>> XMPP message
>>>         delivery  capabilities.
>>>
>>>
>>>         Application Protocol  Stacks
>>>
>>>             * How many protocol stacks does this involve 
>> and more  to
>>>               the point, how many RFCs?
>>>             * For SIP alone, see http://rfc3261.net/. What 
>> will be the
>>>               total  combined RFC count?
>>>
>>>
>>>         [Markus] All what we expect is already pretty 
>> widely  supported
>>>         and even available as open source: basic SIP VoIP 
>> and XMPP IM
>>>         and  presence is enough to start with.   
>>>
>>>
>>>         NAT  Traversal
>>>
>>>             * Will STUN/TURN/ICE work the same for SIP and 
>> for  XMPP?
>>>
>>>         [Markus] In our proposal SIP would be used for 
>> setting up  voice
>>>         & video sessions. So, you would need NAT traversal 
>> for that. How
>>>          that's done is outside the scope but either B2BUA 
>> or ICE based
>>>         mechanisms  would work as far as they work in 
>> general. For XMPP
>>>         we would not need to worry  about NAT traversal 
>> beyond how its
>>>         done in that protocol by default. For a  client there is a
>>>         caveat that it has to keepalive the TCP connections 
>> with  BOTH
>>>         SIP and XMPP servers so it would be recommended to 
>> synchronize
>>>         those  (send them at the same time) for instance in 
>> a battery
>>>         powered  device.
>>>
>>>
>>>         Return  on Investment (ROI): Cost and effort by 
>> developers. More
>>>         network  infrastructure, not even counting B2B NEs.
>>>
>>>             * What is the additional time/money required 
>> for  developers
>>>               that know SIP to also learn XMPP and vice 
>> versa? This is
>>>               all  about return on investment.
>>>
>>>
>>>         [Markus] This is a problem for sure, even more so  for the
>>>         people writing standard specs than code :)  Hopefully the
>>>          developer would just have add a couple of simple 
>> extensions to
>>>         existing  SIP and XMPP implementations.
>>>
>>>
>>>         Support for a REST-full  architecture
>>>
>>>         The WWW has a REST architecture was formulated in 
>> 2000,  after
>>>         the emergence of SIP and XMMP.  As a result of its 
>> architecture,
>>>          the WWW has had a transformational effect on the following
>>>         (this is not a  complete list). Some SIP work 
>> recommends already
>>>         recommends REST, such as in  RFC 4240 and in
>>>         http://tools.ietf.org/html/draft-griffin-bliss-rest-00
>>>
>>>             * Web applications from mail to office apps  
>>>             * Internet applications previously served by dedicated
>>>                network applications protocols
>>>             * Many other industries, from banking to newsprint to
>>>                travel. And Oh, social networks, they also have
>>>                registrars...
>>>
>>>
>>>
>>>         Now please keep in mind the WWW architecture uses basically
>>>          only 3 components*: The URI for addressing, HTTP 
>> for transport
>>>         and HTML for  data rendering, augmented by some 
>> other languages
>>>         (XML) and namespaces. Can we  benefit something from the WWW
>>>         architecture? If yes what  specifically?
>>>
>>>         Heresy: If the web developers could use HTTP only, 
>> what  other
>>>         application level protocols do we really need 
>> except RTP/UDP?
>>>         This  includes in my mind TLS and DTLS for secure 
>> transport or
>>>         even better IMO: HIP  for many reasons.
>>>
>>>
>>>         [Markus] Personally I agree that REST+HTTP+Javascript is a
>>>         superior  toolset compared to SIP, XMPP, IMAP and so on to
>>>         develop many applications.  However, there are 
>> still plenty of
>>>         SIP based VoIP and XMPP based IM and  presence 
>> deployments out
>>>         there and I don't expect the web apps to replace 
>> all  of them in
>>>         the near future :)
>>>
>>>          
>>>         * T.V. Raman: "Toward 2W,  Beyond Web 2.0", 
>> Communications of
>>>         the ACM, February 2009
>>>
>>>         Many or most  of these points may be moot or make 
>> no sense, but
>>>         maybe should be cleared up,  in case there are more 
>> naive people
>>>         like me that may be interested.
>>>
>>>         My  strict personal 2 cents.
>>>
>>>         FRANK comments are most  welcome.
>>>
>>>         Henry
>>>
>>>
>>>
>>>         On 9/4/09 7:09 AM, "Simo.Veikkolainen@nokia.com"
>>>         <Simo.Veikkolainen@nokia.com>  wrote:
>>>
>>>          
>>>
>>>             Hi,
>>>              
>>>             We would like to propose a new DISPATCH topic 
>> on  how SIP
>>>             and XMPP can be used together in a 
>> complementary  fashion.
>>>              
>>>             We have been working on a solution which 
>> combines SIP  based
>>>             VoIP and XMPP based IM and Presence. The 
>> requirements and a
>>>             proposed  solution is outlined in
>>>             
>> _http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01_
>>>              
>> <http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01>
>>>              .
>>>              
>>>             The motivation for this work comes from experience which
>>>              shows that most standards based VoIP 
>> deployments use SIP.
>>>             At the same time,  the Extensible Messaging and Presence
>>>             Protocol (XMPP) is widely used in  instant messaging and
>>>             presence services. An interesting scenario 
>> arises when  a
>>>             SIP based voice (and video) service is combined together
>>>             with an XMPP  based instant messaging and 
>> presence service.
>>>              
>>>             Note that the  goal in this work is not 
>> SIP-XMPP protocol
>>>             translation, but to define  protocol extensions and best
>>>             practises such that SIP based VoIP and XMPP  
>> based IM and
>>>             presence could be seamlessly combined and 
>> offered to the end
>>>              user. For rapid deployment, we assume no changes in the
>>>             server  infrastructure, and instead impose new 
>> requirements
>>>             and protocol extensions  only to the endpoints.
>>>              
>>>             We would like to get some discussion  going on 
>> the use case
>>>             itself as well as on the solution. Also, we 
>> would like  to
>>>             hear you thoughts on DISPATCH or a new 
>> "dispatched" mini-WG
>>>             as the home  for such work.
>>>              
>>>             Exact charter proposal and problem statemen is  
>> to follow.
>>>              
>>>             Regards,
>>>             Simo and  Markus
>>>              
>>>              
>>>
>>>
>>>
>>>
>> --------------------------------------------------------------
>> ----------
>>> _______________________________________________
>>> 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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Subject: [dispatch] Charter proposal: Session Recording Protocol
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Hi: The following charter proposal is a continuation of the
work presented on the same topic (SIP Session Recording)
at the Stockholm IETF.

Regards,

The Session Recording Protocol (SRP) working group is chartered to 
define a SIP-based protocol for controlling a session (media) recorder.

Session recording is a critical requirement in many business 
communications environments such as call centers and financial trading 
floors.  In some of these environments, all calls must be recorded for 
regulatory and compliance reasons.  In others, calls may be recorded for 
quality control, business analytics, or consumer protection.  Recording 
is typically done by sending a copy of the media to the recording 
devices.  The working group will produce a specification for a protocol 
that will manage delivery of media from an end-point that originates 
media or that has access to it to a recording device. PBX and recording 
vendors today implement proprietary, incompatible mechanisms to 
facilitate recording. A standard protocol will reduce the complexity and 
cost of providing such recording services.

The Session Recording problem presents certain unique requirements that 
are not addressed in the current SIP protocol specification. These 
include requirements such as the need for a distinction between the 
session that is being recorded versus the session that has been 
established for recording.

The SRP Working Group will thoroughly identify use cases, provide 
example system architectures and deployment scenarios, and define 
requirements.

The scope of the activity includes:

  * Recorder Control
  * Session metadata content and format
  * Security mechanisms, including transport and media encryption
  * Negotiation of recording media streams

Privacy and security of conversations are significant concerns. The 
group will define these issues and rationalize with IETF standards and 
practices. This includes privacy concerns (including RAVEN), encryption, 
NAT traversal, SIP-enabled firewalls, authorization, and security.

The group will produce:

  * Updated Requirements, Use Cases, Architecture draft
  * Specification for Session Recording Protocol

Timeline:

   Apr 2010   Requirements, Use Cases, Architecture to IESG as 
Informational Draft

   Nov 2010   Submit protocol draft to IESG as standards track

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/

From pkyzivat@cisco.com  Mon Sep 21 08:31:13 2009
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From: Paul Kyzivat <pkyzivat@cisco.com>
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Cc: IETF Dispatch <dispatch@ietf.org>, Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>
Subject: Re: [dispatch] Charter Proposal for Disaggregated Media
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I agree with the utility of such work. Comments inline.

Salvatore Loreto wrote:
> Hi there,
> 
> below is a charter proposal for a WG on Disaggregated Media.
> 
> All comments, thoughts, feedbacks are very welcome!
> 
> cheers
> Sal
> 
> 
> 
> Disaggregated Media WG Charter
> ------------------------------- Disaggregated media refers to the 
> ability for a user to create a
> multimedia session combining different media streams, coming from
> different devices under his or her control, so that they are treated
> by the far end of the session as a single media session.
> 
> Generally, a given participant uses a single device to establish (or
> participate in) a given multimedia session.  Consequently, the SIP
> signaling to manage the multimedia session and the actual media
> streams are typically co-located in the same device.  In scenarios
> involving disaggregated media, a user wants to establish a single
> multimedia session combining different media streams coming from
> different devices under his or her control.  This creates a need to
> coordinate the exchange of the those media streams within the media
> session.
> 
> The Message Bus (Mbus) [RFC3259] can be used by an user to coordinate
> the different devices under his control to involve them in the call.
> The different devices can communicate with one another using Mbus 
> messages, and then let only one device, a call control engine, to 
> initiate, manage and terminate call control relations to other SIP 
> endpoints. All the different user's devices need to support the Mbus 
> protocol.
> 
> The Megaco [RFC3525] protocol can be used in a disaggregated media 
> scenario to let one of the user's devices act as a Media Gateway 
> Controller,
> coordinating all the other devices under the user's control, which in 
> this case
> will act as Media Gateways. In this case all the different user's 
> devices need to support the Megaco protocol.
> 
> The SIP protocol provides 3pcc [RFC3725] as a possible mechanism
> to coordinate the exchange of media streams coming from different 
> devices under the control of the same user, in the case at least one 
> among the different devices under his control supports this
> mechanism and is able to become a Controller for the other in the call.
> 
> All the existing mechanisms only cover a subset of the interesting 
> scenarios that involve disaggregated media. Scenarios not covered by 
> existing mechanisms include the loosely-coupled one where none of the 
> nodes can act as a controller
> because it does not support the necessary functionality or because it 
> will not participate in the whole session.

I don't understand the above. The disaggregated multimedia session will 
not be established unless *something* takes the initiative to establish 
it. IMO that thing, whatever it is, is acting as a "controller".

The mechanism you have proposed before distributes the "controller" 
functionality across multiple nodes. There is something that causes a 
new stream to be established as a separate sip session and causes that 
to announce correlation with some other session. And then there is are 
the devices at the far end, that maintain the correlation and do 
required maintenance of the correlation as changes happen.

Can you provide more definition of what you mean by "loosely coupled" 
while recognizing that something needs to know enough to do the 
initiation and maintenance of the correlation?

> The objective of the proposed working group is to develop a framework
> for Disaggregated Media that include both improvements on existing 
> mechanisms (e.g. 3pcc) and the develop of one or more new mechanisms
> like a loosely coupled mechanism that does not require the 
> implementation of complex logic on any of the terminals.

I would argue that it is premature to assert that new mechanisms are 
required.

> Specifically, the proposed working group will develop the following
> deliverables: 1. A framework document describing key design 
> considerations for Disaggregated   mediamechanism.

> 2. A specification for a loosely couple mechanism.

> 3. One or more specifications to improve existing mechanism to coordinate
>   disaggregated media.

Are both (2) and (3) *specifications*? Or is (2) a requirements document 
that is then satisfied by (3)?

If (2) is requirements, then this makes sense to me, though it might be 
possible to combine that with (1). If both (2) and (3) are mechanism 
specs then I don't get it.

	Thanks,
	Paul

From Even.roni@huawei.com  Mon Sep 21 09:25:14 2009
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Date: Mon, 21 Sep 2009 19:23:44 +0300
From: Roni Even <Even.roni@huawei.com>
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Cc: 'Gonzalo Camarillo' <gonzalo.camarillo@ericsson.com>
Subject: Re: [dispatch] Charter Proposal for Disaggregated Media and combining SIP and XMPP
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Hi,
What is the relation between this proposal and the SIP and XMPP proposal.
To me it looks like both are trying to combine two separate call legs to be
as one and have similar requirements.

Roni Even

> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Salvatore Loreto
> Sent: Monday, September 21, 2009 1:52 PM
> To: IETF Dispatch
> Cc: Gonzalo Camarillo
> Subject: [dispatch] Charter Proposal for Disaggregated Media
> 
> Hi there,
> 
> below is a charter proposal for a WG on Disaggregated Media.
> 
> All comments, thoughts, feedbacks are very welcome!
> 
> cheers
> Sal
> 
> 
> 
> Disaggregated Media WG Charter
> -------------------------------
> Disaggregated media refers to the ability for a user to create a
> multimedia session combining different media streams, coming from
> different devices under his or her control, so that they are treated
> by the far end of the session as a single media session.
> 
> Generally, a given participant uses a single device to establish (or
> participate in) a given multimedia session.  Consequently, the SIP
> signaling to manage the multimedia session and the actual media
> streams are typically co-located in the same device.  In scenarios
> involving disaggregated media, a user wants to establish a single
> multimedia session combining different media streams coming from
> different devices under his or her control.  This creates a need to
> coordinate the exchange of the those media streams within the media
> session.
> 
> The Message Bus (Mbus) [RFC3259] can be used by an user to coordinate
> the different devices under his control to involve them in the call.
> The different devices can communicate with one another using Mbus
> messages, and then let only one device, a call control engine, to
> initiate, manage and terminate call control relations to other SIP
> endpoints.
> All the different user's devices need to support the Mbus protocol.
> 
> The Megaco [RFC3525] protocol can be used in a disaggregated media
> scenario to let one of the user's devices act as a Media Gateway
> Controller,
> coordinating all the other devices under the user's control, which in
> this case
> will act as Media Gateways. In this case all the different user's
> devices
> need to support the Megaco protocol.
> 
> The SIP protocol provides 3pcc [RFC3725] as a possible mechanism
> to coordinate the exchange of media streams coming from different
> devices under the control of the same user, in the case at least
> one among the different devices under his control supports this
> mechanism and is able to become a Controller for the other in the call.
> 
> 
> All the existing mechanisms only cover a subset of the interesting
> scenarios
> that involve disaggregated media. Scenarios not covered by existing
> mechanisms
> include the loosely-coupled one where none of the nodes can act as a
> controller
> because it does not support the necessary functionality or because it
> will not
> participate in the whole session.
> 
> 
> The objective of the proposed working group is to develop a framework
> for Disaggregated Media that include both improvements on existing
> mechanisms (e.g. 3pcc) and the develop of one or more new mechanisms
> like a loosely coupled mechanism that does not require the
> implementation
> of complex logic on any of the terminals.
> 
> 
> Specifically, the proposed working group will develop the following
> deliverables:
> 1. A framework document describing key design considerations for
> Disaggregated
>    mediamechanism.
> 2. A specification for a loosely couple mechanism.
> 3. One or more specifications to improve existing mechanism to
> coordinate
>    disaggregated media.
> 
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


From salvatore.loreto@ericsson.com  Mon Sep 21 09:39:14 2009
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Date: Mon, 21 Sep 2009 19:38:59 +0300
From: Salvatore Loreto <salvatore.loreto@ericsson.com>
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Cc: 'IETF Dispatch' <dispatch@ietf.org>, 'Gonzalo Camarillo' <gonzalo.camarillo@ericsson.com>
Subject: Re: [dispatch] Charter Proposal for Disaggregated Media and combining SIP and XMPP
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Hi Roni,

actually there isn't any relation between this two proposals, each of 
them has started from a different use case.

however if you consider XMPP as a XML streaming, then combining SIP and 
XMPP can be considered as particular case where
"a user wants to establish a single multimedia session combining 
different media streams coming from
different devices under his or her control."

/Sal

Roni Even wrote:
> Hi,
> What is the relation between this proposal and the SIP and XMPP proposal.
> To me it looks like both are trying to combine two separate call legs to be
> as one and have similar requirements.
>
> Roni Even
>
>   
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>> Behalf Of Salvatore Loreto
>> Sent: Monday, September 21, 2009 1:52 PM
>> To: IETF Dispatch
>> Cc: Gonzalo Camarillo
>> Subject: [dispatch] Charter Proposal for Disaggregated Media
>>
>> Hi there,
>>
>> below is a charter proposal for a WG on Disaggregated Media.
>>
>> All comments, thoughts, feedbacks are very welcome!
>>
>> cheers
>> Sal
>>
>>
>>
>> Disaggregated Media WG Charter
>> -------------------------------
>> Disaggregated media refers to the ability for a user to create a
>> multimedia session combining different media streams, coming from
>> different devices under his or her control, so that they are treated
>> by the far end of the session as a single media session.
>>
>> Generally, a given participant uses a single device to establish (or
>> participate in) a given multimedia session.  Consequently, the SIP
>> signaling to manage the multimedia session and the actual media
>> streams are typically co-located in the same device.  In scenarios
>> involving disaggregated media, a user wants to establish a single
>> multimedia session combining different media streams coming from
>> different devices under his or her control.  This creates a need to
>> coordinate the exchange of the those media streams within the media
>> session.
>>
>> The Message Bus (Mbus) [RFC3259] can be used by an user to coordinate
>> the different devices under his control to involve them in the call.
>> The different devices can communicate with one another using Mbus
>> messages, and then let only one device, a call control engine, to
>> initiate, manage and terminate call control relations to other SIP
>> endpoints.
>> All the different user's devices need to support the Mbus protocol.
>>
>> The Megaco [RFC3525] protocol can be used in a disaggregated media
>> scenario to let one of the user's devices act as a Media Gateway
>> Controller,
>> coordinating all the other devices under the user's control, which in
>> this case
>> will act as Media Gateways. In this case all the different user's
>> devices
>> need to support the Megaco protocol.
>>
>> The SIP protocol provides 3pcc [RFC3725] as a possible mechanism
>> to coordinate the exchange of media streams coming from different
>> devices under the control of the same user, in the case at least
>> one among the different devices under his control supports this
>> mechanism and is able to become a Controller for the other in the call.
>>
>>
>> All the existing mechanisms only cover a subset of the interesting
>> scenarios
>> that involve disaggregated media. Scenarios not covered by existing
>> mechanisms
>> include the loosely-coupled one where none of the nodes can act as a
>> controller
>> because it does not support the necessary functionality or because it
>> will not
>> participate in the whole session.
>>
>>
>> The objective of the proposed working group is to develop a framework
>> for Disaggregated Media that include both improvements on existing
>> mechanisms (e.g. 3pcc) and the develop of one or more new mechanisms
>> like a loosely coupled mechanism that does not require the
>> implementation
>> of complex logic on any of the terminals.
>>
>>
>> Specifically, the proposed working group will develop the following
>> deliverables:
>> 1. A framework document describing key design considerations for
>> Disaggregated
>>    mediamechanism.
>> 2. A specification for a loosely couple mechanism.
>> 3. One or more specifications to improve existing mechanism to
>> coordinate
>>    disaggregated media.
>>
>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>     
>
>   


From Even.roni@huawei.com  Mon Sep 21 09:57:09 2009
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Date: Mon, 21 Sep 2009 19:55:46 +0300
From: Roni Even <Even.roni@huawei.com>
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To: 'Salvatore Loreto' <salvatore.loreto@ericsson.com>
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Cc: 'IETF Dispatch' <dispatch@ietf.org>, 'Gonzalo Camarillo' <gonzalo.camarillo@ericsson.com>
Subject: Re: [dispatch] Charter Proposal for Disaggregated Media and combining SIP and XMPP
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Hi,
This is what I meant, you can look at XMPP and the voice streams as two
media streams that you want to handle as one multimedia session.
Roni Even

> -----Original Message-----
> From: Salvatore Loreto [mailto:salvatore.loreto@ericsson.com]
> Sent: Monday, September 21, 2009 7:39 PM
> To: Roni Even
> Cc: 'IETF Dispatch'; 'Gonzalo Camarillo'
> Subject: Re: [dispatch] Charter Proposal for Disaggregated Media and
> combining SIP and XMPP
> 
> Hi Roni,
> 
> actually there isn't any relation between this two proposals, each of
> them has started from a different use case.
> 
> however if you consider XMPP as a XML streaming, then combining SIP and
> XMPP can be considered as particular case where
> "a user wants to establish a single multimedia session combining
> different media streams coming from
> different devices under his or her control."
> 
> /Sal
> 
> Roni Even wrote:
> > Hi,
> > What is the relation between this proposal and the SIP and XMPP
> proposal.
> > To me it looks like both are trying to combine two separate call legs
> to be
> > as one and have similar requirements.
> >
> > Roni Even
> >
> >
> >> -----Original Message-----
> >> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org]
> On
> >> Behalf Of Salvatore Loreto
> >> Sent: Monday, September 21, 2009 1:52 PM
> >> To: IETF Dispatch
> >> Cc: Gonzalo Camarillo
> >> Subject: [dispatch] Charter Proposal for Disaggregated Media
> >>
> >> Hi there,
> >>
> >> below is a charter proposal for a WG on Disaggregated Media.
> >>
> >> All comments, thoughts, feedbacks are very welcome!
> >>
> >> cheers
> >> Sal
> >>
> >>
> >>
> >> Disaggregated Media WG Charter
> >> -------------------------------
> >> Disaggregated media refers to the ability for a user to create a
> >> multimedia session combining different media streams, coming from
> >> different devices under his or her control, so that they are treated
> >> by the far end of the session as a single media session.
> >>
> >> Generally, a given participant uses a single device to establish (or
> >> participate in) a given multimedia session.  Consequently, the SIP
> >> signaling to manage the multimedia session and the actual media
> >> streams are typically co-located in the same device.  In scenarios
> >> involving disaggregated media, a user wants to establish a single
> >> multimedia session combining different media streams coming from
> >> different devices under his or her control.  This creates a need to
> >> coordinate the exchange of the those media streams within the media
> >> session.
> >>
> >> The Message Bus (Mbus) [RFC3259] can be used by an user to
> coordinate
> >> the different devices under his control to involve them in the call.
> >> The different devices can communicate with one another using Mbus
> >> messages, and then let only one device, a call control engine, to
> >> initiate, manage and terminate call control relations to other SIP
> >> endpoints.
> >> All the different user's devices need to support the Mbus protocol.
> >>
> >> The Megaco [RFC3525] protocol can be used in a disaggregated media
> >> scenario to let one of the user's devices act as a Media Gateway
> >> Controller,
> >> coordinating all the other devices under the user's control, which
> in
> >> this case
> >> will act as Media Gateways. In this case all the different user's
> >> devices
> >> need to support the Megaco protocol.
> >>
> >> The SIP protocol provides 3pcc [RFC3725] as a possible mechanism
> >> to coordinate the exchange of media streams coming from different
> >> devices under the control of the same user, in the case at least
> >> one among the different devices under his control supports this
> >> mechanism and is able to become a Controller for the other in the
> call.
> >>
> >>
> >> All the existing mechanisms only cover a subset of the interesting
> >> scenarios
> >> that involve disaggregated media. Scenarios not covered by existing
> >> mechanisms
> >> include the loosely-coupled one where none of the nodes can act as a
> >> controller
> >> because it does not support the necessary functionality or because
> it
> >> will not
> >> participate in the whole session.
> >>
> >>
> >> The objective of the proposed working group is to develop a
> framework
> >> for Disaggregated Media that include both improvements on existing
> >> mechanisms (e.g. 3pcc) and the develop of one or more new mechanisms
> >> like a loosely coupled mechanism that does not require the
> >> implementation
> >> of complex logic on any of the terminals.
> >>
> >>
> >> Specifically, the proposed working group will develop the following
> >> deliverables:
> >> 1. A framework document describing key design considerations for
> >> Disaggregated
> >>    mediamechanism.
> >> 2. A specification for a loosely couple mechanism.
> >> 3. One or more specifications to improve existing mechanism to
> >> coordinate
> >>    disaggregated media.
> >>
> >>
> >> _______________________________________________
> >> dispatch mailing list
> >> dispatch@ietf.org
> >> https://www.ietf.org/mailman/listinfo/dispatch
> >>
> >
> >


From L.Liess@telekom.de  Mon Sep 21 10:44:20 2009
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Subject: [dispatch] Charter proposal:  SIP Alert-Info URN
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Hello,

Please find below our charter proposal for the SIP Alert-Info URN.

Comments are very welcome!

Kind regards,
Laura=20



The SIP Alert-Info URN (SAIU) working group is chartered to define a new
URN-scheme which allows SIP to convey a particular alert indication
using a name instead of a referenced URI. The Alert-Info URN solves
interoperability problems which we encounter today around the use of the
Alert-Info header field for services as call waiting, call forwarding,
blind transfer, automatic callback, call hold or emergency announcements
sent over PBX systems.   =20

RFC 3261 allows a UAS or proxy to provide a specific ring-/ringback-tone
as a reference (e.g. HTTP URI) which can be played to the user in the
Alert-Info header.=20
This mechanism does not ensure interoperability on the semantic level
when there is no common understanding of the referenced content
(different countries, hearing impaired) or when the user has his own
tones configured in the end device. There are a variety of URI
conventions and proprietary Alert-Info header field parameters to
provide this today, all of which are non-interoperable.  A standardized
set of Alert-Info URNs will increase SIP interoperability for this
header field by replacing proprietary conventions.

The group will produce:

- A specification which describes the problem statement, the Alert-Info
URN-scheme and defines the specific Alert-Info URNs identifiers to solve
the known service interoperability problems.=20

Goals and Milestones
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

Dec 10 - Alert-Info URN specification to IESG (PS)









From stpeter@stpeter.im  Mon Sep 21 14:32:14 2009
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From: Peter Saint-Andre <stpeter@stpeter.im>
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Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM and	presence
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

On 9/21/09 6:28 AM, Simo.Veikkolainen@nokia.com wrote:
> Hello,
> 
> Below is the draft charter text for our proposal on combining SIP
> voice with XMPP IM and presence.
> 
> Comments are very welcome!

Hi Simo,

Do you think that this proposed working group would incorporate or
complete work on the existing I-Ds I have listed below?

http://tools.ietf.org/html/draft-saintandre-sip-xmpp-core
http://tools.ietf.org/html/draft-saintandre-sip-xmpp-presence
http://tools.ietf.org/html/draft-saintandre-sip-xmpp-im
http://tools.ietf.org/html/draft-saintandre-sip-xmpp-chat
http://tools.ietf.org/html/draft-saintandre-sip-xmpp-groupchat
http://tools.ietf.org/html/draft-saintandre-sip-xmpp-media

It would be nice to find a home for those I-Ds, and a focused group of
people who want to finish them.

Peter

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


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Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM and	presence
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At 04:33 PM 9/21/2009, Peter Saint-Andre wrote:
>-----BEGIN PGP SIGNED MESSAGE-----
>Hash: SHA1
>
>On 9/21/09 6:28 AM, Simo.Veikkolainen@nokia.com wrote:
> > Hello,
> >
> > Below is the draft charter text for our proposal on combining SIP
> > voice with XMPP IM and presence.
> >
> > Comments are very welcome!
>
>Hi Simo,
>
>Do you think that this proposed working group would incorporate or
>complete work on the existing I-Ds I have listed below?
>
>http://tools.ietf.org/html/draft-saintandre-sip-xmpp-core
>http://tools.ietf.org/html/draft-saintandre-sip-xmpp-presence
>http://tools.ietf.org/html/draft-saintandre-sip-xmpp-im
>http://tools.ietf.org/html/draft-saintandre-sip-xmpp-chat
>http://tools.ietf.org/html/draft-saintandre-sip-xmpp-groupchat
>http://tools.ietf.org/html/draft-saintandre-sip-xmpp-media
>
>It would be nice to find a home for those I-Ds, and a focused group of
>people who want to finish them.

+1 to what Peter asked.

I'd like to add a comment about the charter as written emphasizing 
"SIP voice" at the expense of "SIP video" (i.e., there is no mention 
of video)...

Since SIP doesn't care which codec(s) are in the offer/answer 
exchange, video needs to be stated explicitly if voice is called out.

James


>Peter
>
>- --
>Peter Saint-Andre
>https://stpeter.im/
>
>
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From dean.willis@softarmor.com  Mon Sep 21 20:17:21 2009
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From: Dean Willis <dean.willis@softarmor.com>
To: "<L.Liess@telekom.de>" <L.Liess@telekom.de>
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Subject: Re: [dispatch] Charter proposal:  SIP Alert-Info URN
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On Sep 21, 2009, at 12:45 PM, <L.Liess@telekom.de>  
<L.Liess@telekom.de> wrote:

> Hello,
>
> Please find below our charter proposal for the SIP Alert-Info URN.
>
> Comments are very welcome!
>
> Kind regards,
> Laura
>
>
>
> The SIP Alert-Info URN (SAIU) working group is chartered to define a  
> new
> URN-scheme which allows SIP to convey a particular alert indication
> using a name instead of a referenced URI. The Alert-Info URN solves
> interoperability problems which we encounter today around the use of  
> the
> Alert-Info header field for services as call waiting, call forwarding,
> blind transfer, automatic callback, call hold or emergency  
> announcements
> sent over PBX systems.
>
> RFC 3261 allows a UAS or proxy to provide a specific ring-/ringback- 
> tone
> as a reference (e.g. HTTP URI) which can be played to the user in the
> Alert-Info header.
> This mechanism does not ensure interoperability on the semantic level
> when there is no common understanding of the referenced content
> (different countries, hearing impaired) or when the user has his own
> tones configured in the end device. There are a variety of URI
> conventions and proprietary Alert-Info header field parameters to
> provide this today, all of which are non-interoperable.  A  
> standardized
> set of Alert-Info URNs will increase SIP interoperability for this
> header field by replacing proprietary conventions.
>
> The group will produce:
>
> - A specification which describes the problem statement, the Alert- 
> Info
> URN-scheme and defines the specific Alert-Info URNs identifiers to  
> solve
> the known service interoperability problems.
>

I think this is a very good idea for a short-lived, tightly-scoped  
working group, and I think the problem is valid and needs to be solved  
(indeed, every time one of the mobile standards groups tried to do  
something silly with ringtones, I suggested they look at alert-info).

I'd like the proposed charter text to g into a little more depth about  
the "known service interoperability problems." Where are these being  
drawn from? Will the WG just reference an external spec for a list, or  
do we need to invent our own list and come to consensus on it?

--
Dean




From Even.roni@huawei.com  Mon Sep 21 23:18:06 2009
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Date: Tue, 22 Sep 2009 09:16:41 +0300
From: Roni Even <Even.roni@huawei.com>
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Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM and	presence
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Hi,
I see similarity in the objectives to the disaggregated media discussed in a
separate thread, see my inline comments in the objectives. I believe that
there is a general case here and not only a specific use case. 
 
The objective of this Working Group is to develop solutions for combining
SIP based voice with XMPP based IM and Presence such that a unified user
experience can be offered to end user. Specifically, solutions are needed on
- addressing; end users should be able to initiate sessions to a user
identity in either SIP or XMPP domain. The corresponding user identity in
the other protocol domain needs to be learned automatically.

Roni: Same in disaggregated media where you have two separate call legs that
you want to look like one

- session correlation; endpoints need to be able to correlate voice sessions
with IM/Presence such that they can be presented to the end user in a
seamless fashion

Roni: need to correlate for example between a call on a Phone device and
video from a conference room video system. The audio will be on the phone
and the video on the room system with two SIP call legs.

- presence; it should be possible to change the XMPP presence status based
on the user's activity in the SIP domain.

Roni: In the disaggregated media case the calling user should be able to
know if the called party can have a separate video call before trying to
start the second call leg. 



Roni Even

> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Simo.Veikkolainen@nokia.com
> Sent: Monday, September 21, 2009 3:29 PM
> To: dispatch@ietf.org
> Subject: [dispatch] Charter proposal on combining SIP voice with XMPP
> IM and presence
> 
> Hello,
> 
> Below is the draft charter text for our proposal on combining SIP voice
> with XMPP IM and presence.
> 
> Comments are very welcome!
> 
> Regards,
> Simo
> 
> 
> -----------------
> 
> 
> Combining SIP VoIP and XMPP IM/Presence
> =======================================
> 
> Currently most standards-based Voice over IP (VoIP) deployments use the
> Session Initiation Protocol (SIP).  In addition to providing basic
> voice service SIP has an extensive support for more advanced telephony
> features including interworking with the traditional Public Switched
> Telephone Network (PSTN).  SIP is also gaining popularity in the field
> of video communication. At the same time, the Extensible Messaging and
> Presence Protocol (XMPP) is enjoying widespread usage in instant
> messaging and presence services.
> 
> Both SIP and XMPP offer extensions for voice as well as IM and presence
> (XMPP voice via the Jingle extension, and SIP IM/presence via SIMPLE
> protocols). However, widespread deployment of these extensions has not
> so far taken place. In order to speed up deployment of integrated VoIP
> and IM systems, SIP based voice and XMPP based IM/Presence could be
> combined in endpoints to offer a voice, IM and presence service without
> any changes to existing SIP and XMPP service infrastrucure.
> 
> The objective of this Working Group is to develop solutions for
> combining SIP based voice with XMPP based IM and Presence such that a
> unified user experience can be offered to end user. Specifically,
> solutions are needed on
> - addressing; end users should be able to initiate sessions to a user
> identity in either SIP or XMPP domain. The corresponding user identity
> in the other protocol domain needs to be learned automatically.
> - session correlation; endpoints need to be able to correlate voice
> sessions with IM/Presence such that they can be presented to the end
> user in a seamless fashion
> - presence; it should be possible to change the XMPP presence status
> based on the user's activity in the SIP domain.
> 
> The goal is to rely on existing service infrastructre, and new
> requirements should be imposed only to the endpoint.
> 
> Protocol interworking, that is, translation from one protocol to
> another, is out of scope of this WG.
> 
> Milestones
> Feb 2010 Problem statement and use cases submitted to IESG as
> Informational
> Dec 2010 Specification on combining SIP based voice and XMPP based
> IM/Presence in a seamless manner submitted to IESG as Proposed
> Standard.
> 
> -----------------
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


From Even.roni@huawei.com  Mon Sep 21 23:21:40 2009
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Date: Tue, 22 Sep 2009 09:20:25 +0300
From: Roni Even <Even.roni@huawei.com>
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Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM and	presence
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Hi,
The charter looks good, what I am missing is the multiparty issue. XMPP is
used for a multiparty chat and I think that in that case the users will
expect an audio/video conference to run in parallel

Roni Even

> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Simo.Veikkolainen@nokia.com
> Sent: Monday, September 21, 2009 3:29 PM
> To: dispatch@ietf.org
> Subject: [dispatch] Charter proposal on combining SIP voice with XMPP
> IM and presence
> 
> Hello,
> 
> Below is the draft charter text for our proposal on combining SIP voice
> with XMPP IM and presence.
> 
> Comments are very welcome!
> 
> Regards,
> Simo
> 
> 
> -----------------
> 
> 
> Combining SIP VoIP and XMPP IM/Presence
> =======================================
> 
> Currently most standards-based Voice over IP (VoIP) deployments use the
> Session Initiation Protocol (SIP).  In addition to providing basic
> voice service SIP has an extensive support for more advanced telephony
> features including interworking with the traditional Public Switched
> Telephone Network (PSTN).  SIP is also gaining popularity in the field
> of video communication. At the same time, the Extensible Messaging and
> Presence Protocol (XMPP) is enjoying widespread usage in instant
> messaging and presence services.
> 
> Both SIP and XMPP offer extensions for voice as well as IM and presence
> (XMPP voice via the Jingle extension, and SIP IM/presence via SIMPLE
> protocols). However, widespread deployment of these extensions has not
> so far taken place. In order to speed up deployment of integrated VoIP
> and IM systems, SIP based voice and XMPP based IM/Presence could be
> combined in endpoints to offer a voice, IM and presence service without
> any changes to existing SIP and XMPP service infrastrucure.
> 
> The objective of this Working Group is to develop solutions for
> combining SIP based voice with XMPP based IM and Presence such that a
> unified user experience can be offered to end user. Specifically,
> solutions are needed on
> - addressing; end users should be able to initiate sessions to a user
> identity in either SIP or XMPP domain. The corresponding user identity
> in the other protocol domain needs to be learned automatically.
> - session correlation; endpoints need to be able to correlate voice
> sessions with IM/Presence such that they can be presented to the end
> user in a seamless fashion
> - presence; it should be possible to change the XMPP presence status
> based on the user's activity in the SIP domain.
> 
> The goal is to rely on existing service infrastructre, and new
> requirements should be imposed only to the endpoint.
> 
> Protocol interworking, that is, translation from one protocol to
> another, is out of scope of this WG.
> 
> Milestones
> Feb 2010 Problem statement and use cases submitted to IESG as
> Informational
> Dec 2010 Specification on combining SIP based voice and XMPP based
> IM/Presence in a seamless manner submitted to IESG as Proposed
> Standard.
> 
> -----------------
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


From oej@edvina.net  Mon Sep 21 23:56:52 2009
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Subject: Re: [dispatch] New topic proposal for DISPATCH
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21 sep 2009 kl. 15.21 skrev Henry Sinnreich:

> On 9/20/09 10:20 PM, "Dan Wing" <dwing@Cisco.COM> wrote:
>
> >So XMPP has to use E.164's (like SIP deployments do), or SIP
> >will have to switch to user@host.  I don't forsee either happening.
> >If this is deemed necessary then we will have to map between them
> >-- somehow.  Send an XMPP message asking "what is your phone
> >number" (in a computer-parsable manner), perhaps?
>
> Now why could not the application query the address book to get both  
> the E.164 addresses and user@host addresses?
Most jabber servers have vcard support that could include both SIP and  
XMPP uri's.

I don't know if SIP has it, but one could certainly involve LDAP for  
finding stuff like this.

/O


From lorenzo@meetecho.com  Tue Sep 22 02:10:56 2009
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From: Lorenzo Miniero <lorenzo@meetecho.com>
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Subject: Re: [dispatch] Charter proposal: Session Recording Protocol
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On Mon, 21 Sep 2009 08:33:40 -0500
"Vijay K. Gurbani" <vkg@alcatel-lucent.com> wrote:

> Hi: The following charter proposal is a continuation of the
> work presented on the same topic (SIP Session Recording)
> at the Stockholm IETF.
> 
> Regards,
> 
> The Session Recording Protocol (SRP) working group is chartered to 
> define a SIP-based protocol for controlling a session (media) recorder.
> 
> Session recording is a critical requirement in many business 
> communications environments such as call centers and financial trading 
> floors.  In some of these environments, all calls must be recorded for 
> regulatory and compliance reasons.  In others, calls may be recorded for 
> quality control, business analytics, or consumer protection.  Recording 
> is typically done by sending a copy of the media to the recording 
> devices.  The working group will produce a specification for a protocol 
> that will manage delivery of media from an end-point that originates 
> media or that has access to it to a recording device. PBX and recording 
> vendors today implement proprietary, incompatible mechanisms to 
> facilitate recording. A standard protocol will reduce the complexity and 
> cost of providing such recording services.
> 
> The Session Recording problem presents certain unique requirements that 
> are not addressed in the current SIP protocol specification. These 
> include requirements such as the need for a distinction between the 
> session that is being recorded versus the session that has been 
> established for recording.
> 
> The SRP Working Group will thoroughly identify use cases, provide 
> example system architectures and deployment scenarios, and define 
> requirements.
> 
> The scope of the activity includes:
> 
>   * Recorder Control
>   * Session metadata content and format
>   * Security mechanisms, including transport and media encryption
>   * Negotiation of recording media streams
> 
> Privacy and security of conversations are significant concerns. The 
> group will define these issues and rationalize with IETF standards and 
> practices. This includes privacy concerns (including RAVEN), encryption, 
> NAT traversal, SIP-enabled firewalls, authorization, and security.
> 
> The group will produce:
> 
>   * Updated Requirements, Use Cases, Architecture draft
>   * Specification for Session Recording Protocol
> 
> Timeline:
> 
>    Apr 2010   Requirements, Use Cases, Architecture to IESG as 
> Informational Draft
> 
>    Nov 2010   Submit protocol draft to IESG as standards track
> 
> - vijay
> -- 
> Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
> 1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
> Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
> Web:   http://ect.bell-labs.com/who/vkg/
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 


Hi Vijay,

as a group we find this subject very interesting, and so we definitely
support the charter proposal. We have a related draft on session
recording (even though it's quite focused on conferencing at the
moment)

http://tools.ietf.org/html/draft-romano-dcon-recording-00

so we'd be glad to contribute to the work to be done in here.

Lorenzo

-- 
Lorenzo Miniero <lorenzo@meetecho.com>

From Markus.Isomaki@nokia.com  Tue Sep 22 04:15:58 2009
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From: <Markus.Isomaki@nokia.com>
To: <jmpolk@cisco.com>, <stpeter@stpeter.im>, <Simo.Veikkolainen@nokia.com>
Date: Tue, 22 Sep 2009 13:16:29 +0200
Thread-Topic: [dispatch] Charter proposal on combining SIP voice with XMPP IM and	presence
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Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM and	presence
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Hi James,=20

James M. Polk wrote:
>
>I'd like to add a comment about the charter as written=20
>emphasizing "SIP voice" at the expense of "SIP video" (i.e.,=20
>there is no mention of video)...
>
>Since SIP doesn't care which codec(s) are in the offer/answer=20
>exchange, video needs to be stated explicitly if voice is called out.
>
>James
>

I agree. I think we can pretty much replace "voice" with "voice and video" =
throughout the charter.

The reason voice was emphasized was simply due to its dominance over video =
in current SIP deployments. However, if we look at the potential framework =
for combined use of SIP and XMPP, it's quite clear that the setup of video =
sessions would belong into the domain of SIP. There's no technical differen=
ce between voice and video in this regard.

Markus

From emil@sip-communicator.org  Tue Sep 22 04:59:44 2009
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Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM and	presence
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Hey Simo,

We've been considering and playing a similar mode of operation for a
couple of years now and some of the most delicate issues that we've come
across include "disambiguation of overlapping services and information"
and "behaviour in cases of "half-failure". Here's what I mean:

What happens for example if a mixed client receives a message over
SIMPLE rather than XMPP. Should it be accepted and displayed to the
user? If yes, then where should a client reply (i.e. SIMPLE or XMPP)? If
we choose SIMPLE for how long should the client privilege SIMPLE over
XMPP and under what conditions.

What happens if a mixed client detects presence agent capabilities in a
SIP service? Should it ignore them and only go for XMPP, use them for
certain destinations only (e.g. destinations that we only have a SIP
address for such as conventional SIP phones), or duplicate all
information over the two protocols in which case: How should clients
handle conflicting presence information received via the two protocols?

How should we handle failures for only one of the services. For example:
 I am a mixed UA and am unable to connect to a SIP registrar but can
still use XMPP. Should I declare complete failure to the user? Should I
declare success but warn the user that some features won't work? If
possible, should I try to make up for the missing SIP services using
XMPP (and vice versa).

So, if you think the above are worth considering how about adding
something along these lines:

disambiguation: both XMPP and SIP define semantics that allow each of
the protocols to offer the complete set of services discussed in this
charter (i.e. voice, video, messaging, and presence). In environments
where one or more of these services are supported with both protocols
clients should be able to use a common procedure for determining which
one to use and under what conditions.

half-failures: in cases where use of one of the protocols becomes
impossible (e.g. due to a server failure) clients should be able to
follow clear procedures on how to behave, which services could they try
to reuse over the protocol that is still available and under what
conditions.

WDYT?

Emil


Simo.Veikkolainen@nokia.com wrote:
> Hello,
> 
> Below is the draft charter text for our proposal on combining SIP voice with XMPP IM and presence.
> 
> Comments are very welcome!
> 
> Regards,
> Simo
> 
> 
> -----------------
> 
> 
> Combining SIP VoIP and XMPP IM/Presence
> =======================================
> 
> Currently most standards-based Voice over IP (VoIP) deployments use the Session Initiation Protocol (SIP).  In addition to providing basic voice service SIP has an extensive support for more advanced telephony features including interworking with the traditional Public Switched Telephone Network (PSTN).  SIP is also gaining popularity in the field of video communication. At the same time, the Extensible Messaging and Presence Protocol (XMPP) is enjoying widespread usage in instant messaging and presence services.  
> 
> Both SIP and XMPP offer extensions for voice as well as IM and presence (XMPP voice via the Jingle extension, and SIP IM/presence via SIMPLE protocols). However, widespread deployment of these extensions has not so far taken place. In order to speed up deployment of integrated VoIP and IM systems, SIP based voice and XMPP based IM/Presence could be combined in endpoints to offer a voice, IM and presence service without any changes to existing SIP and XMPP service infrastrucure. 
> 
> The objective of this Working Group is to develop solutions for combining SIP based voice with XMPP based IM and Presence such that a unified user experience can be offered to end user. Specifically, solutions are needed on 
> - addressing; end users should be able to initiate sessions to a user identity in either SIP or XMPP domain. The corresponding user identity in the other protocol domain needs to be learned automatically.
> - session correlation; endpoints need to be able to correlate voice sessions with IM/Presence such that they can be presented to the end user in a seamless fashion
> - presence; it should be possible to change the XMPP presence status based on the user's activity in the SIP domain.
> 
> The goal is to rely on existing service infrastructre, and new requirements should be imposed only to the endpoint. 
> 
> Protocol interworking, that is, translation from one protocol to another, is out of scope of this WG.
> 
> Milestones
> Feb 2010 Problem statement and use cases submitted to IESG as Informational
> Dec 2010 Specification on combining SIP based voice and XMPP based IM/Presence in a seamless manner submitted to IESG as Proposed Standard.
> 
> -----------------
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

-- 
Emil Ivov, Ph.D.                               67000 Strasbourg,
Project Lead                                   France
SIP Communicator
emcho@sip-communicator.org                     PHONE: +33.1.77.62.43.30
http://sip-communicator.org                    FAX:   +33.1.77.62.47.31

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Cc: Simo.Veikkolainen@nokia.com, dispatch@ietf.org
Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM and	presence
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The issues Emil raises are good ones.
I think they should be in scope for this effort.

	Thanks,
	Paul

Emil Ivov wrote:
> Hey Simo,
> 
> We've been considering and playing a similar mode of operation for a
> couple of years now and some of the most delicate issues that we've come
> across include "disambiguation of overlapping services and information"
> and "behaviour in cases of "half-failure". Here's what I mean:
> 
> What happens for example if a mixed client receives a message over
> SIMPLE rather than XMPP. Should it be accepted and displayed to the
> user? If yes, then where should a client reply (i.e. SIMPLE or XMPP)? If
> we choose SIMPLE for how long should the client privilege SIMPLE over
> XMPP and under what conditions.
> 
> What happens if a mixed client detects presence agent capabilities in a
> SIP service? Should it ignore them and only go for XMPP, use them for
> certain destinations only (e.g. destinations that we only have a SIP
> address for such as conventional SIP phones), or duplicate all
> information over the two protocols in which case: How should clients
> handle conflicting presence information received via the two protocols?
> 
> How should we handle failures for only one of the services. For example:
>  I am a mixed UA and am unable to connect to a SIP registrar but can
> still use XMPP. Should I declare complete failure to the user? Should I
> declare success but warn the user that some features won't work? If
> possible, should I try to make up for the missing SIP services using
> XMPP (and vice versa).
> 
> So, if you think the above are worth considering how about adding
> something along these lines:
> 
> disambiguation: both XMPP and SIP define semantics that allow each of
> the protocols to offer the complete set of services discussed in this
> charter (i.e. voice, video, messaging, and presence). In environments
> where one or more of these services are supported with both protocols
> clients should be able to use a common procedure for determining which
> one to use and under what conditions.
> 
> half-failures: in cases where use of one of the protocols becomes
> impossible (e.g. due to a server failure) clients should be able to
> follow clear procedures on how to behave, which services could they try
> to reuse over the protocol that is still available and under what
> conditions.
> 
> WDYT?
> 
> Emil
> 
> 
> Simo.Veikkolainen@nokia.com wrote:
>> Hello,
>>
>> Below is the draft charter text for our proposal on combining SIP voice with XMPP IM and presence.
>>
>> Comments are very welcome!
>>
>> Regards,
>> Simo
>>
>>
>> -----------------
>>
>>
>> Combining SIP VoIP and XMPP IM/Presence
>> =======================================
>>
>> Currently most standards-based Voice over IP (VoIP) deployments use the Session Initiation Protocol (SIP).  In addition to providing basic voice service SIP has an extensive support for more advanced telephony features including interworking with the traditional Public Switched Telephone Network (PSTN).  SIP is also gaining popularity in the field of video communication. At the same time, the Extensible Messaging and Presence Protocol (XMPP) is enjoying widespread usage in instant messaging and presence services.  
>>
>> Both SIP and XMPP offer extensions for voice as well as IM and presence (XMPP voice via the Jingle extension, and SIP IM/presence via SIMPLE protocols). However, widespread deployment of these extensions has not so far taken place. In order to speed up deployment of integrated VoIP and IM systems, SIP based voice and XMPP based IM/Presence could be combined in endpoints to offer a voice, IM and presence service without any changes to existing SIP and XMPP service infrastrucure. 
>>
>> The objective of this Working Group is to develop solutions for combining SIP based voice with XMPP based IM and Presence such that a unified user experience can be offered to end user. Specifically, solutions are needed on 
>> - addressing; end users should be able to initiate sessions to a user identity in either SIP or XMPP domain. The corresponding user identity in the other protocol domain needs to be learned automatically.
>> - session correlation; endpoints need to be able to correlate voice sessions with IM/Presence such that they can be presented to the end user in a seamless fashion
>> - presence; it should be possible to change the XMPP presence status based on the user's activity in the SIP domain.
>>
>> The goal is to rely on existing service infrastructre, and new requirements should be imposed only to the endpoint. 
>>
>> Protocol interworking, that is, translation from one protocol to another, is out of scope of this WG.
>>
>> Milestones
>> Feb 2010 Problem statement and use cases submitted to IESG as Informational
>> Dec 2010 Specification on combining SIP based voice and XMPP based IM/Presence in a seamless manner submitted to IESG as Proposed Standard.
>>
>> -----------------
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
> 

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Cc: Simo.Veikkolainen@nokia.com, dispatch@ietf.org
Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM and	presence
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Roni Even wrote:
> Hi,
> The charter looks good, what I am missing is the multiparty issue. XMPP is
> used for a multiparty chat and I think that in that case the users will
> expect an audio/video conference to run in parallel

Agree.

Similarly, other features will also be expected to work, such as transfer.

	Thanks,
	Paul

> Roni Even
> 
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>> Behalf Of Simo.Veikkolainen@nokia.com
>> Sent: Monday, September 21, 2009 3:29 PM
>> To: dispatch@ietf.org
>> Subject: [dispatch] Charter proposal on combining SIP voice with XMPP
>> IM and presence
>>
>> Hello,
>>
>> Below is the draft charter text for our proposal on combining SIP voice
>> with XMPP IM and presence.
>>
>> Comments are very welcome!
>>
>> Regards,
>> Simo
>>
>>
>> -----------------
>>
>>
>> Combining SIP VoIP and XMPP IM/Presence
>> =======================================
>>
>> Currently most standards-based Voice over IP (VoIP) deployments use the
>> Session Initiation Protocol (SIP).  In addition to providing basic
>> voice service SIP has an extensive support for more advanced telephony
>> features including interworking with the traditional Public Switched
>> Telephone Network (PSTN).  SIP is also gaining popularity in the field
>> of video communication. At the same time, the Extensible Messaging and
>> Presence Protocol (XMPP) is enjoying widespread usage in instant
>> messaging and presence services.
>>
>> Both SIP and XMPP offer extensions for voice as well as IM and presence
>> (XMPP voice via the Jingle extension, and SIP IM/presence via SIMPLE
>> protocols). However, widespread deployment of these extensions has not
>> so far taken place. In order to speed up deployment of integrated VoIP
>> and IM systems, SIP based voice and XMPP based IM/Presence could be
>> combined in endpoints to offer a voice, IM and presence service without
>> any changes to existing SIP and XMPP service infrastrucure.
>>
>> The objective of this Working Group is to develop solutions for
>> combining SIP based voice with XMPP based IM and Presence such that a
>> unified user experience can be offered to end user. Specifically,
>> solutions are needed on
>> - addressing; end users should be able to initiate sessions to a user
>> identity in either SIP or XMPP domain. The corresponding user identity
>> in the other protocol domain needs to be learned automatically.
>> - session correlation; endpoints need to be able to correlate voice
>> sessions with IM/Presence such that they can be presented to the end
>> user in a seamless fashion
>> - presence; it should be possible to change the XMPP presence status
>> based on the user's activity in the SIP domain.
>>
>> The goal is to rely on existing service infrastructre, and new
>> requirements should be imposed only to the endpoint.
>>
>> Protocol interworking, that is, translation from one protocol to
>> another, is out of scope of this WG.
>>
>> Milestones
>> Feb 2010 Problem statement and use cases submitted to IESG as
>> Informational
>> Dec 2010 Specification on combining SIP based voice and XMPP based
>> IM/Presence in a seamless manner submitted to IESG as Proposed
>> Standard.
>>
>> -----------------
>> _______________________________________________
>> 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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Cc: Simo.Veikkolainen@nokia.com, dispatch@ietf.org
Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM and	presence
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Roni Even wrote:
> Hi,
> I see similarity in the objectives to the disaggregated media discussed in a
> separate thread, see my inline comments in the objectives. I believe that
> there is a general case here and not only a specific use case. 
>  
> The objective of this Working Group is to develop solutions for combining
> SIP based voice with XMPP based IM and Presence such that a unified user
> experience can be offered to end user. Specifically, solutions are needed on
> - addressing; end users should be able to initiate sessions to a user
> identity in either SIP or XMPP domain. The corresponding user identity in
> the other protocol domain needs to be learned automatically.
> 
> Roni: Same in disaggregated media where you have two separate call legs that
> you want to look like one
> 
> - session correlation; endpoints need to be able to correlate voice sessions
> with IM/Presence such that they can be presented to the end user in a
> seamless fashion
> 
> Roni: need to correlate for example between a call on a Phone device and
> video from a conference room video system. The audio will be on the phone
> and the video on the room system with two SIP call legs.
> 
> - presence; it should be possible to change the XMPP presence status based
> on the user's activity in the SIP domain.
> 
> Roni: In the disaggregated media case the calling user should be able to
> know if the called party can have a separate video call before trying to
> start the second call leg. 

I've been arguing that in the all-sip case the burden of aggregating the 
disparate media/devices should fall on the end that wants it, not on the 
other end that is happy to have a single multimedia session. That 
eliminates the question of whether the called party can support a 
separate video call.

That of course doesn't work when aggregating two protocols that 
different approaches to session initiation, such as SIP and XMPP. 
(Unless one can function as a subordinate to the other, such as 
negotiating an "XMPP session" using sip.

	Thanks,
	Paul

> Roni Even
> 
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>> Behalf Of Simo.Veikkolainen@nokia.com
>> Sent: Monday, September 21, 2009 3:29 PM
>> To: dispatch@ietf.org
>> Subject: [dispatch] Charter proposal on combining SIP voice with XMPP
>> IM and presence
>>
>> Hello,
>>
>> Below is the draft charter text for our proposal on combining SIP voice
>> with XMPP IM and presence.
>>
>> Comments are very welcome!
>>
>> Regards,
>> Simo
>>
>>
>> -----------------
>>
>>
>> Combining SIP VoIP and XMPP IM/Presence
>> =======================================
>>
>> Currently most standards-based Voice over IP (VoIP) deployments use the
>> Session Initiation Protocol (SIP).  In addition to providing basic
>> voice service SIP has an extensive support for more advanced telephony
>> features including interworking with the traditional Public Switched
>> Telephone Network (PSTN).  SIP is also gaining popularity in the field
>> of video communication. At the same time, the Extensible Messaging and
>> Presence Protocol (XMPP) is enjoying widespread usage in instant
>> messaging and presence services.
>>
>> Both SIP and XMPP offer extensions for voice as well as IM and presence
>> (XMPP voice via the Jingle extension, and SIP IM/presence via SIMPLE
>> protocols). However, widespread deployment of these extensions has not
>> so far taken place. In order to speed up deployment of integrated VoIP
>> and IM systems, SIP based voice and XMPP based IM/Presence could be
>> combined in endpoints to offer a voice, IM and presence service without
>> any changes to existing SIP and XMPP service infrastrucure.
>>
>> The objective of this Working Group is to develop solutions for
>> combining SIP based voice with XMPP based IM and Presence such that a
>> unified user experience can be offered to end user. Specifically,
>> solutions are needed on
>> - addressing; end users should be able to initiate sessions to a user
>> identity in either SIP or XMPP domain. The corresponding user identity
>> in the other protocol domain needs to be learned automatically.
>> - session correlation; endpoints need to be able to correlate voice
>> sessions with IM/Presence such that they can be presented to the end
>> user in a seamless fashion
>> - presence; it should be possible to change the XMPP presence status
>> based on the user's activity in the SIP domain.
>>
>> The goal is to rely on existing service infrastructre, and new
>> requirements should be imposed only to the endpoint.
>>
>> Protocol interworking, that is, translation from one protocol to
>> another, is out of scope of this WG.
>>
>> Milestones
>> Feb 2010 Problem statement and use cases submitted to IESG as
>> Informational
>> Dec 2010 Specification on combining SIP based voice and XMPP based
>> IM/Presence in a seamless manner submitted to IESG as Proposed
>> Standard.
>>
>> -----------------
>> _______________________________________________
>> 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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Subject: Re: [dispatch] New topic proposal for DISPATCH
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AFAIK this is only a solution to the extent that I possess such info 
about the other party in the call.

IM has evolved in such a way that typically you are only allowed to IM 
people you "know", so info is available locally about the other party.

But the same is not true of telephony, where often the *only* 
information you have about the other party is the phone number. (Or AOR.)

So while looking things up in your address book could provide some 
useful fallback behavior, I don't think it is a complete solution.

	Thanks,
	Paul

Olle E. Johansson wrote:
> 
> 21 sep 2009 kl. 15.21 skrev Henry Sinnreich:
> 
>> On 9/20/09 10:20 PM, "Dan Wing" <dwing@Cisco.COM> wrote:
>>
>> >So XMPP has to use E.164's (like SIP deployments do), or SIP
>> >will have to switch to user@host.  I don't forsee either happening.
>> >If this is deemed necessary then we will have to map between them
>> >-- somehow.  Send an XMPP message asking "what is your phone
>> >number" (in a computer-parsable manner), perhaps?
>>
>> Now why could not the application query the address book to get both 
>> the E.164 addresses and user@host addresses?
> Most jabber servers have vcard support that could include both SIP and 
> XMPP uri's.
> 
> I don't know if SIP has it, but one could certainly involve LDAP for 
> finding stuff like this.
> 
> /O
> 
> 

From dromasca@avaya.com  Tue Sep 22 06:42:22 2009
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From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
To: "Peter Saint-Andre" <stpeter@stpeter.im>, <Simo.Veikkolainen@nokia.com>
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Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM and	presence
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How would this work relate to the item in the XMPP WG charter? =20

> Many of the core and extended features of XMPP have also been
implemented in technologies based on the Session Initiation Protocol
(SIP). To ensure interworking between XMPP systems and SIP systems, a
number of Internet-Drafts (draft-saintandre-sip-xmpp-*) have been
produced. The group will define a framework within which this work could
be completed.

Dan


> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Peter Saint-Andre
> Sent: Tuesday, September 22, 2009 12:33 AM
> To: Simo.Veikkolainen@nokia.com
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] Charter proposal on combining SIP=20
> voice with XMPP IM and presence
>=20
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>=20
> On 9/21/09 6:28 AM, Simo.Veikkolainen@nokia.com wrote:
> > Hello,
> >=20
> > Below is the draft charter text for our proposal on combining SIP=20
> > voice with XMPP IM and presence.
> >=20
> > Comments are very welcome!
>=20
> Hi Simo,
>=20
> Do you think that this proposed working group would=20
> incorporate or complete work on the existing I-Ds I have listed below?
>=20
> http://tools.ietf.org/html/draft-saintandre-sip-xmpp-core
> http://tools.ietf.org/html/draft-saintandre-sip-xmpp-presence
> http://tools.ietf.org/html/draft-saintandre-sip-xmpp-im
> http://tools.ietf.org/html/draft-saintandre-sip-xmpp-chat
> http://tools.ietf.org/html/draft-saintandre-sip-xmpp-groupchat
> http://tools.ietf.org/html/draft-saintandre-sip-xmpp-media
>=20
> It would be nice to find a home for those I-Ds, and a focused=20
> group of people who want to finish them.
>=20
> Peter
>=20
> - --
> Peter Saint-Andre
> https://stpeter.im/
>=20
>=20
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> _______________________________________________
> dispatch mailing list
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> https://www.ietf.org/mailman/listinfo/dispatch
>=20

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Hi,=20

Paul Kyzivat wrote:
>
>Roni Even wrote:
>> Hi,
>> The charter looks good, what I am missing is the multiparty issue.=20
>> XMPP is used for a multiparty chat and I think that in that case the=20
>> users will expect an audio/video conference to run in parallel
>
>Agree.
>
>Similarly, other features will also be expected to work, such=20
>as transfer.
>

We could add such things on the charter at later dates. I think it would be=
 important to get something concrete solved first, and preferably even impl=
emented and deployed before going forward. Initially we should IMHO only ha=
ve features that can be relatively easily implemented by using existing SIP=
 and XMPP client codebases. Otherwise we loose the whole point of this work=
.=20

In general I agree that we need to have a way for the combined client to be=
 able to take part into a SIP based voice/video conference and XMPP based m=
ultiparty IM chat and if the server side supports the proper extensions, ac=
tually know that these two are part of the same multimedia conference. Perh=
aps even the rosters can be correlated. But at least we shouldn't require t=
he full XCON suite for this.

Markus=

From Hannes.Tschofenig@gmx.net  Tue Sep 22 06:50:00 2009
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Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM and	presence
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I have only followed parts of the email thread but I would favor of keeping
the two efforts separately since the problem statements are quite different.
Don't re-write them in a way that nobody understands them anymore. 

The SIP / XMPP interworking is also a topic of relevance today, i.e. not a
research topic. 


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From: <Markus.Isomaki@nokia.com>
To: <dromasca@avaya.com>, <stpeter@stpeter.im>, <Simo.Veikkolainen@nokia.com>
Date: Tue, 22 Sep 2009 15:51:02 +0200
Thread-Topic: [dispatch] Charter proposal on combining SIP voice with XMPP IM	and	presence
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Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM	and	presence
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Hi,

That's exactly the same work. I guess Peter is asking if that SIP-related w=
ork could be moved from XMPP WG to happen together with the proposed SIP-XM=
PP combined use work, as a new (mini-)WG for instance.

To me Peter's request makes sense. Of course only if we manage to get the n=
ew SIP/XMPP group setup.) It's a bit odd thing currently in XMPP WG. That's=
 the same reason we are bringing the SIP/XMPP combined use proposal to DISP=
ATCH rather than XMPP WG, i.e. it does not really fit that well with the ot=
her items XMPP WG is working on.

Markus


>-----Original Message-----
>From: dispatch-bounces@ietf.org=20
>[mailto:dispatch-bounces@ietf.org] On Behalf Of ext Romascanu,=20
>Dan (Dan)
>Sent: 22 September, 2009 16:43
>To: Peter Saint-Andre; Veikkolainen Simo (Nokia-D/Helsinki)
>Cc: dispatch@ietf.org
>Subject: Re: [dispatch] Charter proposal on combining SIP=20
>voice with XMPP IM and presence
>
>How would this work relate to the item in the XMPP WG charter? =20
>
>> Many of the core and extended features of XMPP have also been
>implemented in technologies based on the Session Initiation=20
>Protocol (SIP). To ensure interworking between XMPP systems=20
>and SIP systems, a number of Internet-Drafts=20
>(draft-saintandre-sip-xmpp-*) have been produced. The group=20
>will define a framework within which this work could be completed.
>
>Dan
>
>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Peter Saint-Andre
>> Sent: Tuesday, September 22, 2009 12:33 AM
>> To: Simo.Veikkolainen@nokia.com
>> Cc: dispatch@ietf.org
>> Subject: Re: [dispatch] Charter proposal on combining SIP voice with=20
>> XMPP IM and presence
>>=20
>> -----BEGIN PGP SIGNED MESSAGE-----
>> Hash: SHA1
>>=20
>> On 9/21/09 6:28 AM, Simo.Veikkolainen@nokia.com wrote:
>> > Hello,
>> >=20
>> > Below is the draft charter text for our proposal on combining SIP=20
>> > voice with XMPP IM and presence.
>> >=20
>> > Comments are very welcome!
>>=20
>> Hi Simo,
>>=20
>> Do you think that this proposed working group would incorporate or=20
>> complete work on the existing I-Ds I have listed below?
>>=20
>> http://tools.ietf.org/html/draft-saintandre-sip-xmpp-core
>> http://tools.ietf.org/html/draft-saintandre-sip-xmpp-presence
>> http://tools.ietf.org/html/draft-saintandre-sip-xmpp-im
>> http://tools.ietf.org/html/draft-saintandre-sip-xmpp-chat
>> http://tools.ietf.org/html/draft-saintandre-sip-xmpp-groupchat
>> http://tools.ietf.org/html/draft-saintandre-sip-xmpp-media
>>=20
>> It would be nice to find a home for those I-Ds, and a=20
>focused group of=20
>> people who want to finish them.
>>=20
>> Peter
>>=20
>> - --
>> Peter Saint-Andre
>> https://stpeter.im/
>>=20
>>=20
>> -----BEGIN PGP SIGNATURE-----
>> Version: GnuPG v1.4.8 (Darwin)
>> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/
>>=20
>> iEYEARECAAYFAkq38RoACgkQNL8k5A2w/vwgiACfUzJTMSaaH7hc+9u8JrPwgQ2S
>> QQEAoMcAOMhkYNVHxNxJs7GIJye5XPoT
>> =3DQMet
>> -----END PGP SIGNATURE-----
>> _______________________________________________
>> 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 Markus.Isomaki@nokia.com  Tue Sep 22 07:04:45 2009
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From: <Markus.Isomaki@nokia.com>
To: <Hannes.Tschofenig@gmx.net>, <pkyzivat@cisco.com>, <Even.roni@huawei.com>
Date: Tue, 22 Sep 2009 16:05:29 +0200
Thread-Topic: [dispatch] Charter proposal on combining SIP voice with XMPP IM	and	presence
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Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM	and	presence
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Hannes,

Which two efforts are you talking about? At least three efforts have been b=
rought up that have some relationship:

1. SIP and XMPP combined use (the charter proposed in this thread, draft-ve=
ikkolainen-...)=20
2. SIP - XMPP interworking (already a WG item in XMPP WG, draft-saintandre-=
sip-...)
3. Disaggregated media proposal (sent to DISPATCH by Salvatore Loreto)

My take is that 1. and 2. are relatively orthogonal but probably the same s=
et of people are interested in them. 1 and 3. are related in a sense that i=
f we define some kind of session correlation header in SIP (to correlate tw=
o or more SIP dialogs/sessions) we might ideally use the same for SIP/XMPP =
correlation purposes. But that's a minor optimization and either work shoul=
d not get stuck behind the other. There's still room for new SIP headers, r=
ight :-)

Markus


>-----Original Message-----
>From: dispatch-bounces@ietf.org=20
>[mailto:dispatch-bounces@ietf.org] On Behalf Of ext Hannes Tschofenig
>Sent: 22 September, 2009 16:54
>To: 'Paul Kyzivat'; 'Roni Even'
>Cc: Veikkolainen Simo (Nokia-D/Helsinki); dispatch@ietf.org
>Subject: Re: [dispatch] Charter proposal on combining SIP=20
>voice with XMPP IM and presence
>
>I have only followed parts of the email thread but I would=20
>favor of keeping the two efforts separately since the problem=20
>statements are quite different.
>Don't re-write them in a way that nobody understands them anymore.=20
>
>The SIP / XMPP interworking is also a topic of relevance=20
>today, i.e. not a research topic.=20
>
>_______________________________________________
>dispatch mailing list
>dispatch@ietf.org
>https://www.ietf.org/mailman/listinfo/dispatch
>=

From Hannes.Tschofenig@gmx.net  Tue Sep 22 07:07:22 2009
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Cc: Simo.Veikkolainen@nokia.com, dispatch@ietf.org
Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IMand	presence
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Sorry, I should have been more precise. I was referring to "SIP and XMPP
combined use" and "Disaggregated media proposal". Even if we envision
solutions to be similar (some form of SIP header) does not mean that they
need to be mashed together. 

>-----Original Message-----
>From: Markus.Isomaki@nokia.com [mailto:Markus.Isomaki@nokia.com] 
>Sent: 22 September, 2009 17:05
>To: Hannes.Tschofenig@gmx.net; pkyzivat@cisco.com; Even.roni@huawei.com
>Cc: Simo.Veikkolainen@nokia.com; dispatch@ietf.org
>Subject: RE: [dispatch] Charter proposal on combining SIP 
>voice with XMPP IMand presence
>
>Hannes,
>
>Which two efforts are you talking about? At least three 
>efforts have been brought up that have some relationship:
>
>1. SIP and XMPP combined use (the charter proposed in this 
>thread, draft-veikkolainen-...) 2. SIP - XMPP interworking 
>(already a WG item in XMPP WG,  draft-saintandre-sip-...) 3. 
>Disaggregated media proposal (sent to DISPATCH by Salvatore Loreto)
>
>My take is that 1. and 2. are relatively orthogonal but 
>probably the same set of people are interested in them. 1 and 
>3. are related in a sense that if we define some kind of 
>session correlation header in SIP (to correlate two or more 
>SIP dialogs/sessions) we might ideally use the same for 
>SIP/XMPP correlation purposes. But that's a minor optimization 
>and either work should not get stuck behind the other. There's 
>still room for new SIP headers, right :-)
>
>Markus
>
>
>>-----Original Message-----
>>From: dispatch-bounces@ietf.org
>>[mailto:dispatch-bounces@ietf.org] On Behalf Of ext Hannes Tschofenig
>>Sent: 22 September, 2009 16:54
>>To: 'Paul Kyzivat'; 'Roni Even'
>>Cc: Veikkolainen Simo (Nokia-D/Helsinki); dispatch@ietf.org
>>Subject: Re: [dispatch] Charter proposal on combining SIP voice with 
>>XMPP IM and presence
>>
>>I have only followed parts of the email thread but I would favor of 
>>keeping the two efforts separately since the problem statements are 
>>quite different.
>>Don't re-write them in a way that nobody understands them anymore. 
>>
>>The SIP / XMPP interworking is also a topic of relevance today, i.e. 
>>not a research topic.
>>
>>_______________________________________________
>>dispatch mailing list
>>dispatch@ietf.org
>>https://www.ietf.org/mailman/listinfo/dispatch
>>
>


From L.Liess@telekom.de  Tue Sep 22 07:11:33 2009
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Dean,

Thank you for supporting the work. Please see answers in-line.=20

>=20
> I think this is a very good idea for a short-lived, tightly-scoped =20
> working group, and I think the problem is valid and needs to=20
> be solved =20
> (indeed, every time one of the mobile standards groups tried to do =20
> something silly with ringtones, I suggested they look at alert-info).
>=20
> I'd like the proposed charter text to g into a little more=20
> depth about =20
> the "known service interoperability problems." Where are these being =20
> drawn from?

The cases I know of are mainly the use-cases for which indications and
sub-indications have been defined in=20
Sections 3 and 4. of
http://www.ietf.org/id/draft-alexeitsev-bliss-alert-info-urns-02.txt .


- For the call waiting supplementary service provided by carriers, there
is a need for the callee's proxy informs the caller that the call is
waiting. The callee's proxy must provide this information at a semantic
to the caller's end device. A URI will not do it for different
countries, hearing impaired and so on.=20
=20
- All other use cases in the draft came from Alan's implementation
experience at Avaya and interoperability issues around the use of the=20
Alert-Info header field when not-using an external ring file. Here is
Alan's text, which is not in the current version of the draft:     =20
=20
For example, consider the PBX special ringtone for an external (to the
PBX)=20
caller.  Different vendors use different approaches such as:

   Alert-Info: <file://ring.pcm>;alert=3Dnormal  where ring.pcm is a =
dummy

file

or:

   Alert-Info: <file://normal.ring.pcm>

or:

   Alert-Info: <sip:normal-ringtone@example.com>

As a result, Alert-Info currently only works when the same vendor=20
provides proxy and UA, as only then is the same "fake" proprietary URI=20
convention is used.

Also draft-ietf-bliss-shared-appearances needs to specify a default
ringtone when using appearance Alert-Info parameter. =20

> Will the WG just reference an external spec for a=20
> list, or =20
> do we need to invent our own list and come to consensus on it?

I intend to include the text above into the next version of the draft.
Maybe a separate section in the draft on the problem statement and use
cases would make sense.=20

My understanding from your mail way that you know of other use cases
from other mobile standards group.  Is that correct? If yes, does it
make sense to consider them here? The syntax is open so people can add
new labels.=20

In his message
http://www.ietf.org/mail-archive/web/bliss/current/msg01020.html   Adam
raised the issue of the tones defined in TIA/EIA-41-D and 3GPP2 A.S0014.
Personally, I don't have any experience with them. But, as John pointed
out in http://www.ietf.org/mail-archive/web/bliss/current/msg01025.html,
we should try to define names that reflect the meaning, rather than the
representation, of a tone.  I don't know anything about the meaning of
Short-Short2. For my feeling,  this are quite US specific and for my
feeling URIs could be used in this case.=20

=20

Thanks a lot
Laura












>=20
> --
> Dean
>=20
>=20
>=20
>=20

From enrico.marocco@telecomitalia.it  Tue Sep 22 07:36:55 2009
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Subject: Re: [dispatch] Charter proposed: CBUS
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Christer, I see value in a general mechanism for selectively retrieve
lists of URIs, but am wondering whether this is actually a charter
proposal (in such case I guess the text requires heavy rewording,
defining acronyms as a start) or just an event package registration
request according to rfc3427bis, S.4.1?

Enrico

Christer Holmberg wrote:
> 
> Hi,
> 
> Below is the proposed charter for CBUS.
> 
> Regards,
> 
> Christer
> 
> -----------------------------------------
> 
> CBUS Charter.
> =============
> 
> Introduction:
> -------------
> 
> Various OMA service enablers need to be able to retrieve a list of
> addresses (URIs), where the users associated with the URIs fulfil
> certain conditions.  User information is evaluated against the
> conditions, and the matches form a URI list.
> 
> The need for the functionality originates from the OMA PoC service.
> However, there is ongoing work in OMA to define a common mechanism,
> called CBUS enabler (OMA-RD-CBUS-V1_0-20090203-C), to provide the
> functionality, so that it can be used for various types of services
> (e.g. messaging, gaming, conferencing and advertisement).
> 
> The CBUS enabler provides the following functions:
> 
> - Support for requestor initiated condition-based URIs selection requests.
> 
> - Administration of conditions for URI selection.
> 
> - Interaction with other enablers for retrieval of individual user's
>   information (e.g. presence information, location information).
> 
> - Evaluation of conditions and selection of matching individual users
>   based on one time evaluation.
> 
> - Re-evaluation of conditions and re-selection of matching individual
>   users based on monitoring.
> 
> - Aggregation of one-time evaluation results and monitoring results
>   from different information sources.
> 
> - Notification of evaluation results to requestor.
> 
> The conditions can be based on user information which changes over time
> (e.g. presence information or geographical location), but they can also
> be based on more static user information (e.g. hobbies or personal
> interests).
> 
> The entity which acts as requester, and provides the conditions to be
> used for the user information evaluation, is called CBUS client.  The
> CBUS client usually resides on the user equipment, or an application
> server, which supports the CBUS enabler.
> 
> The selection of URIs, based on the provided conditions, may be
> performed by taking a snapshot, i.e. by making a one-time evaluation of
> the user information to find out whether the conditions are fulfilled or
> not.  The URI selection may also be performed by continous monitoring of
> the user information.
> 
> 
> Work and deliverables:
> ----------------------
> 
> The purpose of the work is to, based on the OMA CBUS requirements (OMA-
> RD-CBUS-V1_0-20090203-C), procude the SIP protocol information elements
> which are needed to implement the interface between the CBUS client and
> CBUS server, using the SIP SUBSCRIBE/NOTIFY framework (RFC 3265).
> 
> 
> The primary task for the work is to:
> 
> 1. Procude a mechanism for a subscriber (CBUS client) to, when
> subscribing to an event package which contains a list of URI, provide
> conditions, candidate URIs and evaluation parameters to the notifier
> (CBUS server).
> 
> The mechanism may be used on defining an XML based message body type,
> and SIP header field parameters.
> 
> 2. Produce an event package, or possible re-use an existing event
> package, used to the notifier (CBUS server) to send URI lists (based on
> the conditions and candidate URIs) to the subscriber (CBUS client).
> 
> The CBUS client actions triggered by the received URI list, and the
> interactions between the CBUS server and other enablers, are outside the
> scope of the work.
> 
> 
> The work is planned to take place in, or in co-operaton with, the
> SIPCORE WG.
> 
> 
> Goals and milestones:
> ---------------------
> 
> Jan 2010        Working Group Last Call for the document which contains
> the mechanism for a
>             subscriber to provide the needed CBUS information to the
> notifier, and
>             contains the event package (if a new package is needed) for the
>             notifier to provide the URI list to the subscriber.
> 
> Mar 2010    Submit document to IESG as Proposed Standard
> 

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Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM and	presence
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On 9/22/09 7:43 AM, Romascanu, Dan (Dan) wrote:
> How would this work relate to the item in the XMPP WG charter?  
> 
>> Many of the core and extended features of XMPP have also been
> implemented in technologies based on the Session Initiation Protocol
> (SIP). To ensure interworking between XMPP systems and SIP systems, a
> number of Internet-Drafts (draft-saintandre-sip-xmpp-*) have been
> produced. The group will define a framework within which this work could
> be completed.

Right, that refers to the drafts I noted in my previous message. As Simo
points out, the same people would be interested in both the "combined
SIP/XMPP" use cases and the "SIP/XMPP interworking" use cases. IMHO it
would be more productive to complete the interworking tasks in a WG that
is dedicated to SIP and XMPP, rather than in the XMPP WG.

Peter

> Dan
> 
> 
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org 
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Peter Saint-Andre
>> Sent: Tuesday, September 22, 2009 12:33 AM
>> To: Simo.Veikkolainen@nokia.com
>> Cc: dispatch@ietf.org
>> Subject: Re: [dispatch] Charter proposal on combining SIP 
>> voice with XMPP IM and presence
>>
> On 9/21/09 6:28 AM, Simo.Veikkolainen@nokia.com wrote:
>>>> Hello,
>>>>
>>>> Below is the draft charter text for our proposal on combining SIP 
>>>> voice with XMPP IM and presence.
>>>>
>>>> Comments are very welcome!
> Hi Simo,
> 
> Do you think that this proposed working group would 
> incorporate or complete work on the existing I-Ds I have listed below?
> 
> http://tools.ietf.org/html/draft-saintandre-sip-xmpp-core
> http://tools.ietf.org/html/draft-saintandre-sip-xmpp-presence
> http://tools.ietf.org/html/draft-saintandre-sip-xmpp-im
> http://tools.ietf.org/html/draft-saintandre-sip-xmpp-chat
> http://tools.ietf.org/html/draft-saintandre-sip-xmpp-groupchat
> http://tools.ietf.org/html/draft-saintandre-sip-xmpp-media
> 
> It would be nice to find a home for those I-Ds, and a focused 
> group of people who want to finish them.
> 
> Peter
> 

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On 9/22/09 7:51 AM, Markus.Isomaki@nokia.com wrote:
> Hi,
> 
> That's exactly the same work. I guess Peter is asking if that
> SIP-related work could be moved from XMPP WG to happen together with
> the proposed SIP-XMPP combined use work, as a new (mini-)WG for
> instance.
> 
> To me Peter's request makes sense. Of course only if we manage to get
> the new SIP/XMPP group setup.) It's a bit odd thing currently in XMPP
> WG. That's the same reason we are bringing the SIP/XMPP combined use
> proposal to DISPATCH rather than XMPP WG, i.e. it does not really fit
> that well with the other items XMPP WG is working on.

Agreed.

Peter

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


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From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
To: "Peter Saint-Andre" <stpeter@stpeter.im>
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=20

> -----Original Message-----
> From: Peter Saint-Andre [mailto:stpeter@stpeter.im]=20

> As Simo points out, the same people would be=20
> interested in both the "combined SIP/XMPP" use cases and the=20
> "SIP/XMPP interworking" use cases.=20

Can you shortly explain what is the difference between the two or point
me to a reference that explains this?=20


> IMHO it would be more=20
> productive to complete the interworking tasks in a WG that is=20
> dedicated to SIP and XMPP, rather than in the XMPP WG.

Yes, but the XMPP WG was (re)chartered just a few months ago, and I am
trying to understand what changed since then, or what happened in the
XMPP WG for this work to look so soon for another home.=20

Thanks and Regards,

Dan

>=20
> Peter
>=20
> > Dan
> >=20

From stpeter@stpeter.im  Tue Sep 22 09:05:51 2009
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On 9/22/09 9:47 AM, Romascanu, Dan (Dan) wrote:
>  
> 
>> -----Original Message-----
>> From: Peter Saint-Andre [mailto:stpeter@stpeter.im] 
> 
>> As Simo points out, the same people would be 
>> interested in both the "combined SIP/XMPP" use cases and the 
>> "SIP/XMPP interworking" use cases. 
> 
> Can you shortly explain what is the difference between the two or point
> me to a reference that explains this? 

Combinations are things like "invoke an XMPP groupchat session from a
SIP UA" or "show XMPP presence in a SIP UA".

Interworkings are things like "gateway instant messages between a SIP
network and an XMPP network".

>> IMHO it would be more 
>> productive to complete the interworking tasks in a WG that is 
>> dedicated to SIP and XMPP, rather than in the XMPP WG.
> 
> Yes, but the XMPP WG was (re)chartered just a few months ago, and I am
> trying to understand what changed since then, or what happened in the
> XMPP WG for this work to look so soon for another home. 

The XMPP WG was the best home we had for the SIP-XMPP interworking
effort at the time and IMHO will not receive a great deal of attention
within the XMPP WG because that WG is prioritizing XMPP-specific tasks
such as:

1. Completion of revisions to RFC 3920 and RFC 3921

2. Replacement of RFC 3923 (end-to-end signing and encryption) with
technologies that will be more widely deployed (probably based on TLS
for encryption and XMLDSIG for signing, though that is yet to be worked
out in detail)

3. Improvement of the server-to-server communications model for secure
federation between large-scale service providers.

IMHO those activities will keep the XMPP WG chugging along for a while
and the SIP-XMPP work will not receive a lot of attention. If we can
move it over to a new, more focused WG then I am in favor, because that
will encourage faster progress, I think.

Peter

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


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From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
To: "Peter Saint-Andre" <stpeter@stpeter.im>
Cc: Simo.Veikkolainen@nokia.com, dispatch@ietf.org
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Got it, thanks for the explanations. The differences between the two
approaches could be caught in text in the proposed charter - BTW. Moving
work from an existing WG to a new one is an interesting exercise,
because you are implicitly requiring some kind of rechartering of the
XMPP WG as well, but (much) better schedules in a better focused effort
are a good argument.=20

Dan
=20

> -----Original Message-----
> From: Peter Saint-Andre [mailto:stpeter@stpeter.im]=20
> Sent: Tuesday, September 22, 2009 7:07 PM
> To: Romascanu, Dan (Dan)
> Cc: Simo.Veikkolainen@nokia.com; dispatch@ietf.org
> Subject: Re: [dispatch] Charter proposal on combining SIP=20
> voice with XMPP IM and presence
>=20
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>=20
> On 9/22/09 9:47 AM, Romascanu, Dan (Dan) wrote:
> > =20
> >=20
> >> -----Original Message-----
> >> From: Peter Saint-Andre [mailto:stpeter@stpeter.im]
> >=20
> >> As Simo points out, the same people would be interested in=20
> both the=20
> >> "combined SIP/XMPP" use cases and the "SIP/XMPP interworking" use=20
> >> cases.
> >=20
> > Can you shortly explain what is the difference between the two or=20
> > point me to a reference that explains this?
>=20
> Combinations are things like "invoke an XMPP groupchat=20
> session from a SIP UA" or "show XMPP presence in a SIP UA".
>=20
> Interworkings are things like "gateway instant messages=20
> between a SIP network and an XMPP network".
>=20
> >> IMHO it would be more
> >> productive to complete the interworking tasks in a WG that is=20
> >> dedicated to SIP and XMPP, rather than in the XMPP WG.
> >=20
> > Yes, but the XMPP WG was (re)chartered just a few months=20
> ago, and I am=20
> > trying to understand what changed since then, or what=20
> happened in the=20
> > XMPP WG for this work to look so soon for another home.
>=20
> The XMPP WG was the best home we had for the SIP-XMPP=20
> interworking effort at the time and IMHO will not receive a=20
> great deal of attention within the XMPP WG because that WG is=20
> prioritizing XMPP-specific tasks such as:
>=20
> 1. Completion of revisions to RFC 3920 and RFC 3921
>=20
> 2. Replacement of RFC 3923 (end-to-end signing and=20
> encryption) with technologies that will be more widely=20
> deployed (probably based on TLS for encryption and XMLDSIG=20
> for signing, though that is yet to be worked out in detail)
>=20
> 3. Improvement of the server-to-server communications model=20
> for secure federation between large-scale service providers.
>=20
> IMHO those activities will keep the XMPP WG chugging along=20
> for a while and the SIP-XMPP work will not receive a lot of=20
> attention. If we can move it over to a new, more focused WG=20
> then I am in favor, because that will encourage faster=20
> progress, I think.
>=20
> Peter
>=20
> - --
> Peter Saint-Andre
> https://stpeter.im/
>=20
>=20
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> Version: GnuPG v1.4.8 (Darwin)
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>=20
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> DdQAnRI/hoskAt5Rja+fH46zo/EqEnnI
> =3DaWhG
> -----END PGP SIGNATURE-----
>=20

From oej@edvina.net  Tue Sep 22 09:32:37 2009
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From: "Olle E. Johansson" <oej@edvina.net>
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To: Peter Saint-Andre <stpeter@stpeter.im>
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Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM and	presence
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22 sep 2009 kl. 18.06 skrev Peter Saint-Andre:
 >Combinations are things like "invoke an XMPP groupchat session from a
 >SIP UA" or "show XMPP presence in a SIP UA".

 >Interworkings are things like "gateway instant messages between a SIP
 >network and an XMPP network".

I think both your combination examples requires interworking. What  
I've seen discussed here are multiprotocol clients so combination  
would rather be something like

"invoke a SIP conference call and XMPP groupchat simultaneously"

or

"Avoid SIP calls if XMPP presence is DND"

> IMHO those activities will keep the XMPP WG chugging along for a while
> and the SIP-XMPP work will not receive a lot of attention. If we can
> move it over to a new, more focused WG then I am in favor, because  
> that
> will encourage faster progress, I think.

Will a new working group mean a whole set of fresh new members with  
more time?

/O

From stpeter@stpeter.im  Tue Sep 22 09:39:16 2009
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

On 9/22/09 10:33 AM, Olle E. Johansson wrote:
> 
> 22 sep 2009 kl. 18.06 skrev Peter Saint-Andre:
> 
>> IMHO those activities will keep the XMPP WG chugging along for a while
>> and the SIP-XMPP work will not receive a lot of attention. If we can
>> move it over to a new, more focused WG then I am in favor, because that
>> will encourage faster progress, I think.
> 
> Will a new working group mean a whole set of fresh new members with more
> time?

Last I checked, fresh new members don't materialize out of thin air. :)

My point is that *if* a new SIP-XMPP group is being chartered anyway, it
would be more productive to have all the SIP-XMPP work happening there,
because (1) most people involved in the XMPP WG are only interested in
XMPP and the SIP-XMPP work is not top-of-mind for them, and (2) we might
be able to attract interest in SIP-XMPP work among implementers for whom
SIP-XMPP interworking/combinations are a pain point (that would include
client developers, server developers, and operators). If we reach out to
those folks and define a problem statement that captures what they are
seeing in the implementation and deployment space, then yes we might be
able to attract some fresh new members.

Peter

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


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From oej@edvina.net  Tue Sep 22 09:42:49 2009
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To: Peter Saint-Andre <stpeter@stpeter.im>
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Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM and	presence
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22 sep 2009 kl. 18.40 skrev Peter Saint-Andre:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> On 9/22/09 10:33 AM, Olle E. Johansson wrote:
>>
>> 22 sep 2009 kl. 18.06 skrev Peter Saint-Andre:
>>
>>> IMHO those activities will keep the XMPP WG chugging along for a  
>>> while
>>> and the SIP-XMPP work will not receive a lot of attention. If we can
>>> move it over to a new, more focused WG then I am in favor, because  
>>> that
>>> will encourage faster progress, I think.
>>
>> Will a new working group mean a whole set of fresh new members with  
>> more
>> time?
>
> Last I checked, fresh new members don't materialize out of thin  
> air. :)
I am aware of that issue... ;-)

>
> My point is that *if* a new SIP-XMPP group is being chartered  
> anyway, it
> would be more productive to have all the SIP-XMPP work happening  
> there,
> because (1) most people involved in the XMPP WG are only interested in
> XMPP and the SIP-XMPP work is not top-of-mind for them, and (2) we  
> might
> be able to attract interest in SIP-XMPP work among implementers for  
> whom
> SIP-XMPP interworking/combinations are a pain point (that would  
> include
> client developers, server developers, and operators). If we reach  
> out to
> those folks and define a problem statement that captures what they are
> seeing in the implementation and deployment space, then yes we might  
> be
> able to attract some fresh new members.

Agreed. I am very interested in the SIP-XMPP work.
I might just materialize at some point. Maybe not out of thin air  
though.

/O

From fluffy@cisco.com  Tue Sep 22 10:30:31 2009
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From: Cullen Jennings <fluffy@cisco.com>
To: rai@ietf.org, dispatch@ietf.org, sipping <sipping@ietf.org>, SIP List <sip@ietf.org>
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Subject: [dispatch] Fwd: Request for community guidance on issue concerning a future meeting of the IETF
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Please don't reply to this message. Go reply to the copy on the ietf@ietf.org 
  thread.

I often shy away from forwarding things to the RAI lists, however, I  
think that people need to be aware that there is conversation going on  
that is probably pretty important around if IETF 79 should be in China  
or Canada. I expect that coming out of this conversation will be not  
just where IETF 76 happens, but also guidelines or lore that impacts  
what type of environments the IETF expects to host meetings in that  
country.

Anyways, if you have any interest in this, go send your 2 cents to ietf@ietf.org 
  mailing list.

Thanks, Cullen


Begin forwarded message:

> From: Marshall Eubanks <tme@americafree.tv>
> Date: September 18, 2009 9:42:00 AM MDT (CA)
> To: IETF Announcement list <ietf-announce@ietf.org>, IETF-Discussion  
> list <ietf@ietf.org>, Working Group Chairs <wgchairs@ietf.org>
> Cc: IAOC Jabberr <iaoc@ietf.org>, IAB IAB <iab@iab.org>, IESG <iesg@ietf.org 
> >, irtf-chair@irtf.org
> Subject: Request for community guidance on issue concerning a future  
> meeting of the IETF
>
> Greetings;
>
> We have received numerous suggestions and requests for an IETF meeting
> in China and the IAOC has been working on a potential China meeting  
> for
> several years. We are now close to making a decision on a potential
> upcoming  meeting in China. However, the following issue has arisen
> and we would appreciate your feedback.
>
> The Chinese government has imposed a rule on all conferences held
> since 2008 regarding political speech. A fundamental law in China
> requires that one not criticize the government. Practically, this
> has reference to public political statements or protest marches, which
> are not the IETF's custom. The government, which is a party to the  
> issue,
> requires that people who attend conferences in China (the IETF being
> but one example) not engage in political speech during their tour
> in China. We consider this to be acceptable, on the basis that the
> IETF intends to abide by the laws of whatever nations it visits and
> we don't believe that this impacts our ability to do technical work.
>
> The rule is implemented in the Hotel agreement and reads (note that
> the "Client" would be the Host, and the "Group" would be the IETF) :
>
>   "Should the contents of the Group's activities, visual or audio
>   presentations at the conference,or printed materials used at the
>   conference (which are within the control of the Client) contain
>   any defamation against the Government of the People's Republic
>   of China, or show any disrespect to the Chinese culture, or
>   violates any laws of the People's Republic of China or feature
>   any topics regarding human rights or religion without prior
>   approval from the Government of the People's Republic of China,
>   the Hotel reserves the right to terminate the event on the spot
>   and/or ask the person(s) who initiates or participates in any or
>   all of the above action to leave the hotel premises immediately.
>
>   The Client will support and assist the Hotel with the necessary
>   actions to handle such situations. Should there be any financial
>   loss incurred to the Hotel or damage caused to the Hotel's
>   reputation as a result of any or all of the above acts, the Hotel
>   will claim compensation from the Client."
>
> What does this condition mean ? The hotel staff would have, in theory,
> the legal right to shut down the meeting and ask the offending
> participants to leave the property immediately. While we do not
> foresee a situation where such action would take place, we feel that
> it is proper to disclose these conditions to the community.
>
> The members of the IAOC, speaking as individuals, do not like this
> condition as a matter of principle. The IAOC does believe that this
> condition would not prevent the IETF from conducting its business.
>
> We note that the Vancouver/Quebec survey conducted earlier this year
> asked for people to suggest venues in Asia; an overwhelming majority
> (94%) of those who mentioned China were in favor of having a meeting
> there.
>
> We are therefore asking for input from the community by two means - by
> commenting on the IETF discussion list, and also by completing a very
> short survey on people's intentions to travel to China, or not,
> subject to these conditions. This survey can be found here :
>
> https://www.surveymonkey.com/s.aspx?sm=h4DUkRUOdG_2bVLqioPcYYHw_3d_3d
>
> All responses received by October 1, 2009 at  9:00 AM EDT  (1300 UTC)
> will be considered by the IAOC in making its decision. We appreciate
> the assistance of the community in providing us with data that will
> help us to make an informed decision.
>
> Regards
> Marshall Eubanks
> (acting for the IAOC)
>


From dean.willis@softarmor.com  Tue Sep 22 11:17:00 2009
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To: "<L.Liess@telekom.de>" <L.Liess@telekom.de>
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Subject: Re: [dispatch] Charter proposal:  SIP Alert-Info URN
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On Sep 22, 2009, at 9:12 AM, <L.Liess@telekom.de> <L.Liess@telekom.de>  
wrote:
>
> My understanding from your mail way that you know of other use cases
> from other mobile standards group.  Is that correct? If yes, does it
> make sense to consider them here? The syntax is open so people can add
> new labels.
>

It's been a while but I may remember conversations in 3GPP, 3GPP2, and  
OMA.

The ring-back for national standard ringers has always seemed like a a  
good application of URN in alert-info. In other words, if you call  
somebody in France, you might get back a 180 with an alert-info with a  
URN that basically means "France standard ring tone".

I vaguely remember discussion about URN-based alert-info for "public  
alert" calls, like storm warnings and evacuation orders.


There's also been discussion of use in forward-ring cases, as in alert- 
info in an INVITE request.

See http://www.ietf.org/mail-archive/web/sip/current/msg01385.html.  
This suggests several additional cases that may need to be considered,  
probably national variants of distinctive-ring indicators.


While we probably don't want to charter every possible use case, it  
would probably be appropriate to consider any published use cases from  
the mobile groups. So perhaps part of the task of the WG would be to  
review the extant standards and see if any of them should be added to  
the initial registry.

--
Dean


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From: "Richard Shockey" <richard@shockey.us>
To: "'Milan Patel'" <milanpa@nortel.com>, "'Gonzalo Camarillo'" <Gonzalo.Camarillo@ericsson.com>, "'DISPATCH'" <dispatch@ietf.org>
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Date: Tue, 22 Sep 2009 14:24:56 -0400
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Subject: Re: [dispatch] Cutoff for charter proposals on Monday
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I want to add some support to this idea but for a totally different reason.
This concept breaks open the issue that there have been some in the IETF
that objected to non-routing parameters used in TEL URI's and frankly this
bizarre restriction has caused me a world of grief.

Case in point.

http://tools.ietf.org/html/draft-ietf-enum-cnam-08


What I'd like to see is this issue settled once and for all. What is and is
not acceptable use of the TEL URI? 

If Calling Party Catagory is acceptable use of the TEL paramameter then I
want CNAM.  


>  -----Original Message-----
>  From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>  Behalf Of Milan Patel
>  Sent: Monday, September 21, 2009 5:35 AM
>  To: Gonzalo Camarillo; DISPATCH
>  Subject: Re: [dispatch] Cutoff for charter proposals on Monday
>  
>  Hi,
>  
>  I would like to add an additional topic to the DISPATCH charter - URI
>  parameters to describe the calling party category and toll class.
>  
>  Previously, Calling Party Category (CPC) URI parameter was defined in
>  a
>  draft by Rohan Mahy in the IPTEL group.
>  There is still a need by carriers for this parameter and it is
>  currently
>  also used within TISPAN and 3GPP.
>  The motivation for defining the CPC URI parameter is to provide a
>  standardized method of providing information about the calling party
>  and
>  the station used to originate a call. Currently, a number of
>  proprietary
>  solutions are being used as a standardized solution is unavailable.
>  
>  My intention is to revise Rohan's draft:
>  http://tools.ietf.org/html/draft-mahy-iptel-cpc-06
>  And to take on board some of the comments received on the old IPTEL
>  list.
>  My proposal is to provide a draft that defines 2 URI parameters: "cpc"
>  and "oli" indicating the category and tolls class of the calling
>  party,
>  allowing mapping to parameters used in ISUP.
>  
>  Best regards,
>  Milan
>  
>  Milan Patel
>  Carrier Networks Core Standards
>  Nortel
>  milanpa@nortel.com
>  Telephone +44 162 843 2381 / ESN 560 2381
>  Mobile +44 774 053 9261 / ESN 748 9261
>  
>  For the Companies listed below, The Institute of Chartered Accountants
>  in England and Wales authorises A R Bloom, S Harris and C Hill to act
>  as
>  Insolvency Practitioners under section 390(2)(a) of the Insolvency Act
>  1986 and the Association of Chartered Certified Accountants authorises
>  A
>  M Hudson to act as an Insolvency Practitioner under section 390(2)(a)
>  of
>  the Insolvency Act 1986.
>  
>  The affairs, business and property of the Companies are being managed
>  by
>  the Joint Administrators, A R Bloom, S Harris, AM Hudson and C Hill
>  who
>  act as agents of the Companies only and without personal liability.
>  
>  The Companies are Nortel Networks UK Limited; Nortel Networks SA;
>  Nortel
>  GmbH; Nortel Networks France SAS; Nortel Networks NV; Nortel Networks
>  SpA; Nortel Networks BV; Nortel Networks Polska SP Zoo; Nortel
>  Networks
>  Hispania SA; Nortel Networks (Austria) GmbH; Nortel Networks sro;
>  Nortel
>  Networks Engineering Service Kft; Nortel Networks Portugal SA; Nortel
>  Networks Slovensko sro; Nortel Networks Oy; Nortel Networks Romania
>  SRL;
>  Nortel Networks AB; Nortel Networks International Finance & Holding BV
>  -----Original Message-----
>  From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>  Behalf Of Gonzalo Camarillo
>  Sent: 16 September 2009 15:13
>  To: DISPATCH
>  Subject: [dispatch] Cutoff for charter proposals on Monday
>  
>  Folks,
>  
>  this is a reminder of our cutoff for charter proposals, which is on
>  Monday (September 21st). Per our previous email: "A charter proposal
>  must consist of a minimum of a problem statement and at least one
>  milestone/deliverable. This deadline will allow time for consideration
>  of proposals that could potentially be "dispatched" prior to the WG
>  session."
>  
>  A few of the topics we received arrived after the previous deadline,
>  which was September 7th. We will be considering even those topics that
>  arrived a bit late because we realize the deadline was quite tough to
>  meet (specially for people that returned from their vacation right
>  before the deadline).
>  
>  We are expecting charter proposals for:
>  
>  o Third-party authorization
>  o CBUS
>  o Session recording
>  o Identity
>  o Disaggregated media
>  o Domain registrations in SIP
>  o SIP - XMPP
>  o Alert-info URNs
>  o Reference to an earlier communication
>  
>  Let the chairs know if there are topics missing from this list.
>  
>  Thanks,
>  
>  Gonzalo
>  DISPATCH co-chair
>  _______________________________________________
>  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 richard@shockey.us  Tue Sep 22 11:36:59 2009
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From: "Richard Shockey" <richard@shockey.us>
To: "'Alan Johnston'" <alan@sipstation.com>, "'Dean Willis'" <dean.willis@softarmor.com>
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Cc: dispatch@ietf.org, 'Gonzalo Camarillo' <gonzalo.camarillo@ericsson.com>, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [dispatch] IETF-76 DISPATCH WG deadlines : New Topic Domain	Registration
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Alan is right ..the problem statement will be easier to explain once we have
a draft in hand. 

This is not really a proposal for a WG. There is only one draft here. 

What we need to explain is what is actually happening in the market with
certain classes of SIP-PBX deployments especially in Small and Medium
business markets.  

There are much larger issue here that should be addressed in the
provisioning of PBX to SSP networks but that is not the topic here.   

>  -----Original Message-----
>  From: Alan Johnston [mailto:alan@sipstation.com]
>  Sent: Friday, September 11, 2009 10:20 AM
>  To: Dean Willis
>  Cc: Richard Shockey; dispatch@ietf.org; 'Hadriel Kaplan'; 'Gonzalo
>  Camarillo'
>  Subject: Re: [dispatch] IETF-76 DISPATCH WG deadlines : New Topic
>  Domain Registration
>  
>  Dean,
>  
>  The plan is not to rewrite the Internet's name architecture nor
>  replace
>  RFC 3263.
>  
>  The plan is to standardize a function that has been shown to be needed
>  in the marketplace for the deployment of SIP in certain applications.
>  
>  And yes, dynamic DNS is an alternative, it does work, but again, there
>  are reasons why it is not being used in this certain application.
>  
>  Anyway, lets start this conversation properly when we have a draft.
>  
>  Thanks,
>  Alan
>  
>  
>  Dean Willis wrote:
>  >
>  > On Sep 4, 2009, at 2:53 PM, Richard Shockey wrote:
>  >
>  >> Dispatch Chairs:  We would like to reserve some time during
>  DISPATCH to
>  >> discuss the topic of domain registration in SIP.
>  >
>  > It seems like you're proposing a deep and fundamental rewrite of the
>  > Internet's name architecture, replacing RFC 3263 and its use of DNS
>  > with some sort of SIP REGISTER technique.
>  >
>  > Or maybe I'm missing it entirely, and what you're proposing is a way
>  > to use SIP REGISTER messages to populate a DNS SRV record within a
>  > registry?
>  >
>  > This raises the question of delegation to that registry -- the
>  > registry has to be authoritative for all of  example.com in order to
>  > be able to host an srv record for example.com. So your mythical SMB
>  is
>  > still going to require a DNS service provider that has to be
>  > meaningfully provisioned. It's just a question of the outsourcing
>  > contract, and I fail to see a need for protocol work here. After
>  all,
>  > we have dynamic DNS, and it works pretty darned well for me today.
>  >
>  > So what you talking 'bout, Shockey?
>  >
>  > --
>  > Dean
>  > _______________________________________________
>  > dispatch mailing list
>  > dispatch@ietf.org
>  > https://www.ietf.org/mailman/listinfo/dispatch
>  >



From christer.holmberg@ericsson.com  Tue Sep 22 12:41:04 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Enrico Marocco" <enrico.marocco@telecomitalia.it>
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Subject: Re: [dispatch] Charter proposed: CBUS
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Hi Enrico,
=20
If an even package registration, in addition to the body used in the =
NOTIFY request, also allows the definition of the XML body (and possilbe =
event header parameters) which is needed in the SUBSCRIBE request (in =
order to provide the conditions etc), then I agree that it may be =
enough.
=20
Regards,
=20
Christer

________________________________

From: Enrico Marocco [mailto:enrico.marocco@telecomitalia.it]
Sent: Tue 22/09/2009 16:38
To: Christer Holmberg
Cc: dispatch@ietf.org; Gonzalo Camarillo
Subject: Re: [dispatch] Charter proposed: CBUS



Christer, I see value in a general mechanism for selectively retrieve
lists of URIs, but am wondering whether this is actually a charter
proposal (in such case I guess the text requires heavy rewording,
defining acronyms as a start) or just an event package registration
request according to rfc3427bis, S.4.1?

Enrico

Christer Holmberg wrote:
>
> Hi,
>
> Below is the proposed charter for CBUS.
>
> Regards,
>
> Christer
>
> -----------------------------------------
>
> CBUS Charter.
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>
> Introduction:
> -------------
>
> Various OMA service enablers need to be able to retrieve a list of
> addresses (URIs), where the users associated with the URIs fulfil
> certain conditions.  User information is evaluated against the
> conditions, and the matches form a URI list.
>
> The need for the functionality originates from the OMA PoC service.
> However, there is ongoing work in OMA to define a common mechanism,
> called CBUS enabler (OMA-RD-CBUS-V1_0-20090203-C), to provide the
> functionality, so that it can be used for various types of services
> (e.g. messaging, gaming, conferencing and advertisement).
>
> The CBUS enabler provides the following functions:
>
> - Support for requestor initiated condition-based URIs selection =
requests.
>
> - Administration of conditions for URI selection.
>
> - Interaction with other enablers for retrieval of individual user's
>   information (e.g. presence information, location information).
>
> - Evaluation of conditions and selection of matching individual users
>   based on one time evaluation.
>
> - Re-evaluation of conditions and re-selection of matching individual
>   users based on monitoring.
>
> - Aggregation of one-time evaluation results and monitoring results
>   from different information sources.
>
> - Notification of evaluation results to requestor.
>
> The conditions can be based on user information which changes over =
time
> (e.g. presence information or geographical location), but they can =
also
> be based on more static user information (e.g. hobbies or personal
> interests).
>
> The entity which acts as requester, and provides the conditions to be
> used for the user information evaluation, is called CBUS client.  The
> CBUS client usually resides on the user equipment, or an application
> server, which supports the CBUS enabler.
>
> The selection of URIs, based on the provided conditions, may be
> performed by taking a snapshot, i.e. by making a one-time evaluation =
of
> the user information to find out whether the conditions are fulfilled =
or
> not.  The URI selection may also be performed by continous monitoring =
of
> the user information.
>
>
> Work and deliverables:
> ----------------------
>
> The purpose of the work is to, based on the OMA CBUS requirements =
(OMA-
> RD-CBUS-V1_0-20090203-C), procude the SIP protocol information =
elements
> which are needed to implement the interface between the CBUS client =
and
> CBUS server, using the SIP SUBSCRIBE/NOTIFY framework (RFC 3265).
>
>
> The primary task for the work is to:
>
> 1. Procude a mechanism for a subscriber (CBUS client) to, when
> subscribing to an event package which contains a list of URI, provide
> conditions, candidate URIs and evaluation parameters to the notifier
> (CBUS server).
>
> The mechanism may be used on defining an XML based message body type,
> and SIP header field parameters.
>
> 2. Produce an event package, or possible re-use an existing event
> package, used to the notifier (CBUS server) to send URI lists (based =
on
> the conditions and candidate URIs) to the subscriber (CBUS client).
>
> The CBUS client actions triggered by the received URI list, and the
> interactions between the CBUS server and other enablers, are outside =
the
> scope of the work.
>
>
> The work is planned to take place in, or in co-operaton with, the
> SIPCORE WG.
>
>
> Goals and milestones:
> ---------------------
>
> Jan 2010        Working Group Last Call for the document which =
contains
> the mechanism for a
>             subscriber to provide the needed CBUS information to the
> notifier, and
>             contains the event package (if a new package is needed) =
for the
>             notifier to provide the URI list to the subscriber.
>
> Mar 2010    Submit document to IESG as Proposed Standard
>



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Date: Tue, 22 Sep 2009 21:11:42 -0400
From: Paul Kyzivat <pkyzivat@cisco.com>
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Cc: dispatch@ietf.org, "<L.Liess@telekom.de>" <L.Liess@telekom.de>
Subject: Re: [dispatch] Charter proposal:  SIP Alert-Info URN
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I agree with Dean here.

The example of the "France standard ring tone" is a case where you have 
a URN that identifies some specific audio signal, but in a semantic way 
rather than by providing an encoding of it. This can be useful for a 
variety of reasons:
- its not biased towards a particular representation,
   which might not be suitable for all devices
- its convenient when you want to store the actual encoding
   locally rather than fetching it
- because it is specified "by name" rather than "by value",
   its easier to make policy decisions about whether to use it or not.
- it facilitates translation for the hearing impaired

	Thanks,
	Paul

Dean Willis wrote:
> 
> On Sep 22, 2009, at 9:12 AM, <L.Liess@telekom.de> <L.Liess@telekom.de> 
> wrote:
>>
>> My understanding from your mail way that you know of other use cases
>> from other mobile standards group.  Is that correct? If yes, does it
>> make sense to consider them here? The syntax is open so people can add
>> new labels.
>>
> 
> It's been a while but I may remember conversations in 3GPP, 3GPP2, and OMA.
> 
> The ring-back for national standard ringers has always seemed like a a 
> good application of URN in alert-info. In other words, if you call 
> somebody in France, you might get back a 180 with an alert-info with a 
> URN that basically means "France standard ring tone".
> 
> I vaguely remember discussion about URN-based alert-info for "public 
> alert" calls, like storm warnings and evacuation orders.
> 
> 
> There's also been discussion of use in forward-ring cases, as in 
> alert-info in an INVITE request.
> 
> See http://www.ietf.org/mail-archive/web/sip/current/msg01385.html. This 
> suggests several additional cases that may need to be considered, 
> probably national variants of distinctive-ring indicators.
> 
> 
> While we probably don't want to charter every possible use case, it 
> would probably be appropriate to consider any published use cases from 
> the mobile groups. So perhaps part of the task of the WG would be to 
> review the extant standards and see if any of them should be added to 
> the initial registry.
> 
> -- 
> Dean
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

From dean.willis@softarmor.com  Tue Sep 22 20:34:47 2009
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Subject: Re: [dispatch] IETF-76 DISPATCH WG deadlines : New Topic Domain	Registration
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On Sep 11, 2009, at 9:20 AM, Alan Johnston wrote:

> Dean,
>
> The plan is not to rewrite the Internet's name architecture nor  
> replace RFC 3263.
>
> The plan is to standardize a function that has been shown to be  
> needed in the marketplace for the deployment of SIP in certain  
> applications.
>
> And yes, dynamic DNS is an alternative, it does work, but again,  
> there are reasons why it is not being used in this certain  
> application.
>
> Anyway, lets start this conversation properly when we have a draft.
>
> Thanks,

Ah, so you're not really "registering a domain with the Internet using  
SIP", you're "registering a domain-wide set of AORs with a routing  
proxy".

So here's the use case I'm envisioning. The DNS SRV record for  
example.org points statically to a service provider's  routing proxy/ 
registrar. This gets provisioned into the DNS when example.org  
contracts with the service provider.

When the example.org PBX powers up, it needs to register with the   
proxy/registrar. But rather than running a separate REGISTER  
transaction for each user in example.org, you want to use a single  
transaction to tell the proxy/registrar "If it's for anybody at  
example.org, send it to this Contact". One way to approach this might  
be to use a to/from of "*.example.org" in the REGISTER method (which  
would be an extension to RFC 3261), along with appropriate  
authentication.

This is something like the bulk-registration problem for gateways,  
where people have hacked REGISTER to indicate a range of phone  
numbers. It has the advantage of being able to use qvalue  
prioritization to have some users at example.org register directly  
with the service provider, bypassing the PBX.

But I worry that it's still a serious misuse of the REGISTER model.  
We're not really registering a contact; what we're really doing is  
exchanging routing information for a whole bunch of contacts.

Where it breaks down is when example,org's node is not a PBX (a UA, or  
even B2BUA) but is instead a real proxy (I know, there's no such  
thing ...). The subtle distinction is in the identity presentation,  
and its coupling to TLS.

I shall try to construct an example for clarification.

Suppose Bob calls "alice@example.org", and the service provider issues  
a 302 with the previously registered Contact value. Bob's UA is now  
expecting to talk to Alice's UA, via SIPS/TLS. What certificate does  
the PBX present at the TLS level? is it the certificate of "alice@example.org 
", or is it the certificate of "pbx.example.org"? Does this induce an  
unanticipated respondent problem for Bob?

Alternatively suppose a 305 was used. Bob's UA now expects to be  
talking SIPS/TLS to a proxy that knows how to find Alice. How is the  
behavior different?

This brings in a whole raft of unclarities around TLS, domain usage,   
certificate/identity coupling, and 302 vs 305 behavior that combine to  
make me a very confused puppy. It might be that the Domain Certs draft  
has all this cleared up and I just missed it. It also relates to that  
long-running "what is an AOR" debate.

But it's certainly more useful if the service provider's proxy/ 
registrar knows it is getting a route towards an AOR instead of a  
contact for that AOR. Sure, we could extend REGISTER to convey this  
information. Once we've done that, can REGISTER be used as a general- 
purpose SIP route exchange mechanism? Or would we be better served by  
building a real routing protocol, ala TRIP?

So given this discussion: I know you like to jump directly to a  
solution, and "Domain Registration" with its similarity to that  
private PBX-club SIP specification hint at the solution you might be  
approaching. But I'd really like to take a step back here and get  
clarity on the problem we're trying to solve before leaping to a  
solution, because I have a strong suspicion that we're leading towards  
the wrong solution to a poorly-defined problem that is itself a subset  
of an interesting more-general problem.

The "domain registration" concept, as I understand it, seems to be a  
subset of the broader SIP routing-exchange problem. This broader  
problem I see as becoming increasingly critical in the era of P2P  
overlays, as it starts to address inter-overlay routing and might even  
address questions like "What happens if some users in example.org are  
in overlay A using CHORD and some are in overlay B using PASTRY". This  
might not be as tactical a solution as hacking REGISTER, but I think  
it's a real problem and that there are tractable solutions that would  
fit into the scope of a working group.

--
Dean

--
Dean

From R.Jesske@telekom.de  Tue Sep 22 21:29:30 2009
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Dear all,
here is a proposal for the Charter. I missed to sent it to the whole
list. I'm sorry for that.
So here for discussion:
=20
The use of the Reason header field in responses is considered in
RFC3326, doing so is not specified for any existing response code.=20
Nevertheless for interoperability reason between SIP and PSTN such
behaviour is needed and already deployed within many networks.
A guideline to permit the use of the Reason header within Response is
needed.
=20
Milestone:
May 2010 Guideline for Reason in Responses  to ISEG
=20
Best Regards
=20
Roland

________________________________

Von: Jesske, Roland=20
Gesendet: Donnerstag, 17. September 2009 14:22
An: 'Mary Barnes'; 'Gonzalo Camarillo'
Betreff: AW: [dispatch] IETF-76 DISPATCH WG deadlines


Hi Mary and Gonzalo,
with regard to the mail of this morning I saw that you have already
mentioned the reason issue.
Looking on this mail it seems to me that I have to write something for
the Charter.
So I try my best:
=20
The use of the Reason header field in responses is considered in
RFC3326, doing so is not specified for any existing response code.=20
Nevertheless for interoperability reason between SIP and PSTN such
behaviour is needed and already deployed within many networks.
A guideline to permit the use of the Reason header within Response is
needed.
=20
Milestone:
May 2010 Guideline for Reason in Responses  to ISEG
=20
Is the milestone correct or do I have to differentiate?
=20
I hope everything is correct. Commons?
=20
Thank you and Best Regards
=20
Roland

________________________________

Von: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] Im
Auftrag von Mary Barnes
Gesendet: Freitag, 21. August 2009 18:25
An: dispatch@ietf.org
Cc: rai-ads@tools.ietf.org
Betreff: [dispatch] IETF-76 DISPATCH WG deadlines



Hi all,=20

As was done for IETF-75, we are setting earlier deadlines for discussing
topics in DISPATCH prior to the WG session at IETF-76. =20

We will again follow the deadlines for BoFs:=20
http://www.ietf.org/meeting/cutoff-dates.html#76
<http://www.ietf.org/meeting/cutoff-dates.html#76> =20

This provides enough time to dispatch topics prior to the meeting as
appropriate - e.g., mini-bofs, official bofs, other WGs, new WGs,
mailing lists, etc. Thus, allowing us to have more focused and
constructive discussions on a smaller set of topics prior to the WG
session.=20

Please note that the dates are much sooner than you might expect as the
time between IETF-75 and IETF-76 is 3 weeks less than the time between
IETF-74 and IETF-75. The document deadlines are 9 and 10 weeks away from
next Monday (August 24th). =20


Thus, the following are the proposed deadlines:=20

Sept 7th  - Cutoff date to notify the chairs and DISPATCH WG mailing
list of plans to submit a proposal.=20
------------------------------------------------------------------------
--------------------------------=20

This will help folks with similar topics to work together before the
meeting. If a preliminary charter proposal is available at this point,
please include it.


September 21st - Cutoff for charter proposals for topics.=20
--------------------------------------------------------=20

A charter proposal must consist of a minimum of a problem statement and
at least one milestone/deliverable. This deadline will allow time for
consideration of proposals that could potentially be "dispatched" prior
to the WG session.


September 28th - Topics that are to be the focus of IETF-76 are
announced.=20
------------------------------------------------------------------------
---=20

The idea here is to focus the discussion over the weeks preceding
IETF-76 on these main topics, noting that any new or updates to any
documents are due 3-4 weeks later.  This will ensure we have
constructive discussions before the meeting and are actually able to
determine consensus as to where work should be progressed (e.g.,
separate BoF, a new WG, an existing WG, individual document, etc.) at
IETF-76. Note, that the exact disposition for a topic may (per the usual
process) require follow-up and confirmation by the ADS and/or IESG
(e.g., for a new WG, Bof, individually sponsored document, etc.) or with
another WG to ensure agreement with the DISPATCH WG consensus for the
topic.


As discussed previously, the intention is not necessarily to preclude
folks submitting documents on other topics, however, it is very unlikely
there would be agenda time allocated to documents/topics submitted after
the deadlines. We can include one or two slides on these topics in the
DISPATCH WG chair charts so that the authors can gather other interested
parties to contribute to the work.=20

Regards,=20
Mary and Gonzalo=20
DISPATCH WG co-chairs=20


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<HTML><HEAD><TITLE>IETF-76 DISPATCH WG deadlines</TITLE>

<META content=3D"MSHTML 6.00.2900.3603" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D282552704-23092009>Dear all,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D282552704-23092009>here is a proposal for the Charter. I missed =
to sent it=20
to the whole list. I'm sorry for that.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D282552704-23092009>So here for discussion:</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D282552704-23092009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D093045311-17092009>The use of the Reason header field in =
responses is=20
considered in RFC3326, doing so is not specified for any existing =
response code.=20
</SPAN></FONT></DIV>
<DIV><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
class=3D093045311-17092009>N</SPAN>evertheless&nbsp;for&nbsp;interoperabi=
lity&nbsp;<SPAN=20
class=3D093045311-17092009>reason=20
</SPAN>between&nbsp;SIP&nbsp;and&nbsp;PSTN&nbsp;such&nbsp;behaviour&nbsp;=
is&nbsp;needed&nbsp;and&nbsp;already&nbsp;deployed&nbsp;within&nbsp;many&=
nbsp;networks.</FONT></FONT></FONT></DIV>
<DIV><SPAN class=3D093045311-17092009></SPAN><SPAN=20
class=3D093045311-17092009></SPAN><FONT face=3DArial><FONT =
color=3D#0000ff><FONT=20
size=3D2>A<SPAN class=3D093045311-17092009> guideline to permit the use =
of the=20
Reason header within Response is =
needed.</SPAN></FONT></FONT></FONT></DIV>
<DIV><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
class=3D093045311-17092009></SPAN></FONT></FONT></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
class=3D093045311-17092009>Milestone:</SPAN></FONT></FONT></FONT></DIV>
<DIV><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
class=3D093045311-17092009>May 2010 Guideline for Reason in =
Responses&nbsp; to=20
ISEG</SPAN></FONT></FONT></FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><SPAN class=3D282552704-23092009></SPAN><FONT face=3DArial><FONT=20
color=3D#0000ff><FONT size=3D2>B<SPAN class=3D282552704-23092009>est=20
Regards</SPAN></FONT></FONT></FONT></DIV>
<DIV><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
class=3D282552704-23092009></SPAN></FONT></FONT></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
class=3D282552704-23092009>Roland</SPAN></FONT></FONT></FONT></DIV>
<DIV><BR></DIV>
<DIV class=3DOutlookMessageHeader lang=3Dde dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>Von:</B> Jesske, Roland =
<BR><B>Gesendet:</B>=20
Donnerstag, 17. September 2009 14:22<BR><B>An:</B> 'Mary Barnes'; =
'Gonzalo=20
Camarillo'<BR><B>Betreff:</B> AW: [dispatch] IETF-76 DISPATCH WG=20
deadlines<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D093045311-17092009>Hi Mary and Gonzalo,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D093045311-17092009>with regard to the mail of this morning I saw =
that you=20
have already mentioned the reason issue.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D093045311-17092009>Looking on this mail it seems to me that I =
have to=20
write something for the Charter.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D093045311-17092009>So I try my best:</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D093045311-17092009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D093045311-17092009>The use of the Reason header field in =
responses is=20
considered in RFC3326, doing so is not specified for any existing =
response code.=20
</SPAN></FONT></DIV>
<DIV><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
class=3D093045311-17092009>N</SPAN>evertheless&nbsp;for&nbsp;interoperabi=
lity&nbsp;<SPAN=20
class=3D093045311-17092009>reason=20
</SPAN>between&nbsp;SIP&nbsp;and&nbsp;PSTN&nbsp;such&nbsp;behaviour&nbsp;=
is&nbsp;needed&nbsp;and&nbsp;already&nbsp;deployed&nbsp;within&nbsp;many&=
nbsp;networks.</FONT></FONT></FONT></DIV>
<DIV><SPAN class=3D093045311-17092009></SPAN><SPAN=20
class=3D093045311-17092009></SPAN><FONT face=3DArial><FONT =
color=3D#0000ff><FONT=20
size=3D2>A<SPAN class=3D093045311-17092009> guideline to permit the use =
of the=20
Reason header within Response is =
needed.</SPAN></FONT></FONT></FONT></DIV>
<DIV><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
class=3D093045311-17092009></SPAN></FONT></FONT></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
class=3D093045311-17092009>Milestone:</SPAN></FONT></FONT></FONT></DIV>
<DIV><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
class=3D093045311-17092009>May 2010 Guideline for Reason in =
Responses&nbsp; to=20
ISEG</SPAN></FONT></FONT></FONT></DIV>
<DIV><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
class=3D093045311-17092009></SPAN></FONT></FONT></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
class=3D093045311-17092009>Is the milestone correct or do I have to=20
differentiate?</SPAN></FONT></FONT></FONT></DIV>
<DIV><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
class=3D093045311-17092009></SPAN></FONT></FONT></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
class=3D093045311-17092009>I hope everything is correct.=20
Commons?</SPAN></FONT></FONT></FONT></DIV>
<DIV><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
class=3D093045311-17092009></SPAN></FONT></FONT></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
class=3D093045311-17092009>Thank you and Best=20
Regards</SPAN></FONT></FONT></FONT></DIV>
<DIV><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
class=3D093045311-17092009></SPAN></FONT></FONT></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial><FONT color=3D#0000ff><FONT size=3D2><SPAN=20
class=3D093045311-17092009>Roland</SPAN></FONT></FONT></FONT></DIV>
<DIV><BR></DIV>
<DIV class=3DOutlookMessageHeader lang=3Dde dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>Von:</B> dispatch-bounces@ietf.org=20
[mailto:dispatch-bounces@ietf.org] <B>Im Auftrag von </B>Mary=20
Barnes<BR><B>Gesendet:</B> Freitag, 21. August 2009 18:25<BR><B>An:</B>=20
dispatch@ietf.org<BR><B>Cc:</B> =
rai-ads@tools.ietf.org<BR><B>Betreff:</B>=20
[dispatch] IETF-76 DISPATCH WG deadlines<BR></FONT><BR></DIV>
<DIV></DIV><!-- Converted from text/rtf format -->
<P><FONT face=3D"Courier New" size=3D2>Hi all,</FONT> </P>
<P><FONT face=3D"Courier New" size=3D2>As was done for IETF-75, we are =
setting=20
earlier deadlines for discussing topics in DISPATCH prior to the WG =
session at=20
IETF-76.&nbsp; </FONT></P>
<P><FONT face=3D"Courier New" size=3D2>We will again follow the =
deadlines for=20
BoFs:</FONT> <BR><A=20
href=3D"http://www.ietf.org/meeting/cutoff-dates.html#76"><U><FONT=20
face=3D"Courier New" color=3D#0000ff=20
size=3D2>http://www.ietf.org/meeting/cutoff-dates.html#76</FONT></U></A> =
</P>
<P><FONT face=3D"Courier New" size=3D2>This provides enough time to =
dispatch topics=20
prior to the meeting as appropriate - e.g., mini-bofs, official bofs, =
other WGs,=20
new WGs,&nbsp; mailing lists, etc. Thus, allowing us to have more =
focused and=20
constructive discussions on a smaller set of topics prior to the WG =
session.=20
</FONT></P>
<P><FONT face=3D"Courier New" size=3D2>Please note that the dates are =
much sooner=20
than you might expect as the time between IETF-75 and IETF-76 is 3 weeks =
less=20
than the time between IETF-74 and IETF-75. The document deadlines are 9 =
and 10=20
weeks away from next Monday (August 24th).&nbsp; </FONT></P><BR>
<P><FONT face=3D"Courier New" size=3D2>Thus, the following are the =
proposed=20
deadlines:</FONT> </P>
<P><FONT face=3D"Courier New" size=3D2>Sept 7th&nbsp; - Cutoff date to =
notify the=20
chairs and DISPATCH WG mailing list of plans to submit a =
proposal.</FONT>=20
<BR><FONT face=3D"Courier New"=20
size=3D2>----------------------------------------------------------------=
----------------------------------------=20
</FONT></P>
<P><FONT face=3D"Courier New" size=3D2>This will help folks with similar =
topics to=20
work together before the meeting. If a preliminary charter proposal is =
available=20
at this point, please include it.</FONT></P><BR>
<P><FONT face=3D"Courier New" size=3D2>September 21st - Cutoff for =
charter proposals=20
for topics. </FONT><BR><FONT face=3D"Courier New"=20
size=3D2>--------------------------------------------------------</FONT> =
</P>
<P><FONT face=3D"Courier New" size=3D2>A charter proposal must consist =
of a minimum=20
of a problem statement and at least one milestone/deliverable. This =
deadline=20
will allow time for consideration of proposals that could potentially be =

"dispatched" prior to the WG session.</FONT></P><BR>
<P><FONT face=3D"Courier New" size=3D2>September 28th - Topics that are =
to be the=20
focus of IETF-76 are announced. </FONT><BR><FONT face=3D"Courier New"=20
size=3D2>----------------------------------------------------------------=
-----------=20
</FONT></P>
<P><FONT face=3D"Courier New" size=3D2>The idea here is to focus the =
discussion over=20
the weeks preceding IETF-76 on these main topics, noting that any new or =
updates=20
to any documents are due 3-4 weeks later.&nbsp; This will ensure we have =

constructive discussions before the meeting and are actually able to =
determine=20
consensus as to where work should be progressed (e.g., separate BoF, a =
new WG,=20
an existing WG, individual document, etc.) at IETF-76. Note, that the =
exact=20
disposition for a topic may (per the usual process) require follow-up =
and=20
confirmation by the ADS and/or IESG (e.g., for a new WG, Bof, =
individually=20
sponsored document, etc.) or with another WG to ensure agreement with =
the=20
DISPATCH WG consensus for the topic.</FONT></P><BR>
<P><FONT face=3D"Courier New" size=3D2>As discussed previously, the =
intention is not=20
necessarily to preclude folks submitting documents on other topics, =
however, it=20
is very unlikely there would be agenda time allocated to =
documents/topics=20
submitted after the deadlines. We can include one or two slides on these =
topics=20
in the DISPATCH WG chair charts so that the authors can gather other =
interested=20
parties to contribute to the work. </FONT></P>
<P><FONT face=3D"Courier New" size=3D2>Regards,</FONT> <BR><FONT =
face=3D"Courier New"=20
size=3D2>Mary and Gonzalo</FONT> <BR><FONT face=3D"Courier New" =
size=3D2>DISPATCH WG=20
co-chairs </FONT></P></BODY></HTML>

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From enrico.marocco@telecomitalia.it  Wed Sep 23 00:11:50 2009
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Subject: Re: [dispatch] Charter proposed: CBUS
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Well, I'm probably not the right person to answer this, but as I
understand the new RAI process, a non-template even package (that as per
3265 includes definitions of NOTIFY and SUBSCRIBE bodies as well as
Event header parameters) that falls outside the scope of any existing WG
may (should?) be submitted to dispatch as an individual contribution.

However, since this would be the first time S.4.1 is run, I'd suggest to
do that under the guidance of RAI management since the very beginning.

Enrico

Christer Holmberg wrote:
> Hi Enrico,
> 
> If an even package registration, in addition to the body used in the
> NOTIFY request, also allows the definition of the XML body (and
> possilbe event header parameters) which is needed in the SUBSCRIBE
> request (in order to provide the conditions etc), then I agree that
> it may be enough.
> 
> Regards,
> 
> Christer
> 
> ________________________________
> 
> From: Enrico Marocco [mailto:enrico.marocco@telecomitalia.it] Sent:
> Tue 22/09/2009 16:38 To: Christer Holmberg Cc: dispatch@ietf.org;
> Gonzalo Camarillo Subject: Re: [dispatch] Charter proposed: CBUS
> 
> 
> 
> Christer, I see value in a general mechanism for selectively retrieve
> lists of URIs, but am wondering whether this is actually a charter 
> proposal (in such case I guess the text requires heavy rewording, 
> defining acronyms as a start) or just an event package registration 
> request according to rfc3427bis, S.4.1?
> 
> Enrico
> 
> Christer Holmberg wrote:
>> Hi,
>> 
>> Below is the proposed charter for CBUS.
>> 
>> Regards,
>> 
>> Christer
>> 
>> -----------------------------------------
>> 
>> CBUS Charter. =============
>> 
>> Introduction: -------------
>> 
>> Various OMA service enablers need to be able to retrieve a list of 
>> addresses (URIs), where the users associated with the URIs fulfil 
>> certain conditions.  User information is evaluated against the 
>> conditions, and the matches form a URI list.
>> 
>> The need for the functionality originates from the OMA PoC service.
>>  However, there is ongoing work in OMA to define a common
>> mechanism, called CBUS enabler (OMA-RD-CBUS-V1_0-20090203-C), to
>> provide the functionality, so that it can be used for various types
>> of services (e.g. messaging, gaming, conferencing and
>> advertisement).
>> 
>> The CBUS enabler provides the following functions:
>> 
>> - Support for requestor initiated condition-based URIs selection
>> requests.
>> 
>> - Administration of conditions for URI selection.
>> 
>> - Interaction with other enablers for retrieval of individual
>> user's information (e.g. presence information, location
>> information).
>> 
>> - Evaluation of conditions and selection of matching individual
>> users based on one time evaluation.
>> 
>> - Re-evaluation of conditions and re-selection of matching
>> individual users based on monitoring.
>> 
>> - Aggregation of one-time evaluation results and monitoring results
>>  from different information sources.
>> 
>> - Notification of evaluation results to requestor.
>> 
>> The conditions can be based on user information which changes over
>> time (e.g. presence information or geographical location), but they
>> can also be based on more static user information (e.g. hobbies or
>> personal interests).
>> 
>> The entity which acts as requester, and provides the conditions to
>> be used for the user information evaluation, is called CBUS client.
>> The CBUS client usually resides on the user equipment, or an
>> application server, which supports the CBUS enabler.
>> 
>> The selection of URIs, based on the provided conditions, may be 
>> performed by taking a snapshot, i.e. by making a one-time
>> evaluation of the user information to find out whether the
>> conditions are fulfilled or not.  The URI selection may also be
>> performed by continous monitoring of the user information.
>> 
>> 
>> Work and deliverables: ----------------------
>> 
>> The purpose of the work is to, based on the OMA CBUS requirements
>> (OMA- RD-CBUS-V1_0-20090203-C), procude the SIP protocol
>> information elements which are needed to implement the interface
>> between the CBUS client and CBUS server, using the SIP
>> SUBSCRIBE/NOTIFY framework (RFC 3265).
>> 
>> 
>> The primary task for the work is to:
>> 
>> 1. Procude a mechanism for a subscriber (CBUS client) to, when 
>> subscribing to an event package which contains a list of URI,
>> provide conditions, candidate URIs and evaluation parameters to the
>> notifier (CBUS server).
>> 
>> The mechanism may be used on defining an XML based message body
>> type, and SIP header field parameters.
>> 
>> 2. Produce an event package, or possible re-use an existing event 
>> package, used to the notifier (CBUS server) to send URI lists
>> (based on the conditions and candidate URIs) to the subscriber
>> (CBUS client).
>> 
>> The CBUS client actions triggered by the received URI list, and the
>>  interactions between the CBUS server and other enablers, are
>> outside the scope of the work.
>> 
>> 
>> The work is planned to take place in, or in co-operaton with, the 
>> SIPCORE WG.
>> 
>> 
>> Goals and milestones: ---------------------
>> 
>> Jan 2010        Working Group Last Call for the document which
>> contains the mechanism for a subscriber to provide the needed CBUS
>> information to the notifier, and contains the event package (if a
>> new package is needed) for the notifier to provide the URI list to
>> the subscriber.
>> 
>> Mar 2010    Submit document to IESG as Proposed Standard

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From christer.holmberg@ericsson.com  Wed Sep 23 00:16:49 2009
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Thread-Topic: [dispatch] Charter proposed: CBUS
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Enrico Marocco" <enrico.marocco@telecomitalia.it>
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Cc: dispatch@ietf.org, Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>
Subject: Re: [dispatch] Charter proposed: CBUS
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Hi,

I guess the chairs can give us guidance on this. 

I don't expect a new WG to be formed for this, but we were asked to provide
a charter, so that's what I did :)

Regards,

Christer 

> -----Original Message-----
> From: Enrico Marocco [mailto:enrico.marocco@telecomitalia.it] 
> Sent: 23. syyskuuta 2009 10:13
> To: Christer Holmberg
> Cc: dispatch@ietf.org; Gonzalo Camarillo
> Subject: Re: [dispatch] Charter proposed: CBUS
> 
> Well, I'm probably not the right person to answer this, but 
> as I understand the new RAI process, a non-template even 
> package (that as per
> 3265 includes definitions of NOTIFY and SUBSCRIBE bodies as 
> well as Event header parameters) that falls outside the scope 
> of any existing WG may (should?) be submitted to dispatch as 
> an individual contribution.
> 
> However, since this would be the first time S.4.1 is run, I'd 
> suggest to do that under the guidance of RAI management since 
> the very beginning.
> 
> Enrico
> 
> Christer Holmberg wrote:
> > Hi Enrico,
> > 
> > If an even package registration, in addition to the body 
> used in the 
> > NOTIFY request, also allows the definition of the XML body (and 
> > possilbe event header parameters) which is needed in the SUBSCRIBE 
> > request (in order to provide the conditions etc), then I 
> agree that it 
> > may be enough.
> > 
> > Regards,
> > 
> > Christer
> > 
> > ________________________________
> > 
> > From: Enrico Marocco [mailto:enrico.marocco@telecomitalia.it] Sent:
> > Tue 22/09/2009 16:38 To: Christer Holmberg Cc: dispatch@ietf.org; 
> > Gonzalo Camarillo Subject: Re: [dispatch] Charter proposed: CBUS
> > 
> > 
> > 
> > Christer, I see value in a general mechanism for 
> selectively retrieve 
> > lists of URIs, but am wondering whether this is actually a charter 
> > proposal (in such case I guess the text requires heavy rewording, 
> > defining acronyms as a start) or just an event package registration 
> > request according to rfc3427bis, S.4.1?
> > 
> > Enrico
> > 
> > Christer Holmberg wrote:
> >> Hi,
> >> 
> >> Below is the proposed charter for CBUS.
> >> 
> >> Regards,
> >> 
> >> Christer
> >> 
> >> -----------------------------------------
> >> 
> >> CBUS Charter. =============
> >> 
> >> Introduction: -------------
> >> 
> >> Various OMA service enablers need to be able to retrieve a list of 
> >> addresses (URIs), where the users associated with the URIs fulfil 
> >> certain conditions.  User information is evaluated against the 
> >> conditions, and the matches form a URI list.
> >> 
> >> The need for the functionality originates from the OMA PoC service.
> >>  However, there is ongoing work in OMA to define a common 
> mechanism, 
> >> called CBUS enabler (OMA-RD-CBUS-V1_0-20090203-C), to provide the 
> >> functionality, so that it can be used for various types of 
> services 
> >> (e.g. messaging, gaming, conferencing and advertisement).
> >> 
> >> The CBUS enabler provides the following functions:
> >> 
> >> - Support for requestor initiated condition-based URIs selection 
> >> requests.
> >> 
> >> - Administration of conditions for URI selection.
> >> 
> >> - Interaction with other enablers for retrieval of 
> individual user's 
> >> information (e.g. presence information, location information).
> >> 
> >> - Evaluation of conditions and selection of matching 
> individual users 
> >> based on one time evaluation.
> >> 
> >> - Re-evaluation of conditions and re-selection of matching 
> individual 
> >> users based on monitoring.
> >> 
> >> - Aggregation of one-time evaluation results and 
> monitoring results  
> >> from different information sources.
> >> 
> >> - Notification of evaluation results to requestor.
> >> 
> >> The conditions can be based on user information which changes over 
> >> time (e.g. presence information or geographical location), 
> but they 
> >> can also be based on more static user information (e.g. hobbies or 
> >> personal interests).
> >> 
> >> The entity which acts as requester, and provides the 
> conditions to be 
> >> used for the user information evaluation, is called CBUS client.
> >> The CBUS client usually resides on the user equipment, or an 
> >> application server, which supports the CBUS enabler.
> >> 
> >> The selection of URIs, based on the provided conditions, may be 
> >> performed by taking a snapshot, i.e. by making a one-time 
> evaluation 
> >> of the user information to find out whether the conditions are 
> >> fulfilled or not.  The URI selection may also be performed by 
> >> continous monitoring of the user information.
> >> 
> >> 
> >> Work and deliverables: ----------------------
> >> 
> >> The purpose of the work is to, based on the OMA CBUS requirements
> >> (OMA- RD-CBUS-V1_0-20090203-C), procude the SIP protocol 
> information 
> >> elements which are needed to implement the interface 
> between the CBUS 
> >> client and CBUS server, using the SIP SUBSCRIBE/NOTIFY 
> framework (RFC 
> >> 3265).
> >> 
> >> 
> >> The primary task for the work is to:
> >> 
> >> 1. Procude a mechanism for a subscriber (CBUS client) to, when 
> >> subscribing to an event package which contains a list of 
> URI, provide 
> >> conditions, candidate URIs and evaluation parameters to 
> the notifier 
> >> (CBUS server).
> >> 
> >> The mechanism may be used on defining an XML based message 
> body type, 
> >> and SIP header field parameters.
> >> 
> >> 2. Produce an event package, or possible re-use an existing event 
> >> package, used to the notifier (CBUS server) to send URI 
> lists (based 
> >> on the conditions and candidate URIs) to the subscriber (CBUS 
> >> client).
> >> 
> >> The CBUS client actions triggered by the received URI 
> list, and the  
> >> interactions between the CBUS server and other enablers, 
> are outside 
> >> the scope of the work.
> >> 
> >> 
> >> The work is planned to take place in, or in co-operaton with, the 
> >> SIPCORE WG.
> >> 
> >> 
> >> Goals and milestones: ---------------------
> >> 
> >> Jan 2010        Working Group Last Call for the document which
> >> contains the mechanism for a subscriber to provide the needed CBUS 
> >> information to the notifier, and contains the event 
> package (if a new 
> >> package is needed) for the notifier to provide the URI list to the 
> >> subscriber.
> >> 
> >> Mar 2010    Submit document to IESG as Proposed Standard
> 

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From Simo.Veikkolainen@nokia.com  Wed Sep 23 00:48:40 2009
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From: <Simo.Veikkolainen@nokia.com>
To: <pkyzivat@cisco.com>, <emcho@sip-communicator.org>
Date: Wed, 23 Sep 2009 09:49:26 +0200
Thread-Topic: [dispatch] Charter proposal on combining SIP voice with XMPP IM and	presence
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Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM and	presence
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Hi Paul and Emil,

>-----Original Message-----
>From: ext Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
>Sent: 22 September, 2009 16:24
>To: Emil Ivov
>Cc: Veikkolainen Simo (Nokia-D/Helsinki); dispatch@ietf.org
>Subject: Re: [dispatch] Charter proposal on combining SIP=20
>voice with XMPP IM and presence
>
>The issues Emil raises are good ones.
>I think they should be in scope for this effort.


I also think the issues Emil raises are very valid, and we would need to co=
nsider those.

>	Thanks,
>	Paul
>
>Emil Ivov wrote:
>> Hey Simo,
>>=20
>> We've been considering and playing a similar mode of operation for a=20
>> couple of years now and some of the most delicate issues that we've=20
>> come across include "disambiguation of overlapping services=20
>and information"
>> and "behaviour in cases of "half-failure". Here's what I mean:
>>=20
>> What happens for example if a mixed client receives a message over=20
>> SIMPLE rather than XMPP. Should it be accepted and displayed to the=20
>> user? If yes, then where should a client reply (i.e. SIMPLE=20
>or XMPP)?=20
>> If we choose SIMPLE for how long should the client privilege SIMPLE=20
>> over XMPP and under what conditions.
>>=20
>> What happens if a mixed client detects presence agent=20
>capabilities in=20
>> a SIP service? Should it ignore them and only go for XMPP, use them=20
>> for certain destinations only (e.g. destinations that we only have a=20
>> SIP address for such as conventional SIP phones), or duplicate all=20
>> information over the two protocols in which case: How should clients=20
>> handle conflicting presence information received via the two=20
>protocols?
>>=20
>> How should we handle failures for only one of the services.=20
>For example:
>>  I am a mixed UA and am unable to connect to a SIP registrar but can=20
>> still use XMPP. Should I declare complete failure to the=20
>user? Should=20
>> I declare success but warn the user that some features won't=20
>work? If=20
>> possible, should I try to make up for the missing SIP services using=20
>> XMPP (and vice versa).

Without addressing each issue separately, I would say we can provide normat=
ive statements for some situations, but most likely some issues listed abov=
e will have to be left for the implementation to deal with.

As has been pointed out in earlier emails, there is no automatic correlatio=
n of SIP and XMPP user id's and thus an AOR alice@example.com may not belon=
g to the same person as the JID alice@example.com. This rules out the possi=
bility of responding to a SIP MESSAGE with an XMPP <message> stanza without=
 some a priori knowledge on Alice's identity in XMPP domain.=20

>From an end user's point of view it makes no difference which protocol is u=
sed. Personally, I would not mind having all my messaging (IM's via differe=
nt service providers, SMS, email etc.) visible in a single application view=
, and let the application hide some of the complexity.=20

We have only covered some use cases in our document, but I agree we probabl=
y need to take a look at other situations as well. We just need to be caref=
ul in not exploding the scope of the work and thus delay the work.=20

Simo

>> So, if you think the above are worth considering how about adding=20
>> something along these lines:
>>=20
>> disambiguation: both XMPP and SIP define semantics that=20
>allow each of=20
>> the protocols to offer the complete set of services=20
>discussed in this=20
>> charter (i.e. voice, video, messaging, and presence). In=20
>environments=20
>> where one or more of these services are supported with both=20
>protocols=20
>> clients should be able to use a common procedure for=20
>determining which=20
>> one to use and under what conditions.
>>=20
>> half-failures: in cases where use of one of the protocols becomes=20
>> impossible (e.g. due to a server failure) clients should be able to=20
>> follow clear procedures on how to behave, which services could they=20
>> try to reuse over the protocol that is still available and=20
>under what=20
>> conditions.
>>=20
>> WDYT?
>>=20
>> Emil
>>=20
>>=20
>> Simo.Veikkolainen@nokia.com wrote:
>>> Hello,
>>>
>>> Below is the draft charter text for our proposal on=20
>combining SIP voice with XMPP IM and presence.
>>>
>>> Comments are very welcome!
>>>
>>> Regards,
>>> Simo
>>>
>>>
>>> -----------------
>>>
>>>
>>> Combining SIP VoIP and XMPP IM/Presence=20
>>> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>>>
>>> Currently most standards-based Voice over IP (VoIP)=20
>deployments use the Session Initiation Protocol (SIP).  In=20
>addition to providing basic voice service SIP has an extensive=20
>support for more advanced telephony features including=20
>interworking with the traditional Public Switched Telephone=20
>Network (PSTN).  SIP is also gaining popularity in the field=20
>of video communication. At the same time, the Extensible=20
>Messaging and Presence Protocol (XMPP) is enjoying widespread=20
>usage in instant messaging and presence services. =20
>>>
>>> Both SIP and XMPP offer extensions for voice as well as IM=20
>and presence (XMPP voice via the Jingle extension, and SIP=20
>IM/presence via SIMPLE protocols). However, widespread=20
>deployment of these extensions has not so far taken place. In=20
>order to speed up deployment of integrated VoIP and IM=20
>systems, SIP based voice and XMPP based IM/Presence could be=20
>combined in endpoints to offer a voice, IM and presence=20
>service without any changes to existing SIP and XMPP service=20
>infrastrucure.=20
>>>
>>> The objective of this Working Group is to develop solutions for=20
>>> combining SIP based voice with XMPP based IM and Presence=20
>such that a=20
>>> unified user experience can be offered to end user. Specifically,=20
>>> solutions are needed on
>>> - addressing; end users should be able to initiate sessions=20
>to a user identity in either SIP or XMPP domain. The=20
>corresponding user identity in the other protocol domain needs=20
>to be learned automatically.
>>> - session correlation; endpoints need to be able to correlate voice=20
>>> sessions with IM/Presence such that they can be presented=20
>to the end=20
>>> user in a seamless fashion
>>> - presence; it should be possible to change the XMPP=20
>presence status based on the user's activity in the SIP domain.
>>>
>>> The goal is to rely on existing service infrastructre, and=20
>new requirements should be imposed only to the endpoint.=20
>>>
>>> Protocol interworking, that is, translation from one=20
>protocol to another, is out of scope of this WG.
>>>
>>> Milestones
>>> Feb 2010 Problem statement and use cases submitted to IESG as=20
>>> Informational Dec 2010 Specification on combining SIP based=20
>voice and XMPP based IM/Presence in a seamless manner=20
>submitted to IESG as Proposed Standard.
>>>
>>> -----------------
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>
>>=20
>=

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Date: Wed, 23 Sep 2009 09:53:05 +0200
Thread-Topic: [dispatch] Charter proposal on combining SIP voice with XMPP IM	and	presence
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Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM	and	presence
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=20

>-----Original Message-----
>From: dispatch-bounces@ietf.org=20
>[mailto:dispatch-bounces@ietf.org] On Behalf Of ext Olle E. Johansson
>Sent: 22 September, 2009 19:44
>To: Peter Saint-Andre
>Cc: dispatch@ietf.org
>Subject: Re: [dispatch] Charter proposal on combining SIP=20
>voice with XMPP IM and presence
>
>
>22 sep 2009 kl. 18.40 skrev Peter Saint-Andre:
>
>> -----BEGIN PGP SIGNED MESSAGE-----
>> Hash: SHA1
>>
>> On 9/22/09 10:33 AM, Olle E. Johansson wrote:
>>>
>>> 22 sep 2009 kl. 18.06 skrev Peter Saint-Andre:
>>>
>>>> IMHO those activities will keep the XMPP WG chugging along for a=20
>>>> while and the SIP-XMPP work will not receive a lot of=20
>attention. If=20
>>>> we can move it over to a new, more focused WG then I am in favor,=20
>>>> because that will encourage faster progress, I think.
>>>
>>> Will a new working group mean a whole set of fresh new members with=20
>>> more time?
>>
>> Last I checked, fresh new members don't materialize out of thin air.=20
>> :)
>I am aware of that issue... ;-)
>
>>
>> My point is that *if* a new SIP-XMPP group is being chartered =20
>> anyway, it
>> would be more productive to have all the SIP-XMPP work happening =20
>> there,
>> because (1) most people involved in the XMPP WG are only=20
>interested in
>> XMPP and the SIP-XMPP work is not top-of-mind for them, and (2) we =20
>> might
>> be able to attract interest in SIP-XMPP work among implementers for =20
>> whom
>> SIP-XMPP interworking/combinations are a pain point (that would =20
>> include
>> client developers, server developers, and operators). If we reach =20
>> out to
>> those folks and define a problem statement that captures=20
>what they are
>> seeing in the implementation and deployment space, then yes=20
>we might =20
>> be
>> able to attract some fresh new members.
>
>Agreed. I am very interested in the SIP-XMPP work.
>I might just materialize at some point. Maybe not out of thin air =20
>though.

I wouldn't expect new resources to pop up to complete the draft-saintandre-=
* documents just by moving that work to another WG (if such was formed), bu=
t otherwise I think it makes sense to concentrate this bulk of work togethe=
r.

Simo


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

From emcho@sip-communicator.org  Wed Sep 23 05:45:25 2009
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Cc: dispatch@ietf.org
Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM and	presence
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Hey Simo,

Simo.Veikkolainen@nokia.com wrote:
> As has been pointed out in earlier emails, there is no automatic 
> correlation of SIP and XMPP user id's and thus an AOR 
> alice@example.com may not belong to the same person as the JID 
> alice@example.com. This rules out the possibility of responding to a 
> SIP MESSAGE with an XMPP <message> stanza without some a priori 
> knowledge on Alice's identity in XMPP domain.

I understand why we would not want to rely on an implicit correlation
like for example JID == AOR. However an explicit correlation such as an
XMPP contact header in SIP messages and a SIP URI in the XMPP vcards
would be very helpful. A XMPP+SIP user receiving a call from someone not
on their contact list is likely to wish to add the remote party to their
roster or send them instant messages. Not being able to do so would feel
awkward to the user.

> From an end user's point of view it makes no difference which
> protocol is used. Personally, I would not mind having all my
> messaging (IM's via different service providers, SMS, email etc.)
> visible in a single application view, and let the application hide
> some of the complexity.

Not sure I understand. Does this mean that you expect messages to be
traveling over both SIMPLE and XMPP? If yes, then we definitely need
some priority rules.

Cheers,
Emil


> 
> Simo
> 
>>> So, if you think the above are worth considering how about adding
>>>  something along these lines:
>>> 
>>> disambiguation: both XMPP and SIP define semantics that
>> allow each of
>>> the protocols to offer the complete set of services
>> discussed in this
>>> charter (i.e. voice, video, messaging, and presence). In
>> environments
>>> where one or more of these services are supported with both
>> protocols
>>> clients should be able to use a common procedure for
>> determining which
>>> one to use and under what conditions.
>>> 
>>> half-failures: in cases where use of one of the protocols becomes
>>>  impossible (e.g. due to a server failure) clients should be able
>>>  to follow clear procedures on how to behave, which services
>>> could they try to reuse over the protocol that is still available
>>> and
>> under what
>>> conditions.
>>> 
>>> WDYT?
>>> 
>>> Emil
>>> 
>>> 
>>> Simo.Veikkolainen@nokia.com wrote:
>>>> Hello,
>>>> 
>>>> Below is the draft charter text for our proposal on
>> combining SIP voice with XMPP IM and presence.
>>>> Comments are very welcome!
>>>> 
>>>> Regards, Simo
>>>> 
>>>> 
>>>> -----------------
>>>> 
>>>> 
>>>> Combining SIP VoIP and XMPP IM/Presence 
>>>> =======================================
>>>> 
>>>> Currently most standards-based Voice over IP (VoIP)
>> deployments use the Session Initiation Protocol (SIP).  In addition
>>  to providing basic voice service SIP has an extensive support for 
>> more advanced telephony features including interworking with the 
>> traditional Public Switched Telephone Network (PSTN).  SIP is also 
>> gaining popularity in the field of video communication. At the same
>>  time, the Extensible Messaging and Presence Protocol (XMPP) is 
>> enjoying widespread usage in instant messaging and presence 
>> services.
>>>> Both SIP and XMPP offer extensions for voice as well as IM
>> and presence (XMPP voice via the Jingle extension, and SIP 
>> IM/presence via SIMPLE protocols). However, widespread deployment 
>> of these extensions has not so far taken place. In order to speed 
>> up deployment of integrated VoIP and IM systems, SIP based voice 
>> and XMPP based IM/Presence could be combined in endpoints to offer 
>> a voice, IM and presence service without any changes to existing 
>> SIP and XMPP service infrastrucure.
>>>> The objective of this Working Group is to develop solutions for
>>>>  combining SIP based voice with XMPP based IM and Presence
>> such that a
>>>> unified user experience can be offered to end user. 
>>>> Specifically, solutions are needed on - addressing; end users 
>>>> should be able to initiate sessions
>> to a user identity in either SIP or XMPP domain. The corresponding 
>> user identity in the other protocol domain needs to be learned 
>> automatically.
>>>> - session correlation; endpoints need to be able to correlate 
>>>> voice sessions with IM/Presence such that they can be presented
>>>> 
>>>> 
>> to the end
>>>> user in a seamless fashion - presence; it should be possible to
>>>>  change the XMPP
>> presence status based on the user's activity in the SIP domain.
>>>> The goal is to rely on existing service infrastructre, and
>> new requirements should be imposed only to the endpoint.
>>>> Protocol interworking, that is, translation from one
>> protocol to another, is out of scope of this WG.
>>>> Milestones Feb 2010 Problem statement and use cases submitted 
>>>> to IESG as Informational Dec 2010 Specification on combining 
>>>> SIP based
>> voice and XMPP based IM/Presence in a seamless manner submitted to 
>> IESG as Proposed Standard.
>>>> ----------------- 
>>>> _______________________________________________ dispatch 
>>>> mailing list dispatch@ietf.org 
>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>> 

-- 
Emil Ivov, Ph.D.                               67000 Strasbourg,
Project Lead                                   France
SIP Communicator
emcho@sip-communicator.org                     PHONE: +33.1.77.62.43.30
http://sip-communicator.org                    FAX:   +33.1.77.62.47.31

From vkg@alcatel-lucent.com  Wed Sep 23 06:51:18 2009
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Subject: Re: [dispatch] Charter proposal: Session Recording Protocol
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Lorenzo Miniero wrote:
> Hi Vijay,
> 
> as a group we find this subject very interesting, and so we definitely
> support the charter proposal. We have a related draft on session
> recording (even though it's quite focused on conferencing at the
> moment)
> 
> http://tools.ietf.org/html/draft-romano-dcon-recording-00
> 
> so we'd be glad to contribute to the work to be done in here.

Lorenzo: Thank you for your note of support.

I, as well as others on the SRP author list, did indeed look at
your draft over the summer; IIRC, a good part of it is
synchronizing other multimedia objects to the recorded source
as well.

We will be releasing an update to the requirements draft shortly,
and it would be great to have you contribute to the discussion
therein.

Once again, thanks.

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/

From salvatore.loreto@ericsson.com  Wed Sep 23 07:51:15 2009
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Cc: IETF Dispatch <dispatch@ietf.org>, Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>
Subject: Re: [dispatch] Charter Proposal for Disaggregated Media
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Hi Paul,

thanks for the comments, please see my answers in line

cheers
Sal


Paul Kyzivat wrote:
> I agree with the utility of such work. Comments inline.
>
<snip>
> I don't understand the above. The disaggregated multimedia session 
> will not be established unless *something* takes the initiative to 
> establish it. IMO that thing, whatever it is, is acting as a 
> "controller".
>
> The mechanism you have proposed before distributes the "controller" 
> functionality across multiple nodes. There is something that causes a 
> new stream to be established as a separate sip session and causes that 
> to announce correlation with some other session. And then there is are 
> the devices at the far end, that maintain the correlation and do 
> required maintenance of the correlation as changes happen.
>
> Can you provide more definition of what you mean by "loosely coupled" 
> while recognizing that something needs to know enough to do the 
> initiation and maintenance of the correlation?
>
the "loosely couple control" is a scenario where you can easily remove, 
at any point, any of several devices involved in the conversation, and 
the other end of the conversation will remain unaffected.

easily and at any point mean that you do not need to start a complicate 
mechanism before you can remove the device (e.g. you do not need to 
choose a new controller and transfer the control to it)

of course loosely couple control still needs some sort of control, you 
are right.
However the control could be easily a "distributed" control with a 
minimal amount of information, where, for example, each device involved 
in the conversation only knows the list of all the other devices 
involved in the conversation, but it does not need to know the full 
state "information" related to the dialog that each device has with the 
far end of the conversation.
>> The objective of the proposed working group is to develop a framework
>> for Disaggregated Media that include both improvements on existing 
>> mechanisms (e.g. 3pcc) and the develop of one or more new mechanisms
>> like a loosely coupled mechanism that does not require the 
>> implementation of complex logic on any of the terminals.
>
> I would argue that it is premature to assert that new mechanisms are 
> required.
right, that is something the community has to decide.
>> Specifically, the proposed working group will develop the following
>> deliverables: 1. A framework document describing key design 
>> considerations for Disaggregated   mediamechanism.
>> 2. A specification for a loosely couple mechanism.
>> 3. One or more specifications to improve existing mechanism to 
>> coordinate
>>   disaggregated media.
>
> Are both (2) and (3) *specifications*? Or is (2) a requirements 
> document that is then satisfied by (3)?
yes (1) and (2) can be combined and then eventually produce a new actual 
mechanism.
>
> If (2) is requirements, then this makes sense to me, though it might 
> be possible to combine that with (1). If both (2) and (3) are 
> mechanism specs then I don't get it.
>
>     Thanks,
>     Paul


From RIFATYU@nortel.com  Wed Sep 23 12:55:59 2009
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From: "Rifaat Shekh-Yusef" <rifatyu@nortel.com>
To: <Simo.Veikkolainen@nokia.com>, <dispatch@ietf.org>, "Mary Barnes" <mary.barnes@nortel.com>, <gonzalo.camarillo@ericsson.com>
Cc: Markus.Isomaki@nokia.com
Subject: Re: [dispatch] New topic proposal for DISPATCH
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Simo, Markus,
=20
I am very interested in this work and I like your approach.
=20
I have a question regarding the open issue you have in section 5.2:
Why do you require the correlation-value to be an end-to-end identifier?
In the case where there is an SBC that creates a new dialog for the
target, would it not be enough for each endpoint to use the Call-ID of
the other dialog as the correlation-value?
=20
Thanks,
 Rifaat
=20

________________________________

From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Simo.Veikkolainen@nokia.com
Sent: Friday, September 04, 2009 8:09 AM
To: dispatch@ietf.org; Barnes, Mary (RICH2:AR00);
gonzalo.camarillo@ericsson.com
Cc: Markus.Isomaki@nokia.com
Subject: [dispatch] New topic proposal for DISPATCH


Hi,
=20
We would like to propose a new DISPATCH topic on how SIP and XMPP can be
used together in a complementary fashion.
=20
We have been working on a solution which combines SIP based VoIP and
XMPP based IM and Presence. The requirements and a proposed solution is
outlined in
http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01
<http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01> .
=20
The motivation for this work comes from experience which shows that most
standards based VoIP deployments use SIP. At the same time, the
Extensible Messaging and Presence Protocol (XMPP) is widely used in
instant messaging and presence services. An interesting scenario arises
when a SIP based voice (and video) service is combined together with an
XMPP based instant messaging and presence service.=20
=20
Note that the goal in this work is not SIP-XMPP protocol translation,
but to define protocol extensions and best practises such that SIP based
VoIP and XMPP based IM and presence could be seamlessly combined and
offered to the end user. For rapid deployment, we assume no changes in
the server infrastructure, and instead impose new requirements and
protocol extensions only to the endpoints.
=20
We would like to get some discussion going on the use case itself as
well as on the solution. Also, we would like to hear you thoughts on
DISPATCH or a new "dispatched" mini-WG as the home for such work.
=20
Exact charter proposal and problem statemen is to follow.=20
=20
Regards,
Simo and Markus
=20
=20

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<META content=3D"text/html; charset=3Dus-ascii" =
http-equiv=3DContent-Type>
<META name=3DGENERATOR content=3D"MSHTML 8.00.6001.18812"><!-- converted =
from rtf -->
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</HEAD>
<BODY>
<DIV><SPAN class=3D443113913-23092009><FONT color=3D#0000ff size=3D2 =
face=3DArial>Simo,=20
Markus,</FONT></SPAN></DIV>
<DIV><SPAN class=3D443113913-23092009><FONT color=3D#0000ff size=3D2=20
face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D443113913-23092009><FONT color=3D#0000ff size=3D2 =
face=3DArial>I am=20
very interested in this work and I like your =
approach.</FONT></SPAN></DIV>
<DIV><SPAN class=3D443113913-23092009><FONT color=3D#0000ff size=3D2=20
face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D443113913-23092009><FONT color=3D#0000ff size=3D2 =
face=3DArial>I have=20
a question regarding the open issue you have in section =
5.2:</FONT></SPAN></DIV>
<DIV><SPAN class=3D443113913-23092009><FONT color=3D#0000ff size=3D2 =
face=3DArial>Why do=20
you require the correlation-value to be an end-to-end=20
identifier?</FONT></SPAN></DIV>
<DIV><SPAN class=3D443113913-23092009><FONT color=3D#0000ff size=3D2 =
face=3DArial>In the=20
case&nbsp;where there is an SBC that creates a new dialog for the =
target,=20
w</FONT></SPAN><SPAN class=3D443113913-23092009><FONT color=3D#0000ff =
size=3D2=20
face=3DArial>ould it not be enough for each endpoint to use the Call-ID =
of the=20
other dialog as the correlation-value?</FONT></SPAN></DIV>
<DIV><SPAN class=3D443113913-23092009><FONT color=3D#0000ff size=3D2=20
face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D443113913-23092009><FONT color=3D#0000ff size=3D2=20
face=3DArial>Thanks,</FONT></SPAN></DIV>
<DIV><SPAN class=3D443113913-23092009><FONT color=3D#0000ff size=3D2=20
face=3DArial>&nbsp;Rifaat</FONT></SPAN></DIV>
<DIV><SPAN class=3D443113913-23092009><FONT color=3D#0000ff size=3D2=20
face=3DArial></FONT></SPAN>&nbsp;</DIV><BR>
<DIV dir=3Dltr lang=3Den-us class=3DOutlookMessageHeader align=3Dleft>
<HR tabIndex=3D-1>
<FONT size=3D2 face=3DTahoma><B>From:</B> dispatch-bounces@ietf.org=20
[mailto:dispatch-bounces@ietf.org] <B>On Behalf Of=20
</B>Simo.Veikkolainen@nokia.com<BR><B>Sent:</B> Friday, September 04, =
2009 8:09=20
AM<BR><B>To:</B> dispatch@ietf.org; Barnes, Mary (RICH2:AR00);=20
gonzalo.camarillo@ericsson.com<BR><B>Cc:</B>=20
Markus.Isomaki@nokia.com<BR><B>Subject:</B> [dispatch] New topic =
proposal for=20
DISPATCH<BR></FONT><BR></DIV>
<DIV></DIV><FONT size=3D2 face=3D"Arial, sans-serif">
<DIV>Hi,</DIV>
<DIV>&nbsp;</DIV>
<DIV>We would like to propose a new DISPATCH topic on how SIP and XMPP =
can be=20
used together in a complementary fashion.</DIV>
<DIV>&nbsp;</DIV>
<DIV>We have been working on a solution which combines SIP based VoIP =
and XMPP=20
based IM and Presence. The requirements and a proposed solution is =
outlined in=20
<A=20
href=3D"http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01=
"><FONT=20
color=3D#0000ff><U>http://tools.ietf.org/html/draft-veikkolainen-sip-voip=
-xmpp-im-01</U></FONT></A>.</DIV>
<DIV>&nbsp;</DIV>
<DIV>The motivation for this work comes from experience which shows that =
most=20
standards based VoIP deployments use SIP. At the same time, the =
Extensible=20
Messaging and Presence Protocol (XMPP) is widely used in instant =
messaging and=20
presence services. An interesting scenario arises when a SIP based voice =
(and=20
video) service is combined together with an XMPP based instant messaging =
and=20
presence service. </DIV>
<DIV>&nbsp;</DIV>
<DIV>Note that the goal in this work is not SIP-XMPP protocol =
translation, but=20
to define protocol extensions and best practises such that SIP based =
VoIP and=20
XMPP based IM and presence could be seamlessly combined and offered to =
the end=20
user. For rapid deployment, we assume no changes in the server =
infrastructure,=20
and instead impose new requirements and protocol extensions only to the=20
endpoints.</DIV>
<DIV>&nbsp;</DIV>
<DIV>We would like to get some discussion going on the use case itself =
as well=20
as on the solution. Also, we would like to hear you thoughts on DISPATCH =
or a=20
new "dispatched" mini-WG as the home for such work.</DIV>
<DIV>&nbsp;</DIV>
<DIV>Exact charter proposal and problem statemen is to follow. </DIV>
<DIV>&nbsp;</DIV>
<DIV>Regards,</DIV>
<DIV>Simo and Markus</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV></FONT></BODY></HTML>

------_=_NextPart_001_01CA3C88.01864ECA--

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Cc: dispatch@ietf.org
Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM and	presence
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Simo,

It would be a good start to *identify* these issues in whatever is 
produced, and discuss their resolution - even if all that is said is 
that a particular issue is ruled out of scope or unsolved in an initial 
release. I'd just like to have them *considered* rather than not even 
being mentioned.

	Thanks,
	Paul

Simo.Veikkolainen@nokia.com wrote:
> Hi Paul and Emil,
> 
>> -----Original Message-----
>> From: ext Paul Kyzivat [mailto:pkyzivat@cisco.com] 
>> Sent: 22 September, 2009 16:24
>> To: Emil Ivov
>> Cc: Veikkolainen Simo (Nokia-D/Helsinki); dispatch@ietf.org
>> Subject: Re: [dispatch] Charter proposal on combining SIP 
>> voice with XMPP IM and presence
>>
>> The issues Emil raises are good ones.
>> I think they should be in scope for this effort.
> 
> 
> I also think the issues Emil raises are very valid, and we would need to consider those.
> 
>> 	Thanks,
>> 	Paul
>>
>> Emil Ivov wrote:
>>> Hey Simo,
>>>
>>> We've been considering and playing a similar mode of operation for a 
>>> couple of years now and some of the most delicate issues that we've 
>>> come across include "disambiguation of overlapping services 
>> and information"
>>> and "behaviour in cases of "half-failure". Here's what I mean:
>>>
>>> What happens for example if a mixed client receives a message over 
>>> SIMPLE rather than XMPP. Should it be accepted and displayed to the 
>>> user? If yes, then where should a client reply (i.e. SIMPLE 
>> or XMPP)? 
>>> If we choose SIMPLE for how long should the client privilege SIMPLE 
>>> over XMPP and under what conditions.
>>>
>>> What happens if a mixed client detects presence agent 
>> capabilities in 
>>> a SIP service? Should it ignore them and only go for XMPP, use them 
>>> for certain destinations only (e.g. destinations that we only have a 
>>> SIP address for such as conventional SIP phones), or duplicate all 
>>> information over the two protocols in which case: How should clients 
>>> handle conflicting presence information received via the two 
>> protocols?
>>> How should we handle failures for only one of the services. 
>> For example:
>>>  I am a mixed UA and am unable to connect to a SIP registrar but can 
>>> still use XMPP. Should I declare complete failure to the 
>> user? Should 
>>> I declare success but warn the user that some features won't 
>> work? If 
>>> possible, should I try to make up for the missing SIP services using 
>>> XMPP (and vice versa).
> 
> Without addressing each issue separately, I would say we can provide normative statements for some situations, but most likely some issues listed above will have to be left for the implementation to deal with.
> 
> As has been pointed out in earlier emails, there is no automatic correlation of SIP and XMPP user id's and thus an AOR alice@example.com may not belong to the same person as the JID alice@example.com. This rules out the possibility of responding to a SIP MESSAGE with an XMPP <message> stanza without some a priori knowledge on Alice's identity in XMPP domain. 
> 
>>From an end user's point of view it makes no difference which protocol is used. Personally, I would not mind having all my messaging (IM's via different service providers, SMS, email etc.) visible in a single application view, and let the application hide some of the complexity. 
> 
> We have only covered some use cases in our document, but I agree we probably need to take a look at other situations as well. We just need to be careful in not exploding the scope of the work and thus delay the work. 
> 
> Simo
> 
>>> So, if you think the above are worth considering how about adding 
>>> something along these lines:
>>>
>>> disambiguation: both XMPP and SIP define semantics that 
>> allow each of 
>>> the protocols to offer the complete set of services 
>> discussed in this 
>>> charter (i.e. voice, video, messaging, and presence). In 
>> environments 
>>> where one or more of these services are supported with both 
>> protocols 
>>> clients should be able to use a common procedure for 
>> determining which 
>>> one to use and under what conditions.
>>>
>>> half-failures: in cases where use of one of the protocols becomes 
>>> impossible (e.g. due to a server failure) clients should be able to 
>>> follow clear procedures on how to behave, which services could they 
>>> try to reuse over the protocol that is still available and 
>> under what 
>>> conditions.
>>>
>>> WDYT?
>>>
>>> Emil
>>>
>>>
>>> Simo.Veikkolainen@nokia.com wrote:
>>>> Hello,
>>>>
>>>> Below is the draft charter text for our proposal on 
>> combining SIP voice with XMPP IM and presence.
>>>> Comments are very welcome!
>>>>
>>>> Regards,
>>>> Simo
>>>>
>>>>
>>>> -----------------
>>>>
>>>>
>>>> Combining SIP VoIP and XMPP IM/Presence 
>>>> =======================================
>>>>
>>>> Currently most standards-based Voice over IP (VoIP) 
>> deployments use the Session Initiation Protocol (SIP).  In 
>> addition to providing basic voice service SIP has an extensive 
>> support for more advanced telephony features including 
>> interworking with the traditional Public Switched Telephone 
>> Network (PSTN).  SIP is also gaining popularity in the field 
>> of video communication. At the same time, the Extensible 
>> Messaging and Presence Protocol (XMPP) is enjoying widespread 
>> usage in instant messaging and presence services.  
>>>> Both SIP and XMPP offer extensions for voice as well as IM 
>> and presence (XMPP voice via the Jingle extension, and SIP 
>> IM/presence via SIMPLE protocols). However, widespread 
>> deployment of these extensions has not so far taken place. In 
>> order to speed up deployment of integrated VoIP and IM 
>> systems, SIP based voice and XMPP based IM/Presence could be 
>> combined in endpoints to offer a voice, IM and presence 
>> service without any changes to existing SIP and XMPP service 
>> infrastrucure. 
>>>> The objective of this Working Group is to develop solutions for 
>>>> combining SIP based voice with XMPP based IM and Presence 
>> such that a 
>>>> unified user experience can be offered to end user. Specifically, 
>>>> solutions are needed on
>>>> - addressing; end users should be able to initiate sessions 
>> to a user identity in either SIP or XMPP domain. The 
>> corresponding user identity in the other protocol domain needs 
>> to be learned automatically.
>>>> - session correlation; endpoints need to be able to correlate voice 
>>>> sessions with IM/Presence such that they can be presented 
>> to the end 
>>>> user in a seamless fashion
>>>> - presence; it should be possible to change the XMPP 
>> presence status based on the user's activity in the SIP domain.
>>>> The goal is to rely on existing service infrastructre, and 
>> new requirements should be imposed only to the endpoint. 
>>>> Protocol interworking, that is, translation from one 
>> protocol to another, is out of scope of this WG.
>>>> Milestones
>>>> Feb 2010 Problem statement and use cases submitted to IESG as 
>>>> Informational Dec 2010 Specification on combining SIP based 
>> voice and XMPP based IM/Presence in a seamless manner 
>> submitted to IESG as Proposed Standard.
>>>> -----------------
>>>> _______________________________________________
>>>> dispatch mailing list
>>>> dispatch@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>

From Simo.Veikkolainen@nokia.com  Wed Sep 23 23:35:08 2009
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From: <Simo.Veikkolainen@nokia.com>
To: <emcho@sip-communicator.org>
Date: Thu, 24 Sep 2009 08:34:17 +0200
Thread-Topic: [dispatch] Charter proposal on combining SIP voice with XMPP IM and	presence
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References: <B23B311878A0B4438F5F09F47E01AAE03B18566B21@NOK-EUMSG-01.mgdnok.nokia.com> <4AB8BC67.3080702@sip-communicator.org> <4AB8CFEC.8020703@cisco.com> <B23B311878A0B4438F5F09F47E01AAE03B1886AC5E@NOK-EUMSG-01.mgdnok.nokia.com> <4ABA18A0.5070603@sip-communicator.org>
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Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM and	presence
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=20

>-----Original Message-----
>From: Emil Ivov [mailto:emil@sip-communicator.org] On Behalf=20
>Of ext Emil Ivov
>Sent: 23 September, 2009 15:46
>To: Veikkolainen Simo (Nokia-D/Helsinki)
>Cc: pkyzivat@cisco.com; dispatch@ietf.org
>Subject: Re: [dispatch] Charter proposal on combining SIP=20
>voice with XMPP IM and presence
>
>Hey Simo,
>
>Simo.Veikkolainen@nokia.com wrote:
>> As has been pointed out in earlier emails, there is no automatic=20
>> correlation of SIP and XMPP user id's and thus an AOR=20
>> alice@example.com may not belong to the same person as the JID=20
>> alice@example.com. This rules out the possibility of responding to a=20
>> SIP MESSAGE with an XMPP <message> stanza without some a priori=20
>> knowledge on Alice's identity in XMPP domain.
>
>I understand why we would not want to rely on an implicit=20
>correlation like for example JID =3D=3D AOR. However an explicit=20
>correlation such as an XMPP contact header in SIP messages and=20
>a SIP URI in the XMPP vcards would be very helpful. A XMPP+SIP=20
>user receiving a call from someone not on their contact list=20
>is likely to wish to add the remote party to their roster or=20
>send them instant messages. Not being able to do so would feel=20
>awkward to the user.

Yes. Explicit correlation (carrying SIP identities in XMPP or vice versa) i=
s what we proposed in our draft, and we think could indeed be useful.

>
>> From an end user's point of view it makes no difference=20
>which protocol=20
>> is used. Personally, I would not mind having all my messaging (IM's=20
>> via different service providers, SMS, email etc.) visible in=20
>a single=20
>> application view, and let the application hide some of the=20
>complexity.
>
>Not sure I understand. Does this mean that you expect messages=20
>to be traveling over both SIMPLE and XMPP? If yes, then we=20
>definitely need some priority rules.

What I meant was that from a UI point of view, all the end user needs to se=
e is a contact name and possibility send a message (or call). A lot of the =
complexity of selecting the actual protocol *could* be hidden by the applic=
ation. This may not be what all application developers wish, but for exampl=
e the phonebooks in some mobile devices seem to be going to that direction,=
 combining IM and SMS in one "conversation" view.

So, for example, if a user wants the send an initial message to a contact w=
hose both SIP and XMPP IDs are known, the app could do the selection of pro=
tocol, depending on the other user's presence information. However, when re=
sponding to a message in a conversation, it probably is better to use the s=
ame protocol that was used when sending, unless the client knows that the o=
ther client's UI can render the messages in the same conversation view.=20

So, I'll I'm saying is we need to carefully consider making normative state=
ments that could potentially limit the application design and usability.=20

Simo

>Cheers,
>Emil
>
>
>>=20
>> Simo
>>=20
>>>> So, if you think the above are worth considering how about adding =20
>>>> something along these lines:
>>>>=20
>>>> disambiguation: both XMPP and SIP define semantics that
>>> allow each of
>>>> the protocols to offer the complete set of services
>>> discussed in this
>>>> charter (i.e. voice, video, messaging, and presence). In
>>> environments
>>>> where one or more of these services are supported with both
>>> protocols
>>>> clients should be able to use a common procedure for
>>> determining which
>>>> one to use and under what conditions.
>>>>=20
>>>> half-failures: in cases where use of one of the protocols becomes =20
>>>> impossible (e.g. due to a server failure) clients should=20
>be able  to=20
>>>> follow clear procedures on how to behave, which services=20
>could they=20
>>>> try to reuse over the protocol that is still available and
>>> under what
>>>> conditions.
>>>>=20
>>>> WDYT?
>>>>=20
>>>> Emil
>>>>=20
>>>>=20
>>>> Simo.Veikkolainen@nokia.com wrote:
>>>>> Hello,
>>>>>=20
>>>>> Below is the draft charter text for our proposal on
>>> combining SIP voice with XMPP IM and presence.
>>>>> Comments are very welcome!
>>>>>=20
>>>>> Regards, Simo
>>>>>=20
>>>>>=20
>>>>> -----------------
>>>>>=20
>>>>>=20
>>>>> Combining SIP VoIP and XMPP IM/Presence=20
>>>>> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>>>>>=20
>>>>> Currently most standards-based Voice over IP (VoIP)
>>> deployments use the Session Initiation Protocol (SIP).  In=20
>addition =20
>>> to providing basic voice service SIP has an extensive support for=20
>>> more advanced telephony features including interworking with the=20
>>> traditional Public Switched Telephone Network (PSTN).  SIP is also=20
>>> gaining popularity in the field of video communication. At=20
>the same =20
>>> time, the Extensible Messaging and Presence Protocol (XMPP) is=20
>>> enjoying widespread usage in instant messaging and presence=20
>services.
>>>>> Both SIP and XMPP offer extensions for voice as well as IM
>>> and presence (XMPP voice via the Jingle extension, and SIP=20
>>> IM/presence via SIMPLE protocols). However, widespread=20
>deployment of=20
>>> these extensions has not so far taken place. In order to speed up=20
>>> deployment of integrated VoIP and IM systems, SIP based voice and=20
>>> XMPP based IM/Presence could be combined in endpoints to offer a=20
>>> voice, IM and presence service without any changes to existing SIP=20
>>> and XMPP service infrastrucure.
>>>>> The objective of this Working Group is to develop solutions for =20
>>>>> combining SIP based voice with XMPP based IM and Presence
>>> such that a
>>>>> unified user experience can be offered to end user.=20
>>>>> Specifically, solutions are needed on - addressing; end users=20
>>>>> should be able to initiate sessions
>>> to a user identity in either SIP or XMPP domain. The corresponding=20
>>> user identity in the other protocol domain needs to be learned=20
>>> automatically.
>>>>> - session correlation; endpoints need to be able to=20
>correlate voice=20
>>>>> sessions with IM/Presence such that they can be presented
>>>>>=20
>>>>>=20
>>> to the end
>>>>> user in a seamless fashion - presence; it should be possible to =20
>>>>> change the XMPP
>>> presence status based on the user's activity in the SIP domain.
>>>>> The goal is to rely on existing service infrastructre, and
>>> new requirements should be imposed only to the endpoint.
>>>>> Protocol interworking, that is, translation from one
>>> protocol to another, is out of scope of this WG.
>>>>> Milestones Feb 2010 Problem statement and use cases submitted to=20
>>>>> IESG as Informational Dec 2010 Specification on combining=20
>SIP based
>>> voice and XMPP based IM/Presence in a seamless manner submitted to=20
>>> IESG as Proposed Standard.
>>>>> -----------------
>>>>> _______________________________________________ dispatch mailing=20
>>>>> list dispatch@ietf.org=20
>>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>>=20
>
>--=20
>Emil Ivov, Ph.D.                               67000 Strasbourg,
>Project Lead                                   France
>SIP Communicator
>emcho@sip-communicator.org                     PHONE: +33.1.77.62.43.30
>http://sip-communicator.org                    FAX:   +33.1.77.62.47.31
>=

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From: <Simo.Veikkolainen@nokia.com>
To: <pkyzivat@cisco.com>
Date: Thu, 24 Sep 2009 08:35:39 +0200
Thread-Topic: [dispatch] Charter proposal on combining SIP voice with XMPP IM and	presence
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References: <B23B311878A0B4438F5F09F47E01AAE03B18566B21@NOK-EUMSG-01.mgdnok.nokia.com> <4AB8BC67.3080702@sip-communicator.org> <4AB8CFEC.8020703@cisco.com> <B23B311878A0B4438F5F09F47E01AAE03B1886AC5E@NOK-EUMSG-01.mgdnok.nokia.com> <4ABA9109.8010702@cisco.com>
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Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM and	presence
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=20

>-----Original Message-----
>From: ext Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
>Sent: 24 September, 2009 00:20
>To: Veikkolainen Simo (Nokia-D/Helsinki)
>Cc: emcho@sip-communicator.org; dispatch@ietf.org
>Subject: Re: [dispatch] Charter proposal on combining SIP=20
>voice with XMPP IM and presence
>
>Simo,
>
>It would be a good start to *identify* these issues in=20
>whatever is produced, and discuss their resolution - even if=20
>all that is said is that a particular issue is ruled out of=20
>scope or unsolved in an initial release. I'd just like to have=20
>them *considered* rather than not even being mentioned.

Agreed. I also think we cannot just bypass these; they are anyways the prac=
tical problems implementors are going to face.

Simo

>	Thanks,
>	Paul
>
>Simo.Veikkolainen@nokia.com wrote:
>> Hi Paul and Emil,
>>=20
>>> -----Original Message-----
>>> From: ext Paul Kyzivat [mailto:pkyzivat@cisco.com]
>>> Sent: 22 September, 2009 16:24
>>> To: Emil Ivov
>>> Cc: Veikkolainen Simo (Nokia-D/Helsinki); dispatch@ietf.org
>>> Subject: Re: [dispatch] Charter proposal on combining SIP=20
>voice with=20
>>> XMPP IM and presence
>>>
>>> The issues Emil raises are good ones.
>>> I think they should be in scope for this effort.
>>=20
>>=20
>> I also think the issues Emil raises are very valid, and we=20
>would need to consider those.
>>=20
>>> 	Thanks,
>>> 	Paul
>>>
>>> Emil Ivov wrote:
>>>> Hey Simo,
>>>>
>>>> We've been considering and playing a similar mode of=20
>operation for a=20
>>>> couple of years now and some of the most delicate issues=20
>that we've=20
>>>> come across include "disambiguation of overlapping services
>>> and information"
>>>> and "behaviour in cases of "half-failure". Here's what I mean:
>>>>
>>>> What happens for example if a mixed client receives a message over=20
>>>> SIMPLE rather than XMPP. Should it be accepted and=20
>displayed to the=20
>>>> user? If yes, then where should a client reply (i.e. SIMPLE
>>> or XMPP)?=20
>>>> If we choose SIMPLE for how long should the client=20
>privilege SIMPLE=20
>>>> over XMPP and under what conditions.
>>>>
>>>> What happens if a mixed client detects presence agent
>>> capabilities in
>>>> a SIP service? Should it ignore them and only go for XMPP,=20
>use them=20
>>>> for certain destinations only (e.g. destinations that we=20
>only have a=20
>>>> SIP address for such as conventional SIP phones), or duplicate all=20
>>>> information over the two protocols in which case: How=20
>should clients=20
>>>> handle conflicting presence information received via the two
>>> protocols?
>>>> How should we handle failures for only one of the services.=20
>>> For example:
>>>>  I am a mixed UA and am unable to connect to a SIP=20
>registrar but can=20
>>>> still use XMPP. Should I declare complete failure to the
>>> user? Should
>>>> I declare success but warn the user that some features won't
>>> work? If
>>>> possible, should I try to make up for the missing SIP=20
>services using=20
>>>> XMPP (and vice versa).
>>=20
>> Without addressing each issue separately, I would say we can=20
>provide normative statements for some situations, but most=20
>likely some issues listed above will have to be left for the=20
>implementation to deal with.
>>=20
>> As has been pointed out in earlier emails, there is no=20
>automatic correlation of SIP and XMPP user id's and thus an=20
>AOR alice@example.com may not belong to the same person as the=20
>JID alice@example.com. This rules out the possibility of=20
>responding to a SIP MESSAGE with an XMPP <message> stanza=20
>without some a priori knowledge on Alice's identity in XMPP domain.=20
>>=20
>>>From an end user's point of view it makes no difference=20
>which protocol is used. Personally, I would not mind having=20
>all my messaging (IM's via different service providers, SMS,=20
>email etc.) visible in a single application view, and let the=20
>application hide some of the complexity.=20
>>=20
>> We have only covered some use cases in our document, but I=20
>agree we probably need to take a look at other situations as=20
>well. We just need to be careful in not exploding the scope of=20
>the work and thus delay the work.=20
>>=20
>> Simo
>>=20
>>>> So, if you think the above are worth considering how about adding=20
>>>> something along these lines:
>>>>
>>>> disambiguation: both XMPP and SIP define semantics that
>>> allow each of
>>>> the protocols to offer the complete set of services
>>> discussed in this
>>>> charter (i.e. voice, video, messaging, and presence). In
>>> environments
>>>> where one or more of these services are supported with both
>>> protocols
>>>> clients should be able to use a common procedure for
>>> determining which
>>>> one to use and under what conditions.
>>>>
>>>> half-failures: in cases where use of one of the protocols becomes=20
>>>> impossible (e.g. due to a server failure) clients should=20
>be able to=20
>>>> follow clear procedures on how to behave, which services=20
>could they=20
>>>> try to reuse over the protocol that is still available and
>>> under what
>>>> conditions.
>>>>
>>>> WDYT?
>>>>
>>>> Emil
>>>>
>>>>
>>>> Simo.Veikkolainen@nokia.com wrote:
>>>>> Hello,
>>>>>
>>>>> Below is the draft charter text for our proposal on
>>> combining SIP voice with XMPP IM and presence.
>>>>> Comments are very welcome!
>>>>>
>>>>> Regards,
>>>>> Simo
>>>>>
>>>>>
>>>>> -----------------
>>>>>
>>>>>
>>>>> Combining SIP VoIP and XMPP IM/Presence=20
>>>>> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>>>>>
>>>>> Currently most standards-based Voice over IP (VoIP)
>>> deployments use the Session Initiation Protocol (SIP).  In addition=20
>>> to providing basic voice service SIP has an extensive support for=20
>>> more advanced telephony features including interworking with the=20
>>> traditional Public Switched Telephone Network (PSTN).  SIP is also=20
>>> gaining popularity in the field of video communication. At the same=20
>>> time, the Extensible Messaging and Presence Protocol (XMPP) is=20
>>> enjoying widespread usage in instant messaging and presence=20
>services.
>>>>> Both SIP and XMPP offer extensions for voice as well as IM
>>> and presence (XMPP voice via the Jingle extension, and SIP=20
>>> IM/presence via SIMPLE protocols). However, widespread=20
>deployment of=20
>>> these extensions has not so far taken place. In order to speed up=20
>>> deployment of integrated VoIP and IM systems, SIP based voice and=20
>>> XMPP based IM/Presence could be combined in endpoints to offer a=20
>>> voice, IM and presence service without any changes to existing SIP=20
>>> and XMPP service infrastrucure.
>>>>> The objective of this Working Group is to develop solutions for=20
>>>>> combining SIP based voice with XMPP based IM and Presence
>>> such that a
>>>>> unified user experience can be offered to end user. Specifically,=20
>>>>> solutions are needed on
>>>>> - addressing; end users should be able to initiate sessions
>>> to a user identity in either SIP or XMPP domain. The corresponding=20
>>> user identity in the other protocol domain needs to be learned=20
>>> automatically.
>>>>> - session correlation; endpoints need to be able to=20
>correlate voice=20
>>>>> sessions with IM/Presence such that they can be presented
>>> to the end
>>>>> user in a seamless fashion
>>>>> - presence; it should be possible to change the XMPP
>>> presence status based on the user's activity in the SIP domain.
>>>>> The goal is to rely on existing service infrastructre, and
>>> new requirements should be imposed only to the endpoint.=20
>>>>> Protocol interworking, that is, translation from one
>>> protocol to another, is out of scope of this WG.
>>>>> Milestones
>>>>> Feb 2010 Problem statement and use cases submitted to IESG as=20
>>>>> Informational Dec 2010 Specification on combining SIP based
>>> voice and XMPP based IM/Presence in a seamless manner submitted to=20
>>> IESG as Proposed Standard.
>>>>> -----------------
>>>>> _______________________________________________
>>>>> dispatch mailing list
>>>>> dispatch@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>>
>=

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From: <Markus.Isomaki@nokia.com>
To: <rifatyu@nortel.com>, <Simo.Veikkolainen@nokia.com>, <dispatch@ietf.org>,  <mary.barnes@nortel.com>, <gonzalo.camarillo@ericsson.com>
Date: Thu, 24 Sep 2009 08:54:03 +0200
Thread-Topic: [dispatch] New topic proposal for DISPATCH - SIP/XMPP
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Hi Rifaat,

In our proposal the correlation works so that:
1. UAs engaged in a SIP session/dialog exchange an identifier related to th=
at session (as well as their XMPP full JIDs)
2. They insert them in the very first XMPP messages they subsequently excha=
nge, in order to correlate that exchange to be related to the SIP session.

The correlation identifier has to be end-to-end, otherwise the other end wo=
n't be able to use it for correlation. Call-ID is often changed enroute. So=
, if one endpoint puts it in an XMPP messages as a reference to a SIP sessi=
on, the other endpoint won't recognize it as it has a different Call-ID for=
 the same session.

Markus

________________________________
From: ext Rifaat Shekh-Yusef [mailto:rifatyu@nortel.com]
Sent: 23 September, 2009 22:57
To: Veikkolainen Simo (Nokia-D/Helsinki); dispatch@ietf.org; Mary Barnes; g=
onzalo.camarillo@ericsson.com
Cc: Isomaki Markus (Nokia-CIC/Espoo)
Subject: RE: [dispatch] New topic proposal for DISPATCH

Simo, Markus,

I am very interested in this work and I like your approach.

I have a question regarding the open issue you have in section 5.2:
Why do you require the correlation-value to be an end-to-end identifier?
In the case where there is an SBC that creates a new dialog for the target,=
 would it not be enough for each endpoint to use the Call-ID of the other d=
ialog as the correlation-value?

Thanks,
 Rifaat


________________________________
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Simo.Veikkolainen@nokia.com
Sent: Friday, September 04, 2009 8:09 AM
To: dispatch@ietf.org; Barnes, Mary (RICH2:AR00); gonzalo.camarillo@ericsso=
n.com
Cc: Markus.Isomaki@nokia.com
Subject: [dispatch] New topic proposal for DISPATCH

Hi,

We would like to propose a new DISPATCH topic on how SIP and XMPP can be us=
ed together in a complementary fashion.

We have been working on a solution which combines SIP based VoIP and XMPP b=
ased IM and Presence. The requirements and a proposed solution is outlined =
in http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01.

The motivation for this work comes from experience which shows that most st=
andards based VoIP deployments use SIP. At the same time, the Extensible Me=
ssaging and Presence Protocol (XMPP) is widely used in instant messaging an=
d presence services. An interesting scenario arises when a SIP based voice =
(and video) service is combined together with an XMPP based instant messagi=
ng and presence service.

Note that the goal in this work is not SIP-XMPP protocol translation, but t=
o define protocol extensions and best practises such that SIP based VoIP an=
d XMPP based IM and presence could be seamlessly combined and offered to th=
e end user. For rapid deployment, we assume no changes in the server infras=
tructure, and instead impose new requirements and protocol extensions only =
to the endpoints.

We would like to get some discussion going on the use case itself as well a=
s on the solution. Also, we would like to hear you thoughts on DISPATCH or =
a new "dispatched" mini-WG as the home for such work.

Exact charter proposal and problem statemen is to follow.

Regards,
Simo and Markus



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<BODY>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D705224406-24092009>Hi Rifaat,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D705224406-24092009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D705224406-24092009>In our proposal the correlation works so=20
that:</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D705224406-24092009>1. UAs engaged in a SIP session/dialog exchange =
an=20
identifier related to that session (as well as their XMPP full=20
JIDs)</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D705224406-24092009>2. They insert them in the very first XMPP messa=
ges=20
they subsequently exchange, in order to correlate that exchange to be relat=
ed to=20
the SIP session.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D705224406-24092009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D705224406-24092009>The correlation identifier has to be end-to-end,=
=20
otherwise the other end won't be able to use it for correlation. Call-ID is=
=20
often changed enroute. So, if one endpoint puts it in an XMPP messages as a=
=20
reference to a SIP session, the other endpoint won't recognize it as it has=
 a=20
different Call-ID for the same session.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D705224406-24092009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D705224406-24092009>Markus</SPAN></FONT></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px soli=
d; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> ext Rifaat Shekh-Yusef=20
  [mailto:rifatyu@nortel.com] <BR><B>Sent:</B> 23 September, 2009=20
  22:57<BR><B>To:</B> Veikkolainen Simo (Nokia-D/Helsinki); dispatch@ietf.o=
rg;=20
  Mary Barnes; gonzalo.camarillo@ericsson.com<BR><B>Cc:</B> Isomaki Markus=
=20
  (Nokia-CIC/Espoo)<BR><B>Subject:</B> RE: [dispatch] New topic proposal fo=
r=20
  DISPATCH<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV><SPAN class=3D443113913-23092009><FONT face=3DArial color=3D#0000ff=
=20
  size=3D2>Simo, Markus,</FONT></SPAN></DIV>
  <DIV><SPAN class=3D443113913-23092009><FONT face=3DArial color=3D#0000ff=
=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D443113913-23092009><FONT face=3DArial color=3D#0000ff =
size=3D2>I am=20
  very interested in this work and I like your approach.</FONT></SPAN></DIV=
>
  <DIV><SPAN class=3D443113913-23092009><FONT face=3DArial color=3D#0000ff=
=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D443113913-23092009><FONT face=3DArial color=3D#0000ff =
size=3D2>I=20
  have a question regarding the open issue you have in section=20
  5.2:</FONT></SPAN></DIV>
  <DIV><SPAN class=3D443113913-23092009><FONT face=3DArial color=3D#0000ff =
size=3D2>Why=20
  do you require the correlation-value to be an end-to-end=20
  identifier?</FONT></SPAN></DIV>
  <DIV><SPAN class=3D443113913-23092009><FONT face=3DArial color=3D#0000ff =
size=3D2>In=20
  the case&nbsp;where there is an SBC that creates a new dialog for the tar=
get,=20
  w</FONT></SPAN><SPAN class=3D443113913-23092009><FONT face=3DArial color=
=3D#0000ff=20
  size=3D2>ould it not be enough for each endpoint to use the Call-ID of th=
e other=20
  dialog as the correlation-value?</FONT></SPAN></DIV>
  <DIV><SPAN class=3D443113913-23092009><FONT face=3DArial color=3D#0000ff=
=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D443113913-23092009><FONT face=3DArial color=3D#0000ff=
=20
  size=3D2>Thanks,</FONT></SPAN></DIV>
  <DIV><SPAN class=3D443113913-23092009><FONT face=3DArial color=3D#0000ff=
=20
  size=3D2>&nbsp;Rifaat</FONT></SPAN></DIV>
  <DIV><SPAN class=3D443113913-23092009><FONT face=3DArial color=3D#0000ff=
=20
  size=3D2></FONT></SPAN>&nbsp;</DIV><BR>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> dispatch-bounces@ietf.org=20
  [mailto:dispatch-bounces@ietf.org] <B>On Behalf Of=20
  </B>Simo.Veikkolainen@nokia.com<BR><B>Sent:</B> Friday, September 04, 200=
9=20
  8:09 AM<BR><B>To:</B> dispatch@ietf.org; Barnes, Mary (RICH2:AR00);=20
  gonzalo.camarillo@ericsson.com<BR><B>Cc:</B>=20
  Markus.Isomaki@nokia.com<BR><B>Subject:</B> [dispatch] New topic proposal=
 for=20
  DISPATCH<BR></FONT><BR></DIV>
  <DIV></DIV><FONT face=3D"Arial, sans-serif" size=3D2>
  <DIV>Hi,</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>We would like to propose a new DISPATCH topic on how SIP and XMPP ca=
n be=20
  used together in a complementary fashion.</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>We have been working on a solution which combines SIP based VoIP and=
 XMPP=20
  based IM and Presence. The requirements and a proposed solution is outlin=
ed in=20
  <A=20
  href=3D"http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01=
"><FONT=20
  color=3D#0000ff><U>http://tools.ietf.org/html/draft-veikkolainen-sip-voip=
-xmpp-im-01</U></FONT></A>.</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>The motivation for this work comes from experience which shows that =
most=20
  standards based VoIP deployments use SIP. At the same time, the Extensibl=
e=20
  Messaging and Presence Protocol (XMPP) is widely used in instant messagin=
g and=20
  presence services. An interesting scenario arises when a SIP based voice =
(and=20
  video) service is combined together with an XMPP based instant messaging =
and=20
  presence service. </DIV>
  <DIV>&nbsp;</DIV>
  <DIV>Note that the goal in this work is not SIP-XMPP protocol translation=
, but=20
  to define protocol extensions and best practises such that SIP based VoIP=
 and=20
  XMPP based IM and presence could be seamlessly combined and offered to th=
e end=20
  user. For rapid deployment, we assume no changes in the server infrastruc=
ture,=20
  and instead impose new requirements and protocol extensions only to the=20
  endpoints.</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>We would like to get some discussion going on the use case itself as=
 well=20
  as on the solution. Also, we would like to hear you thoughts on DISPATCH =
or a=20
  new "dispatched" mini-WG as the home for such work.</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>Exact charter proposal and problem statemen is to follow. </DIV>
  <DIV>&nbsp;</DIV>
  <DIV>Regards,</DIV>
  <DIV>Simo and Markus</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;</DIV></BLOCKQUOTE></FONT></BODY></HTML>

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From gunnar.hellstrom@omnitor.se  Thu Sep 24 02:49:17 2009
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From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
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Cc: dispatch@ietf.org
Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM and	presence
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There is another aspect of text communication that I would like to propose
to bring into this charter.

It is to provide a more "real-timish" text conversational mode, where the
text is sent time-sampled in short chunks during its composition rather than
waiting on user command to send a completed message. The presentation should
then present the chunks consecutively until a delimiter is received. 

We have defined the real-time text mode for transmission with RTP, in RFC
4103, and want of course continue supporting that transport. With RTP it is
realistic to get real real-time transmission, with transmission in 300 ms
intervals. With XMPP, the overead makes it perhaps more realistic to
transmit only once per second, but still get a quite useable real-timish
result, that would be appreciated by the users.

So, I suggest to bring real-timish text transmission of XMPP into this
charter.

Gunnar

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
Of Simo.Veikkolainen@nokia.com
Sent: Thursday, September 24, 2009 8:34 AM
To: emcho@sip-communicator.org
Cc: dispatch@ietf.org
Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM
and presence

 

>-----Original Message-----
>From: Emil Ivov [mailto:emil@sip-communicator.org] On Behalf Of ext 
>Emil Ivov
>Sent: 23 September, 2009 15:46
>To: Veikkolainen Simo (Nokia-D/Helsinki)
>Cc: pkyzivat@cisco.com; dispatch@ietf.org
>Subject: Re: [dispatch] Charter proposal on combining SIP voice with 
>XMPP IM and presence
>
>Hey Simo,
>
>Simo.Veikkolainen@nokia.com wrote:
>> As has been pointed out in earlier emails, there is no automatic 
>> correlation of SIP and XMPP user id's and thus an AOR 
>> alice@example.com may not belong to the same person as the JID 
>> alice@example.com. This rules out the possibility of responding to a 
>> SIP MESSAGE with an XMPP <message> stanza without some a priori 
>> knowledge on Alice's identity in XMPP domain.
>
>I understand why we would not want to rely on an implicit correlation 
>like for example JID == AOR. However an explicit correlation such as an 
>XMPP contact header in SIP messages and a SIP URI in the XMPP vcards 
>would be very helpful. A XMPP+SIP user receiving a call from someone 
>not on their contact list is likely to wish to add the remote party to 
>their roster or send them instant messages. Not being able to do so 
>would feel awkward to the user.

Yes. Explicit correlation (carrying SIP identities in XMPP or vice versa) is
what we proposed in our draft, and we think could indeed be useful.

>
>> From an end user's point of view it makes no difference
>which protocol
>> is used. Personally, I would not mind having all my messaging (IM's 
>> via different service providers, SMS, email etc.) visible in
>a single
>> application view, and let the application hide some of the
>complexity.
>
>Not sure I understand. Does this mean that you expect messages to be 
>traveling over both SIMPLE and XMPP? If yes, then we definitely need 
>some priority rules.

What I meant was that from a UI point of view, all the end user needs to see
is a contact name and possibility send a message (or call). A lot of the
complexity of selecting the actual protocol *could* be hidden by the
application. This may not be what all application developers wish, but for
example the phonebooks in some mobile devices seem to be going to that
direction, combining IM and SMS in one "conversation" view.

So, for example, if a user wants the send an initial message to a contact
whose both SIP and XMPP IDs are known, the app could do the selection of
protocol, depending on the other user's presence information. However, when
responding to a message in a conversation, it probably is better to use the
same protocol that was used when sending, unless the client knows that the
other client's UI can render the messages in the same conversation view. 

So, I'll I'm saying is we need to carefully consider making normative
statements that could potentially limit the application design and
usability. 

Simo

>Cheers,
>Emil
>
>
>> 
>> Simo
>> 
>>>> So, if you think the above are worth considering how about adding 
>>>> something along these lines:
>>>> 
>>>> disambiguation: both XMPP and SIP define semantics that
>>> allow each of
>>>> the protocols to offer the complete set of services
>>> discussed in this
>>>> charter (i.e. voice, video, messaging, and presence). In
>>> environments
>>>> where one or more of these services are supported with both
>>> protocols
>>>> clients should be able to use a common procedure for
>>> determining which
>>>> one to use and under what conditions.
>>>> 
>>>> half-failures: in cases where use of one of the protocols becomes 
>>>> impossible (e.g. due to a server failure) clients should
>be able  to
>>>> follow clear procedures on how to behave, which services
>could they
>>>> try to reuse over the protocol that is still available and
>>> under what
>>>> conditions.
>>>> 
>>>> WDYT?
>>>> 
>>>> Emil
>>>> 
>>>> 
>>>> Simo.Veikkolainen@nokia.com wrote:
>>>>> Hello,
>>>>> 
>>>>> Below is the draft charter text for our proposal on
>>> combining SIP voice with XMPP IM and presence.
>>>>> Comments are very welcome!
>>>>> 
>>>>> Regards, Simo
>>>>> 
>>>>> 
>>>>> -----------------
>>>>> 
>>>>> 
>>>>> Combining SIP VoIP and XMPP IM/Presence 
>>>>> =======================================
>>>>> 
>>>>> Currently most standards-based Voice over IP (VoIP)
>>> deployments use the Session Initiation Protocol (SIP).  In
>addition
>>> to providing basic voice service SIP has an extensive support for 
>>> more advanced telephony features including interworking with the 
>>> traditional Public Switched Telephone Network (PSTN).  SIP is also 
>>> gaining popularity in the field of video communication. At
>the same
>>> time, the Extensible Messaging and Presence Protocol (XMPP) is 
>>> enjoying widespread usage in instant messaging and presence
>services.
>>>>> Both SIP and XMPP offer extensions for voice as well as IM
>>> and presence (XMPP voice via the Jingle extension, and SIP 
>>> IM/presence via SIMPLE protocols). However, widespread
>deployment of
>>> these extensions has not so far taken place. In order to speed up 
>>> deployment of integrated VoIP and IM systems, SIP based voice and 
>>> XMPP based IM/Presence could be combined in endpoints to offer a 
>>> voice, IM and presence service without any changes to existing SIP 
>>> and XMPP service infrastrucure.
>>>>> The objective of this Working Group is to develop solutions for 
>>>>> combining SIP based voice with XMPP based IM and Presence
>>> such that a
>>>>> unified user experience can be offered to end user. 
>>>>> Specifically, solutions are needed on - addressing; end users 
>>>>> should be able to initiate sessions
>>> to a user identity in either SIP or XMPP domain. The corresponding 
>>> user identity in the other protocol domain needs to be learned 
>>> automatically.
>>>>> - session correlation; endpoints need to be able to
>correlate voice
>>>>> sessions with IM/Presence such that they can be presented
>>>>> 
>>>>> 
>>> to the end
>>>>> user in a seamless fashion - presence; it should be possible to 
>>>>> change the XMPP
>>> presence status based on the user's activity in the SIP domain.
>>>>> The goal is to rely on existing service infrastructre, and
>>> new requirements should be imposed only to the endpoint.
>>>>> Protocol interworking, that is, translation from one
>>> protocol to another, is out of scope of this WG.
>>>>> Milestones Feb 2010 Problem statement and use cases submitted to 
>>>>> IESG as Informational Dec 2010 Specification on combining
>SIP based
>>> voice and XMPP based IM/Presence in a seamless manner submitted to 
>>> IESG as Proposed Standard.
>>>>> -----------------
>>>>> _______________________________________________ dispatch mailing 
>>>>> list dispatch@ietf.org 
>>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>> 
>
>-- 
>Emil Ivov, Ph.D.                               67000 Strasbourg,
>Project Lead                                   France
>SIP Communicator
>emcho@sip-communicator.org                     PHONE: +33.1.77.62.43.30
>http://sip-communicator.org                    FAX:   +33.1.77.62.47.31
>
_______________________________________________
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From emcho@sip-communicator.org  Thu Sep 24 03:06:12 2009
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References: <B23B311878A0B4438F5F09F47E01AAE03B18566B21@NOK-EUMSG-01.mgdnok.nokia.com><4AB8BC67.3080702@sip-communicator.org> <4AB8CFEC.8020703@cisco.com><B23B311878A0B4438F5F09F47E01AAE03B1886AC5E@NOK-EUMSG-01.mgdnok.nokia.com><4ABA18A0.5070603@sip-communicator.org> <B23B311878A0B4438F5F09F47E01AAE03B1886B4A7@NOK-EUMSG-01.mgdnok.nokia.com> <038101ca3cfc$738b13c0$e800a8c0@GunnarH>
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Cc: Simo.Veikkolainen@nokia.com, dispatch@ietf.org
Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM and	presence
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Hey Gunnar,

I am not sure I understand how this fits the XMPP+SIP work.
Isn't it something that would be better suited by a pure XMPP entity
such as the XMPP WG or xmpp.org? Or do you think that there's some
aspect in this that can only be handled with SIP and another with XMPP?

IIRC, that's one of the features that appear in Google Wave's demos so
maybe there's already some XMPP related work in that direction.

Cheers,
Emil

Gunnar Hellstrom wrote:
> There is another aspect of text communication that I would like to propose
> to bring into this charter.
> 
> It is to provide a more "real-timish" text conversational mode, where the
> text is sent time-sampled in short chunks during its composition rather than
> waiting on user command to send a completed message. The presentation should
> then present the chunks consecutively until a delimiter is received. 
> 
> We have defined the real-time text mode for transmission with RTP, in RFC
> 4103, and want of course continue supporting that transport. With RTP it is
> realistic to get real real-time transmission, with transmission in 300 ms
> intervals. With XMPP, the overead makes it perhaps more realistic to
> transmit only once per second, but still get a quite useable real-timish
> result, that would be appreciated by the users.
> 
> So, I suggest to bring real-timish text transmission of XMPP into this
> charter.
> 
> Gunnar
> 
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
> Of Simo.Veikkolainen@nokia.com
> Sent: Thursday, September 24, 2009 8:34 AM
> To: emcho@sip-communicator.org
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM
> and presence
> 
>  
> 
>> -----Original Message-----
>> From: Emil Ivov [mailto:emil@sip-communicator.org] On Behalf Of ext 
>> Emil Ivov
>> Sent: 23 September, 2009 15:46
>> To: Veikkolainen Simo (Nokia-D/Helsinki)
>> Cc: pkyzivat@cisco.com; dispatch@ietf.org
>> Subject: Re: [dispatch] Charter proposal on combining SIP voice with 
>> XMPP IM and presence
>>
>> Hey Simo,
>>
>> Simo.Veikkolainen@nokia.com wrote:
>>> As has been pointed out in earlier emails, there is no automatic 
>>> correlation of SIP and XMPP user id's and thus an AOR 
>>> alice@example.com may not belong to the same person as the JID 
>>> alice@example.com. This rules out the possibility of responding to a 
>>> SIP MESSAGE with an XMPP <message> stanza without some a priori 
>>> knowledge on Alice's identity in XMPP domain.
>> I understand why we would not want to rely on an implicit correlation 
>> like for example JID == AOR. However an explicit correlation such as an 
>> XMPP contact header in SIP messages and a SIP URI in the XMPP vcards 
>> would be very helpful. A XMPP+SIP user receiving a call from someone 
>> not on their contact list is likely to wish to add the remote party to 
>> their roster or send them instant messages. Not being able to do so 
>> would feel awkward to the user.
> 
> Yes. Explicit correlation (carrying SIP identities in XMPP or vice versa) is
> what we proposed in our draft, and we think could indeed be useful.
> 
>>> From an end user's point of view it makes no difference
>> which protocol
>>> is used. Personally, I would not mind having all my messaging (IM's 
>>> via different service providers, SMS, email etc.) visible in
>> a single
>>> application view, and let the application hide some of the
>> complexity.
>>
>> Not sure I understand. Does this mean that you expect messages to be 
>> traveling over both SIMPLE and XMPP? If yes, then we definitely need 
>> some priority rules.
> 
> What I meant was that from a UI point of view, all the end user needs to see
> is a contact name and possibility send a message (or call). A lot of the
> complexity of selecting the actual protocol *could* be hidden by the
> application. This may not be what all application developers wish, but for
> example the phonebooks in some mobile devices seem to be going to that
> direction, combining IM and SMS in one "conversation" view.
> 
> So, for example, if a user wants the send an initial message to a contact
> whose both SIP and XMPP IDs are known, the app could do the selection of
> protocol, depending on the other user's presence information. However, when
> responding to a message in a conversation, it probably is better to use the
> same protocol that was used when sending, unless the client knows that the
> other client's UI can render the messages in the same conversation view. 
> 
> So, I'll I'm saying is we need to carefully consider making normative
> statements that could potentially limit the application design and
> usability. 
> 
> Simo
> 
>> Cheers,
>> Emil
>>
>>
>>> Simo
>>>
>>>>> So, if you think the above are worth considering how about adding 
>>>>> something along these lines:
>>>>>
>>>>> disambiguation: both XMPP and SIP define semantics that
>>>> allow each of
>>>>> the protocols to offer the complete set of services
>>>> discussed in this
>>>>> charter (i.e. voice, video, messaging, and presence). In
>>>> environments
>>>>> where one or more of these services are supported with both
>>>> protocols
>>>>> clients should be able to use a common procedure for
>>>> determining which
>>>>> one to use and under what conditions.
>>>>>
>>>>> half-failures: in cases where use of one of the protocols becomes 
>>>>> impossible (e.g. due to a server failure) clients should
>> be able  to
>>>>> follow clear procedures on how to behave, which services
>> could they
>>>>> try to reuse over the protocol that is still available and
>>>> under what
>>>>> conditions.
>>>>>
>>>>> WDYT?
>>>>>
>>>>> Emil
>>>>>
>>>>>
>>>>> Simo.Veikkolainen@nokia.com wrote:
>>>>>> Hello,
>>>>>>
>>>>>> Below is the draft charter text for our proposal on
>>>> combining SIP voice with XMPP IM and presence.
>>>>>> Comments are very welcome!
>>>>>>
>>>>>> Regards, Simo
>>>>>>
>>>>>>
>>>>>> -----------------
>>>>>>
>>>>>>
>>>>>> Combining SIP VoIP and XMPP IM/Presence 
>>>>>> =======================================
>>>>>>
>>>>>> Currently most standards-based Voice over IP (VoIP)
>>>> deployments use the Session Initiation Protocol (SIP).  In
>> addition
>>>> to providing basic voice service SIP has an extensive support for 
>>>> more advanced telephony features including interworking with the 
>>>> traditional Public Switched Telephone Network (PSTN).  SIP is also 
>>>> gaining popularity in the field of video communication. At
>> the same
>>>> time, the Extensible Messaging and Presence Protocol (XMPP) is 
>>>> enjoying widespread usage in instant messaging and presence
>> services.
>>>>>> Both SIP and XMPP offer extensions for voice as well as IM
>>>> and presence (XMPP voice via the Jingle extension, and SIP 
>>>> IM/presence via SIMPLE protocols). However, widespread
>> deployment of
>>>> these extensions has not so far taken place. In order to speed up 
>>>> deployment of integrated VoIP and IM systems, SIP based voice and 
>>>> XMPP based IM/Presence could be combined in endpoints to offer a 
>>>> voice, IM and presence service without any changes to existing SIP 
>>>> and XMPP service infrastrucure.
>>>>>> The objective of this Working Group is to develop solutions for 
>>>>>> combining SIP based voice with XMPP based IM and Presence
>>>> such that a
>>>>>> unified user experience can be offered to end user. 
>>>>>> Specifically, solutions are needed on - addressing; end users 
>>>>>> should be able to initiate sessions
>>>> to a user identity in either SIP or XMPP domain. The corresponding 
>>>> user identity in the other protocol domain needs to be learned 
>>>> automatically.
>>>>>> - session correlation; endpoints need to be able to
>> correlate voice
>>>>>> sessions with IM/Presence such that they can be presented
>>>>>>
>>>>>>
>>>> to the end
>>>>>> user in a seamless fashion - presence; it should be possible to 
>>>>>> change the XMPP
>>>> presence status based on the user's activity in the SIP domain.
>>>>>> The goal is to rely on existing service infrastructre, and
>>>> new requirements should be imposed only to the endpoint.
>>>>>> Protocol interworking, that is, translation from one
>>>> protocol to another, is out of scope of this WG.
>>>>>> Milestones Feb 2010 Problem statement and use cases submitted to 
>>>>>> IESG as Informational Dec 2010 Specification on combining
>> SIP based
>>>> voice and XMPP based IM/Presence in a seamless manner submitted to 
>>>> IESG as Proposed Standard.
>>>>>> -----------------
>>>>>> _______________________________________________ dispatch mailing 
>>>>>> list dispatch@ietf.org 
>>>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>>>
>> -- 
>> Emil Ivov, Ph.D.                               67000 Strasbourg,
>> Project Lead                                   France
>> SIP Communicator
>> emcho@sip-communicator.org                     PHONE: +33.1.77.62.43.30
>> http://sip-communicator.org                    FAX:   +33.1.77.62.47.31
>>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 
> __________ NOD32 4452 (20090924) Information __________
> 
> Detta meddelande dr genomsvkt av NOD32 Antivirus.
> http://www.nod32.com
> 
> 
> 

-- 
Emil Ivov, Ph.D.                               67000 Strasbourg,
Project Lead                                   France
SIP Communicator
emcho@sip-communicator.org                     PHONE: +33.1.77.62.43.30
http://sip-communicator.org                    FAX:   +33.1.77.62.47.31

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Cc: Simo.Veikkolainen@nokia.com, dispatch@ietf.org
Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM and	presence
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

There are XMPP implementations that transmit character-by-character
text, but no one has ever brought a proposal to standardize this feature
to the XMPP Standards Foundation, and the XSF hasn't sought one out.
That might change in the future with the Wave work.

On 9/24/09 4:07 AM, Emil Ivov wrote:
> Hey Gunnar,
> 
> I am not sure I understand how this fits the XMPP+SIP work.
> Isn't it something that would be better suited by a pure XMPP entity
> such as the XMPP WG or xmpp.org? Or do you think that there's some
> aspect in this that can only be handled with SIP and another with XMPP?
> 
> IIRC, that's one of the features that appear in Google Wave's demos so
> maybe there's already some XMPP related work in that direction.
> 
> Cheers,
> Emil
> 
> Gunnar Hellstrom wrote:
>> There is another aspect of text communication that I would like to propose
>> to bring into this charter.
>>
>> It is to provide a more "real-timish" text conversational mode, where the
>> text is sent time-sampled in short chunks during its composition rather than
>> waiting on user command to send a completed message. The presentation should
>> then present the chunks consecutively until a delimiter is received. 
>>
>> We have defined the real-time text mode for transmission with RTP, in RFC
>> 4103, and want of course continue supporting that transport. With RTP it is
>> realistic to get real real-time transmission, with transmission in 300 ms
>> intervals. With XMPP, the overead makes it perhaps more realistic to
>> transmit only once per second, but still get a quite useable real-timish
>> result, that would be appreciated by the users.
>>
>> So, I suggest to bring real-timish text transmission of XMPP into this
>> charter.
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Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM and	presence
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I support this proposal and will be happy to assist in the work.

--Avshalom




From:
<Simo.Veikkolainen@nokia.com>
To:
<dispatch@ietf.org>
Date:
21/09/2009 03:29 PM
Subject:
[dispatch] Charter proposal on combining SIP voice with XMPP IM and 
presence
Sent by:
dispatch-bounces@ietf.org



Hello,

Below is the draft charter text for our proposal on combining SIP voice 
with XMPP IM and presence.

Comments are very welcome!

Regards,
Simo


-----------------


Combining SIP VoIP and XMPP IM/Presence
=======================================

Currently most standards-based Voice over IP (VoIP) deployments use the 
Session Initiation Protocol (SIP).  In addition to providing basic voice 
service SIP has an extensive support for more advanced telephony features 
including interworking with the traditional Public Switched Telephone 
Network (PSTN).  SIP is also gaining popularity in the field of video 
communication. At the same time, the Extensible Messaging and Presence 
Protocol (XMPP) is enjoying widespread usage in instant messaging and 
presence services. 

Both SIP and XMPP offer extensions for voice as well as IM and presence 
(XMPP voice via the Jingle extension, and SIP IM/presence via SIMPLE 
protocols). However, widespread deployment of these extensions has not so 
far taken place. In order to speed up deployment of integrated VoIP and IM 
systems, SIP based voice and XMPP based IM/Presence could be combined in 
endpoints to offer a voice, IM and presence service without any changes to 
existing SIP and XMPP service infrastrucure. 

The objective of this Working Group is to develop solutions for combining 
SIP based voice with XMPP based IM and Presence such that a unified user 
experience can be offered to end user. Specifically, solutions are needed 
on 
- addressing; end users should be able to initiate sessions to a user 
identity in either SIP or XMPP domain. The corresponding user identity in 
the other protocol domain needs to be learned automatically.
- session correlation; endpoints need to be able to correlate voice 
sessions with IM/Presence such that they can be presented to the end user 
in a seamless fashion
- presence; it should be possible to change the XMPP presence status based 
on the user's activity in the SIP domain.

The goal is to rely on existing service infrastructre, and new 
requirements should be imposed only to the endpoint. 

Protocol interworking, that is, translation from one protocol to another, 
is out of scope of this WG.

Milestones
Feb 2010 Problem statement and use cases submitted to IESG as 
Informational
Dec 2010 Specification on combining SIP based voice and XMPP based 
IM/Presence in a seamless manner submitted to IESG as Proposed Standard.

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


--=_alternative 00451A45C2257638_=
Content-Type: text/html; charset="US-ASCII"

<font size=2 face="sans-serif">I support this proposal and will be happy
to assist in the work.</font>
<br><font size=2 face="sans-serif"><br>
--Avshalom<br>
</font>
<br>
<br>
<br>
<table width=100%>
<tr valign=top>
<td><font size=1 color=#5f5f5f face="sans-serif">From:</font>
<td><font size=1 face="sans-serif">&lt;Simo.Veikkolainen@nokia.com&gt;</font>
<tr valign=top>
<td><font size=1 color=#5f5f5f face="sans-serif">To:</font>
<td><font size=1 face="sans-serif">&lt;dispatch@ietf.org&gt;</font>
<tr valign=top>
<td><font size=1 color=#5f5f5f face="sans-serif">Date:</font>
<td><font size=1 face="sans-serif">21/09/2009 03:29 PM</font>
<tr valign=top>
<td><font size=1 color=#5f5f5f face="sans-serif">Subject:</font>
<td><font size=1 face="sans-serif">[dispatch] Charter proposal on combining
SIP voice with XMPP IM and &nbsp; &nbsp; &nbsp; &nbsp;presence</font>
<tr valign=top>
<td><font size=1 color=#5f5f5f face="sans-serif">Sent by:</font>
<td><font size=1 face="sans-serif">dispatch-bounces@ietf.org</font></table>
<br>
<hr noshade>
<br>
<br>
<br><tt><font size=2>Hello,<br>
<br>
Below is the draft charter text for our proposal on combining SIP voice
with XMPP IM and presence.<br>
<br>
Comments are very welcome!<br>
<br>
Regards,<br>
Simo<br>
<br>
<br>
-----------------<br>
<br>
<br>
Combining SIP VoIP and XMPP IM/Presence<br>
=======================================<br>
<br>
Currently most standards-based Voice over IP (VoIP) deployments use the
Session Initiation Protocol (SIP). &nbsp;In addition to providing basic
voice service SIP has an extensive support for more advanced telephony
features including interworking with the traditional Public Switched Telephone
Network (PSTN). &nbsp;SIP is also gaining popularity in the field of video
communication. At the same time, the Extensible Messaging and Presence
Protocol (XMPP) is enjoying widespread usage in instant messaging and presence
services. &nbsp;<br>
<br>
Both SIP and XMPP offer extensions for voice as well as IM and presence
(XMPP voice via the Jingle extension, and SIP IM/presence via SIMPLE protocols).
However, widespread deployment of these extensions has not so far taken
place. In order to speed up deployment of integrated VoIP and IM systems,
SIP based voice and XMPP based IM/Presence could be combined in endpoints
to offer a voice, IM and presence service without any changes to existing
SIP and XMPP service infrastrucure. <br>
<br>
The objective of this Working Group is to develop solutions for combining
SIP based voice with XMPP based IM and Presence such that a unified user
experience can be offered to end user. Specifically, solutions are needed
on <br>
- addressing; end users should be able to initiate sessions to a user identity
in either SIP or XMPP domain. The corresponding user identity in the other
protocol domain needs to be learned automatically.<br>
- session correlation; endpoints need to be able to correlate voice sessions
with IM/Presence such that they can be presented to the end user in a seamless
fashion<br>
- presence; it should be possible to change the XMPP presence status based
on the user's activity in the SIP domain.<br>
<br>
The goal is to rely on existing service infrastructre, and new requirements
should be imposed only to the endpoint. <br>
<br>
Protocol interworking, that is, translation from one protocol to another,
is out of scope of this WG.<br>
<br>
Milestones<br>
Feb 2010 Problem statement and use cases submitted to IESG as Informational<br>
Dec 2010 Specification on combining SIP based voice and XMPP based IM/Presence
in a seamless manner submitted to IESG as Proposed Standard.<br>
<br>
-----------------<br>
_______________________________________________<br>
dispatch mailing list<br>
dispatch@ietf.org<br>
</font></tt><a href=https://www.ietf.org/mailman/listinfo/dispatch><tt><font size=2>https://www.ietf.org/mailman/listinfo/dispatch</font></tt></a><tt><font size=2><br>
</font></tt>
<br>
--=_alternative 00451A45C2257638_=--

From Sebastian.Schumann@t-com.sk  Thu Sep 24 07:25:57 2009
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From: Schumann Sebastian <Sebastian.Schumann@t-com.sk>
To: 'Avshalom Houri' <AVSHALOM@il.ibm.com>, "'Simo.Veikkolainen@nokia.com'" <Simo.Veikkolainen@nokia.com>, "'dispatch@ietf.org'" <dispatch@ietf.org>
Date: Thu, 24 Sep 2009 16:27:00 +0200
Thread-Topic: [dispatch] Charter proposal on combining SIP voice with XMPP IM	and	presence
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Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM	and	presence
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Hello Everyone

>From SIP operator point of view, I see it very important to think about com=
bined architecture exactly like Simo describes them: (More or less :) matur=
e implementations of SIP and standardized and mature instant messaging/pres=
ence where clients exist: XMPP. Both candidates are deployed massively out =
there, however still separately. I see big Telco operator's not using XMPP =
for voice in the future as well as Google and other large XMPP deployments =
considering SIP for their IM services.

>From interworking point of view, I think the efforts should not only go int=
o the direction where several drafts are available (communicate as SIP/XMPP=
 user with XMPP/SIP user) but also the into the direction of merging identi=
ties (what Simo describes: it should be possible to change the XMPP presenc=
e status based on the user's activity in the SIP domain). It is more a prop=
osal of users using both networks and "merging" their states. If you use, e=
.g., XMPP for presence, you might want to merge the user's SIP device prese=
nce state as an actual XMPP presence state (the phone as one resource, the =
user being busy when in a call). In other cases, you do not want him to app=
ear online in the XMPP client, if only a SIP hardphone is registered. It is=
 online, but cannot receive any IM communication...

I think a lot of use cases and scenarios can be created, even more will ari=
se in the future, as both infrastructures are growing. Yet, I see both grow=
ing separately and in different "domains" (internet vs. Telco world).

I would like to participate in moving the consolidation forward, hence I se=
cond the proposal!

Best regards
Sebastian

________________________________
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Avshalom Houri
Sent: Thursday, 24. September 2009 16:10
To: Simo.Veikkolainen@nokia.com; dispatch@ietf.org
Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP I=
M and presence

I support this proposal and will be happy to assist in the work.

--Avshalom



From:   <Simo.Veikkolainen@nokia.com>
To:     <dispatch@ietf.org>
Date:   21/09/2009 03:29 PM
Subject:        [dispatch] Charter proposal on combining SIP voice with XMP=
P IM and        presence
Sent by:        dispatch-bounces@ietf.org

________________________________



Hello,

Below is the draft charter text for our proposal on combining SIP voice wit=
h XMPP IM and presence.

Comments are very welcome!

Regards,
Simo


-----------------


Combining SIP VoIP and XMPP IM/Presence
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

Currently most standards-based Voice over IP (VoIP) deployments use the Ses=
sion Initiation Protocol (SIP).  In addition to providing basic voice servi=
ce SIP has an extensive support for more advanced telephony features includ=
ing interworking with the traditional Public Switched Telephone Network (PS=
TN).  SIP is also gaining popularity in the field of video communication. A=
t the same time, the Extensible Messaging and Presence Protocol (XMPP) is e=
njoying widespread usage in instant messaging and presence services.

Both SIP and XMPP offer extensions for voice as well as IM and presence (XM=
PP voice via the Jingle extension, and SIP IM/presence via SIMPLE protocols=
). However, widespread deployment of these extensions has not so far taken =
place. In order to speed up deployment of integrated VoIP and IM systems, S=
IP based voice and XMPP based IM/Presence could be combined in endpoints to=
 offer a voice, IM and presence service without any changes to existing SIP=
 and XMPP service infrastrucure.

The objective of this Working Group is to develop solutions for combining S=
IP based voice with XMPP based IM and Presence such that a unified user exp=
erience can be offered to end user. Specifically, solutions are needed on
- addressing; end users should be able to initiate sessions to a user ident=
ity in either SIP or XMPP domain. The corresponding user identity in the ot=
her protocol domain needs to be learned automatically.
- session correlation; endpoints need to be able to correlate voice session=
s with IM/Presence such that they can be presented to the end user in a sea=
mless fashion
- presence; it should be possible to change the XMPP presence status based =
on the user's activity in the SIP domain.

The goal is to rely on existing service infrastructre, and new requirements=
 should be imposed only to the endpoint.

Protocol interworking, that is, translation from one protocol to another, i=
s out of scope of this WG.

Milestones
Feb 2010 Problem statement and use cases submitted to IESG as Informational
Dec 2010 Specification on combining SIP based voice and XMPP based IM/Prese=
nce in a seamless manner submitted to IESG as Proposed Standard.

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


--_000_A693B91C76085E49AA39C91A3E0AC7A0B02E0CFAEXCHANGE2stsk_
Content-Type: text/html; charset="us-ascii"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.5848" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D339081614-24092009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Hello Everyone</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D339081614-24092009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D339081614-24092009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>From SIP operator point of view, I see it very imp=
ortant=20
<SPAN lang=3DSK>to think about combined architecture exactly like Simo desc=
ribes=20
them: (More or less :) mature implementations of SIP and standardized and=20
mature&nbsp;instant messaging/presence where clients exist: XMPP. Both=20
candidates are deployed massively out there, however still separately. I se=
e big=20
Telco operator's not using XMPP for voice in the future as well as Google a=
nd=20
other large XMPP deployments considering SIP for their IM=20
services.</SPAN></FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D339081614-24092009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2><SPAN lang=3DSK></SPAN></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial><FONT color=3D#0000ff><SPAN=
=20
class=3D339081614-24092009><FONT size=3D2><SPAN lang=3DSK>From interworking=
 point of=20
view, I think the efforts should not only go into the direction where sever=
al=20
drafts are available (communicate as SIP/XMPP user with XMPP/SIP user) but =
also=20
the into the direction of merging identities (what Simo describes: it shoul=
d be=20
possible to change the XMPP presence status based on the user's activity in=
 the=20
SIP domain). </SPAN></FONT></SPAN><SPAN class=3D339081614-24092009><FONT=20
size=3D2><SPAN lang=3DSK>It is more a proposal of users using both networks=
 and=20
"merging" their states. If you use, e.g., XMPP for presence, you might want=
 to=20
merge the user's SIP device presence state as an actual XMPP presence state=
 (the=20
phone as one resource, the user being busy when in a call). In other cases,=
 you=20
do not want him to appear online in the XMPP client, if only a SIP hardphon=
e is=20
registered. It is online,&nbsp;but cannot receive any IM=20
communication...</SPAN></FONT></SPAN></FONT></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial><FONT color=3D#0000ff><SPAN=
=20
class=3D339081614-24092009><FONT size=3D2><SPAN=20
lang=3DSK></SPAN></FONT></SPAN></FONT></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial><FONT color=3D#0000ff><SPAN=
=20
class=3D339081614-24092009><FONT size=3D2><SPAN lang=3DSK>I think a lot of =
use cases=20
and scenarios can be created, even more will arise&nbsp;in the future, as b=
oth=20
infrastructures are growing. Yet, I see both growing separately and in diff=
erent=20
"domains" (internet vs. Telco world).</SPAN></FONT></SPAN></FONT></FONT></D=
IV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial><FONT color=3D#0000ff><SPAN=
=20
class=3D339081614-24092009><FONT size=3D2><SPAN=20
lang=3DSK></SPAN></FONT></SPAN></FONT></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial><FONT color=3D#0000ff><SPAN=
=20
class=3D339081614-24092009><FONT size=3D2><SPAN lang=3DSK>I would like to p=
articipate=20
in moving the consolidation forward, hence I second the=20
proposal!</SPAN></FONT></SPAN></FONT></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial><FONT color=3D#0000ff><SPAN=
=20
class=3D339081614-24092009><FONT size=3D2><SPAN=20
lang=3DSK></SPAN></FONT></SPAN></FONT></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial><FONT color=3D#0000ff><SPAN=
=20
class=3D339081614-24092009><FONT size=3D2><SPAN lang=3DSK>Best=20
regards</SPAN></FONT></SPAN></FONT></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial><FONT color=3D#0000ff><SPAN=
=20
class=3D339081614-24092009><FONT size=3D2><SPAN=20
lang=3DSK>Sebastian</DIV></SPAN></FONT></SPAN><BR></FONT></FONT>
<BLOCKQUOTE=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px soli=
d; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> dispatch-bounces@ietf.org=20
  [mailto:dispatch-bounces@ietf.org] <B>On Behalf Of </B>Avshalom=20
  Houri<BR><B>Sent:</B> Thursday, 24. September 2009 16:10<BR><B>To:</B>=20
  Simo.Veikkolainen@nokia.com; dispatch@ietf.org<BR><B>Subject:</B> Re:=20
  [dispatch] Charter proposal on combining SIP voice with XMPP IM and=20
  presence<BR></FONT><BR></DIV>
  <DIV></DIV><FONT face=3Dsans-serif size=3D2>I support this proposal and w=
ill be=20
  happy to assist in the work.</FONT> <BR><FONT face=3Dsans-serif=20
  size=3D2><BR>--Avshalom<BR></FONT><BR><BR><BR>
  <TABLE width=3D"100%">
    <TBODY>
    <TR vAlign=3Dtop>
      <TD><FONT face=3Dsans-serif color=3D#5f5f5f size=3D1>From:</FONT>=20
      <TD><FONT face=3Dsans-serif=20
        size=3D1>&lt;Simo.Veikkolainen@nokia.com&gt;</FONT>=20
    <TR vAlign=3Dtop>
      <TD><FONT face=3Dsans-serif color=3D#5f5f5f size=3D1>To:</FONT>=20
      <TD><FONT face=3Dsans-serif size=3D1>&lt;dispatch@ietf.org&gt;</FONT>=
=20
    <TR vAlign=3Dtop>
      <TD><FONT face=3Dsans-serif color=3D#5f5f5f size=3D1>Date:</FONT>=20
      <TD><FONT face=3Dsans-serif size=3D1>21/09/2009 03:29 PM</FONT>=20
    <TR vAlign=3Dtop>
      <TD><FONT face=3Dsans-serif color=3D#5f5f5f size=3D1>Subject:</FONT>=
=20
      <TD><FONT face=3Dsans-serif size=3D1>[dispatch] Charter proposal on=20
        combining SIP voice with XMPP IM and &nbsp; &nbsp; &nbsp;=20
        &nbsp;presence</FONT>=20
    <TR vAlign=3Dtop>
      <TD><FONT face=3Dsans-serif color=3D#5f5f5f size=3D1>Sent by:</FONT>=
=20
      <TD><FONT face=3Dsans-serif=20
    size=3D1>dispatch-bounces@ietf.org</FONT></TR></TBODY></TABLE><BR>
  <HR noShade>
  <BR><BR><BR><TT><FONT size=3D2>Hello,<BR><BR>Below is the draft charter t=
ext for=20
  our proposal on combining SIP voice with XMPP IM and presence.<BR><BR>Com=
ments=20
  are very=20
  welcome!<BR><BR>Regards,<BR>Simo<BR><BR><BR>-----------------<BR><BR><BR>=
Combining=20
  SIP VoIP and XMPP=20
  IM/Presence<BR>=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D<BR><BR>Current=
ly most=20
  standards-based Voice over IP (VoIP) deployments use the Session Initiati=
on=20
  Protocol (SIP). &nbsp;In addition to providing basic voice service SIP ha=
s an=20
  extensive support for more advanced telephony features including interwor=
king=20
  with the traditional Public Switched Telephone Network (PSTN). &nbsp;SIP =
is=20
  also gaining popularity in the field of video communication. At the same =
time,=20
  the Extensible Messaging and Presence Protocol (XMPP) is enjoying widespr=
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To: Schumann Sebastian <Sebastian.Schumann@t-com.sk>
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Cc: "'Simo.Veikkolainen@nokia.com'" <Simo.Veikkolainen@nokia.com>, "'dispatch@ietf.org'" <dispatch@ietf.org>
Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM	and	presence
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Quite agree with you. Seems that there should be integration also in the 
back end and not only on the clients.
The use cases and the requirements work here will determine the direction 
of this work.

--Avshalom




From:
Schumann Sebastian <Sebastian.Schumann@t-com.sk>
To:
Avshalom Houri/Haifa/IBM@IBMIL, "'Simo.Veikkolainen@nokia.com'" 
<Simo.Veikkolainen@nokia.com>, "'dispatch@ietf.org'" <dispatch@ietf.org>
Date:
24/09/2009 05:27 PM
Subject:
RE: [dispatch] Charter proposal on combining SIP voice with XMPP IM and 
presence



Hello Everyone
 
>From SIP operator point of view, I see it very important to think about 
combined architecture exactly like Simo describes them: (More or less :) 
mature implementations of SIP and standardized and mature instant 
messaging/presence where clients exist: XMPP. Both candidates are deployed 
massively out there, however still separately. I see big Telco operator's 
not using XMPP for voice in the future as well as Google and other large 
XMPP deployments considering SIP for their IM services.
 
>From interworking point of view, I think the efforts should not only go 
into the direction where several drafts are available (communicate as 
SIP/XMPP user with XMPP/SIP user) but also the into the direction of 
merging identities (what Simo describes: it should be possible to change 
the XMPP presence status based on the user's activity in the SIP domain). 
It is more a proposal of users using both networks and "merging" their 
states. If you use, e.g., XMPP for presence, you might want to merge the 
user's SIP device presence state as an actual XMPP presence state (the 
phone as one resource, the user being busy when in a call). In other 
cases, you do not want him to appear online in the XMPP client, if only a 
SIP hardphone is registered. It is online, but cannot receive any IM 
communication...
 
I think a lot of use cases and scenarios can be created, even more will 
arise in the future, as both infrastructures are growing. Yet, I see both 
growing separately and in different "domains" (internet vs. Telco world).
 
I would like to participate in moving the consolidation forward, hence I 
second the proposal!
 
Best regards
Sebastian

From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On 
Behalf Of Avshalom Houri
Sent: Thursday, 24. September 2009 16:10
To: Simo.Veikkolainen@nokia.com; dispatch@ietf.org
Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP 
IM and presence

I support this proposal and will be happy to assist in the work. 

--Avshalom



From: 
<Simo.Veikkolainen@nokia.com> 
To: 
<dispatch@ietf.org> 
Date: 
21/09/2009 03:29 PM 
Subject: 
[dispatch] Charter proposal on combining SIP voice with XMPP IM and 
presence 
Sent by: 
dispatch-bounces@ietf.org




Hello,

Below is the draft charter text for our proposal on combining SIP voice 
with XMPP IM and presence.

Comments are very welcome!

Regards,
Simo


-----------------


Combining SIP VoIP and XMPP IM/Presence
=======================================

Currently most standards-based Voice over IP (VoIP) deployments use the 
Session Initiation Protocol (SIP).  In addition to providing basic voice 
service SIP has an extensive support for more advanced telephony features 
including interworking with the traditional Public Switched Telephone 
Network (PSTN).  SIP is also gaining popularity in the field of video 
communication. At the same time, the Extensible Messaging and Presence 
Protocol (XMPP) is enjoying widespread usage in instant messaging and 
presence services. 

Both SIP and XMPP offer extensions for voice as well as IM and presence 
(XMPP voice via the Jingle extension, and SIP IM/presence via SIMPLE 
protocols). However, widespread deployment of these extensions has not so 
far taken place. In order to speed up deployment of integrated VoIP and IM 
systems, SIP based voice and XMPP based IM/Presence could be combined in 
endpoints to offer a voice, IM and presence service without any changes to 
existing SIP and XMPP service infrastrucure. 

The objective of this Working Group is to develop solutions for combining 
SIP based voice with XMPP based IM and Presence such that a unified user 
experience can be offered to end user. Specifically, solutions are needed 
on 
- addressing; end users should be able to initiate sessions to a user 
identity in either SIP or XMPP domain. The corresponding user identity in 
the other protocol domain needs to be learned automatically.
- session correlation; endpoints need to be able to correlate voice 
sessions with IM/Presence such that they can be presented to the end user 
in a seamless fashion
- presence; it should be possible to change the XMPP presence status based 
on the user's activity in the SIP domain.

The goal is to rely on existing service infrastructre, and new 
requirements should be imposed only to the endpoint. 

Protocol interworking, that is, translation from one protocol to another, 
is out of scope of this WG.

Milestones
Feb 2010 Problem statement and use cases submitted to IESG as 
Informational
Dec 2010 Specification on combining SIP based voice and XMPP based 
IM/Presence in a seamless manner submitted to IESG as Proposed Standard.

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


--=_alternative 00514BF2C225763B_=
Content-Type: text/html; charset="US-ASCII"

<font size=2 face="sans-serif">Quite agree with you. Seems that there
should be integration also in the back end and not only on the clients.</font>
<br><font size=2 face="sans-serif">The use cases and the requirements work
here will determine the direction of this work.</font>
<br><font size=2 face="sans-serif"><br>
--Avshalom<br>
</font>
<br>
<br>
<br>
<table width=100%>
<tr valign=top>
<td><font size=1 color=#5f5f5f face="sans-serif">From:</font>
<td><font size=1 face="sans-serif">Schumann Sebastian &lt;Sebastian.Schumann@t-com.sk&gt;</font>
<tr valign=top>
<td><font size=1 color=#5f5f5f face="sans-serif">To:</font>
<td><font size=1 face="sans-serif">Avshalom Houri/Haifa/IBM@IBMIL, &quot;'Simo.Veikkolainen@nokia.com'&quot;
&lt;Simo.Veikkolainen@nokia.com&gt;, &quot;'dispatch@ietf.org'&quot; &lt;dispatch@ietf.org&gt;</font>
<tr valign=top>
<td><font size=1 color=#5f5f5f face="sans-serif">Date:</font>
<td><font size=1 face="sans-serif">24/09/2009 05:27 PM</font>
<tr valign=top>
<td><font size=1 color=#5f5f5f face="sans-serif">Subject:</font>
<td><font size=1 face="sans-serif">RE: [dispatch] Charter proposal on combining
SIP voice with XMPP IM &nbsp; &nbsp; &nbsp; &nbsp;and &nbsp; &nbsp;
&nbsp; &nbsp;presence</font></table>
<br>
<hr noshade>
<br>
<br>
<br><font size=2 color=blue face="Arial">Hello Everyone</font>
<br><font size=3>&nbsp;</font>
<br><font size=2 color=blue face="Arial">From SIP operator point of view,
I see it very important to think about combined architecture exactly like
Simo describes them: (More or less :) mature implementations of SIP and
standardized and mature instant messaging/presence where clients exist:
XMPP. Both candidates are deployed massively out there, however still separately.
I see big Telco operator's not using XMPP for voice in the future as well
as Google and other large XMPP deployments considering SIP for their IM
services.</font>
<br><font size=3>&nbsp;</font>
<br><font size=2 color=blue face="Arial">From interworking point of view,
I think the efforts should not only go into the direction where several
drafts are available (communicate as SIP/XMPP user with XMPP/SIP user)
but also the into the direction of merging identities (what Simo describes:
it should be possible to change the XMPP presence status based on the user's
activity in the SIP domain). It is more a proposal of users using both
networks and &quot;merging&quot; their states. If you use, e.g., XMPP for
presence, you might want to merge the user's SIP device presence state
as an actual XMPP presence state (the phone as one resource, the user being
busy when in a call). In other cases, you do not want him to appear online
in the XMPP client, if only a SIP hardphone is registered. It is online,
but cannot receive any IM communication...</font>
<br><font size=3>&nbsp;</font>
<br><font size=2 color=blue face="Arial">I think a lot of use cases and
scenarios can be created, even more will arise in the future, as both infrastructures
are growing. Yet, I see both growing separately and in different &quot;domains&quot;
(internet vs. Telco world).</font>
<br><font size=3>&nbsp;</font>
<br><font size=2 color=blue face="Arial">I would like to participate in
moving the consolidation forward, hence I second the proposal!</font>
<br><font size=3>&nbsp;</font>
<br><font size=2 color=blue face="Arial">Best regards</font>
<br><font size=2 color=blue face="Arial">Sebastian</font>
<br>
<br>
<hr><font size=2 face="Tahoma"><b>From:</b> dispatch-bounces@ietf.org [</font><a href="mailto:dispatch-bounces@ietf.org"><font size=2 face="Tahoma">mailto:dispatch-bounces@ietf.org</font></a><font size=2 face="Tahoma">]
<b>On Behalf Of </b>Avshalom Houri<b><br>
Sent:</b> Thursday, 24. September 2009 16:10<b><br>
To:</b> Simo.Veikkolainen@nokia.com; dispatch@ietf.org<b><br>
Subject:</b> Re: [dispatch] Charter proposal on combining SIP voice with
XMPP IM and presence</font><font size=3><br>
</font>
<br><font size=2 face="sans-serif">I support this proposal and will be
happy to assist in the work.</font><font size=3> </font><font size=2 face="sans-serif"><br>
<br>
--Avshalom</font><font size=3><br>
<br>
<br>
</font>
<table width=100%>
<tr valign=top>
<td width=10%><font size=1 color=#5f5f5f face="sans-serif">From:</font><font size=3>
</font>
<td width=89%><font size=1 face="sans-serif">&lt;Simo.Veikkolainen@nokia.com&gt;</font><font size=3>
</font>
<tr valign=top>
<td><font size=1 color=#5f5f5f face="sans-serif">To:</font><font size=3>
</font>
<td><font size=1 face="sans-serif">&lt;dispatch@ietf.org&gt;</font><font size=3>
</font>
<tr valign=top>
<td><font size=1 color=#5f5f5f face="sans-serif">Date:</font><font size=3>
</font>
<td><font size=1 face="sans-serif">21/09/2009 03:29 PM</font><font size=3>
</font>
<tr valign=top>
<td><font size=1 color=#5f5f5f face="sans-serif">Subject:</font><font size=3>
</font>
<td><font size=1 face="sans-serif">[dispatch] Charter proposal on combining
SIP voice with XMPP IM and &nbsp; &nbsp; &nbsp; &nbsp;presence</font><font size=3>
</font>
<tr valign=top>
<td><font size=1 color=#5f5f5f face="sans-serif">Sent by:</font><font size=3>
</font>
<td><font size=1 face="sans-serif">dispatch-bounces@ietf.org</font></table>
<br><font size=3><br>
</font>
<hr noshade><font size=3><br>
<br>
</font><tt><font size=2><br>
Hello,<br>
<br>
Below is the draft charter text for our proposal on combining SIP voice
with XMPP IM and presence.<br>
<br>
Comments are very welcome!<br>
<br>
Regards,<br>
Simo<br>
<br>
<br>
-----------------<br>
<br>
<br>
Combining SIP VoIP and XMPP IM/Presence<br>
=======================================<br>
<br>
Currently most standards-based Voice over IP (VoIP) deployments use the
Session Initiation Protocol (SIP). &nbsp;In addition to providing basic
voice service SIP has an extensive support for more advanced telephony
features including interworking with the traditional Public Switched Telephone
Network (PSTN). &nbsp;SIP is also gaining popularity in the field of video
communication. At the same time, the Extensible Messaging and Presence
Protocol (XMPP) is enjoying widespread usage in instant messaging and presence
services. &nbsp;<br>
<br>
Both SIP and XMPP offer extensions for voice as well as IM and presence
(XMPP voice via the Jingle extension, and SIP IM/presence via SIMPLE protocols).
However, widespread deployment of these extensions has not so far taken
place. In order to speed up deployment of integrated VoIP and IM systems,
SIP based voice and XMPP based IM/Presence could be combined in endpoints
to offer a voice, IM and presence service without any changes to existing
SIP and XMPP service infrastrucure. <br>
<br>
The objective of this Working Group is to develop solutions for combining
SIP based voice with XMPP based IM and Presence such that a unified user
experience can be offered to end user. Specifically, solutions are needed
on <br>
- addressing; end users should be able to initiate sessions to a user identity
in either SIP or XMPP domain. The corresponding user identity in the other
protocol domain needs to be learned automatically.<br>
- session correlation; endpoints need to be able to correlate voice sessions
with IM/Presence such that they can be presented to the end user in a seamless
fashion<br>
- presence; it should be possible to change the XMPP presence status based
on the user's activity in the SIP domain.<br>
<br>
The goal is to rely on existing service infrastructre, and new requirements
should be imposed only to the endpoint. <br>
<br>
Protocol interworking, that is, translation from one protocol to another,
is out of scope of this WG.<br>
<br>
Milestones<br>
Feb 2010 Problem statement and use cases submitted to IESG as Informational<br>
Dec 2010 Specification on combining SIP based voice and XMPP based IM/Presence
in a seamless manner submitted to IESG as Proposed Standard.<br>
<br>
-----------------<br>
_______________________________________________<br>
dispatch mailing list<br>
dispatch@ietf.org</font></tt><font size=3 color=blue><u><br>
</u></font><a href=https://www.ietf.org/mailman/listinfo/dispatch><tt><font size=2 color=blue><u>https://www.ietf.org/mailman/listinfo/dispatch</u></font></tt></a><font size=3><br>
</font>
<br>
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Simo,

=20

If I add a B2BUA to the example that you have in section 7.2:

=20

Bob                                                B2BUA
Alice

|                                                      |
|

| (F1) INVITE                                    | (F2) INVITE
|

|--------------------------------------------------
>|---------------------------------------------------> |

| XMPP-Contact: Bob; call-id1           | XMPP-Contact: Bob; call-id1
|

|                                                     |
|

|                                                     |
|

| (F4) 200 OK                                  | (F3) 200 OK
|

|<--------------------------------------------------
|<-------------------------------------------------- |

| XMPP-Contact: Alice; call-id2         | XMPP-Contact: Alice; call-id2
|

|                                                     |
|

=20

=20

Let (F1) INVITE have the correlation-id as the Call-ID of the dialog
between Bob and the B2BUA (call-id1),=20

and (F3) 200 OK have the correlation-id as the Call-ID of the dialog
between the B2BUA and Alice (call-id2).

=20

Would it not be enough for Bob to send call-id2 to Alice as the
<sip-correlation>? and for Alice to send call-id1 to Bob as the
<sip-correlation>?

=20

Regards,

 Rifaat


________________________________

From: Markus.Isomaki@nokia.com [mailto:Markus.Isomaki@nokia.com]=20
Sent: Thursday, September 24, 2009 2:54 AM
To: Shekh-Yusef, Rifaat (BVW:9T16); Simo.Veikkolainen@nokia.com;
dispatch@ietf.org; Barnes, Mary (RICH2:AR00);
gonzalo.camarillo@ericsson.com
Subject: RE: [dispatch] New topic proposal for DISPATCH - SIP/XMPP


Hi Rifaat,
=20
In our proposal the correlation works so that:
1. UAs engaged in a SIP session/dialog exchange an identifier related to
that session (as well as their XMPP full JIDs)
2. They insert them in the very first XMPP messages they subsequently
exchange, in order to correlate that exchange to be related to the SIP
session.
=20
The correlation identifier has to be end-to-end, otherwise the other end
won't be able to use it for correlation. Call-ID is often changed
enroute. So, if one endpoint puts it in an XMPP messages as a reference
to a SIP session, the other endpoint won't recognize it as it has a
different Call-ID for the same session.
=20
Markus


________________________________

	From: ext Rifaat Shekh-Yusef [mailto:rifatyu@nortel.com]=20
	Sent: 23 September, 2009 22:57
	To: Veikkolainen Simo (Nokia-D/Helsinki); dispatch@ietf.org;
Mary Barnes; gonzalo.camarillo@ericsson.com
	Cc: Isomaki Markus (Nokia-CIC/Espoo)
	Subject: RE: [dispatch] New topic proposal for DISPATCH
=09
=09
	Simo, Markus,
	=20
	I am very interested in this work and I like your approach.
	=20
	I have a question regarding the open issue you have in section
5.2:
	Why do you require the correlation-value to be an end-to-end
identifier?
	In the case where there is an SBC that creates a new dialog for
the target, would it not be enough for each endpoint to use the Call-ID
of the other dialog as the correlation-value?
	=20
	Thanks,
	 Rifaat
	=20

________________________________

	From: dispatch-bounces@ietf.org
[mailto:dispatch-bounces@ietf.org] On Behalf Of
Simo.Veikkolainen@nokia.com
	Sent: Friday, September 04, 2009 8:09 AM
	To: dispatch@ietf.org; Barnes, Mary (RICH2:AR00);
gonzalo.camarillo@ericsson.com
	Cc: Markus.Isomaki@nokia.com
	Subject: [dispatch] New topic proposal for DISPATCH
=09
=09
=09
	Hi,
	=20
	We would like to propose a new DISPATCH topic on how SIP and
XMPP can be used together in a complementary fashion.
	=20
	We have been working on a solution which combines SIP based VoIP
and XMPP based IM and Presence. The requirements and a proposed solution
is outlined in
http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01
<http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01> .
	=20
	The motivation for this work comes from experience which shows
that most standards based VoIP deployments use SIP. At the same time,
the Extensible Messaging and Presence Protocol (XMPP) is widely used in
instant messaging and presence services. An interesting scenario arises
when a SIP based voice (and video) service is combined together with an
XMPP based instant messaging and presence service.=20
	=20
	Note that the goal in this work is not SIP-XMPP protocol
translation, but to define protocol extensions and best practises such
that SIP based VoIP and XMPP based IM and presence could be seamlessly
combined and offered to the end user. For rapid deployment, we assume no
changes in the server infrastructure, and instead impose new
requirements and protocol extensions only to the endpoints.
	=20
	We would like to get some discussion going on the use case
itself as well as on the solution. Also, we would like to hear you
thoughts on DISPATCH or a new "dispatched" mini-WG as the home for such
work.
	=20
	Exact charter proposal and problem statemen is to follow.=20
	=20
	Regards,
	Simo and Markus
	=20
	=20


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<P style=3D"MARGIN: 0in 0in 0pt" class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: =
10pt">Simo,<?xml:namespace=20
prefix =3D o ns =3D "urn:schemas-microsoft-com:office:office"=20
/><o:p></o:p></SPAN></P>
<P style=3D"MARGIN: 0in 0in 0pt" class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: =
10pt"><o:p>&nbsp;</o:p></SPAN></P>
<P style=3D"MARGIN: 0in 0in 0pt" class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">If =
I&nbsp;<SPAN=20
class=3D945024615-24092009>add a B2BUA to the&nbsp;</SPAN>example that =
you have in=20
section 7.2:<o:p></o:p></SPAN></P>
<P style=3D"MARGIN: 0in 0in 0pt" class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: =
10pt"><o:p>&nbsp;</o:p></SPAN></P>
<P style=3D"MARGIN: 0in 0in 0pt" class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">Bob<SPAN=20
style=3D"mso-tab-count: =
4">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</SPAN><SPAN=20
style=3D"mso-tab-count: =
1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</SPAN>B2BUA<SPAN=20
style=3D"mso-tab-count: =
3">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
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style=3D"mso-tab-count: =
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style=3D"mso-tab-count: =
1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</SPAN><?xml:namespace=20
prefix =3D st1 ns =3D "urn:schemas-microsoft-com:office:smarttags" =
/><st1:City=20
w:st=3D"on"><st1:place=20
w:st=3D"on">Alice</st1:place></st1:City><o:p></o:p></SPAN></P>
<P style=3D"MARGIN: 0in 0in 0pt" class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">|<SPAN=20
style=3D"mso-tab-count: =
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class=3D945024615-24092009>&nbsp;&nbsp;</SPAN></SPAN>|<o:p></o:p></SPAN><=
/P>
<P style=3D"MARGIN: 0in 0in 0pt" class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">| (F1) =
INVITE<SPAN=20
style=3D"mso-tab-count: =
4">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<SPAN=20
class=3D945024615-24092009>&nbsp; &nbsp;</SPAN></SPAN>| (F2) INVITE<SPAN =

style=3D"mso-tab-count: =
4">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
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class=3D945024615-24092009>&nbsp;&nbsp;&nbsp;</SPAN></SPAN>|<o:p></o:p></=
SPAN></P>
<P style=3D"MARGIN: 0in 0in 0pt" class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: =
10pt">|--------------------------------------------------=20
&gt;|---------------------------------------------------&gt;<SPAN=20
style=3D"mso-tab-count: 1"><SPAN class=3D945024615-24092009>=20
</SPAN></SPAN>|<o:p></o:p></SPAN></P>
<P style=3D"MARGIN: 0in 0in 0pt" class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">| =
XMPP-Contact: Bob;=20
call-id1<SPAN=20
style=3D"mso-tab-count: 1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</SPAN><SPAN=20
style=3D"mso-tab-count: 1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<SPAN=20
class=3D945024615-24092009> </SPAN></SPAN>| XMPP-Contact: Bob; =
call-id1<SPAN=20
style=3D"mso-tab-count: 1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</SPAN><SPAN=20
style=3D"mso-tab-count: 1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
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style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">|<SPAN=20
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<P style=3D"MARGIN: 0in 0in 0pt" class=3DMsoNormal><SPAN=20
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<P style=3D"MARGIN: 0in 0in 0pt" class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">| (F4) 200 =
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p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
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p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
</SPAN>|<o:p></o:p></SPAN></P>
<P style=3D"MARGIN: 0in 0in 0pt" class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: =
10pt">|&lt;--------------------------------------------------<SPAN=20
style=3D"mso-tab-count: 1">=20
</SPAN>|&lt;--------------------------------------------------<SPAN=20
style=3D"mso-tab-count: 1"> </SPAN>|<o:p></o:p></SPAN></P>
<P style=3D"MARGIN: 0in 0in 0pt" class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">| =
XMPP-Contact:=20
<st1:City w:st=3D"on">Alice</st1:City>; call-id2<SPAN=20
style=3D"mso-tab-count: 1">&nbsp;&nbsp;&nbsp;&nbsp;</SPAN><SPAN=20
style=3D"mso-tab-count: 1">&nbsp;&nbsp;&nbsp;&nbsp; </SPAN>| =
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<st1:City w:st=3D"on"><st1:place =
w:st=3D"on">Alice</st1:place></st1:City>;=20
call-id2<SPAN style=3D"mso-tab-count: =
1">&nbsp;&nbsp;&nbsp;&nbsp;</SPAN><SPAN=20
style=3D"mso-tab-count: 1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<SPAN=20
class=3D945024615-24092009> </SPAN></SPAN>|<o:p></o:p></SPAN></P>
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style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">|<SPAN=20
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5">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
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nbsp;&nbsp;&nbsp;&nbsp;=20
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5">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
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nbsp;&nbsp;&nbsp;&nbsp;=20
</SPAN>|<o:p></o:p></SPAN></P>
<P style=3D"MARGIN: 0in 0in 0pt" class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: =
10pt"><o:p>&nbsp;</o:p></SPAN></P>
<P style=3D"MARGIN: 0in 0in 0pt" class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: =
10pt"><o:p>&nbsp;</o:p></SPAN></P>
<P style=3D"MARGIN: 0in 0in 0pt" class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">Let (F1) =
INVITE have=20
the correlation-<SPAN class=3D945024615-24092009>id</SPAN> as the =
Call-ID of the=20
dialog between Bob and the B2BUA (call-id1), </SPAN></P>
<P style=3D"MARGIN: 0in 0in 0pt" class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">and (F3) 200 =
OK have=20
the correlation-<SPAN class=3D945024615-24092009>id </SPAN>as the =
Call-ID of the=20
dialog between the B2BUA and Alice (call-id2).<o:p></o:p></SPAN></P>
<P style=3D"MARGIN: 0in 0in 0pt" class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: =
10pt"><o:p>&nbsp;</o:p></SPAN></P>
<P style=3D"MARGIN: 0in 0in 0pt" class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">Would it not =
be enough=20
for Bob to send call-id2 to <st1:City w:st=3D"on"><st1:place=20
w:st=3D"on">Alice</st1:place></st1:City> as the&nbsp;<SPAN=20
class=3D945024615-24092009>&lt;sip-</SPAN>correlation<SPAN=20
class=3D945024615-24092009>&gt;</SPAN>? and for <st1:place =
w:st=3D"on"><st1:City=20
w:st=3D"on">Alice</st1:City></st1:place> to send call-id1 to Bob as =
the&nbsp;<SPAN=20
class=3D945024615-24092009>&lt;</SPAN><SPAN=20
class=3D945024615-24092009>sip-</SPAN>correlation<SPAN=20
class=3D945024615-24092009>&gt;</SPAN>?<o:p></o:p></SPAN></P>
<P style=3D"MARGIN: 0in 0in 0pt" class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: =
10pt"><o:p>&nbsp;</o:p></SPAN></P>
<P style=3D"MARGIN: 0in 0in 0pt" class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: =
10pt">Regards,<o:p></o:p></SPAN></P>
<P style=3D"MARGIN: 0in 0in 0pt" class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt"><SPAN=20
style=3D"mso-spacerun: =
yes">&nbsp;</SPAN>Rifaat<o:p></o:p></SPAN></P></DIV><BR>
<DIV dir=3Dltr lang=3Den-us class=3DOutlookMessageHeader align=3Dleft>
<HR tabIndex=3D-1>
<FONT size=3D2 face=3DTahoma><B>From:</B> Markus.Isomaki@nokia.com=20
[mailto:Markus.Isomaki@nokia.com] <BR><B>Sent:</B> Thursday, September =
24, 2009=20
2:54 AM<BR><B>To:</B> Shekh-Yusef, Rifaat (BVW:9T16);=20
Simo.Veikkolainen@nokia.com; dispatch@ietf.org; Barnes, Mary =
(RICH2:AR00);=20
gonzalo.camarillo@ericsson.com<BR><B>Subject:</B> RE: [dispatch] New =
topic=20
proposal for DISPATCH - SIP/XMPP<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 =
face=3DArial><SPAN=20
class=3D705224406-24092009>Hi Rifaat,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 =
face=3DArial><SPAN=20
class=3D705224406-24092009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 =
face=3DArial><SPAN=20
class=3D705224406-24092009>In our proposal the correlation works so=20
that:</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 =
face=3DArial><SPAN=20
class=3D705224406-24092009>1. UAs engaged in a SIP session/dialog =
exchange an=20
identifier related to that session (as well as their XMPP full=20
JIDs)</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 =
face=3DArial><SPAN=20
class=3D705224406-24092009>2. They insert them in the very first XMPP =
messages=20
they subsequently exchange, in order to correlate that exchange to be =
related to=20
the SIP session.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 =
face=3DArial><SPAN=20
class=3D705224406-24092009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 =
face=3DArial><SPAN=20
class=3D705224406-24092009>The correlation identifier has to be =
end-to-end,=20
otherwise the other end won't be able to use it for correlation. Call-ID =
is=20
often changed enroute. So, if one endpoint puts it in an XMPP messages =
as a=20
reference to a SIP session, the other endpoint won't recognize it as it =
has a=20
different Call-ID for the same session.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 =
face=3DArial><SPAN=20
class=3D705224406-24092009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 =
face=3DArial><SPAN=20
class=3D705224406-24092009>Markus</SPAN></FONT></DIV><BR>
<BLOCKQUOTE=20
style=3D"BORDER-LEFT: #0000ff 2px solid; PADDING-LEFT: 5px; MARGIN-LEFT: =
5px; MARGIN-RIGHT: 0px"=20
dir=3Dltr>
  <DIV dir=3Dltr lang=3Den-us class=3DOutlookMessageHeader align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT size=3D2 face=3DTahoma><B>From:</B> ext Rifaat Shekh-Yusef=20
  [mailto:rifatyu@nortel.com] <BR><B>Sent:</B> 23 September, 2009=20
  22:57<BR><B>To:</B> Veikkolainen Simo (Nokia-D/Helsinki); =
dispatch@ietf.org;=20
  Mary Barnes; gonzalo.camarillo@ericsson.com<BR><B>Cc:</B> Isomaki =
Markus=20
  (Nokia-CIC/Espoo)<BR><B>Subject:</B> RE: [dispatch] New topic proposal =
for=20
  DISPATCH<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV><SPAN class=3D443113913-23092009><FONT color=3D#0000ff size=3D2=20
  face=3DArial>Simo, Markus,</FONT></SPAN></DIV>
  <DIV><SPAN class=3D443113913-23092009><FONT color=3D#0000ff size=3D2=20
  face=3DArial></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D443113913-23092009><FONT color=3D#0000ff size=3D2 =
face=3DArial>I am=20
  very interested in this work and I like your =
approach.</FONT></SPAN></DIV>
  <DIV><SPAN class=3D443113913-23092009><FONT color=3D#0000ff size=3D2=20
  face=3DArial></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D443113913-23092009><FONT color=3D#0000ff size=3D2 =
face=3DArial>I=20
  have a question regarding the open issue you have in section=20
  5.2:</FONT></SPAN></DIV>
  <DIV><SPAN class=3D443113913-23092009><FONT color=3D#0000ff size=3D2 =
face=3DArial>Why=20
  do you require the correlation-value to be an end-to-end=20
  identifier?</FONT></SPAN></DIV>
  <DIV><SPAN class=3D443113913-23092009><FONT color=3D#0000ff size=3D2 =
face=3DArial>In=20
  the case&nbsp;where there is an SBC that creates a new dialog for the =
target,=20
  w</FONT></SPAN><SPAN class=3D443113913-23092009><FONT color=3D#0000ff =
size=3D2=20
  face=3DArial>ould it not be enough for each endpoint to use the =
Call-ID of the=20
  other dialog as the correlation-value?</FONT></SPAN></DIV>
  <DIV><SPAN class=3D443113913-23092009><FONT color=3D#0000ff size=3D2=20
  face=3DArial></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D443113913-23092009><FONT color=3D#0000ff size=3D2=20
  face=3DArial>Thanks,</FONT></SPAN></DIV>
  <DIV><SPAN class=3D443113913-23092009><FONT color=3D#0000ff size=3D2=20
  face=3DArial>&nbsp;Rifaat</FONT></SPAN></DIV>
  <DIV><SPAN class=3D443113913-23092009><FONT color=3D#0000ff size=3D2=20
  face=3DArial></FONT></SPAN>&nbsp;</DIV><BR>
  <DIV dir=3Dltr lang=3Den-us class=3DOutlookMessageHeader align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT size=3D2 face=3DTahoma><B>From:</B> dispatch-bounces@ietf.org=20
  [mailto:dispatch-bounces@ietf.org] <B>On Behalf Of=20
  </B>Simo.Veikkolainen@nokia.com<BR><B>Sent:</B> Friday, September 04, =
2009=20
  8:09 AM<BR><B>To:</B> dispatch@ietf.org; Barnes, Mary (RICH2:AR00);=20
  gonzalo.camarillo@ericsson.com<BR><B>Cc:</B>=20
  Markus.Isomaki@nokia.com<BR><B>Subject:</B> [dispatch] New topic =
proposal for=20
  DISPATCH<BR></FONT><BR></DIV>
  <DIV></DIV><FONT size=3D2 face=3D"Arial, sans-serif">
  <DIV>Hi,</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>We would like to propose a new DISPATCH topic on how SIP and XMPP =
can be=20
  used together in a complementary fashion.</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>We have been working on a solution which combines SIP based VoIP =
and XMPP=20
  based IM and Presence. The requirements and a proposed solution is =
outlined in=20
  <A=20
  =
href=3D"http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01=
"><FONT=20
  =
color=3D#0000ff><U>http://tools.ietf.org/html/draft-veikkolainen-sip-voip=
-xmpp-im-01</U></FONT></A>.</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>The motivation for this work comes from experience which shows =
that most=20
  standards based VoIP deployments use SIP. At the same time, the =
Extensible=20
  Messaging and Presence Protocol (XMPP) is widely used in instant =
messaging and=20
  presence services. An interesting scenario arises when a SIP based =
voice (and=20
  video) service is combined together with an XMPP based instant =
messaging and=20
  presence service. </DIV>
  <DIV>&nbsp;</DIV>
  <DIV>Note that the goal in this work is not SIP-XMPP protocol =
translation, but=20
  to define protocol extensions and best practises such that SIP based =
VoIP and=20
  XMPP based IM and presence could be seamlessly combined and offered to =
the end=20
  user. For rapid deployment, we assume no changes in the server =
infrastructure,=20
  and instead impose new requirements and protocol extensions only to =
the=20
  endpoints.</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>We would like to get some discussion going on the use case itself =
as well=20
  as on the solution. Also, we would like to hear you thoughts on =
DISPATCH or a=20
  new "dispatched" mini-WG as the home for such work.</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>Exact charter proposal and problem statemen is to follow. </DIV>
  <DIV>&nbsp;</DIV>
  <DIV>Regards,</DIV>
  <DIV>Simo and Markus</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;</DIV></BLOCKQUOTE></FONT></BODY></HTML>

------_=_NextPart_001_01CA3D2F.9FEB0EF0--

From gunnar.hellstrom@omnitor.se  Thu Sep 24 11:07:00 2009
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To: "'Emil Ivov'" <emcho@sip-communicator.org>
References: <B23B311878A0B4438F5F09F47E01AAE03B18566B21@NOK-EUMSG-01.mgdnok.nokia.com><4AB8BC67.3080702@sip-communicator.org> <4AB8CFEC.8020703@cisco.com><B23B311878A0B4438F5F09F47E01AAE03B1886AC5E@NOK-EUMSG-01.mgdnok.nokia.com><4ABA18A0.5070603@sip-communicator.org> <B23B311878A0B4438F5F09F47E01AAE03B1886B4A7@NOK-EUMSG-01.mgdnok.nokia.com> <038101ca3cfc$738b13c0$e800a8c0@GunnarH> <4ABB44CD.6000708@sip-communicator.org>
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Cc: Simo.Veikkolainen@nokia.com, dispatch@ietf.org
Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM and	presence
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Emil,
The link with other real-time media in the SIP part of the call makes it
valid to also discuss the real-time characteristics of the text part.

It may look strange to arrange real-time communication in video and =
audio,
but leave the text part in message-wise performance level. Especially =
now,
when real-time text is appearing as a performance enhancement in various
services as you have noticed yourself.

However, you are right in principle, real-time is a general presentation
function that could be defined in XMPP.=20
So, it could be of interest to define this presentation feature in XMPP, =
and
resolve any interoperability and media offer/answer issues with RFC 4103 =
in
the new group.

For example, how would you express in SDP that you offer RFC 4013 in RTP =
and
real-timish mode of XMPP as equal alternatives to select from, in =
contrast
to offering RFC 4103 as the main real-time text medium in the session, =
and
message-based XMPP for side-bar discussion. Such aspects would need to =
be
defined under the new charter.



Gunnar
-------------------------------------------------------------------
Gunnar Hellstr=F6m
Omnitor
gunnar.hellstrom@omnitor.se
Tel: +46708204288
www.omnitor.se

-----Original Message-----
From: Emil Ivov [mailto:emil@sip-communicator.org] On Behalf Of Emil =
Ivov
Sent: Thursday, September 24, 2009 12:07 PM
To: Gunnar Hellstrom
Cc: Simo.Veikkolainen@nokia.com; dispatch@ietf.org
Subject: Re: [dispatch] Charter proposal on combining SIP voice with =
XMPP IM
and presence

Hey Gunnar,

I am not sure I understand how this fits the XMPP+SIP work.
Isn't it something that would be better suited by a pure XMPP entity =
such as
the XMPP WG or xmpp.org? Or do you think that there's some aspect in =
this
that can only be handled with SIP and another with XMPP?

IIRC, that's one of the features that appear in Google Wave's demos so =
maybe
there's already some XMPP related work in that direction.

Cheers,
Emil

Gunnar Hellstrom wrote:
> There is another aspect of text communication that I would like to=20
> propose to bring into this charter.
>=20
> It is to provide a more "real-timish" text conversational mode, where=20
> the text is sent time-sampled in short chunks during its composition=20
> rather than waiting on user command to send a completed message. The=20
> presentation should then present the chunks consecutively until a
delimiter is received.
>=20
> We have defined the real-time text mode for transmission with RTP, in=20
> RFC 4103, and want of course continue supporting that transport. With=20
> RTP it is realistic to get real real-time transmission, with=20
> transmission in 300 ms intervals. With XMPP, the overead makes it=20
> perhaps more realistic to transmit only once per second, but still get =

> a quite useable real-timish result, that would be appreciated by the
users.
>=20
> So, I suggest to bring real-timish text transmission of XMPP into this =

> charter.
>=20
> Gunnar
>=20
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On=20
> Behalf Of Simo.Veikkolainen@nokia.com
> Sent: Thursday, September 24, 2009 8:34 AM
> To: emcho@sip-communicator.org
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] Charter proposal on combining SIP voice with=20
> XMPP IM and presence
>=20
> =20
>=20
>> -----Original Message-----
>> From: Emil Ivov [mailto:emil@sip-communicator.org] On Behalf Of ext=20
>> Emil Ivov
>> Sent: 23 September, 2009 15:46
>> To: Veikkolainen Simo (Nokia-D/Helsinki)
>> Cc: pkyzivat@cisco.com; dispatch@ietf.org
>> Subject: Re: [dispatch] Charter proposal on combining SIP voice with=20
>> XMPP IM and presence
>>
>> Hey Simo,
>>
>> Simo.Veikkolainen@nokia.com wrote:
>>> As has been pointed out in earlier emails, there is no automatic=20
>>> correlation of SIP and XMPP user id's and thus an AOR=20
>>> alice@example.com may not belong to the same person as the JID=20
>>> alice@example.com. This rules out the possibility of responding to a =

>>> SIP MESSAGE with an XMPP <message> stanza without some a priori=20
>>> knowledge on Alice's identity in XMPP domain.
>> I understand why we would not want to rely on an implicit correlation =

>> like for example JID =3D=3D AOR. However an explicit correlation such =
as=20
>> an XMPP contact header in SIP messages and a SIP URI in the XMPP=20
>> vcards would be very helpful. A XMPP+SIP user receiving a call from=20
>> someone not on their contact list is likely to wish to add the remote =

>> party to their roster or send them instant messages. Not being able=20
>> to do so would feel awkward to the user.
>=20
> Yes. Explicit correlation (carrying SIP identities in XMPP or vice=20
> versa) is what we proposed in our draft, and we think could indeed be
useful.
>=20
>>> From an end user's point of view it makes no difference
>> which protocol
>>> is used. Personally, I would not mind having all my messaging (IM's=20
>>> via different service providers, SMS, email etc.) visible in
>> a single
>>> application view, and let the application hide some of the
>> complexity.
>>
>> Not sure I understand. Does this mean that you expect messages to be=20
>> traveling over both SIMPLE and XMPP? If yes, then we definitely need=20
>> some priority rules.
>=20
> What I meant was that from a UI point of view, all the end user needs=20
> to see is a contact name and possibility send a message (or call). A=20
> lot of the complexity of selecting the actual protocol *could* be=20
> hidden by the application. This may not be what all application=20
> developers wish, but for example the phonebooks in some mobile devices =

> seem to be going to that direction, combining IM and SMS in one
"conversation" view.
>=20
> So, for example, if a user wants the send an initial message to a=20
> contact whose both SIP and XMPP IDs are known, the app could do the=20
> selection of protocol, depending on the other user's presence=20
> information. However, when responding to a message in a conversation,=20
> it probably is better to use the same protocol that was used when=20
> sending, unless the client knows that the other client's UI can render =
the
messages in the same conversation view.
>=20
> So, I'll I'm saying is we need to carefully consider making normative=20
> statements that could potentially limit the application design and=20
> usability.
>=20
> Simo
>=20
>> Cheers,
>> Emil
>>
>>
>>> Simo
>>>
>>>>> So, if you think the above are worth considering how about adding=20
>>>>> something along these lines:
>>>>>
>>>>> disambiguation: both XMPP and SIP define semantics that
>>>> allow each of
>>>>> the protocols to offer the complete set of services
>>>> discussed in this
>>>>> charter (i.e. voice, video, messaging, and presence). In
>>>> environments
>>>>> where one or more of these services are supported with both
>>>> protocols
>>>>> clients should be able to use a common procedure for
>>>> determining which
>>>>> one to use and under what conditions.
>>>>>
>>>>> half-failures: in cases where use of one of the protocols becomes=20
>>>>> impossible (e.g. due to a server failure) clients should
>> be able  to
>>>>> follow clear procedures on how to behave, which services
>> could they
>>>>> try to reuse over the protocol that is still available and
>>>> under what
>>>>> conditions.
>>>>>
>>>>> WDYT?
>>>>>
>>>>> Emil
>>>>>
>>>>>
>>>>> Simo.Veikkolainen@nokia.com wrote:
>>>>>> Hello,
>>>>>>
>>>>>> Below is the draft charter text for our proposal on
>>>> combining SIP voice with XMPP IM and presence.
>>>>>> Comments are very welcome!
>>>>>>
>>>>>> Regards, Simo
>>>>>>
>>>>>>
>>>>>> -----------------
>>>>>>
>>>>>>
>>>>>> Combining SIP VoIP and XMPP IM/Presence=20
>>>>>> =
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>>>>>>
>>>>>> Currently most standards-based Voice over IP (VoIP)
>>>> deployments use the Session Initiation Protocol (SIP).  In
>> addition
>>>> to providing basic voice service SIP has an extensive support for=20
>>>> more advanced telephony features including interworking with the=20
>>>> traditional Public Switched Telephone Network (PSTN).  SIP is also=20
>>>> gaining popularity in the field of video communication. At
>> the same
>>>> time, the Extensible Messaging and Presence Protocol (XMPP) is=20
>>>> enjoying widespread usage in instant messaging and presence
>> services.
>>>>>> Both SIP and XMPP offer extensions for voice as well as IM
>>>> and presence (XMPP voice via the Jingle extension, and SIP=20
>>>> IM/presence via SIMPLE protocols). However, widespread
>> deployment of
>>>> these extensions has not so far taken place. In order to speed up=20
>>>> deployment of integrated VoIP and IM systems, SIP based voice and=20
>>>> XMPP based IM/Presence could be combined in endpoints to offer a=20
>>>> voice, IM and presence service without any changes to existing SIP=20
>>>> and XMPP service infrastrucure.
>>>>>> The objective of this Working Group is to develop solutions for=20
>>>>>> combining SIP based voice with XMPP based IM and Presence
>>>> such that a
>>>>>> unified user experience can be offered to end user.=20
>>>>>> Specifically, solutions are needed on - addressing; end users=20
>>>>>> should be able to initiate sessions
>>>> to a user identity in either SIP or XMPP domain. The corresponding=20
>>>> user identity in the other protocol domain needs to be learned=20
>>>> automatically.
>>>>>> - session correlation; endpoints need to be able to
>> correlate voice
>>>>>> sessions with IM/Presence such that they can be presented
>>>>>>
>>>>>>
>>>> to the end
>>>>>> user in a seamless fashion - presence; it should be possible to=20
>>>>>> change the XMPP
>>>> presence status based on the user's activity in the SIP domain.
>>>>>> The goal is to rely on existing service infrastructre, and
>>>> new requirements should be imposed only to the endpoint.
>>>>>> Protocol interworking, that is, translation from one
>>>> protocol to another, is out of scope of this WG.
>>>>>> Milestones Feb 2010 Problem statement and use cases submitted to=20
>>>>>> IESG as Informational Dec 2010 Specification on combining
>> SIP based
>>>> voice and XMPP based IM/Presence in a seamless manner submitted to=20
>>>> IESG as Proposed Standard.
>>>>>> -----------------
>>>>>> _______________________________________________ dispatch mailing=20
>>>>>> list dispatch@ietf.org=20
>>>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>>>
>> --=20
>> Emil Ivov, Ph.D.                               67000 Strasbourg,
>> Project Lead                                   France
>> SIP Communicator
>> emcho@sip-communicator.org                     PHONE: =
+33.1.77.62.43.30
>> http://sip-communicator.org                    FAX:   =
+33.1.77.62.47.31
>>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20
> __________ NOD32 4452 (20090924) Information __________
>=20
> Detta meddelande dr genomsvkt av NOD32 Antivirus.
> http://www.nod32.com
>=20
>=20
>=20

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

__________ NOD32 4452 (20090924) Information __________

Detta meddelande =E4r genoms=F6kt av NOD32 Antivirus.
http://www.nod32.com



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I support this work, and as we have seen in the past, with multiple =
proposals,
the need is real.

The charter generaly looks OK to me.=20

I am personally much more interested in the "loosely coupled" aspect to =
this,
as opposed to the MBUS/MEGACO/3PCC aspects, but I don't object to the=20
group covering both.

-Francois

> -----Original Message-----
From: Salvatore Loreto <salvatore.loreto at ericsson.com>
To: IETF Dispatch <dispatch at ietf.org>
Cc: Gonzalo Camarillo <gonzalo.camarillo at ericsson.com>
Date: Mon, 21 Sep 2009 13:51:51 +0300
Hi there,

below is a charter proposal for a WG on Disaggregated Media.

All comments, thoughts, feedbacks are very welcome!

cheers
Sal



Disaggregated Media WG Charter
-------------------------------=20
Disaggregated media refers to the ability for a user to create a
multimedia session combining different media streams, coming from
different devices under his or her control, so that they are treated
by the far end of the session as a single media session.

Generally, a given participant uses a single device to establish (or
participate in) a given multimedia session.  Consequently, the SIP
signaling to manage the multimedia session and the actual media
streams are typically co-located in the same device.  In scenarios
involving disaggregated media, a user wants to establish a single
multimedia session combining different media streams coming from
different devices under his or her control.  This creates a need to
coordinate the exchange of the those media streams within the media
session.

The Message Bus (Mbus) [RFC3259] can be used by an user to coordinate
the different devices under his control to involve them in the call.
The different devices can communicate with one another using Mbus =
messages, and then let only one device, a call control engine, to =
initiate, manage and terminate call control relations to other SIP =
endpoints. All the different user's devices need to support the Mbus =
protocol.

The Megaco [RFC3525] protocol can be used in a disaggregated media =
scenario to let one of the user's devices act as a Media Gateway =
Controller,
coordinating all the other devices under the user's control, which in =
this case
will act as Media Gateways. In this case all the different user's =
devices need to support the Megaco protocol.

The SIP protocol provides 3pcc [RFC3725] as a possible mechanism
to coordinate the exchange of media streams coming from different =
devices under the control of the same user, in the case at least one =
among the different devices under his control supports this mechanism =
and is able to become a Controller for the other in the call.

All the existing mechanisms only cover a subset of the interesting =
scenarios that involve disaggregated media. Scenarios not covered by =
existing mechanisms include the loosely-coupled one where none of the =
nodes can act as a controller
because it does not support the necessary functionality or because it =
will not
participate in the whole session.


The objective of the proposed working group is to develop a framework
for Disaggregated Media that include both improvements on existing =
mechanisms (e.g. 3pcc) and the develop of one or more new mechanisms =
like a loosely coupled mechanism that does not require the =
implementation of complex logic on any of the terminals.


Specifically, the proposed working group will develop the following
deliverables: 1. A framework document describing key design =
considerations for Disaggregated mediamechanism.
2. A specification for a loosely couple mechanism.
3. One or more specifications to improve existing mechanism to =
coordinate
  disaggregated media.

From ggb@tid.es  Fri Sep 25 03:11:56 2009
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Hi,

I've been reading the requirements draft, and I agree that it would be 
needed some work in the IETF in order to be able to provide this 
functionality making use of SIP events support.

Somehow this feature is closer to searching than to filtering, and I 
tend to agree that RFC4660 does not fit well in this space (although the 
concept of triggers in RFC4660 could be reused).  Defining other XML 
schema for expressing the query/conditions could be the simplest solution.

BR,

Gustavo García
Telefónica I+D

On 09/14/2009 10:58 AM, Christer Holmberg wrote:
> Hi,
>
> Some time ago I submitted a requirement draft for the OMA work on CBUS.
>
> In short, the idea of CBUS is that a client provides a set of conditions (which can be related to presence, location etc etc) to a CBUS server. Based on the conditions, the CBUS server communicates with presence servers, locations servers etc in order to figure out which users fulfill the conditions. The CBUS server then provides the client with a list of the URIs which fulfilled the conditions. The CBUS can do that on a one-time bases, a time-based basis, or whenever the information changes (called evaluation parameters). In addition, the client can provide a set or URIs, or a reference to URIs, which the CBUS chooses from (called candidate URIs).
>
> The idea is to provide the URIs in a subscription event package, and the associated SUBSCRIBE would contain the conditions (most likely using an XML document).
>
> Dean raised a question whether it would be enough to define the new event package, and re-use RFC4660/1 to provide the conditions.
>
> Due to summer vacations etc it took a while to look into this, but me and some OMA people have now looked into this.
>
> Our initial conclusion is that it is NOT very feasible to re-use RFC 4660/1.
>
> RFC4660/1 provides a filtering mechanism, where the client specifies what type of information he wants to receive. However, in CBUS the type of information is always the same: the URI list. Instead, the CLIENT provides the type of information (conditions) on which the URIs shall be selected (and, in addition to that, the evaluation parameter(s) and candidate URI(s)).
>
> We did look at whether it would be possible to somehow re-use 4660/1, but we thought it would be rather hacky. Some of the 4660/1 XML elements would probably be unused, and there would be a need to define new extensions - so we thought it would be nicer to define a new XML schema instead.
>
> So, I would like to request DISPATCH time to discuss this. We plan to provide an updated version of the draft, with clarifications and more details etc.
>
> Regards,
>
> Christer
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>
>    


From christer.holmberg@ericsson.com  Fri Sep 25 03:16:38 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
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Hi Gustavo,

Thanks you for your feeback!

We did look into using RFC4660, and we came to the same conclusion as =
you. Yes, perhaps it would be possible to use SOME part of 4660, but =
many things would not be needed, and there would be a need to define new =
things anyway. We thought that could become rather hacky and messy.

Regards,

Christer

=20

> -----Original Message-----
> From: ggb [mailto:ggb@tid.es]=20
> Sent: 25. syyskuuta 2009 13:13
> To: Christer Holmberg
> Cc: dispatch@ietf.org; Gonzalo Camarillo
> Subject: Re: [dispatch] IETF-76 DISPATCH WG deadlines : CBUS
>=20
> Hi,
>=20
> I've been reading the requirements draft, and I agree that it=20
> would be needed some work in the IETF in order to be able to=20
> provide this functionality making use of SIP events support.
>=20
> Somehow this feature is closer to searching than to=20
> filtering, and I tend to agree that RFC4660 does not fit well=20
> in this space (although the concept of triggers in RFC4660=20
> could be reused).  Defining other XML schema for expressing=20
> the query/conditions could be the simplest solution.
>=20
> BR,
>=20
> Gustavo Garc=EDa
> Telef=F3nica I+D
>=20
> On 09/14/2009 10:58 AM, Christer Holmberg wrote:
> > Hi,
> >
> > Some time ago I submitted a requirement draft for the OMA=20
> work on CBUS.
> >
> > In short, the idea of CBUS is that a client provides a set=20
> of conditions (which can be related to presence, location etc=20
> etc) to a CBUS server. Based on the conditions, the CBUS=20
> server communicates with presence servers, locations servers=20
> etc in order to figure out which users fulfill the=20
> conditions. The CBUS server then provides the client with a=20
> list of the URIs which fulfilled the conditions. The CBUS can=20
> do that on a one-time bases, a time-based basis, or whenever=20
> the information changes (called evaluation parameters). In=20
> addition, the client can provide a set or URIs, or a=20
> reference to URIs, which the CBUS chooses from (called=20
> candidate URIs).
> >
> > The idea is to provide the URIs in a subscription event=20
> package, and the associated SUBSCRIBE would contain the=20
> conditions (most likely using an XML document).
> >
> > Dean raised a question whether it would be enough to define=20
> the new event package, and re-use RFC4660/1 to provide the conditions.
> >
> > Due to summer vacations etc it took a while to look into=20
> this, but me and some OMA people have now looked into this.
> >
> > Our initial conclusion is that it is NOT very feasible to=20
> re-use RFC 4660/1.
> >
> > RFC4660/1 provides a filtering mechanism, where the client=20
> specifies what type of information he wants to receive.=20
> However, in CBUS the type of information is always the same:=20
> the URI list. Instead, the CLIENT provides the type of=20
> information (conditions) on which the URIs shall be selected=20
> (and, in addition to that, the evaluation parameter(s) and=20
> candidate URI(s)).
> >
> > We did look at whether it would be possible to somehow=20
> re-use 4660/1, but we thought it would be rather hacky. Some=20
> of the 4660/1 XML elements would probably be unused, and=20
> there would be a need to define new extensions - so we=20
> thought it would be nicer to define a new XML schema instead.
> >
> > So, I would like to request DISPATCH time to discuss this.=20
> We plan to provide an updated version of the draft, with=20
> clarifications and more details etc.
> >
> > Regards,
> >
> > Christer
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> >
> >   =20
>=20
>=20
>=20

From salvatore.loreto@ericsson.com  Fri Sep 25 03:32:13 2009
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From: "Salvatore Loreto" <salvatore.loreto@ericsson.com>
To: "Francois Audet" <audet@nortel.com>
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Hi Francois,

thanks for the support to the work.

the idea for the eventual wg is to focus on both the "loosely coupled" =
aspects and the improvements for a centralized approach
using 3PCC or other protocols.

/Sal


-----Original Message-----
From: Francois Audet [mailto:audet@nortel.com]
Sent: Fri 9/25/2009 12:26 AM
To: Salvatore Loreto
Cc: IETF Dispatch; Gonzalo Camarillo
Subject: RE: [dispatch] Charter Proposal for Disaggregated Media
=20
I support this work, and as we have seen in the past, with multiple =
proposals,
the need is real.

The charter generaly looks OK to me.=20

I am personally much more interested in the "loosely coupled" aspect to =
this,
as opposed to the MBUS/MEGACO/3PCC aspects, but I don't object to the=20
group covering both.

-Francois

> -----Original Message-----
From: Salvatore Loreto <salvatore.loreto at ericsson.com>
To: IETF Dispatch <dispatch at ietf.org>
Cc: Gonzalo Camarillo <gonzalo.camarillo at ericsson.com>
Date: Mon, 21 Sep 2009 13:51:51 +0300
Hi there,

below is a charter proposal for a WG on Disaggregated Media.

All comments, thoughts, feedbacks are very welcome!

cheers
Sal



Disaggregated Media WG Charter
-------------------------------=20
Disaggregated media refers to the ability for a user to create a
multimedia session combining different media streams, coming from
different devices under his or her control, so that they are treated
by the far end of the session as a single media session.

Generally, a given participant uses a single device to establish (or
participate in) a given multimedia session.  Consequently, the SIP
signaling to manage the multimedia session and the actual media
streams are typically co-located in the same device.  In scenarios
involving disaggregated media, a user wants to establish a single
multimedia session combining different media streams coming from
different devices under his or her control.  This creates a need to
coordinate the exchange of the those media streams within the media
session.

The Message Bus (Mbus) [RFC3259] can be used by an user to coordinate
the different devices under his control to involve them in the call.
The different devices can communicate with one another using Mbus =
messages, and then let only one device, a call control engine, to =
initiate, manage and terminate call control relations to other SIP =
endpoints. All the different user's devices need to support the Mbus =
protocol.

The Megaco [RFC3525] protocol can be used in a disaggregated media =
scenario to let one of the user's devices act as a Media Gateway =
Controller,
coordinating all the other devices under the user's control, which in =
this case
will act as Media Gateways. In this case all the different user's =
devices need to support the Megaco protocol.

The SIP protocol provides 3pcc [RFC3725] as a possible mechanism
to coordinate the exchange of media streams coming from different =
devices under the control of the same user, in the case at least one =
among the different devices under his control supports this mechanism =
and is able to become a Controller for the other in the call.

All the existing mechanisms only cover a subset of the interesting =
scenarios that involve disaggregated media. Scenarios not covered by =
existing mechanisms include the loosely-coupled one where none of the =
nodes can act as a controller
because it does not support the necessary functionality or because it =
will not
participate in the whole session.


The objective of the proposed working group is to develop a framework
for Disaggregated Media that include both improvements on existing =
mechanisms (e.g. 3pcc) and the develop of one or more new mechanisms =
like a loosely coupled mechanism that does not require the =
implementation of complex logic on any of the terminals.


Specifically, the proposed working group will develop the following
deliverables: 1. A framework document describing key design =
considerations for Disaggregated mediamechanism.
2. A specification for a loosely couple mechanism.
3. One or more specifications to improve existing mechanism to =
coordinate
  disaggregated media.


From lorenzo@meetecho.com  Fri Sep 25 03:49:22 2009
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Date: Fri, 25 Sep 2009 12:49:57 +0200
From: Lorenzo Miniero <lorenzo@meetecho.com>
To: Salvatore Loreto <salvatore.loreto@ericsson.com>
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Cc: IETF Dispatch <dispatch@ietf.org>, Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>
Subject: Re: [dispatch] Charter Proposal for Disaggregated Media
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Hi Salvatore,

I definitely support this proposal, and can see many use cases for it.
Be it just guidelines on the existing solutions, or new mechanisms to
support the required scenarios, we need something to address
disaggregated media, and my feeling is that the charter already looks
quite fine. Talking about it in Hiroshima will probably help refining
it.

L.


On Mon, 21 Sep 2009 13:51:51 +0300
Salvatore Loreto <salvatore.loreto@ericsson.com> wrote:

> Hi there,
> 
> below is a charter proposal for a WG on Disaggregated Media.
> 
> All comments, thoughts, feedbacks are very welcome!
> 
> cheers
> Sal
> 
> 
> 
> Disaggregated Media WG Charter
> ------------------------------- 
> Disaggregated media refers to the ability for a user to create a
> multimedia session combining different media streams, coming from
> different devices under his or her control, so that they are treated
> by the far end of the session as a single media session.
> 
> Generally, a given participant uses a single device to establish (or
> participate in) a given multimedia session.  Consequently, the SIP
> signaling to manage the multimedia session and the actual media
> streams are typically co-located in the same device.  In scenarios
> involving disaggregated media, a user wants to establish a single
> multimedia session combining different media streams coming from
> different devices under his or her control.  This creates a need to
> coordinate the exchange of the those media streams within the media
> session.
> 
> The Message Bus (Mbus) [RFC3259] can be used by an user to coordinate
> the different devices under his control to involve them in the call.
> The different devices can communicate with one another using Mbus 
> messages, and then let only one device, a call control engine, to 
> initiate, manage and terminate call control relations to other SIP endpoints. 
> All the different user's devices need to support the Mbus protocol.
> 
> The Megaco [RFC3525] protocol can be used in a disaggregated media 
> scenario to let one of the user's devices act as a Media Gateway Controller,
> coordinating all the other devices under the user's control, which in this case
> will act as Media Gateways. In this case all the different user's devices 
> need to support the Megaco protocol.
> 
> The SIP protocol provides 3pcc [RFC3725] as a possible mechanism
> to coordinate the exchange of media streams coming from different 
> devices under the control of the same user, in the case at least 
> one among the different devices under his control supports this
> mechanism and is able to become a Controller for the other in the call. 
> 
> 
> All the existing mechanisms only cover a subset of the interesting scenarios 
> that involve disaggregated media. Scenarios not covered by existing mechanisms 
> include the loosely-coupled one where none of the nodes can act as a controller
> because it does not support the necessary functionality or because it will not
> participate in the whole session.
> 
> 
> The objective of the proposed working group is to develop a framework
> for Disaggregated Media that include both improvements on existing 
> mechanisms (e.g. 3pcc) and the develop of one or more new mechanisms
> like a loosely coupled mechanism that does not require the implementation 
> of complex logic on any of the terminals.
> 
> 
> Specifically, the proposed working group will develop the following
> deliverables: 
> 1. A framework document describing key design considerations for Disaggregated 
>    mediamechanism.
> 2. A specification for a loosely couple mechanism.
> 3. One or more specifications to improve existing mechanism to coordinate
>    disaggregated media.
> 
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 


-- 
Lorenzo Miniero <lorenzo@meetecho.com>

From victor.pascual.avila@gmail.com  Mon Sep 28 09:39:58 2009
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Subject: Re: [dispatch] draft-elwell-dispatch-identity-reqs-00 posted
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Dan,

thanks for your support. We encourage folks to take a look at it and
provide comments over the coming weeks so that we can address them
before the next IETF meeting.

Cheers,
-Victor

On Thu, Sep 3, 2009 at 5:19 PM, Dan York <dyork@voxeo.com> wrote:
> Victor and John,
>
> In getting caught up on all that's been going on in DISPATCH over the pas=
t
> bit, I will only say that *yes*, I think this is both helpful and going i=
n
> the right direction. =C2=A0Please do continue your work on it. =C2=A0I do=
 not
> currently have further feedback beyond saying that.
>
> Thanks for doing this,
> Dan
> On Jul 21, 2009, at 12:17 PM, Victor Pascual Avila wrote:
>
>> As mentioned in the below message, the authors would appreciate
>> feedback as to whether this is helpful and going in the right
>> direction.
>>
>> Thanks,
>> -Victor
>>
>> 2009/6/29 Elwell, John <john.elwell@siemens-enterprise.com>:
>>>
>>> There was an extensive discussion at IETF#74 on SIP Identity, where a
>>> good many views were put forward. The only consensus seemed to be to
>>> continue to discuss the topic.
>>>
>>> One of the themes during that discussion was to focus more on
>>> requirements, which the authors have attempted to do in this new draft:
>>>
>>> http://www.ietf.org/internet-drafts/draft-elwell-dispatch-identity-reqs=
-00.txt
>>>
>>> We would appreciate feedback as to whether this is helpful and going in
>>> the right direction, as well as detailed comments.
>>>
>>> I had hoped to do this a lot earlier, to trigger a discussion in time t=
o
>>> get something set up for DISPATCH at IETF#75, but I failed, and also I
>>> failed to talk to the many of you who have interest in the topic and
>>> expressed opinions in the past. But since the deadline is approaching, =
I
>>> thought it best to post the draft and let discussion continue on that b=
asis.
>>>
>>> John
>>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>
> --
> Dan York, Director of Conversations
> Voxeo Corporation =C2=A0 http://www.voxeo.com =C2=A0dyork@voxeo.com
> Phone: +1-407-455-5859 =C2=A0 =C2=A0Skype: danyork
>
> Join the Voxeo conversation:
> Blogs: http://blogs.voxeo.com
> Twitter: http://twitter.com/voxeo =C2=A0http://twitter.com/danyork
> Facebook: http://www.facebook.com/voxeo
>
>
>
>
>
>
>
>
>



--=20
Victor Pascual =C3=81vila

From dworley@nortel.com  Mon Sep 28 13:21:44 2009
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Subject: Re: [dispatch] Charter Proposal for Disaggregated Media
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The general approach of the charter looks good to me, but there seem
to be too many words used to describe the existing methods (which may
or may not be solutions to the problem), and a lack of sharpness in
describing the area we are focusing on.  Let me propose some
alternative text.  I think that this does not change the intention of
any of the text.



Disaggregated Media WG Charter
------------------------------- 
Disaggregated media refers to the ability for a user to create a
multimedia session combining different media streams, coming from
different devices under his or her control, so that they are treated
by the far end of the session as a single media session.

    OK

Generally, a given participant uses a single device to establish (or
participate in) a given multimedia session.  Consequently, the SIP
signaling to manage the multimedia session and the actual media
streams are typically co-located in the same device.  In scenarios
involving disaggregated media, a user wants to establish a single
multimedia session combining different media streams coming from
different devices under his or her control.  This creates a need to
coordinate the exchange of the those media streams within the media
session.

    OK

The Message Bus (Mbus) [RFC3259] can be used by an user to coordinate
the different devices under his control to involve them in the call.
The different devices can communicate with one another using Mbus 
messages, and then let only one device, a call control engine, to 
initiate, manage and terminate call control relations to other SIP endpoints. 
All the different user's devices need to support the Mbus protocol.

The Megaco [RFC3525] protocol can be used in a disaggregated media 
scenario to let one of the user's devices act as a Media Gateway Controller,
coordinating all the other devices under the user's control, which in this case
will act as Media Gateways. In this case all the different user's devices 
need to support the Megaco protocol.

The SIP protocol provides 3pcc [RFC3725] as a possible mechanism
to coordinate the exchange of media streams coming from different 
devices under the control of the same user, in the case at least 
one among the different devices under his control supports this
mechanism and is able to become a Controller for the other in the call. 

All the existing mechanisms only cover a subset of the interesting scenarios 
that involve disaggregated media. Scenarios not covered by existing mechanisms 
include the loosely-coupled one where none of the nodes can act as a controller
because it does not support the necessary functionality or because it will not
participate in the whole session.

    There are a number of protocols now used that are used to
    coordinate a number of devices (e.g., Mbus, Megaco, SIP 3pcc), but
    the common methods of using those protocols all require a
    "master" device that must remain involved in the user's session
    for its entire duration.

The objective of the proposed working group is to develop a framework
for Disaggregated Media that include both improvements on existing 
mechanisms (e.g. 3pcc) and the develop of one or more new mechanisms
like a loosely coupled mechanism that does not require the implementation 
of complex logic on any of the terminals.

    The objective of the proposed working group is to develop a
    framework for Disaggregated Media in "loosely-coupled" scenarios,
    where no single device can remain in the session for its entire
    duration and/or no single device has the necessary functionality
    to coordinate all of the media streams.

    The framework may include improvements on existing mechanisms
    (e.g. 3pcc), the development of one or more new mechanisms, and/or
    new methods of utilizing mechanisms.

Specifically, the proposed working group will develop the following
deliverables: 
1. A framework document describing key design considerations for Disaggregated 
   mediamechanism.
2. A specification for a loosely couple mechanism.
3. One or more specifications to improve existing mechanism to coordinate
   disaggregated media.

    Specifically, the proposed working group will develop the following
    deliverables: 
    1. Use cases and functional requirements for the mechanism(s)
    2. A framework document describing key design considerations for
       disaggregated media mechanism(s)
    3. One or more specifications for new mechanism(s), to improve
       existing mechanism(s), and/or improve their utilization(s) to
       coordinate disaggregated media


Dale



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Subject: Re: [dispatch] IETF-76 DISPATCH WG deadlines : New Topic	Domain	Registration
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The whole "problem" seems strange to me.  There is the question of
getting the SIP domain properly mapped to the IP address, but we have
dynamic DNS for that, and our prototypical PBX could coordinate that
with the service provider.  As for "it is impractical and
cost-prohibitive to manually provision their IP Addresses in every SIP
node along the path which needs to reach the SMB IP-PBX customer", of
course you don't manually provision that information:  The Domain Name
System automagically propagates the domain-to-IP-address mapping to the
entire Internet, and the various routing protocols automagically
propagate the IP routing information to "every node along the path".

On Fri, 2009-09-11 at 09:20 -0500, Alan Johnston wrote:
> The plan is to standardize a function that has been shown to be needed 
> in the marketplace for the deployment of SIP in certain applications.
> 
> And yes, dynamic DNS is an alternative, it does work, but again, there 
> are reasons why it is not being used in this certain application.

Might you be able to reveal those reasons?  The techniques I've
described are the ones used by the people I know.

Dale



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From: "Dale Worley" <dworley@nortel.com>
To: R.Jesske@telekom.de
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Organization: Nortel Networks
Date: Mon, 28 Sep 2009 16:50:15 -0400
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Subject: Re: [dispatch] New draft version of	draft-jesske-dispatch-reason-in-responses-00
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On Mon, 2009-09-14 at 13:10 +0200, R.Jesske@telekom.de wrote:
> A since a couple of weeks a new version of the Reason in Responses
> draft is now available.
> 
> http://tools.ietf.org/html/draft-jesske-dispatch-reason-in-responses-00
> 
> I have incorporated all comments and the discussion conclusions
> received for this draft.

There seems to be quite a bit of duplicated text in the Abstract.

Dale



From dean.willis@softarmor.com  Mon Sep 28 21:33:24 2009
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From: Dean Willis <dean.willis@softarmor.com>
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Subject: Re: [dispatch] IETF-76 DISPATCH WG deadlines : New Topic	Domain	Registration
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On Sep 28, 2009, at 3:37 PM, Dale Worley wrote:

> The whole "problem" seems strange to me.  There is the question of
> getting the SIP domain properly mapped to the IP address, but we have
> dynamic DNS for that, and our prototypical PBX could coordinate that
> with the service provider.  As for "it is impractical and
> cost-prohibitive to manually provision their IP Addresses in every SIP
> node along the path which needs to reach the SMB IP-PBX customer", of
> course you don't manually provision that information:  The Domain Name
> System automagically propagates the domain-to-IP-address mapping to  
> the
> entire Internet, and the various routing protocols automagically
> propagate the IP routing information to "every node along the path".

I struggled with this as well. But now that I'm thinking about the  
problem as trying to dynamically construct a SIP route between the  
proxy-of-record (as indicated by DNS) and a local proxy or multi-user  
UA (read "PBX"), I'm finding the problem significantly more tractable.

The reason DDNS doesn't work for this is that there might be private  
topography between the proxy-of-record and the multiuser UA (think  
"outbound". Now think massively-shared outbound connections).

--
Dean

From Simo.Veikkolainen@nokia.com  Wed Sep 30 02:07:57 2009
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From: <Simo.Veikkolainen@nokia.com>
To: <rifatyu@nortel.com>, <Markus.Isomaki@nokia.com>, <dispatch@ietf.org>, <mary.barnes@nortel.com>, <gonzalo.camarillo@ericsson.com>
Date: Wed, 30 Sep 2009 11:09:01 +0200
Thread-Topic: [dispatch] New topic proposal for DISPATCH - SIP/XMPP
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Hello Rifaat,

sorry for the late reply.

If I understood your scenario correctly, you are proposing to use separete =
correlation id's in each direction. That would work as well, but I don't se=
e the reason why Alice would choose call-id2 as her correlation id, if in a=
ny case call-id1 gets to her UA in F2.

Simo

________________________________
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of ext Rifaat Shekh-Yusef
Sent: 24 September, 2009 18:57
To: Isomaki Markus (Nokia-CIC/Espoo); Veikkolainen Simo (Nokia-D/Helsinki);=
 dispatch@ietf.org; Mary Barnes; gonzalo.camarillo@ericsson.com
Subject: Re: [dispatch] New topic proposal for DISPATCH - SIP/XMPP

Simo,

If I add a B2BUA to the example that you have in section 7.2:

Bob                                                B2BUA                   =
                       Alice
|                                                      |                   =
                                  |
| (F1) INVITE                                    | (F2) INVITE             =
                       |
|-------------------------------------------------- >|---------------------=
------------------------------> |
| XMPP-Contact: Bob; call-id1           | XMPP-Contact: Bob; call-id1      =
     |
|                                                     |                    =
                                  |
|                                                     |                    =
                                  |
| (F4) 200 OK                                  | (F3) 200 OK               =
                    |
|<-------------------------------------------------- |<--------------------=
------------------------------ |
| XMPP-Contact: Alice; call-id2         | XMPP-Contact: Alice; call-id2    =
      |
|                                                     |                    =
                                 |


Let (F1) INVITE have the correlation-id as the Call-ID of the dialog betwee=
n Bob and the B2BUA (call-id1),
and (F3) 200 OK have the correlation-id as the Call-ID of the dialog betwee=
n the B2BUA and Alice (call-id2).

Would it not be enough for Bob to send call-id2 to Alice as the <sip-correl=
ation>? and for Alice to send call-id1 to Bob as the <sip-correlation>?

Regards,
 Rifaat

________________________________
From: Markus.Isomaki@nokia.com [mailto:Markus.Isomaki@nokia.com]
Sent: Thursday, September 24, 2009 2:54 AM
To: Shekh-Yusef, Rifaat (BVW:9T16); Simo.Veikkolainen@nokia.com; dispatch@i=
etf.org; Barnes, Mary (RICH2:AR00); gonzalo.camarillo@ericsson.com
Subject: RE: [dispatch] New topic proposal for DISPATCH - SIP/XMPP

Hi Rifaat,

In our proposal the correlation works so that:
1. UAs engaged in a SIP session/dialog exchange an identifier related to th=
at session (as well as their XMPP full JIDs)
2. They insert them in the very first XMPP messages they subsequently excha=
nge, in order to correlate that exchange to be related to the SIP session.

The correlation identifier has to be end-to-end, otherwise the other end wo=
n't be able to use it for correlation. Call-ID is often changed enroute. So=
, if one endpoint puts it in an XMPP messages as a reference to a SIP sessi=
on, the other endpoint won't recognize it as it has a different Call-ID for=
 the same session.

Markus

________________________________
From: ext Rifaat Shekh-Yusef [mailto:rifatyu@nortel.com]
Sent: 23 September, 2009 22:57
To: Veikkolainen Simo (Nokia-D/Helsinki); dispatch@ietf.org; Mary Barnes; g=
onzalo.camarillo@ericsson.com
Cc: Isomaki Markus (Nokia-CIC/Espoo)
Subject: RE: [dispatch] New topic proposal for DISPATCH

Simo, Markus,

I am very interested in this work and I like your approach.

I have a question regarding the open issue you have in section 5.2:
Why do you require the correlation-value to be an end-to-end identifier?
In the case where there is an SBC that creates a new dialog for the target,=
 would it not be enough for each endpoint to use the Call-ID of the other d=
ialog as the correlation-value?

Thanks,
 Rifaat


________________________________
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Simo.Veikkolainen@nokia.com
Sent: Friday, September 04, 2009 8:09 AM
To: dispatch@ietf.org; Barnes, Mary (RICH2:AR00); gonzalo.camarillo@ericsso=
n.com
Cc: Markus.Isomaki@nokia.com
Subject: [dispatch] New topic proposal for DISPATCH

Hi,

We would like to propose a new DISPATCH topic on how SIP and XMPP can be us=
ed together in a complementary fashion.

We have been working on a solution which combines SIP based VoIP and XMPP b=
ased IM and Presence. The requirements and a proposed solution is outlined =
in http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-01.

The motivation for this work comes from experience which shows that most st=
andards based VoIP deployments use SIP. At the same time, the Extensible Me=
ssaging and Presence Protocol (XMPP) is widely used in instant messaging an=
d presence services. An interesting scenario arises when a SIP based voice =
(and video) service is combined together with an XMPP based instant messagi=
ng and presence service.

Note that the goal in this work is not SIP-XMPP protocol translation, but t=
o define protocol extensions and best practises such that SIP based VoIP an=
d XMPP based IM and presence could be seamlessly combined and offered to th=
e end user. For rapid deployment, we assume no changes in the server infras=
tructure, and instead impose new requirements and protocol extensions only =
to the endpoints.

We would like to get some discussion going on the use case itself as well a=
s on the solution. Also, we would like to hear you thoughts on DISPATCH or =
a new "dispatched" mini-WG as the home for such work.

Exact charter proposal and problem statemen is to follow.

Regards,
Simo and Markus



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<DIV dir=3Dltr align=3Dleft><SPAN class=3D992390609-30092009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Hello Rifaat,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D992390609-30092009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D992390609-30092009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>sorry for the late reply.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D992390609-30092009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D992390609-30092009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>If I understood your scenario correctly, you are p=
roposing=20
to use separete correlation id's in each direction. That would work as well=
, but=20
I don't see the reason why Alice would choose call-id2 as her correlation i=
d, if=20
in any case call-id1 gets to her UA in F2.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D992390609-30092009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D992390609-30092009><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Simo</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px soli=
d; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> dispatch-bounces@ietf.org=20
  [mailto:dispatch-bounces@ietf.org] <B>On Behalf Of </B>ext Rifaat=20
  Shekh-Yusef<BR><B>Sent:</B> 24 September, 2009 18:57<BR><B>To:</B> Isomak=
i=20
  Markus (Nokia-CIC/Espoo); Veikkolainen Simo (Nokia-D/Helsinki);=20
  dispatch@ietf.org; Mary Barnes;=20
  gonzalo.camarillo@ericsson.com<BR><B>Subject:</B> Re: [dispatch] New topi=
c=20
  proposal for DISPATCH - SIP/XMPP<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV>
  <P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">Simo,<o:p></o:=
p></SPAN></P>
  <P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial"><o:p>&nbsp;</o=
:p></SPAN></P>
  <P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">If I&nbsp;<SPA=
N=20
  class=3D945024615-24092009>add a B2BUA to the&nbsp;</SPAN>example that yo=
u have=20
  in section 7.2:<o:p></o:p></SPAN></P>
  <P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial"><o:p>&nbsp;</o=
:p></SPAN></P>
  <P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">Bob<SPAN=20
  style=3D"mso-tab-count: 4">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</SPAN><SPAN=20
  style=3D"mso-tab-count: 1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</SPAN>B2B=
UA<SPAN=20
  style=3D"mso-tab-count: 3">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</SPAN><SPAN=20
  style=3D"mso-tab-count: 1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;</SPAN><SPAN=20
  style=3D"mso-tab-count: 1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</SPAN><st1:City=
=20
  w:st=3D"on"><st1:place=20
  w:st=3D"on">Alice</st1:place></st1:City><o:p></o:p></SPAN></P>
  <P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">|<SPAN=20
  style=3D"mso-tab-count: 4">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;</SPAN><SPAN=20
  style=3D"mso-tab-count: 1">&nbsp;&nbsp;<SPAN=20
  class=3D945024615-24092009>&nbsp;&nbsp;&nbsp;&nbsp; </SPAN></SPAN>|<SPAN=
=20
  style=3D"mso-tab-count: 4">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</SPAN><SPAN=
=20
  style=3D"mso-tab-count: 1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;<SPAN=20
  class=3D945024615-24092009>&nbsp;&nbsp;</SPAN></SPAN>|<o:p></o:p></SPAN><=
/P>
  <P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">| (F1) INVITE<=
SPAN=20
  style=3D"mso-tab-count: 4">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;<SPAN=20
  class=3D945024615-24092009>&nbsp; &nbsp;</SPAN></SPAN>| (F2) INVITE<SPAN=
=20
  style=3D"mso-tab-count: 4">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;<SPAN=20
  class=3D945024615-24092009>&nbsp;&nbsp;&nbsp;</SPAN></SPAN>|<o:p></o:p></=
SPAN></P>
  <P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">|-------------=
-------------------------------------=20
  &gt;|---------------------------------------------------&gt;<SPAN=20
  style=3D"mso-tab-count: 1"><SPAN class=3D945024615-24092009>=20
  </SPAN></SPAN>|<o:p></o:p></SPAN></P>
  <P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">| XMPP-Contact=
: Bob;=20
  call-id1<SPAN=20
  style=3D"mso-tab-count: 1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</SPAN><SPAN=20
  style=3D"mso-tab-count: 1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<SPAN=20
  class=3D945024615-24092009> </SPAN></SPAN>| XMPP-Contact: Bob; call-id1<S=
PAN=20
  style=3D"mso-tab-count: 1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</SPAN><SPAN=20
  style=3D"mso-tab-count: 1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  </SPAN>|<o:p></o:p></SPAN></P>
  <P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">|<SPAN=20
  style=3D"mso-tab-count: 4">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</SPAN><SPAN=20
  style=3D"mso-tab-count: 1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;=20
  </SPAN>|<SPAN=20
  style=3D"mso-tab-count: 4">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;</SPAN><SPAN=20
  style=3D"mso-tab-count: 1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  </SPAN>|<o:p></o:p></SPAN></P>
  <P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">|<SPAN=20
  style=3D"mso-tab-count: 5">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  </SPAN>|<SPAN=20
  style=3D"mso-tab-count: 5">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  </SPAN>|<o:p></o:p></SPAN></P>
  <P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">| (F4) 200 OK<=
SPAN=20
  style=3D"mso-tab-count: 4">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;</SPAN>|=20
  (F3) 200 OK<SPAN=20
  style=3D"mso-tab-count: 4">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;=20
  </SPAN>|<o:p></o:p></SPAN></P>
  <P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">|&lt;---------=
-----------------------------------------<SPAN=20
  style=3D"mso-tab-count: 1">=20
  </SPAN>|&lt;--------------------------------------------------<SPAN=20
  style=3D"mso-tab-count: 1"> </SPAN>|<o:p></o:p></SPAN></P>
  <P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">| XMPP-Contact=
:=20
  <st1:City w:st=3D"on">Alice</st1:City>; call-id2<SPAN=20
  style=3D"mso-tab-count: 1">&nbsp;&nbsp;&nbsp;&nbsp;</SPAN><SPAN=20
  style=3D"mso-tab-count: 1">&nbsp;&nbsp;&nbsp;&nbsp; </SPAN>| XMPP-Contact=
:=20
  <st1:City w:st=3D"on"><st1:place w:st=3D"on">Alice</st1:place></st1:City>=
;=20
  call-id2<SPAN style=3D"mso-tab-count: 1">&nbsp;&nbsp;&nbsp;&nbsp;</SPAN><=
SPAN=20
  style=3D"mso-tab-count: 1">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<SPAN=20
  class=3D945024615-24092009> </SPAN></SPAN>|<o:p></o:p></SPAN></P>
  <P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">|<SPAN=20
  style=3D"mso-tab-count: 5">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  </SPAN>|<SPAN=20
  style=3D"mso-tab-count: 5">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  </SPAN>|<o:p></o:p></SPAN></P>
  <P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial"><o:p>&nbsp;</o=
:p></SPAN></P>
  <P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial"><o:p>&nbsp;</o=
:p></SPAN></P>
  <P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">Let (F1) INVIT=
E have=20
  the correlation-<SPAN class=3D945024615-24092009>id</SPAN> as the Call-ID=
 of the=20
  dialog between Bob and the B2BUA (call-id1), </SPAN></P>
  <P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">and (F3) 200 O=
K have=20
  the correlation-<SPAN class=3D945024615-24092009>id </SPAN>as the Call-ID=
 of the=20
  dialog between the B2BUA and Alice (call-id2).<o:p></o:p></SPAN></P>
  <P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial"><o:p>&nbsp;</o=
:p></SPAN></P>
  <P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">Would it not b=
e=20
  enough for Bob to send call-id2 to <st1:City w:st=3D"on"><st1:place=20
  w:st=3D"on">Alice</st1:place></st1:City> as the&nbsp;<SPAN=20
  class=3D945024615-24092009>&lt;sip-</SPAN>correlation<SPAN=20
  class=3D945024615-24092009>&gt;</SPAN>? and for <st1:place w:st=3D"on"><s=
t1:City=20
  w:st=3D"on">Alice</st1:City></st1:place> to send call-id1 to Bob as=20
  the&nbsp;<SPAN class=3D945024615-24092009>&lt;</SPAN><SPAN=20
  class=3D945024615-24092009>sip-</SPAN>correlation<SPAN=20
  class=3D945024615-24092009>&gt;</SPAN>?<o:p></o:p></SPAN></P>
  <P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial"><o:p>&nbsp;</o=
:p></SPAN></P>
  <P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial">Regards,<o:p><=
/o:p></SPAN></P>
  <P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: Arial"><SPAN=20
  style=3D"mso-spacerun: yes">&nbsp;</SPAN>Rifaat<o:p></o:p></SPAN></P></DI=
V><BR>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> Markus.Isomaki@nokia.com=20
  [mailto:Markus.Isomaki@nokia.com] <BR><B>Sent:</B> Thursday, September 24=
,=20
  2009 2:54 AM<BR><B>To:</B> Shekh-Yusef, Rifaat (BVW:9T16);=20
  Simo.Veikkolainen@nokia.com; dispatch@ietf.org; Barnes, Mary (RICH2:AR00)=
;=20
  gonzalo.camarillo@ericsson.com<BR><B>Subject:</B> RE: [dispatch] New topi=
c=20
  proposal for DISPATCH - SIP/XMPP<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><=
SPAN=20
  class=3D705224406-24092009>Hi Rifaat,</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><=
SPAN=20
  class=3D705224406-24092009></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><=
SPAN=20
  class=3D705224406-24092009>In our proposal the correlation works so=20
  that:</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><=
SPAN=20
  class=3D705224406-24092009>1. UAs engaged in a SIP session/dialog exchang=
e an=20
  identifier related to that session (as well as their XMPP full=20
  JIDs)</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><=
SPAN=20
  class=3D705224406-24092009>2. They insert them in the very first XMPP mes=
sages=20
  they subsequently exchange, in order to correlate that exchange to be rel=
ated=20
  to the SIP session.</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><=
SPAN=20
  class=3D705224406-24092009></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><=
SPAN=20
  class=3D705224406-24092009>The correlation identifier has to be end-to-en=
d,=20
  otherwise the other end won't be able to use it for correlation. Call-ID =
is=20
  often changed enroute. So, if one endpoint puts it in an XMPP messages as=
 a=20
  reference to a SIP session, the other endpoint won't recognize it as it h=
as a=20
  different Call-ID for the same session.</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><=
SPAN=20
  class=3D705224406-24092009></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><=
SPAN=20
  class=3D705224406-24092009>Markus</SPAN></FONT></DIV><BR>
  <BLOCKQUOTE dir=3Dltr=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px so=
lid; MARGIN-RIGHT: 0px">
    <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
    <HR tabIndex=3D-1>
    <FONT face=3DTahoma size=3D2><B>From:</B> ext Rifaat Shekh-Yusef=20
    [mailto:rifatyu@nortel.com] <BR><B>Sent:</B> 23 September, 2009=20
    22:57<BR><B>To:</B> Veikkolainen Simo (Nokia-D/Helsinki); dispatch@ietf=
.org;=20
    Mary Barnes; gonzalo.camarillo@ericsson.com<BR><B>Cc:</B> Isomaki Marku=
s=20
    (Nokia-CIC/Espoo)<BR><B>Subject:</B> RE: [dispatch] New topic proposal =
for=20
    DISPATCH<BR></FONT><BR></DIV>
    <DIV></DIV>
    <DIV><SPAN class=3D443113913-23092009><FONT face=3DArial color=3D#0000f=
f=20
    size=3D2>Simo, Markus,</FONT></SPAN></DIV>
    <DIV><SPAN class=3D443113913-23092009><FONT face=3DArial color=3D#0000f=
f=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D443113913-23092009><FONT face=3DArial color=3D#0000f=
f size=3D2>I=20
    am very interested in this work and I like your=20
approach.</FONT></SPAN></DIV>
    <DIV><SPAN class=3D443113913-23092009><FONT face=3DArial color=3D#0000f=
f=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D443113913-23092009><FONT face=3DArial color=3D#0000f=
f size=3D2>I=20
    have a question regarding the open issue you have in section=20
    5.2:</FONT></SPAN></DIV>
    <DIV><SPAN class=3D443113913-23092009><FONT face=3DArial color=3D#0000f=
f=20
    size=3D2>Why do you require the correlation-value to be an end-to-end=20
    identifier?</FONT></SPAN></DIV>
    <DIV><SPAN class=3D443113913-23092009><FONT face=3DArial color=3D#0000f=
f size=3D2>In=20
    the case&nbsp;where there is an SBC that creates a new dialog for the=20
    target, w</FONT></SPAN><SPAN class=3D443113913-23092009><FONT face=3DAr=
ial=20
    color=3D#0000ff size=3D2>ould it not be enough for each endpoint to use=
 the=20
    Call-ID of the other dialog as the correlation-value?</FONT></SPAN></DI=
V>
    <DIV><SPAN class=3D443113913-23092009><FONT face=3DArial color=3D#0000f=
f=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D443113913-23092009><FONT face=3DArial color=3D#0000f=
f=20
    size=3D2>Thanks,</FONT></SPAN></DIV>
    <DIV><SPAN class=3D443113913-23092009><FONT face=3DArial color=3D#0000f=
f=20
    size=3D2>&nbsp;Rifaat</FONT></SPAN></DIV>
    <DIV><SPAN class=3D443113913-23092009><FONT face=3DArial color=3D#0000f=
f=20
    size=3D2></FONT></SPAN>&nbsp;</DIV><BR>
    <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
    <HR tabIndex=3D-1>
    <FONT face=3DTahoma size=3D2><B>From:</B> dispatch-bounces@ietf.org=20
    [mailto:dispatch-bounces@ietf.org] <B>On Behalf Of=20
    </B>Simo.Veikkolainen@nokia.com<BR><B>Sent:</B> Friday, September 04, 2=
009=20
    8:09 AM<BR><B>To:</B> dispatch@ietf.org; Barnes, Mary (RICH2:AR00);=20
    gonzalo.camarillo@ericsson.com<BR><B>Cc:</B>=20
    Markus.Isomaki@nokia.com<BR><B>Subject:</B> [dispatch] New topic propos=
al=20
    for DISPATCH<BR></FONT><BR></DIV>
    <DIV></DIV><FONT face=3D"Arial, sans-serif" size=3D2>
    <DIV>Hi,</DIV>
    <DIV>&nbsp;</DIV>
    <DIV>We would like to propose a new DISPATCH topic on how SIP and XMPP =
can=20
    be used together in a complementary fashion.</DIV>
    <DIV>&nbsp;</DIV>
    <DIV>We have been working on a solution which combines SIP based VoIP a=
nd=20
    XMPP based IM and Presence. The requirements and a proposed solution is=
=20
    outlined in <A=20
    href=3D"http://tools.ietf.org/html/draft-veikkolainen-sip-voip-xmpp-im-=
01"><FONT=20
    color=3D#0000ff><U>http://tools.ietf.org/html/draft-veikkolainen-sip-vo=
ip-xmpp-im-01</U></FONT></A>.</DIV>
    <DIV>&nbsp;</DIV>
    <DIV>The motivation for this work comes from experience which shows tha=
t=20
    most standards based VoIP deployments use SIP. At the same time, the=20
    Extensible Messaging and Presence Protocol (XMPP) is widely used in ins=
tant=20
    messaging and presence services. An interesting scenario arises when a =
SIP=20
    based voice (and video) service is combined together with an XMPP based=
=20
    instant messaging and presence service. </DIV>
    <DIV>&nbsp;</DIV>
    <DIV>Note that the goal in this work is not SIP-XMPP protocol translati=
on,=20
    but to define protocol extensions and best practises such that SIP base=
d=20
    VoIP and XMPP based IM and presence could be seamlessly combined and of=
fered=20
    to the end user. For rapid deployment, we assume no changes in the serv=
er=20
    infrastructure, and instead impose new requirements and protocol extens=
ions=20
    only to the endpoints.</DIV>
    <DIV>&nbsp;</DIV>
    <DIV>We would like to get some discussion going on the use case itself =
as=20
    well as on the solution. Also, we would like to hear you thoughts on=20
    DISPATCH or a new "dispatched" mini-WG as the home for such work.</DIV>
    <DIV>&nbsp;</DIV>
    <DIV>Exact charter proposal and problem statemen is to follow. </DIV>
    <DIV>&nbsp;</DIV>
    <DIV>Regards,</DIV>
    <DIV>Simo and Markus</DIV>
    <DIV>&nbsp;</DIV>
    <DIV>&nbsp;</DIV></BLOCKQUOTE></BLOCKQUOTE></FONT></BODY></HTML>

--_000_B23B311878A0B4438F5F09F47E01AAE03B18AD248DNOKEUMSG01mgd_--

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Thread-Topic: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM and presence
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From: "Smyth, Colm (Colm)" <smythc@avaya.com>
To: <dispatch@ietf.org>
Cc: Simo.Veikkolainen@nokia.com
Subject: Re: [dispatch] Charter proposal on combining SIP voice with XMPP IM and presence
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Hi,
=20
I'm interested in contributing to this working group.
=20
The charter looks good, I'd like to tweak or clarify some aspects of the
description.
=20
I agree that it's useful to focus is on the user experience of a hybrid
SIP/XMPP client, and leave out aspects of protocol-level interworking
(for example to define how to achieve an end-to-end voice session
between an XMPP/Jingle client and a SIP voice endpoint).
=20
1. addressing
A hybrid SIP/XMPP client must have addresses in both protocols, but it's
not clear to me that it's mandatory that a hybrid SIP/XMPP client must
be able to automatically learn the corresponding address of other users
in the other protocol. It may be sufficient in simple clients for the
user to manually associate the other address with one of their contacts.
The other address could be discovered in a shared directory, or in a
private contacts store. It might also define ways for a user to
"publish" their address in the other protocol (e.g. sending an XMPP
message that contains their SIP address), that might allow the client to
discover and (automatically or at the user's discretion) register that
address.
=20
2. session correlation
This topic seems to be entirely around operations and elements in the
UI. It's not so much about sessions as about allowing some operations to
span the two protocols (e.g. create/join/exit a conference that spans
SIP voice and XMPP chat, or "escalating" a conference in one protocol to
include the other, or "downgrading" it to exclude the other).
=20
3. presence
It would make sense to be able to change session in either protocol
based on activity in the other, at the discretion of the client and the
user. I agree with the view that the WG should exclude details of
protocol-level aspects, but to be useful, I think the WG should address
the user experience but also provide concrete guidelines on implementing
cross-protocol presence spanning or synchronization, including mapping
parts of an XMPP presence stanza to/from a (SIP) PIDF document. This
group should address if the SIP and XMPP endpoints are to be considered
as separate points of presence, or if they are to be 100% synchronized.
Ideally we would cover both scenarios (e.g. discrete PIDF <tuple>
elements and possibly a second XMPP session to a different resource for
the same JID).

Peter St Andre referred to several drafts that do relate to SIP/XMPP
protocol inter-working:

http://tools.ietf.org/html/draft-saintandre-sip-xmpp-core
http://tools.ietf.org/html/draft-saintandre-sip-xmpp-presence
http://tools.ietf.org/html/draft-saintandre-sip-xmpp-im
http://tools.ietf.org/html/draft-saintandre-sip-xmpp-chat
http://tools.ietf.org/html/draft-saintandre-sip-xmpp-groupchat
http://tools.ietf.org/html/draft-saintandre-sip-xmpp-media

I would also like to see these drafts progress to RFCs, but I think that
is outside the charter of this group. Related to presence, I would like
to see this group agree on a richer mapping than the one in Table 6 of
draft-saintandre-sip-xmpp-presence, specifically to define a standard
way to map bidirectionally between the different <show> states and PIDF
elements.
=20
Colm.

From dworley@nortel.com  Wed Sep 30 14:00:50 2009
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Subject: Re: [dispatch] Charter Proposal for Disaggregated Media
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On Mon, 2009-09-28 at 16:22 -0400, Dale Worley wrote:
> The objective of the proposed working group is to develop a framework
> for Disaggregated Media that include both improvements on existing 
> mechanisms (e.g. 3pcc) and the develop of one or more new mechanisms
> like a loosely coupled mechanism that does not require the implementation 
> of complex logic on any of the terminals.
> 
>     The objective of the proposed working group is to develop a
>     framework for Disaggregated Media in "loosely-coupled" scenarios,
>     where no single device can remain in the session for its entire
>     duration and/or no single device has the necessary functionality
>     to coordinate all of the media streams.

We could add "and/or the session is discontinuous in time" to that list
-- that was a case discussed in the past, although we may have decided
to exclude that from consideration.

Dale



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From: "Dale Worley" <dworley@nortel.com>
To: "Szilagyi, Mike" <Mike.Szilagyi@inin.com>
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Subject: Re: [dispatch] [sipcore] Requests sent within early dialog -- CSeq mgmt
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On Wed, 2009-08-26 at 17:10 -0400, Szilagyi, Mike wrote:
> I=E2=80=99ve not been able to find definitive text regarding this issue s=
o I=E2=80=99m
> hoping someone can provide clarification.  If a request (INFO or
> UPDATE) is sent on a dialog in the early state, how are the CSeq
> numbers managed by the UAS.  Here=E2=80=99s an example to demonstrate my
> confusion:

The model is that a CANCEL is not an independent transaction from the
INVITE, in the way that the UPDATE is an independent transaction from
the INVITE.  Indeed, CANCEL isn't even routed the same way that the
UPDATE is.  A CANCEL transaction is sort of a "second half" of the
INVITE transaction that it is canceling.  That is why the CANCEL has the
same CSeq as the INVITE.

(Also, your example shows the UPDATE being sent by the UAC before it
receives a non-100 response.  That can't be done, since the UAC doesn't
know the to-tag to use; there isn't an early dialog established at that
point.)

Dale




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Subject: Re: [dispatch] draft-elwell-dispatch-identity-reqs-00 posted
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John,

I think this is going the right direction, but few comments.

Req 5 is too vague to be of any use. If this was replaced with  
something like "The solution must allow for media steering", that  
would make sense to me

On Req 4 - I don't think it is just the sink/source of the media. It  
seems that just about anything that is meant to be end to end you want  
bound to the identity. Imagine an page mode IM. In that case the  
"media" is the message and you want that bound to the caller id.


On Jul 21, 2009, at 9:17 , Victor Pascual Avila wrote:

>
> 2009/6/29 Elwell, John <john.elwell@siemens-enterprise.com>:
>> There was an extensive discussion at IETF#74 on SIP Identity, where  
>> a good many views were put forward. The only consensus seemed to be  
>> to continue to discuss the topic.
>>
>> One of the themes during that discussion was to focus more on  
>> requirements, which the authors have attempted to do in this new  
>> draft:
>> http://www.ietf.org/internet-drafts/draft-elwell-dispatch-identity-reqs-00.txt
>>
>> We would appreciate feedback as to whether this is helpful and  
>> going in the right direction, as well as detailed comments.
>>
>> I had hoped to do this a lot earlier, to trigger a discussion in  
>> time to get something set up for DISPATCH at IETF#75, but I failed,  
>> and also I failed to talk to the many of you who have interest in  
>> the topic and expressed opinions in the past. But since the  
>> deadline is approaching, I thought it best to post the draft and  
>> let discussion continue on that basis.
>>
>> John
>>
> _____________________


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From: Cullen Jennings <fluffy@cisco.com>
To: "Vijay K. Gurbani" <vkg@alcatel-lucent.com>
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Subject: Re: [dispatch] TCP and SCTP connection reuse draft
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The security of this draft seems, uh, bad. It's so bad that I am  
having a really hard time believing I actually am understanding it  
correctly. Help me make sure that I have this right.

Cisco and ACME are part of the AT&T trust domain and get sip trunking  
from AT&T. Cisco sends a via that has ACME on the alias to AT&T. There  
is nothing AT&T can do to validate this. Now Cisco has hijacked all of  
ACME's call. If the trust domain consisted of just one party, well I  
see how this might be secure, sort of, but it seems to fall apart as  
the trust domain expands.

Cullen <with my individual contributor hat on>


On Sep 9, 2009, at 9:09 , Vijay K. Gurbani wrote:

> Folks: Before the summer, we had released a new
> TCP/SCTP connection reuse draft in the (now defunct) SIP
> WG.  With the formation of dispatch, we were advised to
> submit it to dispatch for a decision on where to handle
> this work.
>
> Essentially, the draft documents the practice of reusing TCP
> (and SCTP) connections that is employed by some SIP
> implementations today.  TCP and SCTP transport reuse were
> originally a part of the draft-ietf-sip-connect-reuse draft,
> but were taken out before we progressed that draft to the IESG.
>
> Can we kindly ask the dispatch WG to review and comment on
> the the TCP and SCTP connection reuse draft?  All comments
> received from the first iteration of the draft in the SIP WG
> have been duly addressed in this version.
>
> It will also be helpful if the WG and chairs could provide
> some guidance on how to move this work forward (i.e.,
> AD-sponsored individual draft, take it to sipcore, etc.)
>
> The draft can be retrieved from:
>
> http://tools.ietf.org/html/draft-jain-dispatch-sip-transport-connection-reuse-00
>
> Thank you,
>
> - vijay
> -- 
> Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
> 1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
> Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
> Web:   http://ect.bell-labs.com/who/vkg/
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


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From: "DOLLY, MARTIN C, ATTLABS" <mdolly@att.com>
To: <fluffy@cisco.com>, <vkg@alcatel-lucent.com>
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Subject: Re: [dispatch] TCP and SCTP connection reuse draft
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